205 files changed, 95574 insertions, 57977 deletions

diff --git a/usr/src/Makefile.lint b/usr/src/Makefile.lint
index 7c3ce96683..c4adef0aea 100644
--- a/usr/src/Makefile.lint
+++ b/usr/src/Makefile.lint
@@ -35,6 +35,8 @@ COMMON_SUBDIRS = \
 	cmd/acctadm \
 	cmd/asa \
 	cmd/amt \
+	cmd/audio/audiotest \
+	cmd/audio/mixerctl \
 	cmd/audit \
 	cmd/auditconfig \
 	cmd/auditd \
diff --git a/usr/src/cmd/allocate/Makefile b/usr/src/cmd/allocate/Makefile
index cd572ed03e..5691423a21 100644
--- a/usr/src/cmd/allocate/Makefile
+++ b/usr/src/cmd/allocate/Makefile
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 
 ETCTSOLFILES =	devalloc_defaults
 
@@ -33,13 +31,14 @@ include		../Makefile.cmd
 ROOTSEC =	$(ROOTETC)/security
 ROOTSECDEV =	$(ROOTSEC)/dev
 ROOTSECLIB =	$(ROOTSEC)/lib
-ROOTDIRS =	$(ROOTSECDEV) $(ROOTSECLIB)
+ROOTSECAUD =	$(ROOTSEC)/audio
+ROOTDIRS =	$(ROOTSECDEV) $(ROOTSECLIB) $(ROOTSECAUD)
 
 RTLCKS =	audio fd0 sr0 st0 st1
 CLEANfd =	fd_clean
 CLEANsr =	sr_clean
 CLEANst =	st_clean
-CLEANaudio =	audio_clean_wrapper
+CLEANaudio =	audio_clean
 CLEANdisk =	disk_clean
 CLEAN_SCRIPTS =	$(CLEANfd) $(CLEANsr) $(CLEANst) $(CLEANaudio) $(CLEANdisk)
 WDW_SCRIPTS =	wdwmsg wdwwrapper
@@ -50,10 +49,9 @@ ALLSCRIPTS =	allscripts.sh
 PROGalloc =	allocate
 PROGmkdevalloc =	mkdevalloc
 PROGdminfo =	dminfo
-PROGaudio =	audio_clean
 PROGaddalloc =	add_allocatable
 PROG =		$(PROGalloc) $(PROGmkdevalloc) $(PROGdminfo) \
-	$(PROGaudio) $(PROGaddalloc)
+	$(PROGaddalloc)
 
 LINKPROGalloc =	deallocate list_devices
 LINKPROGmkdevalloc = mkdevmaps
@@ -72,7 +70,6 @@ ROOTWDWLINKS =	$(WDW_LINKS:%=$(ROOTSECLIB)/%.windowing)
 ROOTPROG =	$(PROGallocate:%=$(ROOTUSRSBIN)/%) \
 		$(PROGmkdevalloc:%=$(ROOTUSRSBIN)/%) \
 		$(PROGdminfo:%=$(ROOTUSRSBIN)/%) \
-		$(PROGaudio:%=$(ROOTSECLIB)/%) \
 		$(PROGaddaloc:%=$(ROOTUSRSBIN)/%)
 ROOTLINKalloc =	$(LINKPROGalloc:%=$(ROOTUSRSBIN)/%)
 ROOTLINKmkdevalloc = $(LINKPROGmkdevalloc:%=$(ROOTUSRSBIN)/%)
@@ -82,13 +79,11 @@ ROOTLINKS =	$(ROOTLINKalloc) $(ROOTLINKmkdevalloc) $(ROOTLINKaddalloc)
 PROGallocOBJS =		allocate.o allocate3.o
 PROGmkdevallocOBJS =	mkdevalloc.o
 PROGdminfoOBJS =	dminfo.o
-PROGaudioOBJS =		audio_clean.o
 PROGaddallocOBJS =	add_allocatable.o
 
 OBJS =		$(PROGallocOBJS) \
 		$(PROGmkdevallocOBJS) \
 		$(PROGdminfoOBJS) \
-		$(PROGaudioOBJS) \
 		$(PROGaddallocOBJS)
 
 SRCS =		$(OBJS:%.o=%.c)
@@ -109,7 +104,6 @@ lint :=			LDLIBS += -lbsm -lsec -lsecdb -ldevinfo -ltsol
 $(PROGalloc) :=		LDLIBS += -lbsm -lsec -lsecdb -ldevinfo -ltsol
 $(PROGmkdevalloc) :=	LDLIBS += -lbsm
 $(PROGdminfo) :=	LDLIBS += -lbsm
-$(PROGaudio) :=		LDLIBS += -lbsm
 $(PROGaddalloc) :=	LDLIBS += -lbsm -lsecdb -ltsol
 
 CLOBBERFILES +=	$(SCRIPTS)
@@ -143,10 +137,6 @@ $(PROGdminfo) :	$(PROGdminfoOBJS)
 		$(LINK.c) $(PROGdminfoOBJS) -o $@ $(LDLIBS)
 		$(POST_PROCESS)
 
-$(PROGaudio) :	$(PROGaudioOBJS)
-		$(LINK.c) $(PROGaudioOBJS) -o $@ $(LDLIBS)
-		$(POST_PROCESS)
-
 $(ROOTDIRS) :
 		$(INS.dir)
 
@@ -157,10 +147,6 @@ $(ROOTSECLIB)/%: %
 		$(RM) $@
 		$(INS.file)
 
-$(ROOTSECLIB)/audio_clean :	audio_clean
-				$(RM) $@
-				$(INS.file) $(@F)
-
 $(ROOTLINKalloc) :	$(PROGalloc:%=$(ROOTUSRSBIN)/%)
 			$(RM) $@
 			$(LN) $(PROGalloc:%=$(ROOTUSRSBIN)/%) $@
diff --git a/usr/src/cmd/allocate/audio_clean.c b/usr/src/cmd/allocate/audio_clean.c
deleted file mode 100644
index 24753b8569..0000000000
--- a/usr/src/cmd/allocate/audio_clean.c
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * audio_clean - Clear any residual data that may be residing in the in the
- *	audio device driver or chip.
- *
- *	Usage: audio_clean [-Iisf] device_name
- *
- *	Note: The same operation is performed for any of the "ifs" flags.
- *		The "I" flag is a silent form of "i".  Support for the flags
- *		is provided so that the framework is place if added
- *		functionality is required.
- *
- *	Note: The AUDIO_SETINFO ioctl is used instead of the low level
- *		AUDIOSETREG command which is used to perform low level
- *		operations on the device.  If a process had previously used
- *		AUDIOSETREG to monkey with the device, then the driver would
- *		have reset the chip when the process performed a close,
- *		so this can just clear the info structure.
- */
-
-#include <ctype.h>
-#include <errno.h>
-#include <fcntl.h>
-#include <libintl.h>
-#include <locale.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <strings.h>
-#include <stropts.h>
-#include <unistd.h>
-#include <bsm/devices.h>
-#include <sys/audioio.h>
-#include <sys/file.h>
-#include <sys/ioctl.h>
-#include <sys/param.h>
-#include <sys/stat.h>
-#include <sys/types.h>
-
-#ifndef	TEXT_DOMAIN
-#define	TEXT_DOMAIN	"SUNW_OST_OSCMD"
-#endif
-
-#ifdef	DEBUG
-static void print_info(audio_info_t *);
-static void print_prinfo(audio_prinfo_t *);
-#endif	/* DEBUG */
-
-static void
-usage(char *prog)
-{
-	(void) fprintf(stderr, "%s%s", prog,
-	    gettext(" : usage:[-I|-s|-f|-i] device\n"));
-}
-
-int
-main(int argc, char **argv)
-{
-	int		err = 0;
-	int		Audio_fd;
-	int		forced = 0;		/* Command line options */
-	int		initial = 0;
-	int		standard = 0;
-	int		verbose = 1;		/* default is to be verbose */
-	int		c;
-	char		*prog, *devname, *devpath;
-	devmap_t	*dm;
-	struct stat	st;
-	audio_info_t	info;
-
-	(void) setlocale(LC_ALL, "");
-	(void) textdomain(TEXT_DOMAIN);
-
-	prog = argv[0];		/* save program initiation name */
-
-	/*
-	 * Parse arguments.  Currently i, s and f all do the
-	 * the same thing.
-	 */
-
-	while ((c = getopt(argc, argv, "Iifs")) != -1) {
-		switch (c) {
-		case 'I':
-			verbose = 0;
-			initial++;
-			if (standard || forced)
-				err++;
-			break;
-		case 'i':
-			initial++;
-			if (standard || forced)
-				err++;
-			break;
-		case 's':
-			standard++;
-			if (initial || forced)
-				err++;
-			break;
-		case 'f':
-			forced++;
-			if (initial || standard)
-				err++;
-			break;
-		case '?':
-			err++;
-			break;
-		default:
-			err++;
-			break;
-		}
-		if (err) {
-			if (verbose)
-				usage(prog);
-			exit(1);
-		}
-	}
-
-	if ((argc - optind) != 1) {
-		if (verbose)
-			usage(prog);
-		exit(1);
-	} else {
-		devname = argv[optind];
-	}
-
-	setdmapent();
-	if ((dm = getdmapnam(devname)) == NULL) {
-		enddmapent();
-		if (verbose)
-			(void) fprintf(stderr, "%s%s",
-			    devname,
-			    gettext(" : No such allocatable device\n"));
-			exit(1);
-	}
-	enddmapent();
-	if (dm->dmap_devarray == NULL || dm->dmap_devarray[0] == NULL) {
-		if (verbose)
-			(void) fprintf(stderr, "%s%s",
-			    devname,
-			    gettext(" : No such allocatable device\n"));
-			exit(1);
-	}
-	devpath = strdup(dm->dmap_devarray[0]);
-	freedmapent(dm);
-
-	/*
-	 * Validate and open the audio device
-	 */
-	err = stat(devpath, &st);
-
-	if (err < 0) {
-		if (verbose) {
-			(void) fprintf(stderr, gettext("%s: cannot stat "),
-			    prog);
-			perror("audio_clean");
-		}
-		exit(1);
-	}
-	if (!S_ISCHR(st.st_mode)) {
-		if (verbose)
-			(void) fprintf(stderr,
-			    gettext("%s: %s is not an audio device\n"), prog,
-			    devpath);
-		exit(1);
-	}
-
-	/*
-	 * Since the device /dev/audio can suspend if someone else is
-	 * using it we check to see if we're going to hang before we
-	 * do anything.
-	 */
-	Audio_fd = open(devpath, O_WRONLY | O_NDELAY);
-
-	if ((Audio_fd < 0) && (errno == EBUSY)) {
-		if (verbose)
-			(void) fprintf(stderr, gettext("%s: waiting for %s..."),
-			    prog, devpath);
-		exit(0);
-	} else if (Audio_fd < 0) {
-		if (verbose) {
-			(void) fprintf(stderr, gettext("%s: error opening "),
-			    prog);
-			perror(devpath);
-		}
-		exit(1);
-	}
-
-#ifdef	DEBUG
-	/*
-	 * Read the audio_info structure.
-	 */
-
-	if (ioctl(Audio_fd, AUDIO_GETINFO, &info) != 0)  {
-		perror("Ioctl AUDIO_GETINFO error");
-		(void) close(Audio_fd);
-		exit(1);
-	}
-
-	print_info(&info);
-#endif	/* DEBUG */
-
-/* LINTED */
-	AUDIO_INITINFO(&info);	/* clear audit info structure */
-
-	if (ioctl(Audio_fd, AUDIO_SETINFO, &info) != 0) {
-		if (verbose)
-			perror(gettext("Ioctl AUDIO_SETINFO error"));
-		(void) close(Audio_fd);
-		exit(1);
-	}
-
-#ifdef	DEBUG
-	if (ioctl(Audio_fd, AUDIO_GETINFO, &info) != 0)  {
-		perror("Ioctl AUDIO_GETINFO-2 error");
-		(void) close(Audio_fd);
-		exit(1);
-	}
-
-	print_info(&info);
-#endif	/* DEBUG */
-
-	return (0);
-}
-
-
-#ifdef	DEBUG
-void
-print_info(audio_info_t *info)
-{
-	print_prinfo(&info->play);
-	print_prinfo(&info->record);
-	(void) printf("monitor_gain %d\n", info->monitor_gain);
-	(void) fflush(stdout);
-}
-
-
-void
-print_prinfo(audio_prinfo_t *prinfo)
-{
-	/* The following values decribe audio data encoding: */
-	(void) printf("sample_rate %d\n",	prinfo->sample_rate);
-	(void) printf("channels %d\n", 	prinfo->channels);
-	(void) printf("precision %d\n", 	prinfo->precision);
-	(void) printf("encoding %d\n", 	prinfo->encoding);
-
-	/* The following values control audio device configuration */
-	(void) printf("gain %d\n", 	prinfo->gain);
-	(void) printf("port %d\n", 	prinfo->port);
-	(void) printf("avail_ports %d\n", 	prinfo->avail_ports);
-	(void) printf("mod_ports %d\n", 	prinfo->mod_ports);
-
-	/* These are Reserved for future use, but we clear them  */
-	(void) printf("_xxx %d\n", 	prinfo->_xxx);
-
-	(void) printf("buffer_size %d\n", 	prinfo->buffer_size);
-
-	/* The following values describe driver state */
-	(void) printf("samples %d\n", 	prinfo->samples);
-	(void) printf("eof %d\n", 	prinfo->eof);
-	(void) printf("pause %d\n", 	prinfo->pause);
-	(void) printf("error %d\n", 	prinfo->error);
-	(void) printf("waiting %d\n", 	prinfo->waiting);
-	(void) printf("balance %d\n", 	prinfo->balance);
-
-	/* The following values are read-only state flags */
-	(void) printf("open %d\n", 	prinfo->open);
-	(void) printf("active %d\n", 	prinfo->active);
-}
-#endif	/* DEBUG */
diff --git a/usr/src/cmd/allocate/audio_clean_wrapper.sh b/usr/src/cmd/allocate/audio_clean.sh
index 2a7820c098..24ccea7fdb 100644
--- a/usr/src/cmd/allocate/audio_clean_wrapper.sh
+++ b/usr/src/cmd/allocate/audio_clean.sh
@@ -19,12 +19,10 @@
 #
 # CDDL HEADER END
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
-# This is a wrapper for the audio_clean program.
+# This is the audio_clean program.
 # 
 # Following is the syntax for calling the script:
 #	scriptname [-s|-f|-i|-I] devicename [-A|-D] [username] [zonename]
@@ -56,16 +54,15 @@
 # dialogs.  To present dialogs to the user, the dtksh script
 # /etc/security/lib/wdwmsg may be used.
 #
-# This particular script, audio_clean_wrapper, will work using stdin/stdout, or
-# using dialogs.  A symbolic link audio_clean_wrapper.windowing points to
-# audio_clean_wrapper.
+# This particular script, audio_clean, will work using stdin/stdout, or
+# using dialogs.  A symbolic link audio_clean.windowing points to
+# audio_clean.
 
 
 trap "" INT TERM QUIT TSTP ABRT
 
 USAGE="usage: $0 [-s|-f|-i|-I] devicename [-A|-D][username][zonename][zonepath]"
 PATH="/usr/bin:/usr/sbin"
-CLEAN_PROG="/etc/security/lib/audio_clean"
 WDWMSG="/etc/security/lib/wdwmsg"
 MODE="allocate"
 
@@ -92,22 +89,13 @@ msg() {
 	fi
 }
 
-alloc_msg() {
-	msg "Audio device allocated in zone $ZONENAME." \
-	"\nTurn on microphone if audio recording is to be performed." \
-	"\nTurn microphone off when not recording."
-}
-
-dealloc_msg() {
-	msg "Please make sure the microphone is turned off."
-}
-
 fail_msg() {
 	if [ "$MODE" = "allocate" ]; then
 		msg "$0: Allocate of $DEVICE failed."
 	else
 		msg "$0: Deallocate of $DEVICE failed."
 	fi
+	exit 1
 }
 
 #
@@ -147,18 +135,33 @@ if [ "$MODE" != "allocate" -a "$MODE" != "deallocate" ]; then
 fi
 ZONENAME=$4
 ZONEPATH=$5
-
-$CLEAN_PROG -$FLAG $DEVICE
-
-if [ $? -ne 0 ]; then
-	fail_msg
-	exit 1
+SAVEDIR=/etc/security/audio
+MAP=`dminfo -v -n $AUDIO`
+DEVICE=`echo $MAP | cut -f1 -d:`
+TYPE=`echo $MAP | cut -f2 -d:`
+FILES=`echo $MAP | cut -f3 -d:`
+
+if [ ! -d ${SAVEDIR} ]
+then
+    /usr/bin/mkdir -m 0755 -p ${SAVEDIR} || fail_msg
+    /usr/bin/chown root:sys ${SAVEDIR} || fail_msg
 fi
 
-if [ "$MODE" = "allocate" ]; then
-	alloc_msg
+for d in $FILES
+do
+    x="`expr $d : '/dev/mixer[0-9][0-9]*'`"
+    if [ "$x" -ne 0 ] ; then
+	DEVNM=$d
+	break
+    fi
+done
+SAVEFILE="${SAVEDIR}/`basename ${DEVNM}`"
+
+if [ "${FLAG}" = "i" -a ! -r "${SAVEFILE}" ]
+then
+    /usr/sbin/mixerctl -d ${DEVNM} -f -s ${SAVEFILE} || fail_msg
 else
-	dealloc_msg
+    /usr/sbin/mixerctl -d ${DEVNM} -r ${SAVEFILE} || fail_msg
 fi
 
 exit 0
diff --git a/usr/src/cmd/allocate/devalloc_defaults b/usr/src/cmd/allocate/devalloc_defaults
index 56ef8ce53a..98cca2668d 100644
--- a/usr/src/cmd/allocate/devalloc_defaults
+++ b/usr/src/cmd/allocate/devalloc_defaults
@@ -19,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Default device_allocation attributes for device types.
 # Device types supported -
 #     audio, fd, sr, st, rmdisk
@@ -39,7 +37,7 @@
 # e.g.: minlabel="TS A B";maxlabel="TS AB";auths=solaris.device.allocate;\
 #       cleanscript=/bin/true
 #
-audio:minlabel=admin_low;maxlabel=admin_high;auths=solaris.device.allocate;cleanscript=/etc/security/lib/audio_clean_wrapper
+audio:minlabel=admin_low;maxlabel=admin_high;auths=solaris.device.allocate;cleanscript=/etc/security/lib/audio_clean
 fd:minlabel=admin_low;maxlabel=admin_high;auths=solaris.device.allocate;cleanscript=/etc/security/lib/disk_clean
 sr:minlabel=admin_low;maxlabel=admin_high;auths=solaris.device.allocate;cleanscript=/etc/security/lib/disk_clean
 st:minlabel=admin_low;maxlabel=admin_high;auths=solaris.device.allocate;cleanscript=/etc/security/lib/tape_clean
diff --git a/usr/src/cmd/audio/Makefile b/usr/src/cmd/audio/Makefile
index 73a66ce21b..20fdea8d51 100644
--- a/usr/src/cmd/audio/Makefile
+++ b/usr/src/cmd/audio/Makefile
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # cmd/audio/Makefile
 #
 
@@ -32,6 +29,7 @@ include ../Makefile.cmd
 
 SUBDIRS =			\
 		mixerctl	\
+		audiotest	\
 		utilities .WAIT \
 		audioplay	\
 		audiorecord	\
diff --git a/usr/src/cmd/audio/audioplay/audioplay.c b/usr/src/cmd/audio/audioplay/audioplay.c
index c898626a80..882c4d9087 100644
--- a/usr/src/cmd/audio/audioplay/audioplay.c
+++ b/usr/src/cmd/audio/audioplay/audioplay.c
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,12 +19,10 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 1993-2003 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 /* Command-line audio play utility */
 
 #include <stdio.h>
@@ -64,15 +61,12 @@
 /* Local variables */
 static char *prog;
 
-static char prog_opts[] =	"VEiv:b:d:p:?";	/* getopt() flags */
+static char prog_opts[] =	"VEiv:d:?";	/* getopt() flags */
 
 static char			*Stdin;
 
 #define	MAX_GAIN		(100)		/* maximum gain */
 
-#define	LEFT_BAL		(-100)		/* min/max balance */
-#define	MID_BAL			(0)
-#define	RIGHT_BAL		(100)
 /*
  * This defines the tolerable sample rate error as a ratio between the
  * sample rates of the audio data and the audio device.
@@ -86,12 +80,7 @@ static char			*Stdin;
 
 static unsigned		Volume = INT_MAX;	/* output volume */
 static double		Savevol;		/* saved volume level */
-static unsigned int	Balance = INT_MAX;	/* output balance */
-static unsigned int	Savebal;		/* saved output balance */
-static unsigned int	Port = INT_MAX;
-			/* output port (spkr, line, hp-jack) */
-static unsigned int	Saveport = 0;
-			/* save prev. val so we can restore */
+
 static int		Verbose = FALSE;	/* verbose messages */
 static int		Immediate = FALSE;
 			/* don't hang waiting for device */
@@ -153,7 +142,6 @@ static void sigint(int sig);
 static void open_audio(void);
 static int path_open(char *fname, int flags, mode_t mode, char *path);
 static int parse_unsigned(char *str, unsigned *dst, char *flag);
-static int scale_balance(int g);
 static int reconfig(void);
 static void initmux(int unitsz, int unitsp);
 static void demux(int unitsz, int cnt);
@@ -165,21 +153,15 @@ static void
 usage(void)
 {
 	Error(stderr, MGET("Play an audio file -- usage:\n"
-	    "\t%s [-iV] [-v vol] [-b bal]\n"
-	    "\t%.*s [-p speaker|headphone|line|aux1|aux2|spdif]\n"
-	    "\t%.*s [-d dev] [file ...]\n"
+	    "\t%s [-iV] [-v vol] [-d dev] [file ...]\n"
 	    "where:\n"
 	    "\t-i\tDon't hang if audio device is busy\n"
 	    "\t-V\tPrint verbose warning messages\n"
 	    "\t-v\tSet output volume (0 - %d)\n"
-	    "\t-b\tSet output balance (%d=left, %d=center, %d=right)\n"
-	    "\t-p\tSpecify output port\n"
 	    "\t-d\tSpecify audio device (default: /dev/audio)\n"
 	    "\tfile\tList of files to play\n"
 	    "\t\tIf no files specified, read stdin\n"),
-	    prog, strlen(prog), "                    ",
-	    strlen(prog), "                          ",
-	    MAX_GAIN, LEFT_BAL, MID_BAL, RIGHT_BAL);
+	    prog, MAX_GAIN);
 	exit(1);
 }
 
@@ -193,10 +175,6 @@ sigint(int sig)
 		/* restore saved parameters */
 		if (Volume != INT_MAX)
 			(void) audio_set_play_gain(Audio_fd, &Savevol);
-		if (Balance != INT_MAX)
-			(void) audio_set_play_balance(Audio_fd, &Savebal);
-		if (Port != INT_MAX)
-			(void) audio_set_play_port(Audio_fd, &Saveport);
 		if ((Audio_ctlfd >= 0) &&
 		    (audio_cmp_hdr(&Save_hdr, &Dev_hdr) != 0)) {
 			(void) audio_set_play_config(Audio_fd, &Save_hdr);
@@ -263,29 +241,6 @@ open_audio(void)
 			exit(1);
 		}
 	}
-
-	if (Balance != INT_MAX) {
-		(void) audio_get_play_balance(Audio_fd, &Savebal);
-		err = audio_set_play_balance(Audio_fd, &Balance);
-		if (err != AUDIO_SUCCESS) {
-			Error(stderr,
-			    MGET("%s: could not set output balance for %s\n"),
-			    prog, Audio_dev);
-			exit(1);
-		}
-	}
-
-	/* If -p flag, set the output port now */
-	if (Port != INT_MAX) {
-		(void) audio_get_play_port(Audio_fd, &Saveport);
-		err = audio_set_play_port(Audio_fd, &Port);
-		if (err != AUDIO_SUCCESS) {
-			Error(stderr,
-			    MGET("%s: could not set output port %s\n"),
-			    prog, Audio_dev);
-			exit(1);
-		}
-	}
 }
 
 /* Play a list of audio files. */
@@ -304,7 +259,6 @@ main(int argc, char **argv) {
 	int		ifd;
 	int		stdinseen;
 	int		regular;
-	int		bal;
 	int		swapBytes;
 	int		frame;
 	char		*outbuf;
@@ -342,16 +296,6 @@ main(int argc, char **argv) {
 					err++;
 				}
 				break;
-			case 'b':
-				bal = atoi(optarg);
-				if ((bal > RIGHT_BAL) || (bal < LEFT_BAL)) {
-					Error(stderr, MGET("%s: invalid value "
-					    "for -b\n"), prog);
-					err++;
-				} else {
-					Balance = (unsigned)scale_balance(bal);
-				}
-				break;
 			case 'd':
 				Audio_dev = optarg;
 				break;
@@ -364,32 +308,6 @@ main(int argc, char **argv) {
 			case 'i':
 				Immediate = TRUE;
 				break;
-			case 'p':
-				/* a partial match is OK */
-				if (strncmp(optarg, "speaker",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_SPEAKER;
-				} else if (strncmp(optarg, "headphone",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_HEADPHONE;
-				} else if (strncmp(optarg, "line",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_LINE_OUT;
-				} else if (strncmp(optarg, "aux1",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_AUX1_OUT;
-				} else if (strncmp(optarg, "aux2",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_AUX2_OUT;
-				} else if (strncmp(optarg, "spdif",
-				    strlen(optarg)) == 0) {
-					Port = AUDIO_SPDIF_OUT;
-				} else {
-					Error(stderr, MGET("%s: invalid value "
-					    "for -p\n"), prog);
-					err++;
-				}
-				break;
 			case '?':
 				usage();
 		/*NOTREACHED*/
@@ -416,13 +334,8 @@ main(int argc, char **argv) {
 
 	/* This should probably use audio_cntl instead of open_audio */
 	if ((argc <= 0) && isatty(fileno(stdin))) {
-		if (Verbose) {
-			Error(stderr,
-		    MGET("%s: No files - setting audio device parameters.\n"),
-			    prog);
-		}
-		open_audio();
-		exit(0);
+		Error(stderr, MGET("%s: No files and stdin is a tty.\n"), prog);
+		exit(1);
 	}
 
 	/* Check on the -i status now. */
@@ -438,7 +351,7 @@ main(int argc, char **argv) {
 	Audio_fd = -1;
 
 	/* Try to open the control device and save the current format */
-	(void) sprintf(ctldev, "%sctl", Audio_dev);
+	(void) snprintf(ctldev, sizeof (ctldev), "%sctl", Audio_dev);
 	Audio_ctlfd = open(ctldev, O_RDWR);
 	if (Audio_ctlfd >= 0) {
 		/*
@@ -521,7 +434,7 @@ main(int argc, char **argv) {
 			File_hdr.samples_per_unit = 1;
 			File_hdr.bytes_per_unit = 1;
 			adpcm_state = (struct audio_g72x_state *)malloc
-				(sizeof (*adpcm_state) * File_hdr.channels);
+			    (sizeof (*adpcm_state) * File_hdr.channels);
 			for (i = 0; i < File_hdr.channels; i++) {
 				g721_init_state(&adpcm_state[i]);
 			}
@@ -533,7 +446,7 @@ main(int argc, char **argv) {
 			File_hdr.samples_per_unit = 1;
 			File_hdr.bytes_per_unit = 1;
 			adpcm_state = (struct audio_g72x_state *)malloc
-				(sizeof (*adpcm_state) * File_hdr.channels);
+			    (sizeof (*adpcm_state) * File_hdr.channels);
 			for (i = 0; i < File_hdr.channels; i++) {
 				g723_init_state(&adpcm_state[i]);
 			}
@@ -911,10 +824,6 @@ nextfile:;
 
 	if (Volume != INT_MAX)
 		(void) audio_set_play_gain(Audio_fd, &Savevol);
-	if (Balance != INT_MAX)
-		(void) audio_set_play_balance(Audio_fd, &Savebal);
-	if (Port != INT_MAX)
-		(void) audio_set_play_port(Audio_fd, &Saveport);
 	if ((Audio_ctlfd >= 0) && (audio_cmp_hdr(&Save_hdr, &Dev_hdr) != 0)) {
 		(void) audio_set_play_config(Audio_fd, &Save_hdr);
 	}
@@ -958,7 +867,8 @@ reconfig(void)
 			if (ratio <= SAMPLE_RATE_THRESHOLD) {
 				if (Verbose) {
 					Error(stderr,
-			MGET("%s: WARNING: %s sampled at %d, playing at %d\n"),
+					    MGET("%s: WARNING: %s sampled at "
+					    "%d, playing at %d\n"),
 					    prog, Ifile, File_hdr.sample_rate,
 					    Dev_hdr.sample_rate);
 				}
@@ -1067,7 +977,8 @@ path_open(char *fname, int flags, mode_t mode, char *path)
 				/* got a match! */
 				if (Verbose) {
 					Error(stderr,
-				    MGET("%s: Found %s in path at %s\n"),
+					    MGET("%s: Found %s in path "
+					    "at %s\n"),
 					    prog, fname, fullpath);
 				}
 				return (open(fullpath, flags, mode));
@@ -1085,14 +996,6 @@ path_open(char *fname, int flags, mode_t mode, char *path)
 }
 
 
-/* Convert local balance into device parameters */
-static int
-scale_balance(int g)
-{
-	return (int)(((g + RIGHT_BAL) / (double)(RIGHT_BAL - LEFT_BAL)) *
-	    (double)AUDIO_RIGHT_BALANCE);
-}
-
 /*
  * initmux()
  *
@@ -1119,47 +1022,41 @@ initmux(int unitsz, int unitsp)
 	out_ch_size = in_ch_size * unitsp / unitsz;
 
 	/* Allocate pointers to input channels */
-	in_ch_data = (unsigned char **)malloc(sizeof (unsigned char *)
-		* File_hdr.channels);
+	in_ch_data = malloc(sizeof (unsigned char *) * File_hdr.channels);
 
 	if (in_ch_data == NULL) {
-		Error(stderr, MGET("%s: couldn't allocate %dK "
-			"buf\n"), prog, sizeof (unsigned char *) *
-			File_hdr.channels / 1000);
+		Error(stderr, MGET("%s: couldn't allocate %dK buf\n"),
+		    prog, sizeof (unsigned char *) * File_hdr.channels / 1000);
 		exit(1);
 	}
 
 	/* Allocate input channels */
 	for (c = 0; c < File_hdr.channels; c++) {
-		in_ch_data[c] = (unsigned char *)malloc
-			(sizeof (unsigned char) * in_ch_size);
+		in_ch_data[c] = malloc(sizeof (unsigned char) * in_ch_size);
 
 		if (in_ch_data[c] == NULL) {
-			Error(stderr, MGET("%s: couldn't allocate %dK "
-				"buf\n"), prog, in_ch_size / 1000);
+			Error(stderr, MGET("%s: couldn't allocate %dK buf\n"),
+			    prog, in_ch_size / 1000);
 			exit(1);
 		}
 	}
 
 	/* Allocate pointers to output channels */
-	out_ch_data = (unsigned char **)malloc(sizeof (unsigned char *)
-		* File_hdr.channels);
+	out_ch_data = malloc(sizeof (unsigned char *) * File_hdr.channels);
 
 	if (out_ch_data == NULL) {
-		Error(stderr, MGET("%s: couldn't allocate %dK "
-			"buf\n"), prog, sizeof (unsigned char *) *
-			File_hdr.channels / 1000);
+		Error(stderr, MGET("%s: couldn't allocate %dK buf\n"),
+		    prog, sizeof (unsigned char *) * File_hdr.channels / 1000);
 		exit(1);
 	}
 
 	/* Allocate output channels */
 	for (c = 0; c < File_hdr.channels; c++) {
-		out_ch_data[c] = (unsigned char *)malloc
-			(sizeof (unsigned char) * out_ch_size);
+		out_ch_data[c] = malloc(sizeof (unsigned char) * out_ch_size);
 
 		if (out_ch_data[c] == NULL) {
-			Error(stderr, MGET("%s: couldn't allocate %dK "
-				"buf\n"), prog, out_ch_size / 1000);
+			Error(stderr, MGET("%s: couldn't allocate %dK buf\n"),
+			    prog, out_ch_size / 1000);
 			exit(1);
 		}
 	}
diff --git a/usr/src/cmd/audio/audiorecord/audiorecord.c b/usr/src/cmd/audio/audiorecord/audiorecord.c
index dff332466a..d8b681d777 100644
--- a/usr/src/cmd/audio/audiorecord/audiorecord.c
+++ b/usr/src/cmd/audio/audiorecord/audiorecord.c
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,12 +19,10 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 /* Command-line audio record utility */
 
 #include <stdio.h>
@@ -62,16 +59,11 @@
 #define	TEXT_DOMAIN "SYS_TEST"	/* Use this only if it weren't */
 #endif
 
-/* Defined until I get a copy of the apropriate audioio.h file. */
-#ifndef AUDIO_CD_IN
-#define	AUDIO_CD_IN 0x04	/* input from the internal CD player */
-#endif
-
 #define	Error		(void) fprintf
 
 /* Local variables */
 static char	*prog;
-static char	prog_opts[] = "aft:v:b:m:d:p:i:e:s:c:T:?"; /* getopt() flags */
+static char	prog_opts[] = "aft:v:d:i:e:s:c:T:?"; /* getopt() flags */
 static char	*Stdout;
 
 /* XXX - the input buffer size should depend on sample_rate */
@@ -82,20 +74,10 @@ static char 		swapBuf[AUDIO_BUFSIZ];	/* for byte swapping */
 
 #define	MAX_GAIN		(100)	/* maximum gain */
 
-#define	LEFT_BAL		(-100)	/* min/max balance */
-#define	MID_BAL			(0)
-#define	RIGHT_BAL		(100)
-
 static char	*Info = NULL;		/* pointer to info data */
 static unsigned	Ilen = 0;		/* length of info data */
 static unsigned	Volume = INT_MAX;	/* record volume */
 static double	Savevol;		/* saved  volume */
-static unsigned	Monvol = INT_MAX;	/* monitor volume */
-static double	Savemonvol;		/* saved monitor volume */
-static unsigned int	Balance = INT_MAX;	/* input balance */
-static unsigned int	Savebal;		/* saved input balance */
-static unsigned	Port = INT_MAX;		/* Input port (line, mic) */
-static unsigned	Saveport = 0;		/* restore value of input port */
 static unsigned	Sample_rate = 0;
 static unsigned	Channels = 0;
 static unsigned	Precision = 0;		/* based on encoding */
@@ -130,7 +112,6 @@ extern char *optarg;
 /* Local Functions */
 static void usage(void);
 static void sigint(int sig);
-static int scale_balance(int g);
 static int parse_unsigned(char *str, unsigned *dst, char *flag);
 static int parse_sample_rate(char *s, unsigned *rate);
 
@@ -139,17 +120,13 @@ static void
 usage(void)
 {
 	Error(stderr, MGET("Record an audio file -- usage:\n"
-	    "\t%s [-af] [-v vol] [-b bal] [-m monvol]\n"
-	    "\t%.*s [-p mic|line|cd|aux1|aux2|spdif]\n"
+	    "\t%s [-af] [-v vol]\n"
 	    "\t%.*s [-c channels] [-s rate] [-e encoding]\n"
 	    "\t%.*s [-t time] [-i info] [-d dev] [-T au|wav|aif[f]] [file]\n"
 	    "where:\n"
 	    "\t-a\tAppend to output file\n"
 	    "\t-f\tIgnore sample rate differences on append\n"
 	    "\t-v\tSet record volume (0 - %d)\n"
-	    "\t-b\tSet record balance (%d=left, %d=center, %d=right)\n"
-	    "\t-m\tSet monitor volume (0 - %d)\n"
-	    "\t-p\tSpecify input port\n"
 	    "\t-c\tSpecify number of channels to record\n"
 	    "\t-s\tSpecify rate in samples per second\n"
 	    "\t-e\tSpecify encoding (ulaw | alaw | [u]linear | linear8 )\n"
@@ -164,8 +141,7 @@ usage(void)
 	    prog,
 	    strlen(prog), "                    ",
 	    strlen(prog), "                    ",
-	    strlen(prog), "                    ",
-	    MAX_GAIN, LEFT_BAL, MID_BAL, RIGHT_BAL, MAX_GAIN);
+	    MAX_GAIN);
 	exit(1);
 }
 
@@ -185,12 +161,6 @@ sigint(int sig)
 	if (Audio_fd >= 0) {
 		if (Volume != INT_MAX)
 			(void) audio_set_record_gain(Audio_fd, &Savevol);
-		if (Balance != INT_MAX)
-			(void) audio_set_record_balance(Audio_fd, &Savebal);
-		if (Monvol != INT_MAX)
-			(void) audio_set_monitor_gain(Audio_fd, &Savemonvol);
-		if (Port != INT_MAX)
-			(void) audio_set_record_port(Audio_fd, &Saveport);
 		if (audio_cmp_hdr(&Save_hdr, &Dev_hdr) != 0) {
 			(void) audio_set_record_config(Audio_fd, &Save_hdr);
 		}
@@ -211,11 +181,9 @@ main(int argc, char **argv)
 	int		ofd;
 	int 		swapBytes = FALSE;
 	double		vol;
-	int		bal;
 	struct stat	st;
 	struct pollfd	pfd;
 	char		*cp;
-	char		ctldev[MAXPATHLEN];
 
 	(void) setlocale(LC_ALL, "");
 	(void) textdomain(TEXT_DOMAIN);
@@ -250,25 +218,6 @@ main(int argc, char **argv)
 				err++;
 			}
 			break;
-		case 'b':
-			bal = atoi(optarg);
-			if ((bal > RIGHT_BAL) || (bal < LEFT_BAL)) {
-				Error(stderr, MGET("%s: invalid value for "
-				"-b\n"), prog);
-				err++;
-			} else {
-				Balance = (unsigned)scale_balance(bal);
-			}
-			break;
-		case 'm':
-			if (parse_unsigned(optarg, &Monvol, "-m")) {
-				err++;
-			} else if (Monvol > MAX_GAIN) {
-				Error(stderr, MGET("%s: invalid value for "
-				"-m\n"), prog);
-				err++;
-			}
-			break;
 		case 't':
 			Time = audio_str_to_secs(optarg);
 			if ((Time == HUGE_VAL) || (Time < 0.)) {
@@ -280,33 +229,6 @@ main(int argc, char **argv)
 		case 'd':
 			Audio_dev = optarg;
 			break;
-		case 'p':
-			/* a partial match is OK */
-			if (strncmp(optarg, "microphone",
-			    strlen(optarg)) == 0) {
-				Port = AUDIO_MICROPHONE;
-			} else if (strncmp(optarg, "line",
-			    strlen(optarg)) == 0) {
-				Port = AUDIO_LINE_IN;
-			} else if ((strncmp(optarg, "cd",
-			    strlen(optarg)) == 0) || (strncmp(optarg,
-			    "internal-cd", strlen(optarg)) == 0)) {
-				Port = AUDIO_CD_IN;
-			} else if (strncmp(optarg, "aux1",
-			    strlen(optarg)) == 0) {
-				Port = AUDIO_AUX1_IN;
-			} else if (strncmp(optarg, "aux2",
-			    strlen(optarg)) == 0) {
-				Port = AUDIO_AUX2_IN;
-			} else if (strncmp(optarg, "spdif",
-			    strlen(optarg)) == 0) {
-				Port = AUDIO_SPDIF_IN;
-			} else {
-				Error(stderr, MGET("%s: invalid value for "
-				"-p\n"), prog);
-				err++;
-			}
-			break;
 		case 'f':
 			Force = TRUE;
 			break;
@@ -703,44 +625,10 @@ openinput:
 		}
 	}
 
-	if (Balance != INT_MAX) {
-		(void) audio_get_record_balance(Audio_fd, &Savebal);
-		err = audio_set_record_balance(Audio_fd, &Balance);
-		if (err != AUDIO_SUCCESS) {
-			Error(stderr,
-			    MGET("%s: could not set record balance for %s\n"),
-			    prog, Audio_dev);
-			exit(1);
-		}
-	}
-
-	/* If -m flag, set monitor volume now */
-	if (Monvol != INT_MAX) {
-		vol = (double)Monvol / (double)MAX_GAIN;
-		(void) audio_get_monitor_gain(Audio_fd, &Savemonvol);
-		err = audio_set_monitor_gain(Audio_fd, &vol);
-		if (err != AUDIO_SUCCESS) {
-			Error(stderr,
-			    MGET("%s: could not set monitor volume for %s\n"),
-			    prog, Audio_dev);
-			exit(1);
-		}
-	}
-
-	/* If -p flag, set the input port */
-	if (Port != INT_MAX) {
-		(void) audio_get_record_port(Audio_fd, &Saveport);
-		err = audio_set_record_port(Audio_fd, &Port);
-		if (err != AUDIO_SUCCESS) {
-			Error(stderr,
-			    MGET("%s: could not set input port %s\n"),
-			    prog, Audio_dev);
-			exit(1);
-		}
-	}
-
 	if (isatty(ofd)) {
-		exit(0);
+		Error(stderr, MGET("%s: No files and stdout is a tty\n"),
+		    prog);
+		exit(1);
 	}
 
 	/* Set up SIGINT handler so that final buffers may be flushed */
@@ -823,15 +711,9 @@ openinput:
 		Error(stderr, MGET("%s: WARNING: Data overflow occurred\n"),
 		    prog);
 
-	/* Reset record volume, balance, monitor volume, port, encoding */
+	/* Reset record volume, encoding */
 	if (Volume != INT_MAX)
 		(void) audio_set_record_gain(Audio_fd, &Savevol);
-	if (Balance != INT_MAX)
-		(void) audio_set_record_balance(Audio_fd, &Savebal);
-	if (Monvol != INT_MAX)
-		(void) audio_set_monitor_gain(Audio_fd, &Savemonvol);
-	if (Port != INT_MAX)
-		(void) audio_set_record_port(Audio_fd, &Saveport);
 	if (audio_cmp_hdr(&Save_hdr, &Dev_hdr) != 0) {
 		(void) audio_set_record_config(Audio_fd, &Save_hdr);
 	}
@@ -886,7 +768,8 @@ parse_sample_rate(char *s, unsigned *rate)
 		 * XXX bug alert: could have multiple "." in string
 		 * and mess things up.
 		 */
-		for (cp = s; *cp && (isdigit(*cp) || (*cp == '.')); cp++);
+		for (cp = s; *cp && (isdigit(*cp) || (*cp == '.')); cp++)
+			/* NOP */;
 		if (*cp != NULL) {
 			if ((*cp == 'k') || (*cp == 'K')) {
 				drate *= 1000.0;
@@ -903,11 +786,3 @@ parse_sample_rate(char *s, unsigned *rate)
 	*rate = irint(drate);
 	return (0);
 }
-
-/* Convert local balance into device parameters */
-static int
-scale_balance(int g)
-{
-	return (int)(((g + RIGHT_BAL) / (double)(RIGHT_BAL - LEFT_BAL)) *
-	    (double)AUDIO_RIGHT_BALANCE);
-}
diff --git a/usr/src/cmd/audio/audiotest/Makefile b/usr/src/cmd/audio/audiotest/Makefile
new file mode 100644
index 0000000000..ad3033716c
--- /dev/null
+++ b/usr/src/cmd/audio/audiotest/Makefile
@@ -0,0 +1,50 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Makefile for cmd/audio/audiotest
+
+PROG= audiotest
+OBJS= audiotest.o wavedata.o
+SRCS= $(OBJS:%.o=%.c)
+
+include ../../Makefile.cmd
+
+CFLAGS += $(CCVERBOSE)
+XGETFLAGS += -a -x audiotest.xcl
+
+.KEEP_STATE:
+
+all: $(PROG)
+
+$(PROG):	$(OBJS)
+	$(LINK.c) $(OBJS) -o $@
+	$(POST_PROCESS)
+
+install: all $(ROOTPROG)
+
+clean:
+
+lint:	lint_SRCS
+
+include ../../Makefile.targ
diff --git a/usr/src/cmd/audio/audiotest/audiotest.c b/usr/src/cmd/audio/audiotest/audiotest.c
new file mode 100644
index 0000000000..c995ea16e9
--- /dev/null
+++ b/usr/src/cmd/audio/audiotest/audiotest.c
@@ -0,0 +1,575 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * This program is a general purpose test facility for audio output.
+ * It does not test record.
+ *
+ * The wavedata.c and wavedata.h files contain the actual samples compressed
+ * using the MS ADPCM algorithm.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/utsname.h>
+#include <sys/soundcard.h>
+#include <inttypes.h>
+#include <locale.h>
+
+#if !defined(TEXT_DOMAIN)	/* Should be defined by cc -D */
+#define	TEXT_DOMAIN "SYS_TEST"	/* Use this only if it weren't */
+#endif
+
+#define	_(s)	gettext(s)
+
+/*
+ * Channel selectors
+ */
+#define	CH_LEFT		(1 << 0)
+#define	CH_RIGHT	(1 << 1)
+#define	CH_LREAR4	(1 << 2)	/* quadraphonic */
+#define	CH_RREAR4	(1 << 3)	/* quadraphonic */
+#define	CH_CENTER	(1 << 2)
+#define	CH_LFE		(1 << 3)
+#define	CH_LSURR	(1 << 4)
+#define	CH_RSURR	(1 << 5)
+#define	CH_LREAR	(1 << 6)
+#define	CH_RREAR	(1 << 7)
+#define	CH_STEREO	(CH_LEFT|CH_RIGHT)
+#define	CH_4		(CH_STEREO | CH_LREAR4 | CH_RREAR4)
+#define	CH_5		(CH_STEREO | CH_CENTER | CH_LSURR | CH_RSURR)
+#define	CH_7		(CH_5 | CH_LREAR | CH_RREAR)
+
+typedef struct chancfg {
+	int		mask;
+	const char	*name;
+	unsigned	flags;
+	int16_t		*data;
+	int		len;
+} chancfg_t;
+
+typedef struct testcfg {
+	int		nchan;
+	chancfg_t	*tests[16];
+} testcfg_t;
+
+#define	CFLAG_LFE	0x1	/* lfe channel - not full range */
+
+/*
+ * TRANSLATION_NOTE : The following strings are displayed during progress.
+ * Its important for alignment that they have the same displayed length.
+ */
+#define	NM_LEFT		"\t<left> ................"
+#define	NM_RIGHT	"\t<right> ..............."
+#define	NM_LREAR	"\t<left rear> ..........."
+#define	NM_RREAR	"\t<right rear> .........."
+#define	NM_LSIDE	"\t<left side> ..........."
+#define	NM_RSIDE	"\t<right side> .........."
+#define	NM_CENTER	"\t<center> .............."
+#define	NM_LFE		"\t<lfe> ................."
+#define	NM_STEREO	"\t<stereo> .............."
+#define	NM_40		"\t<4.0 surround> ........"
+#define	NM_50		"\t<5.0 surround> ........"
+#define	NM_70		"\t<7.0 surround> ........"
+
+chancfg_t ch_left = { CH_LEFT, NM_LEFT, 0 };
+chancfg_t ch_right = { CH_RIGHT, NM_RIGHT, 0 };
+chancfg_t ch_stereo = { CH_STEREO, NM_STEREO, 0 };
+
+chancfg_t ch_center = { CH_CENTER, NM_CENTER, 0 };
+chancfg_t ch_lfe = { CH_LFE, NM_LFE, CFLAG_LFE };
+
+chancfg_t ch_lsurr_4 = { (1 << 2), NM_LREAR, 0 };
+chancfg_t ch_rsurr_4 = { (1 << 3), NM_RREAR, 0 };
+chancfg_t ch_4 = { CH_4, NM_40, 0 };
+
+chancfg_t ch_lsurr_5 = { CH_LSURR, NM_LREAR, 0 };
+chancfg_t ch_rsurr_5 = { CH_RSURR, NM_RREAR, 0 };
+chancfg_t ch_5 = { CH_5, NM_50, 0 };
+
+chancfg_t ch_lsurr_7 = { CH_LSURR, NM_LSIDE, 0 };
+chancfg_t ch_rsurr_7 = { CH_RSURR, NM_RSIDE, 0 };
+chancfg_t ch_lrear_7 = { CH_LREAR, NM_LREAR, 0 };
+chancfg_t ch_rrear_7 = { CH_RREAR, NM_RREAR, 0 };
+chancfg_t ch_7 = { CH_7, NM_70, 0 };
+
+testcfg_t test_stereo = {
+	2, { &ch_left, &ch_right, &ch_stereo, NULL }
+};
+
+testcfg_t test_quad = {
+	4, { &ch_left, &ch_right, &ch_stereo,
+	&ch_lsurr_4, &ch_rsurr_4, &ch_4, NULL }
+};
+
+testcfg_t test_51 = {
+	6, { &ch_left, &ch_right, &ch_stereo,
+	&ch_lsurr_5, &ch_rsurr_5, &ch_center, &ch_lfe, &ch_5, NULL }
+};
+
+testcfg_t test_71 = {
+	8, { &ch_left, &ch_right, &ch_stereo,
+	&ch_lsurr_7, &ch_rsurr_7, &ch_lrear_7, &ch_rrear_7,
+	&ch_center, &ch_lfe, &ch_7, NULL }
+};
+
+/*
+ * uncompress_wave() is defined in wavedata.c. It expands the audio
+ * samples stored in wavedata.h and returns the lenghth of the
+ * uncompressed version in bytes.
+ *
+ * The uncompressed wave data format is 16 bit (native) stereo
+ * recorded at 48000 Hz.
+ */
+extern int uncompress_wave(short *outbuf);
+
+static int data_len;
+
+#define	MAXDEVICE   64
+extern void describe_error(int);
+
+#define	SAMPLE_RATE 48000
+
+/*
+ * Operating mode flags (set from the command line).
+ */
+#define	TF_LOOP		0x00000010	/* Loop until interrupted */
+
+static int mixerfd;
+static int num_devices_tested = 0;
+
+static short *sample_buf;
+
+void
+prepare(testcfg_t *tcfg)
+{
+	int	nsamples;
+	int	i;
+	chancfg_t	*ccfg;
+	if ((sample_buf = malloc(2000000)) == NULL) {
+		perror("malloc");
+		exit(-1);
+	}
+
+	data_len = uncompress_wave(sample_buf);
+	nsamples = (data_len / sizeof (int16_t)) / 2;
+
+	for (i = 0; (ccfg = tcfg->tests[i]) != NULL; i++) {
+		int16_t		*src, *dst;
+		int		ch;
+		int		samp;
+
+		src = sample_buf;
+
+		if (ccfg->flags != CFLAG_LFE) {
+			ccfg->len = nsamples * tcfg->nchan * sizeof (int16_t);
+			ccfg->data = malloc(ccfg->len);
+			if ((dst = ccfg->data) == NULL) {
+				perror("malloc");
+				exit(-1);
+			}
+			for (samp = 0; samp < nsamples; samp++) {
+				for (ch = 0; ch < tcfg->nchan; ch++) {
+					*dst = ((1U << ch) & ccfg->mask) ?
+					    *src : 0;
+					dst++;
+				}
+				src += 2;
+			}
+		} else {
+			/* Skip LFE for now */
+			ccfg->len = 0;
+		}
+	}
+}
+
+/*
+ * The testdsp() routine checks the capabilities of a given audio device number
+ * (parameter n) and decides if the test sound needs to be played.
+ */
+
+/*ARGSUSED*/
+int
+testdsp(int hd, int flags, testcfg_t *tcfg)
+{
+	float ratio;
+	struct timeval t1, t2;
+	unsigned long t;
+	int sample_rate;
+	int delay;
+	long long total_bytes = 0;
+	unsigned int tmp, caps;
+	int i;
+	chancfg_t *ccfg;
+
+	caps = 0;
+	if (ioctl(hd, SNDCTL_DSP_GETCAPS, &caps) == -1) {
+		perror("SNDCTL_DSP_GETCAPS");
+		return (-1);
+	}
+
+	/*
+	 * Setup the sample format. Since OSS will support AFMT_S16_NE
+	 * regardless of the device we do not need to support any
+	 * other formats.
+	 */
+
+	tmp = AFMT_S16_NE;
+	if (ioctl(hd, SNDCTL_DSP_SETFMT, &tmp) == -1 || tmp != AFMT_S16_NE) {
+		(void) printf(_("Device doesn't support native 16-bit PCM\n"));
+		return (-1);
+	}
+
+	/*
+	 * Setup the device for channels. Once again we can simply
+	 * assume that stereo will always work before OSS takes care
+	 * of this by emulation if necessary.
+	 */
+	tmp = tcfg->nchan;
+	if (ioctl(hd, SNDCTL_DSP_CHANNELS, &tmp) == -1 || tmp != tcfg->nchan) {
+		(void) printf(_("The device doesn't support %d channels\n"),
+		    tcfg->nchan);
+		return (-2);
+	}
+
+	/*
+	 * Set up the sample rate.
+	 */
+
+	tmp = SAMPLE_RATE;
+	if (ioctl(hd, SNDCTL_DSP_SPEED, &tmp) == -1) {
+		perror("SNDCTL_DSP_SPEED");
+		return (-3);
+	}
+
+	sample_rate = tmp;
+	if (sample_rate != SAMPLE_RATE) {
+		(void) printf(_("The device doesn't support %d Hz\n"),
+		    SAMPLE_RATE);
+		return (-3);
+	}
+	(void) printf("\n");
+
+	/*
+	 * This program will measure the real sampling rate by
+	 * computing the total time required to play the sample.
+	 *
+	 * This is not terribly presice with short test sounds but it
+	 * can be used to detect if the sampling rate badly
+	 * wrong. Errors of few percents is more likely to be caused
+	 * by poor accuracy of the system clock rather than problems
+	 * with the sampling rate.
+	 */
+	(void) gettimeofday(&t1, NULL);
+
+	for (i = 0; (ccfg = tcfg->tests[i]) != NULL; i++) {
+		(void) fputs(_(ccfg->name), stdout);
+		(void) fflush(stdout);
+		if (ccfg->flags & CFLAG_LFE) {
+			(void) printf(_("SKIPPED\n"));
+			continue;
+		}
+
+		if (write(hd, ccfg->data, ccfg->len) < 0) {
+			(void) printf(_("ERROR: %s\n"),
+			    strerror(errno));
+			return (-3);
+		}
+		(void) printf(_("OK\n"));
+		total_bytes += ccfg->len;
+	}
+
+	(void) gettimeofday(&t2, NULL);
+	delay = 0;
+	(void) ioctl(hd, SNDCTL_DSP_GETODELAY, &delay);	/* Ignore errors */
+
+	/*
+	 * Perform the time computations using milliseconds.
+	 */
+
+	t = t2.tv_sec - t1.tv_sec;
+	t *= 1000;
+
+	t += t2.tv_usec / 1000;
+	t -= t1.tv_usec / 1000;
+
+	total_bytes -= delay;
+	total_bytes *= 1000;
+
+	total_bytes /= t;
+	total_bytes /= (tcfg->nchan * sizeof (int16_t));
+
+	ratio = ((float)total_bytes / (float)sample_rate) * 100.0;
+	(void) printf(_("\t<measured sample rate %8.2f Hz (%4.2f%%)>\n"),
+	    (float)sample_rate * ratio / 100.0, ratio - 100.0);
+	num_devices_tested++;
+
+	return (1);
+}
+
+static int
+find_num_devices(void)
+{
+	oss_sysinfo info;
+	struct utsname un;
+	/*
+	 * Find out the number of available audio devices by calling
+	 * SNDCTL_SYSINFO.
+	 */
+
+	if (ioctl(mixerfd, SNDCTL_SYSINFO, &info) == -1) {
+		if (errno == ENXIO) {
+			(void) fprintf(stderr,
+			    _("No supported sound hardware detected.\n"));
+			exit(-1);
+		} else {
+			perror("SNDCTL_SYSINFO");
+			(void) printf(_("Cannot get system information.\n"));
+			exit(-1);
+		}
+	}
+	(void) printf(_("Sound subsystem and version: %s %s (0x%08X)\n"),
+	    info.product, info.version, info.versionnum);
+
+	if (uname(&un) != -1)
+		(void) printf(_("Platform: %s %s %s %s\n"),
+		    un.sysname, un.release, un.version, un.machine);
+
+	return (info.numaudios);
+}
+
+/*
+ * The test_device() routine checks certain information about the device
+ * and calls testdsp() to play the test sound.
+ */
+
+int
+test_device(char *dn, int flags, testcfg_t *tcfg)
+{
+	oss_audioinfo ainfo;
+	int code;
+	int fd;
+
+	fd = open(dn, O_WRONLY, 0);
+	if (fd == -1) {
+		int err = errno;
+		perror(dn);
+		errno = err;
+		describe_error(errno);
+		return (0);
+	}
+
+	ainfo.dev = -1;
+	if (ioctl(fd, SNDCTL_AUDIOINFO, &ainfo) == -1) {
+		perror("SNDCTL_AUDIOINFO");
+		(void) close(fd);
+		return (1);
+	}
+
+	(void) printf(_("\n*** Scanning sound adapter #%d ***\n"),
+	    ainfo.card_number);
+
+	(void) printf(_("%s (audio engine %d): %s\n"), ainfo.devnode, ainfo.dev,
+	    ainfo.name);
+
+	if (!ainfo.enabled) {
+		(void) printf(_("  - Device not present - Skipping\n"));
+		(void) close(fd);
+		return (1);
+	}
+
+	if (!(ainfo.caps & PCM_CAP_OUTPUT)) {
+		(void) printf(_("  - Skipping input only device\n"));
+		(void) close(fd);
+		return (1);
+	}
+
+	(void) printf(_("  - Performing audio playback test... "));
+	(void) fflush(stdout);
+
+	code = testdsp(fd, flags, tcfg);
+	(void) close(fd);
+
+	return (code == 1);
+}
+
+void
+describe_error(int err)
+{
+	switch (err) {
+	case ENODEV:
+		(void) fprintf(stderr,
+		    _("The device file was found in /dev but\n"
+		    "the driver was not loaded.\n"));
+		break;
+
+	case ENXIO:
+		(void) fprintf(stderr,
+		    _("There are no sound devices available.\n"
+		    "The most likely reason is that the device you have\n"
+		    "is malfunctioning or it's not supported.\n"
+		    "It's also possible that you are trying to use the wrong "
+		    "device file.\n"));
+		break;
+
+	case ENOSPC:
+		(void) fprintf(stderr,
+		    _("Your system cannot allocate memory for the device\n"
+		    "buffers. Reboot your machine and try again.\n"));
+		break;
+
+	case ENOENT:
+		(void) fprintf(stderr,
+		    _("The device file is missing from /dev.\n"));
+		break;
+
+
+	case EBUSY:
+		(void) fprintf(stderr,
+		    _("The device is busy. There is some other application\n"
+		    "using it.\n"));
+		break;
+
+	default:
+		break;
+	}
+}
+
+int
+main(int argc, char *argv[])
+{
+	int t, i;
+	int maxdev;
+	int flags = 0;
+	int status = 0;
+	int numdev;
+	extern int optind;
+	testcfg_t	*tcfg;
+
+	(void) setlocale(LC_ALL, "");
+	(void) textdomain(TEXT_DOMAIN);
+
+	tcfg = &test_stereo;
+
+	/*
+	 * Simple command line switch handling.
+	 */
+
+	while ((i = getopt(argc, argv, "l2457")) != EOF) {
+		switch (i) {
+		case 'l':
+			flags |= TF_LOOP;
+			break;
+		case '2':
+			tcfg = &test_stereo;
+			break;
+		case '4':
+			tcfg = &test_quad;
+			break;
+		case '5':
+			tcfg = &test_51;
+			break;
+		case '7':
+			tcfg = &test_71;
+			break;
+		default:
+			(void) printf(_("Usage: %s [options...] [device]\n"
+			    "	-2	Stereo test\n"
+			    "	-4	Quadraphonic 4.0 test\n"
+			    "	-5	Surround 5.1 test\n"
+			    "	-7	Surround 7.1 test\n"
+			    "	-l	Loop test\n"), argv[0]);
+			exit(-1);
+		}
+	}
+
+	/*
+	 * Open the mixer device used for calling SNDCTL_SYSINFO and
+	 * SNDCTL_AUDIOINFO.
+	 */
+	if ((mixerfd = open("/dev/mixer", O_RDWR, 0)) == -1) {
+		int err = errno;
+		perror("/dev/mixer");
+		errno = err;
+		describe_error(errno);
+		exit(-1);
+	}
+
+	prepare(tcfg);			/* Prepare the wave data */
+
+	/*
+	 * Enumerate all devices and play the test sounds.
+	 */
+	maxdev = find_num_devices();
+	if (maxdev < 1) {
+		(void) printf(_("\n*** No audio hardware available ***\n"));
+		exit(-1);
+	}
+
+	numdev = (argc - optind);
+	do {
+		char *dn;
+		oss_audioinfo	ainfo;
+
+		if (numdev > 0) {
+			for (t = 0; t < numdev; t++) {
+				dn = argv[optind + t];
+				if (!test_device(dn, flags, tcfg))
+					status++;
+			}
+		} else {
+			for (t = 0; t < maxdev; t++) {
+				ainfo.dev = t;
+				if (ioctl(mixerfd, SNDCTL_AUDIOINFO,
+				    &ainfo) == -1) {
+					perror("SNDCTL_AUDIOINFO");
+					status++;
+					continue;
+				}
+				dn = ainfo.devnode;
+				if (!test_device(dn, flags, tcfg))
+					status++;
+			}
+		}
+
+		if (status == 0)
+			(void) printf(_("\n*** All tests completed OK ***\n"));
+		else
+			(void) printf(_("\n*** Errors were detected ***\n"));
+
+	} while (flags & TF_LOOP);
+
+	(void) close(mixerfd);
+
+	return (status);
+}
diff --git a/usr/src/cmd/mdb/sparc/v9/audiosup/Makefile b/usr/src/cmd/audio/audiotest/audiotest.xcl
index 33e20a4a01..42057d7af3 100644
--- a/usr/src/cmd/mdb/sparc/v9/audiosup/Makefile
+++ b/usr/src/cmd/audio/audiotest/audiotest.xcl
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -19,18 +18,31 @@
 #
 # CDDL HEADER END
 #
-#
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-
-MODULE = audiosup.so
-MDBTGT = kvm
 
-MODSRCS = audiosup.c
-
-include ../../../../Makefile.cmd
-include ../../../../Makefile.cmd.64
-include ../../Makefile.sparcv9
-include ../../../Makefile.module
+#
+# separating newline
+#
+msgid "\n"
+#
+# getopt switches
+#
+msgid "l2457"
+#
+# file name
+#
+msgid "/dev/mixer"
+#
+# ioctls and system calls used with perror
+#
+msgid "malloc"
+msgid "SNDCTL_DSP_GETCAPS"
+msgid "SNDCTL_DSP_SPEED"
+msgid "SNDCTL_SYSINFO"
+msgid "SNDCTL_AUDIOINFO"
+#
+# argument to setlocale
+#
+msgid ""
diff --git a/usr/src/cmd/audio/audiotest/wavedata.c b/usr/src/cmd/audio/audiotest/wavedata.c
new file mode 100644
index 0000000000..94a05f51ea
--- /dev/null
+++ b/usr/src/cmd/audio/audiotest/wavedata.c
@@ -0,0 +1,194 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * Purpose: Test sounds for osstest
+ *
+ * Nodoc:
+ */
+
+#include <string.h>
+
+#include "wavedata.h"
+
+static int
+le_int(const unsigned char *p, int l)
+{
+	int i, val;
+
+	val = 0;
+
+	for (i = l - 1; i >= 0; i--) {
+		val = (val << 8) | p[i];
+	}
+
+	return (val);
+}
+
+int
+uncompress_wave(short *outbuf)
+{
+#define	WAVE_FORMAT_ADPCM		0x0002
+
+	int i, n, dataleft, x, l = sizeof (inbuf);
+	const unsigned char *hdr = inbuf;
+	typedef struct {
+		int coeff1, coeff2;
+	}
+	adpcm_coeff;
+
+	adpcm_coeff coeff[32];
+	static int AdaptionTable[] = { 230, 230, 230, 230, 307, 409, 512, 614,
+	    768, 614, 512, 409, 307, 230, 230, 230
+	};
+
+	unsigned char buf[4096];
+
+	int channels = 1;
+	int p = 12, outp = 0;
+	int nBlockAlign = 2048;
+	int wSamplesPerBlock = 2036, wNumCoeff = 7;
+	int nib;
+	int ppp;
+
+	/* filelen = le_int(&hdr[4], 4); */
+
+	while (p < l - 16 && memcmp(&hdr[p], "data", 4) != 0) {
+		n = le_int(&hdr[p + 4], 4);
+
+		if (memcmp(&hdr[p], "fmt ", 4) == 0) {
+
+			/* fmt = le_int(&hdr[p + 8], 2); */
+			channels = le_int(&hdr[p + 10], 2);
+			/* speed = le_int(&hdr[p + 12], 4); */
+			nBlockAlign = le_int(&hdr[p + 20], 2);
+			/* bytes_per_sample = le_int(&hdr[p + 20], 2); */
+
+			wSamplesPerBlock = le_int(&hdr[p + 26], 2);
+			wNumCoeff = le_int(&hdr[p + 28], 2);
+
+			x = p + 30;
+
+			for (i = 0; i < wNumCoeff; i++) {
+				coeff[i].coeff1 = (short)le_int(&hdr[x], 2);
+				x += 2;
+				coeff[i].coeff2 = (short)le_int(&hdr[x], 2);
+				x += 2;
+			}
+		}
+
+		p += n + 8;
+	}
+
+	if (p < l - 16 && memcmp(&hdr[p], "data", 4) == 0) {
+
+		dataleft = n = le_int(&hdr[p + 4], 4);
+		p += 8;
+
+/*
+ * Playback procedure
+ */
+#define	OUT_SAMPLE(s) {				\
+		if (s > 32767)			\
+			s = 32767;		\
+		else if (s < -32768)		\
+			s = -32768;		\
+		outbuf[outp++] = s;		\
+		n += 2;				\
+		}
+
+#define	GETNIBBLE					\
+		((nib == 0) ?				\
+		(buf[x + nib++] >> 4) & 0x0f : buf[x++ + --nib] & 0x0f)
+
+		outp = 0;
+
+		ppp = p;
+		while (dataleft > nBlockAlign) {
+			int predictor[2], delta[2], samp1[2], samp2[2];
+
+			int x = 0;
+
+			(void) memcpy(buf, &inbuf[ppp], nBlockAlign);
+			ppp += nBlockAlign;
+			dataleft -= nBlockAlign;
+
+			nib = 0;
+			n = 0;
+
+			for (i = 0; i < channels; i++) {
+				predictor[i] = buf[x];
+				x++;
+			}
+
+			for (i = 0; i < channels; i++) {
+				delta[i] = (short)le_int(&buf[x], 2);
+				x += 2;
+			}
+
+			for (i = 0; i < channels; i++) {
+				samp1[i] = (short)le_int(&buf[x], 2);
+				x += 2;
+				OUT_SAMPLE(samp1[i]);
+			}
+
+			for (i = 0; i < channels; i++) {
+				samp2[i] = (short)le_int(&buf[x], 2);
+				x += 2;
+				OUT_SAMPLE(samp2[i]);
+			}
+
+			while (n < (wSamplesPerBlock * 2 * channels))
+				for (i = 0; i < channels; i++) {
+					int pred, new, error_delta, i_delta;
+
+					pred = ((samp1[i] *
+					    coeff[predictor[i]].coeff1)
+					    + (samp2[i] *
+					    coeff[predictor[i]].coeff2)) / 256;
+					i_delta = error_delta = GETNIBBLE;
+
+					/* Convert to signed */
+					if (i_delta & 0x08)
+						i_delta -= 0x10;
+
+					new = pred + (delta[i] * i_delta);
+					OUT_SAMPLE(new);
+
+					delta[i] = delta[i] *
+					    AdaptionTable[error_delta] / 256;
+					if (delta[i] < 16)
+						delta[i] = 16;
+
+					samp2[i] = samp1[i];
+					samp1[i] = new;
+				}
+		}
+
+	}
+
+	return (outp * 2);
+}
diff --git a/usr/src/cmd/audio/audiotest/wavedata.h b/usr/src/cmd/audio/audiotest/wavedata.h
new file mode 100644
index 0000000000..5e98da8b70
--- /dev/null
+++ b/usr/src/cmd/audio/audiotest/wavedata.h
@@ -0,0 +1,42792 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	WAVEDATA_H
+#define	WAVEDATA_H
+
+static const unsigned char inbuf[] = {
+	0x52, 0x49, 0x46, 0x46, 0x26, 0x38, 0x05, 0x00, 0x57, 0x41, 0x56, 0x45,
+	0x66, 0x6d, 0x74, 0x20,
+	0x32, 0x00, 0x00, 0x00, 0x02, 0x00, 0x02, 0x00, 0x80, 0xbb, 0x00, 0x00,
+	0x9b, 0xbc, 0x00, 0x00,
+	0x00, 0x08, 0x04, 0x00, 0x20, 0x00, 0xf4, 0x07, 0x07, 0x00, 0x00, 0x01,
+	0x00, 0x00, 0x00, 0x02,
+	0x00, 0xff, 0x00, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x40, 0x00, 0xf0, 0x00,
+	0x00, 0x00, 0xcc, 0x01,
+	0x30, 0xff, 0x88, 0x01, 0x18, 0xff, 0x66, 0x61, 0x63, 0x74, 0x04, 0x00,
+	0x00, 0x00, 0x00, 0x30,
+	0x05, 0x00, 0x64, 0x61, 0x74, 0x61, 0xd4, 0x37, 0x05, 0x00, 0x06, 0x06,
+	0x10, 0x00, 0x10, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00,
+	0x00, 0x00, 0x01, 0x00,
+	0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xf0, 0x01, 0x10, 0x0f,
+	0xf0, 0x00, 0x01, 0x00,
+	0x00, 0x0f, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x10,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x10,
+	0x00, 0xf0, 0x00, 0x00, 0x00, 0x10, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0xf0, 0x00, 0x10, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0xf0, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xf0, 0x00,
+	0x11, 0x00, 0x0f, 0xf0,
+	0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x00, 0x01,
+	0x00, 0x0f, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x0f, 0xf0, 0x00, 0x00, 0x10, 0x00,
+	0x00, 0x0f, 0x00, 0x00,
+	0x00, 0x00, 0x0f, 0x10, 0x00, 0xf0, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x0f, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x10, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10,
+	0x00, 0x01, 0x00, 0x00,
+	0x00, 0x00, 0x01, 0x00, 0x01, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x01, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x01, 0x00, 0x0f, 0x00, 0x01, 0x00, 0x0f, 0x00, 0xf1, 0x00, 0x00,
+	0x00, 0xf1, 0x00, 0x00,
+	0x00, 0xf1, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0xf0, 0x00, 0x01, 0x00,
+	0x01, 0x00, 0x00, 0x00,
+	0xf0, 0x01, 0x11, 0x00, 0xf0, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x0f,
+	0x00, 0x00, 0x00, 0x01,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x10, 0x0f,
+	0xf0, 0x10, 0x00, 0x10,
+	0x00, 0x00, 0x00, 0x20, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x01, 0x00,
+	0x0f, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0xf0, 0x00, 0x00, 0x10, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x0f,
+	0x00, 0xf0, 0x00, 0x00, 0x00, 0x0f, 0xff, 0x00, 0x0f, 0x00, 0xf0, 0x00,
+	0x0f, 0x10, 0xf0, 0x00,
+	0x0f, 0x00, 0x10, 0x00, 0x00, 0xff, 0x00, 0x00, 0x00, 0x1f, 0x01, 0xf0,
+	0x0f, 0x0f, 0x00, 0x01,
+	0x10, 0x00, 0xff, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x01, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x0f, 0x00, 0x10, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x0f, 0x10, 0x00, 0xe0, 0x00, 0x10, 0x00, 0x00, 0xff, 0x00, 0x00, 0x11,
+	0x00, 0x00, 0x10, 0x00,
+	0x01, 0x00, 0x10, 0x00, 0x01, 0x10, 0x00, 0xf0, 0x10, 0x00, 0x01, 0xf0,
+	0x0f, 0x10, 0x01, 0x00,
+	0xf0, 0x0f, 0x00, 0x02, 0x00, 0x0f, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x01, 0x10, 0x0f,
+	0x00, 0x01, 0x00, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x01, 0x10, 0x00, 0x00,
+	0x00, 0x00, 0x01, 0x10, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x10, 0x00, 0xf0, 0x00,
+	0x10, 0x00, 0x0f, 0x01, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x0f, 0x00,
+	0xf0, 0x00, 0x10, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x00, 0x00, 0x00, 0x0f,
+	0x20, 0x00, 0xf0, 0x0f,
+	0x00, 0x20, 0x00, 0x00, 0x01, 0x0f, 0x00, 0x00, 0x00, 0x10, 0x01, 0x00,
+	0x00, 0x00, 0xf0, 0x10,
+	0x11, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x10, 0x00, 0xf0, 0xf0, 0x01, 0x10,
+	0x00, 0x00, 0x00, 0x00,
+	0xf0, 0x00, 0x00, 0x10, 0x00, 0xf0, 0xf0, 0x00, 0x00, 0x00, 0x01, 0x00,
+	0x00, 0xf0, 0x00, 0xf0,
+	0x01, 0x20, 0x00, 0xf0, 0xf0, 0x00, 0x00, 0x20, 0x00, 0xf0, 0xf0, 0x00,
+	0x00, 0x00, 0x00, 0xf0,
+	0x00, 0xf0, 0x00, 0x00, 0xf0, 0x00, 0x00, 0x00, 0xf0, 0xf0, 0x00, 0x20,
+	0xf0, 0xf0, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x02, 0x00, 0x0f, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x0f, 0x00, 0x00,
+	0x00, 0x10, 0x00, 0xf0, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, 0x10,
+	0x00, 0xf0, 0x10, 0x20,
+	0x00, 0xf0, 0x00, 0x00, 0x0f, 0x01, 0x01, 0x0f, 0xf0, 0xf0, 0x00, 0x00,
+	0x00, 0x01, 0xf0, 0x00,
+	0x00, 0x00, 0xe0, 0x10, 0x21, 0xf0, 0xef, 0x00, 0x00, 0x10, 0x00, 0x00,
+	0xf0, 0x01, 0x0f, 0x1f,
+	0x00, 0xf2, 0x00, 0x1e, 0x10, 0x01, 0x00, 0x0f, 0x0f, 0x10, 0x01, 0x00,
+	0x0f, 0x0f, 0xf0, 0x00,
+	0x11, 0x0f, 0xf0, 0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x00,
+	0x10, 0x00, 0x00, 0x1f,
+	0x00, 0x00, 0x00, 0x00, 0x1f, 0x00, 0x01, 0x00, 0x0f, 0x00, 0x00, 0x11,
+	0x00, 0xf0, 0x10, 0x0f,
+	0x00, 0x00, 0x02, 0x10, 0x0e, 0x00, 0x10, 0xf0, 0x00, 0x10, 0x20, 0xf0,
+	0x0f, 0xff, 0x10, 0x10,
+	0x10, 0x0f, 0x0f, 0x0f, 0x00, 0x01, 0x1f, 0x0f, 0x00, 0x00, 0x00, 0x00,
+	0x0f, 0x00, 0x01, 0x0f,
+	0x0f, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x00, 0xe0, 0x00, 0x21, 0x00,
+	0xf1, 0xf0, 0x00, 0x01,
+	0x00, 0x00, 0x01, 0xf1, 0x00, 0x00, 0x00, 0x00, 0x01, 0xf0, 0x00, 0x00,
+	0x00, 0x00, 0xef, 0x00,
+	0x10, 0x01, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0xe0,
+	0xf0, 0x10, 0x0f, 0x00,
+	0xf1, 0xf0, 0x00, 0x00, 0xf0, 0x00, 0x21, 0xf0, 0xf0, 0x00, 0x00, 0x20,
+	0x00, 0x00, 0xf0, 0x00,
+	0x10, 0x10, 0x00, 0x00, 0x01, 0x00, 0x10, 0x10, 0x00, 0x00, 0xf1, 0x1f,
+	0x30, 0x00, 0xe0, 0x1f,
+	0x10, 0x20, 0x02, 0x0f, 0x1f, 0x00, 0x01, 0x20, 0x0f, 0x00, 0x01, 0x00,
+	0x00, 0x0f, 0x10, 0x00,
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+	0x01, 0x00, 0x10, 0x00,
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+	0x21, 0x0f, 0xff, 0x00,
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+	0x02, 0xf1, 0xd1, 0x00,
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+	0xf0, 0xf0, 0xff, 0x00,
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+	0x01, 0x10, 0x01, 0xf1,
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+	0x10, 0x11, 0x00, 0x00,
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+	0xe0, 0xff, 0x10, 0x22,
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+	0xf0, 0xf0, 0x01, 0x10,
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+	0x10, 0xff, 0xe0, 0x00,
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+	0x00, 0x10, 0x11, 0x00,
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+	0xf0, 0x1f, 0x20, 0xe1,
+	0xe0, 0x00, 0x11, 0x30, 0xf0, 0xff, 0x0f, 0x0f, 0x1e, 0x11, 0x01, 0x0f,
+	0x0d, 0x00, 0x0f, 0x10,
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+	0x0f, 0x00, 0x00, 0x01,
+	0x30, 0xe0, 0xe0, 0xf0, 0x10, 0x20, 0x00, 0xf0, 0xe1, 0x1f, 0x10, 0x02,
+	0x00, 0x00, 0x20, 0x00,
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+	0x00, 0x00, 0x2e, 0x21,
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+	0x01, 0x21, 0x02, 0x00,
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+	0xe1, 0x02, 0x00, 0x1e,
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+	0x1f, 0x0d, 0xf0, 0xe2,
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+	0x00, 0x00, 0xf1, 0x10,
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+	0x01, 0x10, 0x10, 0x00,
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+	0xee, 0x00, 0x00, 0x10,
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+	0x0f, 0x10, 0x1f, 0x10,
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+	0x0f, 0x00, 0x01, 0x00,
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+	0x00, 0x01, 0xff, 0x00,
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+	0x11, 0x00, 0x0f, 0x00,
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+	0x0e, 0x00, 0x13, 0x00,
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+	0x01, 0x01, 0x1f, 0x0e,
+	0xf1, 0x02, 0x0f, 0x2e, 0x00, 0xf0, 0x03, 0x00, 0x0f, 0x10, 0x10, 0x00,
+	0xef, 0x0f, 0x01, 0x01,
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+	0x00, 0xdf, 0x00, 0x01,
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+	0x0d, 0x00, 0xef, 0xf0,
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+	0x10, 0x00, 0x02, 0x00,
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+	0x10, 0x00, 0x10, 0x00,
+	0xd4, 0xff, 0xfc, 0xff, 0xf1, 0xff, 0xf6, 0xff, 0x0e, 0x0e, 0x01, 0x10,
+	0x0f, 0xee, 0x00, 0x10,
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+	0x20, 0x0f, 0xdf, 0x00,
+	0x11, 0x2f, 0x0e, 0xf1, 0xf1, 0x0f, 0x10, 0x10, 0x0e, 0xef, 0x1f, 0x2f,
+	0x01, 0x01, 0x2e, 0x1c,
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+	0x10, 0x20, 0x12, 0xf0,
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+	0x01, 0x11, 0x00, 0x02,
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+	0xe0, 0x0f, 0x11, 0x00,
+	0x2e, 0x1d, 0x00, 0x03, 0x10, 0x2f, 0x0e, 0x00, 0x10, 0x00, 0x00, 0x01,
+	0x01, 0x10, 0x0e, 0xf0,
+	0x11, 0x13, 0xf0, 0xd0, 0x00, 0x2f, 0x00, 0xd1, 0x00, 0x00, 0xf0, 0xdf,
+	0x0e, 0x2f, 0x02, 0xf3,
+	0xdf, 0x0d, 0x42, 0x03, 0xd0, 0x00, 0x10, 0x32, 0x02, 0xd0, 0x00, 0x1f,
+	0x11, 0x13, 0xf0, 0xf0,
+	0x0e, 0x30, 0x01, 0xd3, 0x00, 0x1f, 0x2f, 0x00, 0xe0, 0x03, 0x00, 0x1f,
+	0x0e, 0x00, 0x00, 0x02,
+	0x00, 0xff, 0x00, 0x00, 0x00, 0x10, 0x10, 0xe0, 0xf0, 0x00, 0x00, 0xff,
+	0x00, 0x00, 0x00, 0x10,
+	0xf0, 0x00, 0xff, 0x01, 0x00, 0x10, 0x22, 0xe0, 0xed, 0x01, 0x12, 0x0e,
+	0xef, 0x02, 0x01, 0x0f,
+	0x1f, 0xf0, 0xd1, 0xd0, 0x1f, 0x40, 0x00, 0xff, 0xcf, 0xff, 0x1f, 0x2e,
+	0x00, 0xd1, 0xf0, 0x0f,
+	0x0d, 0x00, 0x03, 0x01, 0x00, 0x0e, 0xf1, 0x03, 0x20, 0xe0, 0xe1, 0x10,
+	0x30, 0xf0, 0xd0, 0x00,
+	0x22, 0x01, 0xfe, 0x0d, 0x10, 0x04, 0xd0, 0x00, 0x2d, 0x00, 0xe2, 0x0f,
+	0x00, 0x13, 0x00, 0xff,
+	0xd0, 0x10, 0x3f, 0x2f, 0xf0, 0xd2, 0x11, 0x3f, 0x1b, 0xf0, 0x00, 0x32,
+	0x1f, 0x0f, 0x00, 0x30,
+	0x0e, 0x00, 0x21, 0x12, 0x01, 0x0f, 0x0e, 0xf1, 0x12, 0x11, 0x0f, 0x10,
+	0x11, 0xf3, 0xfe, 0x0d,
+	0x20, 0x23, 0xe0, 0xd0, 0x1d, 0x31, 0xe2, 0xef, 0x10, 0x12, 0xe1, 0xe0,
+	0x1e, 0x21, 0xf4, 0xe1,
+	0x1e, 0x0e, 0xf1, 0xf2, 0x01, 0x2e, 0xe0, 0xd0, 0x00, 0x3f, 0x0e, 0xd2,
+	0xe1, 0x0f, 0x3d, 0x00,
+	0xff, 0xd1, 0x01, 0x1f, 0x10, 0x0f, 0xef, 0x00, 0x31, 0x02, 0xf0, 0xff,
+	0x10, 0x02, 0xff, 0x00,
+	0x10, 0x01, 0xe3, 0x00, 0x0f, 0x0d, 0x20, 0x03, 0x02, 0x0f, 0x1d, 0x00,
+	0xf2, 0xe1, 0x1f, 0x3d,
+	0x00, 0xc2, 0xf0, 0xf0, 0x00, 0x1e, 0x00, 0xd2, 0x12, 0x1d, 0x0e, 0xd0,
+	0x0f, 0x10, 0x1e, 0x0d,
+	0xfe, 0x01, 0x0f, 0x1d, 0x0f, 0xe1, 0x20, 0x2c, 0x1f, 0x00, 0x13, 0x0f,
+	0x0c, 0x00, 0x31, 0x20,
+	0x0e, 0xf0, 0xf2, 0x01, 0x0e, 0x1d, 0x20, 0xe3, 0xd2, 0x01, 0x00, 0x10,
+	0x00, 0xee, 0x01, 0x21,
+	0x01, 0xd2, 0x0f, 0x2c, 0x00, 0x00, 0x04, 0x01, 0x0f, 0xfe, 0x01, 0x13,
+	0x30, 0x00, 0xe1, 0x10,
+	0x10, 0x11, 0x00, 0xf0, 0x20, 0x31, 0x01, 0xff, 0x0e, 0x10, 0x44, 0x20,
+	0x0f, 0x00, 0x01, 0x10,
+	0x30, 0x10, 0x01, 0x00, 0xdf, 0x00, 0x41, 0x01, 0xd0, 0xd0, 0x12, 0x30,
+	0x0f, 0xce, 0x03, 0x13,
+	0x50, 0xfe, 0xc0, 0x00, 0x20, 0x23, 0xd2, 0xf1, 0x00, 0x00, 0x02, 0x12,
+	0x22, 0xf0, 0xf0, 0x01,
+	0x40, 0x01, 0xd1, 0x00, 0x4e, 0x00, 0xf0, 0xe0, 0x00, 0x10, 0x20, 0x02,
+	0xe1, 0x0f, 0x1f, 0x11,
+	0x02, 0xf0, 0x0f, 0x3f, 0xf1, 0xc2, 0x00, 0x5f, 0x00, 0xff, 0xf0, 0x10,
+	0x40, 0xff, 0xc0, 0x01,
+	0x50, 0x00, 0xf0, 0xdf, 0x10, 0x12, 0x11, 0xf0, 0xdf, 0x00, 0x10, 0x22,
+	0xe0, 0xee, 0x01, 0x02,
+	0x00, 0x0d, 0x00, 0xf2, 0xf0, 0x20, 0x1d, 0xf2, 0xf2, 0x0f, 0x2d, 0x02,
+	0x02, 0x0e, 0xee, 0x02,
+	0x12, 0xfe, 0xed, 0x21, 0x14, 0xe0, 0xec, 0x00, 0x21, 0x10, 0xf0, 0xd0,
+	0xe0, 0x0f, 0x2e, 0x1f,
+	0xf2, 0xe2, 0x0f, 0x1d, 0xf0, 0x01, 0x32, 0x10, 0x0e, 0xe0, 0x01, 0x01,
+	0x0e, 0x1f, 0x20, 0xe2,
+	0xd0, 0x0f, 0x2d, 0x1f, 0x02, 0xd1, 0x10, 0x2f, 0x00, 0xd0, 0x0e, 0x42,
+	0x11, 0xff, 0xc0, 0x10,
+	0x10, 0x2e, 0xf0, 0xf0, 0xf1, 0x10, 0x01, 0x00, 0xdd, 0x0f, 0x12, 0x03,
+	0xfe, 0x0c, 0x00, 0x00,
+	0xdf, 0xf1, 0x10, 0x3e, 0xf0, 0xc0, 0x00, 0x1f, 0x10, 0xf3, 0x02, 0x1e,
+	0x20, 0x01, 0x03, 0x22,
+	0x02, 0x00, 0x0f, 0x30, 0x10, 0xe2, 0xe2, 0x11, 0x30, 0xfc, 0xd0, 0xe3,
+	0x10, 0x3f, 0x10, 0xdf,
+	0xdf, 0x00, 0x00, 0xf1, 0x00, 0x20, 0xd0, 0xdf, 0xf1, 0x21, 0x12, 0xf0,
+	0xd0, 0x0f, 0x00, 0x00,
+	0x00, 0x02, 0xe1, 0x0f, 0x0c, 0x00, 0xe2, 0x10, 0x01, 0xd1, 0x00, 0xf1,
+	0x01, 0xff, 0xd0, 0x04,
+	0x02, 0x0e, 0xee, 0xf1, 0x43, 0x02, 0xef, 0xeb, 0x20, 0x40, 0x01, 0xd0,
+	0x10, 0x1f, 0x0f, 0x0e,
+	0x10, 0x10, 0xe5, 0xfe, 0x2d, 0x21, 0xf2, 0xfe, 0x1e, 0x21, 0x10, 0xfd,
+	0x0f, 0x10, 0x00, 0x1d,
+	0x1e, 0x0f, 0x00, 0x00, 0x20, 0xef, 0xff, 0x40, 0x30, 0x00, 0xdf, 0x0f,
+	0x10, 0x63, 0x1e, 0xfd,
+	0x00, 0x00, 0x0f, 0x0f, 0x20, 0x40, 0xfc, 0xef, 0xf1, 0x24, 0x2e, 0x0d,
+	0x00, 0x14, 0x00, 0x0d,
+	0xd2, 0x11, 0x4d, 0x0d, 0xe1, 0x05, 0x3f, 0xfc, 0xe0, 0x04, 0x30, 0x0d,
+	0x0d, 0xd1, 0x02, 0x20,
+	0x1e, 0xef, 0xdf, 0xfe, 0x1f, 0x30, 0xf2, 0xc0, 0x00, 0x5e, 0x00, 0x02,
+	0xd2, 0x30, 0x7d, 0x02,
+	0xf5, 0x01, 0x30, 0x0e, 0x02, 0x16, 0x11, 0xff, 0x00, 0xd3, 0x05, 0x2f,
+	0x10, 0xe1, 0xd1, 0x00,
+	0x10, 0x13, 0x02, 0x0e, 0xdd, 0x00, 0x14, 0x10, 0x1e, 0xff, 0xf2, 0x00,
+	0x3e, 0x1e, 0xe1, 0xe5,
+	0x10, 0x4b, 0xf0, 0xb1, 0x01, 0x20, 0x0f, 0x0e, 0xc1, 0x00, 0x00, 0x2b,
+	0x00, 0x00, 0x30, 0x0f,
+	0x0f, 0x20, 0x50, 0x1e, 0x00, 0x21, 0x04, 0x3f, 0x4b, 0x11, 0xf3, 0x00,
+	0x3e, 0x0f, 0x13, 0x13,
+	0xe0, 0x00, 0x30, 0x1f, 0xd2, 0xd4, 0x2e, 0x2e, 0x02, 0xa5, 0x00, 0x1e,
+	0x20, 0xc5, 0xe1, 0x02,
+	0x10, 0x21, 0xf2, 0xd0, 0xf0, 0x04, 0xf0, 0x10, 0xef, 0xc0, 0xf0, 0x1f,
+	0x10, 0xc5, 0xc0, 0x0c,
+	0x2f, 0x41, 0xc3, 0xb0, 0x00, 0x10, 0x20, 0x02, 0xe1, 0xe2, 0xe1, 0x0f,
+	0x00, 0x13, 0xe2, 0x0f,
+	0x0e, 0xf0, 0x00, 0xe1, 0xf1, 0x00, 0x2f, 0x0e, 0xe0, 0x00, 0xf1, 0xfe,
+	0x0c, 0x00, 0x01, 0xe3,
+	0x0c, 0x0b, 0x00, 0x00, 0xe3, 0x0b, 0xff, 0x00, 0x30, 0x11, 0xff, 0xec,
+	0x1f, 0x10, 0x11, 0x00,
+	0xfc, 0x20, 0x3e, 0xfe, 0xc1, 0x22, 0x4e, 0x0e, 0xc2, 0x00, 0x2e, 0x2c,
+	0x02, 0xf4, 0x1f, 0x2b,
+	0x00, 0xe0, 0x12, 0x20, 0x10, 0x0e, 0x0f, 0x00, 0x0f, 0x1d, 0x20, 0x33,
+	0x00, 0xda, 0x00, 0x2f,
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+	0xfc, 0x00, 0x12, 0x00,
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+	0x21, 0x3e, 0x3e, 0xe4,
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+	0x1f, 0xe0, 0xd3, 0x30,
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+	0x20, 0x2f, 0x2e, 0x02,
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+	0xec, 0x20, 0x24, 0x32,
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+	0x32, 0xb0, 0xdc, 0x0f,
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+	0x3e, 0x20, 0xf0, 0xce,
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+	0x0d, 0xce, 0x02, 0x50,
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+	0x3e, 0x00, 0x0e, 0x0e,
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+	0x02, 0x45, 0xe0, 0xcd,
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+	0x10, 0x01, 0xc0, 0x0c,
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+	0xe0, 0x23, 0x3f, 0xea,
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+	0x00, 0xd2, 0xe0, 0x21,
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+	0x03, 0x00, 0x00, 0xe2,
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+	0xfe, 0x51, 0x17, 0xe2,
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+	0x24, 0x21, 0xf4, 0xf1,
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+	0xd1, 0xe4, 0xbf, 0xee,
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+	0xf2, 0xe1, 0x22, 0x00,
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+	0x00, 0xf0, 0xcd, 0x00,
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+	0xfb, 0xbf, 0x02, 0x00,
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+	0x01, 0xf3, 0x0b, 0x7e,
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+	0x2f, 0x2d, 0x30, 0x13,
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+	0x42, 0x00, 0x00, 0x02,
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+	0xf6, 0x0f, 0x3c, 0xf0,
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+	0xeb, 0x60, 0x34, 0xe0,
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+	0xc0, 0x00, 0x12, 0x00,
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+	0x0f, 0xd2, 0x00, 0x4c,
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+	0x03, 0x06, 0x1f, 0xba,
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+	0x1f, 0x6c, 0xe1, 0x02,
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+	0x02, 0x0f, 0x0c, 0x10,
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+	0x22, 0x0d, 0x0f, 0xdf,
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+	0xc0, 0x2f, 0x7f, 0x00,
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+	0xf4, 0x01, 0x0d, 0x6f,
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+	0xf2, 0x14, 0x3f, 0xfb,
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+	0x00, 0xd1, 0xdd, 0x2d,
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+	0x30, 0xef, 0xae, 0x00,
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+	0xbf, 0x02, 0x51, 0xec,
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+	0xb1, 0xcd, 0xf1, 0x14,
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+	0xd2, 0xe0, 0xfd, 0x0f,
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+	0xf0, 0xe0, 0x00, 0x08,
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+	0x51, 0x04, 0xe0, 0x10,
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+	0x22, 0x1d, 0x58, 0x11,
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+	0x0f, 0x1f, 0x5f, 0x0f,
+	0x0e, 0x3f, 0x4e, 0xfe, 0x12, 0x72, 0x2d, 0xc0, 0xf5, 0x33, 0x3b, 0xff,
+	0xe6, 0x02, 0x3f, 0xff,
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+	0x17, 0x12, 0x2e, 0xce,
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+	0xab, 0x0e, 0x22, 0x41,
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+	0xcd, 0xbd, 0x21, 0x34,
+	0xbf, 0xd1, 0x02, 0x6c, 0x0f, 0xa0, 0x01, 0x0f, 0x1d, 0xef, 0xe0, 0x01,
+	0x1f, 0x0c, 0xed, 0xfd,
+	0x13, 0x07, 0x9f, 0x2e, 0x3f, 0xde, 0xd5, 0x53, 0x1b, 0xf0, 0xc3, 0x32,
+	0x3e, 0xf0, 0xf3, 0x10,
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+	0xc4, 0xfe, 0x4e, 0xe1,
+	0xa5, 0x0f, 0x0f, 0x00, 0xf2, 0x0f, 0xef, 0xa3, 0x2f, 0x4c, 0xe3, 0xc2,
+	0x09, 0x2f, 0x00, 0xe2,
+	0x00, 0xff, 0xee, 0x0a, 0x00, 0xd0, 0xd2, 0x3c, 0x0d, 0x01, 0xe1, 0xbc,
+	0xed, 0x3e, 0x72, 0xf2,
+	0xbc, 0x0c, 0x30, 0x13, 0x00, 0xf9, 0x00, 0x00, 0x10, 0x2f, 0x00, 0xd5,
+	0xc0, 0x5c, 0x40, 0xe0,
+	0xb4, 0x01, 0x1e, 0x5a, 0x10, 0xd3, 0xe1, 0xff, 0x3e, 0x6f, 0xf1, 0xc5,
+	0x10, 0x3e, 0x0f, 0xd0,
+	0x13, 0x11, 0x2e, 0x1c, 0xe0, 0xf0, 0x72, 0x14, 0x04, 0xfc, 0x4c, 0x21,
+	0x06, 0x00, 0x71, 0x01,
+	0xf0, 0x00, 0x30, 0x25, 0x20, 0xf0, 0xee, 0x52, 0x36, 0xf0, 0xfd, 0x73,
+	0x03, 0xff, 0xff, 0x42,
+	0x22, 0xfd, 0x00, 0x17, 0x02, 0x00, 0x5c, 0x00, 0xd2, 0x31, 0x02, 0xe0,
+	0xfc, 0x5f, 0x20, 0xc0,
+	0xf3, 0x0f, 0x19, 0xf0, 0xb3, 0xf1, 0x0e, 0x30, 0xf4, 0x90, 0x1a, 0x01,
+	0xd6, 0x00, 0x4c, 0xc0,
+	0xd3, 0x32, 0xfe, 0xcd, 0x01, 0x03, 0xf2, 0x2c, 0x3d, 0xb2, 0xe7, 0x00,
+	0x3c, 0x31, 0xd1, 0xb0,
+	0x1d, 0x00, 0xe5, 0xe0, 0xfc, 0x6f, 0x01, 0xa0, 0x0e, 0x10, 0x30, 0xbd,
+	0x00, 0x10, 0x1d, 0xcd,
+	0x00, 0x40, 0xe2, 0xcd, 0x0f, 0x55, 0x0f, 0xf9, 0xf0, 0x00, 0x06, 0x02,
+	0x11, 0x00, 0x18, 0x00,
+	0x8e, 0xff, 0xe0, 0xff, 0xde, 0xff, 0x74, 0xff, 0x34, 0x15, 0x80, 0xff,
+	0x20, 0x30, 0xdd, 0xdc,
+	0x30, 0x56, 0x03, 0x10, 0x0c, 0xed, 0xf0, 0x30, 0x4c, 0x0b, 0xc0, 0xe2,
+	0x20, 0x7f, 0x00, 0xf1,
+	0xeb, 0x4c, 0x20, 0xe3, 0x23, 0x73, 0xf1, 0xef, 0x4d, 0x4d, 0x0c, 0xe0,
+	0x17, 0x03, 0xf1, 0x30,
+	0x30, 0x9e, 0xff, 0x40, 0x02, 0xc6, 0xf3, 0x31, 0x0e, 0x0d, 0xd0, 0xf0,
+	0xff, 0x30, 0x05, 0xb1,
+	0xfc, 0x1a, 0x1d, 0xbf, 0xe3, 0x16, 0x41, 0x0f, 0x1e, 0x2f, 0x00, 0xc0,
+	0x0f, 0x60, 0x42, 0x01,
+	0xb4, 0x03, 0x2e, 0x08, 0x0e, 0xf0, 0xe1, 0x20, 0x11, 0xd5, 0xd3, 0xf0,
+	0x0f, 0x10, 0xfe, 0x28,
+	0x2e, 0xd0, 0xe0, 0x30, 0x32, 0x33, 0x54, 0xf3, 0x91, 0x11, 0x63, 0x04,
+	0xf0, 0xed, 0x1b, 0x20,
+	0x03, 0xf4, 0xb2, 0x00, 0x3d, 0x20, 0xe5, 0xe1, 0x0f, 0x0e, 0x6c, 0x0f,
+	0x06, 0x03, 0xf0, 0x0c,
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+	0xd0, 0x40, 0x30, 0xd1,
+	0xb1, 0xd3, 0x07, 0x32, 0xd0, 0x80, 0x03, 0x01, 0xe0, 0xe3, 0x05, 0xd3,
+	0x02, 0x30, 0x83, 0xe6,
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+	0xc0, 0xf9, 0x4d, 0xde,
+	0x80, 0x01, 0x1e, 0xe8, 0x0e, 0x12, 0xd1, 0xdf, 0x00, 0x31, 0x20, 0xfe,
+	0x0f, 0xe1, 0xe5, 0x55,
+	0x44, 0x03, 0x01, 0x5b, 0xea, 0x0e, 0x21, 0x23, 0x01, 0x00, 0x11, 0xc3,
+	0x00, 0x5b, 0x0d, 0xdd,
+	0xdf, 0x7f, 0x2f, 0x02, 0xd4, 0x00, 0x3e, 0x7d, 0x18, 0x0e, 0x00, 0x11,
+	0x60, 0x10, 0x10, 0x1e,
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+	0x10, 0x0c, 0xfd, 0x46,
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+	0xfc, 0xd2, 0xf5, 0xb0,
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+	0x1d, 0x52, 0xe4, 0xf0,
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+	0xe0, 0x33, 0x75, 0xf2,
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+	0xed, 0x3e, 0x2f, 0xc1,
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+	0x30, 0x90, 0x00, 0x2f,
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+	0xa1, 0x01, 0x20, 0xf0,
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+	0xec, 0xeb, 0x2f, 0x3f,
+	0xa0, 0x03, 0x43, 0x0f, 0xfc, 0x2c, 0x3b, 0xfc, 0x0e, 0x70, 0x10, 0xcf,
+	0x50, 0x21, 0xff, 0xfe,
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+	0xd2, 0x70, 0x4e, 0xde,
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+	0x01, 0x01, 0xff, 0x11,
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+	0x04, 0x00, 0x0f, 0x10,
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+	0xff, 0xa0, 0xf9, 0x2e,
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+	0x0e, 0xee, 0x2e, 0x61,
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+	0xe1, 0x02, 0x00, 0xcf,
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+	0x31, 0x1d, 0x09, 0x60,
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+	0x1e, 0x0f, 0x10, 0x20,
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+	0x00, 0x01, 0x0f, 0x0f,
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+	0x03, 0xd1, 0xc0, 0xc0,
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+	0x11, 0x8b, 0xdc, 0x11,
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+	0x00, 0x24, 0x00, 0xff,
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+	0xdd, 0xed, 0x32, 0x53,
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+	0x01, 0xac, 0xff, 0x00,
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+	0x65, 0x00, 0x00, 0x0f,
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+	0xf5, 0x47, 0x14, 0xdf,
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+	0x54, 0xec, 0xcd, 0x0f,
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+	0x32, 0x31, 0x00, 0xfd,
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+	0xfe, 0x98, 0xfe, 0x21,
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+	0xdc, 0xfe, 0x33, 0x27,
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+	0xb0, 0x01, 0xd0, 0xad,
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+	0x00, 0xdd, 0xdf, 0xe0,
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+	0x23, 0xfe, 0xdc, 0x12,
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+	0x32, 0x00, 0xc8, 0xef,
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+	0x30, 0x51, 0x21, 0x00,
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+	0xed, 0xfe, 0x0e, 0x42,
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+	0xdf, 0x9a, 0x00, 0x57,
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+	0x12, 0xcf, 0xef, 0xff,
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+	0xe0, 0x03, 0xbe, 0xb9,
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+	0x57, 0x11, 0xcf, 0xde,
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+	0x11, 0x00, 0xfd, 0xbb,
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+	0x76, 0x10, 0xee, 0xff,
+	0xff, 0xff, 0x00, 0xcc, 0xff, 0x00, 0xcf, 0x0f, 0x00, 0x2e, 0x0b, 0x2f,
+	0x31, 0x01, 0xfe, 0xaa,
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+	0x03, 0x06, 0x76, 0x01,
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+	0x46, 0x22, 0xff, 0xee,
+	0x30, 0x52, 0x00, 0xff, 0x23, 0x47, 0xf0, 0xac, 0x00, 0x21, 0x00, 0xbe,
+	0x01, 0x02, 0x00, 0xf0,
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+	0x00, 0x21, 0x00, 0x00,
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+	0x0e, 0xfa, 0x0f, 0xfe,
+	0x0d, 0x72, 0x21, 0x0e, 0x33, 0x44, 0xfc, 0xea, 0x00, 0x41, 0x01, 0xff,
+	0x0f, 0xde, 0xce, 0xfe,
+	0xef, 0xf0, 0x02, 0xfd, 0xb9, 0xdd, 0x52, 0x44, 0xec, 0xee, 0x13, 0x22,
+	0x00, 0x0e, 0xfd, 0xfe,
+	0x21, 0xf0, 0x88, 0xee, 0x02, 0x56, 0x21, 0xff, 0xff, 0x02, 0x01, 0xb0,
+	0x14, 0x23, 0x00, 0x01,
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+	0x1f, 0x0e, 0x63, 0x45,
+	0x0d, 0xff, 0x13, 0x02, 0x0f, 0xfe, 0xde, 0xb9, 0x00, 0x47, 0x21, 0xe0,
+	0xbe, 0xf0, 0x31, 0x31,
+	0x32, 0x55, 0x00, 0x00, 0x11, 0x00, 0xf0, 0x00, 0x55, 0x22, 0xff, 0xdc,
+	0x00, 0x41, 0xcc, 0xdc,
+	0x40, 0x53, 0x00, 0xec, 0xcd, 0xfd, 0x11, 0x20, 0xfb, 0x00, 0x20, 0x00,
+	0x0f, 0x30, 0x0d, 0x0a,
+	0x70, 0x43, 0x0e, 0x0e, 0x0f, 0xee, 0x11, 0x20, 0xc8, 0xef, 0x21, 0xe0,
+	0x8d, 0xfe, 0xee, 0x00,
+	0x20, 0x1e, 0x01, 0x00, 0xde, 0xda, 0x0f, 0x42, 0x32, 0x0f, 0xdc, 0xac,
+	0xff, 0x00, 0x21, 0x55,
+	0xff, 0xcd, 0x00, 0x04, 0x01, 0x13, 0x24, 0xf0, 0x03, 0xf6, 0x80, 0xf0,
+	0xf1, 0x03, 0x03, 0xf2,
+	0xcf, 0xde, 0xf0, 0x37, 0x42, 0x00, 0xdf, 0xff, 0x00, 0x43, 0x42, 0x22,
+	0xfe, 0xd9, 0xf0, 0x23,
+	0x64, 0x10, 0xdd, 0xbc, 0xef, 0xe0, 0x02, 0x57, 0x22, 0xff, 0x0f, 0x01,
+	0x32, 0xff, 0xef, 0x45,
+	0x54, 0x00, 0xf0, 0xff, 0xf0, 0x00, 0x43, 0x10, 0xfd, 0xfd, 0x00, 0x0f,
+	0x0d, 0x52, 0x13, 0xfd,
+	0xca, 0xda, 0xcc, 0x10, 0x32, 0xee, 0x0e, 0x40, 0x0f, 0xdb, 0x41, 0x61,
+	0x0e, 0x10, 0x30, 0x0f,
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+	0x10, 0x1f, 0xea, 0xef,
+	0x11, 0x22, 0x00, 0xcd, 0x0e, 0x20, 0x0e, 0x8a, 0xee, 0x12, 0x00, 0xfd,
+	0x32, 0x75, 0x0f, 0xef,
+	0x13, 0x16, 0x01, 0x01, 0xf0, 0xdf, 0x14, 0x45, 0xc0, 0xaf, 0x01, 0x13,
+	0x00, 0xdf, 0xe0, 0x00,
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+	0x00, 0xf0, 0xef, 0x00,
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+	0xf1, 0x01, 0x00, 0x22,
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+	0xef, 0xfe, 0x50, 0x0f,
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+	0x30, 0x10, 0x42, 0x00,
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+	0xed, 0x30, 0x3f, 0x0b,
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+	0x0c, 0xeb, 0x0e, 0x3f,
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+	0x0f, 0xba, 0xee, 0xac,
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+	0x34, 0x13, 0xd0, 0x90,
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+	0xe2, 0xc2, 0x04, 0x02,
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+	0xdf, 0x27, 0x01, 0xa1,
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+	0x47, 0x11, 0xf0, 0x00,
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+	0x1f, 0x00, 0x00, 0x2f,
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+	0x32, 0x10, 0xdd, 0xfe,
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+	0x0e, 0xfc, 0x1f, 0x0d,
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+	0x03, 0xe0, 0xbe, 0xd0,
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+	0xf0, 0xf1, 0xf1, 0xe0,
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+	0x8f, 0xff, 0x14, 0x11,
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+	0x03, 0x34, 0xff, 0xbf,
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+	0xfb, 0x00, 0x50, 0x31,
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+	0x10, 0x71, 0x0d, 0x08,
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+	0x1d, 0x1c, 0x60, 0x0d,
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+	0x8d, 0x05, 0x07, 0x00,
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+	0xe1, 0xc0, 0xf0, 0x62,
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+	0x03, 0xe1, 0xf0, 0x21,
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+	0x75, 0xef, 0xcc, 0x00,
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+	0x2e, 0x1d, 0x0c, 0x20,
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+	0xc8, 0x00, 0xbc, 0x00,
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+	0xef, 0x01, 0x72, 0x09,
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+	0x45, 0xff, 0xdd, 0x01,
+	0x01, 0x00, 0x01, 0x00, 0xcb, 0x12, 0x54, 0xff, 0xee, 0x00, 0x14, 0xff,
+	0xee, 0x32, 0x76, 0xef,
+	0xcd, 0x11, 0x10, 0x10, 0x12, 0xf0, 0xee, 0xde, 0x34, 0x52, 0xed, 0x00,
+	0x02, 0xce, 0x22, 0x02,
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+	0xf0, 0x07, 0x34, 0xfe,
+	0x31, 0x12, 0x9a, 0xff, 0xf1, 0x13, 0x2e, 0xda, 0x31, 0x74, 0xef, 0xdd,
+	0x11, 0x43, 0xec, 0x00,
+	0x54, 0xff, 0xda, 0x22, 0x10, 0x00, 0x0f, 0x0f, 0xcf, 0x42, 0x12, 0xfe,
+	0x0e, 0x30, 0xce, 0xed,
+	0x67, 0x12, 0x0e, 0xde, 0x00, 0x00, 0x01, 0xff, 0xed, 0x10, 0x22, 0x0f,
+	0xc8, 0xef, 0x51, 0x32,
+	0x0f, 0x0d, 0x30, 0x31, 0xbf, 0xdb, 0x13, 0x43, 0x00, 0xef, 0xb9, 0x11,
+	0x12, 0xf0, 0xf0, 0xe0,
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+	0x0f, 0xe0, 0x22, 0xc0,
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+	0xae, 0x14, 0x46, 0xed,
+	0xfe, 0x24, 0x00, 0xce, 0x30, 0x67, 0x00, 0xde, 0x00, 0x12, 0x21, 0xdf,
+	0xf0, 0x20, 0x10, 0xff,
+	0xe0, 0x01, 0xdf, 0x0d, 0x72, 0x33, 0xdd, 0xce, 0xfe, 0x37, 0x55, 0xfc,
+	0xef, 0x00, 0x1f, 0x31,
+	0x02, 0xac, 0x01, 0x42, 0xfb, 0x00, 0x00, 0xfe, 0x01, 0x34, 0x0d, 0x10,
+	0xfe, 0xce, 0x67, 0x02,
+	0xba, 0x00, 0x21, 0x01, 0xff, 0xdd, 0x01, 0x41, 0x00, 0xff, 0xeb, 0x41,
+	0x44, 0xcd, 0x0e, 0x33,
+	0x00, 0xcb, 0x11, 0x32, 0xf0, 0xd0, 0x10, 0x0f, 0xde, 0x22, 0xc0, 0xfe,
+	0x66, 0x0f, 0xcf, 0x13,
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+	0x21, 0x0f, 0x0f, 0x10,
+	0xde, 0xdd, 0x57, 0x12, 0xce, 0x00, 0x0f, 0x0e, 0x02, 0xe3, 0xe0, 0x10,
+	0x00, 0x0e, 0x74, 0x43,
+	0xed, 0xf1, 0x17, 0x15, 0xff, 0x0f, 0x0f, 0x00, 0x23, 0x22, 0xbc, 0xfe,
+	0x44, 0xf0, 0xbe, 0x10,
+	0x30, 0xf0, 0xf1, 0x04, 0xfc, 0x0e, 0x74, 0x00, 0x0f, 0x00, 0xc9, 0x00,
+	0x54, 0x00, 0xfd, 0xf0,
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+	0x10, 0x12, 0x10, 0xe1,
+	0xff, 0x0c, 0xee, 0x22, 0x23, 0x9a, 0x00, 0x40, 0x3f, 0x1f, 0xef, 0xf0,
+	0xe2, 0xfe, 0x3f, 0x54,
+	0xff, 0xee, 0xf0, 0xcf, 0x20, 0x31, 0xbb, 0xe1, 0x16, 0x00, 0xf0, 0xe0,
+	0x42, 0x02, 0xcc, 0x22,
+	0x13, 0x8c, 0x01, 0x17, 0x1f, 0xed, 0x23, 0x25, 0xbd, 0xfe, 0x21, 0x22,
+	0x00, 0xe0, 0x00, 0x01,
+	0xcf, 0xef, 0x23, 0x00, 0xcb, 0xf0, 0xf2, 0x33, 0x20, 0xf0, 0x10, 0xeb,
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+	0x71, 0x36, 0xa0, 0x03,
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+	0x30, 0x3f, 0xcc, 0xff,
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+	0xe0, 0x00, 0x10, 0x4f,
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+	0x43, 0x46, 0x03, 0x14,
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+	0xfe, 0x0e, 0x2d, 0x30,
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+	0x01, 0x21, 0x2f, 0x01,
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+	0x2e, 0x20, 0x47, 0x20,
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+	0xce, 0x0c, 0x3b, 0x7f,
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+	0xe0, 0x03, 0x14, 0xd0,
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+	0x9d, 0x10, 0x33, 0x01,
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+	0x0d, 0xfe, 0xd1, 0x10,
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+	0x61, 0x00, 0x0d, 0x20,
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+	0x2f, 0x2f, 0x0e, 0x30,
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+	0x24, 0x00, 0x50, 0x17,
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+	0x14, 0x33, 0xce, 0xcc,
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+	0x30, 0xf0, 0x01, 0x34,
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+	0xd0, 0xdf, 0x0f, 0x7f,
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+	0xdf, 0x00, 0xf2, 0xb0,
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+	0x50, 0xee, 0x00, 0x00,
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+	0x21, 0x22, 0x23, 0x03,
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+	0xe0, 0x0d, 0xdf, 0xd4,
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+	0xff, 0xb0, 0xff, 0x10,
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+	0x2f, 0x45, 0xf4, 0xb1,
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+	0xda, 0x30, 0x73, 0x00,
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+	0x02, 0x0f, 0x0a, 0x20,
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+	0xef, 0xed, 0x0c, 0x1d,
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+	0xde, 0x00, 0x10, 0xe1,
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+	0xd1, 0xd3, 0x17, 0x20,
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+	0x34, 0x01, 0xf4, 0x33,
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+	0x04, 0x41, 0x11, 0xdf,
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+	0x1d, 0x3f, 0x20, 0x00,
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+	0xdd, 0xbd, 0xff, 0x7e,
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+	0xf2, 0x02, 0x31, 0xb0,
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+	0xf0, 0x0e, 0xe6, 0xc2,
+	0xe0, 0xee, 0xf0, 0xf1, 0x00, 0x20, 0xc3, 0xe1, 0x42, 0x01, 0x02, 0xe7,
+	0x11, 0x1f, 0x60, 0x27,
+	0xd1, 0xb0, 0x0f, 0x23, 0x01, 0x35, 0x41, 0x90, 0x01, 0x03, 0xee, 0x01,
+	0xf4, 0xdf, 0xf0, 0x27,
+	0xf0, 0xde, 0x10, 0x30, 0x7f, 0x41, 0x0f, 0xdd, 0x34, 0x53, 0x0f, 0xfa,
+	0x20, 0x42, 0x00, 0xee,
+	0x11, 0x00, 0xde, 0xeb, 0x00, 0x0f, 0xcc, 0xe0, 0x0f, 0x0d, 0x6e, 0x32,
+	0x20, 0x1e, 0x6f, 0x1e,
+	0x0a, 0xf0, 0x35, 0x00, 0xdd, 0xac, 0x1e, 0x4f, 0x0b, 0xec, 0x10, 0x2f,
+	0x0d, 0x1f, 0x20, 0xd9,
+	0x0f, 0x10, 0x10, 0xd3, 0xdf, 0x30, 0xa0, 0xc0, 0x3e, 0x3d, 0x00, 0xe6,
+	0x00, 0x01, 0x75, 0x00,
+	0xac, 0x00, 0x11, 0x11, 0x12, 0x03, 0xfd, 0xad, 0xf1, 0x06, 0xe2, 0xf1,
+	0xe0, 0xff, 0x10, 0x00,
+	0xbc, 0xbf, 0x04, 0x37, 0x23, 0x0f, 0xfd, 0xde, 0x0f, 0x25, 0x02, 0xd2,
+	0x20, 0x71, 0x00, 0x00,
+	0xe3, 0xf2, 0x01, 0x47, 0x01, 0xa0, 0x00, 0x00, 0xf2, 0xd2, 0xf0, 0xe1,
+	0x00, 0x70, 0x04, 0x0e,
+	0x2e, 0xfd, 0xfe, 0x12, 0x57, 0x12, 0x4d, 0x0b, 0x20, 0x00, 0x16, 0x00,
+	0x0d, 0xed, 0xd2, 0xc5,
+	0xbe, 0x0c, 0x3f, 0x01, 0xd1, 0xbf, 0x1c, 0x70, 0x62, 0x01, 0xd0, 0x0d,
+	0xfc, 0x15, 0x34, 0x1e,
+	0x0b, 0xdf, 0xef, 0x02, 0x11, 0x4e, 0x0b, 0x00, 0xd0, 0x1f, 0x4e, 0x20,
+	0xdf, 0xb2, 0x21, 0x3b,
+	0xf0, 0xa0, 0xf2, 0x10, 0x7c, 0x0e, 0xef, 0xe7, 0x06, 0x4f, 0x1e, 0xe0,
+	0xee, 0x00, 0x40, 0x23,
+	0xe3, 0xcb, 0x0e, 0x11, 0x01, 0xf2, 0xb0, 0x00, 0x0f, 0x1f, 0x05, 0x05,
+	0x88, 0x00, 0xa4, 0x00,
+	0x6a, 0xec, 0xb1, 0x07, 0x5a, 0xef, 0xc1, 0x01, 0xd2, 0x6a, 0x1e, 0xff,
+	0xc5, 0xf3, 0x22, 0x6f,
+	0x0b, 0xf0, 0xe1, 0x01, 0x02, 0xf2, 0xa0, 0xde, 0x00, 0x25, 0x20, 0x02,
+	0x00, 0x44, 0x72, 0x0e,
+	0x1e, 0x03, 0xb4, 0x00, 0x0d, 0x1f, 0x0f, 0xe5, 0x02, 0x0d, 0x0d, 0xfe,
+	0xff, 0x61, 0x37, 0x26,
+	0x0e, 0x0e, 0x0f, 0x40, 0x11, 0xe3, 0xe0, 0x0c, 0xfe, 0x10, 0x21, 0x9f,
+	0xdd, 0x20, 0x2f, 0x28,
+	0x10, 0xcf, 0xf0, 0x14, 0x40, 0x0b, 0xd0, 0x02, 0xf0, 0x2f, 0x2b, 0x50,
+	0x1f, 0xd0, 0xc0, 0x10,
+	0x4e, 0x2c, 0xff, 0x91, 0xf7, 0xe0, 0xfc, 0x00, 0x40, 0x21, 0x00, 0xe3,
+	0x0f, 0xec, 0x0f, 0x00,
+	0x02, 0xd1, 0xc1, 0xc0, 0x2d, 0x40, 0x7f, 0x23, 0xf1, 0xc2, 0xb6, 0x00,
+	0x10, 0x30, 0x21, 0xd2,
+	0xe2, 0x04, 0xe0, 0xcb, 0xe0, 0xe3, 0xd1, 0xff, 0x30, 0x74, 0x11, 0x24,
+	0x33, 0x00, 0x00, 0x07,
+	0xf6, 0x02, 0x20, 0x0f, 0x1f, 0x21, 0xf2, 0x90, 0xb0, 0xe5, 0x03, 0x00,
+	0x4b, 0xff, 0xdf, 0xf0,
+	0x02, 0x20, 0x5d, 0x12, 0x07, 0x00, 0xdd, 0x0d, 0x70, 0x10, 0x12, 0x13,
+	0x1e, 0x38, 0x10, 0xff,
+	0xd0, 0xe2, 0xe1, 0x0e, 0x29, 0x2f, 0xe0, 0xb0, 0xe0, 0xe1, 0x0f, 0x7b,
+	0x20, 0x1f, 0x50, 0x22,
+	0x03, 0x8e, 0xfd, 0x00, 0x00, 0x04, 0x02, 0xe0, 0x0e, 0x2c, 0x5b, 0x00,
+	0xde, 0x00, 0xe1, 0x05,
+	0x00, 0x0f, 0x2d, 0x3e, 0x00, 0xe2, 0xc4, 0xc6, 0xf2, 0xff, 0x1c, 0x2f,
+	0x53, 0x33, 0xf3, 0xe1,
+	0xd0, 0x00, 0x00, 0xc1, 0x82, 0xe2, 0x07, 0x02, 0x10, 0x7e, 0x00, 0xff,
+	0xe2, 0xf6, 0x22, 0x41,
+	0x32, 0x02, 0xf0, 0xd0, 0xdc, 0x00, 0x04, 0x34, 0x33, 0x4d, 0x0a, 0x9f,
+	0x00, 0x00, 0x50, 0x33,
+	0x0f, 0x0b, 0xe0, 0xf3, 0x02, 0xf1, 0x0f, 0x1a, 0xff, 0x10, 0x3f, 0x10,
+	0x23, 0xb0, 0xb8, 0x0f,
+	0x70, 0x20, 0x0e, 0xf0, 0xe1, 0x00, 0x0c, 0x0f, 0x2f, 0x30, 0x2d, 0x00,
+	0x0f, 0x4c, 0xcc, 0xdc,
+	0xee, 0xc0, 0xc1, 0xf6, 0xe0, 0x50, 0x4c, 0xef, 0xff, 0x02, 0xe0, 0xe0,
+	0x02, 0x12, 0x1f, 0xfb,
+	0xfe, 0x50, 0x14, 0x01, 0xe3, 0xbd, 0xdb, 0xe0, 0xd1, 0x21, 0x52, 0x30,
+	0x0e, 0x00, 0xb2, 0xd5,
+	0xe0, 0xd0, 0xd0, 0xd6, 0x01, 0xe3, 0x00, 0x19, 0xd0, 0x2e, 0x73, 0x67,
+	0x00, 0x00, 0x0f, 0x00,
+	0x00, 0x23, 0x22, 0x00, 0xf1, 0xd0, 0x01, 0xf0, 0xf0, 0xe0, 0xf3, 0x00,
+	0x79, 0x7f, 0x20, 0x00,
+	0xb2, 0xf0, 0x03, 0x02, 0x1a, 0x1d, 0x20, 0x03, 0xe1, 0xd0, 0x0e, 0x20,
+	0x3c, 0x7f, 0x10, 0x03,
+	0x27, 0x10, 0x0f, 0xee, 0xff, 0xc1, 0xd1, 0xd1, 0x20, 0x7f, 0x0b, 0x2c,
+	0x0c, 0xee, 0xf0, 0xf1,
+	0xe7, 0x10, 0x2f, 0x4d, 0x4f, 0x22, 0xc4, 0xe1, 0xf0, 0xed, 0xee, 0xf0,
+	0xe0, 0xe0, 0x2e, 0x7f,
+	0x6f, 0x01, 0x03, 0xd9, 0xde, 0x00, 0x05, 0x23, 0x73, 0x1f, 0x18, 0xf0,
+	0xf0, 0xf0, 0x05, 0x01,
+	0xe0, 0xef, 0xfd, 0xd0, 0xb0, 0xe1, 0xf1, 0x00, 0x4d, 0x4f, 0x00, 0xd0,
+	0xb1, 0xe7, 0x41, 0x51,
+	0x2e, 0x0e, 0x03, 0xd7, 0xe2, 0xf0, 0xff, 0x2e, 0x40, 0x50, 0x00, 0x02,
+	0xf1, 0xf0, 0x2b, 0x50,
+	0x10, 0xf1, 0xd0, 0x03, 0x1f, 0x1a, 0x0f, 0xe0, 0x90, 0xf0, 0xe0, 0x20,
+	0x40, 0x5e, 0x0a, 0x0d,
+	0xff, 0xc2, 0x04, 0x02, 0x12, 0x4c, 0x3c, 0xfd, 0xb0, 0x13, 0x00, 0x2b,
+	0x1e, 0x0f, 0xd1, 0xf3,
+	0x03, 0xeb, 0x0f, 0x41, 0x25, 0xe2, 0x8e, 0xed, 0xff, 0x00, 0x15, 0x60,
+	0x23, 0x00, 0x00, 0x0d,
+	0xd0, 0x12, 0x12, 0x07, 0x06, 0xff, 0x0e, 0x0f, 0x40, 0x00, 0x01, 0x03,
+	0xe0, 0xf1, 0x2f, 0xf0,
+	0x80, 0xa4, 0x02, 0x0f, 0x0f, 0x00, 0x00, 0xe4, 0xe2, 0xee, 0x00, 0x2f,
+	0x70, 0x1f, 0x30, 0x05,
+	0x07, 0xd4, 0xef, 0x0f, 0x01, 0x00, 0x0e, 0x10, 0x1e, 0x00, 0xf0, 0x11,
+	0x5c, 0x28, 0x3d, 0x7f,
+	0x44, 0x01, 0x0c, 0x0d, 0xdf, 0x00, 0xc3, 0x03, 0x0e, 0x39, 0x7f, 0x1f,
+	0x0e, 0x00, 0xff, 0xf0,
+	0x03, 0x02, 0xfe, 0xfb, 0x2e, 0x5e, 0x11, 0xe1, 0xd3, 0xe0, 0x08, 0xcc,
+	0xcd, 0xf1, 0x13, 0x11,
+	0x10, 0x4f, 0x0b, 0x9f, 0xde, 0xf4, 0xe5, 0xd1, 0x0f, 0x7f, 0x3f, 0xff,
+	0xb2, 0xe3, 0x07, 0xe0,
+	0xff, 0x00, 0x0f, 0xe1, 0xd1, 0xd2, 0x21, 0x71, 0x5f, 0x1f, 0x04, 0xf7,
+	0xc2, 0xff, 0x0e, 0x01,
+	0x21, 0xf0, 0xc1, 0xe2, 0xc1, 0xf0, 0xff, 0xfd, 0x0e, 0x2d, 0x10, 0xf5,
+	0x12, 0x52, 0x32, 0x00,
+	0xf2, 0xc2, 0xf7, 0xf7, 0xf0, 0x4f, 0x5e, 0x00, 0xc0, 0xe1, 0x00, 0x00,
+	0x4b, 0x6e, 0x1d, 0x1f,
+	0xff, 0xed, 0x0e, 0x00, 0x10, 0x1f, 0x0a, 0xfe, 0x0d, 0x0d, 0x3c, 0x70,
+	0x60, 0x01, 0xdb, 0xfd,
+	0x0f, 0x20, 0x13, 0x34, 0x2f, 0x3d, 0x0e, 0xc9, 0xf0, 0x0f, 0x0e, 0x30,
+	0x31, 0x3d, 0x0d, 0xad,
+	0xff, 0xf1, 0x04, 0x00, 0xed, 0xf1, 0x05, 0xff, 0x0c, 0x20, 0x23, 0x60,
+	0x23, 0x14, 0xc4, 0xb5,
+	0xf0, 0x0f, 0x2e, 0x42, 0x37, 0x14, 0x01, 0x10, 0x20, 0xf2, 0xa4, 0xe2,
+	0xb0, 0x0f, 0x01, 0x06,
+	0x01, 0xa2, 0xf0, 0xdf, 0xf0, 0xe0, 0xf2, 0x14, 0x20, 0x20, 0x0e, 0xf0,
+	0xd5, 0x02, 0x10, 0x40,
+	0x32, 0xe6, 0xc1, 0xf0, 0xde, 0x8e, 0xf2, 0xe7, 0x03, 0xf0, 0x0c, 0x1f,
+	0x4f, 0x3f, 0x3f, 0x20,
+	0x02, 0xa1, 0xee, 0x08, 0x0f, 0x20, 0x10, 0x20, 0x2e, 0x6d, 0x3d, 0x2c,
+	0xfd, 0xaf, 0xf0, 0x02,
+	0x1f, 0x5b, 0x3f, 0x4f, 0xff, 0xd0, 0xff, 0x1b, 0x4e, 0x5e, 0x2e, 0x2f,
+	0xff, 0xe0, 0x00, 0x1d,
+	0x3d, 0x10, 0x04, 0xf0, 0xcf, 0xed, 0x10, 0x6e, 0x30, 0x30, 0x07, 0xb7,
+	0xd0, 0xe2, 0xf1, 0xfe,
+	0x0d, 0x41, 0x15, 0xf5, 0xd2, 0x00, 0x40, 0x50, 0x01, 0x02, 0xf3, 0x04,
+	0x54, 0x10, 0x32, 0x04,
+	0xa2, 0xf5, 0xc1, 0xd2, 0xe1, 0x01, 0x10, 0x10, 0xe0, 0x8e, 0xf0, 0x07,
+	0x32, 0x22, 0x0e, 0x1e,
+	0xf1, 0x04, 0xe7, 0xf2, 0x03, 0xe0, 0xf0, 0xef, 0xc0, 0x03, 0x00, 0x7d,
+	0x2e, 0x0e, 0x0d, 0xdd,
+	0xce, 0xff, 0xfc, 0x0f, 0x42, 0x3d, 0x18, 0x0d, 0xe0, 0xe0, 0xfc, 0x1d,
+	0x2c, 0x70, 0x21, 0x02,
+	0xd9, 0x0d, 0x5f, 0x3f, 0x30, 0x02, 0xfd, 0x08, 0x3f, 0x3d, 0x60, 0x2f,
+	0x1f, 0x0e, 0x0f, 0x00,
+	0xdf, 0xfe, 0x0a, 0x1f, 0x1d, 0x30, 0x72, 0x41, 0x20, 0xee, 0xfc, 0xd1,
+	0xa2, 0xe1, 0xff, 0x00,
+	0x20, 0x45, 0x00, 0xb0, 0xe2, 0xe2, 0x03, 0x33, 0x43, 0x31, 0xf0, 0xc2,
+	0xf3, 0x05, 0x07, 0xe7,
+	0x82, 0xd1, 0xde, 0xee, 0x00, 0x00, 0xe4, 0xb1, 0xf2, 0x02, 0x4f, 0x1f,
+	0x20, 0x03, 0x07, 0x03,
+	0xf4, 0xb1, 0xd0, 0xe2, 0xd4, 0xe3, 0xf4, 0xd0, 0x00, 0x71, 0x31, 0x2f,
+	0xf0, 0xd2, 0xb0, 0x00,
+	0x13, 0x26, 0x61, 0x0d, 0x0d, 0xfc, 0xe0, 0x10, 0x21, 0x0c, 0x5b, 0x2e,
+	0xfd, 0xef, 0xef, 0x10,
+	0x50, 0x2b, 0x30, 0x04, 0xc0, 0x0d, 0x09, 0x40, 0x4d, 0x5f, 0x00, 0x03,
+	0x00, 0x0a, 0x0f, 0x2e,
+	0x0f, 0x1e, 0x0f, 0x0e, 0x2f, 0x50, 0x2e, 0x0b, 0xee, 0xdf, 0xf1, 0xe3,
+	0xe0, 0x0b, 0x1d, 0x4f,
+	0xdf, 0xa0, 0xc6, 0xd2, 0x01, 0x2f, 0x7d, 0x3f, 0x00, 0xf0, 0xf6, 0xf4,
+	0x03, 0x30, 0xff, 0xbe,
+	0xe1, 0xa3, 0xf4, 0xd3, 0x00, 0xef, 0xee, 0xc2, 0xe5, 0xf2, 0xfe, 0x0c,
+	0x7f, 0x12, 0x16, 0xe2,
+	0xd1, 0xd1, 0x00, 0x05, 0x05, 0x62, 0x11, 0x10, 0xef, 0xf0, 0x03, 0x03,
+	0x12, 0xf0, 0xe1, 0xe0,
+	0xd0, 0x00, 0xfe, 0x3f, 0x7e, 0x14, 0x13, 0x0e, 0x08, 0x1e, 0x5e, 0x30,
+	0x20, 0x00, 0x00, 0x05,
+	0x30, 0x5f, 0x29, 0x1f, 0xf0, 0xd0, 0x3e, 0x7e, 0x7d, 0x3f, 0x0f, 0xd0,
+	0xff, 0x0c, 0x1e, 0x20,
+	0x3c, 0x0c, 0x1d, 0x0d, 0x00, 0x21, 0x04, 0x32, 0x4d, 0x18, 0xfe, 0xdf,
+	0x0e, 0x20, 0x0e, 0xa0,
+	0xc5, 0xa0, 0xdd, 0x0d, 0x1d, 0x50, 0x11, 0xf5, 0xf0, 0x0f, 0x3e, 0x02,
+	0x17, 0x00, 0xd0, 0xde,
+	0xc2, 0xd5, 0xd2, 0x90, 0xdd, 0xc0, 0xe3, 0xe2, 0xe2, 0x10, 0x60, 0x2c,
+	0x0e, 0xc0, 0xe4, 0xe7,
+	0x02, 0xd0, 0x11, 0x13, 0x22, 0xd3, 0x91, 0xf0, 0x01, 0x07, 0x31, 0x01,
+	0x9e, 0x0f, 0x01, 0x01,
+	0x10, 0x41, 0x03, 0x01, 0x04, 0xe0, 0x0e, 0x0a, 0x0e, 0x20, 0x20, 0x02,
+	0x10, 0xe0, 0xec, 0x0f,
+	0x2f, 0x01, 0x00, 0x2e, 0x70, 0x70, 0x5a, 0x10, 0x0f, 0x0b, 0x02, 0x07,
+	0x11, 0x1e, 0x2b, 0x2e,
+	0x3e, 0x3f, 0xf0, 0xc2, 0x00, 0x3c, 0x6e, 0x2e, 0x30, 0x00, 0xce, 0xff,
+	0x00, 0x24, 0x01, 0xc0,
+	0xde, 0xaf, 0xfe, 0x0d, 0x20, 0x74, 0x11, 0x0c, 0xfe, 0xf3, 0x02, 0x30,
+	0x2e, 0x23, 0x06, 0xd0,
+	0xc0, 0xde, 0x84, 0xf1, 0xe0, 0x07, 0x01, 0x0f, 0xfe, 0xfd, 0x08, 0x0e,
+	0x2f, 0x22, 0xc7, 0xd1,
+	0xe2, 0x20, 0x3e, 0x40, 0x03, 0xa6, 0x03, 0xf2, 0x0f, 0x1e, 0x00, 0x02,
+	0xb0, 0xff, 0xe2, 0x05,
+	0x41, 0x20, 0xee, 0xbe, 0xfe, 0xcd, 0x0e, 0x1e, 0x6c, 0x3b, 0x0f, 0xee,
+	0xe0, 0x0f, 0x13, 0x24,
+	0x2f, 0x1b, 0x0d, 0x10, 0x23, 0x12, 0x02, 0xf0, 0x2d, 0x3b, 0x0f, 0xd4,
+	0x01, 0x7a, 0x7e, 0x1e,
+	0x00, 0xe3, 0x06, 0x20, 0x3e, 0x4e, 0x30, 0x20, 0x01, 0x01, 0xf1, 0x01,
+	0x0e, 0xcc, 0xb0, 0xd5,
+	0xd2, 0xf0, 0xfa, 0x00, 0x5f, 0x01, 0xb0, 0x03, 0x03, 0x74, 0x30, 0x00,
+	0x11, 0x04, 0xe1, 0xf0,
+	0xc1, 0xd7, 0xe5, 0x00, 0xd0, 0xef, 0x0f, 0xc0, 0xef, 0x0d, 0x20, 0x2f,
+	0x0f, 0xa0, 0xc0, 0x00,
+	0x02, 0x10, 0x20, 0x27, 0x26, 0xd0, 0xa2, 0xe0, 0x1d, 0x1f, 0x11, 0x53,
+	0x04, 0xf1, 0xe1, 0x00,
+	0x21, 0x00, 0xcf, 0xef, 0x0b, 0x2e, 0x6a, 0x3e, 0x4e, 0x00, 0xd2, 0xc0,
+	0xeb, 0xed, 0x00, 0x2e,
+	0x5c, 0x0d, 0xdb, 0xf0, 0x02, 0x51, 0x30, 0x3b, 0x2e, 0x0e, 0x0f, 0x00,
+	0x43, 0x41, 0x2f, 0xf9,
+	0xf0, 0xd0, 0x93, 0xf0, 0x1d, 0x6e, 0x3f, 0x0f, 0xe0, 0xd4, 0xf2, 0x0d,
+	0x3f, 0x37, 0xf4, 0xa0,
+	0xe0, 0xc0, 0x0f, 0x0f, 0x11, 0x06, 0x03, 0x12, 0x3e, 0x2c, 0x70, 0x11,
+	0x12, 0x01, 0xc4, 0xf4,
+	0xe3, 0xd0, 0xe0, 0xf4, 0xc1, 0xd0, 0xc0, 0xe1, 0x15, 0x41, 0xf0, 0xde,
+	0xeb, 0xdf, 0x00, 0x53,
+	0x27, 0x30, 0x10, 0x0e, 0xa0, 0xf4, 0xf3, 0x1f, 0x3c, 0x6f, 0x00, 0x03,
+	0xd2, 0xf0, 0x0f, 0x0f,
+	0x0f, 0x00, 0x00, 0x5a, 0x2d, 0x0b, 0x3e, 0x5f, 0x20, 0x12, 0xd2, 0x0c,
+	0x0a, 0x3f, 0x4d, 0x1f,
+	0x1f, 0x11, 0x12, 0x50, 0x2d, 0x0e, 0x0c, 0x2d, 0x20, 0x60, 0x0f, 0xfc,
+	0xff, 0xcd, 0x09, 0xfe,
+	0xff, 0x11, 0x50, 0x7e, 0x3a, 0x3f, 0x00, 0xf0, 0xe0, 0xf6, 0x51, 0x11,
+	0x0f, 0x0f, 0xe0, 0xa0,
+	0xd3, 0xc3, 0xe1, 0x01, 0x14, 0x31, 0x10, 0xfd, 0xd3, 0x67, 0x33, 0x44,
+	0x02, 0xf3, 0xd2, 0xf1,
+	0xe3, 0xc4, 0x07, 0x03, 0xe1, 0xd0, 0xbe, 0xf0, 0xe0, 0x01, 0xef, 0x0f,
+	0xe0, 0xf0, 0xf1, 0x27,
+	0x46, 0xd0, 0x90, 0xff, 0x01, 0xf4, 0x11, 0x61, 0x30, 0x0f, 0xf0, 0xd2,
+	0xd7, 0x12, 0x00, 0x0c,
+	0x6f, 0x20, 0xf0, 0xde, 0x0f, 0x3f, 0x70, 0x1e, 0xfc, 0xee, 0xfb, 0x0e,
+	0x3e, 0x1d, 0x5e, 0x1e,
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+	0xe0, 0xf2, 0xd2, 0xe1,
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+	0x2d, 0x3f, 0x4d, 0x10,
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+	0x4f, 0x01, 0xd2, 0xf0,
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+	0x3e, 0x30, 0x01, 0x20,
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+	0xf0, 0xde, 0x0e, 0x00,
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+	0xde, 0xc0, 0xef, 0xe1,
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+	0x00, 0xd5, 0x01, 0x02,
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+	0x40, 0x37, 0x11, 0xd0,
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+	0x16, 0x01, 0xff, 0x7c,
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+	0x1e, 0x4e, 0x20, 0x5f,
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+	0x01, 0x0f, 0x3a, 0x70,
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+	0xe1, 0x0e, 0x0e, 0x0e,
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+	0x55, 0x23, 0x24, 0x03,
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+	0x4a, 0x10, 0x22, 0xe2,
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+	0x12, 0x00, 0x0e, 0x00,
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+	0x0b, 0xfc, 0x00, 0x42,
+	0x50, 0x30, 0x1e, 0xff, 0xdf, 0xde, 0xd2, 0xf6, 0xb4, 0x00, 0x00, 0xdc,
+	0xff, 0x01, 0x03, 0x20,
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+	0x00, 0xf0, 0xfe, 0xb1,
+	0xe3, 0xef, 0x08, 0x0e, 0x0e, 0x10, 0xf2, 0xb1, 0x0e, 0x0d, 0x30, 0x67,
+	0x23, 0x05, 0x00, 0x20,
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+	0x03, 0xd0, 0x0f, 0x6e,
+	0x3b, 0x0c, 0xfc, 0xee, 0x02, 0x24, 0xd2, 0x80, 0xeb, 0x0f, 0x01, 0x64,
+	0x11, 0x52, 0x10, 0x10,
+	0x30, 0xfd, 0xe0, 0xf7, 0xe3, 0x00, 0x00, 0x0d, 0xdd, 0xec, 0xc0, 0xd0,
+	0x01, 0x1e, 0x78, 0x7f,
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+	0x05, 0x0e, 0x3d, 0x4c,
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+	0xf0, 0xfc, 0xb0, 0xf3,
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+	0x20, 0x04, 0x07, 0xe1,
+	0xf1, 0x01, 0x0f, 0x2d, 0x0d, 0xee, 0xbd, 0xe2, 0xe7, 0xf5, 0x01, 0x1f,
+	0x1d, 0xff, 0xbf, 0xf0,
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+	0x11, 0xf3, 0xe3, 0xf2,
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+	0x4e, 0x2c, 0x3c, 0x2f,
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+	0x0f, 0xd0, 0x0d, 0x00,
+	0xee, 0x0f, 0x50, 0x73, 0x64, 0x0f, 0x00, 0xf0, 0x17, 0x07, 0x00, 0xe1,
+	0xff, 0x0f, 0xe0, 0x02,
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+	0xe1, 0xf6, 0xef, 0xdd,
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+	0xff, 0x01, 0xf4, 0xd2,
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+	0x40, 0x11, 0x20, 0x0e,
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+	0x04, 0xe2, 0xef, 0xff,
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+	0xf0, 0x21, 0x00, 0xde,
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+	0x21, 0x54, 0xfe, 0xd8,
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+	0xdd, 0xd0, 0xe3, 0xd2,
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+	0x8e, 0x01, 0x07, 0x00,
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+	0x32, 0xfd, 0xdd, 0x22,
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+	0x22, 0xde, 0x11, 0x67,
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+	0x11, 0xcc, 0xef, 0x43,
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+	0x22, 0xbd, 0xdf, 0x57,
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+	0x01, 0x12, 0xee, 0x01,
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+	0x23, 0x30, 0xe0, 0x0f,
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+	0x43, 0x10, 0xfa, 0xef,
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+	0x10, 0x32, 0x00, 0xbc,
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+	0x01, 0x21, 0xde, 0xdd,
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+	0x22, 0xce, 0xef, 0xf0,
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+	0xfe, 0x02, 0x01, 0x02,
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+	0x02, 0xac, 0xf0, 0x03,
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+	0xfe, 0xed, 0x42, 0x47,
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+	0xee, 0x11, 0x22, 0x00,
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+	0x23, 0xfe, 0xf0, 0x21,
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+	0xf0, 0xaa, 0x00, 0x44,
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+	0x10, 0xff, 0x01, 0x33,
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+	0xff, 0xdc, 0x00, 0x11,
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+	0x42, 0x42, 0xfe, 0x0d,
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+	0x01, 0xcf, 0xef, 0xf1,
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+	0x13, 0x33, 0xef, 0xb8,
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+	0x00, 0x01, 0xef, 0x00,
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+	0xcd, 0x00, 0x77, 0x00,
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+	0x10, 0x40, 0x10, 0xff,
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+	0x01, 0xff, 0xed, 0x11,
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+	0xce, 0x01, 0x14, 0x01,
+	0xdf, 0xf0, 0x01, 0x37, 0x24, 0x8d, 0xff, 0x01, 0x03, 0x20, 0x02, 0x10,
+	0x00, 0x01, 0xf3, 0x02,
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+	0x12, 0x10, 0xfc, 0xf0,
+	0x13, 0x42, 0xff, 0xfa, 0x11, 0x63, 0x0f, 0x1f, 0x21, 0xef, 0xeb, 0x33,
+	0x65, 0x0f, 0xfd, 0xf0,
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+	0x41, 0x0f, 0x0f, 0x25,
+	0x42, 0x00, 0x1d, 0x00, 0x23, 0x45, 0x11, 0xd0, 0x0f, 0x23, 0x41, 0xf0,
+	0xcc, 0x13, 0x27, 0xf0,
+	0xf0, 0x0e, 0x73, 0x22, 0x00, 0xf0, 0x12, 0x00, 0x03, 0x07, 0xd0, 0xf1,
+	0x01, 0x9f, 0xf0, 0x37,
+	0x43, 0xcf, 0xe0, 0xe0, 0xd0, 0x77, 0x12, 0xcd, 0x00, 0x54, 0x00, 0xe0,
+	0x00, 0x35, 0x00, 0xd0,
+	0xff, 0x21, 0x24, 0xff, 0xc0, 0x11, 0x23, 0xfd, 0xdf, 0x43, 0x36, 0xbe,
+	0xed, 0x44, 0x11, 0xde,
+	0xfe, 0x33, 0x73, 0xfc, 0x0f, 0x21, 0x11, 0xfd, 0x00, 0x31, 0xfd, 0x0c,
+	0x31, 0x02, 0x64, 0x0e,
+	0x0b, 0x10, 0x22, 0xff, 0x0f, 0x1d, 0x0c, 0x51, 0x00, 0xc8, 0x0f, 0x61,
+	0x00, 0x0c, 0x20, 0x11,
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+	0xce, 0x13, 0x03, 0xcb,
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+	0xde, 0xee, 0xf1, 0x14,
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+	0x13, 0x15, 0xff, 0xbe,
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+	0xe2, 0x0f, 0xdc, 0x00,
+	0x23, 0xee, 0x2d, 0x20, 0xda, 0xee, 0x61, 0x01, 0xdb, 0x0d, 0x40, 0x51,
+	0xee, 0xfe, 0x00, 0x41,
+	0x0f, 0xf8, 0x0f, 0x00, 0x0f, 0x0e, 0xde, 0xcd, 0xff, 0x4f, 0x1d, 0xea,
+	0x0e, 0x1e, 0x40, 0x10,
+	0x0b, 0x20, 0x10, 0xff, 0x6e, 0x30, 0x0e, 0x02, 0xcf, 0x98, 0x00, 0xe0,
+	0xe0, 0xff, 0xef, 0xdd,
+	0x2f, 0x40, 0xed, 0xdf, 0x01, 0xd0, 0x88, 0x00, 0x11, 0xdf, 0x01, 0x01,
+	0x69, 0x00, 0x46, 0x00,
+	0x14, 0xd0, 0xa9, 0xce, 0x1b, 0xd2, 0x80, 0xcb, 0x24, 0xcb, 0xbf, 0x02,
+	0x02, 0x00, 0x03, 0x0d,
+	0xe8, 0x1f, 0x76, 0x60, 0xfe, 0x0f, 0x22, 0x00, 0xef, 0x0e, 0x02, 0x11,
+	0x0d, 0x10, 0x00, 0xff,
+	0x10, 0x77, 0x0e, 0xcb, 0xff, 0x00, 0x03, 0xe0, 0xda, 0x51, 0x52, 0x00,
+	0xdd, 0x0f, 0x52, 0x03,
+	0xfd, 0xfb, 0x10, 0x32, 0x01, 0xbd, 0x00, 0x02, 0x43, 0x01, 0xee, 0xff,
+	0x03, 0x02, 0x1e, 0xdb,
+	0x04, 0x03, 0x0c, 0x9b, 0x00, 0x33, 0x0f, 0xfd, 0xf0, 0x14, 0x00, 0x02,
+	0x03, 0x6f, 0x1f, 0x17,
+	0xf3, 0xce, 0xf0, 0xf4, 0xf2, 0x8d, 0xfd, 0x15, 0x10, 0xee, 0x0e, 0x11,
+	0x22, 0xff, 0x01, 0xf0,
+	0xfb, 0x21, 0x42, 0xfd, 0x0d, 0x13, 0x04, 0xf4, 0x0f, 0xd9, 0xb0, 0xf3,
+	0xe4, 0xce, 0xce, 0x00,
+	0x60, 0x02, 0x0f, 0x40, 0x13, 0x1f, 0x3d, 0x0b, 0x00, 0x14, 0x32, 0xbd,
+	0x0d, 0x74, 0x21, 0x2f,
+	0xcb, 0xef, 0x01, 0x12, 0xf8, 0xb0, 0x00, 0x10, 0x20, 0xdd, 0xe0, 0x20,
+	0x71, 0x0f, 0x0e, 0x0f,
+	0x0b, 0x2e, 0x32, 0x01, 0xae, 0x0f, 0x34, 0x22, 0x0f, 0xf0, 0x27, 0x03,
+	0xad, 0xff, 0xf1, 0x03,
+	0x0f, 0xde, 0x0c, 0x0b, 0xf0, 0xe1, 0xff, 0x0d, 0x20, 0xf5, 0x33, 0x5e,
+	0x3b, 0x75, 0x75, 0xff,
+	0xcd, 0x01, 0x04, 0x01, 0xf0, 0xee, 0x00, 0x00, 0x00, 0xe0, 0xf0, 0x03,
+	0x20, 0xf9, 0x10, 0x44,
+	0xcf, 0xff, 0x30, 0x50, 0xfe, 0xe0, 0x26, 0x00, 0xc0, 0xf0, 0xff, 0x00,
+	0xf2, 0xb2, 0xef, 0xfc,
+	0x20, 0x75, 0x30, 0x0c, 0xf0, 0xf1, 0x01, 0x00, 0x0c, 0x41, 0x33, 0x20,
+	0x0c, 0x00, 0x24, 0x10,
+	0x00, 0xca, 0xd0, 0xff, 0x00, 0x9e, 0xfe, 0x24, 0x02, 0xe0, 0x2d, 0x70,
+	0x0a, 0x20, 0x31, 0xd0,
+	0xcc, 0xff, 0x00, 0x71, 0x12, 0xf9, 0xf0, 0x61, 0x05, 0x0e, 0xc0, 0x12,
+	0x01, 0x0e, 0x0d, 0x02,
+	0x03, 0x10, 0xef, 0xcc, 0xee, 0x00, 0x01, 0xdf, 0x0e, 0x54, 0x6f, 0x0f,
+	0x05, 0x02, 0xf0, 0x00,
+	0x04, 0x05, 0xd0, 0xd0, 0xff, 0x04, 0xcf, 0xfd, 0x10, 0x12, 0xe3, 0xb0,
+	0xfb, 0x00, 0x30, 0x0e,
+	0x50, 0x03, 0xff, 0xed, 0x71, 0x12, 0x00, 0xc6, 0xef, 0xf0, 0x04, 0xe0,
+	0xe3, 0xf3, 0x06, 0x50,
+	0x0c, 0x00, 0x65, 0x00, 0xdb, 0x00, 0x22, 0xfe, 0x0f, 0x74, 0x61, 0xfe,
+	0xed, 0x03, 0x34, 0x09,
+	0xef, 0x10, 0x12, 0xdd, 0xee, 0x03, 0xf3, 0x00, 0x0c, 0x0c, 0x00, 0x53,
+	0x0f, 0xfb, 0x00, 0x22,
+	0x0d, 0x0d, 0x70, 0x23, 0x10, 0x2c, 0x30, 0x02, 0xb0, 0xe1, 0x07, 0x57,
+	0xf0, 0xda, 0x41, 0x11,
+	0xd0, 0xb0, 0xff, 0xf0, 0x0f, 0xe0, 0xdd, 0x21, 0x53, 0x0e, 0x10, 0x40,
+	0x10, 0x21, 0xe6, 0xe3,
+	0x00, 0xff, 0x0e, 0x34, 0xe3, 0x9f, 0xdb, 0xe0, 0x13, 0x12, 0xef, 0x9e,
+	0x00, 0xf0, 0xfc, 0x0f,
+	0x51, 0x14, 0x20, 0x0f, 0x0f, 0x2f, 0xed, 0xd3, 0xe7, 0xf0, 0xbf, 0xef,
+	0x14, 0x43, 0x20, 0xd0,
+	0xee, 0x01, 0x1f, 0x0a, 0x01, 0xe1, 0x20, 0x7f, 0x33, 0x05, 0x30, 0x71,
+	0x20, 0x0a, 0x0f, 0x00,
+	0xff, 0xc0, 0x12, 0x23, 0xca, 0xce, 0x00, 0x31, 0x11, 0x0c, 0xed, 0xf0,
+	0x23, 0x10, 0xcb, 0x0f,
+	0x6e, 0x3c, 0x1d, 0x21, 0x04, 0x00, 0xdb, 0x30, 0x55, 0x11, 0x00, 0x31,
+	0x11, 0x00, 0xbe, 0x1f,
+	0x23, 0x84, 0xcd, 0xdb, 0x11, 0x20, 0x0f, 0xfb, 0x01, 0x10, 0x20, 0x51,
+	0x11, 0x13, 0xe0, 0xf1,
+	0xf3, 0xe1, 0x02, 0x21, 0xd0, 0xe7, 0x07, 0x90, 0xcb, 0xf0, 0xf3, 0xe0,
+	0xd0, 0xff, 0x31, 0x5f,
+	0x5d, 0x42, 0x02, 0xed, 0xcf, 0x04, 0xf1, 0xfa, 0xee, 0x24, 0x01, 0xaf,
+	0x00, 0x07, 0x31, 0xff,
+	0xed, 0x20, 0x00, 0xe0, 0xf1, 0x23, 0x3f, 0x0e, 0x7e, 0x75, 0x23, 0x0f,
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+	0x1b, 0x21, 0xd0, 0xce,
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+	0xef, 0xe2, 0xe6, 0xaf,
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+	0xe0, 0xf0, 0xf0, 0x00,
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+	0x2f, 0x38, 0x2e, 0x41,
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+	0x03, 0xe2, 0x81, 0xf0,
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+	0x0f, 0xde, 0x8f, 0x0e,
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+	0xb0, 0xfe, 0xf0, 0xe5,
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+	0x42, 0x00, 0x28, 0x0f,
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+	0x1b, 0x00, 0x22, 0x00,
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+	0x10, 0x0f, 0xfe, 0xc2,
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+	0x0f, 0xff, 0xde, 0xff,
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+	0x34, 0x02, 0xfd, 0x0d,
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+	0x00, 0xe0, 0x0f, 0x2d,
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+	0xdd, 0x0f, 0x12, 0x53,
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+	0x1e, 0xf1, 0xb3, 0x00,
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+	0x32, 0x74, 0x40, 0x0e,
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+	0x92, 0xe0, 0xfc, 0xe0,
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+	0x31, 0x77, 0x57, 0x00,
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+	0x03, 0x1a, 0x4c, 0x3e,
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+	0x11, 0x00, 0xfe, 0xff,
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+	0xf0, 0x0d, 0x5e, 0x30,
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+	0x21, 0x11, 0x02, 0x00,
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+	0xd0, 0x80, 0x06, 0x88,
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+	0x40, 0xed, 0x2f, 0x77,
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+	0x31, 0x0f, 0x00, 0x23,
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+	0xaf, 0x0f, 0xe3, 0x01,
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+	0x32, 0x45, 0x20, 0x42,
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+	0x6c, 0x3d, 0x01, 0x13,
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+	0x05, 0x00, 0xbd, 0x00,
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+	0x61, 0x54, 0x24, 0x02,
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+	0x01, 0x04, 0x0c, 0x00,
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+	0xf3, 0xf4, 0xe2, 0xe2,
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+	0x32, 0x07, 0xd1, 0xd0,
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+	0x33, 0x43, 0x0e, 0x0b,
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+	0xd0, 0x0d, 0xf1, 0x57,
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+	0xc0, 0x00, 0x2d, 0x3d,
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+	0xde, 0xec, 0x30, 0x74,
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+	0xff, 0xfd, 0x00, 0x13,
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+	0x17, 0x31, 0xac, 0x00,
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+	0x26, 0x00, 0xee, 0x00,
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+	0x21, 0x0e, 0x1f, 0x00,
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+	0x70, 0x01, 0x23, 0xec,
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+	0xe2, 0xd0, 0xd0, 0x00,
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+	0x77, 0xff, 0xdd, 0x10,
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+	0x0f, 0x21, 0x22, 0x23,
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+	0xef, 0xd0, 0x80, 0xef,
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+	0x88, 0x0d, 0x32, 0x22,
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+	0xdd, 0xdd, 0x0f, 0x21,
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+	0x0e, 0xdb, 0x12, 0x11,
+	0xff, 0x0d, 0x32, 0x41, 0xee, 0xcb, 0x00, 0x32, 0x21, 0xc0, 0xd1, 0xc0,
+	0xb8, 0x10, 0x64, 0xf0,
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+	0xce, 0x00, 0x54, 0x02,
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+	0x1e, 0x01, 0xe3, 0xe0,
+	0xfc, 0x0f, 0x0f, 0x00, 0x15, 0x01, 0x8b, 0x10, 0x62, 0x02, 0xe0, 0x00,
+	0x0f, 0x0c, 0xff, 0x2f,
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+	0x36, 0xf0, 0xcd, 0x10,
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+	0x20, 0x72, 0x23, 0xdc,
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+	0x01, 0xca, 0x00, 0x51,
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+	0xce, 0x46, 0xf0, 0x0f,
+	0x01, 0xf0, 0xe0, 0xff, 0x00, 0x63, 0x00, 0x0d, 0x04, 0x21, 0xf0, 0x1e,
+	0x50, 0x00, 0xfa, 0x01,
+	0x24, 0x23, 0xfe, 0xcc, 0x00, 0x33, 0x77, 0xff, 0xdd, 0x40, 0x63, 0x00,
+	0xfe, 0xdd, 0x13, 0x37,
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+	0x00, 0xe0, 0x12, 0x36,
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+	0x10, 0x20, 0x00, 0xdf,
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+	0xd0, 0x10, 0x12, 0x11,
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+	0xec, 0xbc, 0x33, 0x43,
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+	0x21, 0xbd, 0xf0, 0x05,
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+	0x15, 0xfe, 0xdd, 0xf0,
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+	0x52, 0x42, 0xd0, 0x0e,
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+	0xc0, 0x88, 0xde, 0x73,
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+	0xfe, 0x23, 0x23, 0x14,
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+	0x23, 0x13, 0xf4, 0xf4,
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+	0x30, 0x21, 0x01, 0x50,
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+	0x00, 0x00, 0x30, 0x30,
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+	0x33, 0x50, 0x10, 0x20,
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+	0x3f, 0xfb, 0x8e, 0x1e,
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+	0xa8, 0xfd, 0xee, 0x0d,
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+	0xf2, 0x03, 0x03, 0xb0,
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+	0xff, 0x0e, 0x1e, 0xdd,
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+	0xb5, 0xe3, 0xf2, 0xf1,
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+	0x0f, 0xfd, 0xef, 0xde,
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+	0xa0, 0xe1, 0xf2, 0x04,
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+	0x61, 0x00, 0x46, 0x00,
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+	0xf0, 0x04, 0x02, 0xd1,
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+	0xb2, 0xf0, 0x0e, 0xef,
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+	0x20, 0x71, 0x1d, 0x1e,
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+	0x0e, 0xf0, 0xff, 0xfd,
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+	0xce, 0xc0, 0xe0, 0xd0,
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+	0x11, 0xc0, 0x91, 0xf2,
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+	0xc0, 0xeb, 0xef, 0xdf,
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+	0x0a, 0x50, 0x22, 0x35,
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+	0xdd, 0xe0, 0x03, 0x10,
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+	0x0e, 0x2f, 0x51, 0x20,
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+	0x0d, 0x20, 0x37, 0x03,
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+	0xef, 0x0f, 0xe0, 0x02,
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+	0xf0, 0xe7, 0xf0, 0x12,
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+	0x4e, 0x10, 0x02, 0xa5,
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+	0x02, 0x12, 0x21, 0x11,
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+	0xc8, 0xdf, 0xc0, 0x12,
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+	0xfd, 0xfd, 0x00, 0x0e,
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+	0xaf, 0xcf, 0xdf, 0xc8,
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+	0x32, 0x00, 0xef, 0xf0,
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+	0xf0, 0xf1, 0xfb, 0xeb,
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+	0xf1, 0xe0, 0xd0, 0xb0,
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+	0x00, 0x3f, 0x0d, 0x00,
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+	0x0d, 0x2c, 0x7e, 0x3d,
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+	0x0f, 0x3d, 0x42, 0xd2,
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+	0x00, 0x00, 0xf0, 0x20,
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+	0xfd, 0xe0, 0x03, 0x12,
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+	0x0f, 0x13, 0x10, 0x1f,
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+	0x1e, 0x02, 0x27, 0x37,
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+	0x0b, 0x10, 0x3e, 0x50,
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+	0xfc, 0x0d, 0x16, 0x42,
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+	0x32, 0x53, 0x0f, 0xed,
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+	0x01, 0x12, 0x00, 0x30,
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+	0x42, 0x32, 0xec, 0xdb,
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+	0xdb, 0x00, 0x30, 0x20,
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+	0x70, 0x02, 0x0f, 0x0f,
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+	0x80, 0xf2, 0xf1, 0xcf,
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+	0xdc, 0x0f, 0x31, 0x10,
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+	0x11, 0x13, 0x10, 0xdf,
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+	0xf1, 0xf1, 0xd0, 0xf4,
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+	0xfe, 0x11, 0x23, 0xff,
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+	0x11, 0x12, 0x21, 0x01,
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+	0x03, 0xf1, 0xff, 0xff,
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+	0x02, 0x13, 0x2f, 0x0c,
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+	0x51, 0x76, 0x62, 0x00,
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+	0x90, 0x0f, 0x00, 0x1d,
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+	0x02, 0x21, 0x20, 0x1e,
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+	0xd1, 0xe1, 0xc0, 0x00,
+	0xf4, 0x0e, 0x7b, 0x0d, 0x00, 0x00, 0x0b, 0xcd, 0x01, 0xf3, 0xf1, 0xcd,
+	0x0d, 0x13, 0x56, 0x31,
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+	0xc6, 0xe3, 0xd0, 0xb0,
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+	0x19, 0x00, 0x1e, 0x7f,
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+	0xf1, 0xe3, 0xe0, 0x0d,
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+	0x14, 0x44, 0x40, 0x2e,
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+	0xf0, 0x20, 0x10, 0xee,
+	0xce, 0x04, 0x06, 0xb0, 0xeb, 0x0f, 0x31, 0xe3, 0xb0, 0x01, 0x15, 0xc3,
+	0xd2, 0xe3, 0xd1, 0xd1,
+	0xc6, 0xf3, 0xe2, 0x80, 0xc0, 0xbd, 0xdf, 0xf0, 0x0d, 0xed, 0x18, 0x6b,
+	0x40, 0x40, 0x20, 0x03,
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+	0x00, 0xf1, 0xfc, 0x0b,
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+	0xe1, 0xe4, 0xf0, 0x04,
+	0x43, 0x52, 0x6f, 0x2a, 0x3f, 0x3f, 0x3e, 0x2a, 0x4c, 0x2d, 0x00, 0x1f,
+	0x00, 0xd1, 0xce, 0xfd,
+	0xeb, 0xdf, 0xf2, 0x30, 0x3a, 0xe0, 0xd0, 0x11, 0x00, 0x8e, 0xdc, 0xb0,
+	0xe2, 0xee, 0xee, 0xe3,
+	0x17, 0x07, 0xf1, 0xf0, 0xd1, 0xe3, 0xa2, 0xe0, 0xe0, 0x00, 0xf0, 0x0f,
+	0x20, 0x70, 0x30, 0xc0,
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+	0x10, 0xfd, 0xdb, 0xfd,
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+	0x1b, 0x08, 0xee, 0x20,
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+	0x3a, 0x5d, 0x1a, 0x30,
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+	0xdf, 0xd0, 0xe2, 0xf0,
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+	0x0f, 0xdb, 0x4f, 0x70,
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+	0x04, 0x13, 0x02, 0x20,
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+	0xe0, 0xce, 0xe0, 0x12,
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+	0xee, 0xd0, 0xe3, 0xcd,
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+	0x10, 0x00, 0x15, 0x11,
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+	0x25, 0x16, 0xc1, 0xd2,
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+	0x6b, 0x3b, 0x4e, 0x2e,
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+	0xeb, 0x0f, 0x21, 0x46,
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+	0x01, 0xf4, 0x02, 0x41,
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+	0x31, 0x12, 0x1d, 0x4d,
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+	0xe6, 0xe4, 0xd1, 0xd2,
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+	0xaf, 0xf0, 0xde, 0xfe,
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+	0xf3, 0x04, 0x0e, 0xef,
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+	0xee, 0xad, 0xfd, 0xbf,
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+	0x30, 0x11, 0x4d, 0x0d,
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+	0x00, 0xe0, 0xfa, 0xff,
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+	0x3d, 0x3e, 0x70, 0x3f,
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+	0xd6, 0xc3, 0xdf, 0xd0,
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+	0x13, 0x30, 0x2e, 0x5c,
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+	0xce, 0xfe, 0x0c, 0x30,
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+	0xf1, 0xa1, 0xf5, 0xf2,
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+	0x22, 0x02, 0x15, 0x21,
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+	0x29, 0x1c, 0x0e, 0x1e,
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+	0xdf, 0xf6, 0xe7, 0x02,
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+	0x4f, 0x2f, 0x1e, 0x11,
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+	0xfc, 0xf0, 0xef, 0x32,
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+	0xd0, 0xd0, 0xd1, 0x04,
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+	0xb1, 0xef, 0xc0, 0xf0,
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+	0xf0, 0xd0, 0xe7, 0xd1,
+	0xef, 0xfe, 0xdf, 0xdf, 0xca, 0x0f, 0x2e, 0x6f, 0x2f, 0x4f, 0x60, 0x1f,
+	0x2d, 0x0f, 0x0b, 0x00,
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+	0xd1, 0xf0, 0x02, 0x03,
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+	0x2d, 0x0c, 0x0f, 0x1f,
+	0x19, 0xee, 0x0f, 0x01, 0xef, 0x0b, 0x3f, 0x20, 0x2f, 0x0f, 0xdd, 0xbb,
+	0xff, 0xff, 0x00, 0xb0,
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+	0x0f, 0x40, 0x77, 0x21,
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+	0xae, 0xeb, 0xfc, 0x0f,
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+	0x20, 0x3f, 0x5d, 0x3a,
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+	0x01, 0xd0, 0x9e, 0xfd,
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+	0xe7, 0xf1, 0xf2, 0xe0,
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+	0x70, 0x2e, 0x30, 0x01,
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+	0xe0, 0x0f, 0xf1, 0x00,
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+	0xf3, 0xd4, 0xf5, 0xe6,
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+	0x32, 0x3f, 0x2b, 0x20,
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+	0xd4, 0xdf, 0x0c, 0x01,
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+	0x21, 0x0f, 0x2b, 0x0d,
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+	0x02, 0x03, 0x00, 0x70,
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+	0xbd, 0xeb, 0xe0, 0xf2,
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+	0xc0, 0xef, 0xf0, 0x06,
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+	0xf0, 0xf0, 0x02, 0x00,
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+	0x1f, 0x3a, 0x4f, 0x70,
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+	0xee, 0xd2, 0xd1, 0xc4,
+	0xf7, 0xc3, 0xe0, 0xc2, 0xe4, 0xe3, 0xf3, 0x02, 0x13, 0x60, 0x22, 0x30,
+	0x01, 0x14, 0x11, 0x00,
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+	0x7d, 0x70, 0x0e, 0xfe,
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+	0x4d, 0x0d, 0x1e, 0x3d,
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+	0xc0, 0xff, 0x00, 0x3e,
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+	0xf1, 0xd1, 0xa5, 0x02,
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+	0xe3, 0x00, 0xed, 0xfe,
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+	0x10, 0x3f, 0x67, 0x20,
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+	0xd6, 0xc1, 0xcf, 0xcc,
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+	0x30, 0x00, 0x31, 0x00,
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+	0xf0, 0x0d, 0x10, 0xa0,
+	0x91, 0xe5, 0xd0, 0xcf, 0xe1, 0x03, 0x00, 0xe1, 0x07, 0x04, 0xf1, 0xb1,
+	0x00, 0x00, 0x76, 0x00,
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+	0xf0, 0x12, 0x01, 0x20,
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+	0x01, 0xfb, 0xbe, 0xf1,
+	0xf1, 0x04, 0x11, 0x1f, 0x1a, 0xfd, 0x00, 0x2b, 0xfc, 0x30, 0x30, 0xbb,
+	0x00, 0x7f, 0x10, 0x0d,
+	0x00, 0x31, 0x21, 0xf2, 0xe1, 0xd3, 0x21, 0x3c, 0x0e, 0x70, 0x62, 0x00,
+	0x09, 0xe1, 0x02, 0x12,
+	0xe1, 0xe3, 0xf6, 0xf0, 0xef, 0x30, 0x57, 0xf3, 0xb1, 0xff, 0xf0, 0xe0,
+	0x03, 0x13, 0x21, 0x3d,
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+	0x01, 0x17, 0xd1, 0xaf,
+	0xc0, 0xe0, 0xe4, 0xb0, 0xdd, 0xc0, 0x03, 0xd3, 0xf0, 0x00, 0xff, 0x00,
+	0x44, 0x02, 0x02, 0x27,
+	0x67, 0x03, 0x21, 0x22, 0x20, 0x00, 0xf1, 0xd2, 0x0f, 0x0c, 0x30, 0x34,
+	0x20, 0x4b, 0x2d, 0x2f,
+	0xcd, 0x0f, 0x23, 0xc0, 0x9c, 0xf0, 0xdf, 0xe0, 0xfa, 0x1f, 0x3f, 0xff,
+	0xcc, 0x00, 0x20, 0x54,
+	0x33, 0x20, 0x1f, 0x41, 0x72, 0x67, 0x16, 0x00, 0x0e, 0xd0, 0xf1, 0xd1,
+	0x12, 0x10, 0x0c, 0x3e,
+	0x4f, 0x4e, 0x1c, 0x2f, 0x40, 0x31, 0x32, 0x0c, 0xfa, 0x60, 0x22, 0xf1,
+	0xc2, 0xd4, 0xb4, 0xd3,
+	0xc3, 0xd5, 0xf2, 0xf1, 0x20, 0x10, 0x64, 0x02, 0xee, 0x0c, 0xfd, 0x0e,
+	0xf0, 0x01, 0xcc, 0xcb,
+	0xff, 0xf0, 0x0d, 0x0e, 0xee, 0xc1, 0x02, 0x64, 0x32, 0x63, 0x15, 0x2f,
+	0x1e, 0x29, 0x1f, 0x0f,
+	0x0c, 0xed, 0x2f, 0x73, 0x0b, 0xcb, 0x02, 0x43, 0xe1, 0xe0, 0x05, 0x11,
+	0xe0, 0xd0, 0xb2, 0xe2,
+	0xdf, 0xcd, 0xed, 0xe0, 0xef, 0x14, 0x1d, 0xe9, 0x10, 0x62, 0x20, 0x63,
+	0x32, 0x2f, 0x1c, 0x20,
+	0x20, 0x3f, 0xfb, 0xf0, 0x10, 0x40, 0x2d, 0x29, 0x10, 0x40, 0x00, 0xd1,
+	0xe1, 0xc0, 0xd0, 0x02,
+	0x04, 0x30, 0x37, 0xf7, 0xb3, 0xcf, 0xb0, 0xe1, 0xd3, 0xd1, 0xc0, 0xf1,
+	0x00, 0xfc, 0xff, 0x21,
+	0x42, 0x0e, 0x1a, 0x6f, 0x0e, 0xef, 0xcd, 0x01, 0x00, 0x91, 0x07, 0xf2,
+	0xfe, 0xec, 0x11, 0x24,
+	0x50, 0x6f, 0x1e, 0x2f, 0x0f, 0x10, 0x00, 0xe0, 0xfd, 0xfd, 0xb8, 0xee,
+	0xde, 0xf0, 0x02, 0x00,
+	0x1a, 0x30, 0x70, 0x5f, 0x1f, 0x0f, 0x1b, 0x00, 0x00, 0xb1, 0x01, 0x10,
+	0x0e, 0xc2, 0x07, 0xe3,
+	0xfe, 0xba, 0x10, 0x30, 0x30, 0x7f, 0x30, 0x12, 0x20, 0x00, 0x21, 0x0f,
+	0x0c, 0xce, 0xff, 0xec,
+	0xda, 0xbe, 0xdf, 0xc4, 0xd6, 0xb1, 0xe2, 0xd0, 0x00, 0x01, 0x12, 0x7f,
+	0x2f, 0x0e, 0x20, 0x10,
+	0x26, 0x21, 0x00, 0x1d, 0x1c, 0x2d, 0x0d, 0x1b, 0x0c, 0x10, 0xff, 0xcf,
+	0xf2, 0xc4, 0xc3, 0xf6,
+	0xd0, 0xb0, 0xd1, 0xa3, 0xd0, 0xf2, 0xe0, 0xc0, 0x00, 0x10, 0x0b, 0x0e,
+	0x12, 0x17, 0x00, 0xf0,
+	0x13, 0x25, 0xe3, 0xe2, 0x22, 0x40, 0x00, 0xf0, 0x73, 0x40, 0x2b, 0x2f,
+	0x10, 0x1f, 0xfb, 0xbf,
+	0x3f, 0x40, 0x0e, 0x0f, 0x5d, 0xf8, 0xba, 0x00, 0xfe, 0x00, 0xf0, 0xef,
+	0x0f, 0x0e, 0x70, 0x4e,
+	0x3d, 0x00, 0x24, 0x03, 0x0c, 0xdc, 0xe0, 0x16, 0x23, 0x00, 0xff, 0xd0,
+	0x11, 0x11, 0xe1, 0x43,
+	0x74, 0x72, 0x20, 0x1f, 0x1e, 0x1d, 0x3d, 0x04, 0xf2, 0xcf, 0xd0, 0xc3,
+	0xe7, 0xc1, 0xd1, 0xde,
+	0xef, 0xd0, 0xee, 0x10, 0xfc, 0xee, 0x44, 0x44, 0x25, 0x22, 0xe0, 0x90,
+	0xef, 0xf2, 0xc6, 0xc1,
+	0xb0, 0xf1, 0xd7, 0xf1, 0x01, 0x00, 0x40, 0x31, 0x40, 0x3b, 0x3e, 0x1f,
+	0x40, 0x1e, 0x1f, 0xf0,
+	0xef, 0x13, 0x57, 0x1c, 0x40, 0x66, 0x00, 0xdb, 0x01, 0x04, 0x0f, 0xae,
+	0x0f, 0x01, 0x00, 0xef,
+	0xde, 0x00, 0xd0, 0xcf, 0xdc, 0xf1, 0xf1, 0xf3, 0x60, 0x3f, 0x39, 0x0f,
+	0x00, 0x21, 0x31, 0x41,
+	0x32, 0x51, 0x1f, 0x3b, 0x2f, 0x2e, 0x5e, 0x21, 0x0d, 0xc8, 0x1c, 0x50,
+	0xfe, 0x0d, 0x2f, 0x0f,
+	0xef, 0xcf, 0xf1, 0xe5, 0xc0, 0xd2, 0xd7, 0xe2, 0xd2, 0xdf, 0x1f, 0x71,
+	0x33, 0x21, 0x0f, 0x1f,
+	0x1d, 0x1e, 0xf0, 0x91, 0xd2, 0xd2, 0xc7, 0xd3, 0xce, 0xfe, 0xf6, 0x03,
+	0x03, 0x00, 0xed, 0x30,
+	0x74, 0x7e, 0x1c, 0x0e, 0x0f, 0x00, 0xfe, 0xe5, 0xf7, 0x04, 0xe2, 0xe1,
+	0x12, 0x10, 0x20, 0x00,
+	0x22, 0xfb, 0xff, 0x20, 0x0d, 0x3c, 0x72, 0x50, 0xfb, 0xe0, 0x0f, 0xe0,
+	0xb2, 0xed, 0x0e, 0x01,
+	0x0f, 0x3d, 0x6e, 0x18, 0x2f, 0x0e, 0x3e, 0x70, 0x1d, 0x0f, 0x0f, 0x0e,
+	0x00, 0x02, 0x0e, 0x20,
+	0x25, 0x02, 0x14, 0xe0, 0xc3, 0xf3, 0x17, 0xf1, 0x80, 0xf1, 0xf5, 0x02,
+	0x0e, 0xdd, 0x01, 0xe1,
+	0xf0, 0x1b, 0x70, 0x7f, 0x0a, 0x40, 0x2d, 0x1b, 0x2f, 0x11, 0x00, 0xee,
+	0xff, 0xd0, 0xb0, 0xf0,
+	0xe5, 0xe1, 0xd0, 0xbc, 0xde, 0xf4, 0xc2, 0xef, 0x0f, 0x20, 0x42, 0x00,
+	0x0f, 0x03, 0xd1, 0xe7,
+	0x07, 0xd2, 0xb2, 0xf2, 0xf4, 0x02, 0x40, 0x1f, 0x12, 0x41, 0xf0, 0x4f,
+	0x50, 0x0e, 0x3f, 0x54,
+	0x00, 0xff, 0x01, 0xef, 0xf9, 0xdf, 0xf2, 0xb1, 0xc8, 0xfe, 0x00, 0x30,
+	0x2e, 0x0f, 0x50, 0x2f,
+	0xfd, 0xbd, 0xff, 0xff, 0xe1, 0x03, 0x06, 0x02, 0x14, 0xf2, 0x04, 0x04,
+	0x91, 0x00, 0x44, 0x00,
+	0x5c, 0xe4, 0x1c, 0x04, 0x9b, 0xe5, 0xbf, 0x02, 0xb4, 0xe3, 0xe4, 0xf1,
+	0x02, 0x10, 0x30, 0x5d,
+	0x4d, 0x3b, 0x5b, 0x2c, 0x3d, 0x3b, 0x3d, 0x4c, 0x2d, 0x3d, 0x3e, 0x50,
+	0x30, 0x42, 0x31, 0x41,
+	0x32, 0x42, 0x22, 0x32, 0x41, 0x23, 0x34, 0x32, 0x22, 0x13, 0x13, 0x17,
+	0x06, 0x03, 0x03, 0x05,
+	0x02, 0x14, 0x13, 0x02, 0x04, 0x02, 0xe3, 0xd2, 0xb1, 0xd1, 0xe1, 0x01,
+	0x10, 0x71, 0x31, 0x55,
+	0x55, 0x33, 0x43, 0x44, 0x33, 0x43, 0x33, 0x45, 0x33, 0x24, 0x24, 0x12,
+	0x03, 0xf4, 0xc2, 0xd4,
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+	0xd2, 0xc2, 0xd2, 0xd2,
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+	0x34, 0xf0, 0xdf, 0x01,
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+	0x21, 0x53, 0x00, 0xdc,
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+	0x21, 0xeb, 0xfe, 0x42,
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+	0x8b, 0x8b, 0x00, 0x13,
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+	0xfe, 0xff, 0x00, 0xff,
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+	0x0e, 0xcc, 0xee, 0x57,
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+	0x15, 0xf0, 0xf0, 0x00,
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+	0x22, 0xd0, 0x8c, 0xee,
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+	0xef, 0x9a, 0x00, 0x23,
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+	0x2f, 0x1f, 0x01, 0x01,
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+	0x12, 0xef, 0xd0, 0x33,
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+	0xee, 0x1d, 0x2b, 0x0d,
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+	0x37, 0x03, 0xbd, 0xff,
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+	0xeb, 0x2f, 0x70, 0x21,
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+	0x00, 0x30, 0x00, 0xe2,
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+	0x76, 0x22, 0xc0, 0xf3,
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+	0xef, 0x0d, 0x00, 0xe1,
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+	0xef, 0xce, 0xe0, 0x02,
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+	0xb3, 0xe1, 0xfd, 0x0d,
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+	0xf0, 0xf3, 0x00, 0xf0,
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+	0x1e, 0x0e, 0x56, 0x01,
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+	0x33, 0x20, 0xcc, 0x23,
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+	0x0d, 0x10, 0x27, 0x23,
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+	0x45, 0x0f, 0x0f, 0x63,
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+	0x1f, 0x10, 0x26, 0x32,
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+	0x75, 0x10, 0x0a, 0xfe,
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+	0x0e, 0x0b, 0x41, 0x12,
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+	0x01, 0x0e, 0xfc, 0xf1,
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+	0xce, 0x24, 0x54, 0x0e,
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+	0x01, 0xa0, 0xf0, 0x14,
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+	0x00, 0x1d, 0x30, 0xf2,
+	0xc1, 0x26, 0x72, 0x3f, 0xda, 0xcf, 0xff, 0x30, 0x11, 0xf0, 0x03, 0x32,
+	0x2c, 0x09, 0x0f, 0x0f,
+	0xb0, 0xf1, 0x0f, 0x70, 0xfe, 0xcd, 0x01, 0x00, 0xed, 0x37, 0x74, 0x00,
+	0xee, 0x00, 0x3f, 0x00,
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+	0x40, 0x01, 0xe0, 0xc1,
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+	0x0f, 0x0e, 0x34, 0x32,
+	0xdd, 0x0e, 0x3f, 0x3f, 0xf0, 0xf2, 0xe3, 0x8b, 0x00, 0xe3, 0xdd, 0xda,
+	0x01, 0x13, 0x00, 0x3d,
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+	0xd3, 0xc0, 0x2c, 0x3f,
+	0xcc, 0xe0, 0x03, 0xd0, 0x0c, 0x0f, 0xdb, 0xdf, 0x60, 0x51, 0x00, 0xeb,
+	0x00, 0xe0, 0xdf, 0x10,
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+	0xbd, 0x98, 0x00, 0x00,
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+	0xf0, 0x00, 0x0d, 0xef,
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+	0xed, 0xef, 0x02, 0x03,
+	0xd0, 0x0f, 0x01, 0xc5, 0xe5, 0xc0, 0x8e, 0xf0, 0xf3, 0xc0, 0xf0, 0xf3,
+	0xdc, 0xdd, 0x25, 0x11,
+	0xdf, 0xee, 0xf1, 0xf2, 0x12, 0x3c, 0x3d, 0xde, 0x80, 0xf0, 0x17, 0x72,
+	0xed, 0xd8, 0x10, 0x42,
+	0xf0, 0xde, 0x0e, 0x01, 0xbf, 0x10, 0x40, 0xec, 0xf1, 0x00, 0x0b, 0x3e,
+	0x1e, 0x10, 0x10, 0xfd,
+	0xdf, 0x01, 0x33, 0x3f, 0x88, 0x0e, 0x62, 0x14, 0xfe, 0xcc, 0xff, 0x0e,
+	0x21, 0x02, 0xe0, 0xfb,
+	0x30, 0x0d, 0xff, 0x0f, 0x0e, 0xdc, 0xdf, 0x00, 0x23, 0xf0, 0x89, 0x8e,
+	0x0f, 0x00, 0x13, 0x02,
+	0xff, 0x0e, 0x01, 0x04, 0x02, 0x10, 0x40, 0x00, 0xa8, 0x00, 0x35, 0x00,
+	0xbf, 0x00, 0x01, 0x02,
+	0xdf, 0xff, 0x11, 0x00, 0x23, 0x04, 0x9d, 0xec, 0x07, 0x53, 0xff, 0x0f,
+	0xf0, 0xe0, 0xe0, 0x22,
+	0x62, 0x12, 0x0f, 0xfd, 0x00, 0x43, 0x3f, 0x1c, 0x00, 0x24, 0x10, 0x2d,
+	0x30, 0x0e, 0x0c, 0xd2,
+	0x12, 0x41, 0x30, 0x08, 0xbd, 0x00, 0x75, 0x40, 0x0c, 0x00, 0x11, 0x13,
+	0x22, 0x23, 0x1d, 0x1a,
+	0x30, 0x20, 0x1f, 0x40, 0x44, 0xcf, 0xeb, 0x31, 0x50, 0xfd, 0xf0, 0xf0,
+	0x1f, 0x0f, 0x01, 0x03,
+	0x00, 0xbf, 0x00, 0x00, 0x47, 0x57, 0xe0, 0xdc, 0x21, 0x40, 0x01, 0xf1,
+	0x01, 0x0d, 0x60, 0x04,
+	0xe0, 0xe0, 0x03, 0x35, 0xf0, 0xac, 0x00, 0x13, 0x0f, 0xe1, 0x14, 0xed,
+	0xbe, 0x11, 0x32, 0x0d,
+	0xe0, 0x07, 0xbf, 0xec, 0x33, 0x11, 0xce, 0x00, 0x52, 0xf1, 0xe0, 0x03,
+	0xf0, 0xea, 0x03, 0x36,
+	0xff, 0xde, 0x40, 0x31, 0xf1, 0xf2, 0x13, 0x2f, 0x1f, 0x52, 0x00, 0x03,
+	0x07, 0x61, 0x20, 0xfe,
+	0xf2, 0x17, 0x31, 0x4f, 0x00, 0xe0, 0xb3, 0x42, 0x43, 0xf0, 0xce, 0x10,
+	0x53, 0x04, 0xf4, 0x17,
+	0x21, 0xfe, 0x00, 0x65, 0x00, 0xe0, 0x02, 0x33, 0xf0, 0xc1, 0x02, 0x11,
+	0x10, 0x05, 0x00, 0xb0,
+	0x0f, 0xf0, 0x15, 0x52, 0x00, 0xcc, 0x01, 0x42, 0x03, 0x27, 0x00, 0x10,
+	0x0e, 0xe0, 0x23, 0x05,
+	0xe0, 0xc2, 0xe0, 0x00, 0x73, 0x23, 0xbe, 0xf0, 0x13, 0x12, 0x00, 0x02,
+	0xee, 0xd8, 0x0f, 0x13,
+	0xa0, 0xee, 0x0f, 0xdf, 0x20, 0x77, 0xff, 0xdd, 0x22, 0x20, 0x21, 0x24,
+	0x0e, 0xfb, 0x10, 0x52,
+	0x10, 0x21, 0xde, 0xab, 0x10, 0x56, 0x20, 0xec, 0xff, 0x10, 0x40, 0x0e,
+	0xf2, 0x03, 0xed, 0xec,
+	0x10, 0x40, 0xe0, 0xa0, 0x0f, 0xfd, 0x00, 0x07, 0x22, 0xed, 0xfa, 0x50,
+	0x21, 0x0f, 0x2e, 0x1f,
+	0x1f, 0x0f, 0x20, 0x51, 0xea, 0xbc, 0x0f, 0x00, 0x00, 0x12, 0x00, 0xcd,
+	0xfa, 0x00, 0x61, 0x11,
+	0x00, 0x00, 0x0f, 0x0f, 0x21, 0x23, 0x20, 0x31, 0x00, 0xfd, 0x13, 0x24,
+	0xc9, 0xde, 0x30, 0x01,
+	0x90, 0x0f, 0xed, 0x9c, 0xfd, 0x00, 0x11, 0xde, 0xcd, 0xfd, 0x23, 0x64,
+	0xfa, 0xcb, 0x00, 0x1e,
+	0x2e, 0x00, 0x14, 0xee, 0xed, 0x11, 0x61, 0x00, 0x01, 0x07, 0x0f, 0xcd,
+	0xff, 0x1f, 0x1d, 0x10,
+	0xf2, 0x91, 0x0d, 0x08, 0x0e, 0x0f, 0x20, 0x0f, 0xec, 0xf1, 0x20, 0x2f,
+	0x1c, 0x20, 0xdf, 0xf0,
+	0xdd, 0xbc, 0x00, 0x0d, 0xda, 0xae, 0x2f, 0x51, 0x0f, 0xcd, 0x0f, 0x11,
+	0x00, 0x0f, 0x32, 0x34,
+	0xc0, 0xeb, 0x10, 0x27, 0xe6, 0x80, 0x01, 0x30, 0x00, 0x01, 0x02, 0xb0,
+	0x00, 0x04, 0xe2, 0x02,
+	0x02, 0x10, 0xe2, 0xd2, 0x01, 0x3f, 0x24, 0x17, 0xe5, 0x8f, 0xff, 0x01,
+	0x33, 0x02, 0xe3, 0x01,
+	0x32, 0x90, 0xdf, 0xf3, 0x03, 0xd0, 0x02, 0xf7, 0xd2, 0xf2, 0xcf, 0x1f,
+	0x74, 0x32, 0xf0, 0x33,
+	0x46, 0xe0, 0xc0, 0x1f, 0x40, 0xf2, 0x04, 0xf1, 0xfd, 0x11, 0x77, 0x11,
+	0x00, 0x1e, 0x00, 0x57,
+	0x00, 0xee, 0x31, 0x00, 0xcd, 0x50, 0x13, 0xee, 0xe0, 0x07, 0xe0, 0xfd,
+	0x1e, 0x70, 0x13, 0xf0,
+	0x00, 0x10, 0x00, 0xdd, 0x67, 0x22, 0xdf, 0x00, 0x51, 0x02, 0xe0, 0x00,
+	0xff, 0x0f, 0x21, 0x17,
+	0x11, 0xed, 0x1f, 0x20, 0x02, 0xf3, 0x01, 0xfe, 0x8c, 0x0f, 0x27, 0x05,
+	0x80, 0x00, 0x00, 0x0f,
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+	0x22, 0x10, 0xce, 0x01,
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+	0x04, 0x41, 0x00, 0x0d,
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+	0x33, 0x41, 0x1f, 0x21,
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+	0x1e, 0x4e, 0x2d, 0x20,
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+	0x00, 0x10, 0xec, 0xe0,
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+	0xff, 0x1a, 0x3c, 0xfc,
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+	0x0e, 0xed, 0x3e, 0x70,
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+	0x00, 0x21, 0x10, 0x2d,
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+	0x0f, 0x0e, 0x1f, 0xcf,
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+	0x0d, 0xb0, 0xe0, 0xf0,
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+	0xd0, 0x0e, 0x20, 0x63,
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+	0x21, 0x00, 0x72, 0x14,
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+	0x75, 0x51, 0xe0, 0xd0,
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+	0xc2, 0x00, 0x32, 0x00,
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+	0x0f, 0x73, 0x72, 0x32,
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+	0xb0, 0x11, 0x47, 0x22,
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+	0x11, 0x11, 0x50, 0x33,
+	0x14, 0xff, 0x0f, 0x42, 0x72, 0x13, 0xff, 0xdc, 0xf3, 0x17, 0x05, 0x05,
+	0x19, 0x00, 0x1e, 0x00,
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+	0x03, 0xaf, 0x0e, 0x34,
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+	0x09, 0x0e, 0x70, 0x73,
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+	0x1d, 0x0b, 0x10, 0x30,
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+	0x1d, 0x1c, 0x2e, 0x2e,
+	0x3e, 0x4e, 0xfc, 0xef, 0xdc, 0x0c, 0x7e, 0x41, 0xef, 0xda, 0x00, 0xd0,
+	0xdd, 0xcd, 0x0c, 0x00,
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+	0x40, 0xf1, 0xc0, 0xee,
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+	0xdf, 0xcf, 0x0f, 0x5d,
+	0xf8, 0xdd, 0xfe, 0x14, 0x04, 0xb0, 0xdb, 0xf0, 0xf2, 0xf0, 0xff, 0xa0,
+	0xc0, 0xe0, 0x10, 0x11,
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+	0x46, 0xcc, 0x19, 0x70,
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+	0xed, 0xcf, 0x01, 0x02,
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+	0xae, 0xff, 0xf2, 0xf5,
+	0xee, 0xff, 0xff, 0x02, 0xd0, 0x0e, 0x2f, 0x3f, 0x00, 0x06, 0xe3, 0x9f,
+	0x0a, 0x62, 0x32, 0xeb,
+	0xf1, 0x37, 0xb0, 0xcb, 0x22, 0x22, 0x9e, 0xfe, 0x03, 0x04, 0xff, 0xee,
+	0x01, 0x03, 0x00, 0x11,
+	0x00, 0x04, 0xf0, 0x10, 0x10, 0x00, 0xd3, 0x02, 0x01, 0xff, 0x00, 0x63,
+	0xf3, 0xbe, 0x00, 0x13,
+	0x26, 0x00, 0xcf, 0x20, 0x42, 0x2f, 0x20, 0xfa, 0x20, 0x77, 0x0f, 0x1e,
+	0x40, 0x23, 0x1f, 0x2f,
+	0x64, 0x02, 0x10, 0x00, 0x30, 0x36, 0x10, 0x1f, 0x20, 0xf3, 0x02, 0x2c,
+	0x4f, 0x31, 0x00, 0x0f,
+	0x00, 0x10, 0x5e, 0x3b, 0x30, 0x52, 0x10, 0x20, 0x0d, 0x0b, 0x70, 0x23,
+	0x26, 0x12, 0x20, 0x0e,
+	0x30, 0x52, 0x11, 0x04, 0x01, 0xf0, 0x00, 0x25, 0x20, 0x10, 0x00, 0xd7,
+	0x04, 0x71, 0x70, 0x01,
+	0x13, 0xf0, 0xf1, 0x06, 0x01, 0xf1, 0x21, 0x00, 0xde, 0xd0, 0xf4, 0xf0,
+	0xef, 0x5f, 0x53, 0x25,
+	0x10, 0x1f, 0x30, 0x51, 0x2e, 0x2d, 0x1f, 0x10, 0x17, 0x30, 0x30, 0x0e,
+	0xef, 0x33, 0x66, 0x31,
+	0x30, 0x40, 0x61, 0x10, 0x1f, 0x01, 0x10, 0x62, 0x34, 0x00, 0x2c, 0x0f,
+	0x36, 0xf1, 0xdc, 0x3f,
+	0x70, 0x13, 0xf0, 0xf0, 0x0f, 0x3d, 0x23, 0x27, 0x03, 0xed, 0xdc, 0x41,
+	0x52, 0x01, 0x0c, 0x0e,
+	0xe2, 0x05, 0x30, 0x3e, 0x40, 0x73, 0x54, 0x0d, 0xdd, 0x03, 0x27, 0x00,
+	0x01, 0x00, 0xee, 0x91,
+	0xf6, 0x01, 0x20, 0x1f, 0xe2, 0xf6, 0x00, 0x20, 0x4f, 0x50, 0x1e, 0x00,
+	0xf0, 0x00, 0x1d, 0x11,
+	0x21, 0xaf, 0xdc, 0xee, 0x00, 0xf1, 0xf0, 0xbf, 0xdc, 0xe2, 0x00, 0x10,
+	0x1e, 0x00, 0x25, 0x1b,
+	0xca, 0xb0, 0x57, 0x33, 0xcb, 0x0f, 0x72, 0x02, 0xff, 0xde, 0x00, 0x21,
+	0x01, 0xef, 0x00, 0x20,
+	0x01, 0xf0, 0xf1, 0xff, 0x1f, 0x00, 0x01, 0x14, 0x62, 0x0e, 0x1d, 0x52,
+	0x10, 0x04, 0xd5, 0x21,
+	0x51, 0x20, 0x1e, 0x11, 0xf7, 0xc1, 0x0f, 0x0c, 0xf0, 0x02, 0xf1, 0xaf,
+	0xfd, 0x01, 0x25, 0x0f,
+	0x2a, 0x50, 0x00, 0x01, 0x22, 0x43, 0x3e, 0x3f, 0x55, 0x01, 0x11, 0x23,
+	0x0e, 0x0c, 0x32, 0x05,
+	0xc2, 0xbf, 0xff, 0x01, 0x3e, 0xde, 0xa4, 0x04, 0xff, 0x2a, 0x30, 0x20,
+	0xd0, 0x9f, 0xdc, 0xee,
+	0x0e, 0x11, 0x24, 0xb0, 0xcf, 0x0e, 0x00, 0x0e, 0xfe, 0xbd, 0x02, 0xe7,
+	0xf0, 0xfc, 0xdc, 0x0f,
+	0x0f, 0x2f, 0x20, 0x41, 0x23, 0xde, 0xc8, 0xf0, 0xe0, 0x00, 0x00, 0x2c,
+	0xc0, 0xac, 0xd0, 0xc0,
+	0xc0, 0xde, 0xef, 0xef, 0xee, 0xd0, 0xfd, 0xfa, 0xbf, 0xff, 0x00, 0xea,
+	0x9d, 0x0e, 0x20, 0x21,
+	0xcc, 0xbc, 0xfe, 0xee, 0xf0, 0xc2, 0xfe, 0xfa, 0x0f, 0xed, 0x0d, 0x10,
+	0x11, 0xf0, 0xce, 0x0c,
+	0x2e, 0x2f, 0x1e, 0xdd, 0x8f, 0xd6, 0xf4, 0x00, 0xff, 0xd2, 0xc1, 0xc1,
+	0xe7, 0xd2, 0x33, 0x11,
+	0x91, 0xe6, 0xe0, 0xbe, 0xf0, 0xf1, 0xf2, 0xe0, 0xd0, 0xc0, 0xf2, 0x00,
+	0x0e, 0x1e, 0x0c, 0xe3,
+	0xd7, 0xc0, 0xed, 0xec, 0x20, 0x13, 0x81, 0xaf, 0xf0, 0x0c, 0x00, 0x23,
+	0xf0, 0xd0, 0x0c, 0x3a,
+	0x4e, 0x1e, 0x0d, 0xfd, 0x00, 0x2f, 0x0a, 0x1e, 0xce, 0xbd, 0xfe, 0xf8,
+	0xee, 0xef, 0xcf, 0xcd,
+	0x0f, 0x0f, 0x9e, 0xec, 0x0e, 0x1d, 0x40, 0x22, 0x00, 0xdc, 0x0c, 0x3c,
+	0x6d, 0x0d, 0xfd, 0x0d,
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+	0x1f, 0xfe, 0xed, 0xbb,
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+	0x3d, 0x8d, 0xed, 0xf2,
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+	0xf2, 0xfe, 0x1b, 0x4d,
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+	0xce, 0xaf, 0xe5, 0x03,
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+	0x20, 0x31, 0xf2, 0xb1,
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+	0x13, 0x0a, 0x01, 0x11,
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+	0x05, 0xaf, 0x00, 0x15,
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+	0x00, 0x50, 0x0e, 0x11,
+	0x02, 0x04, 0x34, 0xcd, 0xcd, 0x37, 0x37, 0xa0, 0xe0, 0xf0, 0xf0, 0x01,
+	0x01, 0xc1, 0xd1, 0x02,
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+	0x20, 0x60, 0x14, 0xdc,
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+	0x36, 0x00, 0xee, 0x0f,
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+	0xf0, 0xd0, 0xf1, 0xf2,
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+	0x40, 0x41, 0x00, 0xec,
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+	0xd2, 0xe0, 0xdf, 0xac,
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+	0x4e, 0x1d, 0x0d, 0x0d,
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+	0x0d, 0x0a, 0x3f, 0x4f,
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+	0xaf, 0xe2, 0xc1, 0xfd,
+	0xed, 0xef, 0xb0, 0xf6, 0xf1, 0x0c, 0x1e, 0x50, 0x00, 0x00, 0xe1, 0xea,
+	0xfd, 0xff, 0xad, 0xdc,
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+	0x8b, 0x00, 0x02, 0x00,
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+	0xce, 0xbf, 0xf1, 0xdf,
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+	0x21, 0x0e, 0x08, 0xff,
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+	0xfe, 0xdc, 0x0f, 0x4e,
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+	0x02, 0xd7, 0xb1, 0x11,
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+	0x20, 0x51, 0x31, 0xe1,
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+	0x4f, 0x3c, 0x51, 0x27,
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+	0x35, 0x05, 0x11, 0x01,
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+	0x02, 0x10, 0x03, 0x00,
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+	0x7f, 0x7f, 0x2e, 0x0d,
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+	0xd2, 0xf7, 0xe7, 0xe0,
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+	0xf3, 0xd5, 0xd3, 0xf3,
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+	0x0b, 0xed, 0xef, 0xdd,
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+	0x2f, 0x34, 0x14, 0x01,
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+	0x41, 0x16, 0x10, 0x61,
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+	0x25, 0x03, 0xe1, 0x01,
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+	0xe4, 0xd1, 0x91, 0xe4,
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+	0xd0, 0xcb, 0xdf, 0xb0,
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+	0x00, 0xec, 0xfa, 0xdf,
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+	0xbd, 0x0d, 0x00, 0xf0,
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+	0xef, 0xe0, 0xef, 0xf2,
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+	0xee, 0xce, 0xb0, 0xe1,
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+	0xcd, 0xd0, 0xbd, 0xb8,
+	0xef, 0xcf, 0xe0, 0xcf, 0xfd, 0xd9, 0x0f, 0x00, 0x10, 0x0f, 0xfd, 0xfb,
+	0x1f, 0x3f, 0x0e, 0xde,
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+	0xce, 0xfc, 0xfc, 0xee,
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+	0x1d, 0x10, 0xe0, 0xbe,
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+	0x2f, 0x31, 0x13, 0x2f,
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+	0xd0, 0x93, 0xe6, 0x01,
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+	0x20, 0x41, 0x34, 0x02,
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+	0x1e, 0x50, 0x27, 0x20,
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+	0xcf, 0xf0, 0xc0, 0xec,
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+	0x20, 0x0d, 0x4e, 0x4f,
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+	0xd2, 0x00, 0x50, 0x61,
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+	0x23, 0x33, 0x03, 0xc2,
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+	0x1f, 0x2f, 0x1f, 0x10,
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+	0xd1, 0xb2, 0xd5, 0xc5,
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+	0xd3, 0xc1, 0xd1, 0xf3,
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+	0xa1, 0xf3, 0x06, 0x34,
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+	0x32, 0x20, 0x00, 0x12,
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+	0x20, 0x7c, 0x2a, 0x4d,
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+	0xee, 0xfd, 0xdd, 0xed,
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+	0xc4, 0xe4, 0xc3, 0xc3,
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+	0xd3, 0xd2, 0xe1, 0xd1,
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+	0xc1, 0xd1, 0xf3, 0xd3,
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+	0xdd, 0xe0, 0xf3, 0xe1,
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+	0xcd, 0xde, 0x0d, 0x30,
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+	0x45, 0x0d, 0xdb, 0x1f,
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+	0x00, 0x00, 0xff, 0xdc,
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+	0x02, 0xf0, 0x15, 0x25,
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+	0x01, 0xd0, 0x23, 0xc0,
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+	0x22, 0x31, 0xce, 0xdd,
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+	0x0d, 0x0b, 0x00, 0x20,
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+	0x24, 0xf0, 0xcd, 0x01,
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+	0x02, 0xd0, 0xf0, 0x03,
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+	0x00, 0xf0, 0x22, 0x30,
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+	0x23, 0xef, 0xdc, 0x10,
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+	0x0d, 0xe8, 0x00, 0x50,
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+	0x0d, 0xdd, 0x0f, 0x30,
+	0xff, 0xca, 0xef, 0x1f, 0x10, 0x0e, 0x30, 0x52, 0xed, 0xf0, 0x66, 0x11,
+	0xdd, 0xfe, 0x34, 0x32,
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+	0xef, 0x04, 0x02, 0xad,
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+	0x01, 0x44, 0x01, 0xdf,
+	0xe0, 0x34, 0x33, 0xcf, 0xdf, 0x11, 0x02, 0xef, 0xee, 0x10, 0x43, 0x22,
+	0xfd, 0xed, 0x42, 0x57,
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+	0x11, 0xdd, 0x00, 0x66,
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+	0x0f, 0x00, 0x33, 0x00,
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+	0x11, 0x10, 0x22, 0x0e,
+	0x0c, 0x32, 0x76, 0x00, 0x0b, 0xdd, 0x00, 0x44, 0x01, 0x9b, 0xff, 0x04,
+	0x00, 0xcf, 0x01, 0x13,
+	0xe0, 0xe0, 0xf1, 0xf1, 0x03, 0x02, 0xf0, 0x00, 0x03, 0xe0, 0x03, 0x37,
+	0xd0, 0x8e, 0x15, 0x14,
+	0xf0, 0xcf, 0xf1, 0x16, 0xe0, 0x8f, 0xf0, 0x04, 0x01, 0xdf, 0xff, 0x11,
+	0x23, 0xee, 0x0f, 0x12,
+	0xfd, 0xfc, 0x50, 0x31, 0x1d, 0x3d, 0x3c, 0x09, 0x4f, 0x60, 0x10, 0x0d,
+	0x1d, 0x20, 0x20, 0xfa,
+	0x0f, 0x30, 0x50, 0xda, 0xee, 0x71, 0x00, 0x0d, 0xfc, 0x0f, 0x20, 0x41,
+	0x00, 0xe9, 0xff, 0x51,
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+	0x33, 0x22, 0xca, 0x00,
+	0x72, 0x21, 0xeb, 0x0f, 0x30, 0x21, 0xfd, 0xbb, 0x00, 0x33, 0x00, 0x9c,
+	0x00, 0x03, 0x00, 0xce,
+	0x02, 0x07, 0xe0, 0xbf, 0x01, 0xf4, 0xd0, 0xef, 0xe0, 0xe0, 0xef, 0xf0,
+	0x21, 0xcb, 0xff, 0x20,
+	0xbd, 0xdd, 0x02, 0x01, 0xcd, 0xed, 0x00, 0xca, 0xbb, 0xee, 0x03, 0x11,
+	0xb8, 0xff, 0x20, 0x31,
+	0x0f, 0x0e, 0x20, 0x50, 0x20, 0x20, 0x30, 0x20, 0x20, 0x30, 0x13, 0xff,
+	0xdd, 0x24, 0x13, 0x8c,
+	0x00, 0x27, 0x00, 0xad, 0x02, 0x37, 0x00, 0x00, 0x10, 0x00, 0x0f, 0x00,
+	0x75, 0x23, 0x00, 0xee,
+	0x01, 0x45, 0x10, 0x01, 0x20, 0x21, 0x42, 0x22, 0x00, 0xf0, 0x10, 0x11,
+	0xfd, 0xcd, 0xee, 0x47,
+	0x24, 0xae, 0xdf, 0xe0, 0x03, 0x03, 0xe2, 0xf4, 0xb0, 0xe0, 0xe0, 0xf0,
+	0x01, 0xd0, 0xbd, 0x02,
+	0x26, 0xba, 0xcd, 0x23, 0x13, 0xdb, 0xce, 0x00, 0x23, 0x10, 0xff, 0xfc,
+	0x20, 0x61, 0x00, 0x0b,
+	0x10, 0x30, 0x0f, 0x20, 0x63, 0xfd, 0xd8, 0x00, 0x52, 0x10, 0xff, 0xdc,
+	0xff, 0x23, 0x21, 0xde,
+	0xef, 0xff, 0xf0, 0x0f, 0x00, 0xf1, 0x9d, 0xf0, 0x02, 0xef, 0xdb, 0x01,
+	0x45, 0xff, 0xdd, 0xbc,
+	0xf0, 0x12, 0x10, 0xec, 0x0d, 0x30, 0x64, 0x10, 0xf9, 0x20, 0x50, 0x20,
+	0x2f, 0x0e, 0x51, 0x22,
+	0x12, 0x23, 0x36, 0x11, 0x11, 0x23, 0x12, 0x45, 0x34, 0x00, 0xe0, 0x43,
+	0x45, 0x00, 0x00, 0xe0,
+	0xdd, 0x74, 0x73, 0xfc, 0xfe, 0x13, 0x02, 0xff, 0xe0, 0x06, 0x02, 0xc0,
+	0xcf, 0x01, 0x06, 0xf0,
+	0xd0, 0x01, 0x13, 0x00, 0xdf, 0x23, 0x55, 0x00, 0xcc, 0x21, 0x33, 0xde,
+	0xbb, 0x00, 0x11, 0xff,
+	0xde, 0x10, 0x10, 0xdd, 0xf0, 0x53, 0x0e, 0x1e, 0x3f, 0x2e, 0x50, 0x30,
+	0x3d, 0x40, 0x52, 0x22,
+	0x22, 0x2c, 0x18, 0x43, 0x23, 0xfb, 0xff, 0x53, 0x10, 0x0e, 0x00, 0x00,
+	0xfc, 0x00, 0x04, 0xf0,
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+	0x10, 0x43, 0x34, 0x10,
+	0xff, 0x10, 0x76, 0x11, 0xf0, 0xff, 0x00, 0x04, 0x01, 0xd0, 0xe0, 0x15,
+	0x02, 0xc0, 0xe1, 0x02,
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+	0xe0, 0xe1, 0xd1, 0xe2,
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+	0xd0, 0x8f, 0x0f, 0x03,
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+	0xc0, 0xae, 0x0f, 0x12,
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+	0x53, 0x11, 0x43, 0x25,
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+	0xf2, 0xd1, 0x06, 0xe3,
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+	0xfa, 0xcd, 0x01, 0x47,
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+	0xdf, 0x03, 0xe0, 0x89,
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+	0xe1, 0xa0, 0xf3, 0xf4,
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+	0x40, 0x19, 0x1e, 0x30,
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+	0xee, 0x10, 0xef, 0x00,
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+	0xed, 0x0d, 0x31, 0x47,
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+	0x10, 0x77, 0xfe, 0xdd,
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+	0x01, 0xdd, 0xcd, 0x00,
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+	0x34, 0xff, 0x00, 0x21,
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+	0x00, 0x12, 0x21, 0x00,
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+	0x00, 0x01, 0xf0, 0xef,
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+	0xed, 0x10, 0x00, 0xcd,
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+	0x00, 0x34, 0x10, 0xff,
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+	0x43, 0xef, 0xf0, 0x45,
+	0x33, 0x24, 0xff, 0xff, 0x77, 0x20, 0xed, 0x31, 0x64, 0x0e, 0x00, 0x67,
+	0x21, 0x00, 0xcc, 0x0f,
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+	0xdc, 0xef, 0x21, 0x11,
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+	0x11, 0x00, 0xf1, 0x14,
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+	0xf0, 0xbd, 0xff, 0x45,
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+	0x23, 0x23, 0x11, 0xfe,
+	0x00, 0x77, 0x12, 0xec, 0x21, 0x77, 0x11, 0x11, 0x00, 0x23, 0x32, 0xff,
+	0xff, 0x53, 0x00, 0xb8,
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+	0x54, 0x32, 0xfe, 0xfc,
+	0x0e, 0x0f, 0xcb, 0xde, 0x10, 0x62, 0x32, 0x23, 0x32, 0xff, 0x88, 0xdc,
+	0x11, 0x21, 0xdf, 0xdd,
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+	0x57, 0x31, 0x01, 0x11,
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+	0x32, 0x22, 0x00, 0x00,
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+	0x20, 0x77, 0x00, 0xcc,
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+	0x10, 0xf0, 0xff, 0xbd,
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+	0xf1, 0x88, 0xce, 0x01,
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+	0x01, 0x12, 0x01, 0xde,
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+	0x01, 0xdc, 0x9a, 0xee,
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+	0x21, 0x10, 0xff, 0xbb,
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+	0x00, 0x00, 0x0f, 0xcc,
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+	0x00, 0x23, 0x02, 0xe0,
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+	0x22, 0x64, 0x00, 0xfe,
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+	0x31, 0x43, 0x0e, 0x0b,
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+	0xec, 0xdf, 0x01, 0x12,
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+	0x13, 0xdc, 0xce, 0x07,
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+	0x11, 0x10, 0x11, 0x07,
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+	0x00, 0xf0, 0x02, 0x10,
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+	0xe9, 0x0f, 0x10, 0xff,
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+	0xfc, 0xaf, 0xce, 0x00,
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+	0x04, 0x04, 0xe3, 0xa0,
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+	0xe0, 0xdc, 0xbd, 0xfd,
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+	0xee, 0xff, 0xef, 0xcd,
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+	0xcc, 0x0c, 0x0f, 0x0c,
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+	0xfe, 0xcd, 0x0c, 0x4d,
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+	0xc2, 0xb2, 0xd0, 0xb1,
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+	0x70, 0x10, 0x19, 0x0e,
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+	0x04, 0xf4, 0xd2, 0xe2,
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+	0x01, 0x21, 0x20, 0x20,
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+	0x03, 0xee, 0xfd, 0xc0,
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+	0x05, 0x52, 0x43, 0x31,
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+	0x1e, 0x10, 0x03, 0x17,
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+	0x42, 0x21, 0x22, 0x13,
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+	0xd2, 0xc0, 0xf1, 0x00,
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+	0xff, 0x0f, 0xef, 0xb0,
+	0xdc, 0xdd, 0x0e, 0xeb, 0xed, 0xdf, 0xbc, 0xac, 0xed, 0xde, 0xee, 0x00,
+	0x1c, 0xfc, 0xee, 0x10,
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+	0xfb, 0x0d, 0x2e, 0x10,
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+	0x7f, 0x5f, 0x10, 0x30,
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+	0xe1, 0xf5, 0xff, 0xee,
+	0xef, 0x0f, 0x0f, 0x4a, 0x20, 0x30, 0xfb, 0xae, 0x00, 0x0f, 0xe0, 0xc0,
+	0xcf, 0xed, 0xed, 0xdc,
+	0xfc, 0xdb, 0xd0, 0xd0, 0xe3, 0xdd, 0xcd, 0x00, 0x16, 0x31, 0x31, 0x00,
+	0xdb, 0x9f, 0x00, 0x21,
+	0x51, 0x31, 0x64, 0x23, 0x30, 0x40, 0x20, 0x30, 0x3b, 0x10, 0x23, 0x37,
+	0x01, 0xe0, 0x32, 0x5f,
+	0x30, 0x41, 0x47, 0x12, 0x13, 0x23, 0x25, 0x74, 0x53, 0x20, 0x41, 0x21,
+	0x33, 0x22, 0x21, 0x31,
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+	0x33, 0x13, 0x64, 0x23,
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+	0xf2, 0xe5, 0xd1, 0xd3,
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+	0xf0, 0x2f, 0x50, 0x00,
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+	0xbc, 0xde, 0xdd, 0xed,
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+	0x8a, 0xde, 0xff, 0xeb,
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+	0x1f, 0xdd, 0xca, 0x1e,
+	0x1e, 0x0c, 0x1d, 0x3b, 0x1f, 0xbf, 0xdc, 0x0f, 0x60, 0x10, 0xef, 0xbe,
+	0xff, 0xbd, 0xde, 0xce,
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+	0x0f, 0xef, 0xbe, 0xde,
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+	0xee, 0xed, 0xc0, 0xb0,
+	0xf0, 0xdf, 0xdf, 0xec, 0x0d, 0x4d, 0x1e, 0x10, 0x0d, 0xcc, 0xfe, 0xff,
+	0x0c, 0x20, 0x1d, 0xd9,
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+	0x10, 0x20, 0x5d, 0x3d,
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+	0x15, 0xf1, 0x9f, 0xee,
+	0xcf, 0xef, 0x02, 0x07, 0x01, 0xb0, 0xef, 0x00, 0xe0, 0x00, 0x00, 0x27,
+	0x24, 0x04, 0xb1, 0xc2,
+	0xf0, 0xd3, 0xe0, 0xd1, 0xc1, 0xf4, 0xd5, 0x91, 0xe0, 0xe0, 0xc1, 0xe0,
+	0x02, 0x07, 0x02, 0xa0,
+	0xd0, 0xe0, 0xef, 0xd0, 0x02, 0x02, 0x10, 0x0c, 0xe0, 0xf1, 0xf7, 0xd1,
+	0xdf, 0xcd, 0xa0, 0xde,
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+	0x2d, 0x4f, 0x2a, 0x1c,
+	0x0f, 0x3e, 0x1d, 0x1c, 0x2c, 0x0c, 0x1d, 0x3d, 0x3d, 0x5d, 0x50, 0x2e,
+	0x09, 0x0e, 0x3d, 0x2d,
+	0x3e, 0x5d, 0x0d, 0xfc, 0xcc, 0xfd, 0x0c, 0x7d, 0x7e, 0x2e, 0x1d, 0x1d,
+	0x10, 0x20, 0x0d, 0x1b,
+	0x00, 0xd3, 0xd0, 0xe3, 0x06, 0xe0, 0xdd, 0xdd, 0x02, 0x33, 0x01, 0xd6,
+	0xe4, 0x23, 0x70, 0x71,
+	0x74, 0x10, 0xf0, 0x01, 0xe5, 0x01, 0xf0, 0xe0, 0x02, 0x12, 0x22, 0x31,
+	0xe3, 0xc2, 0xe7, 0x12,
+	0xe0, 0xae, 0x04, 0x05, 0x02, 0x23, 0x61, 0x11, 0x00, 0x21, 0x36, 0x52,
+	0x24, 0x42, 0x32, 0x22,
+	0x32, 0x22, 0x21, 0x30, 0x5e, 0x2e, 0x44, 0x27, 0x21, 0x1f, 0x20, 0x24,
+	0x12, 0x33, 0x76, 0x42,
+	0x31, 0x01, 0x22, 0x53, 0x10, 0x40, 0x32, 0x34, 0x42, 0x14, 0x33, 0x11,
+	0x20, 0x10, 0x40, 0x27,
+	0x44, 0x12, 0x13, 0x14, 0x02, 0xd0, 0x02, 0x42, 0x54, 0x02, 0x45, 0x02,
+	0xe1, 0x02, 0x12, 0x14,
+	0x44, 0x34, 0x13, 0x04, 0x01, 0x04, 0x22, 0x02, 0x32, 0x44, 0x11, 0x12,
+	0x12, 0x43, 0x13, 0x22,
+	0x03, 0x33, 0x27, 0x24, 0x61, 0x33, 0x30, 0x02, 0x00, 0x71, 0x73, 0x73,
+	0x04, 0x03, 0x14, 0x00,
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+	0x1f, 0x2e, 0x24, 0x03,
+	0x3f, 0x6f, 0x53, 0x20, 0x3d, 0x30, 0x46, 0x00, 0x2b, 0x0c, 0x2d, 0x0b,
+	0x20, 0x52, 0x11, 0x1f,
+	0x1b, 0x0e, 0x1e, 0x4f, 0x1f, 0xfd, 0xcb, 0xed, 0x0d, 0x1f, 0x7f, 0x78,
+	0x3e, 0x0b, 0xef, 0x0f,
+	0x00, 0x0f, 0x00, 0xfe, 0x1f, 0x2f, 0x11, 0xe0, 0xb0, 0xdf, 0xcf, 0xdc,
+	0xd0, 0xf0, 0xff, 0x3d,
+	0x60, 0x00, 0x1a, 0x11, 0x00, 0xbb, 0xef, 0x00, 0x11, 0x00, 0x0d, 0x08,
+	0xde, 0xae, 0xcb, 0xff,
+	0xef, 0xed, 0xbe, 0xdd, 0xde, 0xcd, 0xb9, 0xef, 0xec, 0xdf, 0xee, 0xf0,
+	0xcd, 0xf0, 0xde, 0xcd,
+	0xdb, 0x00, 0x21, 0x31, 0x0f, 0x0c, 0xcd, 0xcd, 0xcb, 0xfd, 0x0e, 0x1d,
+	0x0e, 0xc0, 0xd1, 0xc0,
+	0xbb, 0xed, 0xfe, 0x02, 0x03, 0x00, 0x0f, 0xc1, 0xfe, 0xe8, 0x1f, 0x70,
+	0x50, 0x1d, 0x22, 0x12,
+	0x08, 0xfd, 0x00, 0x1f, 0x30, 0xf0, 0x0e, 0xf1, 0xf2, 0xf2, 0xef, 0xcb,
+	0xe0, 0xe0, 0xe0, 0x10,
+	0x51, 0x32, 0x1d, 0x0c, 0xef, 0xe6, 0x17, 0x32, 0x22, 0x81, 0xc0, 0xd2,
+	0x14, 0x31, 0xd1, 0x20,
+	0x44, 0x13, 0x05, 0xd2, 0xc0, 0xb0, 0xd2, 0x90, 0xf0, 0xf3, 0xe4, 0xef,
+	0x00, 0x25, 0x32, 0x01,
+	0x10, 0x32, 0x01, 0xb0, 0xbf, 0x0f, 0xde, 0x0e, 0xec, 0xdd, 0xa0, 0xd0,
+	0xef, 0xee, 0xdb, 0xba,
+	0xde, 0xcd, 0xe0, 0xf5, 0xed, 0xe9, 0x00, 0x10, 0x0d, 0xee, 0xee, 0xaf,
+	0xee, 0xfd, 0x1e, 0x5b,
+	0x0d, 0x0b, 0xfe, 0xdd, 0xbe, 0xcd, 0xe0, 0xdf, 0x0f, 0xfc, 0xdc, 0xed,
+	0xfd, 0x1b, 0xfb, 0xee,
+	0xbc, 0x0e, 0x0e, 0x5f, 0x72, 0x2f, 0x0c, 0x0e, 0x0e, 0xfe, 0xef, 0x00,
+	0xd3, 0xe0, 0xbc, 0xf0,
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+	0xe2, 0xed, 0xd9, 0xff,
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+	0xdf, 0x80, 0xd1, 0xf4,
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+	0x01, 0xed, 0x1c, 0x7f,
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+	0x2e, 0x1c, 0x00, 0xc0,
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+	0x10, 0x0e, 0x0c, 0xe0,
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+	0x0c, 0x1e, 0x10, 0x22,
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+	0xdd, 0xec, 0x2b, 0x7d,
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+	0x7c, 0x4e, 0x2d, 0x0c,
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+	0x05, 0xb3, 0xd3, 0xc2,
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+	0xf3, 0xf1, 0xd0, 0xb0,
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+	0x3e, 0x3d, 0x4f, 0x3f,
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+	0x20, 0x0e, 0xee, 0xbf,
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+	0x52, 0x23, 0x11, 0x12,
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+	0xee, 0x0b, 0x0d, 0x2b,
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+	0x1c, 0x1c, 0xfb, 0xcc,
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+	0x2f, 0x1f, 0x10, 0x1c,
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+	0xd2, 0xd1, 0xd1, 0xa2,
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+	0x03, 0xe1, 0xb2, 0xc3,
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+	0x12, 0x00, 0x02, 0x03,
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+	0x3d, 0x2c, 0x0d, 0xed,
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+	0xe0, 0xf0, 0x00, 0x34,
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+	0xc2, 0xd1, 0xb3, 0xe3,
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+	0xf1, 0xb1, 0xf0, 0x01,
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+	0x2e, 0x2c, 0x4b, 0x3d,
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+	0x3e, 0x2d, 0x0d, 0x02,
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+	0x41, 0x13, 0x36, 0x23,
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+	0x04, 0x03, 0x12, 0x00,
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+	0xe0, 0xff, 0xfd, 0x00,
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+	0x51, 0x77, 0x10, 0xee,
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+	0x1e, 0x0e, 0x0e, 0x20,
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+	0xec, 0x0e, 0x30, 0x11,
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+	0xe0, 0xa0, 0xd0, 0xf5,
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+	0xf0, 0xd0, 0xb3, 0xe3,
+	0xe6, 0xf2, 0xc1, 0xd1, 0xa0, 0xe0, 0x02, 0x01, 0x01, 0x22, 0x01, 0xff,
+	0xdc, 0xca, 0x11, 0x77,
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+	0x22, 0x00, 0xbc, 0xba,
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+	0x0d, 0xfb, 0x00, 0x21,
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+	0xef, 0xfd, 0x50, 0x3d,
+	0x61, 0x27, 0xff, 0x00, 0x24, 0xed, 0xf0, 0x14, 0xb0, 0xc0, 0x27, 0xf1,
+	0xaf, 0xff, 0xcd, 0x00,
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+	0xe1, 0xe1, 0xc0, 0xc0,
+	0xdf, 0xbc, 0x13, 0x26, 0xef, 0xdf, 0x01, 0x12, 0x10, 0x1f, 0x00, 0xd0,
+	0x04, 0x74, 0x41, 0x0f,
+	0xd9, 0x51, 0x77, 0x00, 0xef, 0x00, 0x00, 0xf0, 0x00, 0x00, 0x00, 0x10,
+	0xfd, 0xcb, 0x10, 0x22,
+	0xef, 0xec, 0x30, 0x50, 0xfd, 0xec, 0x20, 0x51, 0x0e, 0x0e, 0x21, 0x0f,
+	0xfa, 0x20, 0x72, 0x0f,
+	0xb9, 0x00, 0x33, 0x11, 0xdd, 0xee, 0x11, 0x31, 0x00, 0x0f, 0xfd, 0x00,
+	0x54, 0xfe, 0xef, 0x11,
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+	0x10, 0x10, 0x54, 0x12,
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+	0x02, 0x00, 0x0f, 0x00,
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+	0xde, 0xe0, 0x14, 0x00,
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+	0x20, 0x40, 0x0d, 0x0d,
+	0x00, 0x41, 0x21, 0xfc, 0xdc, 0x0e, 0x73, 0x56, 0x0f, 0xec, 0x00, 0x21,
+	0x0f, 0x00, 0x43, 0x00,
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+	0xd0, 0x9e, 0x04, 0x17,
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+	0xa0, 0x00, 0x14, 0x00,
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+	0x05, 0xe0, 0xde, 0x32,
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+	0x73, 0x00, 0xfb, 0xee,
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+	0x72, 0x00, 0xf9, 0xfe,
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+	0xff, 0x33, 0x52, 0x0e,
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+	0xf0, 0x13, 0xff, 0xdc,
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+	0x02, 0x06, 0xe0, 0xcd,
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+	0x2e, 0x74, 0xec, 0xdb,
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+	0x0f, 0x1e, 0x20, 0x0c,
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+	0xde, 0x00, 0x00, 0x00,
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+	0x23, 0x23, 0xdc, 0xec,
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+	0xfa, 0xfe, 0x51, 0x10,
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+	0xee, 0x20, 0x76, 0x00,
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+	0xde, 0xf0, 0x45, 0xcf,
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+	0x01, 0x20, 0x0f, 0x0f,
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+	0x45, 0x00, 0xab, 0xed,
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+	0x0e, 0x00, 0x12, 0x00,
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+	0xff, 0x0f, 0x20, 0x22,
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+	0xf0, 0xf0, 0xee, 0x05,
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+	0x01, 0xf2, 0x05, 0xf1,
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+	0x0d, 0xeb, 0x0f, 0x31,
+	0x53, 0xfe, 0x0e, 0x00, 0xdc, 0xbb, 0x20, 0x41, 0xfd, 0xfd, 0x0e, 0xfb,
+	0x1f, 0x61, 0x31, 0x20,
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+	0xe8, 0x9c, 0x21, 0x43,
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+	0xad, 0xf3, 0x07, 0xe1,
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+	0xee, 0x06, 0x02, 0xce,
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+	0x22, 0x12, 0x43, 0xef,
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+	0x0e, 0x55, 0x21, 0xda,
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+	0x72, 0x10, 0x1e, 0x47,
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+	0xdf, 0x02, 0x21, 0x01,
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+	0xd0, 0x8f, 0x03, 0x27,
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+	0x0f, 0x0f, 0x27, 0x22,
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+	0x88, 0xee, 0x71, 0x21,
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+	0x0e, 0x52, 0x11, 0x00,
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+	0x0d, 0x20, 0x11, 0xfd,
+	0x00, 0x0f, 0x0f, 0xf0, 0xab, 0xdd, 0x00, 0xfe, 0x13, 0x47, 0xcf, 0xba,
+	0x00, 0x21, 0x21, 0x01,
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+	0xf3, 0xf5, 0x03, 0xf2,
+	0x8e, 0xbc, 0xf0, 0xf2, 0xf1, 0x16, 0x11, 0xde, 0xef, 0x11, 0x00, 0xbb,
+	0x01, 0x57, 0x11, 0xe0,
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+	0x0e, 0x20, 0x0c, 0xdb,
+	0xff, 0x41, 0x31, 0xfe, 0xed, 0x00, 0x53, 0x00, 0xec, 0x0f, 0xed, 0xdb,
+	0x0f, 0x2f, 0x2d, 0x60,
+	0x1f, 0x09, 0xff, 0x61, 0x22, 0x0e, 0x30, 0x2e, 0x0d, 0x52, 0x0f, 0xb8,
+	0x0e, 0x34, 0x22, 0xee,
+	0xff, 0xbc, 0x00, 0x23, 0x0f, 0xfe, 0xff, 0x0e, 0x23, 0x67, 0x00, 0xef,
+	0xed, 0xf0, 0x01, 0x22,
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+	0xf3, 0xf2, 0xd1, 0xf4,
+	0x47, 0x02, 0xc0, 0xf0, 0x13, 0xce, 0xef, 0x77, 0x24, 0xe0, 0xf0, 0x12,
+	0x02, 0xf0, 0x01, 0xe1,
+	0xcf, 0x00, 0x01, 0x00, 0x0f, 0x0f, 0x00, 0xd9, 0x0e, 0x72, 0x52, 0x0c,
+	0xfc, 0x10, 0x20, 0x0e,
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+	0x40, 0x63, 0x0f, 0x2f,
+	0x20, 0x0d, 0x0d, 0x0e, 0x2f, 0x1f, 0x1f, 0x1e, 0xea, 0xee, 0xef, 0xce,
+	0xcb, 0x10, 0x41, 0x0f,
+	0x0f, 0x00, 0xed, 0xee, 0x13, 0x01, 0xbf, 0xc0, 0xc0, 0x07, 0x16, 0xe0,
+	0xe0, 0x12, 0x23, 0xe1,
+	0x8f, 0xf2, 0x37, 0x02, 0xb0, 0x00, 0xf1, 0x02, 0xf3, 0xf3, 0xf1, 0xe1,
+	0xd0, 0xc0, 0xf2, 0x17,
+	0x01, 0xb0, 0xe0, 0x12, 0x37, 0xf1, 0xa0, 0xe0, 0x01, 0x13, 0x11, 0x00,
+	0x00, 0xee, 0x44, 0x32,
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+	0x10, 0x20, 0x2f, 0xcc,
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+	0x64, 0x23, 0x0e, 0x0d,
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+	0x64, 0x36, 0x00, 0xef,
+	0x02, 0xf0, 0x9c, 0x01, 0x26, 0x12, 0x02, 0x01, 0xf2, 0xd2, 0xe5, 0xf2,
+	0xc1, 0x90, 0xd0, 0x03,
+	0xf2, 0xd0, 0xf4, 0xf1, 0xd0, 0xd0, 0xf2, 0xe0, 0xba, 0x00, 0x53, 0x13,
+	0x00, 0x25, 0x12, 0xf1,
+	0xc0, 0xf1, 0x02, 0xcd, 0xd0, 0x17, 0x01, 0xbf, 0x00, 0x00, 0xcd, 0xdc,
+	0x0f, 0x72, 0x1e, 0x08,
+	0x0f, 0xfe, 0x00, 0x63, 0x00, 0xfb, 0x00, 0x00, 0xfd, 0xee, 0xe0, 0xed,
+	0xcc, 0x0e, 0x00, 0x20,
+	0x0c, 0x0d, 0x50, 0x31, 0x1f, 0x3e, 0x4d, 0x3d, 0x41, 0x47, 0x21, 0x0e,
+	0xfd, 0xfd, 0xee, 0xcf,
+	0xe1, 0xf0, 0xfb, 0x0e, 0x1d, 0x62, 0x14, 0xed, 0xdd, 0xef, 0xfe, 0x0e,
+	0xf0, 0xbd, 0xef, 0x47,
+	0x25, 0x01, 0x35, 0xf0, 0x8c, 0x00, 0x02, 0x04, 0xe0, 0xf0, 0x01, 0xdf,
+	0x8c, 0x01, 0x16, 0xff,
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+	0x13, 0x01, 0xf2, 0xe2,
+	0xe4, 0xe2, 0xe1, 0xcf, 0x10, 0x30, 0xfd, 0x10, 0x41, 0xf8, 0x4d, 0x73,
+	0x1c, 0xea, 0x22, 0x22,
+	0xee, 0xdd, 0xf0, 0xbc, 0x00, 0x23, 0xbc, 0xdc, 0x00, 0x00, 0x10, 0x2f,
+	0x0e, 0x50, 0x31, 0x20,
+	0x73, 0x20, 0x1c, 0x10, 0x23, 0x21, 0x00, 0x00, 0xff, 0xed, 0x37, 0x06,
+	0xd0, 0x21, 0x62, 0x00,
+	0xef, 0xee, 0xff, 0x00, 0x66, 0x44, 0x00, 0xff, 0xf0, 0x12, 0x22, 0x00,
+	0x27, 0x42, 0x00, 0xf2,
+	0xf1, 0xe1, 0x15, 0x02, 0x80, 0xe1, 0xf3, 0xc0, 0xcf, 0xe0, 0xce, 0xe0,
+	0x05, 0x00, 0xce, 0xf0,
+	0x14, 0x00, 0x0f, 0xff, 0xd0, 0x07, 0x56, 0x01, 0xf0, 0xe1, 0xd0, 0xe2,
+	0xf1, 0x90, 0xf0, 0x03,
+	0xff, 0xcd, 0x00, 0x20, 0x09, 0x1e, 0x20, 0x30, 0x61, 0x10, 0xfb, 0x0e,
+	0x51, 0x0e, 0xfa, 0x0f,
+	0x10, 0x1e, 0xec, 0xbc, 0x0f, 0x10, 0x1e, 0x1e, 0x5e, 0x40, 0x2e, 0x1b,
+	0x4f, 0x1e, 0x20, 0x51,
+	0x1e, 0x09, 0x10, 0x31, 0xfe, 0xdb, 0x20, 0x70, 0x20, 0x1f, 0x0f, 0x0e,
+	0x00, 0x12, 0x00, 0xff,
+	0x12, 0xf0, 0xd0, 0x27, 0x13, 0xd0, 0xe0, 0xf2, 0x56, 0x12, 0xc0, 0x12,
+	0x75, 0x12, 0xc0, 0xf2,
+	0x05, 0xe0, 0xa0, 0xf0, 0x02, 0xf2, 0xdf, 0xe0, 0x04, 0xd0, 0xdf, 0x11,
+	0x31, 0x02, 0xf6, 0xe2,
+	0xd1, 0x75, 0x01, 0xd0, 0xf0, 0x03, 0xe1, 0x23, 0x66, 0xef, 0xde, 0x00,
+	0x30, 0x61, 0x21, 0x10,
+	0x0d, 0x2e, 0x0c, 0xfe, 0x32, 0x55, 0x0f, 0x0e, 0x0e, 0x0c, 0xed, 0xbd,
+	0xc9, 0x10, 0x63, 0x0e,
+	0xfc, 0x20, 0x2e, 0x1e, 0x71, 0x22, 0x0f, 0x1f, 0x1c, 0x1f, 0x21, 0x00,
+	0x0c, 0x0f, 0x00, 0x0f,
+	0xff, 0x13, 0x0e, 0xea, 0x00, 0x33, 0x72, 0x00, 0xd8, 0x00, 0x75, 0x30,
+	0xe0, 0xe0, 0x00, 0x01,
+	0xf0, 0xf1, 0x02, 0x05, 0x02, 0x02, 0xa0, 0xf2, 0x05, 0xc0, 0xbf, 0x04,
+	0xf2, 0xb0, 0x01, 0xf1,
+	0xcd, 0xe0, 0x16, 0x01, 0xf0, 0xf0, 0xf0, 0xe0, 0x02, 0x11, 0xdf, 0x03,
+	0x27, 0xf1, 0xb0, 0xf1,
+	0xf2, 0xd1, 0x9d, 0xfe, 0x27, 0x00, 0xed, 0x0f, 0x3e, 0x4f, 0x40, 0x20,
+	0x1f, 0x0d, 0x22, 0x63,
+	0x1b, 0x2f, 0x10, 0x0c, 0x0f, 0x10, 0x0b, 0xfd, 0xec, 0xde, 0x0e, 0x1f,
+	0x5d, 0x3c, 0x7f, 0x3f,
+	0x3e, 0x0c, 0x0b, 0x0f, 0x2e, 0x0e, 0x30, 0x40, 0x0d, 0x0a, 0x0f, 0x30,
+	0x1e, 0xaa, 0xf0, 0x40,
+	0x10, 0xed, 0xfe, 0x10, 0xcc, 0xee, 0x22, 0xe0, 0x89, 0x00, 0x10, 0x00,
+	0x00, 0xff, 0xef, 0x02,
+	0x34, 0x01, 0xf1, 0x02, 0xa0, 0xc0, 0xf5, 0xf3, 0xb0, 0xe0, 0xd0, 0xe0,
+	0xf1, 0xd0, 0xdd, 0xef,
+	0x02, 0x04, 0xbe, 0xdf, 0x07, 0x04, 0xf1, 0xc2, 0xe3, 0xe7, 0xa1, 0xb0,
+	0xf2, 0x03, 0x02, 0xf1,
+	0xde, 0xff, 0x00, 0x12, 0x43, 0x1f, 0x1d, 0x47, 0x21, 0xec, 0x30, 0x72,
+	0x0f, 0xfc, 0x01, 0x02,
+	0xce, 0xf0, 0xeb, 0xfe, 0x00, 0x1e, 0x0d, 0x2d, 0x5e, 0x5d, 0x2c, 0x2c,
+	0x40, 0x31, 0x2d, 0x2a,
+	0x50, 0x1f, 0x0a, 0x00, 0x43, 0x0f, 0xfc, 0x10, 0x00, 0xeb, 0x0f, 0x60,
+	0x11, 0xfe, 0x0c, 0x20,
+	0x30, 0xfd, 0x0e, 0x20, 0x3f, 0x3d, 0x60, 0x22, 0x0f, 0xfc, 0x13, 0x47,
+	0x01, 0xe2, 0xc2, 0xf6,
+	0x02, 0x91, 0xd1, 0xd1, 0xd1, 0xf2, 0xf4, 0xa0, 0xe0, 0x03, 0xf1, 0x03,
+	0x14, 0xff, 0x04, 0x37,
+	0x01, 0x02, 0xf2, 0x81, 0x07, 0x01, 0xb0, 0xe0, 0x02, 0x02, 0xe0, 0xd0,
+	0xf0, 0x02, 0x12, 0x00,
+	0xdd, 0x40, 0x77, 0x00, 0x0f, 0x32, 0x20, 0xfc, 0xfe, 0x00, 0xff, 0xce,
+	0x00, 0xea, 0xee, 0x30,
+	0xfd, 0x0c, 0x70, 0x1e, 0x0d, 0x21, 0x3f, 0x0b, 0x3e, 0x40, 0x3e, 0x2d,
+	0x2b, 0x50, 0x2f, 0x0c,
+	0x1d, 0x4e, 0x4f, 0x0d, 0x1f, 0x2f, 0x0b, 0xed, 0x40, 0x20, 0xce, 0xcd,
+	0xee, 0xda, 0xff, 0xfe,
+	0xee, 0xef, 0xcf, 0xc0, 0x26, 0x51, 0x00, 0x00, 0x13, 0x06, 0xd2, 0xc2,
+	0xd0, 0xc1, 0xd2, 0xe4,
+	0xa0, 0xd0, 0xf2, 0xf2, 0xbf, 0xf0, 0x01, 0xe1, 0x05, 0x05, 0xf1, 0xf1,
+	0x24, 0x03, 0xc4, 0xd2,
+	0x8f, 0xdf, 0x14, 0x12, 0xff, 0xf0, 0x03, 0xff, 0xea, 0x10, 0x50, 0x00,
+	0x1c, 0x40, 0x50, 0x20,
+	0x1e, 0xf8, 0x0f, 0x0f, 0xf0, 0xef, 0xee, 0xfd, 0xdb, 0x20, 0x40, 0xfd,
+	0x0d, 0x5f, 0x7f, 0x3e,
+	0x1d, 0x30, 0x20, 0x3f, 0x4e, 0x31, 0x21, 0x20, 0x14, 0x02, 0x43, 0x57,
+	0x00, 0x02, 0x13, 0x01,
+	0xe0, 0xf0, 0x26, 0x21, 0xef, 0x22, 0x23, 0xbf, 0xe0, 0x57, 0x42, 0xfe,
+	0x23, 0x57, 0x01, 0xf0,
+	0x01, 0x04, 0x13, 0x03, 0x24, 0x03, 0xa2, 0xd0, 0xe4, 0xe2, 0xc1, 0xd2,
+	0xc1, 0x02, 0x00, 0xf0,
+	0xf0, 0xde, 0x00, 0x77, 0x16, 0xf0, 0x00, 0x01, 0xd0, 0x02, 0x26, 0xd0,
+	0x90, 0x00, 0x00, 0xd0,
+	0xee, 0xe0, 0x01, 0xd0, 0xee, 0x61, 0x53, 0x1e, 0x1f, 0x2f, 0x0d, 0xf9,
+	0xee, 0x00, 0x2e, 0xc9,
+	0xee, 0x10, 0x0f, 0xbb, 0x00, 0x00, 0xdc, 0x1f, 0x70, 0x1f, 0x0d, 0x10,
+	0x22, 0x2f, 0x1c, 0x2d,
+	0x1e, 0x2f, 0x30, 0x21, 0x0e, 0x2d, 0x5f, 0x30, 0x09, 0x0e, 0x72, 0x1f,
+	0xea, 0x00, 0x21, 0xdf,
+	0xbe, 0x00, 0xff, 0xfd, 0x00, 0x12, 0x00, 0xcb, 0xf0, 0x34, 0x00, 0x9d,
+	0x17, 0x44, 0xf0, 0x01,
+	0x03, 0xcf, 0xf0, 0x02, 0xd2, 0xe2, 0xe0, 0xed, 0x20, 0x0f, 0xc9, 0x0f,
+	0x20, 0x01, 0x00, 0x21,
+	0x01, 0xf2, 0x57, 0x03, 0x80, 0x02, 0x33, 0x13, 0xf2, 0xc1, 0xa0, 0xf0,
+	0xfe, 0x0f, 0x24, 0x20,
+	0x0d, 0x20, 0x35, 0x20, 0x0d, 0x1e, 0x3d, 0x0e, 0xcd, 0xbe, 0xe0, 0xcd,
+	0xb8, 0xeb, 0x00, 0xf1,
+	0xfd, 0xdc, 0xfd, 0x0d, 0x5e, 0x50, 0x1f, 0x1d, 0x30, 0x2c, 0x3b, 0x50,
+	0x22, 0x0c, 0x0e, 0x20,
+	0x30, 0x2e, 0x2e, 0x20, 0x11, 0x0f, 0x0f, 0x00, 0x0e, 0xb8, 0xfe, 0x33,
+	0x22, 0x00, 0x50, 0x40,
+	0x2c, 0x2e, 0x22, 0x04, 0x01, 0x75, 0x23, 0x05, 0x05, 0x01, 0xb0, 0xe4,
+	0xd3, 0xc3, 0xd2, 0xb2,
+	0xc0, 0xc0, 0xe0, 0x02, 0x02, 0xf2, 0xfe, 0xdc, 0xe4, 0xf6, 0x01, 0xf1,
+	0x14, 0x24, 0xf1, 0xc1,
+	0xc2, 0xa0, 0xd2, 0xf4, 0xb1, 0xdf, 0xd0, 0x01, 0xf1, 0x02, 0x17, 0x42,
+	0x11, 0x20, 0x0f, 0x1f,
+	0x31, 0x10, 0xfb, 0x10, 0x30, 0x0e, 0xd9, 0x0f, 0x0f, 0xad, 0x05, 0x05,
+	0x8b, 0x00, 0x6c, 0x00,
+	0x9c, 0xbe, 0x1e, 0x3a, 0xc5, 0xbd, 0xb5, 0x3c, 0xbc, 0x2e, 0x0b, 0xdf,
+	0x54, 0x32, 0xfc, 0xfd,
+	0x0e, 0x5e, 0xb8, 0x0e, 0x54, 0x0f, 0x2f, 0x41, 0x0e, 0xb8, 0x00, 0x21,
+	0x10, 0x0f, 0x00, 0x21,
+	0xed, 0xbb, 0x20, 0x43, 0xfe, 0xfc, 0x10, 0x31, 0xff, 0x0e, 0x41, 0x0f,
+	0x0f, 0x42, 0x10, 0xe0,
+	0xf1, 0x04, 0x10, 0xb8, 0x10, 0x34, 0xbe, 0xee, 0x10, 0x72, 0xf0, 0xdb,
+	0x10, 0x13, 0xf0, 0xe0,
+	0x10, 0x0e, 0x24, 0x35, 0xdf, 0x00, 0xe0, 0x9e, 0x00, 0x06, 0xe0, 0xd0,
+	0x01, 0x0f, 0x40, 0x34,
+	0xc0, 0xdd, 0x03, 0x41, 0x0e, 0xf0, 0x35, 0xe0, 0xcf, 0x02, 0xe2, 0x9a,
+	0x0f, 0x11, 0x14, 0x00,
+	0xff, 0xdd, 0x00, 0x32, 0x52, 0xfe, 0xdd, 0x30, 0x2f, 0x0f, 0x00, 0xf1,
+	0x2f, 0x52, 0x02, 0xd8,
+	0x7c, 0x20, 0x10, 0x15, 0x00, 0xdc, 0x0f, 0x51, 0x21, 0xfc, 0x20, 0x55,
+	0xde, 0xdc, 0x31, 0x42,
+	0x0e, 0x1e, 0x30, 0x02, 0x0e, 0x50, 0x32, 0x11, 0x0e, 0x02, 0x01, 0xe1,
+	0x05, 0x03, 0xfe, 0xae,
+	0xf5, 0x31, 0x0e, 0xb9, 0xf0, 0x03, 0xee, 0x10, 0x43, 0xd0, 0x00, 0x62,
+	0x11, 0xef, 0x04, 0x47,
+	0xd0, 0x8b, 0x01, 0x12, 0xf0, 0xde, 0x14, 0x11, 0xdd, 0x12, 0x56, 0xdf,
+	0xfe, 0x13, 0x21, 0x00,
+	0x43, 0xf1, 0xbe, 0x00, 0x37, 0xef, 0xfe, 0x13, 0x00, 0xc0, 0x00, 0x15,
+	0x00, 0xd0, 0x1d, 0x72,
+	0x01, 0xce, 0x01, 0x04, 0x0f, 0x44, 0x10, 0xea, 0x11, 0x62, 0xff, 0xff,
+	0x10, 0x22, 0x20, 0x00,
+	0x0d, 0x0f, 0x00, 0x11, 0x10, 0x0b, 0x23, 0x2f, 0xb8, 0x0e, 0x22, 0x10,
+	0x0e, 0x70, 0x20, 0xfa,
+	0x00, 0x30, 0x00, 0xff, 0xd0, 0xf1, 0xdf, 0x72, 0x67, 0xfe, 0xcc, 0x2f,
+	0x01, 0x00, 0xf0, 0xfd,
+	0xee, 0x0f, 0x00, 0x0f, 0x0f, 0x24, 0xde, 0xcf, 0x23, 0x24, 0x8c, 0xda,
+	0x00, 0x01, 0x04, 0xe0,
+	0x8b, 0xff, 0x01, 0x10, 0x00, 0x46, 0x00, 0xab, 0x00, 0x11, 0x01, 0x01,
+	0x10, 0x0e, 0xbf, 0x00,
+	0x02, 0x01, 0xeb, 0xf0, 0x04, 0x0f, 0xdd, 0x21, 0x21, 0xce, 0xff, 0x00,
+	0x10, 0x00, 0x0f, 0x2f,
+	0x0e, 0xe0, 0xff, 0xda, 0x20, 0x21, 0xe0, 0x0f, 0x20, 0xbe, 0xdf, 0xff,
+	0x1a, 0x7f, 0x00, 0xde,
+	0xdc, 0x31, 0x00, 0xe9, 0x2f, 0xed, 0x0d, 0x74, 0x70, 0x3d, 0x0b, 0x00,
+	0x00, 0x02, 0xf2, 0x00,
+	0x00, 0x20, 0x20, 0x00, 0xce, 0xf1, 0xbe, 0x0b, 0x22, 0xbd, 0xcc, 0x35,
+	0x20, 0xdc, 0x12, 0x55,
+	0x00, 0xcd, 0x00, 0x03, 0x03, 0x00, 0xf0, 0xcd, 0xef, 0xd4, 0xe4, 0xd1,
+	0xf1, 0xd0, 0xb0, 0x10,
+	0x47, 0xc4, 0xe0, 0x31, 0x1e, 0x30, 0x25, 0x80, 0xfd, 0x00, 0x03, 0x11,
+	0x03, 0xfe, 0xee, 0x1f,
+	0x03, 0xd1, 0xcc, 0x00, 0x41, 0xcb, 0xee, 0xf0, 0x24, 0x1e, 0xc9, 0x30,
+	0x70, 0x0f, 0xfe, 0x00,
+	0x40, 0x12, 0x0d, 0xeb, 0x2f, 0x40, 0x0d, 0x31, 0x00, 0xa8, 0x0f, 0x00,
+	0x00, 0xcc, 0x0f, 0x43,
+	0x1e, 0x08, 0x20, 0x10, 0xf0, 0xe0, 0x2e, 0x20, 0xaf, 0x01, 0x43, 0x10,
+	0xfe, 0xf0, 0x23, 0x00,
+	0xdc, 0xb1, 0xf1, 0xed, 0x1f, 0x57, 0xe0, 0xcd, 0x10, 0x21, 0xf2, 0x04,
+	0x54, 0xff, 0xde, 0x15,
+	0x10, 0xe0, 0x03, 0xd4, 0xae, 0x11, 0x04, 0xd0, 0x02, 0x42, 0x02, 0xe1,
+	0xde, 0x51, 0x57, 0x00,
+	0xee, 0x23, 0x14, 0xe0, 0xfe, 0x00, 0x2f, 0x00, 0xe0, 0xd0, 0x00, 0x70,
+	0x20, 0xc8, 0xef, 0x27,
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+	0x10, 0x53, 0x00, 0x0c,
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+	0xce, 0xfb, 0x11, 0x14,
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+	0x60, 0x65, 0x00, 0xdb,
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+	0x00, 0x10, 0x53, 0x00,
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+	0x11, 0x8f, 0x9f, 0xf2,
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+	0x0e, 0x0f, 0x05, 0x00,
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+	0x53, 0x00, 0x4c, 0x00,
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+	0x12, 0xbe, 0xef, 0x27,
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+	0x30, 0x26, 0x00, 0xde,
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+	0x01, 0x01, 0xdd, 0x0e,
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+	0xfc, 0x00, 0xee, 0x0c,
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+	0x01, 0x13, 0x00, 0xd0,
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+	0x62, 0x10, 0xf0, 0xe0,
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+	0xa8, 0x0c, 0x13, 0x00,
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+	0x10, 0x42, 0x1f, 0x1c,
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+	0x20, 0x02, 0x04, 0xd0,
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+	0x02, 0x20, 0x01, 0x24,
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+	0x42, 0xf0, 0xc0, 0xf0,
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+	0x00, 0xff, 0xf1, 0x13,
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+	0xfd, 0x0b, 0x41, 0x00,
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+	0xa8, 0xff, 0xf1, 0x22,
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+	0x22, 0x9e, 0xff, 0x0f,
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+	0x00, 0x1f, 0x20, 0x2e,
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+	0x0e, 0x20, 0x01, 0xad,
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+	0x0e, 0xfe, 0x8c, 0xef,
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+	0x40, 0x11, 0x00, 0xff,
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+	0xff, 0x0f, 0xff, 0xf0,
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+	0x02, 0x11, 0x0e, 0x01,
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+	0x0c, 0x27, 0x11, 0x01,
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+	0x10, 0x32, 0x1e, 0xf0,
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+	0x40, 0x51, 0x00, 0xf0,
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+	0x10, 0x40, 0x00, 0xbf,
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+	0xa2, 0xe2, 0x05, 0x12,
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+	0x2f, 0x0e, 0x31, 0x13,
+	0xff, 0xce, 0xf2, 0x04, 0x45, 0x00, 0xb2, 0xe3, 0xe0, 0x03, 0x77, 0x24,
+	0x0f, 0xfe, 0x01, 0x01,
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+	0x44, 0xc3, 0xd1, 0xd5,
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+	0x24, 0x21, 0x01, 0xf0,
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+	0xd4, 0xf1, 0x34, 0x72,
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+	0xb1, 0xcc, 0x36, 0x33,
+	0xde, 0xef, 0x00, 0x00, 0x25, 0xff, 0xcc, 0x60, 0x43, 0x00, 0xfd, 0x20,
+	0x20, 0x00, 0xee, 0x01,
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+	0x1f, 0xfa, 0x40, 0x31,
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+	0xd0, 0x03, 0xf1, 0xae,
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+	0x00, 0xff, 0x04, 0x12,
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+	0x20, 0x12, 0x0e, 0x00,
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+	0x9a, 0x0f, 0x00, 0x00,
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+	0x0f, 0x11, 0x10, 0xec,
+	0x0c, 0x7f, 0x3f, 0x10, 0xfc, 0xbe, 0x00, 0x0d, 0x0c, 0x1f, 0x2d, 0x00,
+	0xd0, 0xba, 0x24, 0x2f,
+	0xd8, 0x73, 0x51, 0xab, 0x01, 0x10, 0x01, 0x01, 0xed, 0xf0, 0x01, 0xf0,
+	0xee, 0x00, 0x13, 0x33,
+	0x0c, 0x20, 0x21, 0x04, 0x20, 0x00, 0x00, 0x32, 0x11, 0x01, 0x33, 0xb0,
+	0xce, 0xf0, 0xf3, 0x03,
+	0x27, 0x77, 0xff, 0xcd, 0x23, 0x13, 0xee, 0x12, 0x34, 0xbc, 0xf0, 0x23,
+	0x20, 0x0f, 0x1f, 0x30,
+	0x24, 0x67, 0xe0, 0xce, 0x00, 0x32, 0x22, 0xf1, 0xf0, 0x0f, 0xbf, 0xe0,
+	0x02, 0x11, 0x0d, 0xf8,
+	0x20, 0x52, 0xbf, 0xfe, 0x40, 0x30, 0x0e, 0xfe, 0x02, 0xe1, 0xfd, 0xfc,
+	0x3c, 0x50, 0x00, 0xcd,
+	0xea, 0x0e, 0x0e, 0x6f, 0x24, 0x00, 0xc9, 0xef, 0x00, 0x0f, 0x10, 0x31,
+	0xfe, 0xef, 0xdc, 0x0e,
+	0x41, 0xef, 0x8c, 0x00, 0x1f, 0xfc, 0xcf, 0x1f, 0x3e, 0xe9, 0xef, 0x20,
+	0x00, 0xad, 0x00, 0x00,
+	0x10, 0x30, 0x0f, 0xee, 0xdd, 0x00, 0x35, 0x10, 0x21, 0x00, 0x44, 0x13,
+	0xe0, 0x3c, 0x7e, 0x42,
+	0xf7, 0xb1, 0xee, 0x0e, 0x00, 0xfe, 0x14, 0x65, 0xef, 0xdc, 0x01, 0x22,
+	0x12, 0x00, 0xe0, 0x0f,
+	0x23, 0x34, 0x20, 0x1e, 0x7e, 0x77, 0x40, 0xde, 0x0f, 0x0f, 0x00, 0x0d,
+	0xdd, 0xeb, 0xef, 0xf0,
+	0x21, 0x3d, 0xfa, 0xe0, 0x0f, 0x3f, 0x71, 0x71, 0x0d, 0x0d, 0x02, 0x20,
+	0xf8, 0x0b, 0x40, 0x00,
+	0xcd, 0x00, 0xe1, 0x01, 0x30, 0x0c, 0xe0, 0xe0, 0xdf, 0xd0, 0xf1, 0xf0,
+	0x8c, 0xc9, 0xf0, 0x03,
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+	0x01, 0xf0, 0xff, 0xf2,
+	0xf4, 0xe0, 0xe2, 0xf0, 0x05, 0x24, 0x00, 0x12, 0xf0, 0xce, 0x10, 0x64,
+	0x01, 0x25, 0x01, 0xef,
+	0xff, 0x22, 0x63, 0x02, 0x02, 0x02, 0xf3, 0xd5, 0x13, 0x21, 0x00, 0x17,
+	0xd0, 0x90, 0x57, 0x03,
+	0xdd, 0x20, 0x50, 0x1f, 0x43, 0xe0, 0xdf, 0x0f, 0x44, 0x11, 0xff, 0xff,
+	0xf0, 0xe0, 0x23, 0x03,
+	0xcc, 0xd0, 0x21, 0x1d, 0x21, 0x03, 0xd0, 0x63, 0x1b, 0x1e, 0x21, 0x05,
+	0xff, 0x0a, 0x71, 0x71,
+	0x0e, 0xea, 0x02, 0xf3, 0xee, 0xed, 0xdf, 0x0f, 0xff, 0xce, 0x00, 0xdf,
+	0xfd, 0xdb, 0xfd, 0x10,
+	0x10, 0x3d, 0x50, 0x1e, 0x44, 0x44, 0x1f, 0x00, 0x1f, 0x0d, 0x0e, 0xe5,
+	0x06, 0xdf, 0xbe, 0x00,
+	0xf3, 0xf3, 0x22, 0x13, 0x8f, 0x9c, 0x00, 0x14, 0x47, 0x0e, 0xfd, 0x01,
+	0x22, 0x0e, 0x21, 0x10,
+	0xfa, 0x11, 0x23, 0xd0, 0xc0, 0xd0, 0x10, 0x73, 0x11, 0xfd, 0xd1, 0x07,
+	0x01, 0x1f, 0x7f, 0x0f,
+	0x0e, 0xef, 0xf1, 0x01, 0x31, 0x23, 0x1d, 0x00, 0xd2, 0xd3, 0x27, 0x77,
+	0x43, 0x00, 0xd0, 0xf1,
+	0xe4, 0xd1, 0xe0, 0x01, 0x61, 0x02, 0x15, 0x00, 0x01, 0x20, 0x34, 0x65,
+	0x00, 0x00, 0xb1, 0xf0,
+	0xd0, 0xf1, 0xf1, 0xee, 0xcd, 0x0f, 0x00, 0xed, 0xef, 0xb8, 0x00, 0x14,
+	0x60, 0x20, 0x1f, 0x1c,
+	0x11, 0x03, 0xfd, 0xec, 0xff, 0x22, 0x34, 0x00, 0xcf, 0x02, 0x14, 0xf0,
+	0x07, 0x03, 0xaf, 0xed,
+	0xf3, 0x07, 0x01, 0xcf, 0x89, 0x33, 0x34, 0x9d, 0x0e, 0x00, 0x21, 0x23,
+	0x00, 0xec, 0x12, 0x11,
+	0x10, 0x00, 0x00, 0x11, 0x00, 0xed, 0x03, 0xf4, 0xee, 0x0e, 0x72, 0x24,
+	0xfe, 0xee, 0x31, 0x21,
+	0xee, 0xeb, 0x00, 0x63, 0x00, 0x0c, 0x00, 0xe8, 0xd0, 0x63, 0x21, 0xee,
+	0xde, 0xff, 0xce, 0xfd,
+	0x45, 0x45, 0xed, 0xdc, 0x41, 0x31, 0x00, 0xed, 0x30, 0x53, 0xff, 0xca,
+	0x20, 0x20, 0xe0, 0x9f,
+	0x0e, 0xef, 0xeb, 0x00, 0x11, 0x0e, 0xaa, 0x01, 0x32, 0x0f, 0xfd, 0xf3,
+	0x12, 0x1d, 0x30, 0x54,
+	0x23, 0x0e, 0x00, 0xe2, 0x13, 0x52, 0x00, 0x0d, 0xd4, 0xf3, 0xc4, 0xe2,
+	0x03, 0x01, 0x32, 0x01,
+	0xb2, 0xef, 0xcb, 0x02, 0x30, 0x0e, 0x23, 0x33, 0xc8, 0xed, 0x10, 0x63,
+	0x33, 0x46, 0x12, 0xff,
+	0xd9, 0x20, 0x72, 0x00, 0xfd, 0xdf, 0x0d, 0x0e, 0x0d, 0x02, 0x00, 0xd8,
+	0x00, 0x12, 0x01, 0xff,
+	0xbc, 0x1e, 0x70, 0x40, 0x03, 0xe0, 0xfa, 0x1f, 0x20, 0x30, 0x1f, 0xea,
+	0xe0, 0x10, 0x10, 0xcf,
+	0xbd, 0x00, 0x03, 0x22, 0xbc, 0xed, 0xee, 0xd0, 0xf4, 0xe1, 0xce, 0xcb,
+	0x00, 0x21, 0x20, 0x0f,
+	0x75, 0x21, 0x0f, 0x13, 0x03, 0xd2, 0xf1, 0xc0, 0xf4, 0x07, 0x33, 0x62,
+	0x04, 0xc0, 0x03, 0x04,
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+	0x62, 0x11, 0x0f, 0x43,
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+	0x03, 0x31, 0x40, 0x20,
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+	0x6f, 0x2e, 0x09, 0xdb,
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+	0x20, 0x10, 0x4f, 0x3d,
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+	0x00, 0x0c, 0x10, 0x30,
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+	0x17, 0xf1, 0x80, 0xf2,
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+	0xe1, 0xf2, 0x01, 0x02,
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+	0xd0, 0xe2, 0xf1, 0x35,
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+	0x10, 0xfd, 0x0f, 0x10,
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+	0x0b, 0x5f, 0x40, 0x2d,
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+	0x99, 0x10, 0x64, 0x00,
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+	0xef, 0xbc, 0x12, 0x36,
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+	0xf1, 0xa0, 0x02, 0x26,
+	0x01, 0xf0, 0x12, 0x33, 0xc0, 0x01, 0x36, 0xe0, 0xb0, 0x01, 0x02, 0xf1,
+	0x02, 0xd1, 0x14, 0x54,
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+	0x21, 0xec, 0x00, 0x32,
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+	0x1f, 0x0b, 0x1f, 0x40,
+	0x1e, 0x2d, 0x2e, 0x3e, 0x2e, 0x0b, 0x1d, 0x4f, 0x0c, 0xdd, 0x0d, 0x51,
+	0x0e, 0xda, 0x8c, 0x10,
+	0x35, 0xff, 0xcd, 0xf0, 0x00, 0x00, 0x00, 0x00, 0xbd, 0xed, 0x13, 0x25,
+	0xbc, 0xee, 0x01, 0x23,
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+	0xd1, 0x8e, 0xe0, 0x06,
+	0x02, 0xdf, 0xef, 0x01, 0x25, 0x01, 0xd0, 0xcf, 0x11, 0x43, 0xf0, 0xce,
+	0x00, 0x15, 0xf0, 0xbf,
+	0x01, 0x02, 0xcd, 0xf0, 0x16, 0xef, 0xcd, 0x00, 0x23, 0xff, 0xdd, 0xca,
+	0x00, 0x00, 0x10, 0x51,
+	0xdc, 0xcb, 0x31, 0x63, 0x0f, 0x0d, 0x1f, 0x20, 0x43, 0x33, 0x0d, 0x0b,
+	0x30, 0x42, 0x0e, 0x0b,
+	0x31, 0x31, 0xfd, 0x0d, 0x1f, 0x2d, 0x3e, 0x40, 0xfc, 0x43, 0x54, 0xb8,
+	0xfe, 0x32, 0x11, 0xdd,
+	0x0f, 0x10, 0xef, 0xe0, 0x13, 0xcf, 0x9a, 0xf0, 0x33, 0x35, 0xef, 0xcd,
+	0x02, 0x01, 0xe0, 0x13,
+	0x02, 0xe0, 0x45, 0x12, 0xc0, 0xef, 0x13, 0xf1, 0xe0, 0x26, 0xe0, 0xcf,
+	0xf0, 0xe0, 0xe1, 0xe0,
+	0x8c, 0xdc, 0x23, 0x56, 0xf0, 0xaa, 0xff, 0x23, 0x32, 0xde, 0xcd, 0x01,
+	0x45, 0x00, 0xdd, 0xff,
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+	0x10, 0x0e, 0x41, 0x32,
+	0x0f, 0x00, 0x77, 0x46, 0x00, 0x00, 0x31, 0x44, 0x00, 0x00, 0x20, 0x41,
+	0x32, 0x21, 0x00, 0x1f,
+	0x63, 0x24, 0x0f, 0x00, 0x23, 0x00, 0x00, 0x22, 0x00, 0x02, 0x36, 0xce,
+	0xff, 0x77, 0x22, 0xef,
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+	0x00, 0x89, 0x01, 0x25,
+	0xdf, 0x8b, 0x00, 0x04, 0xef, 0xde, 0x02, 0x14, 0xdf, 0xbe, 0x02, 0x00,
+	0x00, 0x01, 0x00, 0xe0,
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+	0xbd, 0xee, 0x02, 0x00,
+	0xcc, 0xee, 0x00, 0xf0, 0xb8, 0x00, 0x10, 0x00, 0xee, 0x10, 0x40, 0x10,
+	0x0e, 0x0e, 0x41, 0x10,
+	0xfc, 0x10, 0x31, 0xca, 0xee, 0x33, 0x10, 0x8a, 0xee, 0x22, 0x11, 0xed,
+	0xdd, 0x0f, 0x00, 0x10,
+	0x30, 0x0e, 0xfb, 0x00, 0x52, 0x11, 0xcd, 0xfd, 0x02, 0x02, 0xba, 0xf0,
+	0x13, 0x02, 0x89, 0xf0,
+	0x13, 0x01, 0xdc, 0x11, 0x34, 0x00, 0xbd, 0x01, 0x46, 0x00, 0xef, 0xff,
+	0x43, 0x22, 0xee, 0xed,
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+	0xbb, 0x00, 0x32, 0x00,
+	0xec, 0xff, 0x63, 0x00, 0xfd, 0x0f, 0x13, 0x01, 0xcb, 0x00, 0x25, 0x01,
+	0xef, 0xce, 0x00, 0x43,
+	0x00, 0xd9, 0x0f, 0x30, 0xfe, 0x20, 0x70, 0x0c, 0x20, 0x53, 0x0f, 0x10,
+	0x22, 0x30, 0x20, 0x1f,
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+	0x20, 0x44, 0x32, 0x08,
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+	0xa2, 0xd3, 0xc4, 0xd2,
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+	0xde, 0xec, 0xde, 0xcf,
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+	0xcd, 0xed, 0xce, 0xdb,
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+	0xc3, 0xe7, 0xc7, 0xd1,
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+	0xce, 0xce, 0xdf, 0xd1,
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+	0x55, 0x43, 0x53, 0x43,
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+	0x33, 0x33, 0x35, 0x23,
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+	0x1d, 0x6d, 0x4d, 0x3d,
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+	0x0f, 0x50, 0x32, 0x10,
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+	0x2e, 0x1d, 0x0a, 0x20,
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+	0x0e, 0x21, 0x53, 0x0e,
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+	0xce, 0x04, 0x56, 0x00,
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+	0x1e, 0x1a, 0x1f, 0x41,
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+	0xfe, 0x30, 0x3e, 0x0c,
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+	0xff, 0x13, 0x37, 0x00,
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+	0x0f, 0xe9, 0xce, 0x10,
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+	0xef, 0xbb, 0x10, 0x01,
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+	0x0f, 0x00, 0x20, 0x10,
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+	0x03, 0x33, 0x0f, 0xb9,
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+	0x01, 0xd0, 0xbf, 0x00,
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+	0x32, 0xf0, 0xf0, 0x24,
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+	0x42, 0x31, 0x21, 0xee,
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+	0x0f, 0x50, 0x20, 0x0d,
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+	0xcc, 0x00, 0x1f, 0xcc,
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+	0x51, 0x21, 0x00, 0x11,
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+	0x02, 0xb9, 0xde, 0xe0,
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+	0x13, 0xf0, 0xdd, 0x03,
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+	0x61, 0x64, 0x1d, 0xf9,
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+	0x42, 0x44, 0x0f, 0x0d,
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+	0x21, 0x22, 0x01, 0xff,
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+	0x20, 0x00, 0x0f, 0x00,
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+	0x00, 0x04, 0x12, 0xf0,
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+	0xe0, 0x53, 0x23, 0x00,
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+	0x0e, 0x2f, 0x10, 0x00,
+	0xfc, 0xfe, 0x0e, 0xe8, 0xdc, 0x50, 0x63, 0x0f, 0x1e, 0x21, 0x10, 0x1f,
+	0x3f, 0x20, 0x30, 0x43,
+	0x11, 0xfe, 0xe0, 0xf0, 0xdb, 0x0f, 0x67, 0x12, 0x00, 0x00, 0x10, 0x00,
+	0x62, 0x0f, 0x10, 0x67,
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+	0x10, 0x20, 0x02, 0x02,
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+	0x0f, 0x00, 0x10, 0x21,
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+	0x26, 0x21, 0xbf, 0xe0,
+	0x01, 0x00, 0xe1, 0x01, 0xbf, 0xcd, 0x22, 0x20, 0xd8, 0xfd, 0x30, 0x20,
+	0x0f, 0x3f, 0x0c, 0xcc,
+	0xed, 0xfe, 0x71, 0x00, 0xb9, 0xff, 0x31, 0x20, 0xce, 0x00, 0x21, 0x00,
+	0x20, 0x20, 0x0d, 0x12,
+	0x47, 0xb0, 0xcd, 0x01, 0x12, 0xff, 0xf0, 0x00, 0x00, 0xf0, 0x0f, 0x1f,
+	0x41, 0x01, 0xbb, 0x02,
+	0x57, 0xf2, 0x9b, 0xf0, 0x00, 0xf1, 0xf1, 0xe2, 0xe0, 0xe0, 0xf1, 0x01,
+	0x02, 0x0e, 0xee, 0x00,
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+	0x24, 0x14, 0x8c, 0xed,
+	0x13, 0x33, 0x0f, 0xfc, 0xfd, 0xc8, 0xfe, 0x40, 0x31, 0xee, 0x0f, 0x42,
+	0x00, 0xf0, 0x11, 0x22,
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+	0x3d, 0x4d, 0x50, 0x44,
+	0x2f, 0x08, 0x1f, 0x1f, 0xed, 0x0e, 0x70, 0x0f, 0xfc, 0x0f, 0x00, 0xe8,
+	0x0f, 0x60, 0x00, 0x00,
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+	0x60, 0x23, 0x10, 0x10,
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+	0x31, 0x00, 0x10, 0x02,
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+	0x00, 0xd0, 0xbd, 0xf0,
+	0xe0, 0xcb, 0xf1, 0x06, 0x02, 0xe0, 0xae, 0x0f, 0x21, 0x23, 0x24, 0x11,
+	0x00, 0x00, 0x22, 0x46,
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+	0x0d, 0x3f, 0x1e, 0xcb,
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+	0x0f, 0xfe, 0x0f, 0xfe,
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+	0x21, 0x21, 0x0f, 0xff,
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+	0x24, 0x01, 0xef, 0xf0,
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+	0xda, 0x8b, 0x0f, 0x20,
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+	0x32, 0x0e, 0x2f, 0x20,
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+	0xf1, 0x02, 0xf2, 0x04,
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+	0x12, 0x00, 0x12, 0xd1,
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+	0x13, 0x62, 0x0f, 0xfe,
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+	0xee, 0x20, 0x31, 0x63,
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+	0x13, 0xee, 0x0d, 0x32,
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+	0xdd, 0x63, 0x16, 0x0f,
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+	0x00, 0xac, 0x10, 0x24,
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+	0x35, 0xdf, 0x20, 0x43,
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+	0x06, 0xce, 0x0f, 0x40,
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+	0x00, 0xac, 0x00, 0x00,
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+	0x33, 0xfe, 0x8a, 0xde,
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+	0x00, 0xfa, 0x10, 0x50,
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+	0x11, 0x10, 0x01, 0x31,
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+	0xef, 0xef, 0x31, 0x11,
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+	0x00, 0xe0, 0x35, 0x13,
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+	0x00, 0x54, 0x00, 0xff,
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+	0x01, 0x57, 0xee, 0xff,
+	0x32, 0x01, 0xde, 0x20, 0x30, 0xf1, 0x31, 0x2f, 0x0e, 0x40, 0x0e, 0xc8,
+	0x10, 0x75, 0x40, 0xdb,
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+	0x21, 0xdd, 0xee, 0x76,
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+	0xef, 0xcd, 0x15, 0xdf,
+	0xa9, 0x21, 0x32, 0xbe, 0xfe, 0x00, 0x00, 0xdf, 0x10, 0x0f, 0xfd, 0x01,
+	0xdf, 0xcc, 0x00, 0x03,
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+	0xdd, 0xed, 0x03, 0x00,
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+	0xf0, 0x01, 0xe0, 0x9c,
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+	0x0f, 0x00, 0xbd, 0x00,
+	0xff, 0xdc, 0x21, 0x00, 0x8b, 0x00, 0x0e, 0x0e, 0x0e, 0xdd, 0xfc, 0x11,
+	0x10, 0xcb, 0x20, 0x13,
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+	0x12, 0xfe, 0xbc, 0x10,
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+	0x02, 0xef, 0xcd, 0x0f,
+	0x13, 0x03, 0xf9, 0x0e, 0x31, 0xd1, 0x9f, 0x0f, 0x01, 0xf0, 0xed, 0xf0,
+	0x04, 0xdd, 0x9b, 0x01,
+	0x12, 0xde, 0xef, 0x00, 0x10, 0x0d, 0xed, 0xcd, 0x23, 0x17, 0x04, 0x0e,
+	0xbc, 0xff, 0x00, 0x01,
+	0x04, 0x02, 0x0d, 0x2b, 0x11, 0x31, 0x10, 0xeb, 0x5f, 0x63, 0xe0, 0xcd,
+	0x20, 0x32, 0xf0, 0xf0,
+	0x0f, 0xcc, 0x00, 0x77, 0x00, 0x00, 0xef, 0xec, 0x20, 0x75, 0x00, 0xee,
+	0x00, 0x23, 0x00, 0xee,
+	0x40, 0x24, 0xfe, 0x1e, 0x52, 0x26, 0x01, 0x0e, 0xfe, 0x10, 0x21, 0x13,
+	0x04, 0x11, 0x0f, 0xed,
+	0x31, 0x74, 0x1d, 0x10, 0x23, 0x10, 0xed, 0x01, 0x17, 0x01, 0xcd, 0x10,
+	0x32, 0x01, 0x22, 0x21,
+	0xf0, 0x73, 0x77, 0x10, 0x00, 0x24, 0x10, 0x0f, 0x36, 0x54, 0x00, 0xee,
+	0x34, 0x10, 0x90, 0x00,
+	0x47, 0x00, 0xbd, 0x13, 0x32, 0xef, 0x0e, 0x42, 0x14, 0x00, 0x23, 0x23,
+	0x10, 0x10, 0x00, 0x46,
+	0x10, 0x02, 0x11, 0x01, 0x75, 0x10, 0x01, 0x00, 0xf0, 0x11, 0x46, 0x00,
+	0x0f, 0x62, 0x11, 0x00,
+	0x35, 0x44, 0x00, 0x10, 0x60, 0x11, 0x0e, 0x00, 0x24, 0x12, 0x00, 0xff,
+	0x23, 0x30, 0x0f, 0x02,
+	0x23, 0x1d, 0x21, 0x77, 0x55, 0x0f, 0x0f, 0x02, 0x45, 0x0f, 0xfd, 0x34,
+	0x10, 0xf0, 0x21, 0x63,
+	0xff, 0xfe, 0x01, 0x35, 0x11, 0x1f, 0x0e, 0xe0, 0x26, 0x51, 0xf0, 0x00,
+	0x12, 0x22, 0x22, 0xff,
+	0x54, 0x57, 0xef, 0xee, 0x13, 0x53, 0x00, 0x04, 0x11, 0x0f, 0xd0, 0x11,
+	0x16, 0x00, 0xbf, 0x0f,
+	0x23, 0xf0, 0xce, 0x42, 0x34, 0xef, 0x10, 0x32, 0xbb, 0x22, 0x02, 0xe0,
+	0x00, 0xf1, 0x03, 0x25,
+	0xbe, 0xee, 0x02, 0x10, 0xef, 0x0d, 0x1f, 0xfe, 0x0e, 0x73, 0x30, 0xd8,
+	0x20, 0x73, 0x10, 0x00,
+	0x0f, 0xee, 0x2c, 0x0b, 0x10, 0x74, 0x0d, 0xec, 0x12, 0x10, 0x0e, 0xfd,
+	0x2f, 0x0d, 0x3f, 0x33,
+	0x35, 0xfe, 0xea, 0xcf, 0x50, 0x41, 0x00, 0xe0, 0xef, 0xf2, 0xf2, 0x67,
+	0x42, 0xef, 0xe0, 0x35,
+	0x0f, 0x0e, 0x12, 0xf1, 0xaf, 0x00, 0x02, 0x23, 0x00, 0xcc, 0x0f, 0xf0,
+	0x04, 0x02, 0x0d, 0xfe,
+	0x07, 0x34, 0x22, 0x90, 0xef, 0x14, 0x01, 0xd0, 0xa0, 0xe6, 0xa1, 0x00,
+	0xf3, 0x00, 0xe3, 0xef,
+	0xec, 0x02, 0x46, 0x01, 0x01, 0x00, 0xff, 0x00, 0xf2, 0x54, 0x21, 0xde,
+	0x00, 0xce, 0xa8, 0x01,
+	0x13, 0xf0, 0xde, 0xe0, 0x1f, 0x0e, 0x20, 0x52, 0xef, 0x0a, 0x30, 0xf0,
+	0xef, 0x51, 0x40, 0x0e,
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+	0x0f, 0xfe, 0x20, 0x11,
+	0x0e, 0x30, 0x2e, 0x30, 0x60, 0x0e, 0xda, 0x00, 0x53, 0x02, 0x2e, 0x0e,
+	0x12, 0xd0, 0x00, 0x27,
+	0x20, 0x32, 0x61, 0xdc, 0xbb, 0xf0, 0x05, 0x0f, 0x00, 0x01, 0x03, 0xbc,
+	0xde, 0x13, 0x32, 0xef,
+	0x00, 0x45, 0xee, 0xac, 0x01, 0x13, 0x00, 0x11, 0x43, 0xd0, 0xcc, 0xe1,
+	0x07, 0xf0, 0xbc, 0x12,
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+	0x04, 0xf0, 0x8f, 0xbb,
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+	0xcb, 0xf0, 0x10, 0x42,
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+	0x7f, 0x10, 0xe0, 0x20,
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+	0x31, 0x11, 0x80, 0x09,
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+	0x11, 0x12, 0x0f, 0xf0,
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+	0x11, 0x40, 0xed, 0xe0,
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+	0x03, 0x14, 0x1d, 0xff,
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+	0xfd, 0x2c, 0xbb, 0xdb,
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+	0x50, 0x15, 0x0f, 0xeb,
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+	0x00, 0x10, 0x00, 0x02,
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+	0x00, 0x06, 0xe2, 0xff,
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+	0x31, 0x62, 0xff, 0xca,
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+	0xf5, 0x16, 0xf0, 0xfc,
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+	0x74, 0x12, 0xcc, 0xec,
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+	0xcc, 0xfb, 0x32, 0x45,
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+	0x40, 0x20, 0xf1, 0xd3,
+	0xec, 0x0c, 0x00, 0x00, 0xdd, 0xae, 0x00, 0x15, 0x70, 0x3f, 0x10, 0xbb,
+	0xff, 0x10, 0x00, 0xe3,
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+	0x02, 0xff, 0x70, 0x41,
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+	0x12, 0x31, 0xd0, 0x0e,
+	0x10, 0x0d, 0xff, 0xc0, 0x45, 0x10, 0x0e, 0x30, 0x20, 0x01, 0xe0, 0x8e,
+	0xff, 0x07, 0x42, 0xf2,
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+	0x11, 0xf8, 0xeb, 0x10,
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+	0x01, 0x2d, 0xeb, 0xd1,
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+	0xff, 0xe0, 0x1f, 0x21,
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+	0x0e, 0xeb, 0xf0, 0x54,
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+	0x00, 0xac, 0xe0, 0x13,
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+	0x30, 0x30, 0xfa, 0xff,
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+	0xdb, 0x00, 0x42, 0x11,
+	0x51, 0x2e, 0x4f, 0x3e, 0x0b, 0x10, 0x02, 0x31, 0x1b, 0x5f, 0x42, 0x01,
+	0x0f, 0xff, 0xd2, 0xf5,
+	0xf2, 0xef, 0x8a, 0xcf, 0x10, 0x31, 0x21, 0xee, 0xce, 0x0e, 0xf0, 0x05,
+	0x24, 0x31, 0xdc, 0xbf,
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+	0x12, 0x24, 0xfe, 0x0d,
+	0x1f, 0x7f, 0xe0, 0xa5, 0x60, 0x0b, 0xb0, 0x25, 0x00, 0x1f, 0x0d, 0xbf,
+	0x10, 0x32, 0xf3, 0xb5,
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+	0x00, 0x00, 0x7f, 0x01,
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+	0x0f, 0xdc, 0x0a, 0xfc,
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+	0x00, 0xaf, 0x0d, 0x01,
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+	0x01, 0x06, 0xf0, 0x00,
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+	0x00, 0xfe, 0x41, 0x43,
+	0xf0, 0x0d, 0xfd, 0xcf, 0x7b, 0x1f, 0x00, 0xd0, 0x01, 0x27, 0xf3, 0xcc,
+	0x0f, 0x00, 0xe1, 0xe0,
+	0xeb, 0xbf, 0xf0, 0x01, 0x00, 0x10, 0x32, 0x73, 0x25, 0x10, 0xed, 0x0e,
+	0x77, 0x25, 0xe0, 0x0d,
+	0x20, 0x0f, 0xf0, 0x02, 0xef, 0xdd, 0xee, 0xfd, 0x00, 0xcf, 0x05, 0x45,
+	0x71, 0x32, 0xe0, 0xeb,
+	0xe0, 0xfe, 0x20, 0x11, 0x1f, 0x00, 0x00, 0x40, 0x52, 0x01, 0xe1, 0xe1,
+	0x0d, 0xc8, 0x01, 0x26,
+	0x00, 0xcf, 0xdd, 0x0f, 0x20, 0x60, 0x13, 0x03, 0xfe, 0xda, 0x1f, 0x60,
+	0x01, 0x03, 0xf0, 0x0e,
+	0x30, 0x30, 0x00, 0xcc, 0xe1, 0xe7, 0x60, 0xef, 0x82, 0x00, 0xf0, 0x0c,
+	0x10, 0x25, 0x00, 0xff,
+	0xdf, 0xef, 0x50, 0x23, 0xa0, 0x0d, 0x0c, 0x31, 0x03, 0x00, 0xdc, 0x24,
+	0x25, 0x0e, 0x00, 0x10,
+	0xde, 0xd0, 0x03, 0x13, 0xe0, 0x8c, 0xcf, 0x12, 0x24, 0x14, 0x00, 0x52,
+	0xf2, 0xbc, 0x30, 0x20,
+	0x02, 0xde, 0x01, 0x37, 0x02, 0xad, 0xff, 0x03, 0x10, 0x09, 0xee, 0x0f,
+	0x10, 0x04, 0x00, 0x0d,
+	0x0c, 0x72, 0x34, 0x0e, 0xdc, 0x53, 0x42, 0x0f, 0x0e, 0x12, 0x00, 0x02,
+	0x17, 0x23, 0x0d, 0xda,
+	0xf0, 0x24, 0x01, 0x9f, 0xdd, 0x0d, 0x10, 0x64, 0x11, 0x00, 0x1b, 0x10,
+	0x1f, 0xef, 0x00, 0x42,
+	0x00, 0xfc, 0xfd, 0xaf, 0xd0, 0xe5, 0x01, 0x43, 0x43, 0x10, 0x01, 0x01,
+	0x14, 0x00, 0x11, 0x00,
+	0x61, 0xf7, 0xaa, 0xff, 0xf3, 0xf3, 0xc3, 0xff, 0xf9, 0x8e, 0xf0, 0x36,
+	0x73, 0x1c, 0xc8, 0xee,
+	0x00, 0x35, 0x20, 0xdf, 0xa0, 0x00, 0xe2, 0xf0, 0x00, 0xd0, 0x12, 0x04,
+	0xef, 0x8f, 0xd0, 0x10,
+	0x2c, 0xdb, 0xd0, 0x63, 0x02, 0x1f, 0x1b, 0x00, 0x21, 0xd0, 0x12, 0x37,
+	0xf2, 0x00, 0xdc, 0xe0,
+	0xb0, 0x01, 0x90, 0xb1, 0x33, 0x21, 0xeb, 0xde, 0x1f, 0x30, 0x02, 0x20,
+	0x1c, 0x0f, 0x53, 0x47,
+	0x20, 0xfd, 0xef, 0x20, 0x2f, 0x0e, 0xf0, 0xcf, 0x00, 0x20, 0x10, 0xbf,
+	0x14, 0x65, 0x0f, 0xfa,
+	0xf0, 0xdf, 0x0f, 0x00, 0x00, 0xef, 0x52, 0x27, 0x21, 0x0e, 0x5e, 0x10,
+	0x01, 0xb0, 0x0f, 0x0c,
+	0xe0, 0x02, 0x03, 0x01, 0xdd, 0xec, 0x53, 0x27, 0xe0, 0xf2, 0x21, 0x00,
+	0x20, 0x31, 0x23, 0x12,
+	0x9e, 0xdf, 0xce, 0x0e, 0x1e, 0x1e, 0x00, 0x30, 0xb5, 0xf6, 0x53, 0x81,
+	0x1d, 0x41, 0xa3, 0x17,
+	0x03, 0x0d, 0x0d, 0xdf, 0x1e, 0x40, 0x07, 0xe5, 0xfd, 0x2c, 0x30, 0x04,
+	0xa0, 0x1f, 0x30, 0xf0,
+	0xe0, 0x0f, 0x0e, 0x20, 0x15, 0x01, 0xde, 0xef, 0x00, 0xdd, 0xb9, 0x00,
+	0x02, 0xe2, 0x70, 0x09,
+	0xef, 0x00, 0x3f, 0x31, 0x02, 0x01, 0x2f, 0xfe, 0xfe, 0x01, 0x63, 0x20,
+	0x00, 0xee, 0xfe, 0x11,
+	0x03, 0x00, 0x33, 0x2f, 0xe8, 0xf8, 0x00, 0x60, 0x44, 0xd0, 0xd0, 0xe0,
+	0xcf, 0x20, 0x24, 0xe0,
+	0xfd, 0x9f, 0xc1, 0x12, 0x41, 0xe0, 0x06, 0x21, 0xdd, 0xca, 0x00, 0x22,
+	0x21, 0xe0, 0xe0, 0x0e,
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+	0x14, 0x12, 0xdd, 0xbe,
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+	0xf0, 0xe0, 0x0d, 0xe5,
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+	0xec, 0x20, 0x50, 0x11,
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+	0x24, 0x11, 0xef, 0x0e,
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+	0x23, 0x13, 0x01, 0x24,
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+	0x10, 0x0e, 0xd0, 0x13,
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+	0xff, 0x80, 0xa1, 0x00,
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+	0x0b, 0xec, 0xdd, 0x0e,
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+	0x60, 0x21, 0x0c, 0xdd,
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+	0x4e, 0x02, 0xe2, 0x8f,
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+	0x02, 0x02, 0x12, 0x03,
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+	0xe0, 0x8e, 0xff, 0x00,
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+	0x42, 0x20, 0x0e, 0x0f,
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+	0x20, 0x31, 0xef, 0x00,
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+	0xe2, 0xd0, 0x0e, 0x01,
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+	0x31, 0xe2, 0xe1, 0x0f,
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+	0x32, 0xf1, 0x88, 0x9d,
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+	0x23, 0x22, 0xfb, 0xfe,
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+	0xef, 0x01, 0x16, 0xbd,
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+	0x0c, 0xee, 0x31, 0x42,
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+	0xea, 0xbf, 0x10, 0x73,
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+	0xeb, 0xde, 0x00, 0x54,
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+	0x42, 0xd0, 0xff, 0x00,
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+	0xd0, 0x45, 0xdf, 0x9a,
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+	0x42, 0x10, 0xeb, 0x00,
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+	0x30, 0x12, 0x0e, 0xfa,
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+	0x23, 0xf1, 0xcb, 0x20,
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+	0x0f, 0xcf, 0xfc, 0x0c,
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+	0x2f, 0x3f, 0x04, 0x03,
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+	0xf1, 0x12, 0x1e, 0x1b,
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+	0x01, 0x17, 0x53, 0x20,
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+	0xcc, 0x0f, 0x4f, 0x13,
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+	0x02, 0xe0, 0x27, 0x72,
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+	0x8e, 0xe1, 0xff, 0xe9,
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+	0x2e, 0x1b, 0x1f, 0x00,
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+	0xf3, 0xed, 0xcd, 0x00,
+	0x32, 0x01, 0x0f, 0x00, 0x26, 0x41, 0x0d, 0xec, 0x40, 0x77, 0x12, 0xec,
+	0xff, 0x00, 0xfe, 0xee,
+	0x13, 0x33, 0xe3, 0xd0, 0xdf, 0x1f, 0x40, 0xdd, 0xcb, 0xe0, 0xcd, 0xcd,
+	0x0c, 0x0a, 0x00, 0xff,
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+	0x27, 0x35, 0x1f, 0x0f,
+	0xf0, 0xe0, 0x02, 0xf0, 0x20, 0x75, 0xd4, 0xbe, 0x0b, 0xef, 0xe0, 0x04,
+	0x61, 0x21, 0xfe, 0xe1,
+	0x12, 0x40, 0x10, 0xf1, 0xcc, 0xfb, 0x04, 0x64, 0x00, 0xed, 0x00, 0x03,
+	0x02, 0x11, 0x0d, 0x1f,
+	0x73, 0x57, 0x03, 0x0d, 0xfe, 0x01, 0x14, 0xff, 0xed, 0xf2, 0x02, 0xee,
+	0xec, 0xf0, 0xec, 0x22,
+	0x77, 0x25, 0xcb, 0xff, 0x0f, 0x50, 0x20, 0xf0, 0xff, 0x0f, 0x0e, 0x0d,
+	0x02, 0x34, 0x10, 0x01,
+	0xe1, 0xd0, 0x1e, 0x60, 0xf5, 0xb0, 0x0f, 0x22, 0xff, 0xb8, 0x1f, 0x2f,
+	0x30, 0xf2, 0xdf, 0xfd,
+	0xdc, 0xe1, 0x04, 0x44, 0x10, 0x8e, 0xe0, 0x00, 0x23, 0xd0, 0xe1, 0x37,
+	0x72, 0x21, 0xee, 0xed,
+	0x21, 0x21, 0xe0, 0x17, 0x01, 0xab, 0x00, 0xde, 0xf0, 0xef, 0x0f, 0x74,
+	0x13, 0xfc, 0xde, 0x12,
+	0x02, 0xaf, 0x01, 0x13, 0x0d, 0x00, 0xf4, 0x21, 0x42, 0x0e, 0xfb, 0x00,
+	0x01, 0x32, 0x0f, 0x0a,
+	0x0f, 0x00, 0x00, 0x00, 0x20, 0x74, 0x22, 0x00, 0xec, 0xbc, 0xff, 0x31,
+	0x54, 0xcb, 0xef, 0x00,
+	0xfd, 0xdc, 0x00, 0x20, 0x11, 0x10, 0x40, 0xfc, 0xdc, 0x6f, 0x21, 0x0f,
+	0xfa, 0x30, 0x76, 0x20,
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+	0x30, 0x0e, 0xdf, 0xb1,
+	0x16, 0x40, 0xe0, 0xcd, 0x00, 0x00, 0xe0, 0xf1, 0x02, 0xef, 0xe3, 0xcf,
+	0xb8, 0xeb, 0x00, 0x15,
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+	0xef, 0x11, 0x50, 0x21,
+	0x22, 0x01, 0xf0, 0x40, 0x4d, 0xfb, 0xe0, 0x37, 0x22, 0xbe, 0xee, 0x00,
+	0x12, 0xe0, 0xe0, 0xe2,
+	0xbe, 0x2e, 0x70, 0x52, 0x04, 0x00, 0xe9, 0x2f, 0x52, 0x0f, 0xd8, 0x00,
+	0x23, 0x12, 0xed, 0xfe,
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+	0xff, 0x0e, 0x00, 0x20,
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+	0xce, 0x00, 0x26, 0x72,
+	0x00, 0xca, 0x10, 0x34, 0x21, 0xf0, 0xbd, 0x01, 0x03, 0x00, 0xf0, 0x15,
+	0xef, 0xc0, 0x01, 0xf1,
+	0xac, 0xef, 0x01, 0x14, 0x50, 0x01, 0x0d, 0x00, 0x17, 0x33, 0x0d, 0xbd,
+	0x00, 0x00, 0xfd, 0x00,
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+	0x2d, 0x0d, 0x1e, 0x4c,
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+	0x67, 0x23, 0xdd, 0xbc,
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+	0xcc, 0xef, 0x00, 0x0f,
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+	0xfe, 0xdc, 0x0f, 0xff,
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+	0x0f, 0x77, 0x12, 0xed,
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+	0xdf, 0x15, 0xf0, 0xf0,
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+	0x22, 0x12, 0xaa, 0x12,
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+	0x00, 0x00, 0x00, 0xee,
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+	0x11, 0xdc, 0xbb, 0x00,
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+	0xbd, 0x00, 0x33, 0x01,
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+	0xff, 0x22, 0x01, 0xf0,
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+	0x10, 0xfe, 0xfe, 0x10,
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+	0x33, 0x12, 0xff, 0xcc,
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+	0x66, 0x00, 0xee, 0x00,
+	0x01, 0x22, 0x32, 0xcc, 0xed, 0x55, 0x10, 0xdd, 0xf0, 0x00, 0x21, 0x10,
+	0xff, 0x01, 0x00, 0xdd,
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+	0xb8, 0xee, 0x00, 0x00,
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+	0xa8, 0x00, 0x22, 0x43,
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+	0x32, 0xec, 0xbb, 0xff,
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+	0x00, 0xfe, 0x0f, 0x00,
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+	0xdb, 0xcb, 0x41, 0x65,
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+	0xde, 0xbd, 0x02, 0x77,
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+	0x25, 0x00, 0xc0, 0xe0,
+	0x12, 0x02, 0xcf, 0xef, 0x46, 0x11, 0xce, 0x13, 0x67, 0x0f, 0xee, 0x11,
+	0x23, 0x12, 0x10, 0x0f,
+	0xcd, 0x11, 0x77, 0x00, 0xee, 0x00, 0x23, 0x01, 0xed, 0xff, 0xff, 0x10,
+	0x41, 0x13, 0xff, 0xf0,
+	0x22, 0xfa, 0x9b, 0x10, 0x55, 0x10, 0x00, 0x11, 0x00, 0xfe, 0x00, 0x20,
+	0x00, 0x54, 0x23, 0x00,
+	0x0f, 0x0f, 0xec, 0xfe, 0x77, 0x23, 0x0f, 0x00, 0x01, 0xff, 0x02, 0x11,
+	0xef, 0x34, 0x77, 0x00,
+	0x01, 0x11, 0x00, 0x10, 0x23, 0x22, 0x22, 0x22, 0x01, 0x0f, 0x01, 0x67,
+	0x11, 0x12, 0x00, 0xdf,
+	0x44, 0x32, 0x00, 0x11, 0xf1, 0xf2, 0x77, 0x42, 0xf0, 0xf0, 0xf0, 0x01,
+	0x24, 0x44, 0x01, 0xf0,
+	0xf0, 0xe0, 0x01, 0x66, 0x12, 0x11, 0x00, 0xdf, 0xcf, 0x00, 0x67, 0x32,
+	0x00, 0xe0, 0x21, 0x12,
+	0x32, 0x75, 0x00, 0xf0, 0x12, 0x54, 0x00, 0xfe, 0x33, 0x34, 0x12, 0x32,
+	0x11, 0x0f, 0xee, 0xfe,
+	0x77, 0x43, 0xfe, 0xdc, 0x22, 0x34, 0xfe, 0xee, 0x11, 0x00, 0x00, 0x40,
+	0x0e, 0x10, 0x77, 0xff,
+	0xdc, 0x10, 0x21, 0x10, 0x0f, 0x0f, 0x00, 0x20, 0x00, 0x0f, 0x40, 0x23,
+	0xee, 0xdb, 0x76, 0x32,
+	0xbc, 0xde, 0x00, 0x10, 0x00, 0x00, 0xf0, 0xff, 0xff, 0xfe, 0x00, 0xfe,
+	0x98, 0xff, 0x00, 0x42,
+	0x11, 0x0f, 0xde, 0xa9, 0xee, 0x10, 0x22, 0xef, 0x0f, 0xee, 0x00, 0x33,
+	0xcd, 0xa8, 0x00, 0x32,
+	0x10, 0x00, 0xdf, 0xdd, 0xdc, 0x41, 0x33, 0xcc, 0xdc, 0x00, 0x0f, 0xbd,
+	0x00, 0x11, 0x0e, 0xee,
+	0xcc, 0xff, 0x00, 0x0f, 0xcb, 0x0f, 0x31, 0xed, 0xcd, 0x00, 0x20, 0xfa,
+	0xce, 0x30, 0x22, 0xbb,
+	0xee, 0x00, 0x9c, 0xee, 0x11, 0x01, 0xcb, 0xff, 0x01, 0xdc, 0xac, 0x00,
+	0x01, 0xf0, 0xf0, 0x00,
+	0xff, 0xdc, 0xbb, 0xef, 0x11, 0x22, 0xee, 0x9b, 0xed, 0xfe, 0x00, 0x01,
+	0x12, 0x11, 0xdc, 0x9a,
+	0xf0, 0x0f, 0xff, 0x00, 0x10, 0x00, 0xef, 0xdc, 0xac, 0x0f, 0x01, 0xff,
+	0xff, 0xff, 0xde, 0xdb,
+	0x00, 0x13, 0xce, 0xbb, 0x00, 0x00, 0x00, 0x00, 0xff, 0xcc, 0x10, 0x22,
+	0xdc, 0xed, 0x00, 0xbd,
+	0xfe, 0x35, 0x53, 0x0f, 0xb8, 0xef, 0x10, 0x00, 0x10, 0x00, 0x00, 0x00,
+	0xe3, 0x01, 0x8f, 0x01,
+	0x04, 0xd4, 0xf5, 0xd5, 0x25, 0xd7, 0x93, 0xd8, 0x32, 0x67, 0xf0, 0xcd,
+	0x22, 0x24, 0x00, 0x11,
+	0x10, 0x0f, 0x00, 0x00, 0x01, 0x55, 0x21, 0x31, 0x21, 0xf0, 0xac, 0x00,
+	0x25, 0x10, 0x41, 0x45,
+	0xf0, 0xde, 0xdc, 0xdd, 0x30, 0x77, 0x20, 0xef, 0xf0, 0xff, 0x00, 0x11,
+	0x0f, 0xef, 0x43, 0x55,
+	0xef, 0xbc, 0xff, 0x00, 0x23, 0x01, 0x00, 0x00, 0x00, 0xdf, 0x01, 0x14,
+	0xf0, 0xdd, 0x01, 0x52,
+	0x00, 0xca, 0xff, 0x44, 0x42, 0xdb, 0xef, 0x53, 0x33, 0xee, 0xff, 0x02,
+	0xff, 0x01, 0x77, 0x00,
+	0xff, 0xdd, 0x00, 0x57, 0x01, 0xdd, 0xcd, 0x22, 0x55, 0x00, 0x00, 0xee,
+	0xfe, 0x33, 0x11, 0xfe,
+	0x00, 0xfd, 0xed, 0x77, 0xf0, 0xec, 0x12, 0x31, 0xde, 0x0f, 0x12, 0x00,
+	0xf0, 0xf0, 0x10, 0x77,
+	0x00, 0xcd, 0xff, 0x00, 0x44, 0x22, 0x00, 0xbb, 0x0f, 0x22, 0x11, 0xfe,
+	0x00, 0xff, 0x11, 0x31,
+	0x00, 0x12, 0x00, 0xc8, 0xce, 0x31, 0x53, 0x00, 0xff, 0x21, 0xfe, 0xb9,
+	0x00, 0x54, 0x23, 0xef,
+	0xed, 0x12, 0x44, 0x00, 0xfe, 0x02, 0x02, 0x44, 0x43, 0xff, 0xed, 0x00,
+	0x11, 0xff, 0xb9, 0x00,
+	0x22, 0x54, 0x10, 0xed, 0xbb, 0x12, 0x33, 0xed, 0x00, 0x24, 0x00, 0xf0,
+	0xf0, 0xcd, 0x11, 0x77,
+	0x01, 0xff, 0xee, 0x02, 0x00, 0xde, 0x00, 0x34, 0x00, 0xf0, 0x35, 0xbf,
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+	0x41, 0x30, 0xf8, 0xde,
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+	0x09, 0xf0, 0x01, 0xde,
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+	0x0f, 0xed, 0x10, 0x10,
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+	0x20, 0x40, 0x50, 0x27,
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+	0x04, 0x01, 0x40, 0x3e,
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+	0x1e, 0x0b, 0x21, 0x01,
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+	0xe0, 0xdb, 0x1f, 0x41,
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+	0xf0, 0xb0, 0x0d, 0x0d,
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+	0xb0, 0xb1, 0x03, 0x01,
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+	0x27, 0xe5, 0xcf, 0x0d,
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+	0x0f, 0xda, 0x00, 0x4f,
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+	0x02, 0xc0, 0xdf, 0x0d,
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+	0x30, 0x3b, 0x10, 0x06,
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+	0xf1, 0xc0, 0x01, 0x00,
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+	0xe0, 0x12, 0x61, 0x2d,
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+	0x2f, 0xfb, 0xe0, 0x15,
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+	0x01, 0xd3, 0xed, 0x08,
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+	0x0f, 0xde, 0xff, 0x00,
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+	0x54, 0x00, 0xfb, 0xe0,
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+	0x0f, 0x10, 0x01, 0x01,
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+	0xe0, 0xe7, 0xf0, 0x0a,
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+	0x00, 0xe2, 0x03, 0x0d,
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+	0x43, 0x00, 0xad, 0xe0,
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+	0x31, 0x45, 0xef, 0xed,
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+	0xcf, 0xff, 0xf0, 0x1f,
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+	0xed, 0x20, 0x20, 0x30,
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+	0x10, 0xf1, 0xbf, 0x2b,
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+	0x22, 0xe9, 0x8d, 0xe0,
+	0x15, 0x41, 0x00, 0xf0, 0x02, 0x3f, 0xfc, 0xcf, 0xe0, 0xf1, 0x56, 0x52,
+	0x0e, 0xca, 0x0e, 0x00,
+	0x11, 0x05, 0xd0, 0xf0, 0x2e, 0x20, 0x00, 0xc1, 0xe2, 0xf1, 0x33, 0x72,
+	0xfc, 0xcb, 0x00, 0x22,
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+	0x20, 0x04, 0x02, 0x0d,
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+	0xed, 0xfa, 0x0f, 0x0f,
+	0x00, 0x0f, 0x0d, 0xbf, 0xf2, 0x27, 0x40, 0x0e, 0x0e, 0x13, 0x16, 0xf0,
+	0xff, 0xdb, 0x01, 0xe4,
+	0x12, 0x40, 0xec, 0xdf, 0x30, 0x05, 0x80, 0x9f, 0x1e, 0x21, 0x14, 0x00,
+	0x0d, 0xd0, 0xf0, 0x01,
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+	0xf0, 0xc1, 0xf3, 0x10,
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+	0x02, 0x24, 0xee, 0xee,
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+	0x06, 0x44, 0x2f, 0x29,
+	0xe0, 0xf0, 0xe2, 0x40, 0x60, 0x20, 0x0f, 0xdf, 0x1f, 0x31, 0xd2, 0xb2,
+	0x0d, 0x3b, 0x20, 0xc4,
+	0xc1, 0xfc, 0xcf, 0xf0, 0x02, 0x22, 0x01, 0xbe, 0x0b, 0x3f, 0x20, 0xff,
+	0xc0, 0x00, 0x77, 0x20,
+	0x31, 0xd0, 0xd0, 0x23, 0x44, 0x00, 0xe2, 0xf3, 0x01, 0x5e, 0xfe, 0xc0,
+	0xf1, 0xea, 0x0b, 0x2f,
+	0x3e, 0xf0, 0x90, 0x04, 0x0f, 0x19, 0x00, 0x00, 0xd3, 0x01, 0x40, 0x00,
+	0xee, 0x0e, 0x50, 0x11,
+	0x03, 0x00, 0x34, 0xdd, 0xb8, 0x0c, 0x52, 0x64, 0x11, 0x0f, 0xee, 0x00,
+	0x01, 0x01, 0x01, 0xe0,
+	0x00, 0xf1, 0xf6, 0x32, 0x20, 0xac, 0x0f, 0x40, 0xe1, 0xb3, 0x04, 0x52,
+	0x0f, 0xd8, 0x0f, 0x00,
+	0x4f, 0x00, 0x02, 0xc3, 0xd0, 0x00, 0x00, 0x40, 0x10, 0x2b, 0x11, 0x07,
+	0xb0, 0x00, 0xff, 0x0e,
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+	0x01, 0xd7, 0xf5, 0xff,
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+	0xfe, 0xe8, 0x20, 0x32,
+	0x40, 0x00, 0x8f, 0xcc, 0x01, 0x22, 0x22, 0x21, 0x00, 0xec, 0xdc, 0xb0,
+	0x02, 0x36, 0x30, 0x0c,
+	0xe0, 0x01, 0x11, 0x61, 0x00, 0x2f, 0x1d, 0xd0, 0xe0, 0x3e, 0x6d, 0x0f,
+	0xde, 0xf5, 0x07, 0xf1,
+	0x0e, 0xed, 0xd0, 0x25, 0x72, 0x00, 0xbc, 0xdf, 0x00, 0x55, 0x10, 0x20,
+	0xf0, 0xc0, 0xeb, 0x00,
+	0x30, 0x32, 0xf2, 0xb1, 0x0f, 0x3a, 0xf0, 0xf0, 0x00, 0xf0, 0xf0, 0xb7,
+	0xe0, 0x00, 0x7e, 0x1f,
+	0x02, 0xe5, 0x00, 0xef, 0x32, 0x71, 0x20, 0xec, 0xe0, 0x01, 0xf4, 0x12,
+	0x4f, 0x0c, 0xbf, 0x01,
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+	0x0f, 0xd0, 0xd0, 0x00,
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+	0x50, 0x00, 0xc2, 0x01,
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+	0x11, 0xf0, 0xdd, 0x10,
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+	0x81, 0xc0, 0xfc, 0xff,
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+	0xe9, 0xe0, 0x00, 0x00,
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+	0x75, 0x51, 0x2d, 0xcd,
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+	0x22, 0xe0, 0xb1, 0xf0,
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+	0xe1, 0x00, 0x1f, 0x00,
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+	0xcd, 0x00, 0x40, 0x11,
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+	0xde, 0xff, 0x34, 0x34,
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+	0xf0, 0xee, 0xef, 0x10,
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+	0xe6, 0xf1, 0x0e, 0x1e,
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+	0x2e, 0xec, 0xd7, 0x22,
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+	0x0a, 0xcf, 0xe0, 0x03,
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+	0xe0, 0x04, 0x30, 0x1c,
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+	0xd0, 0xa0, 0x0a, 0x1e,
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+	0x01, 0xde, 0x0d, 0x30,
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+	0xe0, 0xbc, 0x0f, 0x1e,
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+	0xef, 0xec, 0x1e, 0x02,
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+	0x7f, 0x20, 0x05, 0xa2,
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+	0xe1, 0x00, 0xf0, 0x02,
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+	0xdb, 0xcf, 0xf0, 0x10,
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+	0x01, 0x03, 0x20, 0x00,
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+	0x12, 0xe4, 0xe0, 0xfe,
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+	0xb1, 0x20, 0x1d, 0x0e,
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+	0x01, 0x3e, 0x1f, 0xe0,
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+	0x00, 0x0e, 0xef, 0x1f,
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+	0xb0, 0xc4, 0xf0, 0x09,
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+	0x01, 0xc0, 0x0e, 0xfd,
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+	0x72, 0x33, 0xf3, 0xd1,
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+	0xfc, 0xcb, 0xe1, 0x06,
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+	0x56, 0xee, 0x0f, 0x51,
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+	0xd0, 0x15, 0x75, 0xcd,
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+	0x33, 0xbb, 0x00, 0x74,
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+	0x02, 0x10, 0x0b, 0x71,
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+	0x37, 0x30, 0xcd, 0xde,
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+	0xd2, 0xea, 0xfd, 0x0e,
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+	0xc0, 0xeb, 0x0f, 0x50,
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+	0xde, 0xc0, 0x05, 0x31,
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+	0x0d, 0xe0, 0xd2, 0xcf,
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+	0x3b, 0x0e, 0x0e, 0x12,
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+	0x24, 0x00, 0x38, 0x00,
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+	0x00, 0xd4, 0xe0, 0xdd,
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+	0xc0, 0xf2, 0x11, 0x3d,
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+	0x73, 0x32, 0x03, 0xe7,
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+	0x02, 0x93, 0xd3, 0xb2,
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+	0x00, 0x60, 0x3d, 0x3d,
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+	0xcd, 0xee, 0xdd, 0xcc,
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+	0x1f, 0x30, 0x00, 0xf0,
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+	0x1f, 0xf0, 0xc4, 0xe4,
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+	0xfe, 0x32, 0x11, 0xee,
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+	0x11, 0x41, 0xe1, 0xc0,
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+	0xc9, 0xde, 0xe3, 0xf7,
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+	0x30, 0x31, 0xf1, 0xd0,
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+	0x00, 0xde, 0x00, 0x62,
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+	0x1f, 0x00, 0x00, 0x10,
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+	0x2d, 0x0b, 0x1f, 0x00,
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+	0xc0, 0xe4, 0xf7, 0x03,
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+	0x76, 0x23, 0x01, 0x01,
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+	0x0f, 0x70, 0x30, 0x3f,
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+	0x1d, 0x0d, 0xfb, 0x0c,
+	0x0d, 0xfd, 0xfe, 0xdd, 0xac, 0xdf, 0xf0, 0x2e, 0x2a, 0x1d, 0xfe, 0xce,
+	0xaf, 0xcb, 0xed, 0x2d,
+	0x2e, 0xfd, 0xcc, 0xdd, 0xdd, 0xfe, 0x0e, 0xde, 0x8d, 0xed, 0xee, 0x0e,
+	0x00, 0x04, 0xd3, 0xc0,
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+	0x10, 0x20, 0x01, 0xc1,
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+	0xf0, 0x42, 0x74, 0x74,
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+	0x21, 0xee, 0x10, 0x64,
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+	0x1f, 0x40, 0x10, 0xff,
+	0x0f, 0x01, 0x31, 0x75, 0x00, 0x0f, 0x01, 0x02, 0x37, 0x05, 0xe0, 0x22,
+	0x23, 0xe3, 0x81, 0x01,
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+	0xf0, 0x01, 0x23, 0x01,
+	0xe1, 0xe4, 0xf3, 0xd1, 0xd2, 0xe4, 0x80, 0x9f, 0xe0, 0xf0, 0xf0, 0xee,
+	0xda, 0xe0, 0xe0, 0xbd,
+	0xfc, 0x10, 0x30, 0x0c, 0x0c, 0x10, 0x30, 0x00, 0x30, 0x5f, 0x4e, 0x40,
+	0x1c, 0x0a, 0x1e, 0x10,
+	0x30, 0x51, 0x31, 0x23, 0x22, 0x0e, 0x31, 0x56, 0x12, 0x00, 0x22, 0x35,
+	0x21, 0x00, 0x51, 0x20,
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+	0xbd, 0xdb, 0xfe, 0x50,
+	0x13, 0xd1, 0xb1, 0xe1, 0x17, 0x57, 0x11, 0x11, 0x01, 0x11, 0x32, 0x53,
+	0x12, 0x03, 0x35, 0x33,
+	0x11, 0x00, 0x02, 0xd2, 0xb1, 0xe1, 0xe2, 0xd1, 0xd3, 0xc3, 0x9f, 0xeb,
+	0x00, 0x03, 0xe2, 0xd0,
+	0xf0, 0x00, 0x30, 0x0b, 0xec, 0x0f, 0x2e, 0x1d, 0x3e, 0xfa, 0x8c, 0xde,
+	0x0f, 0x20, 0x0f, 0x9a,
+	0xdd, 0xde, 0xee, 0xfe, 0xd9, 0xee, 0x0e, 0xde, 0xdd, 0xfd, 0xfe, 0xdd,
+	0x0c, 0x0d, 0xdb, 0xbd,
+	0xee, 0xfd, 0xfd, 0xda, 0x1e, 0x7f, 0x1e, 0xfc, 0xde, 0xde, 0xce, 0xed,
+	0x0d, 0xee, 0x8c, 0xac,
+	0xfe, 0xff, 0xee, 0xcc, 0xef, 0xdd, 0xef, 0xde, 0xba, 0xfe, 0x0e, 0xff,
+	0xee, 0xc0, 0xe0, 0xec,
+	0xd9, 0x0f, 0xdf, 0xf0, 0xf0, 0xe0, 0x00, 0xe0, 0xcf, 0x02, 0x23, 0x77,
+	0x77, 0xbc, 0xdc, 0x74,
+	0x01, 0xee, 0x00, 0x33, 0xff, 0xca, 0x21, 0x33, 0xf0, 0xcd, 0x0f, 0x21,
+	0x11, 0xef, 0xff, 0x0f,
+	0x00, 0x00, 0x00, 0x0f, 0x01, 0x01, 0xdf, 0x11, 0x22, 0x01, 0xe0, 0x8d,
+	0x03, 0x07, 0xef, 0xfd,
+	0x00, 0xee, 0xe0, 0x04, 0x01, 0x00, 0xde, 0xde, 0xf2, 0x90, 0xbd, 0xf0,
+	0xf0, 0x04, 0xf1, 0xd0,
+	0xde, 0xef, 0xe3, 0xdf, 0xcf, 0xdb, 0xb8, 0xf0, 0xf0, 0xfe, 0x10, 0x21,
+	0xef, 0xed, 0x0f, 0x64,
+	0x11, 0xfc, 0x21, 0x23, 0x00, 0x17, 0x12, 0xce, 0x0f, 0xe0, 0xd3, 0xb4,
+	0x9f, 0xef, 0xff, 0xfd,
+	0x01, 0x12, 0x14, 0x14, 0xfc, 0x0c, 0x73, 0x45, 0x1f, 0x2e, 0x2c, 0x40,
+	0x45, 0x42, 0x20, 0x1e,
+	0x2e, 0x20, 0x11, 0x11, 0x0b, 0x4e, 0x51, 0x01, 0x02, 0x00, 0xf9, 0x1f,
+	0x30, 0xfe, 0xff, 0xe0,
+	0xfe, 0x0c, 0xfb, 0x50, 0x0f, 0xcc, 0xfe, 0x00, 0x60, 0x21, 0x11, 0x0f,
+	0x1e, 0x21, 0x57, 0x15,
+	0x01, 0x00, 0x02, 0x02, 0xf2, 0x03, 0x35, 0x02, 0xe0, 0x72, 0x77, 0x12,
+	0x11, 0x21, 0x21, 0x41,
+	0x23, 0x24, 0x22, 0x01, 0x02, 0x05, 0xe1, 0xf1, 0x03, 0xc2, 0xa0, 0xc0,
+	0xe1, 0x04, 0xe0, 0xe0,
+	0xef, 0x00, 0x43, 0x10, 0xe8, 0x2e, 0x71, 0x11, 0x0f, 0x3f, 0x40, 0x31,
+	0x1e, 0x10, 0x34, 0x00,
+	0xed, 0x2e, 0x6f, 0x2d, 0x1b, 0x1f, 0x10, 0x1d, 0x0d, 0x2d, 0x5d, 0x64,
+	0x27, 0x10, 0x10, 0x30,
+	0x43, 0x13, 0x00, 0x20, 0x35, 0x42, 0x42, 0x3f, 0x19, 0x2f, 0x01, 0x00,
+	0x10, 0x40, 0xff, 0xc0,
+	0xf3, 0xf0, 0xf3, 0xe5, 0xe3, 0xd2, 0xa1, 0x07, 0xe6, 0xc1, 0x00, 0x33,
+	0xd2, 0xd3, 0xd2, 0x64,
+	0x44, 0x43, 0x23, 0x9a, 0x33, 0x25, 0xce, 0x10, 0x01, 0x54, 0x11, 0xbb,
+	0x00, 0x33, 0x02, 0xfe,
+	0xff, 0x22, 0x01, 0xee, 0x00, 0x42, 0x23, 0x10, 0x0f, 0x0e, 0x00, 0x10,
+	0x2f, 0x20, 0x0e, 0x1d,
+	0x3f, 0x21, 0x15, 0xe0, 0x00, 0x77, 0xef, 0xeb, 0x50, 0x11, 0xfe, 0xfe,
+	0x62, 0x23, 0xe8, 0xed,
+	0x40, 0x52, 0x00, 0xfc, 0x0f, 0x11, 0x00, 0xfe, 0x1e, 0x40, 0x00, 0x0d,
+	0xed, 0x00, 0x00, 0xce,
+	0xde, 0x0f, 0xee, 0x98, 0xf0, 0xe1, 0xab, 0xed, 0x02, 0xf1, 0xcd, 0xff,
+	0xef, 0xfe, 0xed, 0x17,
+	0x24, 0xbf, 0xee, 0x01, 0x24, 0x22, 0x11, 0xe1, 0xc2, 0x04, 0x03, 0xf4,
+	0xf3, 0xe1, 0xbf, 0xe1,
+	0xd5, 0xc3, 0xd1, 0xd0, 0x00, 0x01, 0x00, 0xe2, 0xe1, 0xbb, 0x2e, 0x71,
+	0x10, 0x0d, 0x1b, 0x3f,
+	0x1b, 0x1d, 0x6f, 0x1f, 0x2f, 0x0b, 0x0d, 0x30, 0x3d, 0x3f, 0x31, 0x02,
+	0x01, 0x42, 0x33, 0x00,
+	0xdc, 0x1f, 0x00, 0x88, 0x0f, 0x61, 0x10, 0x0b, 0x0d, 0xff, 0x01, 0x13,
+	0x0f, 0xfc, 0x0d, 0x1f,
+	0xbc, 0xbe, 0xf1, 0xf0, 0x0d, 0x0b, 0xfe, 0xff, 0xfe, 0x0e, 0x30, 0x50,
+	0x23, 0x12, 0x01, 0x17,
+	0x77, 0x32, 0x20, 0x00, 0x42, 0x25, 0xf1, 0xc1, 0x01, 0x01, 0xe0, 0xc0,
+	0xc2, 0xa0, 0xe0, 0xe1,
+	0xf1, 0xf0, 0xef, 0xdb, 0xde, 0xe3, 0xe5, 0x8a, 0xf0, 0x35, 0xe0, 0xdf,
+	0x20, 0x41, 0x00, 0xce,
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+	0xef, 0xbb, 0xcc, 0xdc,
+	0xff, 0x0f, 0x0e, 0xdb, 0xfe, 0xff, 0xbc, 0xed, 0x0d, 0x1d, 0xdc, 0x2f,
+	0x20, 0xab, 0xfe, 0x10,
+	0x30, 0xff, 0xde, 0x00, 0x02, 0xa0, 0xf1, 0xf2, 0xe0, 0xdf, 0x14, 0xf3,
+	0x8e, 0xef, 0x03, 0x01,
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+	0x12, 0xf2, 0x8c, 0xdc,
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+	0x10, 0x10, 0x21, 0x21,
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+	0x07, 0x11, 0x0f, 0x7f,
+	0x33, 0x00, 0xf0, 0x35, 0x56, 0x42, 0x11, 0x10, 0x32, 0x15, 0x23, 0x43,
+	0x21, 0x00, 0xe0, 0xe0,
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+	0xf1, 0x15, 0x02, 0x02,
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+	0x02, 0x02, 0xf2, 0xf3,
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+	0xcb, 0x10, 0x20, 0x20,
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+	0x0e, 0x0f, 0x1e, 0x0f,
+	0x24, 0x77, 0x53, 0x20, 0x2e, 0x3d, 0x4e, 0x3e, 0x3c, 0x2c, 0x4c, 0x3c,
+	0x3e, 0x10, 0x10, 0x50,
+	0x5e, 0x2e, 0x22, 0x14, 0x01, 0xf0, 0xf0, 0xdd, 0xeb, 0x0f, 0x0f, 0xfb,
+	0x1e, 0x60, 0x35, 0x03,
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+	0x13, 0x83, 0xd2, 0x03,
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+	0x76, 0x01, 0xfe, 0x10,
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+	0x00, 0x10, 0x00, 0x10,
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+	0xdc, 0xdd, 0x50, 0x41,
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+	0xee, 0x01, 0xc0, 0xab,
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+	0x10, 0x03, 0xbf, 0x00,
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+	0x31, 0x0f, 0x00, 0x25,
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+	0x53, 0x13, 0x10, 0x02,
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+	0x2e, 0x4d, 0x2d, 0x3d,
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+	0x04, 0xd1, 0xd0, 0x02,
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+	0x23, 0x04, 0x03, 0x14,
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+	0xfd, 0x0e, 0xff, 0x01,
+	0x15, 0x0e, 0x0a, 0xfd, 0xfb, 0x2f, 0x1d, 0x3c, 0x0e, 0xdc, 0x0e, 0x40,
+	0xae, 0x88, 0x20, 0x30,
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+	0x00, 0x0e, 0x32, 0x11,
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+	0x20, 0xfe, 0xf0, 0xde,
+	0x34, 0x44, 0xf0, 0x77, 0x63, 0xdc, 0x01, 0x47, 0x32, 0x00, 0x01, 0x02,
+	0x12, 0x02, 0x12, 0x02,
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+	0xe1, 0xe1, 0xe2, 0xe1,
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+	0xee, 0xca, 0xad, 0xcc,
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+	0xdd, 0xfc, 0x4e, 0x5e,
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+	0xdc, 0x89, 0xed, 0xfe,
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+	0x00, 0x00, 0xcd, 0x00,
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+	0x15, 0x77, 0x03, 0xca,
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+	0x00, 0x30, 0x0f, 0x0e,
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+	0x53, 0xf0, 0xd0, 0x02,
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+	0xcf, 0xdd, 0xd0, 0xc2,
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+	0x54, 0x33, 0x21, 0x33,
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+	0x00, 0x01, 0xf1, 0xab,
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+	0x0f, 0x00, 0x21, 0x11,
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+	0xfd, 0xfc, 0x2e, 0x50,
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+	0xff, 0x0e, 0x47, 0x12,
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+	0x44, 0x02, 0x00, 0x02,
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+	0x00, 0x33, 0xf0, 0xe0,
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+	0x0e, 0xeb, 0xee, 0x20,
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+	0x00, 0x63, 0x24, 0x10,
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+	0x0d, 0x0c, 0xef, 0xd0,
+	0xa0, 0xe5, 0xf0, 0xfe, 0xeb, 0xbd, 0x02, 0x17, 0x03, 0xcf, 0x01, 0x47,
+	0x01, 0x01, 0x02, 0x03,
+	0x03, 0xf2, 0xe2, 0x01, 0x00, 0x43, 0x77, 0x12, 0x32, 0x01, 0xd0, 0xf0,
+	0x13, 0x34, 0x00, 0x0f,
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+	0x34, 0x26, 0x22, 0x10,
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+	0x40, 0x0e, 0x0d, 0x63,
+	0x77, 0x04, 0xf8, 0x2f, 0x50, 0x01, 0x00, 0x0e, 0x00, 0xef, 0xee, 0x01,
+	0x20, 0x1d, 0x09, 0x0e,
+	0x21, 0xf1, 0x80, 0xef, 0xdf, 0xef, 0x00, 0xff, 0xfd, 0xfc, 0xcd, 0xdd,
+	0xee, 0x02, 0xe0, 0xba,
+	0x10, 0x0f, 0xde, 0x12, 0xf2, 0x05, 0x33, 0xa0, 0xd0, 0xd1, 0x8f, 0xf3,
+	0x07, 0xf5, 0xf1, 0xe1,
+	0x01, 0x03, 0xf1, 0xc0, 0xf3, 0x02, 0x10, 0x42, 0x00, 0xed, 0x0f, 0x10,
+	0x2c, 0x72, 0x2f, 0xf8,
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+	0x10, 0x0c, 0x0a, 0x30,
+	0x10, 0x01, 0x65, 0x00, 0xef, 0x20, 0x2f, 0x0d, 0x0f, 0xbf, 0x02, 0xe0,
+	0x0d, 0x3c, 0x1b, 0x50,
+	0xfe, 0x0e, 0x77, 0x00, 0x0c, 0x0e, 0x20, 0x01, 0x02, 0xf0, 0x32, 0x11,
+	0xef, 0xef, 0xd2, 0x91,
+	0xf1, 0x11, 0x74, 0x11, 0x0d, 0x05, 0x26, 0x12, 0x32, 0x0f, 0xf0, 0x04,
+	0xd2, 0xc2, 0xe3, 0xe1,
+	0xf0, 0x80, 0x92, 0xe4, 0xd5, 0x01, 0xe0, 0xde, 0xc0, 0xd1, 0xd3, 0xf2,
+	0xfd, 0x08, 0x2e, 0x51,
+	0x25, 0x11, 0x20, 0x40, 0x2c, 0x1c, 0x4f, 0x3f, 0x2e, 0x1d, 0x05, 0x05,
+	0x20, 0x00, 0x1d, 0x00,
+	0xa3, 0x0c, 0x2a, 0x12, 0x69, 0x0d, 0x11, 0x14, 0x8d, 0x02, 0xeb, 0x08,
+	0x71, 0x02, 0xbb, 0xff,
+	0x34, 0x3f, 0x2b, 0x11, 0x03, 0xd0, 0xce, 0x10, 0x21, 0x35, 0xab, 0xda,
+	0x74, 0x10, 0xcc, 0x02,
+	0x11, 0x00, 0xfd, 0xfe, 0x43, 0x02, 0xdd, 0xed, 0x02, 0x22, 0xfe, 0xdb,
+	0x34, 0x11, 0xbf, 0x0f,
+	0xec, 0x00, 0x73, 0xff, 0xdc, 0x12, 0xfe, 0xec, 0x23, 0x40, 0xda, 0xaf,
+	0x00, 0x12, 0xfb, 0xf0,
+	0x00, 0xd0, 0xe0, 0xf0, 0x10, 0x1f, 0xdb, 0xe0, 0xc0, 0xd0, 0x00, 0xdd,
+	0x1c, 0x60, 0xf2, 0xbc,
+	0xf0, 0x00, 0xbc, 0x00, 0x47, 0xff, 0xac, 0x01, 0x01, 0xe0, 0xec, 0xf0,
+	0x00, 0x01, 0x00, 0x00,
+	0x0e, 0x0d, 0x00, 0x10, 0x2f, 0x30, 0x51, 0xfc, 0x0d, 0x03, 0xff, 0xdb,
+	0x02, 0xf1, 0xab, 0xdf,
+	0x00, 0xdf, 0x00, 0x01, 0xfd, 0xe9, 0x20, 0x02, 0x8e, 0xdd, 0x60, 0x00,
+	0x0e, 0x00, 0xf1, 0xfb,
+	0xee, 0x44, 0x33, 0x00, 0xbc, 0x02, 0x03, 0xec, 0xfe, 0x02, 0xe1, 0xf1,
+	0xf0, 0xed, 0x47, 0xe3,
+	0x8d, 0xbf, 0x01, 0x04, 0x11, 0x00, 0x00, 0x0d, 0x00, 0x03, 0x05, 0x02,
+	0xdc, 0xdc, 0x13, 0x55,
+	0xfe, 0xce, 0xf0, 0xf3, 0x01, 0x1c, 0xfe, 0xce, 0xf6, 0xb3, 0xc1, 0x00,
+	0xff, 0x0b, 0x00, 0x02,
+	0xd0, 0xc0, 0xe0, 0x0f, 0xbf, 0xd8, 0x20, 0x30, 0x20, 0x10, 0x00, 0xe0,
+	0xff, 0x0e, 0x7f, 0x2c,
+	0xff, 0x00, 0x10, 0x20, 0xfd, 0xf3, 0xee, 0x5c, 0x5a, 0x0c, 0x00, 0x03,
+	0x20, 0xf9, 0xf0, 0xf0,
+	0x44, 0x00, 0x30, 0x73, 0x4d, 0xcd, 0x17, 0x01, 0xee, 0x01, 0x76, 0x0f,
+	0xdd, 0x21, 0x54, 0x00,
+	0xf0, 0xdd, 0xf0, 0x53, 0x00, 0x0f, 0x01, 0x35, 0x01, 0xbd, 0x10, 0x37,
+	0xef, 0xcf, 0x10, 0x22,
+	0x20, 0x11, 0xf0, 0xdf, 0x00, 0x01, 0xe2, 0xe1, 0xbd, 0x0d, 0x66, 0x00,
+	0xdf, 0x00, 0x03, 0x35,
+	0x1f, 0x0c, 0x73, 0x01, 0xde, 0x04, 0x1e, 0xec, 0x11, 0x10, 0x0e, 0x52,
+	0xf1, 0xef, 0x20, 0x3d,
+	0x3e, 0x02, 0x8a, 0xff, 0x15, 0x01, 0xfe, 0xff, 0xcf, 0xf4, 0x05, 0xbc,
+	0xee, 0x00, 0x10, 0xf0,
+	0xff, 0x0d, 0x70, 0x13, 0x0b, 0x00, 0x01, 0x00, 0x1e, 0x50, 0x0e, 0xec,
+	0xef, 0xef, 0x00, 0x0e,
+	0x40, 0x1f, 0xf8, 0xee, 0x22, 0xf0, 0xab, 0x10, 0x01, 0x0d, 0x0e, 0x0d,
+	0x75, 0x11, 0xee, 0x15,
+	0x0d, 0x99, 0x21, 0x33, 0x0f, 0xfe, 0x42, 0x10, 0xdd, 0x10, 0x51, 0x0e,
+	0xff, 0x00, 0x00, 0xe1,
+	0xe4, 0x0e, 0x09, 0x40, 0x32, 0xc1, 0xc0, 0x00, 0x00, 0xe0, 0x00, 0x31,
+	0x0a, 0x40, 0x12, 0xc0,
+	0xf0, 0xee, 0x20, 0x51, 0x10, 0x32, 0xe1, 0xad, 0x13, 0x13, 0xf6, 0x04,
+	0x01, 0xee, 0xed, 0xc1,
+	0x17, 0x13, 0xcf, 0xee, 0x02, 0x03, 0x10, 0x30, 0x73, 0x21, 0x33, 0x00,
+	0x01, 0x47, 0x05, 0x00,
+	0x63, 0x21, 0x0e, 0x00, 0xe0, 0x12, 0x22, 0xff, 0xbd, 0x10, 0x44, 0x30,
+	0x06, 0x04, 0x10, 0x00,
+	0xff, 0x00, 0x63, 0x53, 0x22, 0x21, 0x27, 0x32, 0x01, 0x0f, 0xde, 0x23,
+	0x32, 0x41, 0x51, 0x30,
+	0x33, 0xf0, 0xd0, 0x24, 0x31, 0x33, 0x00, 0xad, 0x04, 0x17, 0x21, 0x1f,
+	0x34, 0x34, 0x42, 0x10,
+	0x43, 0x41, 0x2e, 0x31, 0x02, 0xbd, 0x0f, 0x01, 0x05, 0xdf, 0xe4, 0x07,
+	0x57, 0x00, 0xbf, 0x0f,
+	0x01, 0x13, 0x34, 0xbf, 0xef, 0x50, 0x33, 0xe1, 0xd2, 0x31, 0x75, 0x11,
+	0x00, 0x1f, 0x30, 0x44,
+	0x02, 0x00, 0x17, 0x11, 0x00, 0xd0, 0x22, 0x21, 0xe1, 0xa1, 0x05, 0xf3,
+	0xf0, 0xef, 0x41, 0x73,
+	0x53, 0x02, 0xe2, 0x01, 0x01, 0x23, 0x1c, 0x09, 0x0f, 0x13, 0x47, 0x03,
+	0xec, 0x0f, 0x20, 0x00,
+	0x00, 0x01, 0x01, 0x00, 0xee, 0x0f, 0x71, 0x30, 0x5f, 0x0f, 0x04, 0xf4,
+	0xff, 0x0d, 0x2f, 0x33,
+	0x61, 0x2c, 0x20, 0x01, 0x13, 0x1b, 0x3d, 0x0d, 0x1c, 0x00, 0x53, 0xf2,
+	0xbc, 0xeb, 0xdd, 0xed,
+	0xf0, 0xf7, 0xa0, 0xfd, 0x01, 0x01, 0x00, 0x0f, 0xd0, 0xd1, 0xb0, 0xfb,
+	0xec, 0x0d, 0x0f, 0x23,
+	0x90, 0xdc, 0xd0, 0xe0, 0x0b, 0x00, 0x1f, 0xec, 0xd1, 0xc3, 0xd0, 0xe0,
+	0x81, 0x00, 0xe0, 0x00,
+	0x05, 0xe0, 0xe8, 0xee, 0xce, 0xf0, 0xf0, 0xbd, 0x0d, 0x12, 0x64, 0x00,
+	0xe9, 0xcf, 0x0f, 0x10,
+	0x1e, 0x30, 0x30, 0xec, 0xff, 0x11, 0x0e, 0x0d, 0x20, 0xdd, 0xd0, 0xef,
+	0xdc, 0x2d, 0x78, 0x50,
+	0x03, 0xe2, 0xfe, 0xfd, 0x1f, 0x52, 0x03, 0x00, 0x0e, 0x3e, 0x70, 0x2f,
+	0xe8, 0x0f, 0x1f, 0x50,
+	0x21, 0x0d, 0xdc, 0x00, 0x0e, 0x0b, 0xcf, 0xf0, 0xd3, 0x02, 0x53, 0x4f,
+	0xf9, 0xbf, 0x00, 0xf0,
+	0xfe, 0xff, 0x0e, 0xf0, 0x00, 0xd3, 0xe5, 0xac, 0xef, 0xe1, 0xee, 0xea,
+	0xf2, 0xe2, 0x80, 0xd0,
+	0x02, 0xe0, 0xef, 0xdd, 0xf7, 0xe1, 0xe2, 0xde, 0x8a, 0xee, 0xff, 0x00,
+	0x50, 0x00, 0x0f, 0x02,
+	0xe0, 0xf0, 0x0e, 0x08, 0xed, 0xb0, 0xf3, 0xe1, 0xdf, 0xca, 0xf0, 0xee,
+	0x0f, 0x1f, 0xff, 0x90,
+	0xf0, 0xeb, 0x0e, 0xee, 0x0c, 0x1f, 0x20, 0x0f, 0xbe, 0xfb, 0xf0, 0x00,
+	0xee, 0xee, 0x10, 0x10,
+	0xfe, 0x88, 0xff, 0x0f, 0x00, 0xf0, 0xef, 0xfe, 0x10, 0x0f, 0xcd, 0xff,
+	0x01, 0x16, 0xe0, 0xf0,
+	0x50, 0x62, 0x1e, 0x1d, 0x33, 0x02, 0xea, 0x10, 0x5f, 0x41, 0x21, 0xfb,
+	0xff, 0x10, 0x12, 0xf5,
+	0xe1, 0xce, 0xef, 0xe1, 0xc2, 0x02, 0x52, 0x2f, 0xde, 0xa0, 0xea, 0xef,
+	0x01, 0x14, 0x10, 0x1c,
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+	0xe6, 0x07, 0xe0, 0x9f,
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+	0x02, 0x03, 0x04, 0x02,
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+	0xdd, 0x0e, 0xf2, 0xf7,
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+	0xdf, 0xda, 0xf0, 0x10,
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+	0x01, 0x53, 0x20, 0x1c,
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+	0x34, 0x54, 0x01, 0xfe,
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+	0x01, 0x01, 0xbd, 0xcb,
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+	0x21, 0xfe, 0xdd, 0x9a,
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+	0x11, 0x45, 0x21, 0x00,
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+	0x00, 0x11, 0x44, 0x22,
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+	0x11, 0xff, 0xee, 0xff,
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+	0xde, 0xff, 0x11, 0x55,
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+	0x00, 0x11, 0x22, 0x43,
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+	0xee, 0x00, 0x12, 0x0f,
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+	0x55, 0x00, 0xff, 0x0f,
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+	0x22, 0x00, 0x00, 0x76,
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+	0x54, 0x00, 0x0d, 0x24,
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+	0x32, 0x66, 0xee, 0xbc,
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+	0x02, 0x14, 0x01, 0x16,
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+	0xce, 0x00, 0x00, 0x12,
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+	0xad, 0xff, 0x02, 0x24,
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+	0x34, 0x43, 0xfe, 0xed,
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+	0x00, 0xfe, 0xff, 0x10,
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+	0x33, 0x24, 0x89, 0xdd,
+	0x01, 0x55, 0x01, 0x0f, 0xff, 0x00, 0x01, 0x22, 0x00, 0xff, 0xed, 0xba,
+	0xff, 0x00, 0xf0, 0xef,
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+	0xe0, 0x77, 0x31, 0xe0,
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+	0x00, 0xf0, 0x00, 0xe0,
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+	0xf0, 0xdd, 0x10, 0x43,
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+	0x67, 0x00, 0x00, 0x10,
+	0x21, 0x11, 0x23, 0xfd, 0x98, 0xff, 0x10, 0x21, 0x00, 0x0f, 0x0f, 0xf0,
+	0x63, 0x34, 0xfe, 0x0f,
+	0x44, 0x0f, 0xdb, 0x10, 0x75, 0x00, 0x00, 0x33, 0x10, 0xfe, 0xed, 0x0f,
+	0x32, 0x32, 0xec, 0xbc,
+	0xed, 0xfe, 0x44, 0x33, 0xfc, 0x00, 0x53, 0xfe, 0xa8, 0x0f, 0x11, 0x11,
+	0x00, 0x00, 0x00, 0x0f,
+	0x21, 0x10, 0x01, 0x33, 0x31, 0x00, 0xaa, 0xde, 0x10, 0x33, 0x11, 0xff,
+	0x00, 0x00, 0xef, 0xef,
+	0x0f, 0xee, 0x01, 0x31, 0x88, 0xde, 0x01, 0x01, 0xe0, 0xcc, 0x12, 0x43,
+	0xdb, 0xde, 0x23, 0x02,
+	0xf0, 0x34, 0xbe, 0x9b, 0x00, 0xf1, 0xdf, 0xef, 0x02, 0x25, 0x11, 0xee,
+	0x01, 0x77, 0x00, 0xee,
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+	0x10, 0x0e, 0x12, 0x35,
+	0x00, 0x01, 0xce, 0x98, 0xef, 0xdf, 0x10, 0x77, 0x31, 0x00, 0x0f, 0xff,
+	0x10, 0x10, 0xee, 0xcc,
+	0x00, 0x00, 0xcc, 0xf0, 0x32, 0x00, 0x0f, 0x00, 0x32, 0x66, 0x22, 0xff,
+	0xdd, 0xee, 0x10, 0x65,
+	0xff, 0xdb, 0x00, 0x22, 0xf0, 0xad, 0x00, 0x11, 0x23, 0x22, 0x11, 0xff,
+	0x22, 0x33, 0xba, 0xef,
+	0x66, 0x11, 0xcc, 0x00, 0x33, 0x11, 0xdd, 0x0f, 0x12, 0xef, 0xcc, 0x00,
+	0x23, 0x22, 0xe0, 0xee,
+	0xee, 0x0f, 0xde, 0x88, 0xdf, 0x33, 0x56, 0x11, 0x00, 0xdf, 0x00, 0x44,
+	0x00, 0x01, 0x21, 0x00,
+	0xef, 0x00, 0x44, 0x0f, 0xbc, 0xed, 0x10, 0x32, 0x22, 0x55, 0x21, 0x10,
+	0x31, 0x55, 0x20, 0xff,
+	0x0e, 0x0f, 0x00, 0xee, 0xb8, 0xdf, 0xff, 0x00, 0x30, 0x10, 0x00, 0x00,
+	0xce, 0x8a, 0xff, 0xf0,
+	0x00, 0xef, 0xff, 0x00, 0x01, 0x24, 0x22, 0x00, 0x00, 0x53, 0x12, 0x13,
+	0x0f, 0xa8, 0xff, 0x21,
+	0x00, 0x0f, 0x00, 0xff, 0x0f, 0x21, 0x11, 0x33, 0x33, 0xfd, 0x00, 0x65,
+	0x0f, 0xb8, 0xff, 0x00,
+	0x12, 0x42, 0x10, 0x00, 0xfe, 0x00, 0x34, 0x00, 0x0e, 0x50, 0x0f, 0x0e,
+	0x66, 0x12, 0xff, 0x00,
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+	0x00, 0x01, 0x00, 0xde,
+	0xcc, 0x00, 0x42, 0xff, 0xdd, 0x00, 0x44, 0x00, 0xfe, 0x11, 0x0f, 0xde,
+	0x88, 0xce, 0x00, 0x12,
+	0x02, 0x13, 0x01, 0xed, 0xa8, 0xde, 0x22, 0x34, 0x00, 0x22, 0x33, 0x00,
+	0xbb, 0xf0, 0x21, 0x10,
+	0x00, 0x23, 0x32, 0xbe, 0x0f, 0x22, 0xff, 0xee, 0x01, 0x03, 0xf0, 0xe0,
+	0xee, 0xf3, 0xe4, 0x8c,
+	0xfd, 0x14, 0x47, 0x00, 0x00, 0x0e, 0x22, 0x67, 0x00, 0xef, 0x32, 0x43,
+	0xff, 0xff, 0x21, 0x34,
+	0x00, 0xf0, 0x22, 0x76, 0x31, 0xef, 0xee, 0xf0, 0x11, 0xfe, 0xdd, 0x00,
+	0x20, 0x75, 0x0e, 0xed,
+	0x11, 0x65, 0x11, 0x0f, 0x1f, 0x10, 0x0e, 0xea, 0xff, 0x00, 0xed, 0xbd,
+	0x01, 0x02, 0x00, 0x00,
+	0xdc, 0xbb, 0xff, 0x00, 0x00, 0xcc, 0xdd, 0x44, 0x22, 0xbb, 0xcd, 0x0f,
+	0x01, 0x12, 0x00, 0x0f,
+	0x00, 0x00, 0x00, 0x36, 0x64, 0xfe, 0xab, 0x00, 0x01, 0x12, 0x12, 0x0e,
+	0xcb, 0xff, 0x00, 0x21,
+	0x32, 0x34, 0x00, 0xee, 0x01, 0x00, 0x43, 0x22, 0xbc, 0xed, 0x13, 0x13,
+	0xbb, 0xbb, 0x00, 0x23,
+	0x00, 0xee, 0xee, 0x00, 0x15, 0x11, 0x00, 0x33, 0x33, 0x01, 0x00, 0x11,
+	0x63, 0x00, 0xed, 0x11,
+	0x56, 0x11, 0x00, 0xfc, 0xf0, 0x34, 0x33, 0xee, 0xba, 0xff, 0x42, 0x66,
+	0x00, 0xdd, 0x00, 0x0f,
+	0xbc, 0x13, 0x54, 0x00, 0x00, 0x00, 0xef, 0xff, 0x10, 0x12, 0x00, 0xf0,
+	0x0e, 0x1f, 0x44, 0x00,
+	0xa8, 0xee, 0xee, 0xef, 0x00, 0xfe, 0xef, 0x43, 0x34, 0xee, 0xfe, 0x10,
+	0xf0, 0x00, 0x77, 0x31,
+	0xcd, 0xba, 0xee, 0x22, 0x22, 0x00, 0xee, 0xf0, 0xde, 0xff, 0x11, 0x33,
+	0x44, 0x53, 0x21, 0xfe,
+	0xec, 0x32, 0x67, 0x0f, 0xdd, 0x00, 0x42, 0x34, 0x13, 0x22, 0x11, 0x20,
+	0x00, 0xee, 0xee, 0x00,
+	0xdc, 0xfe, 0x72, 0x0f, 0xd8, 0x0f, 0x20, 0xde, 0xce, 0x22, 0x33, 0x10,
+	0x42, 0x00, 0xba, 0x00,
+	0x11, 0xde, 0xf0, 0xf0, 0x03, 0x35, 0x00, 0xde, 0x8d, 0xec, 0xde, 0xed,
+	0x00, 0x45, 0x00, 0xf0,
+	0xdd, 0xed, 0xef, 0x02, 0x77, 0x01, 0x9c, 0xee, 0x01, 0x25, 0x00, 0x00,
+	0x01, 0x35, 0x22, 0xd0,
+	0xce, 0x00, 0x11, 0x22, 0x00, 0xf0, 0xdf, 0xee, 0x20, 0x32, 0xcc, 0x8b,
+	0xff, 0x01, 0xf0, 0x00,
+	0x02, 0x00, 0xe0, 0x47, 0x35, 0x41, 0x32, 0xde, 0x8c, 0x0f, 0x00, 0x11,
+	0x45, 0x33, 0xfe, 0xdd,
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+	0x20, 0xff, 0xfd, 0x10,
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+	0x01, 0x75, 0x21, 0x10,
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+	0x46, 0x31, 0x32, 0x20,
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+	0x12, 0x01, 0x3d, 0x01,
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+	0x00, 0x03, 0x14, 0xdd,
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+	0x35, 0xdf, 0x9a, 0x00,
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+	0x43, 0x21, 0x0d, 0x0f,
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+	0x00, 0xf0, 0x12, 0x41,
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+	0x56, 0x54, 0x00, 0xcd,
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+	0x77, 0x21, 0x0f, 0x0e,
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+	0xee, 0xdf, 0x00, 0x00,
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+	0x0d, 0x10, 0x10, 0x12,
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+	0xaf, 0x12, 0x77, 0x27,
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+	0x0e, 0x54, 0x00, 0x88,
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+	0x00, 0x0f, 0x00, 0x23,
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+	0xa8, 0x00, 0x00, 0xdf,
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+	0x0e, 0xff, 0xe0, 0x00,
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+	0x00, 0x77, 0x00, 0xef,
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+	0x47, 0x42, 0x12, 0x11,
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+	0x42, 0x01, 0x00, 0x41,
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+	0x12, 0xff, 0x01, 0x01,
+	0xcb, 0x00, 0x13, 0xde, 0xaa, 0xef, 0xee, 0x0f, 0x00, 0xde, 0xca, 0x10,
+	0x41, 0x00, 0x00, 0x62,
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+	0x03, 0xd0, 0xd0, 0x00,
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+	0x51, 0x0d, 0xfc, 0x21,
+	0x21, 0xec, 0x41, 0x43, 0xef, 0xf0, 0x01, 0x34, 0x00, 0x99, 0xff, 0x53,
+	0x20, 0xeb, 0xf0, 0x33,
+	0x12, 0xed, 0x00, 0x63, 0x00, 0x00, 0xef, 0x01, 0xed, 0xb8, 0x00, 0x22,
+	0xff, 0xf0, 0xf0, 0xf1,
+	0x23, 0x00, 0x89, 0xee, 0x01, 0xf0, 0xf0, 0x22, 0x10, 0xed, 0x00, 0x00,
+	0x0e, 0x41, 0x43, 0x13,
+	0xff, 0xa8, 0xea, 0x00, 0x54, 0x01, 0xef, 0x00, 0xff, 0xee, 0xff, 0x00,
+	0x20, 0x76, 0x00, 0xcb,
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+	0x0e, 0x00, 0x00, 0x00,
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+	0xff, 0x12, 0x11, 0x00,
+	0x20, 0x10, 0x00, 0x11, 0x52, 0x11, 0x00, 0x20, 0x10, 0x1f, 0x77, 0x77,
+	0x00, 0x00, 0x11, 0x0f,
+	0xef, 0x22, 0x11, 0xfe, 0x00, 0xf0, 0xdc, 0xfc, 0x10, 0x11, 0x01, 0xf1,
+	0xde, 0xbd, 0xef, 0xde,
+	0x00, 0x0c, 0xd8, 0x61, 0x44, 0xe0, 0xef, 0xfe, 0xdc, 0x10, 0x52, 0xef,
+	0xce, 0xf1, 0x27, 0x00,
+	0xf0, 0x31, 0xe0, 0xdf, 0x00, 0xef, 0xde, 0x0e, 0x0f, 0xdf, 0x8d, 0xab,
+	0xf0, 0x05, 0x21, 0x00,
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+	0x01, 0xe2, 0x17, 0x64,
+	0xf0, 0xff, 0x30, 0x20, 0xfd, 0x42, 0x11, 0x98, 0xfe, 0x00, 0x21, 0x11,
+	0x11, 0xef, 0x00, 0x41,
+	0x10, 0x01, 0xf2, 0xef, 0xcb, 0x21, 0x45, 0xff, 0xbe, 0xfe, 0xec, 0xee,
+	0xfe, 0xcc, 0xfe, 0x21,
+	0x65, 0x00, 0xcd, 0xfe, 0x0e, 0x0f, 0x20, 0x11, 0x53, 0x10, 0xea, 0x00,
+	0x03, 0x22, 0x24, 0x01,
+	0xdf, 0x22, 0x55, 0x11, 0x00, 0x11, 0x54, 0x12, 0xee, 0x0d, 0x74, 0x00,
+	0xd8, 0x0f, 0x01, 0xe0,
+	0xcf, 0xff, 0x00, 0x22, 0xf0, 0xce, 0xff, 0xef, 0xde, 0xdb, 0x0f, 0x74,
+	0x66, 0x00, 0xed, 0x00,
+	0x14, 0x01, 0x01, 0x12, 0x32, 0x00, 0x00, 0x0f, 0xdd, 0xff, 0x44, 0x10,
+	0xbb, 0xdd, 0x0f, 0x55,
+	0x21, 0x00, 0x10, 0x31, 0x0e, 0x1e, 0x64, 0x00, 0xfe, 0xfe, 0x00, 0x00,
+	0xfd, 0xde, 0x02, 0x67,
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+	0xfd, 0xbf, 0x07, 0x00,
+	0x10, 0x31, 0xe0, 0xe0, 0x42, 0xf1, 0xbd, 0x10, 0x56, 0x01, 0xef, 0x00,
+	0x23, 0x21, 0x02, 0x00,
+	0x00, 0x77, 0x45, 0xdf, 0x01, 0x32, 0xff, 0xde, 0x13, 0x13, 0x01, 0x02,
+	0xe0, 0xcb, 0x00, 0x23,
+	0xdf, 0x10, 0x50, 0xff, 0x16, 0x45, 0xc0, 0xde, 0x00, 0x34, 0x01, 0xae,
+	0x0f, 0x42, 0x21, 0x00,
+	0x1f, 0x20, 0x1d, 0x0b, 0xdd, 0xcb, 0xce, 0x20, 0x63, 0xfd, 0xb9, 0x0f,
+	0x00, 0x00, 0x10, 0x00,
+	0x10, 0x22, 0xed, 0xff, 0x10, 0x0d, 0xca, 0xce, 0xef, 0xf0, 0x00, 0x0e,
+	0xba, 0x00, 0x62, 0x21,
+	0x0c, 0x10, 0x41, 0x01, 0x0e, 0x0d, 0xee, 0xf0, 0x00, 0xee, 0xee, 0x00,
+	0x01, 0xf0, 0x00, 0x88,
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+	0x33, 0x32, 0x27, 0xf3,
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+	0x11, 0x11, 0x00, 0x0f,
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+	0xbc, 0xf0, 0xf0, 0x9b,
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+	0x32, 0x23, 0x10, 0xef,
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+	0xf0, 0x14, 0x11, 0xff,
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+	0x56, 0x0f, 0xed, 0xee,
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+	0x20, 0x30, 0x00, 0x00,
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+	0x0e, 0xed, 0xab, 0xef,
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+	0x23, 0x02, 0xdf, 0x10,
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+	0xf0, 0x45, 0x12, 0xe0,
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+	0x01, 0x12, 0xe0, 0xd0,
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+	0x00, 0x26, 0x11, 0xe0,
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+	0xef, 0x01, 0x00, 0x11,
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+	0x00, 0x0e, 0x10, 0x00,
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+	0x0f, 0xdc, 0xde, 0xbf,
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+	0x10, 0xfe, 0xdc, 0xf0,
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+	0xfd, 0x2e, 0x4e, 0xf8,
+	0xbe, 0xde, 0xbd, 0xca, 0xef, 0xff, 0xfe, 0xef, 0xce, 0xee, 0xf0, 0xfe,
+	0xba, 0xf0, 0x01, 0x00,
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+	0x03, 0x15, 0x02, 0x01,
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+	0x33, 0x32, 0xed, 0x11,
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+	0x24, 0x12, 0x47, 0x22,
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+	0xcb, 0xbe, 0xef, 0xcc,
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+	0xee, 0x00, 0x54, 0x0f,
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+	0x1e, 0x00, 0xcf, 0xe1,
+	0xe1, 0xdf, 0x03, 0x05, 0x8f, 0xde, 0x0f, 0xee, 0xdd, 0xdd, 0x02, 0xf7,
+	0xb0, 0xe0, 0xd1, 0xd0,
+	0xdd, 0x0b, 0x1f, 0x31, 0x23, 0xf0, 0xbf, 0x00, 0x43, 0x56, 0x00, 0xe0,
+	0xd0, 0x01, 0x21, 0x02,
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+	0x20, 0x30, 0x0f, 0xf0,
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+	0x12, 0xed, 0xee, 0x26,
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+	0xbd, 0x00, 0x02, 0xe0,
+	0xae, 0xff, 0xfc, 0x09, 0x0f, 0x60, 0x00, 0xec, 0xfe, 0x10, 0x10, 0x12,
+	0x22, 0x10, 0x0f, 0x21,
+	0x55, 0x22, 0x02, 0xbf, 0xfd, 0x10, 0x22, 0x00, 0x00, 0x17, 0x05, 0x05,
+	0xcd, 0x00, 0x41, 0x01,
+	0x9a, 0xe3, 0xfc, 0x02, 0x9e, 0xe5, 0xb8, 0x04, 0xff, 0x74, 0x15, 0xee,
+	0xd9, 0x00, 0x31, 0x13,
+	0x40, 0x0d, 0xf0, 0xc0, 0xd0, 0x31, 0x62, 0x0f, 0xff, 0x0f, 0x0d, 0xf3,
+	0xe0, 0xb8, 0x20, 0x31,
+	0x03, 0x10, 0x0e, 0xc8, 0xef, 0xa0, 0xee, 0x22, 0x63, 0x0f, 0xcb, 0x00,
+	0x00, 0xff, 0x02, 0x43,
+	0x0e, 0xcd, 0xfd, 0x01, 0x35, 0x0d, 0xcd, 0x11, 0x33, 0x01, 0x01, 0x0d,
+	0xdd, 0x04, 0x46, 0xe0,
+	0xce, 0x00, 0x20, 0x00, 0x00, 0x0d, 0x1f, 0x10, 0xe3, 0xf2, 0xf4, 0x04,
+	0xfe, 0xaa, 0x20, 0x00,
+	0xe5, 0x01, 0x8a, 0xcf, 0x31, 0x54, 0x00, 0xed, 0xe0, 0x20, 0x77, 0x00,
+	0xfd, 0xf0, 0x02, 0x20,
+	0xee, 0xe0, 0x14, 0x40, 0xff, 0x0d, 0x43, 0x12, 0xee, 0xfd, 0x14, 0x10,
+	0x0f, 0x22, 0x72, 0xfb,
+	0xbd, 0xf0, 0x00, 0x20, 0x0d, 0x04, 0x34, 0xee, 0xbb, 0x11, 0x54, 0x0d,
+	0xdd, 0x00, 0x24, 0x10,
+	0x1e, 0xff, 0xab, 0x22, 0x76, 0x00, 0xfd, 0xe0, 0x0f, 0x1f, 0x01, 0x13,
+	0xfe, 0xdb, 0x21, 0x32,
+	0xef, 0xdd, 0x12, 0x77, 0x00, 0x0e, 0xff, 0xf0, 0x00, 0x14, 0x1f, 0x30,
+	0xff, 0xaf, 0x0e, 0xf0,
+	0x01, 0x22, 0x36, 0x00, 0x01, 0x20, 0xe0, 0xd0, 0xd1, 0x02, 0x23, 0xad,
+	0xab, 0x37, 0x30, 0xdf,
+	0xf0, 0x32, 0x10, 0xdf, 0xb0, 0x21, 0x21, 0x11, 0x01, 0xbe, 0x00, 0x31,
+	0x00, 0x01, 0xdc, 0xe9,
+	0x01, 0x45, 0xf1, 0x00, 0xf0, 0xce, 0x00, 0x00, 0xef, 0xab, 0x02, 0x35,
+	0xef, 0xde, 0x40, 0x00,
+	0x02, 0x74, 0x01, 0xde, 0xce, 0x12, 0x10, 0xcd, 0x01, 0x42, 0xfe, 0xff,
+	0x15, 0x00, 0xeb, 0x20,
+	0x43, 0xdf, 0xbd, 0x20, 0x11, 0x00, 0xcc, 0x0e, 0x56, 0xf1, 0xf0, 0x0f,
+	0x41, 0xff, 0x98, 0x10,
+	0x37, 0x10, 0xee, 0xe0, 0x0f, 0x10, 0x11, 0x21, 0x00, 0x20, 0x11, 0x00,
+	0xdf, 0x0e, 0x73, 0x03,
+	0xba, 0x00, 0x21, 0x0f, 0x20, 0x23, 0x00, 0x00, 0x01, 0xfc, 0x1f, 0x71,
+	0x12, 0xee, 0xe2, 0x02,
+	0x09, 0xdc, 0x0f, 0x30, 0xd0, 0xf7, 0x40, 0xef, 0xde, 0xdf, 0x51, 0x63,
+	0x0f, 0xf0, 0xee, 0xcc,
+	0xfd, 0x00, 0x45, 0x00, 0xed, 0xe0, 0x00, 0x3f, 0x33, 0x01, 0xd0, 0x0d,
+	0x36, 0x73, 0xed, 0xbb,
+	0x01, 0x16, 0x00, 0x0e, 0xdd, 0xf0, 0x76, 0x31, 0xde, 0xde, 0x54, 0x01,
+	0xec, 0xe0, 0x43, 0x00,
+	0xff, 0x01, 0x43, 0xec, 0xff, 0x10, 0x36, 0x20, 0x3f, 0x00, 0xc0, 0xff,
+	0x01, 0x55, 0x42, 0x0f,
+	0xfc, 0x00, 0x23, 0x20, 0x12, 0x73, 0x00, 0xda, 0x00, 0x41, 0x11, 0x0f,
+	0xff, 0xf0, 0x00, 0x34,
+	0x42, 0xfd, 0xdc, 0x13, 0x24, 0x00, 0x40, 0x0d, 0xff, 0xd0, 0x40, 0x01,
+	0x95, 0x02, 0x0d, 0xe9,
+	0xf1, 0x42, 0x3f, 0xec, 0xd0, 0x03, 0xf1, 0x40, 0x40, 0x0e, 0xe0, 0x00,
+	0xdd, 0xff, 0x1e, 0xb8,
+	0x3f, 0x77, 0x11, 0xdf, 0x00, 0x00, 0xac, 0x00, 0xf0, 0xe0, 0xf0, 0xef,
+	0x10, 0x72, 0x00, 0xfb,
+	0xff, 0x24, 0x01, 0xff, 0x0d, 0x0d, 0x00, 0x47, 0xd0, 0xcd, 0x00, 0x00,
+	0x00, 0x3f, 0xef, 0x8f,
+	0xff, 0x02, 0x13, 0xfc, 0xee, 0x76, 0x02, 0x00, 0x0c, 0xde, 0x00, 0x23,
+	0x0f, 0xee, 0xfe, 0x0c,
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+	0x03, 0x32, 0x1b, 0xdd,
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+	0x21, 0x1f, 0x0d, 0xff,
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+	0xee, 0xdf, 0xff, 0xe0,
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+	0xee, 0xda, 0x11, 0x52,
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+	0xef, 0xcb, 0x00, 0x20,
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+	0xef, 0xee, 0x0f, 0x10,
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+	0xee, 0xed, 0x12, 0x37,
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+	0x0c, 0x03, 0x24, 0xbc,
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+	0xb0, 0x00, 0x44, 0xef,
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+	0x32, 0x12, 0xf0, 0x00,
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+	0xfe, 0xfe, 0x00, 0xf0,
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+	0x8d, 0x00, 0x37, 0xe1,
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+	0x00, 0x9c, 0xf1, 0x03,
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+	0xfc, 0x70, 0x45, 0xe0,
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+	0xfd, 0xf0, 0xf0, 0xed,
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+	0x34, 0x00, 0xba, 0x0f,
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+	0x21, 0x21, 0x00, 0xee,
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+	0x0f, 0x00, 0xd0, 0xd1,
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+	0x30, 0x70, 0x54, 0xe3,
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+	0xef, 0xac, 0x11, 0x57,
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+	0x01, 0xf0, 0x2f, 0x42,
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+	0xcc, 0xf0, 0x14, 0x31,
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+	0xfe, 0x11, 0x55, 0x00,
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+	0xe1, 0x00, 0x57, 0x01,
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+	0x14, 0x51, 0x0f, 0xdd,
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+	0xfe, 0xf0, 0x31, 0x10,
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+	0x0c, 0xfe, 0x11, 0x35,
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+	0x00, 0xee, 0xf0, 0x00,
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+	0x37, 0x12, 0xfc, 0xdd,
+	0x11, 0x72, 0x00, 0xf2, 0x25, 0x20, 0x40, 0x13, 0x00, 0x0f, 0x74, 0x34,
+	0xfe, 0x1d, 0x30, 0x01,
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+	0xad, 0x01, 0x35, 0x01,
+	0x0f, 0x1f, 0x2f, 0x0d, 0xdd, 0x03, 0x27, 0x40, 0xfe, 0x0f, 0x72, 0x44,
+	0xdf, 0xda, 0xf0, 0x11,
+	0x41, 0xff, 0xfe, 0x01, 0x00, 0x9c, 0x00, 0x56, 0x00, 0xee, 0xe0, 0x10,
+	0x20, 0x10, 0x02, 0x00,
+	0xab, 0x0f, 0x01, 0x23, 0x2f, 0xdb, 0xf0, 0x44, 0x0e, 0xfd, 0x11, 0x44,
+	0xe0, 0xef, 0x00, 0xf1,
+	0xef, 0x0f, 0xee, 0x00, 0x31, 0x10, 0xec, 0xad, 0x01, 0x77, 0x11, 0xf0,
+	0xcd, 0x0f, 0x30, 0x22,
+	0x13, 0x25, 0xdf, 0xa9, 0x00, 0x41, 0x01, 0x0f, 0xff, 0xf0, 0xc0, 0xe0,
+	0xff, 0x00, 0x24, 0x21,
+	0xce, 0x00, 0x24, 0xff, 0x8a, 0x00, 0x45, 0xff, 0xbd, 0x00, 0x34, 0x0f,
+	0xde, 0x00, 0x14, 0x00,
+	0xee, 0x00, 0x15, 0x00, 0xce, 0xdf, 0x01, 0x21, 0xee, 0xdf, 0xe0, 0xd0,
+	0x9e, 0x03, 0x10, 0xf8,
+	0x01, 0x41, 0xe1, 0x12, 0x00, 0xfc, 0xd0, 0x01, 0xc0, 0x0f, 0x25, 0xf2,
+	0x8c, 0xeb, 0x0f, 0x31,
+	0x02, 0x8f, 0xff, 0x00, 0x01, 0x20, 0x03, 0x01, 0x0e, 0xfc, 0xfe, 0x11,
+	0x40, 0x0a, 0xbf, 0xe0,
+	0x12, 0x74, 0x0f, 0xea, 0x11, 0x44, 0xff, 0xee, 0x10, 0x21, 0xf0, 0xee,
+	0xfd, 0x00, 0x1f, 0x0d,
+	0x00, 0xf0, 0xf9, 0x10, 0x74, 0x12, 0xdf, 0xfd, 0xcd, 0xfc, 0x23, 0x56,
+	0xff, 0xbb, 0x0f, 0x00,
+	0x10, 0xe0, 0x0f, 0x02, 0x12, 0xf0, 0xec, 0xfe, 0xf0, 0x34, 0x44, 0xd1,
+	0xaf, 0x0f, 0x40, 0x20,
+	0xf0, 0xcb, 0x01, 0x34, 0xee, 0xde, 0x01, 0x34, 0xfe, 0xfd, 0x75, 0x33,
+	0xee, 0xbe, 0x00, 0x14,
+	0x24, 0xcb, 0xde, 0x51, 0x11, 0xef, 0xde, 0xf1, 0x02, 0x0f, 0x05, 0x05,
+	0x6f, 0x00, 0xc9, 0x00,
+	0x8f, 0xea, 0xf3, 0xf9, 0xd3, 0xea, 0xc6, 0xfb, 0xf2, 0x36, 0x72, 0x23,
+	0x21, 0x0e, 0x0e, 0x31,
+	0x00, 0x88, 0x10, 0x34, 0x00, 0xfd, 0x32, 0x11, 0xff, 0x00, 0x01, 0x24,
+	0x00, 0xbc, 0xce, 0x44,
+	0x76, 0x0f, 0xdc, 0x0f, 0x00, 0x30, 0x10, 0xf0, 0xd0, 0x40, 0x42, 0x00,
+	0xd0, 0xec, 0x01, 0x01,
+	0xff, 0x1c, 0x3f, 0x30, 0x34, 0xff, 0x1f, 0x71, 0x0e, 0xfe, 0x0d, 0xef,
+	0x11, 0x77, 0x05, 0x0d,
+	0xee, 0x01, 0x13, 0x72, 0x10, 0x01, 0xe0, 0xdc, 0xdb, 0xfe, 0x32, 0x45,
+	0x33, 0x01, 0xbd, 0x00,
+	0x1f, 0x20, 0x25, 0x33, 0x00, 0x0f, 0xee, 0xed, 0xd0, 0xe2, 0xb3, 0xbf,
+	0xda, 0x04, 0x34, 0x10,
+	0x0e, 0x71, 0x47, 0xf4, 0xee, 0x0e, 0x31, 0x00, 0xe0, 0xcf, 0xe0, 0xcf,
+	0x00, 0x21, 0x12, 0x11,
+	0xd2, 0xc0, 0x11, 0x61, 0x02, 0x02, 0x11, 0x0e, 0xa8, 0x0d, 0x01, 0x20,
+	0x0e, 0xbc, 0xe0, 0x01,
+	0x03, 0x2d, 0x1f, 0x06, 0xe0, 0xcc, 0x01, 0x04, 0x31, 0xeb, 0x9c, 0x00,
+	0x04, 0x0f, 0x0f, 0x10,
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+	0xe0, 0x23, 0x22, 0x19,
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+	0x00, 0x44, 0x00, 0xed,
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+	0xf1, 0xed, 0xff, 0x14,
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+	0x30, 0x61, 0x12, 0x0f,
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+	0x20, 0x0c, 0x0c, 0x20,
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+	0xc8, 0xec, 0x22, 0x73,
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+	0x31, 0x32, 0xc0, 0xba,
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+	0xf0, 0x42, 0x04, 0xef,
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+	0xdc, 0x00, 0x05, 0xf0,
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+	0x0e, 0xd0, 0x02, 0x0d,
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+	0xdb, 0xff, 0x20, 0x50,
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+	0x01, 0x01, 0x00, 0xed,
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+	0x32, 0x44, 0x0f, 0xed,
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+	0x00, 0x11, 0x10, 0x00,
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+	0xd0, 0x20, 0x70, 0x10,
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+	0xf0, 0xfc, 0xfb, 0xef,
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+	0x01, 0x9d, 0xfb, 0x00,
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+	0x02, 0xfe, 0xf1, 0x14,
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+	0xfe, 0xff, 0x20, 0x51,
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+	0xe9, 0x00, 0x51, 0xfc,
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+	0x30, 0x24, 0x00, 0xd0,
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+	0x0f, 0xfd, 0xf0, 0xfe,
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+	0xde, 0xfe, 0x5f, 0x23,
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+	0xee, 0xed, 0x12, 0x53,
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+	0x90, 0x01, 0x03, 0x32,
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+	0x1f, 0x35, 0x00, 0xce,
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+	0x0e, 0x01, 0x24, 0x12,
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+	0x02, 0x21, 0xf0, 0xdf,
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+	0xc8, 0xbc, 0xf0, 0x0f,
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+	0xbd, 0xed, 0x40, 0x6e,
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+	0xef, 0x02, 0x45, 0x20,
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+	0xff, 0x02, 0x13, 0x0f,
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+	0xe1, 0xf2, 0xe3, 0xec,
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+	0x0f, 0xdd, 0xce, 0x21,
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+	0xb1, 0x00, 0x00, 0xd2,
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+	0xbf, 0x10, 0x12, 0xff,
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+	0x22, 0x00, 0x43, 0x35,
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+	0xe0, 0x41, 0x53, 0x20,
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+	0x00, 0x73, 0x1d, 0x30,
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+	0x23, 0x20, 0x11, 0x23,
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+	0x12, 0x00, 0xf0, 0x00,
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+	0xe0, 0xb0, 0xc1, 0xf2,
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+	0x44, 0x00, 0xff, 0xe0,
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+	0xfc, 0xce, 0x0f, 0x20,
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+	0x00, 0x75, 0x21, 0x1d,
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+	0xb0, 0x01, 0x37, 0x35,
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+	0xfe, 0xff, 0x03, 0x22,
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+	0xf0, 0x12, 0x52, 0x0f,
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+	0x0d, 0xdd, 0x0f, 0xf0,
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+	0x0f, 0x20, 0xfb, 0xd0,
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+	0x0e, 0xfe, 0x00, 0x21,
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+	0x1a, 0x00, 0x3c, 0x00,
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+	0xe0, 0xd0, 0x00, 0x06,
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+	0xed, 0xdd, 0xd0, 0xe1,
+	0xa0, 0xda, 0xdf, 0xf0, 0x03, 0x11, 0x11, 0x14, 0xfe, 0xfa, 0x71, 0x01,
+	0xcf, 0x00, 0x34, 0x44,
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+	0x10, 0x3f, 0x2f, 0xfe,
+	0xc8, 0xcf, 0x12, 0x34, 0xe0, 0x00, 0x42, 0x01, 0x30, 0x22, 0xf1, 0xf0,
+	0x0b, 0xce, 0xfd, 0x50,
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+	0x00, 0xff, 0x22, 0x40,
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+	0x00, 0x77, 0x32, 0xb9,
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+	0xbc, 0x00, 0x00, 0x10,
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+	0x34, 0x13, 0xeb, 0xcc,
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+	0xcd, 0x00, 0x27, 0x52,
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+	0x01, 0xd2, 0x9e, 0xe0,
+	0x17, 0x27, 0xcd, 0xbf, 0x00, 0xe1, 0xd0, 0xcf, 0xcd, 0xdd, 0xdd, 0x02,
+	0x77, 0x00, 0xef, 0xd0,
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+	0xdd, 0xff, 0x01, 0x23,
+	0xf1, 0x15, 0x52, 0xf0, 0xdf, 0x0f, 0xff, 0xf1, 0x01, 0x2d, 0x2d, 0x0f,
+	0x4e, 0x74, 0x11, 0x0d,
+	0x22, 0x47, 0x25, 0x0e, 0x0e, 0x00, 0x10, 0x72, 0x0e, 0xfc, 0x14, 0x11,
+	0xec, 0xbe, 0x22, 0x36,
+	0x0f, 0x0f, 0x11, 0x64, 0x20, 0x0d, 0x01, 0x14, 0x50, 0x1d, 0x30, 0x32,
+	0x01, 0x0c, 0xfe, 0xf0,
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+	0x0d, 0x00, 0xf1, 0x10,
+	0x2c, 0x2e, 0x43, 0x44, 0x0c, 0xdb, 0xff, 0x21, 0x75, 0x72, 0x40, 0xdd,
+	0x0f, 0x00, 0x01, 0x0e,
+	0x0c, 0xd0, 0x02, 0x32, 0xff, 0xba, 0x00, 0x24, 0xf0, 0xe0, 0xf1, 0xcd,
+	0xfd, 0x27, 0x42, 0xf0,
+	0xaa, 0x00, 0x12, 0xef, 0xcd, 0x01, 0x34, 0xd0, 0xd0, 0x12, 0x13, 0xae,
+	0xdb, 0xf0, 0x01, 0x11,
+	0xff, 0xdc, 0x00, 0x34, 0x00, 0x2e, 0xce, 0xb0, 0xe0, 0x90, 0xec, 0x12,
+	0x37, 0x12, 0xef, 0xde,
+	0xb0, 0x02, 0x12, 0x00, 0xcb, 0x00, 0x13, 0xf1, 0x8b, 0xef, 0x00, 0x02,
+	0x00, 0xf0, 0xfc, 0xd0,
+	0x05, 0x22, 0x0f, 0xbc, 0xd0, 0x00, 0xdf, 0xec, 0xd0, 0x10, 0x20, 0x00,
+	0x66, 0x31, 0x0f, 0xdc,
+	0xa0, 0x00, 0x13, 0x01, 0xf1, 0x22, 0x00, 0xe0, 0xec, 0xcf, 0xdc, 0xf0,
+	0x57, 0x20, 0x00, 0x52,
+	0x33, 0x00, 0x0e, 0x00, 0x01, 0x42, 0x11, 0xc8, 0xcf, 0x10, 0x22, 0x11,
+	0x00, 0xfe, 0xf0, 0xc0,
+	0xf1, 0xf0, 0x00, 0x67, 0x35, 0xfc, 0xe0, 0x1f, 0x20, 0x4e, 0x40, 0x11,
+	0x43, 0x51, 0x0f, 0x0e,
+	0x21, 0x23, 0x0e, 0xed, 0x00, 0x20, 0x46, 0xef, 0xcd, 0x10, 0x74, 0x14,
+	0xe9, 0xcf, 0x20, 0x23,
+	0x0f, 0xdd, 0x10, 0x71, 0x14, 0x0d, 0x0e, 0x11, 0x56, 0x00, 0x0e, 0x1f,
+	0x00, 0x10, 0x52, 0x00,
+	0x0a, 0x00, 0xf0, 0x10, 0x00, 0x23, 0x52, 0x00, 0x08, 0xe0, 0x31, 0x43,
+	0xfe, 0xdd, 0x00, 0x23,
+	0x1e, 0xfa, 0xbf, 0x01, 0x32, 0x21, 0x13, 0x00, 0xcf, 0x13, 0x77, 0x20,
+	0x00, 0x0e, 0x0f, 0xee,
+	0xce, 0xd0, 0x10, 0x72, 0x23, 0xd0, 0xf0, 0x57, 0x43, 0x0f, 0xdc, 0x0f,
+	0xfc, 0x00, 0x24, 0xef,
+	0xee, 0xf0, 0xe0, 0xee, 0xcd, 0x12, 0x47, 0xf0, 0xcf, 0x13, 0x0f, 0x9a,
+	0xc0, 0x01, 0x13, 0xff,
+	0xbc, 0x01, 0x02, 0xf0, 0xce, 0xff, 0x12, 0x1f, 0xd8, 0xdf, 0xd0, 0xde,
+	0xf1, 0x77, 0x41, 0xef,
+	0xce, 0x10, 0x01, 0x00, 0xff, 0x00, 0x44, 0x00, 0xdd, 0x00, 0x15, 0xff,
+	0x90, 0x01, 0x21, 0x1f,
+	0x0f, 0x32, 0x35, 0x00, 0x00, 0x5f, 0x0d, 0x10, 0x00, 0x00, 0x0c, 0xf1,
+	0x07, 0x00, 0x11, 0xf1,
+	0xa0, 0x1e, 0x70, 0x01, 0x00, 0x00, 0x2f, 0x2f, 0x2f, 0x0a, 0xee, 0x00,
+	0x1f, 0x0e, 0x01, 0x77,
+	0x77, 0x01, 0x0e, 0xdd, 0x00, 0x20, 0x03, 0x0f, 0xdc, 0x11, 0x33, 0x00,
+	0x0e, 0x3f, 0xcd, 0x23,
+	0x77, 0x10, 0xdb, 0x00, 0x00, 0x10, 0x10, 0x01, 0x0f, 0xae, 0xfe, 0x01,
+	0x63, 0x10, 0x00, 0x0d,
+	0x32, 0x45, 0xff, 0xfd, 0xff, 0x33, 0x26, 0xf0, 0xcc, 0xff, 0x10, 0x10,
+	0xdd, 0xca, 0x00, 0x34,
+	0x20, 0x20, 0x73, 0x44, 0xee, 0x0d, 0x00, 0xe4, 0x01, 0x01, 0x10, 0x13,
+	0xf0, 0xa9, 0xf0, 0x20,
+	0x32, 0xbd, 0xed, 0x03, 0xf1, 0xd2, 0x37, 0x72, 0x1f, 0xfd, 0x00, 0x01,
+	0xfd, 0xca, 0x00, 0x42,
+	0xf0, 0x8c, 0x00, 0x02, 0x02, 0xe1, 0x01, 0x10, 0xf0, 0xd1, 0x25, 0x1e,
+	0xcb, 0xd0, 0x12, 0x01,
+	0x8b, 0xef, 0x00, 0x02, 0xe0, 0xf2, 0xf2, 0x00, 0xdc, 0xff, 0xec, 0x00,
+	0xef, 0xbf, 0xcd, 0xfe,
+	0x26, 0x43, 0xf0, 0xce, 0x0e, 0x0e, 0xe0, 0xdd, 0x00, 0x37, 0x77, 0xf0,
+	0xcc, 0x11, 0x02, 0xe0,
+	0xdf, 0x0d, 0x00, 0x74, 0x0f, 0x0c, 0x00, 0x24, 0x31, 0x42, 0x1f, 0xec,
+	0xef, 0x12, 0x33, 0xf0,
+	0xc1, 0x20, 0x2f, 0xd9, 0xa0, 0x10, 0x52, 0x00, 0x10, 0x10, 0x67, 0x00,
+	0x0e, 0x01, 0x34, 0x10,
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+	0x56, 0x10, 0x0f, 0x00,
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+	0xc0, 0xe3, 0x04, 0x80,
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+	0xdd, 0xbd, 0xd0, 0xb3,
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+	0x00, 0x22, 0x00, 0x0f,
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+	0x00, 0x0c, 0x1e, 0x4f,
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+	0x2c, 0x2e, 0x2c, 0x1c,
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+	0x32, 0x30, 0x65, 0x35,
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+	0xe3, 0xc3, 0xd3, 0xe6,
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+	0xb0, 0xe0, 0xd3, 0xd5,
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+	0xba, 0xdb, 0xff, 0x2f,
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+	0xce, 0xae, 0xcc, 0xdd,
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+	0xcc, 0xdd, 0x0f, 0x4f,
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+	0xe0, 0xf0, 0xdf, 0xdd,
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+	0x11, 0x21, 0x23, 0x20,
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+	0xdb, 0xfd, 0x1c, 0x0c,
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+	0x4f, 0x31, 0x43, 0x30,
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+	0x43, 0x33, 0x65, 0x44,
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+	0x27, 0x67, 0x11, 0x12,
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+	0x12, 0x01, 0x00, 0x02,
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+	0xe0, 0xdf, 0xd0, 0x13,
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+	0xb8, 0xdc, 0xf0, 0xf0,
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+	0x0f, 0xee, 0xdf, 0x00,
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+	0x10, 0x0f, 0xfc, 0xfe,
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+	0xc8, 0xdd, 0xe0, 0x00,
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+	0xee, 0xdd, 0xee, 0xcc,
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+	0x20, 0x42, 0x47, 0x42,
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+	0xef, 0x03, 0x05, 0xf1,
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+	0x02, 0x01, 0x43, 0x53,
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+	0x55, 0x11, 0x22, 0x01,
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+	0x40, 0x0d, 0x0e, 0x0e,
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+	0x24, 0x02, 0x90, 0xe0,
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+	0x33, 0x0f, 0xcb, 0xfe,
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+	0xdd, 0xed, 0xf0, 0xf1,
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+	0xdd, 0xdc, 0xbb, 0xdc,
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+	0x38, 0x1c, 0x4e, 0x40,
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+	0x20, 0x11, 0x21, 0x41,
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+	0x22, 0x22, 0x14, 0x01,
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+	0xd2, 0xe4, 0xe2, 0xb0,
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+	0x05, 0x24, 0x12, 0xe0,
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+	0x0e, 0x0f, 0x20, 0x1e,
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+	0xdf, 0xf0, 0x02, 0x01,
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+	0x62, 0x21, 0x30, 0x43,
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+	0xe2, 0xe2, 0xf0, 0xf2,
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+	0x0c, 0x09, 0x1f, 0x71,
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+	0x45, 0x10, 0x12, 0x12,
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+	0x20, 0x0d, 0x0c, 0xf0,
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+	0xb2, 0xd2, 0x90, 0xed,
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+	0x0d, 0xff, 0x22, 0x41,
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+	0xf0, 0x00, 0xee, 0xff,
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+	0xdc, 0x30, 0x74, 0x70,
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+	0xe0, 0xd0, 0xee, 0xef,
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+	0x0f, 0x1e, 0x2d, 0x3d,
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+	0x10, 0xff, 0xf0, 0xdf,
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+	0x0d, 0x1e, 0x1c, 0x09,
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+	0x0e, 0xad, 0xcc, 0xf0,
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+	0xcc, 0xdb, 0x00, 0x10,
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+	0xd1, 0xac, 0xe0, 0xf7,
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+	0x07, 0x03, 0xf4, 0x05,
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+	0x23, 0xc1, 0xa2, 0xf4,
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+	0xed, 0xed, 0xfc, 0xdb,
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+	0xed, 0xcd, 0xdc, 0xdc,
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+	0x01, 0xe0, 0xf0, 0x11,
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+	0x0e, 0xfb, 0xdd, 0x0d,
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+	0x45, 0x56, 0x21, 0x11,
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+	0xd2, 0xf6, 0x03, 0x14,
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+	0x13, 0x22, 0x00, 0xff,
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+	0xad, 0xdd, 0xde, 0xdd,
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+	0x00, 0xff, 0xd8, 0xed,
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+	0x0f, 0xf9, 0xce, 0xed,
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+	0xef, 0xed, 0xdb, 0xf0,
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+	0xbf, 0xc8, 0xdf, 0xdf,
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+	0x60, 0x41, 0x1e, 0x31,
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+	0x47, 0x21, 0x02, 0x21,
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+	0x01, 0x01, 0x25, 0x34,
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+	0x01, 0x67, 0x35, 0x32,
+	0x32, 0x32, 0x01, 0x00, 0xf0, 0xd0, 0x90, 0xcf, 0xd0, 0x02, 0x02, 0xf1,
+	0x13, 0x35, 0x74, 0x23,
+	0x32, 0x22, 0x33, 0x56, 0x10, 0x0f, 0x0d, 0x0c, 0xdb, 0xce, 0x0e, 0x10,
+	0x0f, 0xed, 0xfe, 0x20,
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+	0xd2, 0xb0, 0xee, 0xff,
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+	0x8a, 0x12, 0x04, 0xcf,
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+	0x00, 0x00, 0xd0, 0xcf,
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+	0xb0, 0xbc, 0xb8, 0xef,
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+	0x3e, 0x4c, 0x2c, 0x1b,
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+	0x21, 0x1e, 0x2f, 0x53,
+	0x21, 0x1e, 0x20, 0x42, 0x55, 0x45, 0x45, 0x23, 0x21, 0x23, 0x33, 0x35,
+	0x33, 0x34, 0x32, 0x23,
+	0x01, 0xf0, 0xe0, 0xe0, 0xdf, 0xdf, 0x03, 0x57, 0x02, 0xe2, 0xe2, 0x24,
+	0x00, 0xa0, 0xd0, 0xc0,
+	0xf3, 0xd2, 0xd2, 0xe3, 0xe3, 0xf7, 0xe3, 0x80, 0xd0, 0xd2, 0xe4, 0x05,
+	0xe1, 0xc2, 0x02, 0x02,
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+	0x0f, 0x0e, 0x0e, 0x0f,
+	0xff, 0x0e, 0xfe, 0xfc, 0xda, 0xfe, 0xce, 0x8d, 0xed, 0xee, 0xdf, 0xbe,
+	0xef, 0xcf, 0xdf, 0xef,
+	0xdc, 0xc9, 0xff, 0xf0, 0xff, 0xfc, 0xcc, 0xdd, 0xed, 0x1e, 0x1d, 0x0c,
+	0x2e, 0x71, 0x64, 0x00,
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+	0xf0, 0xdf, 0xdf, 0xf4,
+	0x04, 0x04, 0x66, 0x44, 0x12, 0x00, 0xc0, 0xc0, 0xf0, 0xe0, 0xbd, 0xdd,
+	0xee, 0xee, 0xcd, 0xdd,
+	0xce, 0xde, 0xd0, 0xf7, 0xf3, 0xac, 0xee, 0x01, 0x36, 0x30, 0x1d, 0x0b,
+	0x2e, 0x1d, 0x2d, 0x3d,
+	0x5d, 0x1d, 0x0a, 0x30, 0x72, 0x51, 0x10, 0x10, 0x20, 0x1c, 0x0c, 0x30,
+	0x51, 0x20, 0x1f, 0x11,
+	0x30, 0x1d, 0x2c, 0x62, 0x36, 0x31, 0x30, 0x43, 0x65, 0x11, 0x20, 0x11,
+	0x22, 0x10, 0x0c, 0xfb,
+	0xfd, 0x00, 0x00, 0xd0, 0xde, 0xf0, 0x01, 0x05, 0x24, 0x13, 0x13, 0x66,
+	0x22, 0x01, 0xd0, 0xe0,
+	0x03, 0xf2, 0x04, 0xe2, 0x8d, 0xb1, 0x47, 0x26, 0x01, 0xf0, 0xf2, 0x01,
+	0xf1, 0xc0, 0x00, 0x01,
+	0xf0, 0xcf, 0x00, 0x22, 0x10, 0x00, 0x01, 0x66, 0x01, 0xe0, 0xef, 0xe0,
+	0xe1, 0xae, 0xf0, 0x23,
+	0x20, 0x1d, 0x0c, 0xe8, 0x0e, 0x50, 0x0f, 0xdc, 0xcc, 0xbd, 0xee, 0xef,
+	0xdf, 0xde, 0x01, 0xf2,
+	0xce, 0xde, 0xf0, 0xff, 0x0f, 0x0b, 0xcb, 0x0d, 0x50, 0x2f, 0x1d, 0x1e,
+	0x3f, 0x3e, 0x5f, 0x50,
+	0x22, 0x11, 0x1f, 0x1c, 0x1b, 0x0a, 0xcc, 0xbb, 0xde, 0xde, 0xe0, 0xde,
+	0xef, 0x01, 0x43, 0x34,
+	0x51, 0x11, 0xde, 0xcd, 0xf0, 0x02, 0xce, 0xbb, 0xe0, 0xf1, 0xe2, 0xe0,
+	0x02, 0x16, 0x23, 0x04,
+	0xf3, 0xf5, 0xc2, 0xb0, 0xe2, 0xd0, 0xcf, 0xbe, 0xed, 0xdd, 0xea, 0xfe,
+	0x0f, 0xfe, 0x0e, 0x30,
+	0x2f, 0x1d, 0x0c, 0x0f, 0x20, 0x52, 0x32, 0x44, 0x23, 0x33, 0x32, 0x11,
+	0x20, 0x73, 0x67, 0x02,
+	0x01, 0x00, 0x03, 0x07, 0x22, 0x10, 0x11, 0x43, 0x44, 0x75, 0x22, 0x11,
+	0x21, 0x10, 0x21, 0x34,
+	0x43, 0x11, 0x22, 0x11, 0x01, 0x00, 0x03, 0x77, 0x67, 0x03, 0xff, 0xff,
+	0x00, 0xff, 0x00, 0x33,
+	0x21, 0xeb, 0xde, 0xdf, 0xef, 0x77, 0x64, 0x00, 0x00, 0xf0, 0x00, 0x13,
+	0xf0, 0xaa, 0xff, 0x00,
+	0x02, 0x33, 0x10, 0x32, 0x22, 0xfe, 0x9a, 0xee, 0x11, 0x67, 0x22, 0x00,
+	0x00, 0xcb, 0xac, 0xfe,
+	0x34, 0x54, 0x00, 0x0f, 0xff, 0xed, 0x00, 0x00, 0x11, 0x22, 0x00, 0x0f,
+	0xed, 0x00, 0x36, 0x21,
+	0xce, 0x9a, 0xdd, 0x11, 0x77, 0x11, 0xee, 0xf0, 0x00, 0x00, 0x0f, 0x00,
+	0x00, 0xf0, 0x00, 0x44,
+	0x00, 0xfe, 0xef, 0x00, 0xf0, 0x0f, 0x01, 0x00, 0xce, 0xfe, 0x04, 0x04,
+	0x91, 0x02, 0xe9, 0x01,
+	0x57, 0x01, 0x18, 0x09, 0xca, 0x02, 0x2c, 0x08, 0xcc, 0xba, 0xf0, 0x45,
+	0x43, 0x01, 0xee, 0xde,
+	0xee, 0x00, 0x23, 0xed, 0x99, 0x12, 0x55, 0x00, 0xff, 0xef, 0xff, 0x00,
+	0x01, 0xf0, 0xff, 0x00,
+	0xe0, 0xe0, 0x23, 0x23, 0xbc, 0xb9, 0x00, 0x34, 0x00, 0xce, 0xff, 0x00,
+	0x00, 0x12, 0x21, 0xf0,
+	0xed, 0xdd, 0xff, 0x64, 0x23, 0x00, 0xfd, 0xda, 0xef, 0x20, 0x0f, 0x0f,
+	0x52, 0x00, 0xec, 0xdd,
+	0xa9, 0x10, 0x75, 0x11, 0x00, 0x0f, 0xfd, 0x00, 0x22, 0x10, 0xfd, 0x22,
+	0x32, 0x10, 0x54, 0x33,
+	0x00, 0x0f, 0xda, 0xee, 0x76, 0x32, 0xff, 0x00, 0x32, 0x00, 0xef, 0xed,
+	0x00, 0x22, 0x77, 0x10,
+	0xf0, 0x00, 0x00, 0x55, 0x11, 0xde, 0xbc, 0x00, 0x35, 0x11, 0xef, 0x00,
+	0x01, 0x11, 0x01, 0x22,
+	0x01, 0x01, 0x01, 0x32, 0x34, 0x9e, 0x13, 0x57, 0xde, 0xef, 0x43, 0xf0,
+	0xdd, 0x45, 0x33, 0xff,
+	0xdd, 0xab, 0x00, 0x44, 0x00, 0x00, 0x33, 0xff, 0xcb, 0x00, 0x21, 0xde,
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+	0x22, 0x0d, 0x1d, 0x0e,
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+	0xed, 0x0f, 0x0d, 0x0d,
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+	0xee, 0xdd, 0xdc, 0x32,
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+	0xef, 0xe0, 0xcc, 0x11,
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+	0x00, 0x0f, 0x00, 0x00,
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+	0x41, 0x32, 0x0f, 0xda,
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+	0xdd, 0xef, 0xff, 0x00,
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+	0xff, 0xff, 0x00, 0x35,
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+	0x22, 0x77, 0x12, 0x0f,
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+	0xcf, 0xf0, 0x10, 0x01,
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+	0xf0, 0xee, 0xee, 0x32,
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+	0xd0, 0x00, 0x31, 0xc9,
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+	0x12, 0xdd, 0xca, 0xef,
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+	0x56, 0x00, 0xfe, 0x00,
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+	0x00, 0xff, 0x33, 0x11,
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+	0x31, 0x20, 0x11, 0x00,
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+	0x32, 0x11, 0x22, 0x75,
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+	0xed, 0xfe, 0x76, 0x32,
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+	0x77, 0x00, 0xdb, 0x30,
+	0x51, 0x00, 0xfe, 0x0f, 0x10, 0x0f, 0xfc, 0x0f, 0x10, 0x00, 0xcd, 0xdc,
+	0x65, 0x11, 0xcc, 0xed,
+	0x10, 0x10, 0x11, 0x12, 0x00, 0xff, 0xff, 0x00, 0x10, 0xcd, 0x88, 0xfe,
+	0x11, 0x34, 0x00, 0x00,
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+	0x01, 0x16, 0x01, 0xf0,
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+	0x00, 0xff, 0xdc, 0xe0,
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+	0x11, 0xed, 0x00, 0x22,
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+	0x20, 0x20, 0x20, 0xef,
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+	0xeb, 0xdc, 0xfe, 0x0f,
+	0xfe, 0x20, 0x33, 0x00, 0xdd, 0x8a, 0xdd, 0x00, 0x12, 0x0f, 0xff, 0x0f,
+	0xfe, 0xfd, 0x01, 0x36,
+	0x9e, 0xdd, 0x00, 0x00, 0x00, 0xf0, 0xbc, 0xed, 0x15, 0x12, 0xdc, 0xff,
+	0x0f, 0xce, 0x00, 0x57,
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+	0xdd, 0xba, 0x12, 0x01,
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+	0xbc, 0xf0, 0x00, 0x0d,
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+	0x00, 0x11, 0x0f, 0xfe,
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+	0xf0, 0x00, 0xfe, 0xdd,
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+	0xff, 0x00, 0xbc, 0x00,
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+	0x00, 0xee, 0xff, 0x23,
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+	0x36, 0x00, 0x01, 0x00,
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+	0x00, 0xf0, 0x00, 0x00,
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+	0x0f, 0xde, 0x00, 0x77,
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+	0x07, 0x22, 0x10, 0xfd,
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+	0x01, 0x00, 0xeb, 0x11,
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+	0xbe, 0xdc, 0x03, 0x36,
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+	0x26, 0x52, 0xdd, 0xee,
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+	0xd0, 0xff, 0x01, 0x01,
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+	0x10, 0x10, 0x00, 0x51,
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+	0x24, 0x20, 0xed, 0xdb,
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+	0x10, 0x01, 0x12, 0xf0,
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+	0x0e, 0x33, 0x04, 0xde,
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+	0x07, 0x10, 0xee, 0xeb,
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+	0x01, 0xe0, 0x10, 0x31,
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+	0xee, 0x00, 0x10, 0x32,
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+	0x11, 0x74, 0x10, 0x2f,
+	0x51, 0x47, 0x23, 0x0e, 0xfe, 0x00, 0x51, 0x00, 0x1e, 0x11, 0x37, 0x20,
+	0x01, 0xe0, 0x01, 0x43,
+	0x1f, 0x52, 0xf2, 0xe3, 0x27, 0x52, 0x02, 0x00, 0x51, 0x25, 0x00, 0x2f,
+	0x30, 0xf4, 0x04, 0x44,
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+	0x12, 0x00, 0x10, 0x00,
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+	0x00, 0x20, 0x30, 0x20,
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+	0x01, 0x00, 0x10, 0x2b,
+	0x00, 0xd4, 0xee, 0x1e, 0x1a, 0xff, 0x64, 0x01, 0x1f, 0x30, 0xff, 0x8b,
+	0x20, 0x31, 0xd0, 0xcc,
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+	0x2e, 0x0c, 0xf0, 0x0d,
+	0xdc, 0x20, 0x31, 0xee, 0x88, 0x0f, 0x2f, 0xed, 0x9c, 0x01, 0x11, 0xee,
+	0xbd, 0x10, 0x00, 0xdb,
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+	0xf0, 0xe1, 0xbf, 0xdd,
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+	0xd0, 0xd4, 0x1f, 0x3e,
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+	0xf0, 0x01, 0x0d, 0xef,
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+	0xee, 0x20, 0x0d, 0xab,
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+	0xde, 0xee, 0xee, 0xcb,
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+	0xde, 0x0c, 0x30, 0x20,
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+	0xec, 0xf0, 0x23, 0x32,
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+	0x00, 0x26, 0xff, 0xcf,
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+	0x0f, 0x00, 0x90, 0x00,
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+	0xed, 0xa0, 0xcb, 0xee,
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+	0x0f, 0x2f, 0x12, 0x03,
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+	0x00, 0x43, 0xf0, 0x0d,
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+	0x2e, 0x02, 0x04, 0x45,
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+	0x03, 0xd2, 0xa1, 0x0f,
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+	0xb0, 0xf2, 0xf0, 0xfc,
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+	0xde, 0xbd, 0x00, 0x03,
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+	0xbe, 0xe9, 0x00, 0x13,
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+	0xe1, 0x0f, 0xec, 0x11,
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+	0x3f, 0x7f, 0x2c, 0x0f,
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+	0xca, 0xde, 0x0f, 0x3c,
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+	0xed, 0xdb, 0x9d, 0xec,
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+	0x10, 0x0f, 0xef, 0xde,
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+	0x00, 0x10, 0x4e, 0xfd,
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+	0xfe, 0x00, 0x10, 0xfe,
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+	0x0f, 0xde, 0xcf, 0xdc,
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+	0xd0, 0x05, 0x03, 0xf1,
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+	0x01, 0xf2, 0xd3, 0x03,
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+	0xed, 0x0f, 0x00, 0x43,
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+	0xcd, 0xeb, 0xe0, 0xdd,
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+	0xcd, 0xea, 0xf0, 0xf3,
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+	0xc0, 0xff, 0x0f, 0x10,
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+	0x12, 0x52, 0x1b, 0x00,
+	0x05, 0xe2, 0xd0, 0xf0, 0x17, 0x30, 0xdf, 0xdb, 0x00, 0x00, 0x90, 0xe0,
+	0xff, 0x10, 0x71, 0x31,
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+	0x75, 0x72, 0x12, 0x21,
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+	0x01, 0x36, 0x32, 0xd0,
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+	0x0f, 0x53, 0x23, 0x00,
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+	0x00, 0x2e, 0x30, 0x00,
+	0xc2, 0xf4, 0xec, 0xfe, 0xd2, 0xe5, 0xf0, 0x01, 0x57, 0x12, 0x0f, 0xdb,
+	0x70, 0x5f, 0x0f, 0x00,
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+	0x0f, 0x10, 0x1f, 0xf0,
+	0xe4, 0x4f, 0x69, 0x1e, 0x3f, 0x00, 0x01, 0x05, 0x01, 0xf1, 0x11, 0x41,
+	0x01, 0xe0, 0xb1, 0xfe,
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+	0x00, 0x42, 0x01, 0x80,
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+	0x11, 0x13, 0x10, 0xf0,
+	0x04, 0x47, 0x14, 0x22, 0x21, 0x01, 0xbf, 0xc0, 0xe0, 0x01, 0xe0, 0x00,
+	0x20, 0x60, 0x13, 0x02,
+	0x13, 0x21, 0x61, 0x23, 0x00, 0x34, 0x13, 0x2f, 0x2d, 0x08, 0x0d, 0x00,
+	0x42, 0x04, 0xee, 0xcd,
+	0x11, 0x46, 0x20, 0x0f, 0x32, 0x25, 0xfd, 0x0e, 0x40, 0x31, 0x9f, 0xdd,
+	0xf3, 0xf5, 0xf2, 0x02,
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+	0x14, 0x01, 0x1e, 0x2c,
+	0x40, 0x57, 0x03, 0xd1, 0xef, 0x10, 0x55, 0x11, 0x3e, 0x2d, 0xff, 0xfe,
+	0x0e, 0x21, 0x02, 0x0e,
+	0x70, 0x30, 0x00, 0xe7, 0xf7, 0xe1, 0x0f, 0x40, 0x20, 0xb1, 0xe0, 0x0e,
+	0x0d, 0x20, 0x51, 0x02,
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+	0xf0, 0x30, 0x42, 0x32,
+	0x21, 0x60, 0x2f, 0x00, 0xf0, 0x12, 0x51, 0x12, 0x15, 0x35, 0x23, 0xe0,
+	0xd2, 0xc1, 0xc2, 0xf0,
+	0xbf, 0xed, 0xd0, 0x00, 0x01, 0xdf, 0x0a, 0x60, 0x24, 0xf1, 0xbd, 0x0c,
+	0x30, 0x1f, 0x2d, 0x0d,
+	0xfb, 0xee, 0xdb, 0xdf, 0x0d, 0xfe, 0x9c, 0xee, 0xe0, 0x00, 0xed, 0x0a,
+	0x2f, 0x2d, 0xcd, 0xcc,
+	0xdf, 0xdd, 0xfc, 0xde, 0xac, 0xce, 0xdf, 0xcf, 0xdd, 0xde, 0xea, 0x0e,
+	0x4d, 0x50, 0xdc, 0xba,
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+	0x20, 0xf9, 0xce, 0x0f,
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+	0x5b, 0x20, 0x20, 0x2e,
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+	0xfb, 0x1d, 0x4f, 0x32,
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+	0x3e, 0x0e, 0x0c, 0xe9,
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+	0x2e, 0x2d, 0x18, 0x1f,
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+	0x06, 0x21, 0x50, 0x70,
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+	0xd0, 0xee, 0x01, 0xe7,
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+	0x0b, 0xdb, 0xfd, 0x0d,
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+	0xdf, 0x0f, 0xee, 0x2f,
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+	0x34, 0x21, 0xf7, 0x07,
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+	0xf4, 0xc2, 0xe3, 0x02,
+	0xd2, 0xe2, 0xc2, 0xd3, 0xf4, 0xe4, 0x04, 0xf4, 0x02, 0x03, 0xd3, 0x34,
+	0x77, 0x21, 0x21, 0x21,
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+	0xf3, 0x00, 0xe4, 0x03,
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+	0xff, 0x11, 0x57, 0x17,
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+	0x10, 0xf4, 0x15, 0x43,
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+	0x35, 0xd2, 0xf1, 0x21,
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+	0xbd, 0xfe, 0x00, 0x3f,
+	0x10, 0x13, 0x57, 0x11, 0x10, 0x1c, 0x20, 0x31, 0xee, 0x9f, 0x0f, 0x02,
+	0x26, 0x11, 0x00, 0xff,
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+	0xf2, 0xe2, 0xe4, 0xa4,
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+	0x41, 0x10, 0x0d, 0x0f,
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+	0xba, 0x0f, 0x1f, 0x00,
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+	0xad, 0xf0, 0x1e, 0x5e,
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+	0x03, 0x02, 0xf5, 0xf1,
+	0x0f, 0x0e, 0xbb, 0xed, 0x23, 0x56, 0x13, 0x11, 0x20, 0x10, 0x4b, 0x51,
+	0x01, 0xe0, 0xce, 0x03,
+	0xe5, 0xdf, 0xbc, 0xe0, 0xef, 0xcf, 0xf0, 0x06, 0x02, 0xf1, 0x0f, 0x60,
+	0x35, 0x30, 0xe0, 0x00,
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+	0x00, 0x02, 0xe4, 0x02,
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+	0xe2, 0xe0, 0x02, 0x21,
+	0x70, 0x1f, 0xf9, 0x0d, 0x6e, 0x1e, 0x0e, 0xfc, 0xfe, 0x1d, 0x00, 0xfe,
+	0xcd, 0xfe, 0xf0, 0x0f,
+	0x09, 0xed, 0x0c, 0x2e, 0x4c, 0x0c, 0xef, 0xfc, 0x0c, 0xed, 0xdb, 0x2e,
+	0x6d, 0x2d, 0xfe, 0x0e,
+	0x2f, 0x3e, 0x1f, 0x00, 0xeb, 0x8c, 0xde, 0x0e, 0xad, 0xeb, 0xfd, 0xfe,
+	0x2c, 0x3d, 0x5c, 0x5e,
+	0x4f, 0x20, 0x00, 0x00, 0xfc, 0x1e, 0x2e, 0x0d, 0xcc, 0xde, 0xfd, 0xdf,
+	0xfe, 0xcb, 0x1f, 0x5e,
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+	0xed, 0xfe, 0xee, 0xc1,
+	0xf0, 0x20, 0x39, 0x00, 0x10, 0x50, 0x0e, 0x2a, 0x0f, 0x00, 0x0f, 0xff,
+	0x52, 0x71, 0x3f, 0x0d,
+	0xd0, 0x13, 0x37, 0x41, 0x12, 0x03, 0xd0, 0xc2, 0x02, 0x3f, 0x51, 0x17,
+	0x03, 0x12, 0x30, 0xe5,
+	0xa3, 0xf3, 0xe1, 0xc2, 0xd5, 0x04, 0xf4, 0xf2, 0xf1, 0x20, 0x74, 0x41,
+	0x00, 0x00, 0x10, 0x10,
+	0x20, 0xd0, 0xbe, 0xff, 0x01, 0x03, 0xf0, 0xeb, 0x20, 0x52, 0x26, 0x02,
+	0xff, 0xfc, 0xde, 0xe0,
+	0xde, 0x98, 0xfd, 0xef, 0xed, 0x0d, 0x1e, 0x00, 0xee, 0xed, 0x5e, 0x62,
+	0x03, 0xe1, 0xda, 0xde,
+	0x0e, 0x1f, 0xfd, 0xc8, 0x2e, 0x7d, 0x20, 0x1f, 0xf0, 0x22, 0x3e, 0x0a,
+	0x0e, 0xdd, 0xf0, 0x00,
+	0xe0, 0xcd, 0x0e, 0xd1, 0xe0, 0xfe, 0x28, 0x70, 0x72, 0x30, 0x00, 0x20,
+	0x40, 0x0d, 0xfd, 0xe0,
+	0x01, 0xe2, 0xfe, 0x0b, 0xa0, 0xdf, 0xd2, 0xe7, 0x05, 0x00, 0xe1, 0xcf,
+	0xec, 0x00, 0x12, 0x33,
+	0x51, 0x30, 0x54, 0x13, 0x10, 0x30, 0x10, 0x13, 0x01, 0x66, 0x14, 0x02,
+	0xd3, 0xe2, 0x11, 0x42,
+	0x02, 0x23, 0x47, 0x34, 0x11, 0x03, 0xd3, 0xf3, 0xb4, 0xa3, 0xf4, 0xd4,
+	0xc2, 0xc4, 0xe3, 0x04,
+	0x03, 0x02, 0x42, 0x64, 0x22, 0x11, 0x0f, 0x01, 0xe2, 0xf0, 0x22, 0x02,
+	0xb1, 0xff, 0x01, 0xf6,
+	0x05, 0x03, 0x23, 0x03, 0xa0, 0x00, 0x13, 0x01, 0xc0, 0xe8, 0x00, 0x2f,
+	0x10, 0xd1, 0x00, 0x50,
+	0x6e, 0x1f, 0x00, 0xc7, 0x04, 0x10, 0x10, 0xd0, 0xbf, 0xdc, 0x00, 0x0f,
+	0x00, 0x23, 0x10, 0x00,
+	0x00, 0x5f, 0x61, 0x24, 0x03, 0x00, 0x1e, 0x10, 0x02, 0xd0, 0xd1, 0x07,
+	0x06, 0x02, 0x33, 0x00,
+	0x9e, 0x0d, 0x02, 0x25, 0x40, 0x14, 0x32, 0x10, 0xff, 0xce, 0xed, 0xee,
+	0xf1, 0xd4, 0x04, 0x03,
+	0x83, 0xbf, 0xfe, 0xea, 0x0e, 0xfe, 0x0d, 0x00, 0x15, 0x32, 0x5d, 0x0d,
+	0x1e, 0x4e, 0x21, 0x17,
+	0x03, 0xd0, 0xcf, 0x15, 0x23, 0xe1, 0xf1, 0x23, 0x72, 0x50, 0x26, 0x23,
+	0x31, 0x3f, 0x0e, 0x03,
+	0x25, 0x00, 0xb0, 0xd5, 0xc4, 0xe2, 0xa4, 0xe2, 0x03, 0x05, 0x42, 0x10,
+	0x12, 0xf1, 0xdf, 0x0d,
+	0x2e, 0x40, 0xfe, 0xe0, 0x0d, 0xd8, 0x0c, 0x30, 0x43, 0x03, 0xd0, 0x3f,
+	0x6e, 0xde, 0xa0, 0xe0,
+	0x0f, 0x1d, 0x0a, 0x0d, 0xcc, 0x0b, 0x2e, 0x4d, 0xff, 0xb0, 0xef, 0xf0,
+	0x13, 0x0d, 0xdc, 0xef,
+	0xe0, 0xee, 0x98, 0x0f, 0x30, 0x2d, 0x0e, 0xdc, 0x30, 0x74, 0x0c, 0xfd,
+	0x51, 0x20, 0xed, 0x0d,
+	0x01, 0x1e, 0x0b, 0xe1, 0x36, 0xe0, 0xcd, 0x32, 0x01, 0x9c, 0x10, 0x53,
+	0x16, 0x0f, 0xdc, 0x71,
+	0x00, 0x0e, 0x32, 0xf1, 0xef, 0x03, 0x04, 0xdf, 0xfd, 0xe1, 0xe3, 0xef,
+	0xf0, 0xe0, 0xf4, 0x25,
+	0x10, 0xcd, 0xf1, 0x66, 0x31, 0x21, 0xff, 0x00, 0xd3, 0xe0, 0xcf, 0xde,
+	0x01, 0x24, 0x40, 0x72,
+	0x37, 0x02, 0x0f, 0x40, 0x5e, 0x41, 0x24, 0x1f, 0xca, 0xff, 0xff, 0xdf,
+	0xee, 0xf6, 0xa1, 0xd0,
+	0x00, 0x10, 0x02, 0x03, 0xe0, 0x32, 0x70, 0x0e, 0x1e, 0x0c, 0x09, 0x0e,
+	0x30, 0x00, 0xe0, 0xbe,
+	0x0d, 0x0d, 0x0c, 0xff, 0xe0, 0xd2, 0xaf, 0xe8, 0x0f, 0x3f, 0x1d, 0xfc,
+	0x1e, 0x4f, 0xfc, 0x0d,
+	0x1d, 0x2e, 0x50, 0x1c, 0x1d, 0xe0, 0xb2, 0xee, 0x0b, 0x10, 0x12, 0xad,
+	0xec, 0x00, 0x20, 0xda,
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+	0x00, 0xef, 0xef, 0xf0,
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+	0xe0, 0xe0, 0xe2, 0xc1,
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+	0x2d, 0x40, 0x20, 0x20,
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+	0xbe, 0xee, 0xed, 0x10,
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+	0xee, 0xde, 0xf1, 0xe0,
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+	0x50, 0x0a, 0x1c, 0x2f,
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+	0xdd, 0x03, 0x23, 0x2f,
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+	0x24, 0x22, 0x22, 0x36,
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+	0x43, 0x57, 0x11, 0x12,
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+	0x11, 0x01, 0x00, 0x13,
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+	0xd1, 0xb1, 0x04, 0x34,
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+	0xb1, 0xe2, 0x04, 0x03,
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+	0x02, 0xec, 0xbb, 0x00,
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+	0xf0, 0xe0, 0x8d, 0xe3,
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+	0x0d, 0x20, 0x2f, 0x08,
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+	0x03, 0x04, 0x02, 0x12,
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+	0x05, 0xf2, 0xb0, 0xf2,
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+	0xfd, 0x0f, 0x01, 0x9e,
+	0xdc, 0xe0, 0xd2, 0xe3, 0xc2, 0xd4, 0xe7, 0xf3, 0xe0, 0x00, 0x00, 0xed,
+	0x1b, 0x60, 0x3f, 0x2e,
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+	0xfe, 0x00, 0x00, 0x0f,
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+	0x3d, 0x50, 0x1b, 0xd9,
+	0x40, 0x20, 0x8a, 0xff, 0x00, 0x00, 0xdf, 0xee, 0xe0, 0xe1, 0xd1, 0xc0,
+	0xb0, 0xf0, 0x02, 0xdf,
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+	0x11, 0x10, 0x00, 0x23,
+	0x25, 0x00, 0x00, 0x24, 0x01, 0x01, 0xe0, 0xf1, 0x45, 0x11, 0x00, 0x20,
+	0x74, 0x11, 0x00, 0x01,
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+	0x20, 0x00, 0xdb, 0x00,
+	0x2f, 0x1f, 0x1f, 0xcd, 0xbb, 0x10, 0x20, 0x00, 0xed, 0xef, 0xdf, 0xbe,
+	0xef, 0xf0, 0xcf, 0xb9,
+	0x21, 0x43, 0xed, 0xfd, 0x1f, 0x3d, 0x1a, 0x0d, 0x40, 0x20, 0xdc, 0xac,
+	0xe0, 0x01, 0xf0, 0xbc,
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+	0xff, 0x00, 0x00, 0x42,
+	0x22, 0xf0, 0x00, 0x70, 0x6f, 0x0e, 0x1d, 0x08, 0x0e, 0x2e, 0x2c, 0x0c,
+	0xeb, 0x0f, 0xfe, 0x9d,
+	0xff, 0xe0, 0xbf, 0xe0, 0xd0, 0xe1, 0xe5, 0xe3, 0x9f, 0xe0, 0xff, 0xf0,
+	0x02, 0xd8, 0xde, 0x20,
+	0x2f, 0x0d, 0x51, 0x22, 0x20, 0x22, 0x22, 0x43, 0x36, 0x32, 0x35, 0x8c,
+	0xee, 0x77, 0x02, 0xed,
+	0x01, 0x22, 0x00, 0x00, 0x00, 0x33, 0x11, 0x00, 0x23, 0x23, 0x23, 0x14,
+	0x02, 0xe0, 0xf0, 0x27,
+	0x01, 0xf1, 0xb0, 0xee, 0x52, 0x32, 0x0e, 0x20, 0x40, 0x0e, 0x62, 0x21,
+	0x20, 0x20, 0xf8, 0xff,
+	0x0f, 0x0f, 0xee, 0xcd, 0xde, 0xab, 0xdd, 0xef, 0x00, 0x00, 0xb8, 0x1e,
+	0x40, 0x0e, 0x1e, 0x2f,
+	0x2f, 0x1e, 0x0c, 0xff, 0x10, 0xee, 0x88, 0xf0, 0x01, 0xdf, 0xef, 0xef,
+	0xdd, 0x0e, 0x20, 0x20,
+	0x0c, 0xfd, 0x0e, 0x0e, 0x1f, 0x0d, 0xdb, 0xfe, 0x2f, 0x3f, 0x70, 0x30,
+	0x0b, 0x2f, 0x70, 0x1e,
+	0x1a, 0x40, 0x3e, 0x2e, 0x3d, 0x4d, 0x1c, 0x0c, 0xcd, 0xff, 0x02, 0xf3,
+	0xc1, 0xb1, 0xf7, 0xe3,
+	0xa1, 0xe1, 0xf3, 0xc0, 0xe0, 0xf0, 0x00, 0x12, 0x0e, 0x0e, 0x31, 0x2e,
+	0x2e, 0x77, 0x46, 0x21,
+	0x11, 0x11, 0x44, 0x34, 0x11, 0x02, 0x13, 0x22, 0x25, 0x02, 0x01, 0x23,
+	0x12, 0x03, 0x54, 0x44,
+	0x23, 0x32, 0x22, 0x35, 0x53, 0x03, 0x23, 0x24, 0x01, 0xd2, 0xf3, 0x05,
+	0xd0, 0x81, 0x02, 0xf0,
+	0xee, 0x13, 0x11, 0xee, 0x44, 0x34, 0x00, 0x1f, 0x41, 0x45, 0xf0, 0xd0,
+	0xf3, 0x27, 0xd1, 0x80,
+	0xf2, 0x03, 0x13, 0x00, 0xf0, 0x33, 0x54, 0x11, 0x45, 0x22, 0x00, 0xef,
+	0xff, 0x00, 0xf3, 0xf4,
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+	0x11, 0x23, 0xf0, 0xe0,
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+	0x6f, 0x0c, 0x0b, 0xfd,
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+	0x00, 0x31, 0xee, 0x00,
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+	0x12, 0xee, 0xcd, 0xff,
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+	0x00, 0x02, 0x00, 0xbe,
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+	0x00, 0x44, 0x12, 0x00,
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+	0x02, 0x00, 0x0c, 0x0e,
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+	0x43, 0x00, 0xfe, 0x00,
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+	0x36, 0x12, 0xae, 0xef,
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+	0x67, 0x00, 0xbb, 0xbc,
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+	0xbb, 0xf0, 0x03, 0xdd,
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+	0x0f, 0x0f, 0x0f, 0x00,
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+	0x10, 0x63, 0x10, 0x0c,
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+	0xfe, 0x0e, 0x00, 0xd8,
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+	0x2e, 0x77, 0x11, 0x0f,
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+	0x21, 0x00, 0x11, 0x32,
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+	0xc0, 0x33, 0x76, 0x10,
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+	0x12, 0x00, 0x00, 0x00,
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+	0x01, 0x57, 0x12, 0xde,
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+	0x47, 0x23, 0x0f, 0xf0,
+	0xcf, 0xcd, 0x13, 0x67, 0x00, 0xb8, 0x0f, 0x20, 0x0f, 0xfd, 0x30, 0x30,
+	0x0d, 0x0f, 0x10, 0x54,
+	0x21, 0x0f, 0xec, 0x00, 0x53, 0x0f, 0x0f, 0x22, 0x10, 0x1f, 0x22, 0xf0,
+	0x88, 0xee, 0x13, 0x00,
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+	0x00, 0xdf, 0x12, 0xd0,
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+	0x01, 0x00, 0x15, 0x74,
+	0x24, 0x01, 0xd0, 0xf0, 0x03, 0x01, 0x13, 0x22, 0x8c, 0xfd, 0x11, 0x33,
+	0x20, 0x10, 0x0d, 0xfa,
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+	0xed, 0xfe, 0xff, 0xdc,
+	0xdd, 0x00, 0x10, 0x20, 0xfe, 0xdd, 0x0f, 0x00, 0x45, 0xed, 0x88, 0xef,
+	0x32, 0x10, 0xff, 0xee,
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+	0xab, 0xf0, 0x23, 0x0f,
+	0x0d, 0x0f, 0xfa, 0xff, 0x20, 0x2f, 0x0e, 0xee, 0xed, 0xee, 0x10, 0x42,
+	0xff, 0x9b, 0xee, 0xed,
+	0xf0, 0xe0, 0xcf, 0x01, 0x00, 0xdd, 0xcb, 0x0f, 0xed, 0xde, 0x02, 0x55,
+	0x00, 0xca, 0xbd, 0xf0,
+	0x01, 0x00, 0x11, 0x03, 0xd1, 0xf1, 0xe0, 0xe0, 0x02, 0x57, 0x45, 0x22,
+	0xf0, 0xd0, 0x23, 0x12,
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+	0x11, 0x00, 0x21, 0x21,
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+	0x35, 0xe0, 0xf0, 0x00,
+	0x21, 0x62, 0x12, 0x00, 0x0f, 0xff, 0xff, 0xfe, 0x22, 0x45, 0x00, 0xee,
+	0xbb, 0xef, 0x21, 0x20,
+	0x41, 0x11, 0xcb, 0xdd, 0x0f, 0x00, 0x00, 0x00, 0xfc, 0xed, 0x73, 0x20,
+	0x0b, 0x0f, 0x2e, 0x50,
+	0x45, 0x00, 0x0e, 0xff, 0x22, 0x34, 0x23, 0x00, 0xcf, 0xbc, 0x0f, 0x11,
+	0x14, 0x01, 0x00, 0x20,
+	0x44, 0x64, 0x35, 0x20, 0xff, 0xed, 0xf0, 0xff, 0x13, 0x77, 0xf1, 0xbb,
+	0xff, 0xf0, 0xf0, 0xcc,
+	0xee, 0x12, 0x12, 0x0f, 0xdc, 0xee, 0xef, 0x02, 0x35, 0xf0, 0x9d, 0xff,
+	0x02, 0xe0, 0xe0, 0x05,
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+	0x55, 0x01, 0xff, 0x22,
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+	0x1f, 0x9d, 0xfd, 0x21,
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+	0xd9, 0x03, 0x26, 0xe0,
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+	0x32, 0x22, 0x22, 0x22,
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+	0xd2, 0xa0, 0xe2, 0xe2,
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+	0x21, 0xff, 0xee, 0x00,
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+	0xde, 0xd0, 0xe0, 0xf1,
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+	0x0f, 0x0f, 0x0e, 0xfd,
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+	0x3e, 0x1f, 0x22, 0x0f,
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+	0x44, 0x33, 0x00, 0x1f,
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+	0xd0, 0x47, 0x33, 0xdf,
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+	0xd1, 0xf2, 0xf0, 0xef,
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+	0xef, 0x01, 0x03, 0x20,
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+	0x0b, 0x01, 0xf9, 0x00,
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+	0xcd, 0x34, 0x33, 0xdb,
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+	0x11, 0xee, 0xed, 0x57,
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+	0xd0, 0xf0, 0xdf, 0xef,
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+	0xd0, 0x80, 0xc2, 0x06,
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+	0xde, 0xcb, 0x21, 0x22,
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+	0xe0, 0xe0, 0xef, 0xee,
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+	0x20, 0xfd, 0x88, 0x00,
+	0x04, 0x01, 0xff, 0xdf, 0x11, 0x00, 0xdc, 0xd0, 0x66, 0x00, 0x0e, 0xef,
+	0xdf, 0x53, 0x24, 0xff,
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+	0xf0, 0xfe, 0xdc, 0x02,
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+	0x04, 0xf0, 0xed, 0x41,
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+	0xe0, 0xcc, 0x02, 0x21,
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+	0x0f, 0x0e, 0xe0, 0x00,
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+	0xfd, 0xfe, 0xf0, 0x25,
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+	0x0d, 0x74, 0x00, 0xc9,
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+	0xed, 0x30, 0x76, 0x0d,
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+	0x22, 0x65, 0x0e, 0x0f,
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+	0x0e, 0x00, 0x53, 0xe0,
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+	0xde, 0x62, 0x36, 0x00,
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+	0xde, 0x9e, 0x31, 0x41,
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+	0x46, 0xee, 0xdf, 0xd0,
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+	0x32, 0x74, 0x00, 0x0c,
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+	0x02, 0x25, 0x10, 0xdc,
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+	0xef, 0xed, 0xed, 0x50,
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+	0x03, 0x14, 0x0e, 0xee,
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+	0x9a, 0x00, 0x00, 0x01,
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+	0xfd, 0xc8, 0x00, 0x14,
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+	0x06, 0x00, 0x90, 0xfe,
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+	0x0f, 0x00, 0xe0, 0xe1,
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+	0xbd, 0x1f, 0x31, 0xe0,
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+	0xce, 0xf0, 0xf6, 0x01,
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+	0x20, 0x01, 0x0f, 0xef,
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+	0x02, 0x55, 0x0f, 0x00,
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+	0xee, 0xbb, 0x10, 0x22,
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+	0x34, 0x0f, 0x00, 0x73,
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+	0x30, 0x0d, 0x00, 0x12,
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+	0x0f, 0x00, 0x45, 0x21,
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+	0x20, 0x30, 0x22, 0x10,
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+	0x43, 0xe2, 0xbe, 0x0c,
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+	0x12, 0x00, 0x5f, 0x43,
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+	0xed, 0xbe, 0x01, 0x01,
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+	0x10, 0x0f, 0xce, 0x10,
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+	0xed, 0x73, 0x67, 0xd0,
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+	0x03, 0xf0, 0xc8, 0xff,
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+	0x12, 0xff, 0xc8, 0xef,
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+	0x20, 0x40, 0x00, 0x26,
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+	0xfe, 0xee, 0x02, 0x17,
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+	0x02, 0x20, 0x1f, 0xed,
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+	0xef, 0x0f, 0x1d, 0xf0,
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+	0x0f, 0x00, 0x02, 0xc2,
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+	0x00, 0x1e, 0x0f, 0xbb,
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+	0xdf, 0x1e, 0x50, 0x21,
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+	0x37, 0x12, 0xcc, 0xd0,
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+	0x20, 0x00, 0x12, 0x45,
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+	0x42, 0x14, 0xe0, 0x2f,
+	0x62, 0x13, 0xee, 0xde, 0x02, 0x26, 0x2f, 0xbd, 0xf0, 0xf0, 0xe0, 0x00,
+	0x0f, 0xea, 0x30, 0x42,
+	0xaf, 0x00, 0x12, 0x2e, 0x9c, 0x0f, 0x1f, 0xef, 0xd1, 0x41, 0x11, 0xf6,
+	0xd0, 0x25, 0x41, 0xed,
+	0xef, 0xe0, 0xd1, 0x73, 0x21, 0xdf, 0x25, 0x64, 0x0d, 0xfc, 0x03, 0x12,
+	0x1e, 0x00, 0xef, 0x0b,
+	0x01, 0x04, 0x9c, 0xfb, 0x00, 0xf3, 0xe0, 0x01, 0xfd, 0x4f, 0xe0, 0xd0,
+	0x10, 0x10, 0xec, 0x02,
+	0x24, 0xc0, 0xc0, 0x44, 0x7f, 0x2c, 0x03, 0x13, 0x00, 0x3c, 0xdf, 0xf0,
+	0x14, 0x11, 0x20, 0xcd,
+	0x9b, 0xfc, 0xe0, 0x01, 0x24, 0x31, 0x40, 0xfc, 0xdf, 0x54, 0x34, 0x0f,
+	0x0c, 0xc0, 0xe0, 0x23,
+	0x54, 0x00, 0xfc, 0xcd, 0xe0, 0x03, 0x55, 0x0c, 0x00, 0x00, 0x0e, 0xdb,
+	0x13, 0x57, 0x3f, 0xfe,
+	0xe0, 0x13, 0x31, 0x01, 0xbf, 0x1f, 0x21, 0x12, 0xdf, 0xb8, 0xf0, 0x01,
+	0x31, 0x1e, 0xde, 0x33,
+	0x25, 0xff, 0xdc, 0x40, 0x72, 0x01, 0xd1, 0x00, 0xe2, 0x30, 0x1f, 0xe2,
+	0x06, 0x40, 0xcc, 0xb0,
+	0x00, 0x20, 0x16, 0x00, 0x00, 0x30, 0xef, 0xe0, 0xf1, 0xbd, 0xe0, 0xf2,
+	0x14, 0x73, 0x0e, 0x88,
+	0x0f, 0x03, 0x11, 0xfd, 0xdf, 0x2e, 0x2d, 0x00, 0x01, 0x15, 0x2e, 0x1e,
+	0x00, 0xf6, 0x30, 0x00,
+	0x10, 0x70, 0x21, 0x02, 0xe0, 0xfe, 0x1e, 0x10, 0xae, 0xbb, 0xf1, 0xf2,
+	0xfb, 0x10, 0x20, 0xdf,
+	0x0d, 0x7e, 0x1c, 0x0f, 0x00, 0x01, 0x47, 0x02, 0xe0, 0x0d, 0x20, 0xf1,
+	0xcd, 0xdf, 0xf0, 0x02,
+	0xf8, 0xef, 0x62, 0x00, 0xcd, 0x20, 0x46, 0x11, 0xdf, 0x00, 0x24, 0x10,
+	0x50, 0x22, 0x00, 0xb0,
+	0x02, 0x00, 0xdc, 0xcd, 0x1e, 0x1b, 0xf0, 0xa1, 0x04, 0x00, 0x1b, 0x11,
+	0x42, 0xed, 0xec, 0x02,
+	0x46, 0x30, 0x10, 0xb0, 0xfe, 0x00, 0x30, 0xfe, 0xf0, 0x56, 0x10, 0xfd,
+	0xee, 0x04, 0x15, 0x20,
+	0x50, 0x0e, 0xe0, 0x02, 0x03, 0x00, 0xe0, 0xd0, 0x8d, 0x0d, 0x01, 0x04,
+	0xf3, 0xff, 0x0a, 0xce,
+	0x0e, 0x31, 0x41, 0xdb, 0xdd, 0x43, 0x34, 0xee, 0xe0, 0x45, 0x00, 0xff,
+	0x30, 0x7f, 0x2f, 0x31,
+	0x25, 0x13, 0xaf, 0xfe, 0x27, 0x40, 0xde, 0xdd, 0x10, 0x22, 0x0f, 0xba,
+	0xe0, 0x11, 0x45, 0x00,
+	0x0e, 0x61, 0x11, 0x00, 0x0e, 0xb0, 0xbe, 0x3e, 0x2e, 0xa0, 0x13, 0x15,
+	0x0b, 0xfe, 0xf2, 0x03,
+	0x1f, 0x0c, 0x03, 0x20, 0x0c, 0xf0, 0x04, 0x22, 0x1c, 0x0d, 0xff, 0xcf,
+	0x00, 0x24, 0x00, 0xc9,
+	0x00, 0xc0, 0xd1, 0xdf, 0xfd, 0x2e, 0x70, 0xf0, 0xd5, 0xf0, 0x11, 0x40,
+	0xf1, 0x03, 0x27, 0x33,
+	0x0f, 0x0c, 0x41, 0x36, 0xa0, 0xdf, 0xef, 0x31, 0x31, 0x21, 0x20, 0x0f,
+	0xef, 0x3e, 0x71, 0x67,
+	0x00, 0xee, 0xef, 0x12, 0x22, 0x11, 0x14, 0xf2, 0xb0, 0xcd, 0xf3, 0x26,
+	0x30, 0x1d, 0x0f, 0x6e,
+	0x0e, 0xdf, 0x35, 0x21, 0x20, 0x61, 0x10, 0x0c, 0x10, 0x07, 0xf1, 0xff,
+	0x1f, 0x30, 0xcf, 0x8d,
+	0xf1, 0xf3, 0xe0, 0xfd, 0x0d, 0xea, 0xf0, 0x01, 0xf1, 0x30, 0x3f, 0x00,
+	0x04, 0x24, 0x3c, 0x9a,
+	0xff, 0x60, 0x22, 0x00, 0xdf, 0x0c, 0x10, 0x2f, 0xfe, 0xbc, 0x10, 0x16,
+	0xff, 0x0f, 0x50, 0xf1,
+	0xcf, 0xff, 0x00, 0x77, 0x22, 0xee, 0x00, 0xf0, 0xe0, 0x00, 0x03, 0xd9,
+	0xee, 0x10, 0x13, 0xff,
+	0xee, 0xe1, 0x02, 0x0d, 0x5b, 0x40, 0xb1, 0xc0, 0x0f, 0x11, 0xff, 0xfc,
+	0x32, 0x14, 0x02, 0xe2,
+	0xd0, 0xb0, 0x01, 0x21, 0xef, 0x02, 0x77, 0x22, 0x2c, 0x10, 0x34, 0x40,
+	0xff, 0xcb, 0xd0, 0x40,
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+	0x76, 0x33, 0xff, 0x1b,
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+	0x73, 0x00, 0xf1, 0x04,
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+	0x1d, 0x88, 0xed, 0x02,
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+	0x42, 0x00, 0x00, 0x3e,
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+	0x02, 0xd0, 0xd0, 0xdf,
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+	0x10, 0xee, 0x9c, 0x0f,
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+	0x0f, 0xe9, 0x20, 0x42,
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+	0x0f, 0xdc, 0xc0, 0xe2,
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+	0x0e, 0x1c, 0x03, 0xf5,
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+	0x7f, 0x51, 0x01, 0xf0,
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+	0xe1, 0xd0, 0x3b, 0x3d,
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+	0xe0, 0xf1, 0xd2, 0xe0,
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+	0xe9, 0x20, 0x63, 0x30,
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+	0x8f, 0xdc, 0xf0, 0xd0,
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+	0x00, 0xd1, 0x11, 0x13,
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+	0x30, 0x42, 0x15, 0x22,
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+	0xf0, 0x1e, 0x2d, 0x30,
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+	0x53, 0x00, 0xfa, 0x0e,
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+	0xd0, 0xe7, 0xc4, 0xde,
+	0xef, 0xd0, 0xdf, 0x30, 0x74, 0x76, 0x20, 0x0e, 0x0f, 0x11, 0x0f, 0xff,
+	0xfe, 0x02, 0xe1, 0xa9,
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+	0x04, 0xe2, 0xef, 0xfc,
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+	0x01, 0x00, 0x02, 0x54,
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+	0x10, 0x41, 0x41, 0x22,
+	0xfd, 0xe9, 0x30, 0x50, 0x1f, 0xcf, 0xcd, 0xee, 0x0f, 0xeb, 0xf0, 0xe0,
+	0x0e, 0x1e, 0x10, 0x73,
+	0x77, 0x00, 0x00, 0xf1, 0x14, 0x21, 0x0f, 0x00, 0x15, 0x10, 0x01, 0x01,
+	0xf0, 0xfc, 0xee, 0x11,
+	0x74, 0x31, 0x30, 0x0e, 0x21, 0x54, 0x35, 0xe0, 0xd9, 0x10, 0x40, 0xc1,
+	0xc1, 0xcd, 0xdf, 0xe1,
+	0xf1, 0xf0, 0x0e, 0x20, 0x40, 0x71, 0x31, 0x44, 0x14, 0x02, 0x05, 0xf2,
+	0x16, 0x22, 0xf3, 0x8e,
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+	0x0f, 0x00, 0x3d, 0x3f,
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+	0xbf, 0x0d, 0x0a, 0xfc,
+	0xcd, 0x20, 0x50, 0x00, 0xff, 0xcf, 0xf0, 0xf2, 0xe6, 0xe3, 0xc4, 0x90,
+	0xe0, 0xd1, 0xf2, 0xf1,
+	0xf0, 0x0c, 0x3f, 0x6c, 0x2c, 0x0d, 0x10, 0x50, 0x0c, 0x30, 0x23, 0x13,
+	0x2f, 0x1d, 0x17, 0x21,
+	0xef, 0xed, 0xe0, 0x80, 0xd8, 0x0f, 0x11, 0x42, 0xfd, 0xba, 0xff, 0x0f,
+	0x1f, 0x00, 0x00, 0x34,
+	0x11, 0xff, 0x02, 0x77, 0x10, 0xdd, 0xe0, 0x15, 0x00, 0xc0, 0xf2, 0x11,
+	0x02, 0x8f, 0x0f, 0x20,
+	0x10, 0x1e, 0x20, 0x43, 0x14, 0x0e, 0x1c, 0x20, 0x63, 0x21, 0x0d, 0x3d,
+	0x25, 0x00, 0x9a, 0x00,
+	0xe0, 0x00, 0xee, 0xbd, 0xf0, 0x05, 0xf0, 0xcc, 0x00, 0x0f, 0x0e, 0x0e,
+	0x50, 0x20, 0x1c, 0xed,
+	0x02, 0x77, 0x05, 0xc0, 0x00, 0xe0, 0xf1, 0xe4, 0x12, 0x02, 0xce, 0xee,
+	0xd0, 0x20, 0x76, 0x21,
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+	0xc0, 0xac, 0xff, 0xed,
+	0x10, 0x32, 0x02, 0x2c, 0x0b, 0x10, 0x50, 0x35, 0x00, 0xc0, 0x00, 0xf0,
+	0xe0, 0x21, 0x00, 0x9e,
+	0xea, 0x00, 0x1f, 0x51, 0x0f, 0xdb, 0x0f, 0x5f, 0x20, 0x30, 0x0e, 0x33,
+	0x00, 0x98, 0xff, 0x01,
+	0xe2, 0xc0, 0xff, 0xe1, 0xe0, 0xcd, 0xec, 0x01, 0x05, 0x2f, 0x1d, 0x10,
+	0x03, 0x40, 0x31, 0x64,
+	0x12, 0x32, 0x27, 0x14, 0x01, 0x11, 0x01, 0x01, 0xf2, 0x9f, 0xfb, 0xef,
+	0xfe, 0xe0, 0x33, 0x32,
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+	0x11, 0xd0, 0x8b, 0xf0,
+	0x03, 0x1f, 0x0d, 0x00, 0x64, 0x00, 0xde, 0xee, 0xdf, 0x0f, 0x0e, 0x33,
+	0x37, 0xe0, 0xed, 0x52,
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+	0xf2, 0x0e, 0xe9, 0x02,
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+	0x12, 0x1f, 0xfe, 0x0f,
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+	0x0e, 0xdb, 0x71, 0x00,
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+	0xfc, 0xe0, 0x31, 0x21,
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+	0x00, 0x2f, 0x32, 0x51,
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+	0xf0, 0xfe, 0xfd, 0xff,
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+	0x2f, 0x10, 0x01, 0xbd,
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+	0x33, 0xef, 0xbd, 0x20,
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+	0xcc, 0xfe, 0x32, 0x12,
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+	0x00, 0xe3, 0xd1, 0x0b,
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+	0xea, 0x10, 0x21, 0xf0,
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+	0x20, 0x30, 0xdf, 0xb0,
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+	0x00, 0xf0, 0x0d, 0xba,
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+	0x21, 0x01, 0x1f, 0xeb,
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+	0x55, 0x0f, 0x0b, 0x20,
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+	0x00, 0x12, 0x00, 0x10,
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+	0x1e, 0xfb, 0xe0, 0xe0,
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+	0x05, 0xed, 0x2a, 0x10,
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+	0x74, 0x32, 0xbd, 0xbc,
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+	0x1f, 0x13, 0xd0, 0xdf,
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+	0x12, 0x33, 0xf1, 0xfc,
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+	0x0d, 0x14, 0x27, 0x22,
+	0x0c, 0xff, 0xf0, 0xe1, 0x45, 0x20, 0xff, 0x00, 0x00, 0xbe, 0x0d, 0x03,
+	0x15, 0x0f, 0xdb, 0x40,
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+	0xae, 0xf3, 0x05, 0x0e,
+	0x10, 0x64, 0x0f, 0xdd, 0x01, 0x10, 0x12, 0x07, 0x03, 0xf0, 0x3d, 0x20,
+	0xb0, 0xe4, 0x0e, 0xfc,
+	0x31, 0x3f, 0x0d, 0xa5, 0x05, 0x0f, 0x1a, 0x20, 0xe2, 0xc1, 0x10, 0x1e,
+	0xef, 0xef, 0x00, 0x0d,
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+	0x00, 0x22, 0x25, 0xdf,
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+	0x32, 0x10, 0xdc, 0xdd,
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+	0x16, 0xd2, 0xed, 0x1f,
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+	0x20, 0x10, 0x02, 0xdf,
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+	0x20, 0xff, 0xbc, 0x01,
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+	0xf1, 0xf0, 0x23, 0x51,
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+	0xf0, 0xff, 0x12, 0x27,
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+	0xef, 0xe0, 0xa3, 0xdc,
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+	0x94, 0xee, 0xfc, 0x20,
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+	0x47, 0x51, 0x0f, 0xfe,
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+	0x0e, 0x1f, 0x82, 0xc2,
+	0x0f, 0x0f, 0x30, 0xef, 0xfc, 0x2f, 0x1c, 0x2e, 0x70, 0x07, 0x04, 0xe0,
+	0x3f, 0x32, 0x02, 0xf1,
+	0xe2, 0x0f, 0xfd, 0x90, 0x07, 0x22, 0x3d, 0x1b, 0xe0, 0x92, 0xf3, 0x0f,
+	0x5d, 0x41, 0x02, 0xae,
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+	0x01, 0x13, 0xef, 0xbd,
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+	0x20, 0xfc, 0xcf, 0x00,
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+	0x20, 0x42, 0x53, 0x0a,
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+	0xf1, 0xb0, 0x02, 0x03,
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+	0x33, 0xce, 0x0e, 0x05,
+	0x00, 0x00, 0x01, 0x04, 0x0e, 0xab, 0xef, 0x00, 0x53, 0x24, 0x14, 0xf2,
+	0xbd, 0x0b, 0xff, 0x30,
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+	0xec, 0xdf, 0x31, 0x53,
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+	0x36, 0xf1, 0xb0, 0xfc,
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+	0xe0, 0xc7, 0xf1, 0x1f,
+	0x5f, 0x40, 0x2f, 0xe0, 0xbf, 0x02, 0x40, 0x3e, 0xfd, 0xcc, 0xff, 0x00,
+	0x12, 0x30, 0xee, 0xc2,
+	0x04, 0x20, 0xfc, 0x01, 0x33, 0x35, 0xd3, 0xfe, 0x1c, 0xfe, 0x0f, 0x76,
+	0x01, 0x0d, 0xfd, 0xb4,
+	0xd3, 0xff, 0xfb, 0x2f, 0xe0, 0x92, 0xf4, 0xc0, 0xff, 0x2b, 0x50, 0x10,
+	0xd1, 0xc0, 0x01, 0x41,
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+	0x15, 0xf5, 0xc0, 0x0c,
+	0x30, 0x0f, 0xc1, 0x02, 0x00, 0xff, 0xff, 0x11, 0x05, 0xbc, 0xcd, 0x3f,
+	0x55, 0x00, 0xaf, 0x0d,
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+	0x22, 0x0f, 0xfd, 0x0f,
+	0x0e, 0x2f, 0x30, 0x12, 0x04, 0xb3, 0xe1, 0x28, 0x7c, 0x01, 0x05, 0xe1,
+	0xd0, 0x00, 0xe0, 0x0d,
+	0x0f, 0x02, 0xd5, 0xc0, 0xff, 0x0c, 0x5f, 0x2f, 0x00, 0x02, 0x24, 0x0f,
+	0x2e, 0x70, 0x21, 0xf3,
+	0xd0, 0xc0, 0x02, 0x04, 0x4d, 0x0d, 0xcf, 0xf5, 0xf5, 0x30, 0x7c, 0x1d,
+	0xed, 0xff, 0xff, 0xef,
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+	0x11, 0x00, 0xfe, 0x00,
+	0xe1, 0x04, 0x41, 0xf2, 0xc1, 0x01, 0x3c, 0xec, 0xc0, 0x14, 0x00, 0xee,
+	0xfe, 0x2f, 0x50, 0x1d,
+	0x0c, 0xed, 0xd4, 0x47, 0x00, 0x1e, 0x2f, 0x40, 0xf2, 0x82, 0xfd, 0x0c,
+	0x6e, 0x00, 0x07, 0xe4,
+	0x0e, 0x2b, 0x31, 0x03, 0xf1, 0xbe, 0xfe, 0x01, 0x25, 0x00, 0x00, 0xcd,
+	0xe1, 0x02, 0x00, 0xfd,
+	0xdf, 0x42, 0x73, 0x09, 0xcd, 0xe0, 0x03, 0x14, 0x50, 0x1e, 0xf9, 0xd0,
+	0xf2, 0x12, 0x6e, 0x0d,
+	0x0f, 0xd2, 0xf0, 0x0d, 0x0d, 0x10, 0x2f, 0xf2, 0xb4, 0x0f, 0x1e, 0x67,
+	0x17, 0x00, 0xd0, 0x0f,
+	0x10, 0x42, 0x20, 0x0f, 0xc1, 0xe0, 0x0f, 0xff, 0xf3, 0x13, 0x4f, 0x28,
+	0xff, 0xb0, 0xe1, 0xf0,
+	0x0d, 0x4f, 0x30, 0x06, 0xc0, 0xcf, 0xeb, 0x0f, 0x41, 0x01, 0x11, 0x10,
+	0x41, 0xf5, 0xb0, 0x0f,
+	0x0d, 0x40, 0x04, 0x01, 0x40, 0x00, 0xd0, 0x02, 0x2f, 0x0e, 0x86, 0xf1,
+	0x11, 0x41, 0x11, 0x02,
+	0xe4, 0xd1, 0x1d, 0x48, 0x0d, 0xf1, 0xe2, 0x00, 0x0c, 0x02, 0x61, 0x0e,
+	0xdd, 0x00, 0x00, 0x01,
+	0xd5, 0x22, 0x5d, 0x0b, 0xcf, 0xff, 0x02, 0x03, 0x0f, 0x7c, 0x30, 0xf2,
+	0xb0, 0x0a, 0x00, 0x21,
+	0x0e, 0xed, 0xe7, 0xd4, 0xd0, 0x2d, 0x6e, 0x02, 0xd3, 0xb1, 0xd0, 0xff,
+	0x0e, 0x7f, 0x21, 0x0f,
+	0xdb, 0x01, 0x26, 0x20, 0x0b, 0xce, 0x20, 0x35, 0x01, 0xcf, 0x0d, 0x20,
+	0xf0, 0xf0, 0x33, 0x05,
+	0xde, 0x0e, 0x0e, 0xa0, 0xf1, 0x02, 0x0d, 0x0d, 0xe0, 0x0f, 0x2b, 0xb1,
+	0xd5, 0x10, 0x1d, 0x3d,
+	0x41, 0x55, 0x0f, 0xff, 0xf2, 0x14, 0x11, 0xef, 0x8b, 0xd0, 0x20, 0x26,
+	0x00, 0x10, 0x2e, 0x2e,
+	0x60, 0x03, 0x00, 0xeb, 0x1f, 0x30, 0x1d, 0x00, 0xe0, 0x9f, 0xbe, 0xe0,
+	0xf1, 0x24, 0x0f, 0xdf,
+	0xb1, 0xf7, 0x01, 0x50, 0x30, 0x10, 0xfc, 0x0e, 0x14, 0x05, 0x40, 0x6d,
+	0x02, 0xf7, 0xe0, 0x0e,
+	0xf0, 0x11, 0x15, 0x10, 0xc0, 0xb0, 0x00, 0x0f, 0x5f, 0x0e, 0xf0, 0x04,
+	0x05, 0xd0, 0xfe, 0x1b,
+	0x70, 0x51, 0x04, 0xc0, 0x0d, 0x0d, 0x10, 0x32, 0x00, 0xcf, 0x0e, 0x00,
+	0x0f, 0x1e, 0xe1, 0xd4,
+	0x02, 0x24, 0x62, 0x10, 0xfc, 0xe4, 0x03, 0x0c, 0x3d, 0x05, 0x04, 0x20,
+	0x50, 0x00, 0xdf, 0xac,
+	0x00, 0x0e, 0x20, 0x27, 0xe2, 0xbc, 0x0e, 0x00, 0x70, 0x0f, 0xee, 0xf1,
+	0xf1, 0x0f, 0x19, 0x00,
+	0x10, 0x20, 0x10, 0xf0, 0xc0, 0xcb, 0x01, 0x15, 0x51, 0x10, 0x0d, 0xef,
+	0x02, 0x01, 0xf1, 0x0f,
+	0x31, 0xd7, 0xe5, 0x3e, 0x3c, 0x0f, 0x32, 0x23, 0x82, 0x9f, 0x0b, 0x20,
+	0x40, 0xff, 0xc0, 0xf0,
+	0x00, 0x12, 0x00, 0x2f, 0x2e, 0xfc, 0xc0, 0xf4, 0x07, 0xe0, 0xee, 0x1e,
+	0xff, 0x82, 0x04, 0x21,
+	0x70, 0x1e, 0x0f, 0xd1, 0x04, 0x30, 0x0c, 0x0e, 0x40, 0x26, 0x02, 0xc1,
+	0xf0, 0xfd, 0xff, 0x00,
+	0x22, 0x02, 0xbf, 0xee, 0x0e, 0x28, 0x1f, 0x02, 0x24, 0x70, 0x0e, 0xee,
+	0xf0, 0x03, 0x01, 0xef,
+	0xf4, 0x74, 0x40, 0x0c, 0xef, 0xe1, 0x15, 0x13, 0xef, 0xeb, 0x3e, 0x50,
+	0x16, 0x00, 0xd0, 0xbd,
+	0x00, 0x00, 0x03, 0x21, 0x2c, 0x0f, 0xe0, 0x11, 0x7f, 0x7f, 0x2f, 0xe1,
+	0xe3, 0x02, 0xed, 0x0e,
+	0x31, 0x27, 0x03, 0xa0, 0x0b, 0x0f, 0x00, 0x02, 0x02, 0x2c, 0x0b, 0x0e,
+	0xd0, 0xc2, 0x05, 0x4d,
+	0x1b, 0x0f, 0xa1, 0xc3, 0x03, 0x2e, 0x0a, 0xd0, 0xb6, 0xf0, 0x0e, 0x5f,
+	0x20, 0x13, 0xf0, 0xc1,
+	0x00, 0x3e, 0x50, 0x02, 0xd0, 0xfe, 0x00, 0x26, 0x50, 0x0d, 0x00, 0xc5,
+	0x0f, 0x0e, 0xee, 0xf1,
+	0x32, 0x32, 0xf0, 0xca, 0xe0, 0x01, 0x03, 0x1e, 0x29, 0xdf, 0xe2, 0xf3,
+	0xaf, 0x0b, 0x6e, 0x00,
+	0xe3, 0xf2, 0x0e, 0xfb, 0x00, 0x30, 0x00, 0x93, 0x00, 0x00, 0x1c, 0x1e,
+	0x41, 0x07, 0xe1, 0x20,
+	0x6e, 0x10, 0x21, 0x13, 0xff, 0xdd, 0x30, 0x22, 0x26, 0xe0, 0x9f, 0xde,
+	0xd0, 0xf0, 0x01, 0xfe,
+	0xd9, 0x0f, 0x72, 0x04, 0xd0, 0xfb, 0x10, 0x40, 0x11, 0x01, 0x01, 0x00,
+	0x1d, 0xc0, 0xe2, 0x01,
+	0x4d, 0x3c, 0xe2, 0x85, 0x02, 0x30, 0x1e, 0x0f, 0x22, 0xc3, 0xd5, 0x10,
+	0x7d, 0x3a, 0xe1, 0xd1,
+	0x02, 0xe0, 0xef, 0x0d, 0x0e, 0x50, 0x13, 0xff, 0xcb, 0x10, 0x30, 0x16,
+	0x00, 0x01, 0xce, 0xdb,
+	0x00, 0xf1, 0xd3, 0xb0, 0x0f, 0x0c, 0x30, 0x12, 0x10, 0x9f, 0xe2, 0x57,
+	0x53, 0x00, 0x0b, 0x0e,
+	0x2f, 0x00, 0xa4, 0xf1, 0x02, 0x0e, 0x2f, 0x1f, 0xee, 0xfe, 0x04, 0x20,
+	0x2a, 0x00, 0x13, 0x43,
+	0x10, 0xbb, 0xce, 0xee, 0x51, 0x52, 0xf0, 0xd2, 0xe2, 0x0c, 0x2f, 0x30,
+	0xf1, 0xb0, 0x0c, 0x2f,
+	0x22, 0x17, 0x01, 0xdf, 0x0b, 0x00, 0x00, 0x14, 0xf0, 0x8f, 0xff, 0x05,
+	0x52, 0x0b, 0xdd, 0xfe,
+	0x00, 0x64, 0x11, 0x12, 0xf0, 0xff, 0x00, 0xfe, 0x20, 0x14, 0x14, 0xdd,
+	0x9d, 0xf0, 0x05, 0x43,
+	0x01, 0xec, 0xfc, 0x30, 0x40, 0x11, 0xfe, 0xde, 0x02, 0x05, 0xb0, 0x0f,
+	0x4d, 0x30, 0xed, 0xae,
+	0xe2, 0xe7, 0x01, 0x7d, 0x1e, 0x00, 0x07, 0xe0, 0x0d, 0x1f, 0x12, 0x47,
+	0x22, 0xe0, 0xbb, 0x00,
+	0x2f, 0x32, 0x02, 0xe1, 0xbe, 0x00, 0xf5, 0x01, 0x1f, 0x0e, 0x00, 0x01,
+	0xf2, 0xf2, 0x13, 0x60,
+	0x3b, 0x10, 0x00, 0xd0, 0x02, 0xff, 0x1e, 0x30, 0x54, 0x0f, 0xd9, 0xff,
+	0x0e, 0x00, 0xf5, 0xe1,
+	0x0e, 0x0e, 0xe0, 0x3f, 0x7e, 0x70, 0x10, 0xe0, 0x04, 0x12, 0x0f, 0x08,
+	0xff, 0xb0, 0xf3, 0x40,
+	0x32, 0x03, 0x00, 0xdf, 0x9c, 0xdd, 0xee, 0x25, 0x26, 0x00, 0xdb, 0xd0,
+	0xd0, 0x91, 0x02, 0x20,
+	0x30, 0x2f, 0xeb, 0xdf, 0x00, 0x31, 0x63, 0x4f, 0x0b, 0xfe, 0xb0, 0x01,
+	0x26, 0x22, 0x00, 0x0d,
+	0x20, 0xf2, 0xc3, 0xf0, 0x0c, 0x3d, 0x00, 0xb6, 0xe1, 0xdd, 0x0d, 0x5e,
+	0x10, 0xe5, 0xd0, 0x20,
+	0xfc, 0xa0, 0x03, 0x40, 0x0f, 0xd2, 0xe4, 0xff, 0x0b, 0x10, 0x01, 0x01,
+	0x27, 0x00, 0x1a, 0x00,
+	0x74, 0xf3, 0xd6, 0xf7, 0x47, 0xf6, 0xfe, 0xf8, 0xca, 0x00, 0x63, 0x63,
+	0x10, 0x0d, 0xea, 0x00,
+	0x13, 0x44, 0x20, 0x0f, 0x01, 0xf7, 0xd0, 0xe1, 0xe1, 0x00, 0x2c, 0x4f,
+	0xde, 0xbf, 0xf0, 0xd5,
+	0x01, 0x31, 0x7f, 0x0e, 0xef, 0xdb, 0x0f, 0x03, 0x67, 0x00, 0xf0, 0xee,
+	0xf0, 0x02, 0x4f, 0x1b,
+	0x12, 0x26, 0x11, 0xd0, 0xbc, 0x00, 0x11, 0x61, 0x00, 0xff, 0xde, 0xf0,
+	0xf2, 0x01, 0x0b, 0x2d,
+	0x7f, 0x00, 0xd0, 0xef, 0x12, 0x70, 0x2f, 0x0f, 0xf3, 0xf7, 0x03, 0xfa,
+	0x2f, 0x02, 0xe4, 0xe0,
+	0x30, 0x2f, 0x8e, 0xef, 0x02, 0x33, 0x10, 0x09, 0x20, 0x23, 0xb2, 0xff,
+	0x0c, 0x20, 0x16, 0xd0,
+	0xf0, 0x0f, 0x1d, 0x0f, 0xef, 0xc0, 0x00, 0x5f, 0x21, 0x45, 0xfe, 0xde,
+	0x33, 0x76, 0x60, 0xfe,
+	0xe0, 0xd0, 0x10, 0x1f, 0x1a, 0x11, 0x04, 0x00, 0xaf, 0x0d, 0xf0, 0x04,
+	0x00, 0x21, 0x24, 0xcf,
+	0xdd, 0xff, 0x44, 0x33, 0x0e, 0xfe, 0x01, 0xf0, 0x89, 0xee, 0x00, 0x30,
+	0x23, 0x30, 0xfa, 0xaf,
+	0x00, 0x01, 0x30, 0x0e, 0xe0, 0xe1, 0x10, 0x3d, 0xc0, 0xdc, 0x40, 0x51,
+	0xf7, 0xc4, 0xf1, 0x00,
+	0x3c, 0x4d, 0x02, 0x16, 0x43, 0x21, 0xff, 0xce, 0x0e, 0xef, 0xd2, 0xf4,
+	0x20, 0x7c, 0x0e, 0xbf,
+	0xe1, 0xf3, 0x00, 0x54, 0x00, 0x0c, 0x0d, 0x00, 0x01, 0x0d, 0x0a, 0x3f,
+	0x31, 0x01, 0xbe, 0xdb,
+	0xf1, 0x35, 0x53, 0x04, 0xe1, 0xff, 0xfc, 0x10, 0x26, 0x31, 0xef, 0xce,
+	0xc3, 0xf3, 0x6f, 0x3b,
+	0x1d, 0xf0, 0xe1, 0x07, 0xe0, 0x10, 0x70, 0x3e, 0xce, 0xd0, 0x04, 0x03,
+	0xf0, 0x1a, 0x50, 0xf0,
+	0xc0, 0x00, 0x2f, 0x0f, 0x1f, 0x00, 0x0f, 0x10, 0x00, 0x10, 0x70, 0x10,
+	0xff, 0xbf, 0xf2, 0xf3,
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+	0xe3, 0x01, 0x30, 0x2b,
+	0x2d, 0x00, 0xf3, 0x0f, 0xf9, 0x20, 0x5f, 0x53, 0x23, 0x0c, 0x1b, 0xe0,
+	0xc5, 0x02, 0x31, 0x3f,
+	0xac, 0xe0, 0xb0, 0xf1, 0x0e, 0x7b, 0x10, 0xf0, 0xc0, 0xfd, 0x0d, 0x3f,
+	0x0c, 0x0d, 0xd4, 0xe7,
+	0xc6, 0x0f, 0x3e, 0x40, 0x00, 0xd1, 0xf2, 0xf0, 0x3d, 0x70, 0x30, 0xf1,
+	0xed, 0x1c, 0x20, 0xe0,
+	0xf3, 0x20, 0x60, 0x0e, 0xe0, 0xdf, 0x0b, 0xc0, 0xdf, 0x00, 0x00, 0xd5,
+	0x80, 0xdc, 0x2c, 0x5c,
+	0x00, 0xe1, 0x06, 0x50, 0x00, 0xff, 0x00, 0x33, 0x0f, 0x1d, 0x01, 0x06,
+	0xf2, 0xf0, 0xee, 0xf0,
+	0x33, 0x13, 0x81, 0xf1, 0x01, 0x43, 0x07, 0xd5, 0x30, 0x4c, 0x0e, 0x01,
+	0x44, 0x11, 0xcf, 0xcf,
+	0x00, 0x40, 0x3e, 0x0b, 0x0f, 0xe1, 0x05, 0xdf, 0xef, 0x10, 0x60, 0x30,
+	0x20, 0xd0, 0xff, 0x08,
+	0xc0, 0xd2, 0xc3, 0x0c, 0xb0, 0x01, 0x72, 0x5e, 0x0d, 0x04, 0xf5, 0xe0,
+	0x10, 0x0f, 0x32, 0xf4,
+	0xd1, 0x01, 0x3f, 0xca, 0xcf, 0x0f, 0xee, 0x00, 0xf6, 0x00, 0x4d, 0x5e,
+	0x00, 0xd2, 0xdf, 0x2d,
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+	0x00, 0xce, 0xf1, 0x04,
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+	0x04, 0x60, 0x0d, 0x0f,
+	0xff, 0xef, 0xef, 0xe1, 0x07, 0x20, 0x00, 0x0d, 0x8c, 0xff, 0x0f, 0x00,
+	0x03, 0xf4, 0x34, 0x4f,
+	0xf9, 0xaf, 0x01, 0x75, 0x31, 0xf0, 0xd2, 0x1f, 0x5e, 0x20, 0x00, 0xce,
+	0x00, 0x07, 0x01, 0xf0,
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+	0x10, 0x43, 0x0f, 0x2f,
+	0xe1, 0xb4, 0xdb, 0xee, 0x24, 0x76, 0x10, 0xe0, 0xc3, 0x00, 0x10, 0x21,
+	0x13, 0x18, 0x0c, 0x20,
+	0x01, 0xe4, 0xbf, 0x0b, 0x4f, 0x61, 0x04, 0xf0, 0xff, 0x0f, 0x00, 0x13,
+	0x52, 0x00, 0xef, 0xfe,
+	0x0d, 0x20, 0x15, 0xf0, 0xee, 0xfe, 0x0f, 0x30, 0x71, 0x5d, 0x0c, 0xa1,
+	0x03, 0x02, 0x1c, 0x0b,
+	0xde, 0x01, 0x05, 0xff, 0xfd, 0xf0, 0xe0, 0xfc, 0xf2, 0xf4, 0xd5, 0xd2,
+	0x20, 0x70, 0x7f, 0x1d,
+	0x0f, 0xe2, 0xf7, 0xf1, 0x2e, 0x5d, 0x0f, 0xdf, 0xff, 0x01, 0x27, 0x14,
+	0x0e, 0xee, 0xe2, 0xc4,
+	0x0f, 0x2e, 0x4f, 0x3e, 0xfe, 0xb3, 0xf3, 0x00, 0xd8, 0x0f, 0x70, 0x10,
+	0xcf, 0xcf, 0x0d, 0x01,
+	0x16, 0x10, 0x0e, 0x0d, 0xef, 0xe0, 0xf4, 0x10, 0x1d, 0x1c, 0x21, 0x43,
+	0x5f, 0x3c, 0xfe, 0xf0,
+	0xf4, 0xf3, 0x43, 0x71, 0x1c, 0x0e, 0xd0, 0x07, 0x34, 0xd0, 0xdf, 0x0f,
+	0x50, 0x00, 0x90, 0xfe,
+	0xf0, 0x01, 0x0b, 0x1e, 0xf0, 0x12, 0x22, 0xf1, 0xdf, 0x0d, 0x7a, 0x10,
+	0xf5, 0xf1, 0x1e, 0x3e,
+	0xf4, 0x00, 0x0c, 0xbf, 0xd3, 0xf5, 0x1f, 0x10, 0x20, 0x22, 0x5e, 0x0e,
+	0x82, 0xf5, 0x00, 0x3d,
+	0x2e, 0x30, 0x02, 0xbf, 0x0e, 0x0e, 0x00, 0xf4, 0xd1, 0xcf, 0xfd, 0x2f,
+	0x6d, 0xf2, 0xb7, 0x00,
+	0x1c, 0x2b, 0x0f, 0x10, 0x12, 0xd1, 0xc3, 0x04, 0x20, 0x2e, 0x00, 0xdd,
+	0x9b, 0x01, 0x27, 0x52,
+	0x0e, 0x0e, 0x21, 0x52, 0xdf, 0xcf, 0x04, 0x20, 0x2e, 0x00, 0x22, 0xec,
+	0x8d, 0xf2, 0x07, 0x30,
+	0x2b, 0xe0, 0xd0, 0x32, 0x4f, 0x0e, 0x0f, 0x2f, 0x0d, 0xb0, 0xf4, 0x04,
+	0x3f, 0x3d, 0xaf, 0xe1,
+	0xe2, 0x05, 0x24, 0x20, 0x1a, 0x0f, 0xf0, 0x00, 0xc0, 0xde, 0x20, 0x40,
+	0x20, 0x11, 0x22, 0xff,
+	0x8d, 0xe0, 0x03, 0x57, 0x07, 0xf1, 0xee, 0x1f, 0x30, 0x4f, 0x00, 0xd0,
+	0xdd, 0x3e, 0x43, 0x05,
+	0xe1, 0x1e, 0x0d, 0xbf, 0xf5, 0x11, 0x1f, 0x0c, 0x3f, 0xcd, 0x84, 0xb3,
+	0x1d, 0x28, 0x10, 0x12,
+	0x03, 0xff, 0xdd, 0xf0, 0x20, 0x6f, 0x13, 0x37, 0x30, 0x0f, 0xbe, 0xfd,
+	0x0b, 0x10, 0x11, 0xf4,
+	0xe2, 0x03, 0x01, 0x0e, 0xff, 0xc0, 0x05, 0x33, 0x01, 0xb1, 0x01, 0x55,
+	0x02, 0x9e, 0xfb, 0x00,
+	0x7f, 0x22, 0x05, 0x05, 0x10, 0xdd, 0xdc, 0x11, 0x54, 0x12, 0xe0, 0xde,
+	0x1f, 0x3f, 0x00, 0xa4,
+	0x0e, 0x1a, 0x00, 0x1f, 0x21, 0x2f, 0x5b, 0x10, 0x00, 0xf2, 0x22, 0x4f,
+	0xfc, 0xc1, 0x05, 0x10,
+	0x01, 0xc0, 0xee, 0x00, 0xe1, 0xa0, 0x0e, 0xec, 0xc0, 0xd0, 0xf4, 0x5f,
+	0x20, 0xee, 0xed, 0x2d,
+	0x40, 0x12, 0x16, 0x05, 0xb0, 0x00, 0x0e, 0x1f, 0x5c, 0x0f, 0xee, 0xd0,
+	0xf7, 0x02, 0x1e, 0x3d,
+	0x40, 0x00, 0xc1, 0x02, 0x26, 0x71, 0x40, 0xfc, 0xee, 0xf1, 0xf3, 0xd0,
+	0xed, 0x2e, 0x40, 0x04,
+	0x8d, 0xfc, 0xd3, 0x17, 0x30, 0x0f, 0xf0, 0xff, 0x22, 0x15, 0xcc, 0xfa,
+	0x3f, 0x30, 0xf1, 0x03,
+	0x10, 0x20, 0x3d, 0x10, 0x4f, 0x21, 0xe5, 0x94, 0x30, 0x70, 0x20, 0xee,
+	0xcc, 0x00, 0x07, 0xd1,
+	0x2f, 0x3d, 0xe0, 0x90, 0xfd, 0x02, 0x01, 0xff, 0x0d, 0x30, 0x26, 0x21,
+	0x2f, 0x1b, 0xf0, 0xc1,
+	0x07, 0xe2, 0x20, 0x7e, 0x20, 0xe0, 0xd1, 0xc0, 0xad, 0x00, 0x03, 0x13,
+	0x3f, 0x4b, 0x1d, 0xcd,
+	0xd0, 0x03, 0x4d, 0x3b, 0x21, 0x05, 0xe0, 0xec, 0x00, 0x00, 0x05, 0xef,
+	0xdd, 0x70, 0x71, 0x23,
+	0xe0, 0xe0, 0xf1, 0x06, 0x00, 0x2e, 0x30, 0x11, 0x2e, 0xdf, 0xc0, 0xe4,
+	0xc1, 0xfe, 0x08, 0x4f,
+	0x02, 0xf3, 0xe0, 0x0c, 0xc0, 0xc0, 0xf5, 0x00, 0x00, 0x60, 0x20, 0x3e,
+	0x6e, 0x00, 0xc0, 0x05,
+	0x11, 0x20, 0x1c, 0x0f, 0x4c, 0x41, 0xe6, 0x01, 0x4f, 0xdc, 0xb0, 0x02,
+	0x30, 0x31, 0xa1, 0xf0,
+	0x04, 0xff, 0xeb, 0xc0, 0x0f, 0x02, 0x03, 0x0e, 0x6f, 0x30, 0x01, 0xc1,
+	0x05, 0x00, 0x3e, 0x49,
+	0x0f, 0xd0, 0xd1, 0x0f, 0xd0, 0x92, 0xf4, 0xfd, 0x0c, 0x2c, 0x0d, 0x0f,
+	0x00, 0x53, 0x03, 0x2e,
+	0x0c, 0x21, 0x37, 0x61, 0x20, 0xff, 0xd0, 0xf4, 0xc4, 0xc1, 0x00, 0x6e,
+	0x1e, 0xf0, 0xc5, 0x01,
+	0x0f, 0x00, 0x34, 0x00, 0xec, 0x0b, 0x10, 0xf3, 0x01, 0x70, 0x2e, 0xfe,
+	0xc0, 0x02, 0x4e, 0x0a,
+	0xe0, 0x01, 0x13, 0xf1, 0x80, 0x0f, 0x0d, 0x1f, 0x10, 0x01, 0xe5, 0x14,
+	0x44, 0x13, 0x21, 0x0f,
+	0xce, 0x08, 0x0f, 0x01, 0x27, 0x20, 0xfe, 0x8f, 0x00, 0x12, 0x03, 0xe4,
+	0x00, 0x6e, 0x1e, 0xe1,
+	0xd6, 0x0e, 0x2d, 0x1f, 0x00, 0x00, 0x0e, 0x20, 0x33, 0x0b, 0x0c, 0xf0,
+	0xff, 0x0f, 0x60, 0x3f,
+	0x50, 0x00, 0xc5, 0xc3, 0xd0, 0xeb, 0x00, 0x35, 0xf2, 0x80, 0xfd, 0xef,
+	0x0f, 0x30, 0x10, 0xfc,
+	0x0e, 0xe0, 0xf1, 0x07, 0x32, 0x30, 0x0e, 0xff, 0x74, 0x57, 0x02, 0xce,
+	0x0c, 0x10, 0x00, 0x10,
+	0x00, 0xd1, 0xe1, 0x4d, 0x7b, 0x3e, 0xef, 0xa3, 0x02, 0x02, 0x0d, 0x1a,
+	0x0f, 0xe0, 0x01, 0x14,
+	0xc0, 0xbd, 0xff, 0x00, 0x42, 0x01, 0xe3, 0x0f, 0x2d, 0x1e, 0x00, 0x21,
+	0x24, 0x30, 0x2b, 0x0e,
+	0x0d, 0x30, 0xde, 0x8d, 0xae, 0x0f, 0x3e, 0x28, 0xee, 0xc0, 0x06, 0x01,
+	0xef, 0x0f, 0x00, 0x10,
+	0x0d, 0x0c, 0x70, 0x00, 0xe3, 0xcc, 0x0b, 0x01, 0x13, 0x10, 0x3c, 0x30,
+	0x50, 0x14, 0xe4, 0xe4,
+	0xf5, 0x0f, 0x3d, 0x30, 0x01, 0xa3, 0xc1, 0xc0, 0x0c, 0x00, 0x00, 0x00,
+	0x0f, 0x53, 0x47, 0x36,
+	0xee, 0xfe, 0x30, 0x30, 0x06, 0xd0, 0xee, 0x70, 0x10, 0xd0, 0xc0, 0x00,
+	0x01, 0x00, 0xe0, 0x0e,
+	0x4f, 0x3b, 0x30, 0x20, 0xd0, 0x00, 0x40, 0x20, 0x14, 0x02, 0x70, 0x6d,
+	0x04, 0xe3, 0xc0, 0x00,
+	0xf0, 0x11, 0x42, 0x06, 0xbe, 0xd9, 0x0f, 0x40, 0x23, 0xe2, 0xcf, 0x0e,
+	0x14, 0x0f, 0xdb, 0x0f,
+	0x40, 0x42, 0x31, 0xfb, 0xaf, 0x00, 0x00, 0x5b, 0x00, 0x02, 0xc2, 0x0f,
+	0x1d, 0x0d, 0x3c, 0x0f,
+	0xdf, 0xb5, 0x05, 0x00, 0x10, 0xf1, 0x00, 0x4f, 0x70, 0x03, 0xef, 0xdc,
+	0x01, 0x57, 0x04, 0xfd,
+	0xec, 0x1f, 0x3f, 0x01, 0xc0, 0xe1, 0x07, 0x04, 0xef, 0xab, 0xf0, 0x0f,
+	0x61, 0x12, 0xf0, 0xee,
+	0x0f, 0x40, 0x25, 0xe0, 0xdb, 0x0f, 0x61, 0x12, 0xf0, 0xee, 0xf0, 0x06,
+	0x31, 0x0f, 0xbe, 0xf1,
+	0x14, 0x31, 0x40, 0x0e, 0xbc, 0x00, 0x02, 0x1f, 0x0d, 0x00, 0x06, 0x21,
+	0x3e, 0xec, 0xce, 0xf1,
+	0x07, 0x61, 0x21, 0x0f, 0xcc, 0xe0, 0xf2, 0x00, 0x02, 0xe2, 0x12, 0x7f,
+	0x2b, 0x1e, 0xff, 0xe2,
+	0xf4, 0x02, 0x1c, 0x1d, 0x3d, 0xe0, 0xc6, 0x02, 0x00, 0x3d, 0x30, 0x10,
+	0xd0, 0x83, 0xf1, 0x05,
+	0x21, 0x40, 0x1b, 0x00, 0x2f, 0x52, 0x01, 0x30, 0x4c, 0x1e, 0xee, 0x90,
+	0xf3, 0x06, 0x3f, 0x1d,
+	0xed, 0xd0, 0xc1, 0xd2, 0xd3, 0x11, 0x6e, 0x1b, 0x00, 0x02, 0xe2, 0xe0,
+	0x1d, 0x30, 0x1e, 0x9f,
+	0xee, 0x10, 0x41, 0x37, 0x14, 0xf0, 0x82, 0xc2, 0x02, 0x04, 0x4d, 0x09,
+	0xef, 0xd1, 0x14, 0x12,
+	0x1c, 0x0c, 0x6e, 0x3f, 0x11, 0x05, 0xe1, 0x00, 0x2c, 0x10, 0x00, 0x2f,
+	0x20, 0x16, 0xe1, 0x8f,
+	0xde, 0xd0, 0xd2, 0x01, 0x03, 0x2d, 0xfd, 0x9e, 0xf0, 0x10, 0x4d, 0x5b,
+	0x01, 0xf4, 0xe3, 0xe0,
+	0x2d, 0x4e, 0x0f, 0xdd, 0x0b, 0x2e, 0x20, 0x07, 0xe4, 0xc0, 0xff, 0xfa,
+	0x00, 0x1f, 0x22, 0xd2,
+	0x8f, 0x0e, 0x0f, 0x10, 0x20, 0x04, 0x03, 0x12, 0x1d, 0x3c, 0x1f, 0xdc,
+	0x00, 0x43, 0x14, 0xe0,
+	0xca, 0x2f, 0x71, 0x51, 0xb1, 0xf0, 0xff, 0x0c, 0x2f, 0x00, 0xf4, 0xd5,
+	0x05, 0x4f, 0x3d, 0x0f,
+	0xce, 0xe2, 0x03, 0x21, 0x5f, 0x2c, 0x00, 0x0f, 0xe2, 0xf4, 0xe1, 0x12,
+	0x31, 0x51, 0xf2, 0x84,
+	0xf0, 0xfc, 0x08, 0xdf, 0xf0, 0x00, 0x42, 0x00, 0x20, 0x4e, 0x0a, 0x2f,
+	0x0f, 0x50, 0x43, 0x03,
+	0xf2, 0xf0, 0xfd, 0xe1, 0x02, 0x02, 0xe5, 0xb1, 0xed, 0xdb, 0xf0, 0x00,
+	0x03, 0x61, 0x32, 0x0b,
+	0xac, 0x0f, 0x12, 0x53, 0x00, 0xff, 0xfe, 0x00, 0x04, 0x02, 0x12, 0x33,
+	0xf3, 0xd0, 0xed, 0x0b,
+	0xed, 0xa0, 0x0d, 0xfe, 0x2e, 0x23, 0x17, 0x11, 0xb2, 0xd2, 0x01, 0xd1,
+	0x01, 0x39, 0x6d, 0x3f,
+	0x2f, 0x01, 0x11, 0xec, 0xad, 0xfc, 0x00, 0x44, 0x00, 0x03, 0x0d, 0xf8,
+	0xdf, 0x0e, 0xf0, 0x60,
+	0x30, 0xd4, 0xd7, 0xd6, 0xe0, 0xec, 0x1e, 0x70, 0x30, 0xf3, 0xc1, 0xf0,
+	0xff, 0xec, 0xfe, 0x2f,
+	0x70, 0x52, 0x11, 0xf0, 0xdf, 0x10, 0x24, 0x40, 0x0c, 0x1e, 0xf2, 0xe7,
+	0xf1, 0xd0, 0xf0, 0x00,
+	0x32, 0x3e, 0xed, 0xdf, 0xd3, 0x03, 0x13, 0x02, 0xd3, 0x85, 0xd5, 0x22,
+	0x70, 0x4b, 0xdf, 0xd0,
+	0xf5, 0x21, 0x12, 0x0e, 0x2b, 0x20, 0x10, 0xe3, 0xd2, 0x00, 0x2b, 0xaf,
+	0xc0, 0x00, 0xff, 0x10,
+	0x21, 0x14, 0xf1, 0xcd, 0xdf, 0x0e, 0x5f, 0x6f, 0x31, 0x25, 0x02, 0x00,
+	0x0e, 0x5e, 0x0f, 0xa1,
+	0xf2, 0x01, 0x08, 0x1a, 0x00, 0xe3, 0xf1, 0x01, 0x0f, 0x0e, 0x1f, 0x71,
+	0x23, 0xe3, 0xbf, 0x0b,
+	0x1d, 0x2c, 0x0b, 0x0f, 0x00, 0xe4, 0xf6, 0x40, 0x3f, 0x1d, 0x9f, 0xd0,
+	0xe4, 0x02, 0x32, 0x40,
+	0xed, 0xb0, 0x04, 0xe1, 0x0b, 0x1d, 0x1e, 0x3d, 0x10, 0x31, 0x64, 0x1f,
+	0x2d, 0x03, 0x04, 0x03,
+	0x02, 0x03, 0x0c, 0x2b, 0x9f, 0xde, 0xe4, 0xc4, 0x00, 0x10, 0x48, 0x00,
+	0xf1, 0xd1, 0xd0, 0xef,
+	0x0e, 0x0f, 0x40, 0x51, 0x24, 0x50, 0x04, 0xd2, 0xee, 0xed, 0x1d, 0x30,
+	0x17, 0x01, 0x00, 0x21,
+	0x50, 0xdd, 0xde, 0x00, 0x4e, 0x00, 0xa1, 0xf2, 0xd6, 0xb0, 0x0e, 0x0e,
+	0x5f, 0x02, 0x07, 0xe5,
+	0xf0, 0xfd, 0xe0, 0x0f, 0x40, 0x71, 0x20, 0xf0, 0xe0, 0xfe, 0x0f, 0x1f,
+	0x30, 0x40, 0x00, 0xcf,
+	0xee, 0xf0, 0x0e, 0x10, 0x17, 0x07, 0xc2, 0xeb, 0x0e, 0x0f, 0xe1, 0xe1,
+	0x00, 0x6e, 0x0f, 0x3f,
+	0x3e, 0x60, 0x41, 0x10, 0x00, 0x07, 0x15, 0x00, 0x00, 0xff, 0x3f, 0x4e,
+	0xf0, 0x9e, 0xec, 0x00,
+	0xd0, 0x04, 0x20, 0x0f, 0xf2, 0x07, 0xf3, 0xf0, 0xee, 0xbd, 0xe1, 0x02,
+	0x41, 0x73, 0x46, 0x00,
+	0x00, 0xfb, 0xd0, 0xf0, 0x00, 0x0e, 0x2d, 0x3d, 0x00, 0xf2, 0x91, 0xe2,
+	0xff, 0x00, 0x0f, 0xce,
+	0xb1, 0x00, 0xfc, 0x1e, 0x30, 0x07, 0x47, 0x04, 0x1e, 0x1d, 0x30, 0x50,
+	0x00, 0xe1, 0xbf, 0xf0,
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+	0x22, 0x50, 0x2f, 0x2c,
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+	0x0d, 0xfa, 0x00, 0x0e,
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+	0x2f, 0x18, 0x0f, 0x0f,
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+	0x20, 0x1e, 0x00, 0xc3,
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+	0xdd, 0x0b, 0x2d, 0xaf,
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+	0xfe, 0xf0, 0x0f, 0x20,
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+	0xcf, 0xcc, 0xee, 0xd1,
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+	0x0c, 0xfe, 0x0f, 0x30,
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+	0x0f, 0x11, 0x14, 0xf2,
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+	0x5b, 0x20, 0x11, 0x41,
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+	0x02, 0x40, 0x4e, 0x1b,
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+	0xc1, 0xd1, 0xdf, 0xaa,
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+	0x70, 0x30, 0x70, 0x2f,
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+	0xf0, 0x02, 0xf1, 0x02,
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+	0x21, 0x61, 0x2e, 0x38,
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+	0xf3, 0x07, 0x13, 0x40,
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+	0xed, 0x01, 0xf6, 0xe0,
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+	0xed, 0x9f, 0xef, 0xe1,
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+	0xde, 0x08, 0xef, 0x0e,
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+	0x60, 0x11, 0x0d, 0xaf,
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+	0xe0, 0x24, 0x35, 0x05,
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+	0xf1, 0x77, 0x53, 0x35,
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+	0x10, 0x30, 0xf0, 0xff,
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+	0x03, 0x00, 0x31, 0x77,
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+	0xf2, 0xc1, 0xe0, 0x33,
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+	0x6c, 0x4f, 0x0f, 0x0d,
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+	0x33, 0x17, 0xff, 0x2e,
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+	0xc6, 0xe2, 0xed, 0xfc,
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+	0xb1, 0x10, 0x7f, 0x42,
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+	0xde, 0xee, 0xee, 0x0b,
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+	0xd7, 0xb6, 0x10, 0x0f,
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+	0x02, 0x17, 0x42, 0x2e,
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+	0x2c, 0x0e, 0xf0, 0x01,
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+	0xff, 0xd0, 0x00, 0x22,
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+	0xbb, 0xea, 0xee, 0x10,
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+	0x01, 0xd0, 0xc3, 0xe5,
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+	0x41, 0x10, 0x10, 0xfd,
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+	0x10, 0x00, 0xca, 0x1f,
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+	0xe0, 0x00, 0xf1, 0x12,
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+	0x10, 0x45, 0xfd, 0xbc,
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+	0xe0, 0xbe, 0xf1, 0xd3,
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+	0x31, 0x10, 0xa0, 0xe0,
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+	0x0b, 0x1e, 0x0b, 0x0d,
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+	0xc0, 0xd1, 0xd1, 0xe6,
+	0x03, 0xd0, 0xbc, 0x0f, 0x4f, 0x52, 0x13, 0x05, 0x36, 0x32, 0x1f, 0x60,
+	0x27, 0x31, 0x41, 0x50,
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+	0xc1, 0x00, 0xee, 0xf0,
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+	0x33, 0xfd, 0xdb, 0xf1,
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+	0xee, 0x0e, 0x70, 0x10,
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+	0xce, 0x1f, 0x40, 0x44,
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+	0x1d, 0x6d, 0x76, 0x12,
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+	0xfd, 0xfe, 0x03, 0xf2,
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+	0xd2, 0xf0, 0xf0, 0x00,
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+	0x50, 0x22, 0x20, 0x1d,
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+	0xf0, 0xd7, 0xd3, 0xd3,
+	0xd1, 0xb0, 0xe0, 0xc1, 0xe0, 0xf8, 0xff, 0xff, 0xef, 0x1e, 0x40, 0x43,
+	0x27, 0x50, 0x01, 0xf0,
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+	0x04, 0x01, 0xfc, 0x0f,
+	0xef, 0xc0, 0x00, 0x0f, 0xee, 0x00, 0x26, 0x54, 0xe1, 0xa0, 0xf0, 0x14,
+	0x1d, 0x61, 0x47, 0x10,
+	0x10, 0x0f, 0x10, 0x22, 0x20, 0x0b, 0x0e, 0x0f, 0x02, 0xf2, 0xed, 0xcc,
+	0xde, 0x80, 0xf0, 0xd1,
+	0x07, 0x04, 0x3e, 0x4f, 0x30, 0xf0, 0x10, 0x6d, 0x00, 0x2e, 0x00, 0x00,
+	0x9a, 0x00, 0x19, 0x00,
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+	0x4e, 0x2f, 0x22, 0x20,
+	0x13, 0x07, 0x07, 0x04, 0xc1, 0xd3, 0xc3, 0xd4, 0xd3, 0xd4, 0xe4, 0xd4,
+	0xc3, 0xd4, 0xd2, 0xd3,
+	0xc2, 0xd3, 0xb4, 0xd3, 0xc2, 0xd3, 0xd5, 0xe2, 0xf2, 0x03, 0x03, 0x13,
+	0x02, 0xf0, 0xaf, 0xdd,
+	0xcb, 0xcc, 0xbb, 0xde, 0xdc, 0xce, 0xdd, 0xce, 0xfd, 0xfc, 0xfd, 0x0d,
+	0x0d, 0x0d, 0x0d, 0xfe,
+	0xfe, 0x0d, 0x0b, 0xfc, 0xab, 0xcd, 0xcd, 0xbc, 0xcc, 0xdd, 0xcc, 0xde,
+	0xed, 0xdc, 0xee, 0xfd,
+	0xfc, 0xee, 0xee, 0xed, 0xed, 0xfd, 0x0c, 0x1d, 0x5c, 0x2c, 0x3c, 0x4d,
+	0x3c, 0x3d, 0x2c, 0x2d,
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+	0xce, 0xdd, 0xed, 0xdd,
+	0xdb, 0xfc, 0x0d, 0x2b, 0x6e, 0x4d, 0x3d, 0x3d, 0x0c, 0x0d, 0x0c, 0xde,
+	0xce, 0xce, 0xc0, 0xd0,
+	0xa2, 0xd4, 0xd3, 0xd3, 0xb5, 0xe4, 0xe3, 0xf6, 0xf2, 0xd4, 0xd2, 0xd1,
+	0xd2, 0xd2, 0xf1, 0x00,
+	0x01, 0xf0, 0xf2, 0xf0, 0xcf, 0xcd, 0xda, 0xcd, 0xdc, 0xbc, 0xdc, 0xcc,
+	0xbb, 0xcd, 0xdd, 0xcc,
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+	0x2e, 0x3d, 0x4b, 0x3d,
+	0x4c, 0x2c, 0x2c, 0x3d, 0x5d, 0x2d, 0x2d, 0x0e, 0x00, 0xef, 0xee, 0xdf,
+	0xbf, 0xd0, 0xe0, 0xc4,
+	0xc3, 0xe2, 0xe1, 0xf2, 0x11, 0x52, 0x67, 0x45, 0x33, 0x53, 0x36, 0x33,
+	0x45, 0x32, 0x33, 0x43,
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+	0x23, 0x14, 0x13, 0xf3,
+	0xd2, 0xa4, 0xd3, 0xc5, 0xe2, 0xd3, 0xf3, 0x03, 0xf2, 0x02, 0x02, 0x04,
+	0x02, 0xf4, 0x03, 0xe4,
+	0xc4, 0xd2, 0xf2, 0xb3, 0xb3, 0xb4, 0xd1, 0xc3, 0xe3, 0xe3, 0x14, 0x43,
+	0x65, 0x42, 0x42, 0x32,
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+	0xbc, 0xbe, 0xdd, 0xdd,
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+	0xef, 0x00, 0x23, 0x00,
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+	0xce, 0xdd, 0xf0, 0xf0,
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+	0x1f, 0x0c, 0xec, 0x0f,
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+	0x03, 0x10, 0xd8, 0x9b,
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+	0x8c, 0xc8, 0x00, 0x33,
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+	0x33, 0x00, 0xff, 0x23,
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+	0x13, 0x55, 0x00, 0xff,
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+	0x00, 0x22, 0xff, 0x0f,
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+	0xcd, 0x00, 0x21, 0x20,
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+	0xcb, 0xee, 0x01, 0x11,
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+	0x00, 0xf0, 0x00, 0x45,
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+	0x00, 0x53, 0xff, 0x0f,
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+	0xed, 0xff, 0xee, 0x22,
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+	0x00, 0xce, 0x00, 0x24,
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+	0xf0, 0x12, 0x12, 0x00,
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+	0x0f, 0x00, 0x00, 0xed,
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+	0xfd, 0x74, 0x77, 0x02,
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+	0x1e, 0xfe, 0x0e, 0x67,
+	0x11, 0xed, 0x98, 0xcd, 0xfd, 0x00, 0xbb, 0xec, 0x00, 0x00, 0xdd, 0xcb,
+	0x00, 0x00, 0x10, 0x10,
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+	0xef, 0xce, 0xed, 0x25,
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+	0x33, 0x00, 0x01, 0x34,
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+	0x23, 0x00, 0xfe, 0x34,
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+	0xcc, 0x00, 0x33, 0x11,
+	0xcc, 0xdc, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x20, 0x43, 0x23, 0x22,
+	0x12, 0x77, 0x24, 0xff,
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+	0xf0, 0x22, 0x21, 0xf0,
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+	0xdd, 0x01, 0x44, 0xff,
+	0xcd, 0x11, 0x56, 0x11, 0xde, 0xee, 0x31, 0xef, 0xde, 0x77, 0x22, 0xee,
+	0x00, 0x32, 0x01, 0x00,
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+	0x32, 0x20, 0xdc, 0xff,
+	0xfe, 0xce, 0x00, 0x35, 0x21, 0xae, 0xf0, 0x24, 0x56, 0x00, 0xdf, 0xff,
+	0xde, 0xdd, 0x0f, 0x74,
+	0x23, 0x10, 0xfc, 0xff, 0x00, 0xde, 0xcc, 0x66, 0x33, 0xef, 0xbb, 0x01,
+	0x33, 0x01, 0xee, 0x00,
+	0x00, 0xe0, 0xee, 0x00, 0x11, 0x0f, 0x33, 0x65, 0x0f, 0xed, 0xcb, 0xfe,
+	0x00, 0x00, 0x00, 0x10,
+	0x21, 0x11, 0x43, 0x77, 0x01, 0x0e, 0x13, 0x23, 0x00, 0xee, 0xdf, 0xf0,
+	0xee, 0x00, 0x31, 0xca,
+	0x9a, 0xee, 0xee, 0x00, 0x64, 0x00, 0xca, 0xef, 0x10, 0x00, 0xee, 0x0f,
+	0x10, 0x00, 0xde, 0x01,
+	0x66, 0x00, 0xde, 0x00, 0x33, 0x12, 0x00, 0x35, 0x73, 0x12, 0x01, 0x22,
+	0x00, 0xcd, 0xec, 0xff,
+	0x00, 0xec, 0xbc, 0xfe, 0x00, 0x00, 0x0f, 0x55, 0x23, 0x0f, 0xff, 0x11,
+	0x13, 0x00, 0xba, 0xdd,
+	0x00, 0x25, 0x00, 0x8d, 0xed, 0x02, 0x35, 0x22, 0x00, 0xdd, 0x0f, 0x33,
+	0x33, 0x01, 0x34, 0x44,
+	0x21, 0x00, 0x00, 0x77, 0x43, 0x00, 0xff, 0x00, 0x22, 0x21, 0x33, 0x55,
+	0x00, 0xfe, 0xdd, 0xfe,
+	0x0f, 0x0e, 0x22, 0x12, 0xfe, 0x0d, 0x0e, 0x76, 0x65, 0x00, 0x00, 0xfd,
+	0x0e, 0x00, 0x0e, 0xfd,
+	0xec, 0xfe, 0xdd, 0x24, 0x67, 0xdf, 0xab, 0x00, 0x21, 0x24, 0x00, 0xf0,
+	0xee, 0xff, 0x01, 0x43,
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+	0x00, 0x35, 0x43, 0xcf,
+	0xcd, 0x01, 0x14, 0x00, 0x32, 0x11, 0xcd, 0xee, 0x14, 0x23, 0x0e, 0xab,
+	0xde, 0xec, 0x0e, 0x31,
+	0xac, 0xa9, 0x00, 0x00, 0x00, 0x00, 0x00, 0xdc, 0x00, 0x54, 0x11, 0x0f,
+	0xff, 0x00, 0x34, 0x43,
+	0xf0, 0x00, 0x33, 0x22, 0xdc, 0x30, 0x76, 0x00, 0xfd, 0x00, 0x20, 0xff,
+	0xfc, 0x10, 0x20, 0x01,
+	0x0f, 0xce, 0xee, 0x22, 0x54, 0x00, 0x11, 0x11, 0xfe, 0xfe, 0xe0, 0x99,
+	0xfe, 0x00, 0x02, 0x22,
+	0x00, 0xf2, 0x02, 0x77, 0x46, 0xf0, 0xff, 0x22, 0x21, 0xbc, 0x00, 0x44,
+	0x00, 0xf0, 0x33, 0x24,
+	0xde, 0xdd, 0xef, 0x01, 0x24, 0xf0, 0x8a, 0xde, 0xfe, 0x01, 0x22, 0xff,
+	0x0f, 0x33, 0xee, 0x88,
+	0xde, 0x11, 0x01, 0xf0, 0x00, 0x00, 0x00, 0x11, 0x21, 0x00, 0x10, 0x54,
+	0x25, 0xde, 0x89, 0x00,
+	0x21, 0x00, 0xff, 0x00, 0x0f, 0x00, 0xef, 0xfe, 0xee, 0x00, 0x20, 0xdd,
+	0xdb, 0x20, 0x43, 0x00,
+	0xdc, 0x1f, 0x72, 0x0f, 0xec, 0x22, 0x21, 0x00, 0x56, 0x00, 0xbd, 0x00,
+	0x0f, 0xdb, 0xef, 0x21,
+	0x53, 0x33, 0x12, 0x10, 0x56, 0x23, 0x00, 0xf0, 0x33, 0x55, 0x00, 0xcd,
+	0xef, 0xf0, 0x03, 0x13,
+	0x02, 0xef, 0x0f, 0x10, 0x0f, 0x11, 0x35, 0x00, 0xdf, 0xde, 0x89, 0xbc,
+	0xef, 0x00, 0x35, 0x43,
+	0x00, 0xff, 0xef, 0x00, 0x12, 0xef, 0x99, 0x0f, 0x00, 0xde, 0xef, 0x22,
+	0x22, 0x00, 0x0f, 0x31,
+	0x57, 0x22, 0x00, 0xde, 0xde, 0x00, 0x52, 0x00, 0xda, 0xde, 0x31, 0x00,
+	0x8b, 0xff, 0x00, 0x32,
+	0x11, 0x21, 0x00, 0x11, 0x32, 0xec, 0xac, 0x23, 0x23, 0xca, 0x00, 0x54,
+	0x21, 0xee, 0xff, 0x11,
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+	0x71, 0x22, 0x37, 0x0f,
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+	0x13, 0x35, 0x10, 0xf0,
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+	0x22, 0xe0, 0xac, 0x0f,
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+	0xee, 0xea, 0x00, 0x10,
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+	0xee, 0xbc, 0x00, 0x0e,
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+	0xe0, 0x10, 0x4f, 0x0f,
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+	0x06, 0x00, 0xfe, 0x2f,
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+	0xde, 0xd0, 0x03, 0xfb,
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+	0x21, 0xee, 0x9d, 0x02,
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+	0xdf, 0xbe, 0x00, 0x01,
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+	0xeb, 0xe0, 0xf0, 0x12,
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+	0x0d, 0xe0, 0x03, 0x02,
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+	0x00, 0x20, 0x21, 0x21,
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+	0x01, 0xfc, 0x0c, 0x32,
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+	0x17, 0x12, 0x2f, 0x72,
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+	0x07, 0x11, 0x0f, 0xfc,
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+	0x17, 0x20, 0x60, 0x10,
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+	0x10, 0x11, 0x10, 0x02,
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+	0x00, 0x10, 0x00, 0x0f,
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+	0x06, 0xf1, 0xb0, 0x12,
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+	0x11, 0x44, 0xf0, 0xcf,
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+	0x15, 0x21, 0x8d, 0x0e,
+	0x03, 0x04, 0x22, 0xe2, 0xce, 0x0b, 0x1e, 0x10, 0x30, 0x33, 0x08, 0xef,
+	0xf0, 0x41, 0x51, 0x20,
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+	0xbc, 0x00, 0x10, 0x54,
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+	0x21, 0x20, 0xef, 0xcb,
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+	0xee, 0xf0, 0x0f, 0xed,
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+	0xca, 0x11, 0x45, 0xee,
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+	0xdf, 0x00, 0xff, 0xec,
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+	0x04, 0xae, 0xc9, 0xff,
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+	0x00, 0xd8, 0xbf, 0x01,
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+	0x00, 0x01, 0x10, 0xdf,
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+	0xe4, 0x07, 0xf1, 0xde,
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+	0x01, 0xee, 0xff, 0x00,
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+	0x01, 0x43, 0x0e, 0x00,
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+	0x10, 0x73, 0x00, 0xec,
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+	0x23, 0x20, 0xcb, 0xdd,
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+	0x34, 0x00, 0xfe, 0x0f,
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+	0xef, 0xf0, 0x37, 0x01,
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+	0x0f, 0xfe, 0xdd, 0x12,
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+	0x33, 0xfc, 0x0e, 0x0e,
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+	0x10, 0x00, 0xee, 0xab,
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+	0x0f, 0xdc, 0x00, 0x32,
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+	0x14, 0xf0, 0xdf, 0xe1,
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+	0x0f, 0x71, 0x00, 0xfe,
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+	0x67, 0x01, 0xee, 0xff,
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+	0xce, 0x00, 0x10, 0x0f,
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+	0x11, 0x10, 0x01, 0x01,
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+	0x77, 0x77, 0x0f, 0xbc,
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+	0xe0, 0x01, 0x00, 0x32,
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+	0x32, 0x22, 0xbb, 0xfd,
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+	0x04, 0xe0, 0x9f, 0x63,
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+	0xef, 0x14, 0x11, 0x21,
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+	0x13, 0x35, 0x11, 0xbd,
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+	0x43, 0x02, 0x12, 0x01,
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+	0x31, 0x32, 0xad, 0xec,
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+	0xad, 0xb8, 0x00, 0x53,
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+	0x00, 0x11, 0x0f, 0xfc,
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+	0x00, 0x1e, 0x30, 0x30,
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+	0xfe, 0x0d, 0x1e, 0x0f,
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+	0x00, 0x1f, 0x1f, 0x00,
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+	0x00, 0xd0, 0x0f, 0x62,
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+	0x0f, 0x20, 0x46, 0xf0,
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+	0x0f, 0xfe, 0xdf, 0x0f,
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+	0x70, 0x02, 0x47, 0x03,
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+	0xe0, 0xff, 0x1d, 0x03,
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+	0xd2, 0xe0, 0x00, 0xff,
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+	0xe4, 0x00, 0x30, 0x30,
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+	0x1e, 0x7e, 0x3f, 0x03,
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+	0xe3, 0xef, 0xff, 0xf0,
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+	0xf0, 0xfb, 0xcf, 0x00,
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+	0xe2, 0xfc, 0x00, 0x41,
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+	0x02, 0x03, 0x0e, 0x3e,
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+	0x01, 0xf0, 0x0e, 0x30,
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+	0xfb, 0xcf, 0x4f, 0x70,
+	0x1f, 0xff, 0xdd, 0xd0, 0x03, 0x12, 0x1a, 0xde, 0xe3, 0x17, 0x00, 0xed,
+	0xee, 0x1f, 0x12, 0x35,
+	0x10, 0x8b, 0xff, 0x20, 0x40, 0x01, 0x12, 0x30, 0xff, 0xfd, 0x71, 0x27,
+	0xc1, 0xd0, 0x0d, 0x40,
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+	0x31, 0x03, 0xaf, 0x1c,
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+	0xf0, 0xb0, 0x00, 0x7f,
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+	0x01, 0xd6, 0x11, 0x10,
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+	0xf1, 0xcd, 0xec, 0x01,
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+	0x40, 0x30, 0x3c, 0x21,
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+	0x0d, 0x0e, 0x4f, 0x2f,
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+	0x02, 0x25, 0x10, 0xc0,
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+	0xbc, 0xde, 0x00, 0x16,
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+	0xbb, 0xcd, 0xff, 0x10,
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+	0xf1, 0xb3, 0xe2, 0x0f,
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+	0x31, 0x4d, 0x2e, 0x10,
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+	0x12, 0x23, 0x00, 0xe1,
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+	0xb2, 0x01, 0x2f, 0x0a,
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+	0x34, 0x31, 0xfc, 0xcd,
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+	0x14, 0x40, 0x04, 0xf4,
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+	0xe7, 0xe1, 0xef, 0xfc,
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+	0xcd, 0x2d, 0x70, 0x30,
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+	0x42, 0x10, 0xed, 0xfe,
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+	0xfe, 0x02, 0xf2, 0xe6,
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+	0x02, 0x20, 0xf0, 0xfe,
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+	0xca, 0x43, 0x10, 0xad,
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+	0x0f, 0xed, 0xf0, 0x11,
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+	0xdd, 0x03, 0xf1, 0xdf,
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+	0x0d, 0x00, 0xff, 0xae,
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+	0x77, 0xae, 0xfb, 0x22,
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+	0x23, 0x10, 0xde, 0x0e,
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+	0x10, 0x11, 0x11, 0x33,
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+	0xf0, 0x05, 0x03, 0x1d,
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+	0x24, 0x0f, 0xff, 0x21,
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+	0xf1, 0x13, 0x43, 0x21,
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+	0x4d, 0x73, 0x02, 0xd8,
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+	0xef, 0x22, 0x44, 0xff,
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+	0x22, 0xee, 0xed, 0x02,
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+	0x30, 0xe0, 0x9e, 0x00,
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+	0xff, 0xf0, 0xff, 0xfe,
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+	0xcb, 0x00, 0x31, 0x00,
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+	0x9c, 0x0e, 0x03, 0x57,
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+	0xe0, 0xe2, 0xe0, 0xef,
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+	0xf0, 0xea, 0x10, 0x43,
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+	0x47, 0x00, 0xdf, 0x20,
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+	0x00, 0x41, 0x45, 0xff,
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+	0x01, 0xee, 0xcf, 0x24,
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+	0xf0, 0x04, 0x00, 0x20,
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+	0x02, 0xfe, 0x00, 0x05,
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+	0xff, 0xd0, 0x01, 0xf0,
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+	0xfd, 0xcc, 0x00, 0x0f,
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+	0x53, 0x01, 0xe0, 0xe0,
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+	0x1f, 0x9c, 0xe0, 0xe0,
+	0x15, 0x24, 0xc0, 0x8b, 0xfe, 0xf1, 0x04, 0x00, 0x00, 0x30, 0x20, 0x0a,
+	0xee, 0x41, 0x64, 0x10,
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+	0xad, 0xff, 0x01, 0x30,
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+	0x2e, 0x00, 0x5a, 0x00,
+	0xa5, 0x16, 0x9f, 0x21, 0x1e, 0x18, 0x8a, 0x21, 0xd2, 0x26, 0x62, 0x01,
+	0xef, 0xf0, 0xfe, 0xca,
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+	0xec, 0x00, 0x01, 0xf0,
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+	0x10, 0x24, 0x25, 0x13,
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+	0x12, 0x0d, 0xe8, 0x4f,
+	0x70, 0x20, 0x0f, 0x20, 0x20, 0xc9, 0xde, 0x41, 0x53, 0x00, 0xeb, 0xde,
+	0x00, 0x32, 0xff, 0x10,
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+	0xf2, 0xe3, 0xd0, 0xe1,
+	0xe3, 0xce, 0xf1, 0x07, 0x0f, 0xeb, 0x21, 0x1e, 0xe9, 0x51, 0x10, 0xdd,
+	0xde, 0xfe, 0x41, 0x0f,
+	0xa8, 0xf0, 0x00, 0xee, 0xce, 0xef, 0x21, 0x20, 0xdc, 0xbd, 0x01, 0x02,
+	0x00, 0xe0, 0xe0, 0xf2,
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+	0xf1, 0x12, 0x53, 0x34,
+	0x13, 0xf0, 0xe0, 0x74, 0x56, 0x10, 0x10, 0x10, 0x20, 0x61, 0x3f, 0x3d,
+	0x60, 0x20, 0x2b, 0x2d,
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+	0xef, 0xf4, 0xe3, 0xde,
+	0xca, 0xee, 0xee, 0xf0, 0xf0, 0xe0, 0xef, 0xcc, 0xdf, 0xf2, 0x07, 0xc0,
+	0xcf, 0xf0, 0x01, 0xf3,
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+	0x45, 0x12, 0x12, 0x02,
+	0xe0, 0xf5, 0x24, 0x43, 0x02, 0xf1, 0xc1, 0xf3, 0x03, 0x01, 0x00, 0xc9,
+	0xff, 0x0e, 0xca, 0x0f,
+	0x60, 0x2f, 0x0d, 0xdb, 0x0e, 0x0e, 0xfc, 0x0e, 0x2e, 0x1d, 0x1b, 0x1e,
+	0x0d, 0x0d, 0x2c, 0x3d,
+	0x0d, 0xed, 0x1d, 0x7d, 0xea, 0x9e, 0xee, 0xdf, 0xef, 0x00, 0xdf, 0xde,
+	0xde, 0xf0, 0xd1, 0x92,
+	0xc2, 0xd0, 0x9d, 0xee, 0xdd, 0xdc, 0xef, 0xf3, 0xc0, 0xdf, 0xe0, 0x00,
+	0x20, 0x47, 0x12, 0x21,
+	0x46, 0x32, 0x20, 0x10, 0x2e, 0x30, 0x50, 0x2e, 0x20, 0x20, 0x11, 0x06,
+	0x45, 0x32, 0x31, 0x55,
+	0x24, 0x01, 0xd0, 0xcf, 0x02, 0x36, 0xe1, 0xd0, 0xf2, 0xe1, 0xc3, 0x92,
+	0xe3, 0xe4, 0xd1, 0xb0,
+	0xf1, 0x25, 0x43, 0x10, 0xe0, 0xf0, 0xf1, 0x02, 0x03, 0x24, 0x25, 0xe1,
+	0x91, 0x03, 0x57, 0x11,
+	0x00, 0x00, 0x10, 0x41, 0x23, 0x23, 0x25, 0x01, 0xf0, 0x00, 0x53, 0x46,
+	0x12, 0x01, 0x02, 0x65,
+	0x11, 0x11, 0x43, 0x33, 0x23, 0x55, 0x31, 0x32, 0x31, 0x42, 0x32, 0x41,
+	0x53, 0x34, 0x35, 0x22,
+	0x20, 0x20, 0x53, 0x34, 0x23, 0x21, 0x20, 0x1f, 0x1e, 0x10, 0x20, 0x0f,
+	0xea, 0xae, 0xdd, 0xcc,
+	0xfe, 0x00, 0x11, 0xff, 0xa9, 0xee, 0xe0, 0xef, 0xde, 0xbc, 0xdd, 0xcd,
+	0xdf, 0xdf, 0xce, 0xee,
+	0xce, 0xee, 0xe3, 0x07, 0xf3, 0xe0, 0xe0, 0xe0, 0xfe, 0x00, 0x24, 0x0f,
+	0xea, 0x3d, 0x7d, 0x60,
+	0x1f, 0x1a, 0x0d, 0x0b, 0x0e, 0x1e, 0x0d, 0x0e, 0x0f, 0xd0, 0xcf, 0xde,
+	0xbc, 0xff, 0xd0, 0xbf,
+	0xdd, 0xc0, 0xd0, 0xdf, 0xdf, 0xef, 0xee, 0xda, 0x00, 0x0f, 0x0d, 0x21,
+	0x77, 0x26, 0x00, 0xcc,
+	0xff, 0x77, 0x31, 0xef, 0xee, 0x21, 0x63, 0x00, 0x10, 0x20, 0x20, 0x30,
+	0x20, 0x30, 0x31, 0x20,
+	0x2f, 0x30, 0x30, 0x0f, 0xfc, 0x9d, 0xee, 0xef, 0xb8, 0xff, 0x03, 0x11,
+	0xfd, 0xfd, 0xff, 0x03,
+	0x14, 0x0f, 0xeb, 0xee, 0x21, 0x76, 0x32, 0x00, 0xf1, 0xf2, 0x13, 0x24,
+	0x01, 0x20, 0x10, 0x20,
+	0x73, 0x3f, 0x0b, 0x50, 0x42, 0x2f, 0x1c, 0x2e, 0x31, 0x40, 0x2e, 0x1c,
+	0x1d, 0xfc, 0x10, 0x00,
+	0xfb, 0xce, 0xde, 0x02, 0x11, 0xce, 0xbb, 0xde, 0xdd, 0x00, 0x03, 0x0f,
+	0xcb, 0xcd, 0xf0, 0x00,
+	0x0e, 0xea, 0x0f, 0x70, 0x40, 0x2d, 0x1a, 0x3f, 0x50, 0x2f, 0x3e, 0x1e,
+	0x1d, 0x1f, 0x0d, 0x00,
+	0x02, 0xef, 0xdd, 0x05, 0x26, 0x03, 0xc2, 0xd2, 0x03, 0x34, 0x03, 0x05,
+	0x03, 0x82, 0xb3, 0x07,
+	0x33, 0x02, 0xf1, 0x03, 0xf3, 0x92, 0xe1, 0x05, 0x44, 0x01, 0xd1, 0x02,
+	0x44, 0x22, 0x11, 0x43,
+	0x42, 0x33, 0x45, 0x42, 0x21, 0x11, 0x31, 0x32, 0x43, 0x12, 0x35, 0x32,
+	0xde, 0xf0, 0x77, 0x33,
+	0xf0, 0xce, 0x12, 0x22, 0xef, 0x00, 0x03, 0xf0, 0x02, 0x10, 0x0e, 0x20,
+	0xfe, 0xba, 0x23, 0x22,
+	0xee, 0x0e, 0x72, 0x20, 0x1c, 0x2f, 0x40, 0x4d, 0xf8, 0x0a, 0x30, 0x10,
+	0xed, 0xbd, 0x0f, 0x0e,
+	0xdc, 0xcf, 0xde, 0xad, 0xcb, 0xde, 0xff, 0xef, 0xde, 0xab, 0xee, 0xef,
+	0xed, 0xff, 0xcd, 0xca,
+	0xef, 0x00, 0x0f, 0xcd, 0xee, 0xf0, 0x04, 0x11, 0xf0, 0xf0, 0xe0, 0xef,
+	0xee, 0xed, 0x0f, 0x42,
+	0x34, 0x10, 0xd8, 0xfe, 0x71, 0x43, 0x20, 0x10, 0x30, 0x30, 0x11, 0x00,
+	0x74, 0x21, 0x0e, 0xfd,
+	0xdd, 0xf0, 0xdd, 0xd0, 0xd6, 0x8e, 0xdd, 0xe3, 0xe0, 0xb9, 0xcb, 0xf0,
+	0xf1, 0xce, 0xdb, 0xdf,
+	0xef, 0xef, 0xfe, 0xdd, 0xcd, 0x00, 0xff, 0xfe, 0xed, 0xed, 0x1e, 0x30,
+	0x50, 0x3f, 0x28, 0x0b,
+	0x3f, 0x50, 0x30, 0x20, 0x0d, 0xfc, 0xed, 0xff, 0xe0, 0xaf, 0xde, 0xcf,
+	0xde, 0xab, 0xed, 0xf0,
+	0xfe, 0xec, 0x0e, 0x00, 0x13, 0x0d, 0xfa, 0x20, 0x30, 0x00, 0xde, 0xef,
+	0x34, 0x52, 0x00, 0x03,
+	0x47, 0x53, 0x12, 0x0f, 0x3e, 0x50, 0x41, 0x42, 0x40, 0x30, 0x54, 0x21,
+	0x30, 0x3e, 0x2b, 0x30,
+	0x35, 0x10, 0x1f, 0x10, 0x12, 0xef, 0x88, 0xde, 0xff, 0xdd, 0xcb, 0xef,
+	0xe0, 0xe0, 0xd0, 0xc0,
+	0xe2, 0xd1, 0xd1, 0xd4, 0xd3, 0x91, 0xe4, 0xf4, 0xf3, 0x05, 0xe1, 0xf0,
+	0x05, 0x44, 0x11, 0x22,
+	0x53, 0x23, 0x32, 0x22, 0x45, 0x43, 0x21, 0x22, 0x24, 0x02, 0x03, 0x02,
+	0x02, 0xe2, 0x04, 0x24,
+	0x12, 0x03, 0x23, 0x02, 0x12, 0x76, 0x73, 0x11, 0x01, 0x02, 0x01, 0xf1,
+	0x12, 0x1f, 0x0f, 0x00,
+	0xea, 0xdd, 0x3e, 0x7f, 0x50, 0x1c, 0x2d, 0x3d, 0x3d, 0x3c, 0x2a, 0x3d,
+	0x4f, 0x60, 0x2f, 0x1b,
+	0x0d, 0x0d, 0x3e, 0x4e, 0x4e, 0x30, 0x40, 0x2d, 0x2f, 0x20, 0x2f, 0x21,
+	0x20, 0x10, 0x0f, 0xef,
+	0xd2, 0xb3, 0xd7, 0xc1, 0xb0, 0xf4, 0xf2, 0xf0, 0xfd, 0x00, 0x47, 0x36,
+	0x10, 0x1f, 0x4f, 0x4e,
+	0x2e, 0x3e, 0x31, 0x12, 0x30, 0x0e, 0x0b, 0xf0, 0x02, 0xcd, 0x8a, 0xf0,
+	0xf1, 0xd0, 0xcc, 0xcd,
+	0xff, 0x02, 0x01, 0xd0, 0xdf, 0xe0, 0xe7, 0xc1, 0xbf, 0xf0, 0xf5, 0xe2,
+	0x04, 0xe3, 0xe2, 0xf2,
+	0xf2, 0x04, 0x33, 0x21, 0x40, 0x41, 0x40, 0x52, 0x30, 0x1e, 0x2a, 0x1c,
+	0x50, 0x40, 0x1c, 0x1f,
+	0x31, 0x20, 0x31, 0x21, 0x22, 0x02, 0xef, 0x00, 0x77, 0x00, 0xfd, 0x01,
+	0x47, 0x33, 0x12, 0xf0,
+	0x88, 0x00, 0x45, 0x00, 0xcf, 0xf0, 0x00, 0x01, 0x21, 0x00, 0x0e, 0x00,
+	0x53, 0x23, 0x30, 0x30,
+	0x50, 0x33, 0x23, 0x2f, 0x1c, 0x2f, 0x43, 0x24, 0x0f, 0x02, 0x27, 0x13,
+	0xf0, 0x12, 0x64, 0xf0,
+	0x12, 0x57, 0xee, 0xbc, 0x00, 0x23, 0xf0, 0xef, 0xf0, 0xdf, 0xcf, 0xc0,
+	0xe0, 0xd0, 0xd3, 0xbf,
+	0xdc, 0xd0, 0xcf, 0xdb, 0xe0, 0xc0, 0xcd, 0xff, 0xf0, 0xdb, 0xcd, 0xee,
+	0x12, 0x13, 0x0f, 0xfc,
+	0xfd, 0xfb, 0x0f, 0x72, 0x0d, 0x08, 0xfd, 0x10, 0x20, 0xec, 0xde, 0xdf,
+	0xbd, 0xef, 0xce, 0xac,
+	0xee, 0xcb, 0xe0, 0x02, 0xd0, 0xf0, 0xf0, 0xe0, 0xff, 0x01, 0xf5, 0xf0,
+	0xde, 0x0f, 0x33, 0x25,
+	0x00, 0x30, 0x72, 0x62, 0x20, 0x20, 0x41, 0x35, 0x30, 0x30, 0x30, 0x30,
+	0x64, 0x12, 0x1c, 0x0c,
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+	0xff, 0x20, 0x41, 0x0f,
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+	0xe0, 0xff, 0x02, 0x01,
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+	0xdf, 0xde, 0x01, 0x23,
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+	0x0f, 0x41, 0xfd, 0xc8,
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+	0xdf, 0xe0, 0xd0, 0xcf,
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+	0x22, 0x53, 0x47, 0x32,
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+	0x0f, 0xde, 0xdc, 0x00,
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+	0x04, 0x22, 0x02, 0xe1,
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+	0xf0, 0x12, 0x12, 0x0f,
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+	0x00, 0x0d, 0x0f, 0x20,
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+	0x09, 0x0d, 0xcb, 0xdd,
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+	0xf0, 0xce, 0xdf, 0xf0,
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+	0x42, 0x20, 0x11, 0x11,
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+	0x01, 0xdf, 0xbe, 0x13,
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+	0x11, 0x13, 0x24, 0x11,
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+	0x1f, 0x51, 0x0f, 0x03,
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+	0x32, 0x0f, 0xfb, 0x0e,
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+	0x22, 0xcc, 0xdc, 0x51,
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+	0x00, 0xe0, 0xce, 0x03,
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+	0x34, 0x33, 0x23, 0x02,
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+	0x01, 0x32, 0x30, 0xf0,
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+	0x0f, 0x0e, 0x00, 0x42,
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+	0xdc, 0xee, 0x40, 0x72,
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+	0xf0, 0xe0, 0xbc, 0xff,
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+	0x74, 0x23, 0x11, 0x77,
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+	0xed, 0x21, 0x34, 0x00,
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+	0xdd, 0x10, 0x33, 0xff,
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+	0xbe, 0xec, 0x12, 0x33,
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+	0xee, 0xfa, 0x10, 0x0f,
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+	0x37, 0xef, 0xdf, 0x20,
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+	0x00, 0xef, 0x0e, 0x0f,
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+	0x00, 0xed, 0xfe, 0x10,
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+	0x34, 0x00, 0xdf, 0x00,
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+	0xbb, 0xee, 0x22, 0x12,
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+	0x53, 0x00, 0xef, 0x01,
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+	0x00, 0x01, 0xff, 0x22,
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+	0xef, 0xfe, 0x32, 0x00,
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+	0x10, 0x21, 0x21, 0xfe,
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+	0x63, 0x13, 0xcc, 0x00,
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+	0xbd, 0xdb, 0x44, 0x13,
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+	0x0f, 0x52, 0x12, 0xf0,
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+	0xfd, 0x00, 0x43, 0x0f,
+	0xec, 0x10, 0x32, 0xff, 0xee, 0x21, 0x45, 0x00, 0xbe, 0xfe, 0x01, 0x03,
+	0x01, 0x03, 0xf0, 0x9a,
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+	0x00, 0xfe, 0xfe, 0x33,
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+	0x33, 0x0f, 0xef, 0x22,
+	0x31, 0xdd, 0xdd, 0x43, 0x45, 0xff, 0xbc, 0x00, 0x55, 0x00, 0xdd, 0xff,
+	0x00, 0x00, 0xfd, 0xef,
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+	0xdc, 0x10, 0x50, 0xfe,
+	0xdd, 0x0e, 0x10, 0x0e, 0xcb, 0x21, 0x31, 0xcd, 0xcd, 0xff, 0x26, 0x43,
+	0xf0, 0xce, 0x10, 0x13,
+	0xef, 0x12, 0x12, 0xcc, 0x23, 0x12, 0xdf, 0xfe, 0xf0, 0xf1, 0xbd, 0xa8,
+	0x00, 0x23, 0xf0, 0xdc,
+	0xee, 0x01, 0x45, 0xef, 0xff, 0xff, 0xac, 0xee, 0x21, 0x34, 0x10, 0xde,
+	0xed, 0x77, 0x43, 0x00,
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+	0x92, 0x00, 0xbf, 0x00,
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+	0xe0, 0x43, 0x77, 0xef,
+	0xcc, 0x21, 0x22, 0xee, 0xbc, 0x10, 0x22, 0xff, 0xbb, 0x10, 0x33, 0x00,
+	0xbc, 0x0e, 0x12, 0x11,
+	0xee, 0x00, 0x01, 0xf0, 0x00, 0x03, 0x00, 0xff, 0x02, 0xbb, 0xdc, 0x13,
+	0x32, 0xff, 0x00, 0x14,
+	0x76, 0x00, 0xef, 0x23, 0x77, 0x01, 0xee, 0xf0, 0x24, 0xf0, 0xcf, 0x00,
+	0x02, 0x22, 0x34, 0x00,
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+	0x03, 0xdd, 0xee, 0x37,
+	0xd1, 0x8e, 0xe0, 0x47, 0x00, 0xce, 0x00, 0x21, 0x12, 0xdf, 0xcd, 0x21,
+	0x44, 0xff, 0x9a, 0x01,
+	0x12, 0xff, 0xed, 0x00, 0x12, 0xec, 0xdc, 0x21, 0x21, 0xfe, 0xec, 0x1e,
+	0x50, 0x00, 0x2c, 0x2f,
+	0xdd, 0x30, 0x21, 0xbb, 0xf0, 0x21, 0x02, 0x9f, 0xff, 0x00, 0xde, 0xb8,
+	0xff, 0x34, 0x00, 0xab,
+	0x00, 0x44, 0x00, 0xf0, 0x0f, 0x12, 0x22, 0x00, 0x00, 0x24, 0x21, 0x21,
+	0x52, 0x23, 0x00, 0x10,
+	0x45, 0x33, 0x0f, 0x31, 0x72, 0x0e, 0x0c, 0x21, 0x42, 0x0e, 0x10, 0x2f,
+	0xf8, 0x0e, 0x20, 0x73,
+	0x10, 0xf9, 0xee, 0x1f, 0x30, 0xff, 0xfd, 0xfe, 0x0e, 0x20, 0x02, 0xc0,
+	0xdd, 0xf0, 0xf3, 0xb0,
+	0xcc, 0xef, 0x47, 0x14, 0xdf, 0xf0, 0x02, 0x11, 0x10, 0x42, 0x33, 0xf0,
+	0xfe, 0x0e, 0x52, 0x47,
+	0x00, 0x00, 0x75, 0x22, 0xec, 0xff, 0x65, 0x11, 0xfe, 0xff, 0x11, 0x23,
+	0x10, 0x0f, 0x11, 0x56,
+	0x21, 0x00, 0x00, 0x00, 0x23, 0x10, 0x56, 0x22, 0xe0, 0xde, 0x21, 0x64,
+	0x00, 0xfe, 0xdc, 0x11,
+	0x75, 0x0e, 0xed, 0x00, 0x10, 0x01, 0x10, 0x1f, 0x0d, 0xed, 0x0f, 0x0f,
+	0xec, 0xf0, 0x45, 0x44,
+	0x00, 0x00, 0x00, 0x34, 0x43, 0x00, 0xe0, 0x17, 0x63, 0x01, 0xe0, 0xe1,
+	0x35, 0x23, 0x02, 0xe1,
+	0xf2, 0x65, 0x21, 0x01, 0x32, 0x22, 0x00, 0x22, 0x64, 0x11, 0x31, 0x11,
+	0xff, 0x10, 0x75, 0x32,
+	0x00, 0x00, 0xb8, 0x00, 0x53, 0x00, 0xbb, 0x00, 0x11, 0x00, 0xee, 0x00,
+	0x01, 0x00, 0xff, 0xde,
+	0xf0, 0x01, 0xef, 0xcd, 0x03, 0x13, 0xca, 0xcd, 0x12, 0x33, 0xec, 0xee,
+	0x44, 0x02, 0xbe, 0xff,
+	0x11, 0x00, 0xdc, 0xdc, 0x0f, 0xf0, 0x9b, 0xfd, 0x0f, 0xec, 0xec, 0xdc,
+	0xef, 0x20, 0xff, 0xab,
+	0x0f, 0x00, 0x00, 0xff, 0xfe, 0x0f, 0x00, 0xff, 0xbc, 0x10, 0x00, 0xed,
+	0xcd, 0xfe, 0xcc, 0xdc,
+	0xee, 0xe0, 0xbc, 0xcc, 0xe0, 0x01, 0xdd, 0xa9, 0x00, 0x00, 0xd0, 0xcc,
+	0x0f, 0x10, 0xff, 0x0e,
+	0x40, 0x31, 0x0c, 0xcb, 0x30, 0x73, 0x13, 0xfe, 0xeb, 0x10, 0x14, 0xe0,
+	0xad, 0xfe, 0x00, 0xd0,
+	0xcd, 0xe0, 0xf2, 0xef, 0xdd, 0xe2, 0x07, 0x02, 0xdf, 0xcd, 0xed, 0x00,
+	0x20, 0xfe, 0xb8, 0xdd,
+	0xee, 0xdf, 0x10, 0x22, 0xbc, 0xba, 0xff, 0x00, 0x0f, 0xeb, 0xef, 0x2f,
+	0x2f, 0x10, 0x30, 0xee,
+	0xcd, 0xfd, 0xfd, 0xbc, 0xed, 0x1f, 0xfe, 0xaa, 0xcd, 0xfd, 0x00, 0xfe,
+	0x9b, 0xfc, 0x00, 0x00,
+	0xef, 0xee, 0xff, 0xff, 0xef, 0xbc, 0xde, 0xe0, 0x00, 0x02, 0xf2, 0xf0,
+	0x11, 0x45, 0x31, 0x42,
+	0x55, 0x10, 0x31, 0x42, 0x34, 0x32, 0x21, 0x32, 0x33, 0x42, 0x1f, 0x35,
+	0x43, 0x00, 0x0e, 0x43,
+	0x34, 0x0e, 0x0f, 0x22, 0xdc, 0xbe, 0x01, 0x00, 0xb2, 0xd4, 0xe4, 0xf7,
+	0xc1, 0xc0, 0xe1, 0xd4,
+	0x04, 0xe3, 0xd0, 0xc0, 0xf1, 0x15, 0x01, 0xf0, 0x23, 0x66, 0x01, 0xf1,
+	0x00, 0x24, 0x33, 0x03,
+	0x24, 0x12, 0x01, 0x74, 0x65, 0x12, 0x01, 0x10, 0x54, 0x44, 0x00, 0x00,
+	0x32, 0x44, 0x11, 0x00,
+	0x12, 0x46, 0x22, 0x00, 0xf0, 0x01, 0x34, 0x22, 0x00, 0x10, 0x52, 0x10,
+	0x1f, 0x2f, 0x2f, 0x6f,
+	0x3d, 0x3e, 0x2c, 0x3c, 0x5e, 0x3e, 0x2d, 0x3e, 0x40, 0x1a, 0x1d, 0x2d,
+	0x3b, 0x3e, 0x3f, 0x0d,
+	0xfb, 0xfd, 0x0e, 0xde, 0x9b, 0xe0, 0x00, 0xce, 0xbd, 0xef, 0x00, 0xdf,
+	0xcb, 0xf0, 0x02, 0xd1,
+	0xcf, 0xf1, 0xe1, 0xc0, 0x00, 0xf3, 0xf4, 0xf0, 0xef, 0xed, 0x00, 0x2f,
+	0xfb, 0xcd, 0x00, 0x14,
+	0x0f, 0xdd, 0x13, 0x76, 0x00, 0x10, 0x42, 0x35, 0x32, 0x42, 0x34, 0x43,
+	0x54, 0x23, 0x24, 0x36,
+	0x32, 0x00, 0x00, 0x03, 0x14, 0x42, 0x01, 0x02, 0x04, 0xf1, 0x11, 0x73,
+	0x54, 0x01, 0x00, 0x31,
+	0x23, 0x00, 0x10, 0x3f, 0x71, 0x22, 0x1e, 0x0c, 0x10, 0x3e, 0x3e, 0x42,
+	0x33, 0x42, 0x35, 0x21,
+	0x21, 0x31, 0x34, 0x47, 0x24, 0x23, 0x32, 0x11, 0x33, 0x75, 0x31, 0x22,
+	0x77, 0x11, 0x00, 0xfe,
+	0x01, 0x33, 0x10, 0x00, 0x10, 0x10, 0x00, 0x12, 0x31, 0x32, 0x72, 0x11,
+	0x10, 0x44, 0x41, 0x1a,
+	0x0d, 0x40, 0x43, 0x2e, 0x3d, 0x50, 0x20, 0x18, 0x20, 0x42, 0x20, 0x1d,
+	0x20, 0x22, 0x0d, 0xa8,
+	0xfe, 0x01, 0xdf, 0xbc, 0xef, 0xe0, 0xef, 0xdd, 0xbc, 0xe0, 0xd2, 0xbe,
+	0xdb, 0xd0, 0xd0, 0xce,
+	0xd0, 0xe0, 0xe0, 0xed, 0xef, 0x02, 0xf0, 0x0d, 0xeb, 0x0e, 0x20, 0x1e,
+	0x0e, 0x77, 0x33, 0x10,
+	0x1d, 0x43, 0x25, 0x00, 0xbb, 0xef, 0x03, 0x01, 0xbe, 0xf1, 0x05, 0xd0,
+	0xae, 0xee, 0xf3, 0x05,
+	0xe0, 0xd0, 0xe1, 0xf1, 0xf1, 0xd0, 0xdf, 0xce, 0xe9, 0x0e, 0x00, 0xe0,
+	0xef, 0xee, 0x21, 0x77,
+	0x01, 0xee, 0x00, 0x23, 0x33, 0x0f, 0x0e, 0x70, 0x30, 0x1a, 0x2e, 0x50,
+	0x3f, 0x3a, 0x3f, 0x5f,
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+	0xee, 0xcd, 0xee, 0xe0,
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+	0x01, 0xee, 0x0f, 0x22,
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+	0x10, 0x2f, 0xee, 0xbf,
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+	0xcd, 0xde, 0xd1, 0xe2,
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+	0xff, 0xc0, 0xe0, 0xff,
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+	0x03, 0x0e, 0x0e, 0xc0,
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+	0xf0, 0x02, 0x26, 0x01,
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+	0xce, 0xe9, 0x13, 0x74,
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+	0x26, 0x33, 0xfe, 0xfd,
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+	0x40, 0x10, 0x0f, 0xff,
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+	0xa0, 0xfe, 0x56, 0x22,
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+	0x16, 0x40, 0xdc, 0xd0,
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+	0x62, 0x10, 0x0d, 0xce,
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+	0xfe, 0x0e, 0xed, 0x32,
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+	0x33, 0x0e, 0x0d, 0x53,
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+	0x56, 0x00, 0xce, 0x11,
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+	0x00, 0xf0, 0x12, 0x41,
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+	0xec, 0x01, 0x03, 0xff,
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+	0xfe, 0xcb, 0x10, 0x40,
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+	0xdf, 0x0f, 0x13, 0x20,
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+	0x03, 0xdc, 0xdc, 0x01,
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+	0x00, 0x54, 0x00, 0x0e,
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+	0x11, 0xfb, 0xdc, 0x11,
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+	0xef, 0xf2, 0xf4, 0xfd,
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+	0x25, 0x21, 0xac, 0xff,
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+	0xbd, 0x0e, 0x50, 0x00,
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+	0xdf, 0x05, 0x13, 0xf0,
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+	0xdf, 0x14, 0x22, 0x11,
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+	0x57, 0x34, 0xfc, 0x0d,
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+	0x22, 0x0e, 0xfd, 0x11,
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+	0xf3, 0x17, 0xe3, 0xbc,
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+	0x02, 0x0d, 0x0e, 0x00,
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+	0x0e, 0xde, 0xff, 0x11,
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+	0x0d, 0x0c, 0x42, 0x33,
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+	0x31, 0x1f, 0xc8, 0xcf,
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+	0x2f, 0x0a, 0x01, 0x20,
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+	0x00, 0x03, 0x00, 0x00,
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+	0xf0, 0xf0, 0xd0, 0xdf,
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+	0x50, 0x00, 0x8d, 0xdf,
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+	0xd0, 0xe0, 0xb0, 0xdf,
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+	0x00, 0x12, 0xf0, 0x04,
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+	0xcf, 0x00, 0x21, 0x37,
+	0x00, 0x9c, 0x0f, 0x00, 0x33, 0x00, 0xfb, 0x00, 0xf3, 0xff, 0x40, 0x52,
+	0xfd, 0xcb, 0x11, 0x32,
+	0x00, 0x0f, 0x0e, 0xfd, 0x9c, 0x03, 0x27, 0x31, 0xdf, 0xac, 0x00, 0x02,
+	0x56, 0x00, 0x0f, 0xee,
+	0x20, 0x12, 0x11, 0x34, 0x30, 0xfd, 0xfd, 0x02, 0x00, 0x00, 0x53, 0xee,
+	0x89, 0xf0, 0x02, 0x12,
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+	0x0e, 0xce, 0x0e, 0x75,
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+	0x02, 0x10, 0x0d, 0x29,
+	0xff, 0x00, 0x02, 0xfe, 0x61, 0x36, 0x30, 0x0f, 0xca, 0xf0, 0x22, 0x11,
+	0xde, 0xf0, 0x10, 0x01,
+	0xf0, 0x8c, 0xb0, 0x00, 0x27, 0x32, 0xe1, 0xbd, 0x20, 0x45, 0x01, 0x0f,
+	0x0d, 0xe0, 0xff, 0x11,
+	0x41, 0x31, 0x11, 0xe0, 0x11, 0x11, 0x01, 0x37, 0xce, 0xde, 0x01, 0x02,
+	0x74, 0x20, 0x0a, 0xf0,
+	0x60, 0x43, 0x00, 0xca, 0xf0, 0x33, 0x20, 0xdc, 0xe1, 0x24, 0x11, 0x0e,
+	0x7f, 0x34, 0x00, 0x0f,
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+	0x19, 0x00, 0x1d, 0x00,
+	0x02, 0x12, 0x60, 0x11, 0xdc, 0x10, 0xef, 0x10, 0x77, 0x00, 0x00, 0xff,
+	0xc0, 0xea, 0x0e, 0x11,
+	0x12, 0xdc, 0xeb, 0xf0, 0xce, 0xce, 0x4f, 0x70, 0x0d, 0xdd, 0x02, 0x01,
+	0x0b, 0x0e, 0x03, 0x03,
+	0xad, 0xfd, 0x27, 0xf0, 0xaf, 0xfd, 0x03, 0x16, 0x10, 0xfb, 0xe0, 0x20,
+	0x62, 0x11, 0x10, 0x44,
+	0x00, 0xfe, 0x0f, 0x10, 0x00, 0x1d, 0x63, 0x13, 0x0d, 0x0d, 0x21, 0x55,
+	0x10, 0x12, 0x37, 0x33,
+	0x31, 0x0d, 0xec, 0x11, 0x77, 0x11, 0x0f, 0x00, 0xe0, 0xf0, 0x23, 0x00,
+	0x01, 0x14, 0x01, 0x32,
+	0x72, 0x76, 0x22, 0xff, 0xfd, 0x33, 0x46, 0x00, 0xdf, 0xcd, 0x00, 0x45,
+	0x23, 0xf0, 0xbf, 0xff,
+	0x00, 0x47, 0x21, 0xe0, 0xf0, 0x42, 0x33, 0x01, 0xe3, 0x12, 0xfe, 0x99,
+	0xf0, 0x23, 0x44, 0x01,
+	0xdf, 0xfd, 0x0f, 0x02, 0x16, 0x11, 0xe0, 0xa0, 0x00, 0x01, 0xe0, 0xe0,
+	0x02, 0x66, 0x00, 0xbd,
+	0x00, 0xe2, 0x0f, 0xe0, 0xf0, 0xf1, 0x02, 0xfe, 0xff, 0xef, 0x02, 0x77,
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+	0x11, 0x31, 0x00, 0x20,
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+	0xde, 0x00, 0xe5, 0xa0,
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+	0x0e, 0x1e, 0x0b, 0xbd,
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+	0xdc, 0xcb, 0xee, 0x0e,
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+	0xdd, 0xdf, 0xde, 0xca,
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+	0x40, 0x0e, 0xcc, 0xcb,
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+	0xe2, 0xf5, 0xf5, 0xe2,
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+	0x10, 0xfe, 0x0f, 0x45,
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+	0xe0, 0xaa, 0xf0, 0x32,
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+	0xee, 0xee, 0x0f, 0x00,
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+	0x0f, 0x12, 0x12, 0xff,
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+	0xee, 0x00, 0x00, 0x23,
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+	0xee, 0x21, 0x77, 0x00,
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+	0xff, 0x11, 0x43, 0x00,
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+	0x00, 0x54, 0x00, 0xa8,
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+	0x11, 0x22, 0xf0, 0xf0,
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+	0x57, 0x55, 0x00, 0xcc,
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+	0x00, 0xfd, 0x01, 0x44,
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+	0x54, 0x0f, 0xff, 0x10,
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+	0xdd, 0x00, 0x12, 0xf0,
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+	0x65, 0x00, 0xdd, 0xcc,
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+	0xef, 0xee, 0xff, 0x00,
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+	0x00, 0x00, 0xdc, 0x00,
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+	0xcc, 0xee, 0x22, 0x67,
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+	0x76, 0xf0, 0xdd, 0xef,
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+	0x10, 0x00, 0xfe, 0xdd,
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+	0xfe, 0x0f, 0x22, 0x31,
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+	0x0f, 0x00, 0x00, 0xec,
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+	0x01, 0xdf, 0xde, 0x36,
+	0x54, 0xf0, 0xad, 0xee, 0x00, 0x11, 0x01, 0x45, 0x45, 0x0f, 0xcd, 0xff,
+	0x01, 0x22, 0x01, 0xff,
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+	0x10, 0xdd, 0x22, 0x21,
+	0xaa, 0x00, 0x42, 0xff, 0xff, 0x44, 0x00, 0xfe, 0xee, 0x00, 0x46, 0x32,
+	0xf0, 0xf0, 0x01, 0x01,
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+	0x00, 0x00, 0x10, 0x00,
+	0x44, 0x43, 0x0f, 0xed, 0x11, 0x44, 0xfd, 0xdc, 0x21, 0x22, 0xff, 0xca,
+	0xfe, 0x11, 0x64, 0x00,
+	0xfe, 0x0f, 0x00, 0x01, 0x66, 0xff, 0xbb, 0xff, 0x00, 0x12, 0x00, 0x00,
+	0x00, 0xaa, 0xf0, 0x21,
+	0x00, 0xef, 0x00, 0xf0, 0xee, 0x23, 0x0f, 0xdc, 0x00, 0x62, 0x11, 0xeb,
+	0xba, 0xfe, 0x0e, 0x20,
+	0x72, 0x0e, 0xb8, 0x0f, 0x10, 0x00, 0x00, 0x10, 0x00, 0x00, 0xf0, 0x00,
+	0x00, 0x00, 0x22, 0x00,
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+	0x00, 0x12, 0x22, 0x00,
+	0x01, 0x23, 0xcd, 0xdd, 0x34, 0x22, 0x9c, 0xff, 0x23, 0x35, 0x21, 0x00,
+	0xde, 0xf0, 0x34, 0x11,
+	0x00, 0x02, 0x01, 0xcf, 0x22, 0x54, 0x01, 0xf1, 0xf0, 0xf0, 0x00, 0x77,
+	0xf0, 0xa9, 0x33, 0x44,
+	0x00, 0xdd, 0xed, 0x00, 0x64, 0x12, 0x0e, 0xec, 0xfe, 0xed, 0x33, 0x56,
+	0xff, 0xff, 0x11, 0x00,
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+	0x12, 0x0d, 0xa8, 0xfd,
+	0xf0, 0x12, 0x66, 0x00, 0xdc, 0xef, 0x0f, 0x01, 0x00, 0xee, 0xf0, 0x00,
+	0xde, 0x32, 0x45, 0xbf,
+	0xdd, 0xcc, 0x13, 0x23, 0xbb, 0xf0, 0x00, 0xf0, 0x02, 0xf0, 0xba, 0xee,
+	0x00, 0x00, 0xff, 0x22,
+	0x11, 0xee, 0x00, 0x12, 0x56, 0xdf, 0x88, 0xdf, 0x21, 0x53, 0x11, 0x00,
+	0xfe, 0xed, 0xff, 0x00,
+	0x1f, 0x10, 0x32, 0x21, 0xdc, 0xaa, 0x00, 0x20, 0x00, 0xf0, 0x00, 0xdf,
+	0xff, 0x00, 0x21, 0x76,
+	0x11, 0xee, 0xfc, 0x0f, 0x20, 0x43, 0x22, 0x22, 0xfe, 0xcb, 0x00, 0x43,
+	0x12, 0x0f, 0xcc, 0xf0,
+	0xf0, 0xde, 0xdf, 0x02, 0x00, 0x88, 0xee, 0x11, 0x34, 0x00, 0xdf, 0xf0,
+	0xef, 0xdd, 0x00, 0x45,
+	0x43, 0x12, 0x21, 0xef, 0xac, 0x01, 0x11, 0x01, 0x77, 0x25, 0xee, 0xde,
+	0x11, 0x21, 0x00, 0x30,
+	0x32, 0xef, 0xdd, 0x0d, 0x1f, 0x0d, 0x0f, 0x72, 0x10, 0x0c, 0xea, 0x9b,
+	0xff, 0x33, 0x22, 0xf0,
+	0xff, 0xdd, 0xed, 0xff, 0xf0, 0x23, 0x34, 0xdf, 0xbc, 0x01, 0x00, 0x9b,
+	0xfe, 0x00, 0x25, 0x00,
+	0xf0, 0x01, 0x32, 0x44, 0x23, 0xff, 0xde, 0xdf, 0xe0, 0x00, 0x66, 0x22,
+	0xff, 0xf0, 0x01, 0x01,
+	0x12, 0xe0, 0xac, 0x15, 0x11, 0xde, 0xcd, 0x14, 0x45, 0xf0, 0xce, 0xee,
+	0xf0, 0x65, 0x01, 0xce,
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+	0x1e, 0x63, 0x23, 0x0c,
+	0xfa, 0x0f, 0x54, 0x34, 0x0f, 0xec, 0x0f, 0x20, 0x31, 0x21, 0x00, 0x0f,
+	0xfd, 0x00, 0x54, 0x0f,
+	0xda, 0x0f, 0x30, 0x30, 0x31, 0x01, 0x73, 0x56, 0x00, 0xff, 0x00, 0x00,
+	0xdf, 0x01, 0x47, 0x00,
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+	0xbb, 0x22, 0x33, 0xdd,
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+	0x00, 0xde, 0x11, 0x43,
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+	0x23, 0xf0, 0xcb, 0x33,
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+	0xdd, 0x1f, 0x72, 0x56,
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+	0xf0, 0xdd, 0x00, 0x36,
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+	0xf0, 0xf0, 0xf0, 0x03,
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+	0x0f, 0x12, 0x12, 0x0c,
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+	0x00, 0xdc, 0xdd, 0x00,
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+	0x34, 0x21, 0x22, 0x00,
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+	0xac, 0x01, 0x14, 0x11,
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+	0x10, 0x52, 0x00, 0x0f,
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+	0x56, 0x00, 0xcf, 0xff,
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+	0x21, 0x0e, 0xca, 0xff,
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+	0x10, 0x74, 0x11, 0x0f,
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+	0x00, 0x10, 0x0e, 0xda,
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+	0x11, 0x00, 0xe0, 0xd0,
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+	0xda, 0x00, 0x41, 0x20,
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+	0xdd, 0x0f, 0x71, 0x11,
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+	0x4d, 0x00, 0x38, 0x00,
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+	0x26, 0x9e, 0xdd, 0xef,
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+	0x35, 0xe0, 0xbe, 0x00,
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+	0x00, 0xec, 0x51, 0x23,
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+	0x00, 0xd0, 0xbf, 0xde,
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+	0x00, 0x12, 0x00, 0x23,
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+	0xbb, 0xde, 0x10, 0x32,
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+	0x03, 0xf1, 0x89, 0xf0,
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+	0x60, 0x42, 0xeb, 0xcc,
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+	0x10, 0x41, 0x10, 0x00,
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+	0x00, 0x24, 0x11, 0xf0,
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+	0x00, 0x01, 0xf0, 0xe0,
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+	0x26, 0xff, 0xff, 0x30,
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+	0x20, 0x21, 0x31, 0x00,
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+	0x01, 0xf0, 0xe0, 0x45,
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+	0x0e, 0xd8, 0x10, 0x31,
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+	0xaa, 0x00, 0x63, 0x1e,
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+	0xf0, 0x03, 0x00, 0xff,
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+	0xf0, 0x36, 0x00, 0xcf,
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+	0xff, 0x11, 0x00, 0x1f,
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+	0xf0, 0xdd, 0xbc, 0x01,
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+	0x22, 0x01, 0xf0, 0xef,
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+	0x22, 0xe0, 0x00, 0xf0,
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+	0xe0, 0xad, 0xef, 0x23,
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+	0xfc, 0x0f, 0x31, 0x10,
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+	0x00, 0x0e, 0x0e, 0xeb,
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+	0x53, 0xff, 0x88, 0xfe,
+	0x11, 0x33, 0x00, 0xfc, 0xee, 0x00, 0x1f, 0x10, 0x10, 0xfd, 0x10, 0x31,
+	0xff, 0xce, 0xdf, 0xff,
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+	0xe0, 0x00, 0x44, 0x75,
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+	0x02, 0x46, 0x00, 0xc0,
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+	0xad, 0x00, 0x57, 0x00,
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+	0x11, 0x12, 0xf0, 0x01,
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+	0xef, 0x02, 0x33, 0x00,
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+	0x0c, 0xfd, 0x10, 0x43,
+	0x12, 0x21, 0x0f, 0xf9, 0x10, 0x43, 0xef, 0x9a, 0xff, 0x0f, 0x11, 0x31,
+	0x00, 0x0e, 0x1f, 0x4f,
+	0x0b, 0xec, 0x20, 0x62, 0x20, 0x31, 0x10, 0xfd, 0xfe, 0x0e, 0xee, 0x12,
+	0x24, 0xdd, 0xbe, 0xe0,
+	0xc0, 0x05, 0x32, 0xfd, 0x46, 0x76, 0xef, 0xcc, 0xef, 0x23, 0x34, 0xf0,
+	0xde, 0xf0, 0xf0, 0x22,
+	0x23, 0xcf, 0xe0, 0x35, 0x00, 0xc0, 0x33, 0x77, 0x00, 0xef, 0xf0, 0x33,
+	0xf0, 0x9c, 0x21, 0x45,
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+	0xbc, 0xeb, 0x20, 0x63,
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+	0xdb, 0xde, 0x44, 0x32,
+	0xe0, 0x00, 0x36, 0x10, 0xe0, 0x32, 0x74, 0x00, 0xf1, 0x12, 0x11, 0x00,
+	0x01, 0xe0, 0x23, 0x33,
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+	0x0f, 0xef, 0xe0, 0xe0,
+	0x04, 0xf2, 0x8c, 0xec, 0x03, 0x25, 0x00, 0xcf, 0xed, 0xf0, 0x21, 0x23,
+	0xff, 0xa8, 0x00, 0x11,
+	0x21, 0x00, 0x00, 0xee, 0xee, 0x11, 0xcd, 0xed, 0x00, 0x0f, 0x0f, 0xdd,
+	0x98, 0x0e, 0x0f, 0x00,
+	0x63, 0x0e, 0xd8, 0x0f, 0x10, 0x30, 0x20, 0x11, 0x00, 0xed, 0xef, 0x20,
+	0xff, 0xbe, 0x01, 0xee,
+	0xee, 0x22, 0xf0, 0xd0, 0x01, 0x67, 0x22, 0x0f, 0xa9, 0xff, 0x11, 0x00,
+	0x31, 0x44, 0xf0, 0xef,
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+	0xf0, 0xef, 0x00, 0xe0,
+	0xed, 0x75, 0x32, 0x0f, 0x30, 0x00, 0xed, 0xdc, 0x32, 0x76, 0x00, 0xee,
+	0xff, 0x32, 0x43, 0x0e,
+	0x10, 0x20, 0x0c, 0xfe, 0x10, 0x31, 0x52, 0x33, 0xfc, 0xdb, 0x32, 0x66,
+	0x00, 0x00, 0x31, 0xff,
+	0x22, 0x44, 0xbc, 0xde, 0x57, 0x53, 0x00, 0x00, 0xff, 0xef, 0x34, 0x22,
+	0xf0, 0xf0, 0xe0, 0x01,
+	0x56, 0x11, 0xf0, 0xcd, 0xfe, 0x00, 0x11, 0x22, 0x00, 0xee, 0xef, 0xf1,
+	0x27, 0x32, 0xf0, 0xa0,
+	0xe0, 0xe0, 0xc0, 0xe1, 0xf2, 0xef, 0xdd, 0x54, 0x35, 0xcc, 0xee, 0x20,
+	0x00, 0xd9, 0x71, 0x53,
+	0x00, 0xfd, 0x11, 0x11, 0x0f, 0x1f, 0x00, 0xfb, 0xfe, 0x20, 0x0e, 0x0e,
+	0x0f, 0x0e, 0xed, 0xd9,
+	0x71, 0x64, 0x0f, 0xec, 0x00, 0x00, 0x0f, 0x0f, 0x10, 0x1f, 0xec, 0x00,
+	0x00, 0x9b, 0xdd, 0x12,
+	0x54, 0x0f, 0xca, 0x00, 0x12, 0x00, 0x0f, 0x00, 0xf0, 0xcd, 0xef, 0xf0,
+	0x00, 0xff, 0xe0, 0xcc,
+	0xde, 0xf0, 0xd0, 0x9b, 0xf0, 0x15, 0x01, 0xf0, 0x01, 0xd0, 0xd0, 0x05,
+	0x12, 0x00, 0x0f, 0xee,
+	0x11, 0x77, 0x40, 0xff, 0x10, 0x33, 0x10, 0xee, 0xde, 0x75, 0x11, 0xee,
+	0xef, 0xf0, 0xff, 0xf0,
+	0xe0, 0xf0, 0x0f, 0xed, 0xbb, 0x00, 0x24, 0x43, 0x0f, 0xfc, 0x40, 0x63,
+	0x15, 0x10, 0x0d, 0x00,
+	0x45, 0x21, 0x10, 0x00, 0xdc, 0x22, 0x66, 0xff, 0xee, 0x00, 0xf0, 0xce,
+	0xff, 0x11, 0xff, 0xf0,
+	0x00, 0xb9, 0xfe, 0x00, 0x0f, 0x1f, 0x0e, 0x89, 0xee, 0x00, 0x22, 0x00,
+	0xdd, 0xf0, 0x02, 0x03,
+	0xef, 0xcf, 0xce, 0x05, 0x26, 0xe0, 0xaf, 0x00, 0x00, 0xff, 0xee, 0x02,
+	0x12, 0x11, 0xff, 0xfe,
+	0x0e, 0x0f, 0xeb, 0x00, 0x75, 0x10, 0xcd, 0x00, 0xee, 0x88, 0xff, 0x00,
+	0x00, 0x0f, 0xff, 0x0d,
+	0xfd, 0x0f, 0x11, 0x20, 0x0c, 0xec, 0x71, 0x33, 0x0d, 0x20, 0x34, 0x10,
+	0x10, 0x43, 0x12, 0xfd,
+	0xe9, 0x31, 0x76, 0x11, 0x21, 0x33, 0x11, 0x00, 0x11, 0x52, 0x0f, 0xfe,
+	0x24, 0x55, 0x00, 0x10,
+	0x33, 0xf0, 0xef, 0xe1, 0x26, 0x32, 0xc0, 0xcf, 0xbf, 0xf0, 0x27, 0x12,
+	0xcf, 0x00, 0x21, 0x00,
+	0x0f, 0x40, 0x01, 0x00, 0x41, 0x54, 0x24, 0x21, 0x00, 0x21, 0x23, 0x11,
+	0x00, 0x21, 0x01, 0x04,
+	0x67, 0x22, 0xff, 0xef, 0x0e, 0xfd, 0xde, 0xd1, 0x02, 0x00, 0x1e, 0x1c,
+	0x70, 0x66, 0x00, 0x0c,
+	0x00, 0x10, 0x20, 0x11, 0x00, 0x0d, 0x53, 0x35, 0x00, 0x13, 0x23, 0xa9,
+	0xee, 0x21, 0x46, 0x00,
+	0xf0, 0x30, 0x20, 0xde, 0xef, 0x76, 0x31, 0xed, 0xef, 0x20, 0x12, 0xff,
+	0xff, 0x00, 0x01, 0x56,
+	0xe0, 0x8f, 0x00, 0x02, 0xf0, 0xf0, 0xf2, 0xf0, 0xf0, 0x00, 0x00, 0x0e,
+	0x25, 0x00, 0x01, 0x24,
+	0xef, 0xcd, 0x74, 0x12, 0xca, 0x00, 0x16, 0xf0, 0xce, 0x00, 0xef, 0xf0,
+	0x45, 0x00, 0xae, 0x00,
+	0x12, 0x10, 0x10, 0x10, 0x21, 0x23, 0x11, 0x43, 0x43, 0x0f, 0x10, 0x22,
+	0x43, 0x2e, 0x0c, 0x0e,
+	0xfa, 0x21, 0x66, 0x00, 0x02, 0x35, 0x10, 0xf0, 0xe0, 0xef, 0xf0, 0x43,
+	0x74, 0x21, 0xed, 0xca,
+	0x00, 0x10, 0x00, 0x10, 0x42, 0x01, 0xf0, 0xf0, 0xef, 0x24, 0x55, 0x00,
+	0xfe, 0x64, 0x11, 0x0f,
+	0x00, 0x0f, 0xef, 0x23, 0x35, 0xef, 0xde, 0x62, 0x21, 0x0f, 0x0e, 0xed,
+	0xff, 0x37, 0x64, 0x0f,
+	0x0e, 0xfc, 0xff, 0x30, 0x0d, 0xdd, 0x30, 0x1f, 0xb8, 0xf0, 0x20, 0xf0,
+	0xcc, 0xff, 0xff, 0x11,
+	0x42, 0xcf, 0xcc, 0x21, 0x23, 0xbd, 0xef, 0x03, 0x07, 0xe1, 0x8f, 0xff,
+	0x03, 0xf3, 0xe0, 0xf2,
+	0x04, 0xe2, 0xae, 0xfd, 0xf0, 0xe1, 0x17, 0x01, 0xdd, 0xf0, 0x57, 0x00,
+	0xbd, 0x0f, 0x13, 0x22,
+	0x0f, 0xf0, 0xef, 0xcf, 0x02, 0xf0, 0xcd, 0x03, 0xf0, 0xb8, 0x00, 0x23,
+	0xf0, 0x00, 0x00, 0x00,
+	0x22, 0xee, 0xea, 0x71, 0x31, 0xda, 0xff, 0x32, 0x00, 0xca, 0xff, 0x1f,
+	0x10, 0xed, 0xcd, 0x1f,
+	0x0e, 0xab, 0x0e, 0x2f, 0x0c, 0xfe, 0xff, 0xec, 0x3f, 0x4f, 0xdb, 0xde,
+	0x3f, 0x1e, 0x0e, 0x71,
+	0x20, 0xa9, 0xff, 0xf0, 0x00, 0x20, 0x00, 0xdf, 0xff, 0xde, 0xbc, 0xf0,
+	0x13, 0xf0, 0xcc, 0x0f,
+	0xed, 0xcd, 0xde, 0xab, 0xff, 0x24, 0x00, 0xbb, 0xf0, 0x00, 0xee, 0x00,
+	0x44, 0x00, 0xd0, 0xfe,
+	0xfe, 0x22, 0x13, 0xdf, 0xf0, 0x03, 0xe2, 0xd0, 0xf0, 0x23, 0xac, 0xef,
+	0x16, 0x43, 0x01, 0x8a,
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+	0x31, 0x00, 0x00, 0x53,
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+	0x52, 0x00, 0xee, 0xef,
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+	0x01, 0xbe, 0x0e, 0x00,
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+	0x0f, 0x1f, 0x0f, 0x0f,
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+	0x00, 0x35, 0x0b, 0xee,
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+	0x0f, 0x0a, 0x22, 0x01,
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+	0xfd, 0x36, 0x22, 0xee,
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+	0x24, 0xf0, 0xde, 0x00,
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+	0x70, 0x20, 0xf0, 0x00,
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+	0x20, 0x53, 0x00, 0x0c,
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+	0xe0, 0xe0, 0x1f, 0x10,
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+	0xe3, 0x02, 0xb8, 0xf0,
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+	0xcb, 0xcf, 0x10, 0x13,
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+	0x44, 0xf0, 0x8e, 0x0f,
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+	0x50, 0x00, 0xab, 0xe0,
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+	0x03, 0x00, 0x0d, 0x00,
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+	0x10, 0x32, 0xe0, 0xd0,
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+	0x24, 0x0a, 0xfc, 0x0f,
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+	0x21, 0x3f, 0x0e, 0xf2,
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+	0x02, 0x77, 0x00, 0x01,
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+	0xa0, 0x27, 0x34, 0xec,
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+	0xed, 0xfd, 0x06, 0x33,
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+	0x3f, 0x1f, 0xf3, 0xa7,
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+	0xe0, 0xee, 0x00, 0x12,
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+	0xd0, 0x00, 0x3d, 0x02,
+	0xc1, 0xec, 0x1e, 0x1b, 0xf0, 0xf7, 0x57, 0x20, 0xfe, 0xc0, 0x21, 0xff,
+	0x0c, 0x64, 0xf1, 0xe0,
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+	0xf0, 0xf0, 0xed, 0x0d,
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+	0x20, 0x21, 0x00, 0x00,
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+	0xf0, 0x12, 0x52, 0xf0,
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+	0x41, 0xee, 0xed, 0x26,
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+	0xe0, 0x06, 0x32, 0xed,
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+	0x00, 0x70, 0x10, 0xbc,
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+	0x01, 0xff, 0x0f, 0x42,
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+	0x22, 0x00, 0xc0, 0x0f,
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+	0xcf, 0xeb, 0xe0, 0x24,
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+	0x1d, 0x03, 0xf3, 0x11,
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+	0x0f, 0x33, 0x00, 0xf2,
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+	0x02, 0x1f, 0xfe, 0xee,
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+	0x21, 0x12, 0xf3, 0x01,
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+	0xfc, 0x03, 0xd1, 0xdd,
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+	0x04, 0x0f, 0x00, 0x13,
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+	0x0d, 0xfc, 0x2f, 0x61,
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+	0x34, 0x25, 0xdf, 0x8f,
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+	0xe0, 0x72, 0x13, 0xde,
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+	0x75, 0x3f, 0x0e, 0xec,
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+	0x64, 0x11, 0xcc, 0xe0,
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+	0xe3, 0xde, 0x10, 0x11,
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+	0x0f, 0xb8, 0x00, 0x20,
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+	0xca, 0xbf, 0x00, 0x30,
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+	0xfc, 0xc0, 0x13, 0x32,
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+	0xfe, 0x00, 0xe0, 0x12,
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+	0x27, 0x1e, 0xfb, 0x01,
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+	0x21, 0xe3, 0xe4, 0x11,
+	0x71, 0x1f, 0x99, 0xff, 0xf0, 0x22, 0x11, 0x13, 0x00, 0xfa, 0x0e, 0x2f,
+	0x20, 0xd2, 0xf2, 0x33,
+	0x50, 0x0e, 0xdd, 0xff, 0xf0, 0x22, 0x47, 0x03, 0x05, 0xb0, 0x9c, 0x00,
+	0x31, 0x02, 0xf1, 0x02,
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+	0x00, 0x00, 0xed, 0xde,
+	0x13, 0xf0, 0xd8, 0x4f, 0xf0, 0xa0, 0xf3, 0x14, 0x5f, 0x1c, 0xe0, 0xf3,
+	0x36, 0x62, 0xfe, 0xed,
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+	0x0f, 0xec, 0xe0, 0xa0,
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+	0x3f, 0xf0, 0x9c, 0xe0,
+	0x02, 0x45, 0x20, 0xfd, 0xcc, 0xe0, 0x00, 0x46, 0x10, 0xfd, 0xb0, 0x0f,
+	0x20, 0x20, 0x1d, 0x00,
+	0x20, 0xfd, 0xc0, 0x05, 0xf1, 0x3f, 0x5e, 0xed, 0x05, 0x05, 0x0f, 0xac,
+	0x20, 0x72, 0x04, 0xf0,
+	0xe0, 0x00, 0x11, 0x0c, 0xd0, 0x35, 0x51, 0xe0, 0xc0, 0xf0, 0x01, 0x01,
+	0x27, 0x00, 0x13, 0x00,
+	0x74, 0xf4, 0x19, 0xc5, 0x7e, 0xf2, 0x69, 0xc3, 0xf9, 0x8f, 0xee, 0x16,
+	0x03, 0xe0, 0xeb, 0xec,
+	0xf0, 0x02, 0x44, 0xf2, 0xe2, 0xaf, 0x0d, 0x53, 0x02, 0xfc, 0x2e, 0x4e,
+	0x10, 0x07, 0x0f, 0x3d,
+	0x30, 0x0e, 0xcb, 0x11, 0x34, 0xbe, 0xcd, 0xf0, 0xf4, 0x00, 0x41, 0x01,
+	0xa0, 0xfc, 0x21, 0x77,
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+	0x00, 0x90, 0x01, 0x44,
+	0x10, 0xd4, 0xdf, 0xfc, 0x3f, 0x50, 0xf2, 0xf5, 0x03, 0xfd, 0xdf, 0x16,
+	0x22, 0xfd, 0xdb, 0x71,
+	0x32, 0x02, 0xd1, 0xee, 0x0a, 0x20, 0x01, 0xf2, 0x00, 0x20, 0x8a, 0xdf,
+	0xe0, 0x10, 0x0f, 0xe0,
+	0x01, 0x05, 0x01, 0xd0, 0xde, 0x10, 0x1f, 0x10, 0x76, 0x11, 0xae, 0xfc,
+	0x00, 0x25, 0x00, 0x10,
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+	0x2b, 0x20, 0x30, 0x21,
+	0xd1, 0xd0, 0x7d, 0x3e, 0x33, 0xe3, 0xa0, 0x03, 0x2b, 0xc8, 0x0d, 0x34,
+	0xf4, 0xc1, 0x1e, 0x1d,
+	0x0e, 0xf0, 0x14, 0x30, 0x51, 0x10, 0x0c, 0xf0, 0x00, 0x04, 0x01, 0x0e,
+	0xfe, 0x35, 0x91, 0xaf,
+	0x1e, 0x52, 0x00, 0xe0, 0x03, 0x06, 0xef, 0xee, 0x20, 0x33, 0x00, 0x10,
+	0x0c, 0x01, 0xe6, 0xbf,
+	0x0d, 0x30, 0x10, 0xc2, 0xcf, 0x09, 0x10, 0x1f, 0x03, 0xe4, 0xe3, 0x00,
+	0x3b, 0x70, 0x02, 0xf2,
+	0xe0, 0x0d, 0x4f, 0x40, 0x95, 0xee, 0x1b, 0x50, 0x0e, 0xd1, 0x14, 0x1d,
+	0x2e, 0xff, 0xbe, 0x12,
+	0x33, 0x23, 0x27, 0x03, 0xde, 0xcf, 0x0f, 0x0e, 0x33, 0x15, 0x8f, 0xef,
+	0x20, 0x00, 0xfd, 0xfe,
+	0x31, 0x50, 0x04, 0xf5, 0x01, 0x30, 0xfc, 0xdf, 0x16, 0x20, 0xef, 0xcb,
+	0x30, 0x41, 0xf1, 0xf1,
+	0x00, 0x1e, 0x0e, 0xc1, 0x14, 0x33, 0x00, 0x4c, 0x10, 0xa2, 0xd2, 0x00,
+	0x2c, 0xdc, 0xb0, 0x00,
+	0x61, 0x03, 0xdf, 0x98, 0x0d, 0x10, 0x10, 0x15, 0x04, 0xde, 0xdb, 0xff,
+	0x00, 0x03, 0x0f, 0x5f,
+	0x10, 0x21, 0xf0, 0xbd, 0x10, 0x20, 0x30, 0x13, 0x52, 0xe8, 0xbd, 0x0f,
+	0x12, 0x00, 0x0f, 0xde,
+	0x01, 0x34, 0x03, 0xfe, 0x2c, 0x30, 0xc1, 0xf4, 0x05, 0xc0, 0xe1, 0x30,
+	0x70, 0x22, 0x21, 0xe0,
+	0xb0, 0x01, 0x0e, 0x18, 0xef, 0x00, 0x75, 0x20, 0xca, 0x01, 0x02, 0x0f,
+	0xbe, 0x20, 0x23, 0x10,
+	0xde, 0x00, 0x00, 0xf0, 0xd0, 0xff, 0xed, 0x0e, 0x20, 0x77, 0x46, 0xce,
+	0xf0, 0x00, 0x43, 0x44,
+	0xef, 0xdd, 0x10, 0x4f, 0x10, 0xe2, 0xf4, 0xe0, 0xf8, 0x00, 0x0f, 0x20,
+	0xe2, 0xe0, 0x00, 0x1f,
+	0x00, 0x03, 0x70, 0xed, 0xef, 0x26, 0x34, 0xee, 0xdd, 0xf0, 0xc3, 0x20,
+	0x7f, 0x00, 0x01, 0xfc,
+	0xfe, 0x00, 0x50, 0x15, 0x02, 0x0d, 0x3c, 0x0f, 0xe0, 0x10, 0x24, 0xaf,
+	0xfa, 0xf0, 0xcf, 0x00,
+	0x44, 0xde, 0x9c, 0x02, 0x26, 0xf0, 0x0f, 0x40, 0x00, 0xd2, 0x11, 0x0d,
+	0xbc, 0x01, 0x54, 0x20,
+	0x0f, 0xd0, 0xef, 0x10, 0x73, 0x31, 0xdf, 0xdb, 0xfe, 0x20, 0x41, 0x06,
+	0xcf, 0x0d, 0x0f, 0x1e,
+	0x20, 0x00, 0x01, 0x00, 0x0b, 0xf0, 0x0f, 0x42, 0x64, 0x03, 0xa1, 0x04,
+	0x00, 0x2a, 0x02, 0x12,
+	0xf0, 0x0f, 0xf0, 0xc0, 0x00, 0x40, 0x0f, 0xde, 0x01, 0x47, 0x06, 0xdf,
+	0xff, 0x2f, 0x21, 0xe2,
+	0xef, 0x0f, 0xd3, 0x05, 0x00, 0xbc, 0xbd, 0x00, 0x53, 0x12, 0x1d, 0x4d,
+	0x32, 0x05, 0xb2, 0x00,
+	0x0a, 0x40, 0x10, 0xf3, 0xcf, 0x0e, 0x0c, 0xdf, 0xaf, 0x04, 0x33, 0x11,
+	0xf1, 0xdf, 0x3e, 0x61,
+	0x05, 0x04, 0xf1, 0x0d, 0x0c, 0x3d, 0x61, 0x05, 0x10, 0xea, 0xee, 0x00,
+	0x11, 0x31, 0x3f, 0xdf,
+	0xc2, 0xe5, 0x01, 0x30, 0xec, 0xd0, 0x03, 0x53, 0xcf, 0xfd, 0x20, 0x10,
+	0x0e, 0x7d, 0x21, 0xd4,
+	0xf2, 0x0d, 0xeb, 0x04, 0x57, 0xe0, 0xde, 0xe0, 0xbf, 0x02, 0x71, 0x0c,
+	0xce, 0xfc, 0xfc, 0x30,
+	0x47, 0xf0, 0xa0, 0xff, 0x0e, 0x0f, 0x00, 0x34, 0x03, 0x00, 0xec, 0xe0,
+	0xe1, 0x0f, 0xf0, 0xd3,
+	0x02, 0x70, 0x1e, 0xf0, 0xd2, 0x20, 0x0d, 0x10, 0x36, 0x02, 0xdd, 0x0b,
+	0x2f, 0x20, 0x14, 0x10,
+	0x7c, 0x2d, 0xfd, 0x0f, 0x42, 0x07, 0xb1, 0xdd, 0x2d, 0x71, 0x53, 0xef,
+	0x0d, 0x0e, 0xfe, 0xd5,
+	0x46, 0x40, 0x1f, 0xee, 0x00, 0x04, 0x0f, 0xdf, 0x21, 0x66, 0x00, 0xde,
+	0xff, 0x00, 0x12, 0x01,
+	0xe4, 0xe0, 0x10, 0x35, 0xcd, 0xcf, 0x10, 0x45, 0x1f, 0xbe, 0xef, 0x10,
+	0x50, 0x00, 0xbf, 0xfe,
+	0xf0, 0x1f, 0x31, 0x05, 0xeb, 0xdd, 0xf0, 0x11, 0x45, 0xff, 0xed, 0x00,
+	0x03, 0x52, 0x03, 0x9d,
+	0x08, 0x30, 0x44, 0xd2, 0xbf, 0x0c, 0x1f, 0x20, 0x11, 0xe0, 0xdc, 0x0f,
+	0x1f, 0x71, 0x13, 0xff,
+	0xf3, 0x17, 0x0e, 0xff, 0xf0, 0x02, 0x41, 0x23, 0x32, 0x0d, 0xbf, 0x04,
+	0x03, 0x4c, 0xfc, 0xdd,
+	0xfc, 0x10, 0x52, 0x06, 0xc0, 0xd0, 0xed, 0x10, 0x0d, 0x09, 0xf0, 0x33,
+	0x33, 0xde, 0xbd, 0xe0,
+	0xfb, 0x10, 0xf2, 0x05, 0x40, 0xcf, 0xbe, 0x40, 0x40, 0x13, 0xe0, 0xd0,
+	0xe0, 0xf3, 0x21, 0x71,
+	0x30, 0xee, 0xdc, 0x00, 0x10, 0x21, 0xe2, 0xd4, 0x05, 0x0f, 0x0b, 0xc0,
+	0xf0, 0x2f, 0x4d, 0x61,
+	0xf4, 0xc0, 0x00, 0x4d, 0xe8, 0xec, 0x41, 0x12, 0xf0, 0xdf, 0x1f, 0x30,
+	0x00, 0xdf, 0xbe, 0xcf,
+	0x0c, 0x5d, 0x12, 0xd7, 0x07, 0x20, 0x4f, 0x0f, 0x00, 0x00, 0x34, 0x33,
+	0xff, 0xfa, 0xf0, 0x10,
+	0x50, 0x10, 0xf2, 0x84, 0x0f, 0xfd, 0x0f, 0x40, 0x10, 0x04, 0xb0, 0xfd,
+	0xfa, 0x21, 0x54, 0x00,
+	0xcd, 0x00, 0xf0, 0x11, 0x30, 0x00, 0x03, 0xb3, 0xfe, 0x0e, 0x06, 0x02,
+	0x23, 0xfe, 0xdd, 0x63,
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+	0x0c, 0x0c, 0xe0, 0xb0,
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+	0xf3, 0xf4, 0x02, 0x23,
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+	0xe1, 0xf0, 0xfc, 0x0d,
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+	0x24, 0x02, 0xfe, 0xfc,
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+	0x3b, 0x10, 0xd7, 0xd2,
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+	0x06, 0x22, 0x50, 0xcc,
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+	0x11, 0x12, 0xe0, 0xfe,
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+	0x70, 0xe0, 0xd0, 0x00,
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+	0xff, 0x02, 0x01, 0xae,
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+	0x22, 0x47, 0x01, 0xfe,
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+	0x71, 0x30, 0xe0, 0xf0,
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+	0xd1, 0xb2, 0xd0, 0xe0,
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+	0xef, 0xff, 0x0e, 0x30,
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+	0x20, 0x10, 0xf0, 0xf0,
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+	0x10, 0x24, 0xd0, 0x90,
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+	0x1c, 0x50, 0x10, 0x0c,
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+	0x1c, 0x20, 0x11, 0xf4,
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+	0xed, 0xc0, 0xe1, 0xd0,
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+	0xf0, 0x07, 0x56, 0x3e,
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+	0x23, 0x00, 0x3a, 0x00,
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+	0xff, 0x0e, 0x40, 0x50,
+	0x11, 0x00, 0x00, 0xfd, 0xd0, 0xd3, 0xe7, 0x32, 0x70, 0x30, 0xb3, 0xe5,
+	0x00, 0x1e, 0x0e, 0x32,
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+	0xef, 0xee, 0x02, 0x02,
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+	0x10, 0xf6, 0xc3, 0x10,
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+	0x30, 0x00, 0x02, 0x1f,
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+	0x0f, 0x1c, 0x1e, 0x17,
+	0x27, 0x03, 0x0e, 0x2d, 0x21, 0xb4, 0xd0, 0x90, 0xe0, 0x0d, 0x30, 0x73,
+	0x01, 0xef, 0xff, 0x0f,
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+	0xf1, 0xde, 0xfb, 0x1f,
+	0xff, 0xf0, 0x73, 0x30, 0x1f, 0x40, 0x31, 0x23, 0xf2, 0xc5, 0x00, 0xfd,
+	0x32, 0x14, 0xc0, 0xce,
+	0xef, 0xe4, 0xa3, 0xf0, 0x0d, 0xcf, 0xfc, 0x32, 0x77, 0x14, 0x20, 0x00,
+	0x1d, 0x40, 0x00, 0x22,
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+	0xf2, 0xd2, 0xe2, 0x12,
+	0x40, 0x0c, 0xf0, 0xde, 0xff, 0xd0, 0xe0, 0x1d, 0x7c, 0x76, 0x02, 0xed,
+	0x0c, 0x00, 0x33, 0x25,
+	0x22, 0x00, 0xdd, 0x0f, 0x0e, 0x3e, 0x3e, 0x1d, 0x0f, 0xa2, 0xc7, 0xf0,
+	0x5d, 0x73, 0x57, 0xf2,
+	0xdd, 0xfc, 0x10, 0x32, 0x21, 0xe1, 0xcf, 0xf0, 0x03, 0x04, 0x00, 0xef,
+	0xce, 0xef, 0x0f, 0x0a,
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+	0xca, 0xdf, 0x1e, 0x20,
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+	0x10, 0x73, 0x55, 0x00,
+	0x11, 0xf1, 0x00, 0x00, 0x15, 0x12, 0x0d, 0x2e, 0x01, 0xb4, 0xde, 0xfa,
+	0x0f, 0xfe, 0xc0, 0x01,
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+	0xf1, 0xf0, 0x0e, 0xf0,
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+	0x74, 0x70, 0x0f, 0xfe,
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+	0x1f, 0x21, 0xd0, 0xaf,
+	0x06, 0x41, 0x0d, 0x2b, 0x20, 0x34, 0x01, 0x0e, 0x1a, 0x50, 0x00, 0xd0,
+	0xbe, 0x01, 0x44, 0x51,
+	0x0f, 0xcf, 0xf0, 0xf1, 0x07, 0xe1, 0xe0, 0x1c, 0x3f, 0x00, 0x02, 0x30,
+	0xfe, 0x00, 0x00, 0x3e,
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+	0x1f, 0x2f, 0x04, 0xe2,
+	0xed, 0x0f, 0x4d, 0xeb, 0x01, 0x65, 0x31, 0x1f, 0x0d, 0x0f, 0xd0, 0x05,
+	0x10, 0x49, 0x0f, 0xb1,
+	0xc3, 0x02, 0x1d, 0x1d, 0xc1, 0xc7, 0xe1, 0xe0, 0x2e, 0x52, 0x02, 0xd1,
+	0xd2, 0x0f, 0x5c, 0x42,
+	0x13, 0x0d, 0x0a, 0xe3, 0x13, 0x70, 0x1c, 0xf0, 0xc0, 0x00, 0x0c, 0xdc,
+	0xef, 0x1d, 0x20, 0xfd,
+	0x99, 0x0e, 0xef, 0x00, 0x40, 0x1c, 0x00, 0x05, 0x14, 0x20, 0x0c, 0x2f,
+	0x61, 0x16, 0x11, 0xff,
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+	0x6d, 0x70, 0x10, 0x10,
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+	0xe1, 0x00, 0x46, 0x1f,
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+	0xed, 0xde, 0x35, 0x22,
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+	0x24, 0x42, 0x00, 0x0f,
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+	0xe3, 0x14, 0xfd, 0x8a,
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+	0x11, 0x02, 0x00, 0xbd,
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+	0xdd, 0xbe, 0xcc, 0xe0,
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+	0x30, 0xfe, 0xfe, 0x40,
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+	0xc0, 0x02, 0x14, 0xb0,
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+	0x10, 0x00, 0x10, 0x00,
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+	0x12, 0x35, 0xe0, 0xbd,
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+	0x40, 0x31, 0x01, 0x55,
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+	0xdf, 0x00, 0x03, 0xfd,
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+	0xee, 0x8b, 0x01, 0xf0,
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+	0x30, 0x2e, 0x10, 0x24,
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+	0x0f, 0x41, 0x01, 0xdc,
+	0xce, 0x1f, 0x21, 0x02, 0x0f, 0x0f, 0x01, 0x00, 0x02, 0x31, 0x01, 0xf0,
+	0xc2, 0xf6, 0xe0, 0x00,
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+	0xfe, 0xde, 0x02, 0x40,
+	0x0b, 0xfe, 0x13, 0x34, 0xfe, 0xa8, 0x00, 0x60, 0x00, 0xff, 0xcf, 0x03,
+	0x25, 0x0f, 0x00, 0x32,
+	0xff, 0xcc, 0x13, 0x54, 0x0f, 0xfd, 0x55, 0x42, 0x02, 0x00, 0x10, 0x0d,
+	0x00, 0x57, 0x13, 0x0c,
+	0xff, 0xe1, 0x04, 0x01, 0x0e, 0xde, 0x0e, 0x74, 0x03, 0xdc, 0xf0, 0x11,
+	0xfe, 0xdc, 0x31, 0x56,
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+	0x33, 0x05, 0x02, 0xf3,
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+	0xfa, 0x10, 0x04, 0x03,
+	0x01, 0xce, 0xdf, 0xcf, 0xf0, 0xf1, 0x12, 0xd0, 0xba, 0xdc, 0xed, 0x3f,
+	0x0e, 0xbe, 0x10, 0x23,
+	0xde, 0xdb, 0xdf, 0x0e, 0x43, 0x15, 0xd0, 0x00, 0x54, 0x0e, 0x2e, 0xed,
+	0xd8, 0x0e, 0x00, 0x10,
+	0x10, 0xdf, 0x0c, 0x64, 0x37, 0xf0, 0xb0, 0x0f, 0x01, 0x24, 0x00, 0x2e,
+	0x0f, 0xe0, 0xe0, 0x00,
+	0x2d, 0x1f, 0x12, 0x11, 0xfd, 0xce, 0xf1, 0xd1, 0xf4, 0x03, 0x15, 0xe0,
+	0xad, 0xee, 0xed, 0xea,
+	0x0f, 0x3f, 0x3d, 0x2f, 0x10, 0xff, 0x1f, 0x10, 0x00, 0x15, 0x55, 0x00,
+	0xef, 0xff, 0x25, 0x22,
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+	0x10, 0x0b, 0x00, 0xe1,
+	0xbf, 0xf0, 0xf0, 0xd0, 0xb8, 0xed, 0xf0, 0x00, 0x21, 0xdd, 0xf0, 0x33,
+	0xfd, 0xea, 0x00, 0x30,
+	0x24, 0xde, 0xcc, 0x00, 0x1f, 0x20, 0x0e, 0xf0, 0x0f, 0xdb, 0x00, 0x54,
+	0x22, 0x32, 0x44, 0x26,
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+	0xf0, 0xe1, 0xef, 0x00,
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+	0x22, 0xf0, 0x0c, 0x30,
+	0x20, 0xff, 0x00, 0x63, 0x44, 0x13, 0x10, 0x0e, 0x3d, 0x20, 0x10, 0xfe,
+	0xc8, 0xde, 0xe0, 0x0f,
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+	0xd0, 0xde, 0xbd, 0xdc,
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+	0xcd, 0x00, 0xf1, 0xed,
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+	0x55, 0x0f, 0x0c, 0x01,
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+	0xaa, 0x00, 0x03, 0x00,
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+	0x3d, 0x2f, 0x65, 0x00,
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+	0x33, 0xeb, 0xd0, 0xf1,
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+	0x27, 0x67, 0xdf, 0xef,
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+	0xf0, 0x33, 0x21, 0x0f,
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+	0x24, 0xef, 0xc0, 0x0d,
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+	0xfa, 0xde, 0xff, 0xf0,
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+	0xce, 0xed, 0xf4, 0xff,
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+	0xde, 0x0f, 0xf0, 0x12,
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+	0xfc, 0xf0, 0x34, 0x77,
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+	0x23, 0x43, 0xeb, 0xcc,
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+	0x00, 0xff, 0xfd, 0x10,
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+	0xdf, 0xcb, 0x31, 0x73,
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+	0x03, 0x03, 0xef, 0xca,
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+	0xde, 0x0f, 0xfe, 0x0f,
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+	0x20, 0x51, 0xef, 0xca,
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+	0x13, 0xed, 0xf0, 0x32,
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+	0xfd, 0x00, 0x33, 0xfd,
+	0xde, 0x23, 0x57, 0x03, 0xae, 0x0f, 0x12, 0x27, 0x01, 0xf0, 0x0e, 0x52,
+	0x00, 0xc8, 0x20, 0x50,
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+	0x02, 0x21, 0xd5, 0xb0,
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+	0x12, 0x47, 0x12, 0x04,
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+	0x12, 0x2f, 0xe8, 0x10,
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+	0x00, 0xdf, 0x00, 0x21,
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+	0x00, 0xff, 0xf0, 0x02,
+	0x22, 0x11, 0x0f, 0xeb, 0xe0, 0x25, 0x73, 0xed, 0xec, 0x02, 0x01, 0xcf,
+	0xee, 0x24, 0x00, 0x89,
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+	0x1d, 0x0c, 0xcc, 0x12,
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+	0x0d, 0x9a, 0x00, 0x01,
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+	0xe0, 0x9a, 0x00, 0x50,
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+	0xed, 0x0e, 0x27, 0x13,
+	0x10, 0x0f, 0xca, 0xbf, 0xf0, 0x0e, 0xe0, 0xf0, 0xcf, 0x02, 0xdf, 0xf0,
+	0xf6, 0xbf, 0xf0, 0x14,
+	0x12, 0xff, 0xae, 0xf0, 0xf7, 0x01, 0x01, 0xf1, 0xbd, 0xfa, 0x20, 0x76,
+	0x21, 0x02, 0xfe, 0xff,
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+	0x0f, 0x20, 0xdb, 0xde,
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+	0x0b, 0xcc, 0xcd, 0x00,
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+	0x00, 0x20, 0x32, 0x00,
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+	0x0e, 0xfe, 0x00, 0x0e,
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+	0xff, 0x26, 0x00, 0x9f,
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+	0xf0, 0x14, 0x1f, 0xb8,
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+	0xff, 0xdc, 0x10, 0x21,
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+	0x02, 0xfe, 0x1d, 0x42,
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+	0x11, 0xde, 0xbc, 0xef,
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+	0xfe, 0x01, 0x07, 0xf0,
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+	0x01, 0x23, 0x12, 0x32,
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+	0x06, 0x00, 0x88, 0x00,
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+	0x63, 0x0d, 0xd8, 0x0f,
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+	0x21, 0xbc, 0xfc, 0x21,
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+	0x22, 0x0f, 0x1d, 0x56,
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+	0xef, 0xcd, 0x0f, 0x31,
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+	0x02, 0x20, 0x00, 0x0e,
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+	0x42, 0x43, 0xfa, 0xdc,
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+	0xe1, 0x04, 0xbe, 0x00,
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+	0x36, 0x30, 0x00, 0x11,
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+	0xdc, 0x53, 0x10, 0xe8,
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+	0x02, 0x0f, 0xce, 0xff,
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+	0xfd, 0xe0, 0xf0, 0xe0,
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+	0x31, 0x33, 0x00, 0xce,
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+	0x01, 0xf0, 0xdc, 0xfe,
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+	0x27, 0x42, 0x0f, 0xcb,
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+	0xe1, 0xf3, 0xef, 0xf0,
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+	0xf0, 0x65, 0x11, 0x0f,
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+	0xdf, 0x0f, 0x02, 0x40,
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+	0xff, 0x0d, 0x00, 0x0c,
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+	0x32, 0x25, 0x31, 0xf0,
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+	0x03, 0xb2, 0x02, 0x20,
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+	0xdf, 0xcf, 0x11, 0x04,
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+	0x32, 0x65, 0x10, 0x10,
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+	0xf0, 0x0f, 0x50, 0x22,
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+	0x01, 0x03, 0xff, 0x21,
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+	0x0f, 0x10, 0x0f, 0x0f,
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+	0x32, 0x26, 0xdf, 0x01,
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+	0x11, 0x12, 0x00, 0xeb,
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+	0x10, 0x2f, 0xdb, 0xee,
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+	0x10, 0x00, 0x10, 0x00,
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+	0x0e, 0xf1, 0x00, 0xff,
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+	0x0f, 0xde, 0x01, 0x77,
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+	0x64, 0x01, 0x20, 0x12,
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+	0xd0, 0xcc, 0xe0, 0xde,
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+	0x43, 0x0e, 0xe8, 0xce,
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+	0xff, 0xe0, 0x0f, 0xce,
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+	0xef, 0x11, 0x56, 0xf0,
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+	0xec, 0xee, 0x30, 0x65,
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+	0x46, 0x10, 0xde, 0xb9,
+	0x10, 0x11, 0x10, 0xef, 0xff, 0x00, 0xce, 0xed, 0xdd, 0x11, 0x13, 0x00,
+	0xd0, 0x13, 0x0d, 0xeb,
+	0x02, 0x25, 0xff, 0xcc, 0x01, 0x02, 0xee, 0xfe, 0xef, 0xfe, 0xfe, 0x34,
+	0x56, 0x00, 0xde, 0x00,
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+	0x22, 0x02, 0x11, 0x42,
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+	0xfd, 0x11, 0x00, 0xef,
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+	0x10, 0xee, 0xec, 0xed,
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+	0x03, 0xc1, 0xbb, 0xfe,
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+	0x33, 0x24, 0x1f, 0x0c,
+	0xff, 0x11, 0x46, 0x0e, 0xcb, 0x00, 0x53, 0x14, 0xff, 0xdc, 0x00, 0x43,
+	0x21, 0xcd, 0xfd, 0x01,
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+	0x20, 0x40, 0x15, 0x0e,
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+	0x02, 0x01, 0x3f, 0xea,
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+	0x04, 0x00, 0xf0, 0xfe,
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+	0x30, 0x00, 0x10, 0xfd,
+	0xf0, 0x44, 0x32, 0x02, 0x02, 0x33, 0x10, 0x00, 0x03, 0x52, 0x12, 0xe0,
+	0xc0, 0x0f, 0x01, 0x11,
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+	0xfc, 0xde, 0x20, 0x40,
+	0xc9, 0xbc, 0xf0, 0x22, 0x42, 0x00, 0x0f, 0x00, 0x23, 0x10, 0xfb, 0xdc,
+	0x40, 0x12, 0xd0, 0xff,
+	0x32, 0x21, 0xff, 0x8c, 0x0f, 0xf3, 0x05, 0x12, 0x00, 0xe2, 0x01, 0xe0,
+	0xf1, 0xf2, 0x02, 0x01,
+	0xcf, 0xf0, 0xf4, 0x33, 0xfb, 0xad, 0x13, 0x27, 0x10, 0xf1, 0xf1, 0x11,
+	0x20, 0xf2, 0xf1, 0x20,
+	0x42, 0xfd, 0xed, 0x20, 0x0c, 0x0c, 0x02, 0x52, 0xeb, 0xff, 0x21, 0x0e,
+	0x1b, 0x52, 0x11, 0x00,
+	0x09, 0x2f, 0x30, 0x70, 0xd0, 0xe0, 0x70, 0x41, 0xab, 0x1e, 0x11, 0x03,
+	0xf0, 0x02, 0x01, 0x0e,
+	0xfd, 0xff, 0x01, 0x14, 0x11, 0xe1, 0xe0, 0x01, 0x34, 0x0e, 0x99, 0x00,
+	0x11, 0x02, 0x21, 0x11,
+	0xff, 0x03, 0x23, 0x33, 0x47, 0x00, 0x0f, 0x0d, 0x20, 0x32, 0x01, 0x02,
+	0x63, 0x21, 0xed, 0x0f,
+	0x11, 0x56, 0x02, 0x00, 0x10, 0x64, 0xf0, 0xed, 0xf0, 0x04, 0xf0, 0xec,
+	0xbd, 0x00, 0xcd, 0xdc,
+	0x0f, 0xee, 0xf1, 0xf4, 0xd0, 0xcd, 0xfc, 0x20, 0x02, 0xf0, 0x0e, 0x2e,
+	0xdc, 0xfc, 0x1f, 0x73,
+	0x00, 0xf8, 0xee, 0x10, 0x31, 0x01, 0x10, 0x00, 0x53, 0x25, 0xef, 0xde,
+	0x11, 0x03, 0x23, 0x02,
+	0xd0, 0x0f, 0xcd, 0xfe, 0x0f, 0xdc, 0x0f, 0x00, 0x22, 0x11, 0x01, 0x10,
+	0x2f, 0xf0, 0xcf, 0x23,
+	0x36, 0x00, 0xcc, 0xd1, 0x13, 0xef, 0xab, 0x01, 0x24, 0xf0, 0xed, 0xe0,
+	0x45, 0x32, 0xf0, 0xde,
+	0x1f, 0x40, 0x34, 0x30, 0x01, 0x00, 0x01, 0x01, 0xfd, 0xff, 0x00, 0x22,
+	0xee, 0xf1, 0x14, 0x40,
+	0x2d, 0xf0, 0x00, 0x50, 0xfd, 0xf9, 0x00, 0x54, 0x0f, 0xdd, 0x10, 0xee,
+	0xde, 0x20, 0x03, 0xce,
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+	0xcb, 0xfe, 0x01, 0x56,
+	0xf0, 0xef, 0x00, 0x31, 0x04, 0x01, 0x02, 0x20, 0xf0, 0xfe, 0xef, 0x00,
+	0x04, 0x62, 0x10, 0x02,
+	0x01, 0x0f, 0x31, 0x17, 0x12, 0x32, 0x10, 0x1d, 0x03, 0x16, 0x21, 0xfe,
+	0xdf, 0x20, 0x22, 0xf0,
+	0x02, 0xf0, 0xb8, 0xf0, 0x00, 0x02, 0xcd, 0xdf, 0x13, 0x21, 0xed, 0x0b,
+	0x00, 0xec, 0x0d, 0x00,
+	0x20, 0x41, 0xde, 0xa9, 0x30, 0x00, 0xe1, 0x3e, 0x00, 0xe5, 0x01, 0xff,
+	0xda, 0xaf, 0x0f, 0x00,
+	0xf2, 0xdf, 0xfb, 0xc0, 0xc0, 0xed, 0x1f, 0x10, 0xde, 0x00, 0x16, 0x41,
+	0x10, 0xce, 0x00, 0x64,
+	0x02, 0x0e, 0xf9, 0xf0, 0x54, 0x10, 0xcf, 0x3e, 0x02, 0xf2, 0x00, 0x21,
+	0x23, 0xd2, 0x16, 0x42,
+	0xfe, 0xcb, 0x00, 0x01, 0x30, 0x51, 0xff, 0xdc, 0x0f, 0x4e, 0x10, 0x23,
+	0x03, 0xe0, 0xcb, 0xe0,
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+	0xec, 0x20, 0x43, 0x00,
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+	0x20, 0x43, 0xde, 0x10,
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+	0x12, 0x43, 0xb0, 0xf1,
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+	0x02, 0xf2, 0xce, 0xf0,
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+	0x11, 0x1f, 0x09, 0x10,
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+	0xf1, 0x14, 0x32, 0x20,
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+	0x11, 0x43, 0x12, 0x24,
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+	0xe0, 0xd0, 0xd1, 0xf1,
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+	0x14, 0x11, 0x33, 0xfd,
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+	0xe2, 0x01, 0x00, 0xf0,
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+	0x63, 0x0e, 0xff, 0x00,
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+	0xf2, 0xf4, 0xe1, 0xec,
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+	0xff, 0xfc, 0xfd, 0xef,
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+	0x10, 0x23, 0x31, 0xfd,
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+	0xcb, 0xe0, 0xf2, 0xdf,
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+	0x01, 0x25, 0x43, 0xf0,
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+	0x0c, 0x0e, 0x0e, 0x24,
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+	0x00, 0x00, 0x61, 0x01,
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+	0xc0, 0x10, 0x2f, 0xed,
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+	0x01, 0x01, 0x00, 0x03,
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+	0xd9, 0xff, 0x30, 0x01,
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+	0xeb, 0xc0, 0x20, 0x32,
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+	0x31, 0x00, 0x00, 0x20,
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+	0x00, 0xca, 0x00, 0x20,
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+	0x2d, 0x01, 0x56, 0x11,
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+	0x27, 0x12, 0xa0, 0x0e,
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+	0x10, 0x13, 0x00, 0x1c,
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+	0x03, 0xf0, 0x10, 0x0d,
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+	0x0f, 0x40, 0xff, 0xbe,
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+	0x0e, 0xeb, 0xff, 0x2f,
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+	0xed, 0xdd, 0x01, 0xf3,
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+	0xec, 0x01, 0x25, 0x11,
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+	0x11, 0x0e, 0xfe, 0x41,
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+	0x0d, 0xeb, 0x60, 0x36,
+	0x11, 0x0f, 0xff, 0x00, 0x40, 0xe0, 0xed, 0xee, 0x00, 0x00, 0xef, 0xee,
+	0x0d, 0x0e, 0xed, 0xdd,
+	0x11, 0x16, 0xae, 0xed, 0x04, 0xe2, 0xd0, 0x00, 0x01, 0xe0, 0x12, 0x42,
+	0xff, 0xac, 0x13, 0x77,
+	0x10, 0xee, 0x20, 0x3f, 0xf0, 0x04, 0x11, 0x32, 0x00, 0x02, 0x01, 0xf3,
+	0xf1, 0x04, 0xf2, 0xdf,
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+	0x30, 0xa1, 0x0f, 0xfa,
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+	0x0f, 0xdc, 0x1f, 0x10,
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+	0x0e, 0x0d, 0xc1, 0x12,
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+	0x20, 0xf2, 0xe2, 0x12,
+	0x43, 0x01, 0xbe, 0xfb, 0xf0, 0x01, 0x02, 0xf3, 0xe0, 0xfd, 0xff, 0xff,
+	0x33, 0x73, 0x0e, 0xeb,
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+	0xf0, 0x0f, 0x1f, 0x30,
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+	0x22, 0xef, 0xec, 0x0f,
+	0x0f, 0x01, 0x04, 0x02, 0xf0, 0xcf, 0x0f, 0x21, 0x00, 0xc0, 0x11, 0x77,
+	0xf0, 0xeb, 0x00, 0x40,
+	0x13, 0x11, 0x10, 0x0f, 0xf0, 0x10, 0x11, 0x36, 0x12, 0x20, 0xdf, 0xe0,
+	0x25, 0x02, 0xe0, 0xfe,
+	0xdf, 0x10, 0x56, 0x13, 0x10, 0xe0, 0x00, 0xf0, 0xfe, 0x42, 0xe1, 0x90,
+	0x33, 0x34, 0xbc, 0x0b,
+	0xf0, 0x21, 0x22, 0xf0, 0x0e, 0x3e, 0x0f, 0x10, 0x2e, 0xe0, 0x01, 0x0e,
+	0xff, 0xbf, 0xec, 0xfd,
+	0xfd, 0xef, 0xe0, 0x33, 0x0c, 0x8a, 0x00, 0x00, 0x00, 0x0d, 0x01, 0x00,
+	0xcc, 0x0d, 0xff, 0xdf,
+	0xee, 0xef, 0x00, 0x0f, 0xd0, 0x00, 0x10, 0x20, 0x24, 0x41, 0x21, 0x0d,
+	0x3f, 0x43, 0x27, 0x20,
+	0x0f, 0xce, 0x00, 0xf0, 0x00, 0xef, 0xcf, 0x02, 0x12, 0xef, 0xbc, 0x0d,
+	0x0f, 0xef, 0x0f, 0x10,
+	0x35, 0xe0, 0xed, 0x1f, 0x03, 0xf1, 0x02, 0xf2, 0xcf, 0x0b, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0x88, 0x13, 0xa4, 0x0f, 0xc4, 0x14, 0x10, 0x11, 0x42, 0x01, 0xef, 0x4e,
+	0x1e, 0x22, 0x31, 0xf0,
+	0x22, 0x20, 0x01, 0x12, 0xd0, 0xe0, 0xf0, 0xcf, 0x0f, 0x11, 0xf1, 0xef,
+	0xfe, 0x02, 0x12, 0xef,
+	0xf1, 0x01, 0x00, 0xe0, 0xf0, 0x0f, 0x00, 0x14, 0x0f, 0xed, 0x01, 0x2e,
+	0x00, 0x33, 0x23, 0x00,
+	0x10, 0x30, 0xfe, 0xef, 0x11, 0x11, 0x22, 0x02, 0x05, 0xf0, 0x00, 0x24,
+	0x00, 0x00, 0x33, 0x10,
+	0x10, 0x20, 0x30, 0x34, 0x21, 0x12, 0x00, 0xfc, 0x11, 0x25, 0xe0, 0xdf,
+	0xfd, 0xe0, 0xf3, 0xf2,
+	0x01, 0x1f, 0x0d, 0x00, 0x00, 0x0d, 0x10, 0x20, 0x2e, 0x21, 0x0f, 0x0a,
+	0xee, 0xfd, 0x20, 0x45,
+	0x00, 0x0d, 0xbd, 0x01, 0x34, 0x01, 0xbc, 0xee, 0x00, 0x10, 0xdb, 0xff,
+	0xf0, 0xf1, 0x01, 0xde,
+	0xdf, 0x23, 0x24, 0x12, 0x03, 0xf0, 0x04, 0x30, 0x00, 0x05, 0x13, 0xef,
+	0xe0, 0xe1, 0xf0, 0xdc,
+	0xcd, 0xf1, 0x07, 0x01, 0xe0, 0x00, 0xfe, 0xde, 0x02, 0x12, 0x32, 0x00,
+	0x2f, 0x1f, 0x3c, 0x0e,
+	0x41, 0x12, 0x10, 0x2d, 0x0f, 0x0e, 0x00, 0x30, 0x3f, 0x0f, 0x24, 0x42,
+	0x0f, 0x3d, 0x43, 0x33,
+	0x00, 0x0f, 0x00, 0x00, 0x0b, 0x0f, 0xee, 0xe0, 0xf0, 0x23, 0x02, 0xce,
+	0xff, 0xe0, 0x00, 0xbf,
+	0x02, 0xf6, 0xf0, 0x02, 0xc0, 0xdb, 0x00, 0x31, 0x44, 0x0e, 0xfd, 0x2f,
+	0x40, 0x41, 0x57, 0x13,
+	0x10, 0x0e, 0x0f, 0x12, 0x52, 0x05, 0x00, 0x03, 0x20, 0xf0, 0x01, 0xd0,
+	0xde, 0x01, 0x33, 0x02,
+	0xfe, 0xda, 0xff, 0x0f, 0x20, 0x15, 0x11, 0xfe, 0xcb, 0xd0, 0x12, 0x10,
+	0xde, 0xd0, 0xf0, 0xf0,
+	0xf1, 0xed, 0x0d, 0x00, 0x1d, 0xff, 0x0e, 0x12, 0x64, 0x00, 0x2d, 0x00,
+	0xe0, 0xea, 0x10, 0x32,
+	0xf0, 0xef, 0x0d, 0x12, 0x37, 0x00, 0xdf, 0xcd, 0xef, 0x00, 0x01, 0xe0,
+	0xf0, 0xf0, 0x00, 0xe0,
+	0xcd, 0xea, 0x12, 0x43, 0x12, 0xf0, 0x03, 0x20, 0x20, 0x21, 0x34, 0x00,
+	0xf0, 0xd0, 0x01, 0x00,
+	0x00, 0xff, 0xcb, 0xdf, 0x01, 0x15, 0xff, 0xde, 0xee, 0xe0, 0x25, 0x42,
+	0x20, 0x20, 0x2f, 0x10,
+	0x23, 0x00, 0x22, 0x13, 0x01, 0xf0, 0xcd, 0x0e, 0x01, 0x02, 0x12, 0x03,
+	0x0e, 0x0a, 0x00, 0xf0,
+	0x40, 0x20, 0x00, 0x0e, 0x2d, 0x0c, 0x0e, 0xea, 0xde, 0x20, 0x41, 0xf0,
+	0xdd, 0x0e, 0xee, 0xec,
+	0xcf, 0xe0, 0x03, 0x00, 0x0f, 0xde, 0x0f, 0x00, 0x21, 0xf0, 0x0f, 0x10,
+	0x0e, 0xf0, 0x02, 0x34,
+	0x24, 0x12, 0xff, 0xed, 0xff, 0x11, 0x55, 0x13, 0x13, 0xf0, 0xef, 0x25,
+	0x12, 0x02, 0xe0, 0xf2,
+	0x23, 0x11, 0x01, 0x00, 0xee, 0xfc, 0x33, 0x52, 0xff, 0xef, 0x00, 0x00,
+	0x30, 0x0e, 0x01, 0xe2,
+	0x0f, 0x20, 0x00, 0xcf, 0x01, 0x01, 0x0f, 0xee, 0xf0, 0x0c, 0x0e, 0x00,
+	0xff, 0xdb, 0x0e, 0x1f,
+	0x0e, 0xf0, 0xdd, 0x0f, 0x00, 0xe0, 0xf1, 0xdc, 0xbb, 0xff, 0xf0, 0x0f,
+	0xbe, 0xdb, 0x0e, 0xef,
+	0xfe, 0x0f, 0x10, 0x00, 0x20, 0x24, 0x44, 0x23, 0x23, 0x20, 0x21, 0x44,
+	0x02, 0xf0, 0x13, 0x0f,
+	0x1e, 0x00, 0xef, 0xed, 0xfe, 0x00, 0xf1, 0x01, 0x10, 0xe1, 0x02, 0x01,
+	0x00, 0x0e, 0x10, 0x13,
+	0x00, 0xed, 0xff, 0x31, 0x10, 0x00, 0x00, 0xdd, 0xfc, 0xfd, 0x20, 0x32,
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+	0x23, 0x1f, 0x0d, 0x01,
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+	0x00, 0x22, 0x23, 0x12,
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+	0x0b, 0x00, 0x21, 0x11,
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+	0xdf, 0xe0, 0xd0, 0xee,
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+	0x00, 0x00, 0x1f, 0x23,
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+	0xdf, 0x0e, 0x00, 0x21,
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+	0x20, 0xff, 0xff, 0x10,
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+	0xf0, 0xdc, 0xee, 0xe0,
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+	0x00, 0x0f, 0x01, 0x16,
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+	0x1d, 0x00, 0x00, 0xff,
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+	0x24, 0x12, 0xdf, 0xfd,
+	0x20, 0x33, 0x00, 0xcf, 0x00, 0x14, 0xf1, 0x00, 0x00, 0x0d, 0x00, 0x13,
+	0x00, 0x10, 0x10, 0xe0,
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+	0xe1, 0xee, 0x0c, 0x00,
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+	0x10, 0x00, 0x10, 0x00,
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+	0x23, 0x0f, 0xed, 0xfe,
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+	0x11, 0x12, 0x0f, 0xfc,
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+	0x05, 0x33, 0x01, 0x00,
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+	0x33, 0xd0, 0xf0, 0x0f,
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+	0xe0, 0xca, 0xf0, 0x10,
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+	0x14, 0x00, 0xe0, 0x00,
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+	0x00, 0x01, 0x02, 0xff,
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+	0x0f, 0x30, 0xfd, 0xca,
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+	0x20, 0x10, 0x00, 0xc0,
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+	0x0f, 0xf0, 0x15, 0x31,
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+	0xef, 0x35, 0x54, 0x00,
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+	0x02, 0x0f, 0x0f, 0x01,
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+	0xef, 0xfd, 0x0f, 0x13,
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+	0x11, 0x10, 0x1f, 0x00,
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+	0xda, 0x10, 0x20, 0x02,
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+	0x1f, 0x00, 0x00, 0x0e,
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+	0x01, 0x01, 0x03, 0x00,
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+	0x04, 0x20, 0x00, 0xf0,
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+	0xee, 0x10, 0x31, 0x23,
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+	0xec, 0xec, 0xdc, 0xef,
+	0x0e, 0xce, 0xff, 0xdd, 0xec, 0xef, 0xee, 0x00, 0xdc, 0x0e, 0x0e, 0xef,
+	0xfe, 0x10, 0x30, 0x20,
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+	0x00, 0x02, 0x03, 0x1f,
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+	0x00, 0x10, 0xf0, 0xee,
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+	0xff, 0x11, 0x24, 0x00,
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+	0x02, 0xf0, 0xf1, 0x0f,
+	0x0f, 0x00, 0xf1, 0xf0, 0xfd, 0x21, 0x12, 0x10, 0x0c, 0xff, 0x20, 0x10,
+	0x02, 0x01, 0x25, 0x00,
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+	0x04, 0x10, 0xfe, 0xfc,
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+	0x11, 0x0f, 0xf0, 0x23,
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+	0x0c, 0x0c, 0xfe, 0x0f,
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+	0x0e, 0x2e, 0x10, 0x0d,
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+	0xf2, 0xef, 0xdd, 0xdf,
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+	0x20, 0x02, 0x0f, 0x1f,
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+	0x01, 0xf1, 0x02, 0x30,
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+	0xed, 0xbe, 0x01, 0xf0,
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+	0x14, 0x21, 0x02, 0x02,
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+	0xf2, 0x04, 0x00, 0x0b,
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+	0x20, 0x00, 0x2f, 0x12,
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+	0x0d, 0xf0, 0x0e, 0x2f,
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+	0x00, 0x00, 0xf0, 0xfd,
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+	0x00, 0x11, 0x12, 0x01,
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+	0x00, 0x01, 0xf2, 0xf0,
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+	0x12, 0xf2, 0xde, 0x0d,
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+	0x0e, 0x00, 0x10, 0x0f,
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+	0xe0, 0x03, 0x12, 0xf0,
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+	0x03, 0xf3, 0xef, 0xfe,
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+	0x21, 0x0f, 0x0f, 0xfe,
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+	0xfe, 0xce, 0x0f, 0x2e,
+	0xdc, 0xde, 0xfc, 0xf0, 0xdd, 0xeb, 0xde, 0x0d, 0xff, 0xdf, 0x0f, 0x1e,
+	0x0f, 0x10, 0x0c, 0xee,
+	0x11, 0x44, 0x00, 0x0d, 0x23, 0x02, 0xdd, 0xfe, 0xf1, 0x07, 0x11, 0xe0,
+	0xee, 0x01, 0x13, 0x02,
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+	0x0e, 0x0e, 0xfd, 0xdf,
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+	0x44, 0xfd, 0xde, 0x13,
+	0x20, 0x13, 0x11, 0xf1, 0xde, 0x0f, 0xe1, 0xf1, 0xf0, 0xfe, 0xf0, 0xf0,
+	0x01, 0x13, 0x04, 0x23,
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+	0x1f, 0x12, 0x05, 0x0f,
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+	0x31, 0x12, 0x01, 0x00,
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+	0x02, 0x45, 0x1f, 0xfd,
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+	0x21, 0x13, 0x21, 0x00,
+	0xfb, 0xef, 0x0e, 0xfe, 0xcc, 0xec, 0x0f, 0x00, 0x1e, 0xf0, 0x01, 0xde,
+	0x0d, 0x0f, 0x2f, 0x0f,
+	0x10, 0x01, 0x23, 0x0f, 0x20, 0x00, 0xe0, 0xd0, 0x01, 0xf3, 0xf1, 0xdf,
+	0xd0, 0xf3, 0xe0, 0x04,
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+	0x23, 0x0f, 0xfd, 0xf1,
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+	0x1d, 0x30, 0x24, 0x01,
+	0x03, 0x00, 0x0e, 0x0f, 0x0e, 0xef, 0xf1, 0x02, 0x0f, 0xda, 0xff, 0xee,
+	0xff, 0x0e, 0x00, 0x04,
+	0xfd, 0xda, 0xff, 0x0e, 0x22, 0x22, 0x00, 0xdd, 0x1f, 0x20, 0x12, 0x00,
+	0xe0, 0x13, 0x14, 0x00,
+	0x0f, 0xed, 0x04, 0x44, 0x01, 0x13, 0x11, 0xfd, 0xff, 0x02, 0x04, 0x21,
+	0x0f, 0x00, 0x0f, 0xed,
+	0xef, 0x0e, 0x0e, 0xf0, 0x13, 0x1e, 0x0c, 0xff, 0x2f, 0x10, 0x10, 0x00,
+	0xe0, 0x10, 0xf0, 0xef,
+	0x0d, 0xff, 0xf0, 0x02, 0x00, 0x00, 0x1e, 0x30, 0xff, 0xf0, 0x22, 0x22,
+	0x10, 0x0e, 0x10, 0x22,
+	0x22, 0xee, 0xe0, 0x03, 0x00, 0xde, 0xde, 0xed, 0xfe, 0xe0, 0xf0, 0xe1,
+	0xf0, 0xef, 0x0e, 0x00,
+	0x33, 0x00, 0x10, 0x00, 0xe0, 0x00, 0x32, 0x13, 0xf0, 0xe0, 0xf0, 0xe0,
+	0xfe, 0xe0, 0xef, 0xff,
+	0x0f, 0x01, 0x23, 0x00, 0x31, 0x00, 0x00, 0x20, 0x21, 0x01, 0x03, 0x00,
+	0xf0, 0x0f, 0x10, 0xf0,
+	0xfe, 0xf1, 0x01, 0x00, 0x2f, 0x0e, 0x10, 0x44, 0x11, 0x10, 0x11, 0x01,
+	0x0e, 0xff, 0x00, 0x01,
+	0xe3, 0x12, 0x31, 0x00, 0x00, 0x0d, 0xef, 0x01, 0x23, 0x1d, 0xfd, 0x0f,
+	0x1f, 0x0c, 0xff, 0x03,
+	0x26, 0x00, 0xfd, 0x0e, 0x00, 0x02, 0x42, 0x13, 0x2f, 0x0d, 0xdc, 0x00,
+	0x2f, 0x10, 0x12, 0x02,
+	0xfe, 0xff, 0xf0, 0x0e, 0xd0, 0xf1, 0xf2, 0x02, 0xdf, 0x00, 0x02, 0xe0,
+	0xd0, 0x01, 0x1f, 0xef,
+	0xe0, 0x11, 0x02, 0x00, 0xf2, 0x24, 0x10, 0xff, 0x00, 0x10, 0x12, 0x11,
+	0x33, 0x22, 0x30, 0x22,
+	0x00, 0x00, 0x00, 0x44, 0x22, 0x11, 0x31, 0x00, 0x00, 0x01, 0x22, 0x0f,
+	0xcd, 0x0e, 0xfe, 0x01,
+	0x23, 0xdc, 0xed, 0xec, 0xed, 0x0e, 0xff, 0xeb, 0xdd, 0xed, 0x00, 0x00,
+	0xda, 0xef, 0x00, 0x21,
+	0x0f, 0xec, 0x00, 0x00, 0xff, 0x0f, 0x00, 0x20, 0x23, 0xf0, 0xff, 0x02,
+	0xf2, 0xe0, 0xf0, 0x03,
+	0x04, 0x20, 0x01, 0x24, 0x11, 0x10, 0x00, 0xcc, 0xfd, 0x00, 0x44, 0xf0,
+	0xdc, 0xf0, 0x0f, 0xdd,
+	0xfd, 0xf1, 0xe0, 0x00, 0x24, 0x10, 0xed, 0x0f, 0x00, 0x20, 0x10, 0xff,
+	0xfd, 0x00, 0x02, 0x12,
+	0x0f, 0xff, 0x0e, 0xf0, 0x00, 0xf2, 0xe0, 0x00, 0x05, 0x10, 0x00, 0xdf,
+	0xff, 0xed, 0xef, 0x00,
+	0x33, 0x00, 0x0e, 0x0e, 0x0f, 0x00, 0x20, 0x00, 0x22, 0xfe, 0x0f, 0x42,
+	0x01, 0xcf, 0x0f, 0x22,
+	0x01, 0x0f, 0xd0, 0x00, 0x10, 0x01, 0x04, 0x00, 0x0e, 0x10, 0x42, 0x12,
+	0x12, 0x23, 0x21, 0x00,
+	0x0d, 0xef, 0x01, 0x37, 0x11, 0x01, 0x00, 0xe0, 0xff, 0xfe, 0x00, 0x02,
+	0x47, 0x10, 0x00, 0x0f,
+	0x00, 0x00, 0xf0, 0x12, 0x11, 0x10, 0x0e, 0x10, 0x02, 0x34, 0x00, 0xfe,
+	0xf0, 0xff, 0x20, 0x23,
+	0x00, 0x03, 0x22, 0xfe, 0xd9, 0x1f, 0x10, 0xff, 0xde, 0x0f, 0x0f, 0x0f,
+	0xee, 0x0e, 0x1d, 0x0e,
+	0x0e, 0x12, 0x32, 0x00, 0xfe, 0xfe, 0x0f, 0x00, 0xf0, 0xff, 0x02, 0xe1,
+	0xe0, 0xe1, 0xd0, 0xcd,
+	0xfe, 0xe0, 0xe1, 0x03, 0x00, 0x10, 0x00, 0x12, 0x0e, 0xde, 0x21, 0x12,
+	0x10, 0x00, 0x00, 0xef,
+	0x00, 0x02, 0x00, 0xed, 0x10, 0x32, 0x02, 0x41, 0x44, 0x10, 0x21, 0x54,
+	0x01, 0x00, 0x30, 0x02,
+	0x33, 0x02, 0x01, 0x10, 0xf0, 0xdf, 0xfe, 0xe0, 0x00, 0x04, 0x0f, 0xec,
+	0x00, 0xde, 0xe9, 0xf0,
+	0x0f, 0x31, 0x12, 0xff, 0x0e, 0xfd, 0x00, 0x25, 0x12, 0x0d, 0x0d, 0x22,
+	0x21, 0x12, 0x11, 0x10,
+	0x00, 0x1f, 0x10, 0x00, 0x11, 0x15, 0x14, 0x00, 0x01, 0xdf, 0x0f, 0x11,
+	0x02, 0xe4, 0x00, 0x20,
+	0xfd, 0xee, 0x0f, 0x0f, 0x10, 0x32, 0x24, 0x10, 0xf0, 0x00, 0x0f, 0x01,
+	0x00, 0xf0, 0xfe, 0x10,
+	0x00, 0xf0, 0x00, 0x00, 0x2f, 0x0e, 0x0f, 0x2f, 0x40, 0x13, 0x0f, 0x20,
+	0x00, 0x13, 0x23, 0x0f,
+	0x0e, 0x00, 0x03, 0xdf, 0x0f, 0xe0, 0xfe, 0xee, 0xdd, 0xed, 0xf0, 0xe0,
+	0x0f, 0x00, 0x00, 0xf0,
+	0xe0, 0x00, 0xf3, 0x11, 0x12, 0x0e, 0xef, 0xf0, 0x00, 0xf1, 0x02, 0xee,
+	0x0d, 0xf0, 0x01, 0x02,
+	0x01, 0xfd, 0xee, 0xf0, 0x01, 0x32, 0x44, 0x03, 0x00, 0xef, 0x10, 0x00,
+	0x0e, 0x10, 0x0e, 0xf1,
+	0x00, 0xed, 0xfd, 0x00, 0x00, 0x02, 0x20, 0xee, 0xf0, 0x02, 0x32, 0x00,
+	0x0f, 0x00, 0xe0, 0xed,
+	0x00, 0x01, 0x10, 0xff, 0x01, 0x11, 0xfc, 0x00, 0x20, 0x03, 0xfe, 0x0e,
+	0x20, 0x00, 0x11, 0xff,
+	0x0b, 0x00, 0x20, 0x20, 0x0f, 0x0e, 0x10, 0x33, 0x11, 0x0f, 0x0e, 0x00,
+	0x20, 0xf0, 0xf1, 0x00,
+	0x0c, 0xfe, 0x12, 0x13, 0xef, 0xff, 0x12, 0x04, 0xff, 0xcd, 0x00, 0xf1,
+	0xf5, 0x02, 0x02, 0xff,
+	0xdd, 0x00, 0x01, 0xf0, 0xff, 0xe1, 0xf0, 0x0f, 0x0f, 0xfe, 0x00, 0x11,
+	0x32, 0x00, 0x00, 0x1f,
+	0x20, 0x35, 0x12, 0x11, 0x13, 0x00, 0x00, 0x0f, 0x10, 0x10, 0xe0, 0x03,
+	0x13, 0x00, 0x0f, 0x0f,
+	0xff, 0x0e, 0x02, 0xe3, 0xfe, 0x0d, 0xff, 0x0e, 0xfc, 0xff, 0x0e, 0xfd,
+	0xec, 0x0e, 0xfc, 0xdd,
+	0x0e, 0xf0, 0x12, 0x0e, 0xfc, 0xee, 0x00, 0x11, 0x00, 0x02, 0xf3, 0x00,
+	0x1d, 0x00, 0x01, 0x03,
+	0xf0, 0xe1, 0xe1, 0xe0, 0x00, 0x00, 0x22, 0x01, 0x00, 0xf0, 0x00, 0x00,
+	0x00, 0x0f, 0x0f, 0x00,
+	0x02, 0xf0, 0xff, 0x0d, 0xed, 0x0f, 0x10, 0x22, 0x12, 0x00, 0x0c, 0x0f,
+	0x40, 0x21, 0xe0, 0x0e,
+	0x22, 0x22, 0x00, 0xee, 0x11, 0x04, 0x00, 0xdf, 0x02, 0xe4, 0xe0, 0xee,
+	0xff, 0x00, 0xe0, 0x00,
+	0x21, 0xfd, 0x0c, 0x10, 0x21, 0x0f, 0x0e, 0x54, 0x25, 0x00, 0xff, 0xf0,
+	0x00, 0x21, 0xf0, 0xe0,
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+	0x01, 0x13, 0x03, 0x12,
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+	0x10, 0x12, 0x24, 0x10,
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+	0x00, 0x10, 0x35, 0x10,
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+	0x1e, 0x0f, 0xfd, 0x10,
+	0x21, 0x0e, 0xfd, 0xf0, 0x13, 0x21, 0x01, 0xce, 0x00, 0xff, 0xf0, 0xee,
+	0x00, 0x02, 0x00, 0x03,
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+	0xe0, 0x02, 0x1e, 0xfe,
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+	0x12, 0x10, 0x23, 0x01,
+	0x02, 0x34, 0x21, 0x0d, 0xf0, 0x01, 0x01, 0xe0, 0xfe, 0xef, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
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+	0x00, 0xdd, 0xf8, 0xff,
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+	0x01, 0x0e, 0x00, 0x00,
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+	0x23, 0x10, 0x00, 0x10,
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+	0x00, 0x0e, 0x10, 0x02,
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+	0x01, 0xe0, 0x1f, 0x24,
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+	0x02, 0xf0, 0x00, 0xff,
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+	0x02, 0x01, 0xe0, 0x00,
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+	0x00, 0x11, 0x03, 0x20,
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+	0x0f, 0x02, 0x20, 0x0e,
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+	0x03, 0x2f, 0xfb, 0xff,
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+	0x00, 0xec, 0x0d, 0x1f,
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+	0xf1, 0xe1, 0xe0, 0x0f,
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+	0x10, 0x00, 0x23, 0x23,
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+	0x01, 0x01, 0x20, 0xfc,
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+	0x0f, 0xfd, 0xff, 0xf0,
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+	0x11, 0xf2, 0xf0, 0xe0,
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+	0x0e, 0x01, 0xf0, 0xec,
+	0x0f, 0x11, 0x00, 0xfe, 0xeb, 0x0f, 0x11, 0x01, 0xff, 0x00, 0x20, 0x0e,
+	0x10, 0x00, 0x14, 0xee,
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+	0x01, 0x35, 0x1f, 0xfc,
+	0x00, 0x10, 0x31, 0x00, 0x00, 0x00, 0x0f, 0x23, 0x25, 0xff, 0xcf, 0x0e,
+	0x02, 0x01, 0x0d, 0x0d,
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+	0x00, 0xef, 0xff, 0x02,
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+	0x10, 0x11, 0x0f, 0xf0,
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+	0x13, 0x25, 0x00, 0x0f,
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+	0x0f, 0x00, 0x1e, 0x00,
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+	0xf0, 0xd1, 0xf0, 0xf1,
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+	0x10, 0x22, 0x35, 0x30,
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+	0xf1, 0x01, 0x10, 0xd9,
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+	0x10, 0x01, 0x11, 0x00,
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+	0x2f, 0x0f, 0x00, 0xfe,
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+	0x0f, 0xef, 0x10, 0x00,
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+	0x03, 0x02, 0x10, 0xe0,
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+	0x30, 0x0f, 0xfd, 0x10,
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+	0x20, 0xfe, 0xf0, 0x04,
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+	0x01, 0x33, 0x11, 0x00,
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+	0x30, 0x0f, 0x01, 0x10,
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+	0xde, 0xef, 0xf2, 0xf2,
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+	0x11, 0x21, 0x10, 0x00,
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+	0x1e, 0x32, 0x16, 0x11,
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+	0xda, 0xf0, 0x10, 0x30,
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+	0xdd, 0x00, 0x12, 0x00,
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+	0x31, 0x00, 0x0e, 0xff,
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+	0x00, 0x02, 0x23, 0x20,
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+	0x0f, 0xf0, 0x0f, 0x00,
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+	0x00, 0x1c, 0xf0, 0x0f,
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+	0xf2, 0x03, 0x01, 0xfe,
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+	0x20, 0x20, 0x00, 0x02,
+	0x23, 0xe0, 0xfe, 0x02, 0x02, 0x11, 0x14, 0x13, 0x01, 0x00, 0x01, 0xff,
+	0x0c, 0x0e, 0x0f, 0x55,
+	0x13, 0x10, 0x0d, 0x0e, 0x10, 0x10, 0x01, 0x00, 0x0d, 0x00, 0x33, 0x11,
+	0x0e, 0xdb, 0x0f, 0xfe,
+	0x0f, 0x41, 0x01, 0x0c, 0xfe, 0xff, 0x00, 0x0f, 0x00, 0x01, 0x00, 0xee,
+	0xf0, 0x00, 0x02, 0x13,
+	0xf0, 0xcf, 0x00, 0x0e, 0xe0, 0xd0, 0xdf, 0xf0, 0x0f, 0xfe, 0xee, 0xf0,
+	0x03, 0x12, 0x12, 0x21,
+	0x01, 0x0f, 0x00, 0x21, 0xf0, 0x00, 0x11, 0x32, 0x10, 0x10, 0x00, 0x24,
+	0x12, 0x00, 0x13, 0x42,
+	0x01, 0x01, 0x15, 0x11, 0xe0, 0x01, 0x01, 0xee, 0xdd, 0xf0, 0x0f, 0xe0,
+	0x01, 0xfd, 0xdc, 0x0f,
+	0x10, 0x23, 0x10, 0x10, 0x0e, 0xca, 0x0f, 0x00, 0x11, 0x20, 0xef, 0xfe,
+	0x00, 0x12, 0x1f, 0x0e,
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+	0x01, 0x02, 0xf2, 0x01,
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+	0x1f, 0x00, 0x10, 0x00,
+	0xfe, 0xef, 0x10, 0x11, 0x1f, 0x00, 0x20, 0x02, 0x1e, 0x0f, 0x10, 0x0d,
+	0xff, 0x00, 0x20, 0x01,
+	0x12, 0x11, 0x02, 0x00, 0x0f, 0x12, 0x00, 0xe1, 0x00, 0xee, 0x00, 0x00,
+	0x00, 0xf0, 0xf1, 0x00,
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+	0xef, 0xff, 0x00, 0x03,
+	0x11, 0x0e, 0x0f, 0x00, 0x00, 0x01, 0xb0, 0x00, 0xee, 0xfe, 0xff, 0x01,
+	0x33, 0x00, 0x00, 0x34,
+	0x43, 0x0f, 0x1c, 0x22, 0x12, 0x11, 0x12, 0x0f, 0xfd, 0xff, 0x1e, 0x01,
+	0x02, 0x02, 0xf0, 0x0e,
+	0x1e, 0x00, 0x13, 0x12, 0x00, 0x10, 0x23, 0xfe, 0xdc, 0x0f, 0x00, 0x03,
+	0x12, 0x02, 0xee, 0xed,
+	0xe0, 0x22, 0x24, 0x00, 0xff, 0x0d, 0x1e, 0x10, 0x10, 0x1f, 0x20, 0x24,
+	0x22, 0x0e, 0x0d, 0x00,
+	0x53, 0x02, 0x10, 0x0c, 0xf0, 0x00, 0xfb, 0xef, 0x10, 0x00, 0xf1, 0x01,
+	0x01, 0xe0, 0xde, 0xef,
+	0xef, 0xd1, 0x02, 0xe0, 0xf2, 0x00, 0xf0, 0xfe, 0xff, 0x00, 0x02, 0x10,
+	0x1f, 0x10, 0x11, 0x00,
+	0x10, 0x01, 0x01, 0xf0, 0x00, 0x01, 0x32, 0x02, 0x13, 0x10, 0x14, 0x43,
+	0x02, 0x1e, 0xfd, 0x11,
+	0x33, 0x10, 0x00, 0x00, 0x0e, 0x1c, 0x13, 0x32, 0xf0, 0xfd, 0xf0, 0xfd,
+	0xfd, 0xef, 0xff, 0xfe,
+	0xfe, 0x00, 0x11, 0x0d, 0xf9, 0xe0, 0x0e, 0x00, 0xee, 0x00, 0x21, 0x21,
+	0x1f, 0x00, 0xf2, 0x00,
+	0x02, 0x0f, 0xff, 0xd0, 0x04, 0x02, 0xd0, 0xef, 0xd0, 0xde, 0xfd, 0xed,
+	0xf0, 0xf2, 0x11, 0x43,
+	0x13, 0x0f, 0x09, 0xf0, 0x11, 0x00, 0x00, 0x10, 0x11, 0xfc, 0x0e, 0xfe,
+	0xef, 0x01, 0x13, 0x00,
+	0x0e, 0xe0, 0xf0, 0x10, 0x00, 0xe0, 0x1f, 0x12, 0x01, 0xdd, 0xef, 0xff,
+	0x00, 0x00, 0xe0, 0xec,
+	0x00, 0x00, 0x15, 0x21, 0xe0, 0x0f, 0x00, 0x01, 0x0f, 0xee, 0x0d, 0xf0,
+	0x24, 0x33, 0x01, 0x0c,
+	0xdd, 0x00, 0x10, 0x42, 0x11, 0x0d, 0x0c, 0x10, 0x02, 0x22, 0x23, 0x00,
+	0xf0, 0x00, 0x15, 0x01,
+	0xf0, 0xd1, 0xf0, 0xee, 0x00, 0x11, 0x11, 0x00, 0x44, 0x12, 0xfd, 0xfe,
+	0x20, 0x20, 0x35, 0xf0,
+	0x00, 0x01, 0x01, 0xf1, 0xff, 0xff, 0xf0, 0x01, 0x23, 0xf2, 0x01, 0x12,
+	0x01, 0x22, 0x00, 0xda,
+	0x00, 0x00, 0x13, 0x33, 0x1f, 0xfe, 0x02, 0x21, 0x00, 0x10, 0x00, 0x10,
+	0x0f, 0x2f, 0x00, 0x0f,
+	0x0c, 0x00, 0x0d, 0x1e, 0x30, 0x10, 0x00, 0xfc, 0xde, 0x1f, 0x20, 0x10,
+	0x0f, 0xee, 0xf1, 0x02,
+	0x01, 0x0e, 0xec, 0xde, 0xef, 0x02, 0x02, 0xde, 0xed, 0xbe, 0xfd, 0xdf,
+	0xdf, 0xf0, 0xf0, 0x01,
+	0x15, 0x10, 0xfd, 0x0f, 0x02, 0x22, 0x30, 0x23, 0x20, 0x10, 0x20, 0x44,
+	0x0f, 0xe0, 0x03, 0x34,
+	0x42, 0x01, 0x02, 0x35, 0x11, 0x01, 0x01, 0x0f, 0xdd, 0x00, 0x10, 0x21,
+	0x01, 0xdf, 0xf0, 0xff,
+	0x0f, 0xfe, 0xf0, 0xf0, 0x04, 0x00, 0x10, 0x0d, 0xcd, 0x0d, 0x00, 0x31,
+	0x00, 0x0d, 0x01, 0x20,
+	0x10, 0x20, 0x12, 0x21, 0x10, 0x0e, 0x0f, 0x10, 0x32, 0x13, 0x01, 0x31,
+	0x00, 0x0e, 0x01, 0x02,
+	0xf0, 0xf0, 0x01, 0xf1, 0xef, 0x00, 0xe0, 0x00, 0x12, 0x25, 0xf0, 0x02,
+	0x00, 0x0f, 0x0e, 0xfe,
+	0xff, 0x01, 0x43, 0x01, 0x00, 0xfd, 0x0d, 0x10, 0x44, 0x24, 0x00, 0xfd,
+	0x20, 0x32, 0x01, 0x00,
+	0x00, 0xfd, 0x0f, 0x00, 0x21, 0x10, 0x0e, 0x00, 0x32, 0x12, 0x01, 0xef,
+	0xfd, 0x00, 0x00, 0xf3,
+	0xf0, 0x0e, 0xf0, 0xfe, 0xef, 0xdf, 0x10, 0x21, 0x01, 0xdf, 0xee, 0xef,
+	0x00, 0xe0, 0xf2, 0xe0,
+	0x13, 0x13, 0x00, 0xdd, 0xef, 0xf0, 0xfe, 0x00, 0x00, 0xf0, 0x0f, 0x0e,
+	0xee, 0x0f, 0x0f, 0x0f,
+	0x32, 0x10, 0x01, 0x22, 0x00, 0x0d, 0x02, 0x65, 0x00, 0xea, 0x00, 0x00,
+	0xff, 0xdf, 0x10, 0x23,
+	0x00, 0xca, 0x00, 0x22, 0xf0, 0xe0, 0x02, 0x31, 0xff, 0xee, 0xf0, 0x00,
+	0xf1, 0x00, 0x0f, 0x00,
+	0xf0, 0x00, 0xee, 0x11, 0x24, 0x00, 0x00, 0x0f, 0x42, 0x13, 0x2f, 0xfe,
+	0x0f, 0x40, 0x0f, 0x20,
+	0x00, 0x30, 0x10, 0x10, 0x12, 0x00, 0x0c, 0xf0, 0x00, 0xff, 0xf0, 0x00,
+	0x00, 0xf0, 0xe0, 0x02,
+	0xe0, 0xd0, 0x0f, 0xe0, 0xef, 0x0f, 0xf0, 0xff, 0xee, 0xfe, 0x01, 0x25,
+	0x1f, 0xfe, 0xf0, 0x00,
+	0xef, 0xec, 0x00, 0x02, 0x33, 0x10, 0x00, 0x00, 0x02, 0x04, 0x32, 0xf0,
+	0x0f, 0x01, 0x35, 0x13,
+	0x00, 0xfd, 0xfe, 0x12, 0x33, 0x00, 0x0e, 0xf0, 0x12, 0x11, 0x14, 0x00,
+	0xfe, 0xed, 0xfe, 0x00,
+	0xf1, 0xee, 0xfb, 0x0f, 0x10, 0x20, 0x0c, 0xfe, 0x0f, 0x1d, 0x0e, 0x0d,
+	0x0f, 0x22, 0x01, 0x00,
+	0xd0, 0x01, 0x01, 0xe0, 0xef, 0xf1, 0xc0, 0xec, 0xef, 0xf0, 0xf0, 0xee,
+	0xf0, 0xff, 0xde, 0xdf,
+	0x11, 0x44, 0x01, 0xf0, 0xff, 0x00, 0x23, 0x10, 0x0f, 0x0e, 0x11, 0x01,
+	0x00, 0xfc, 0x0f, 0x30,
+	0x13, 0x10, 0x0f, 0x34, 0x02, 0x0f, 0x00, 0x20, 0x0e, 0x0e, 0x31, 0x01,
+	0xde, 0xfe, 0x0f, 0xfd,
+	0xe0, 0x0f, 0xef, 0x01, 0x04, 0x01, 0xd0, 0xfe, 0xf0, 0xf1, 0x00, 0x03,
+	0x33, 0x0e, 0xca, 0x00,
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+	0x22, 0x01, 0x00, 0xef,
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+	0x01, 0x10, 0x00, 0x00,
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+	0x14, 0x10, 0x10, 0x14,
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+	0x20, 0x0e, 0x0f, 0x00,
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+	0x00, 0x10, 0x21, 0x01,
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+	0xff, 0xee, 0xef, 0x0f,
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+	0x13, 0x24, 0x00, 0x20,
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+	0xf1, 0xf1, 0x00, 0x0f,
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+	0x00, 0x01, 0x1f, 0x0c,
+	0x10, 0x0f, 0x20, 0x32, 0x10, 0x1d, 0x00, 0x2f, 0x1f, 0x00, 0x0e, 0x01,
+	0x01, 0x21, 0x00, 0x0e,
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+	0x32, 0xff, 0x0f, 0x01,
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+	0x12, 0x21, 0x20, 0x02,
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+	0x11, 0x00, 0x00, 0xef,
+	0x02, 0xf1, 0xf1, 0x00, 0x0f, 0x00, 0x0e, 0xce, 0xf0, 0x11, 0x44, 0x10,
+	0xff, 0xac, 0x00, 0x00,
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+	0xdf, 0x12, 0x00, 0x01,
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+	0x02, 0x33, 0x12, 0x0d,
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+	0xfd, 0xe0, 0xef, 0xfd,
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+	0x31, 0x00, 0xfc, 0xfd,
+	0x30, 0x51, 0x01, 0x11, 0x30, 0x10, 0x0c, 0x01, 0x00, 0x10, 0x10, 0x02,
+	0xd0, 0x00, 0x01, 0xe0,
+	0xed, 0xff, 0xe0, 0x02, 0xee, 0xfe, 0xf0, 0x10, 0x00, 0xdf, 0xfc, 0x00,
+	0x01, 0x01, 0x0f, 0xf0,
+	0xf0, 0x03, 0x11, 0xde, 0xec, 0xf0, 0x00, 0x45, 0x01, 0x21, 0x0f, 0x00,
+	0x13, 0x31, 0x0e, 0x0f,
+	0x24, 0x33, 0x03, 0x00, 0x01, 0x0f, 0x2f, 0x11, 0x03, 0x01, 0x00, 0x00,
+	0x0e, 0x00, 0x01, 0x00,
+	0x0c, 0xdf, 0x10, 0x20, 0x0b, 0x0f, 0x10, 0x0f, 0x2f, 0xfd, 0x00, 0x2f,
+	0x0d, 0x00, 0x02, 0xf1,
+	0xdd, 0x01, 0x12, 0x00, 0xff, 0xf0, 0x02, 0xf1, 0xd0, 0x10, 0x0e, 0xbd,
+	0xed, 0xf0, 0x02, 0x01,
+	0x00, 0x20, 0xcb, 0xff, 0xf0, 0xf1, 0x02, 0x11, 0x23, 0x00, 0x0b, 0xef,
+	0x00, 0x00, 0x00, 0x00,
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+	0x0f, 0x03, 0x02, 0xfd,
+	0xfc, 0xee, 0xdf, 0x01, 0x02, 0x05, 0x00, 0xdf, 0xdf, 0x0f, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0x8c, 0x11, 0x68, 0x09, 0xce, 0x12, 0x6f, 0x0a, 0x00, 0xf1, 0x00, 0x00,
+	0x0e, 0x0e, 0x10, 0x32,
+	0x22, 0x2f, 0x1f, 0x10, 0x00, 0x02, 0x12, 0x10, 0xfd, 0xe0, 0x00, 0x10,
+	0x03, 0x01, 0x00, 0xf0,
+	0x00, 0x00, 0x00, 0x00, 0x10, 0x23, 0x01, 0xf0, 0x00, 0x11, 0x01, 0x20,
+	0x21, 0x00, 0x01, 0x13,
+	0xfe, 0xfd, 0x01, 0x22, 0xe0, 0xf1, 0x00, 0x00, 0x03, 0x02, 0x00, 0xe0,
+	0xe1, 0x13, 0x23, 0x13,
+	0x10, 0x0d, 0xed, 0x00, 0x00, 0x12, 0x01, 0x1f, 0x31, 0x23, 0x01, 0x0f,
+	0xff, 0x1e, 0x10, 0x00,
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+	0x01, 0x01, 0x02, 0x00,
+	0xf0, 0xed, 0xe0, 0xdf, 0xdd, 0xdd, 0xee, 0xff, 0xff, 0xed, 0xcd, 0xf0,
+	0xef, 0x0f, 0xec, 0xef,
+	0xdf, 0x02, 0x26, 0x00, 0x00, 0x11, 0x30, 0x22, 0x12, 0x00, 0xf0, 0x00,
+	0x31, 0x32, 0x11, 0x10,
+	0x0f, 0x00, 0x05, 0x32, 0x12, 0x22, 0x01, 0xf0, 0x02, 0x22, 0x03, 0x31,
+	0xfd, 0xdc, 0x00, 0x00,
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+	0x00, 0x30, 0x20, 0x31,
+	0x00, 0x0f, 0x5f, 0x20, 0x2f, 0x00, 0x21, 0x31, 0x00, 0xff, 0xf0, 0x00,
+	0xf0, 0x01, 0x03, 0x01,
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+	0x00, 0x0f, 0x23, 0x00,
+	0xbe, 0x0f, 0x0e, 0x22, 0x34, 0x02, 0x0e, 0xfd, 0x00, 0x32, 0x23, 0x22,
+	0x00, 0xf0, 0x0f, 0x33,
+	0x11, 0x0f, 0xfc, 0xff, 0x0f, 0x23, 0x00, 0x0c, 0x20, 0x30, 0x00, 0x00,
+	0x00, 0x00, 0x03, 0x21,
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+	0xed, 0xdf, 0x00, 0xf0,
+	0xf1, 0x01, 0xe0, 0x0e, 0xe1, 0x03, 0x0f, 0xfd, 0xef, 0x0f, 0x00, 0x02,
+	0xf0, 0x00, 0xfd, 0xdc,
+	0x0f, 0x0f, 0x0f, 0x1f, 0x34, 0x12, 0x00, 0x0c, 0xf0, 0x20, 0x31, 0x11,
+	0x00, 0xfc, 0x00, 0x20,
+	0xf0, 0xff, 0xf0, 0x10, 0x12, 0xff, 0xdf, 0xf0, 0x00, 0xe0, 0x00, 0x02,
+	0xff, 0x0d, 0x00, 0x02,
+	0xf2, 0xf0, 0x23, 0x10, 0xec, 0x01, 0x01, 0x02, 0x14, 0x01, 0x20, 0x00,
+	0x1f, 0x0d, 0x0e, 0x20,
+	0x2f, 0x21, 0x53, 0x11, 0x0f, 0x0e, 0x00, 0x13, 0x02, 0xf0, 0x0f, 0xde,
+	0xff, 0xf0, 0xe0, 0x0f,
+	0x00, 0x20, 0xff, 0xed, 0xed, 0x00, 0x01, 0x10, 0x0f, 0xde, 0xf0, 0xef,
+	0xf0, 0xf1, 0x01, 0x00,
+	0x01, 0xf0, 0xff, 0xda, 0x00, 0x10, 0x23, 0x21, 0x0e, 0x0f, 0x21, 0x12,
+	0x00, 0xe1, 0x12, 0x24,
+	0x03, 0x24, 0xe0, 0xff, 0xf0, 0xf0, 0x00, 0x03, 0x30, 0x0d, 0x00, 0x02,
+	0x02, 0x00, 0x0e, 0xfd,
+	0x00, 0x40, 0x01, 0x1f, 0xfe, 0xee, 0x0f, 0x20, 0x2f, 0x0c, 0x00, 0x0f,
+	0xe0, 0x00, 0x10, 0x0e,
+	0xe0, 0xf0, 0x0f, 0xf0, 0xe0, 0x01, 0xf1, 0xfe, 0xeb, 0xcf, 0xde, 0xff,
+	0x01, 0x04, 0x10, 0xff,
+	0xdd, 0xe0, 0xee, 0xe0, 0x00, 0x01, 0x02, 0x03, 0x01, 0x00, 0x0d, 0x00,
+	0x0f, 0x1e, 0x2f, 0x30,
+	0x10, 0x23, 0x12, 0x20, 0x30, 0x0f, 0xef, 0x00, 0x32, 0x01, 0x10, 0x0f,
+	0xdd, 0x00, 0x0d, 0xef,
+	0xff, 0x00, 0xf0, 0xf0, 0x0f, 0x00, 0xce, 0x01, 0xf3, 0xf0, 0x01, 0x00,
+	0xf0, 0x02, 0x30, 0xff,
+	0xed, 0x10, 0x41, 0x33, 0x30, 0x1f, 0x00, 0x00, 0xf0, 0x01, 0x00, 0xfd,
+	0x0f, 0x01, 0xf4, 0x01,
+	0x02, 0x20, 0x0f, 0xff, 0x20, 0x12, 0x21, 0x00, 0x0f, 0x00, 0xf2, 0x00,
+	0x0e, 0x01, 0x45, 0x00,
+	0xf0, 0xf0, 0x01, 0xf0, 0xf1, 0x02, 0x02, 0xf0, 0x10, 0x03, 0x10, 0x0f,
+	0x00, 0x01, 0x00, 0x00,
+	0x00, 0x34, 0x01, 0x00, 0xf0, 0x02, 0x00, 0xf0, 0xff, 0xfd, 0x0e, 0x12,
+	0x56, 0x10, 0x0f, 0xee,
+	0x00, 0x3d, 0x10, 0x20, 0x53, 0x10, 0x0d, 0x00, 0x12, 0x30, 0x30, 0x02,
+	0x00, 0x00, 0xfe, 0x00,
+	0x00, 0x10, 0x00, 0xd2, 0x00, 0xec, 0x0f, 0xff, 0xdc, 0xdf, 0x01, 0xf1,
+	0xe0, 0xf0, 0xfe, 0xff,
+	0x00, 0x00, 0xe0, 0xfe, 0xee, 0xf0, 0x0e, 0x00, 0x33, 0x00, 0x32, 0x33,
+	0x00, 0x00, 0x00, 0x00,
+	0x20, 0x11, 0x13, 0x14, 0x00, 0x00, 0xf1, 0x04, 0xf0, 0xe0, 0x00, 0x33,
+	0x11, 0x11, 0xf0, 0xee,
+	0xf0, 0x12, 0x01, 0xf0, 0xed, 0xee, 0xe0, 0x03, 0xc0, 0xe0, 0x0f, 0xfe,
+	0x0f, 0xdf, 0x00, 0x10,
+	0x40, 0x00, 0x2d, 0x2f, 0x1b, 0x30, 0x20, 0x00, 0x23, 0x10, 0x1e, 0xfd,
+	0xed, 0x00, 0x02, 0x00,
+	0x01, 0x31, 0x00, 0xe0, 0x00, 0x01, 0x00, 0x00, 0x11, 0x00, 0xde, 0x00,
+	0x00, 0x02, 0xf0, 0xed,
+	0x00, 0xf0, 0x11, 0x10, 0xed, 0xee, 0x11, 0x45, 0x01, 0xf0, 0x0d, 0x10,
+	0x24, 0x21, 0xf0, 0xef,
+	0x0e, 0x20, 0x01, 0x01, 0xf0, 0x0e, 0x0f, 0x00, 0x40, 0xf0, 0xeb, 0x00,
+	0x22, 0x21, 0x01, 0x01,
+	0x33, 0x00, 0xef, 0xf1, 0xf0, 0xff, 0x01, 0x31, 0xe0, 0xdd, 0x00, 0xff,
+	0xf0, 0x02, 0x01, 0x00,
+	0x0c, 0xe0, 0x00, 0x01, 0xf2, 0xef, 0x00, 0x01, 0xff, 0xde, 0x01, 0x03,
+	0x0f, 0xed, 0x00, 0x00,
+	0xfe, 0x00, 0x15, 0x10, 0xee, 0xee, 0x0f, 0x42, 0x13, 0xff, 0xf0, 0x10,
+	0x22, 0x42, 0x00, 0x10,
+	0x2f, 0x0f, 0x00, 0x11, 0x1f, 0x2f, 0x00, 0x11, 0x03, 0xe0, 0x02, 0xff,
+	0x0d, 0xf0, 0x00, 0x11,
+	0xd0, 0xf0, 0xf0, 0xf1, 0x10, 0x00, 0x0d, 0xe0, 0xf0, 0xef, 0x00, 0x01,
+	0x30, 0x10, 0x03, 0x41,
+	0x00, 0x1d, 0x11, 0x42, 0x23, 0x01, 0x00, 0xee, 0x00, 0x43, 0xf0, 0xee,
+	0x10, 0x0f, 0xde, 0x00,
+	0x10, 0xff, 0x00, 0x00, 0x01, 0x01, 0xf0, 0xf1, 0xee, 0xed, 0xfe, 0xdd,
+	0xfd, 0x02, 0x15, 0x0f,
+	0x0d, 0xef, 0x00, 0x00, 0x00, 0xfe, 0x0f, 0x10, 0x23, 0x00, 0x21, 0x11,
+	0x12, 0xf0, 0xe1, 0x02,
+	0x00, 0x00, 0x20, 0x14, 0x03, 0x00, 0xf0, 0x00, 0x10, 0x14, 0x1f, 0x10,
+	0x00, 0x0e, 0xed, 0x13,
+	0x43, 0x1e, 0x0c, 0x00, 0x00, 0x20, 0x10, 0x10, 0x0d, 0x00, 0x0e, 0x00,
+	0x02, 0x45, 0xf0, 0x00,
+	0x00, 0x0e, 0xfe, 0xe0, 0xee, 0x00, 0x23, 0x00, 0xfd, 0xf0, 0x00, 0xf0,
+	0xdd, 0x00, 0xf0, 0xf0,
+	0xf0, 0x03, 0x01, 0xce, 0xdc, 0xdc, 0xdf, 0xef, 0xfe, 0xf0, 0x02, 0x12,
+	0x2e, 0x0d, 0x0d, 0x1f,
+	0x31, 0x22, 0x01, 0x0f, 0xf0, 0x0f, 0x2f, 0xff, 0x00, 0x03, 0x02, 0x00,
+	0xfd, 0xed, 0xfe, 0x00,
+	0x10, 0x00, 0x01, 0xf1, 0x00, 0xef, 0xff, 0xf0, 0xe1, 0xd0, 0xd0, 0x01,
+	0x20, 0xe0, 0xff, 0x00,
+	0x00, 0x00, 0x31, 0x20, 0x00, 0x20, 0x32, 0x00, 0x2f, 0x00, 0xef, 0x00,
+	0x10, 0xf0, 0xd0, 0x00,
+	0x02, 0x00, 0x01, 0x10, 0x02, 0x10, 0x01, 0x20, 0x32, 0x21, 0x0f, 0xff,
+	0x10, 0x00, 0xe0, 0x00,
+	0x01, 0x11, 0x00, 0xf0, 0x03, 0xf0, 0xf0, 0x0f, 0x00, 0x11, 0x01, 0x00,
+	0x00, 0x04, 0x32, 0x13,
+	0x00, 0xdb, 0xf0, 0x00, 0xf2, 0x04, 0x21, 0xff, 0xef, 0x01, 0x01, 0x00,
+	0xdc, 0x00, 0x30, 0x00,
+	0x10, 0x20, 0x2d, 0x30, 0x01, 0x00, 0x32, 0x32, 0x20, 0x30, 0x1f, 0x00,
+	0x0e, 0x10, 0x21, 0x10,
+	0x10, 0xfe, 0xfd, 0x01, 0x00, 0xee, 0xfe, 0xfd, 0xff, 0xf0, 0x01, 0xbe,
+	0xfe, 0xef, 0x0e, 0xdc,
+	0xef, 0x0f, 0x00, 0xdf, 0xf0, 0x0f, 0xdd, 0xff, 0xef, 0x00, 0xf0, 0x02,
+	0x14, 0x53, 0x01, 0x0d,
+	0xf0, 0xf0, 0x25, 0x42, 0x34, 0x10, 0x00, 0xfe, 0x11, 0x35, 0x01, 0x21,
+	0x01, 0x0f, 0x10, 0x12,
+	0x02, 0x00, 0xfd, 0x01, 0x00, 0xe0, 0xfe, 0xf1, 0xe3, 0xf1, 0xde, 0xed,
+	0x0f, 0xff, 0xde, 0x00,
+	0x33, 0x13, 0x33, 0x3f, 0x0c, 0x10, 0x20, 0x1f, 0x20, 0x00, 0x2e, 0x10,
+	0x00, 0x1f, 0x00, 0x00,
+	0xf0, 0x01, 0x01, 0x22, 0x31, 0x13, 0x01, 0x00, 0xdf, 0xfe, 0x11, 0x12,
+	0x0e, 0x0f, 0xff, 0x00,
+	0xf0, 0xef, 0xf0, 0x11, 0x35, 0x02, 0x00, 0x01, 0xfe, 0x11, 0x34, 0xf0,
+	0xef, 0x00, 0x42, 0x13,
+	0x0f, 0x1f, 0xfc, 0xfe, 0x00, 0x12, 0x33, 0x0e, 0xfe, 0x10, 0x31, 0x13,
+	0x10, 0xfe, 0xfe, 0x00,
+	0x32, 0x00, 0x00, 0x1e, 0xf0, 0x01, 0xf0, 0x00, 0xef, 0xfe, 0x11, 0x23,
+	0xf0, 0xd0, 0xed, 0xf0,
+	0x14, 0x00, 0xed, 0x01, 0xf3, 0xe0, 0x00, 0xff, 0xec, 0x0e, 0xf0, 0xf2,
+	0xf1, 0xf0, 0xee, 0xff,
+	0x00, 0x11, 0xff, 0xfe, 0x00, 0xf2, 0x00, 0x0e, 0xfd, 0x0f, 0x30, 0x00,
+	0xf0, 0x20, 0x2d, 0x1f,
+	0x20, 0x12, 0x11, 0x0e, 0x0f, 0x11, 0x00, 0xee, 0x00, 0x02, 0xf0, 0xe1,
+	0x00, 0xf0, 0xfe, 0xf0,
+	0x02, 0x01, 0x01, 0x10, 0x00, 0xfe, 0xe0, 0x00, 0x00, 0x01, 0xf0, 0x02,
+	0x13, 0x0d, 0xff, 0xef,
+	0x01, 0x13, 0x45, 0x20, 0x10, 0x2f, 0x00, 0x32, 0x13, 0x1e, 0x1c, 0x00,
+	0x0d, 0x00, 0x31, 0x14,
+	0x01, 0xde, 0xde, 0x0f, 0x00, 0x00, 0xf2, 0xf1, 0x00, 0x0f, 0xce, 0x0d,
+	0xfe, 0xef, 0x0f, 0x0e,
+	0x00, 0x00, 0xec, 0x0f, 0x10, 0x0f, 0xf0, 0xf1, 0x01, 0x20, 0xf0, 0xf0,
+	0x00, 0x11, 0x11, 0x00,
+	0xf0, 0xf2, 0x03, 0x0f, 0x00, 0x01, 0xe2, 0x00, 0x01, 0x11, 0x02, 0x00,
+	0x02, 0x35, 0x00, 0xde,
+	0xfd, 0x10, 0x22, 0x1f, 0xfc, 0xf0, 0x22, 0x22, 0x10, 0x0e, 0xe0, 0x00,
+	0x00, 0x00, 0x20, 0x00,
+	0x10, 0x0f, 0xef, 0x0f, 0x20, 0x20, 0x10, 0x0c, 0xdf, 0x0d, 0x00, 0xf0,
+	0xe3, 0x00, 0x00, 0xef,
+	0xf1, 0xd0, 0xce, 0xff, 0xef, 0xe0, 0xde, 0xed, 0xff, 0xef, 0xfd, 0xe0,
+	0x01, 0xf1, 0x0f, 0x1f,
+	0x30, 0x13, 0x22, 0x01, 0x10, 0x00, 0x00, 0x30, 0x0d, 0x0f, 0x1f, 0x31,
+	0x21, 0x20, 0x0e, 0x0e,
+	0x10, 0x01, 0xf0, 0xe0, 0x0f, 0x21, 0x00, 0xcf, 0xef, 0x0f, 0xff, 0xf0,
+	0xfd, 0xf1, 0x10, 0xf0,
+	0xef, 0x0f, 0x01, 0x01, 0x41, 0x22, 0x00, 0xd0, 0x13, 0x23, 0x3f, 0x0f,
+	0xec, 0xff, 0x00, 0xf1,
+	0x00, 0x0f, 0x02, 0x22, 0x01, 0x00, 0x23, 0x22, 0x10, 0x00, 0x10, 0x14,
+	0x0f, 0xff, 0x00, 0x12,
+	0x11, 0xfe, 0xed, 0x0f, 0x00, 0x00, 0x01, 0x15, 0x21, 0x00, 0xef, 0x22,
+	0x12, 0xf0, 0x04, 0x20,
+	0x00, 0xe2, 0xf0, 0xff, 0xed, 0x02, 0x02, 0xdf, 0x00, 0x00, 0x12, 0x00,
+	0x11, 0x01, 0xf0, 0x0f,
+	0x30, 0x0f, 0x20, 0x00, 0x2e, 0x30, 0x20, 0x10, 0x30, 0x42, 0x12, 0x23,
+	0x20, 0x00, 0x0e, 0x0e,
+	0x10, 0x55, 0xf0, 0xf0, 0x00, 0x00, 0x21, 0x0f, 0xee, 0x11, 0x02, 0x0e,
+	0xde, 0x00, 0xf0, 0xd0,
+	0xec, 0xf0, 0xfe, 0xd0, 0xef, 0xef, 0x10, 0x0d, 0xde, 0xed, 0x00, 0x30,
+	0x01, 0xf0, 0x10, 0x11,
+	0x11, 0x11, 0x01, 0x12, 0x0f, 0xde, 0x22, 0x35, 0x12, 0x00, 0x00, 0x00,
+	0x00, 0x20, 0xf0, 0x0e,
+	0x01, 0x15, 0x01, 0xf2, 0xf0, 0xfe, 0xef, 0x03, 0x02, 0xcf, 0xef, 0x00,
+	0xe2, 0xef, 0xfd, 0xef,
+	0xfe, 0x00, 0x00, 0x20, 0x1e, 0x2e, 0x11, 0x20, 0x21, 0x20, 0x20, 0x0d,
+	0x0c, 0x10, 0x10, 0xff,
+	0xdd, 0x0f, 0x43, 0x02, 0x1f, 0x00, 0x01, 0x02, 0x11, 0x00, 0x0d, 0x01,
+	0xf0, 0xfe, 0x00, 0xfe,
+	0xf0, 0x00, 0x00, 0xf0, 0xfd, 0xde, 0x00, 0x27, 0x31, 0x00, 0xd0, 0x1f,
+	0x10, 0xf3, 0x00, 0x00,
+	0x1f, 0x1e, 0x00, 0x11, 0x00, 0xf0, 0xfc, 0x00, 0x21, 0x33, 0xe0, 0x0d,
+	0x11, 0x12, 0x11, 0x22,
+	0x0f, 0xff, 0xfe, 0xf0, 0x00, 0x10, 0x02, 0xff, 0x0f, 0x0f, 0xfe, 0xfe,
+	0x14, 0x24, 0x01, 0x00,
+	0xfd, 0xff, 0x00, 0xf0, 0x03, 0x20, 0xf1, 0xf0, 0xfd, 0x01, 0x00, 0xee,
+	0xff, 0xff, 0xff, 0xe1,
+	0x02, 0x03, 0xf0, 0xe0, 0xee, 0x01, 0x22, 0x00, 0xf0, 0x0e, 0xfe, 0x0e,
+	0x0e, 0x10, 0x30, 0x0f,
+	0x20, 0x02, 0x22, 0x03, 0xff, 0x0f, 0x10, 0x12, 0x00, 0x03, 0x21, 0x01,
+	0xfe, 0x0f, 0x20, 0x0e,
+	0xdd, 0x00, 0x10, 0x14, 0x22, 0x00, 0xfe, 0xdf, 0xee, 0xf0, 0x00, 0x21,
+	0xff, 0xf0, 0xef, 0x00,
+	0x01, 0xf0, 0x0e, 0x00, 0x25, 0x31, 0x11, 0x10, 0x12, 0x11, 0x21, 0x00,
+	0x1f, 0x10, 0x00, 0x00,
+	0x00, 0x30, 0xf0, 0xf0, 0x00, 0x1f, 0xf0, 0xe0, 0x01, 0x03, 0x00, 0xee,
+	0x0d, 0xff, 0xff, 0xdc,
+	0x0f, 0x1f, 0x0e, 0x00, 0x0f, 0x0d, 0xee, 0xdc, 0x00, 0x21, 0x23, 0x10,
+	0x01, 0x01, 0xf2, 0xf1,
+	0x00, 0xed, 0x01, 0x22, 0x05, 0xff, 0xef, 0xf0, 0x00, 0x22, 0x00, 0x01,
+	0x10, 0x00, 0x24, 0x44,
+	0x00, 0xfe, 0x00, 0x0d, 0xf1, 0x35, 0x1f, 0x1f, 0x10, 0x00, 0xfe, 0x0e,
+	0x11, 0x54, 0x01, 0x10,
+	0x00, 0xeb, 0x01, 0x30, 0x01, 0x00, 0x22, 0x20, 0x00, 0x00, 0xe0, 0x12,
+	0x0f, 0xed, 0xe0, 0xff,
+	0xef, 0x01, 0x03, 0xe0, 0xcb, 0xdf, 0xee, 0x0f, 0xed, 0xdf, 0x00, 0x10,
+	0xf0, 0xec, 0xff, 0xfe,
+	0xfe, 0x00, 0x35, 0x00, 0xee, 0x0e, 0x00, 0x1f, 0x00, 0x12, 0x00, 0x0e,
+	0x0f, 0x0f, 0x20, 0x00,
+	0xe0, 0x00, 0x02, 0x10, 0xff, 0xdd, 0x00, 0x44, 0xf0, 0xbd, 0x01, 0x10,
+	0x00, 0xfd, 0xf0, 0xef,
+	0xef, 0x01, 0x24, 0x00, 0xed, 0x01, 0x11, 0x13, 0x01, 0x01, 0x0e, 0x0f,
+	0x21, 0x00, 0x0e, 0xee,
+	0x00, 0x00, 0x00, 0x31, 0x01, 0x01, 0x02, 0xf1, 0x01, 0x33, 0x00, 0x00,
+	0x0f, 0x00, 0x01, 0x0f,
+	0xff, 0x10, 0x0e, 0x0e, 0x10, 0x00, 0x00, 0x02, 0x13, 0x11, 0x00, 0x02,
+	0x11, 0x12, 0x25, 0x01,
+	0xf0, 0xef, 0x00, 0xff, 0x00, 0x00, 0xf3, 0xee, 0xfe, 0xf0, 0x02, 0xf0,
+	0xf1, 0x00, 0xf0, 0x26,
+	0x00, 0x0e, 0x00, 0x0e, 0x10, 0x10, 0x20, 0x10, 0x20, 0x30, 0x10, 0x1f,
+	0x00, 0x10, 0x20, 0x56,
+	0x1f, 0xff, 0x0e, 0x01, 0x43, 0x03, 0x00, 0x1d, 0x1d, 0xfe, 0x00, 0x31,
+	0xe0, 0xdc, 0x00, 0x00,
+	0xfd, 0x9b, 0x0f, 0xef, 0x00, 0x02, 0xde, 0xec, 0x00, 0x0f, 0xdf, 0xff,
+	0x00, 0xfe, 0x00, 0xe0,
+	0x00, 0x10, 0x01, 0xf2, 0x01, 0x14, 0x32, 0x0d, 0xfe, 0x35, 0x33, 0x0f,
+	0x00, 0x40, 0x22, 0x12,
+	0x21, 0x01, 0x00, 0x12, 0xf1, 0xf0, 0xf2, 0xf1, 0xf0, 0x02, 0xe2, 0x01,
+	0xde, 0xec, 0xff, 0x00,
+	0xde, 0xff, 0x00, 0x10, 0x20, 0x00, 0x20, 0x21, 0xf0, 0xf0, 0x11, 0x20,
+	0x42, 0x10, 0x0e, 0x00,
+	0x0f, 0x0f, 0x0f, 0x21, 0xee, 0xdd, 0x77, 0x32, 0xcb, 0x22, 0x12, 0x0f,
+	0x0f, 0xff, 0x01, 0x12,
+	0x0f, 0xee, 0x00, 0x33, 0x12, 0xdf, 0xed, 0x00, 0x11, 0x02, 0xf1, 0x14,
+	0x10, 0xee, 0x01, 0x24,
+	0x00, 0xe0, 0x00, 0x11, 0x20, 0x0e, 0x01, 0x11, 0x1f, 0x0e, 0x1f, 0x10,
+	0x33, 0x00, 0xe0, 0x10,
+	0x21, 0x22, 0x1f, 0xfd, 0xe0, 0x30, 0x13, 0xff, 0xfc, 0x00, 0x11, 0x13,
+	0x00, 0xfe, 0xf0, 0xfe,
+	0x01, 0x13, 0x00, 0xff, 0xcc, 0x00, 0xf3, 0x21, 0x11, 0xee, 0xe0, 0x00,
+	0xf0, 0xcd, 0x01, 0x24,
+	0xff, 0xbd, 0x01, 0xe2, 0x00, 0xed, 0xef, 0x10, 0x2f, 0x0f, 0x0f, 0x32,
+	0x11, 0x1d, 0x0c, 0x0d,
+	0xf0, 0x20, 0x12, 0x00, 0x00, 0x03, 0xe0, 0x0d, 0xee, 0x0f, 0x00, 0x12,
+	0x23, 0x0e, 0xfc, 0x00,
+	0x10, 0x00, 0x01, 0x00, 0x00, 0xe1, 0x02, 0x01, 0xf0, 0xff, 0xe0, 0xf1,
+	0xef, 0x00, 0x00, 0xf1,
+	0x02, 0x01, 0x00, 0x0f, 0x2e, 0x0d, 0xf0, 0x46, 0x32, 0x01, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0x18, 0x17, 0x77, 0x18, 0xbc, 0x16, 0xd3, 0x17, 0x00, 0x1f, 0x20, 0x32,
+	0x12, 0xed, 0xfe, 0x01,
+	0x11, 0x11, 0x00, 0xe0, 0x00, 0xf0, 0x02, 0xff, 0xee, 0x00, 0x0f, 0x00,
+	0xef, 0x0d, 0x0d, 0x1e,
+	0x0c, 0x0d, 0xe0, 0x20, 0xf0, 0xee, 0x0f, 0x20, 0x01, 0xe2, 0x02, 0xf1,
+	0xf0, 0xff, 0x0f, 0x01,
+	0x00, 0xf0, 0x00, 0x12, 0x00, 0x00, 0xfe, 0x0f, 0xef, 0x02, 0x15, 0x21,
+	0x03, 0xe0, 0xe0, 0x02,
+	0x01, 0xe0, 0xff, 0x00, 0x20, 0x2e, 0x00, 0x0f, 0xfd, 0x10, 0x32, 0x20,
+	0x11, 0x1f, 0x00, 0x01,
+	0x00, 0x10, 0x11, 0x00, 0x0e, 0x31, 0x11, 0x01, 0x1f, 0x0f, 0xf1, 0xff,
+	0xfe, 0x0e, 0xcd, 0xf1,
+	0x17, 0xd0, 0xcd, 0x01, 0xde, 0xcd, 0xee, 0x01, 0xf2, 0xef, 0xf0, 0x0d,
+	0xfd, 0xff, 0x00, 0xef,
+	0x0e, 0x02, 0x22, 0x0f, 0xf0, 0x00, 0x00, 0x01, 0x33, 0x0f, 0xfc, 0x30,
+	0x20, 0x20, 0x0f, 0x00,
+	0x41, 0x00, 0xee, 0x31, 0x0f, 0xce, 0x10, 0x10, 0x02, 0xf0, 0xed, 0xff,
+	0xf0, 0x01, 0x0f, 0xef,
+	0x00, 0x00, 0x0d, 0xef, 0xfe, 0x12, 0x33, 0x22, 0x00, 0xf0, 0xf0, 0x00,
+	0xf0, 0x22, 0x24, 0xef,
+	0xfe, 0x11, 0x12, 0x23, 0x00, 0xff, 0x0e, 0x01, 0x32, 0x00, 0x0f, 0x20,
+	0x13, 0x00, 0xde, 0x00,
+	0x31, 0x10, 0xfe, 0x13, 0x13, 0xfe, 0xff, 0x02, 0x03, 0x10, 0x00, 0xf0,
+	0x03, 0x02, 0x20, 0x00,
+	0x00, 0xef, 0xff, 0x02, 0xe3, 0xf0, 0x11, 0x00, 0xd0, 0xdc, 0xf0, 0x01,
+	0x20, 0x01, 0x34, 0x00,
+	0x0f, 0x00, 0x20, 0x11, 0x2f, 0x0e, 0x2e, 0x20, 0x35, 0x21, 0x00, 0x0e,
+	0x20, 0x20, 0x21, 0x23,
+	0x20, 0x10, 0x22, 0x12, 0x21, 0x21, 0xff, 0x00, 0x30, 0x01, 0xf1, 0xef,
+	0xff, 0x0f, 0xdd, 0xdf,
+	0xf0, 0x00, 0xe0, 0xfd, 0xde, 0x00, 0xfc, 0x00, 0x01, 0xff, 0x0e, 0x00,
+	0x10, 0x0f, 0x11, 0x00,
+	0xdd, 0xee, 0x00, 0x00, 0x11, 0x22, 0xff, 0xf0, 0x03, 0x21, 0x00, 0x1e,
+	0x23, 0x22, 0x01, 0x12,
+	0x23, 0x00, 0xe0, 0xd0, 0x04, 0xd0, 0xd0, 0xe0, 0x02, 0xe0, 0xdd, 0x22,
+	0xf0, 0xcd, 0x01, 0x20,
+	0x00, 0x0e, 0xfd, 0x21, 0x11, 0x10, 0x0f, 0x0f, 0x10, 0x0e, 0x00, 0x21,
+	0x0f, 0x00, 0x31, 0x00,
+	0xfc, 0xff, 0x10, 0x01, 0x00, 0x20, 0x22, 0x2f, 0xfd, 0x00, 0x00, 0x00,
+	0x00, 0x22, 0xf0, 0x0e,
+	0xf0, 0xfd, 0x00, 0x00, 0x01, 0x00, 0x0f, 0xdd, 0xef, 0x24, 0x05, 0x00,
+	0xe0, 0x00, 0x00, 0x12,
+	0x11, 0x00, 0xed, 0x10, 0x14, 0x20, 0x0d, 0xfd, 0x00, 0x00, 0x23, 0x10,
+	0x0f, 0xec, 0xf0, 0x10,
+	0x35, 0x01, 0x00, 0x13, 0x00, 0x00, 0xdb, 0xff, 0x10, 0x31, 0x02, 0xf0,
+	0x2f, 0x0e, 0xfd, 0x02,
+	0x11, 0x00, 0x00, 0x0f, 0xf1, 0xf1, 0x12, 0x01, 0x01, 0xe1, 0xff, 0xff,
+	0x01, 0x01, 0xdf, 0xdc,
+	0x02, 0xf1, 0x00, 0xfe, 0xed, 0x10, 0x12, 0x1f, 0x1e, 0x12, 0x10, 0x00,
+	0x00, 0x0e, 0xf0, 0x03,
+	0x03, 0x0e, 0xff, 0x00, 0xee, 0x0f, 0xf0, 0x11, 0x32, 0x10, 0x23, 0x10,
+	0x00, 0x1f, 0x20, 0x00,
+	0xf0, 0x20, 0x11, 0x00, 0xe0, 0xe0, 0xff, 0xf0, 0xe0, 0xce, 0xfd, 0x02,
+	0x00, 0x00, 0xf1, 0x00,
+	0xff, 0x10, 0x32, 0x1f, 0x0e, 0x0f, 0x01, 0x33, 0x33, 0x10, 0x10, 0x11,
+	0x01, 0x14, 0x11, 0x01,
+	0xf0, 0x0e, 0xf0, 0xf0, 0xe2, 0x01, 0x12, 0x10, 0x0e, 0xff, 0xfe, 0xfd,
+	0xfd, 0x10, 0x0e, 0x2f,
+	0x30, 0xf0, 0x00, 0xfd, 0xdc, 0x00, 0x0f, 0xfd, 0x00, 0x01, 0xd0, 0xee,
+	0x00, 0x12, 0x11, 0x0d,
+	0xf0, 0x11, 0x0e, 0x0f, 0x12, 0x24, 0x23, 0xe0, 0xdc, 0x00, 0x17, 0x00,
+	0xcd, 0x13, 0x10, 0x0f,
+	0x00, 0xf0, 0x25, 0x20, 0x21, 0x00, 0xf0, 0x00, 0x32, 0x22, 0x00, 0xee,
+	0x00, 0x0f, 0x1f, 0x1f,
+	0x33, 0x33, 0xed, 0xfe, 0x53, 0x1f, 0x0e, 0x12, 0x25, 0x20, 0x01, 0x01,
+	0x03, 0xde, 0xfd, 0x00,
+	0xd1, 0xe0, 0xde, 0xed, 0xff, 0xf1, 0x00, 0xdd, 0xfb, 0xee, 0x10, 0x2f,
+	0x00, 0xf0, 0xf0, 0xf0,
+	0xee, 0xf0, 0xee, 0x0f, 0xef, 0xe0, 0xf0, 0xcc, 0x00, 0x1f, 0x00, 0x1f,
+	0x0e, 0x10, 0x21, 0x11,
+	0x20, 0x20, 0x0d, 0x0f, 0x10, 0x02, 0xe2, 0x01, 0x01, 0xff, 0xdc, 0xf0,
+	0x04, 0xff, 0xfe, 0x00,
+	0x00, 0xf0, 0x00, 0xff, 0x20, 0x00, 0xd0, 0x0e, 0x00, 0x11, 0x23, 0x01,
+	0x00, 0xf0, 0xf1, 0x00,
+	0x00, 0x0f, 0xf0, 0x01, 0x01, 0x10, 0xff, 0x00, 0x0f, 0xf0, 0x00, 0x20,
+	0x01, 0x00, 0x0f, 0x20,
+	0x01, 0x10, 0x10, 0x12, 0x22, 0x02, 0x20, 0x00, 0xff, 0x00, 0x0f, 0xff,
+	0x03, 0x23, 0x01, 0x01,
+	0x02, 0x00, 0xf0, 0xf0, 0xf0, 0x12, 0x12, 0xff, 0xdc, 0xe0, 0x04, 0xf0,
+	0xef, 0x0f, 0x0e, 0xfd,
+	0xe0, 0x10, 0x03, 0x21, 0x21, 0x0e, 0x00, 0x00, 0x10, 0x0f, 0x21, 0x44,
+	0x10, 0x0c, 0x0e, 0x10,
+	0x31, 0x44, 0x23, 0x20, 0x0e, 0x2e, 0x00, 0x00, 0x31, 0x24, 0x0f, 0xfe,
+	0x23, 0xdd, 0xcc, 0x00,
+	0x02, 0xfe, 0xbc, 0x00, 0xfe, 0xef, 0xff, 0xf0, 0x00, 0xff, 0xed, 0xe0,
+	0x0f, 0x10, 0xff, 0xef,
+	0x11, 0x14, 0x0e, 0xcb, 0x0f, 0xf0, 0x00, 0x02, 0x01, 0x00, 0x20, 0x46,
+	0x00, 0x21, 0x22, 0x01,
+	0x22, 0x34, 0x11, 0x0e, 0xee, 0xe0, 0x01, 0xf2, 0xdf, 0xee, 0x00, 0x02,
+	0xf1, 0xe1, 0x00, 0xfd,
+	0x00, 0x10, 0x02, 0x22, 0xfe, 0xec, 0x02, 0x31, 0x00, 0x0d, 0xfe, 0x00,
+	0x10, 0x01, 0x32, 0x22,
+	0x00, 0x00, 0x10, 0x1f, 0x00, 0x20, 0x21, 0x33, 0x0f, 0x0f, 0x00, 0x00,
+	0x10, 0x10, 0x11, 0x00,
+	0x10, 0x00, 0xff, 0x00, 0x0f, 0xe0, 0x22, 0x14, 0x00, 0x00, 0xe0, 0xef,
+	0xe0, 0xf0, 0x01, 0x00,
+	0x23, 0x13, 0x10, 0x12, 0x20, 0xee, 0xed, 0x20, 0x10, 0x00, 0x00, 0x12,
+	0x01, 0x32, 0x00, 0x0d,
+	0x0f, 0xf0, 0x00, 0x13, 0x12, 0x10, 0x31, 0x02, 0xff, 0xfd, 0xfe, 0x01,
+	0x31, 0x00, 0x30, 0x0e,
+	0xdc, 0xef, 0x0f, 0x12, 0x22, 0xf0, 0xee, 0xf0, 0x01, 0x03, 0x01, 0xf0,
+	0xed, 0xf0, 0xf1, 0x03,
+	0xf0, 0xd0, 0xfc, 0xef, 0x0f, 0xfd, 0xee, 0x11, 0x35, 0x00, 0x0e, 0x10,
+	0x00, 0x00, 0x20, 0x0f,
+	0x0f, 0x01, 0x00, 0xff, 0xfc, 0x00, 0x0f, 0xee, 0x00, 0xef, 0x0e, 0x10,
+	0x10, 0x22, 0x22, 0x01,
+	0x10, 0x02, 0x10, 0xfe, 0xf0, 0x00, 0x00, 0xc0, 0xef, 0xe0, 0x01, 0x01,
+	0xf1, 0xf2, 0xf0, 0xf0,
+	0xff, 0x0f, 0xfb, 0x01, 0x75, 0x0f, 0xfe, 0x23, 0x10, 0x0f, 0x00, 0x00,
+	0x23, 0x23, 0x00, 0xf0,
+	0x02, 0x23, 0x12, 0xdf, 0xfe, 0x01, 0x00, 0x0f, 0x00, 0x00, 0x20, 0x0f,
+	0x0f, 0x11, 0xfe, 0xfb,
+	0x00, 0x2f, 0x10, 0x00, 0xef, 0x0f, 0x10, 0x00, 0xfd, 0xfd, 0xfe, 0xe0,
+	0x0f, 0xff, 0x00, 0x0f,
+	0x2f, 0xfe, 0xf0, 0x15, 0x22, 0x00, 0xcf, 0x00, 0x02, 0xf0, 0xf1, 0x00,
+	0xf2, 0xf0, 0x0f, 0xef,
+	0x00, 0x01, 0xf0, 0x00, 0xf0, 0x03, 0x30, 0xf1, 0x02, 0xf2, 0x01, 0x00,
+	0x11, 0x0f, 0xfd, 0x0e,
+	0x0d, 0x10, 0x32, 0x10, 0x0f, 0x10, 0x40, 0x11, 0x12, 0x12, 0x23, 0x34,
+	0xf0, 0xee, 0xff, 0xf0,
+	0x02, 0xf0, 0xef, 0x0f, 0xee, 0xdb, 0xfe, 0xef, 0xf0, 0x02, 0x00, 0x0d,
+	0xe0, 0x00, 0xe0, 0xff,
+	0xf0, 0xe0, 0x00, 0x0d, 0xdb, 0xef, 0xfe, 0xe0, 0x02, 0xee, 0xfd, 0x10,
+	0x00, 0x0f, 0x10, 0x55,
+	0x12, 0x10, 0x0d, 0x11, 0x32, 0xfe, 0x0f, 0x1f, 0x10, 0x0f, 0xef, 0x0e,
+	0xef, 0x00, 0x00, 0x00,
+	0xff, 0x00, 0x11, 0x0e, 0xde, 0x00, 0x10, 0x10, 0x12, 0x00, 0xff, 0x10,
+	0x33, 0xe0, 0x0f, 0x00,
+	0x02, 0x00, 0xf1, 0x03, 0x02, 0x01, 0x01, 0xe0, 0x00, 0xfd, 0x00, 0x00,
+	0x20, 0x00, 0x01, 0xff,
+	0x1f, 0x51, 0x01, 0xf1, 0x32, 0xef, 0x00, 0x54, 0x01, 0x10, 0xef, 0xef,
+	0x0f, 0x10, 0x33, 0x00,
+	0xef, 0x0f, 0xf0, 0x15, 0x12, 0xf0, 0x00, 0xf0, 0x03, 0xe0, 0xd0, 0xff,
+	0xef, 0x00, 0x00, 0x00,
+	0x0f, 0xef, 0x0f, 0x22, 0x00, 0xf0, 0x1f, 0x20, 0x10, 0x32, 0x13, 0x20,
+	0x01, 0x00, 0x54, 0x00,
+	0x0e, 0x0c, 0x21, 0x52, 0x13, 0x12, 0x1f, 0x20, 0x20, 0x33, 0x13, 0x1f,
+	0x10, 0x20, 0xf0, 0x02,
+	0x02, 0x01, 0xe0, 0xed, 0xff, 0x0f, 0xde, 0xee, 0xe0, 0xfe, 0x00, 0x0f,
+	0xfd, 0x00, 0x01, 0xf0,
+	0xef, 0x0d, 0x0f, 0x20, 0x00, 0xfc, 0xee, 0xfd, 0x00, 0x01, 0x02, 0x10,
+	0x00, 0x01, 0x14, 0x00,
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+	0x00, 0xf1, 0xd1, 0x03,
+	0x03, 0xe0, 0x0c, 0xf0, 0x00, 0x01, 0x00, 0x0e, 0x00, 0xed, 0x0d, 0x10,
+	0x21, 0x00, 0x01, 0x10,
+	0x10, 0x0e, 0xff, 0x01, 0x11, 0x21, 0x23, 0x20, 0xff, 0xfd, 0x20, 0x3f,
+	0x1e, 0xec, 0x0f, 0x0f,
+	0x1f, 0x21, 0x02, 0x01, 0xf0, 0x0f, 0xe0, 0xf0, 0x00, 0x12, 0x03, 0xf1,
+	0x00, 0xdd, 0x00, 0x00,
+	0xff, 0x0e, 0x00, 0x11, 0x34, 0x01, 0x00, 0x10, 0x00, 0xde, 0xee, 0x47,
+	0x00, 0xae, 0x11, 0x11,
+	0xfe, 0x00, 0x00, 0x00, 0x20, 0x00, 0x20, 0x10, 0x1f, 0x0f, 0x02, 0x34,
+	0x10, 0x00, 0x00, 0x0e,
+	0x00, 0xf1, 0xf0, 0xfc, 0x0f, 0x0e, 0xef, 0x32, 0x12, 0xe0, 0x02, 0x22,
+	0xf1, 0x00, 0xde, 0x00,
+	0x13, 0x00, 0xe0, 0x00, 0xff, 0xff, 0xef, 0x0e, 0xee, 0x0f, 0x23, 0x11,
+	0x0f, 0xde, 0x10, 0x25,
+	0x11, 0x01, 0xdf, 0x20, 0x12, 0xdc, 0xfe, 0x0f, 0x0f, 0x00, 0x10, 0x11,
+	0x21, 0x00, 0x00, 0x11,
+	0x12, 0x10, 0x02, 0x00, 0x00, 0x02, 0x02, 0xf0, 0x0f, 0xcd, 0xed, 0x0f,
+	0x0f, 0xee, 0xff, 0x13,
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+	0x00, 0x01, 0x02, 0x12,
+	0x01, 0x00, 0x00, 0xf1, 0x04, 0x11, 0x00, 0x00, 0x01, 0x12, 0x0e, 0x0f,
+	0xff, 0x0f, 0x20, 0x10,
+	0x0e, 0x0f, 0xfb, 0x00, 0x0f, 0x00, 0x00, 0x00, 0xff, 0xfd, 0xff, 0x0d,
+	0x0f, 0x0f, 0x10, 0x0f,
+	0xfe, 0xed, 0x00, 0x31, 0x11, 0x01, 0x22, 0x00, 0x00, 0xd0, 0x03, 0xf0,
+	0xed, 0x00, 0x01, 0xf1,
+	0xf1, 0x00, 0xe0, 0x03, 0x10, 0x0f, 0xe0, 0x00, 0x21, 0x02, 0x24, 0x11,
+	0x00, 0x0f, 0x10, 0x31,
+	0x00, 0xff, 0x22, 0x00, 0x2f, 0x20, 0x0e, 0xee, 0x31, 0x45, 0x00, 0x02,
+	0x21, 0x54, 0x02, 0xed,
+	0xff, 0x12, 0x00, 0xff, 0x00, 0xe0, 0xf0, 0x0f, 0xed, 0xdd, 0xf0, 0x12,
+	0x21, 0xed, 0xcc, 0x11,
+	0x13, 0xed, 0xed, 0xff, 0x0f, 0xef, 0xee, 0xee, 0xed, 0x0f, 0xfd, 0x00,
+	0xee, 0xee, 0x00, 0x44,
+	0x01, 0x00, 0x1e, 0x0f, 0x0e, 0x11, 0x12, 0x00, 0xfe, 0x0f, 0x0f, 0x11,
+	0x01, 0xcf, 0xfe, 0x00,
+	0x03, 0x00, 0xff, 0xff, 0x01, 0x11, 0xf1, 0x0f, 0x0e, 0x0f, 0x00, 0x30,
+	0x00, 0xef, 0x00, 0x02,
+	0xf0, 0xff, 0x01, 0x01, 0xdf, 0x00, 0xe2, 0x13, 0x00, 0xfe, 0x0e, 0x03,
+	0x00, 0x1f, 0x1f, 0x00,
+	0x10, 0x2f, 0x0e, 0x00, 0xf0, 0x11, 0x24, 0x12, 0x1f, 0x00, 0x01, 0x01,
+	0x01, 0x00, 0x10, 0x1e,
+	0x11, 0x22, 0x01, 0x00, 0xf1, 0xf1, 0x01, 0x10, 0xf0, 0xe1, 0xf2, 0x01,
+	0xf1, 0xe0, 0x0f, 0xed,
+	0xff, 0xee, 0xff, 0x00, 0x10, 0x00, 0x0e, 0x00, 0x00, 0xfe, 0x01, 0x25,
+	0x32, 0x00, 0x0f, 0x1f,
+	0x11, 0x12, 0x32, 0x00, 0x1d, 0x0d, 0x0d, 0x10, 0x42, 0x22, 0x10, 0x21,
+	0x00, 0x42, 0x13, 0x20,
+	0xf0, 0x01, 0x01, 0xf1, 0x00, 0x0e, 0xed, 0x00, 0xff, 0xff, 0xdf, 0x0f,
+	0xe0, 0xfd, 0xed, 0xf0,
+	0x00, 0x01, 0xf1, 0x0e, 0xfe, 0xfd, 0xdf, 0x0f, 0x00, 0x00, 0x0e, 0x10,
+	0x01, 0x10, 0x00, 0x02,
+	0x02, 0x11, 0x11, 0x12, 0x10, 0x02, 0x01, 0x02, 0xf0, 0x01, 0x0f, 0xf0,
+	0xef, 0x02, 0x13, 0x00,
+	0xfe, 0xdf, 0x00, 0x11, 0xf0, 0xfe, 0x02, 0x00, 0x11, 0x11, 0x0e, 0xde,
+	0x1f, 0x00, 0x10, 0x31,
+	0x02, 0xff, 0x00, 0x02, 0x00, 0x00, 0x0f, 0x22, 0x33, 0x10, 0x20, 0x10,
+	0x10, 0x1f, 0xfd, 0x10,
+	0x20, 0x12, 0x0f, 0x00, 0x10, 0x01, 0xf1, 0x02, 0x10, 0x0f, 0x0f, 0xf0,
+	0x03, 0x12, 0x00, 0x0d,
+	0x0e, 0xff, 0x10, 0x22, 0x01, 0xf2, 0x12, 0x00, 0x00, 0x01, 0xe1, 0x01,
+	0xe0, 0x01, 0x00, 0xf0,
+	0x11, 0x00, 0xdd, 0x21, 0x32, 0x20, 0x10, 0x1f, 0xff, 0x00, 0x31, 0x01,
+	0x01, 0x21, 0x22, 0x00,
+	0x0f, 0x0f, 0xef, 0x0f, 0x20, 0x0d, 0xff, 0xfe, 0xff, 0x00, 0x01, 0x22,
+	0x00, 0x00, 0xe0, 0xff,
+	0xe0, 0x13, 0x01, 0xef, 0x0f, 0xff, 0xef, 0x0e, 0x01, 0xf0, 0x0f, 0x00,
+	0xf1, 0x01, 0xf0, 0x0f,
+	0x00, 0xff, 0xf0, 0x00, 0x20, 0x2f, 0x00, 0xff, 0x00, 0x0d, 0x2e, 0x0f,
+	0x1e, 0x10, 0x12, 0x02,
+	0x00, 0xef, 0xf0, 0x12, 0x22, 0xd0, 0xee, 0xf0, 0x01, 0x02, 0x00, 0xec,
+	0xef, 0x00, 0x0f, 0xde,
+	0x00, 0x01, 0x12, 0x00, 0xf0, 0x01, 0x00, 0x00, 0xf0, 0x11, 0x11, 0x10,
+	0x12, 0x10, 0x00, 0x0f,
+	0x10, 0x00, 0x00, 0x12, 0x02, 0xf1, 0x02, 0x02, 0x02, 0x00, 0x20, 0x00,
+	0x02, 0x11, 0x0f, 0xfe,
+	0x2f, 0x1f, 0xfe, 0x0e, 0x20, 0xfe, 0x00, 0x00, 0x00, 0x0e, 0xee, 0xf0,
+	0xf0, 0x00, 0x00, 0x0b,
+	0xff, 0x00, 0x1e, 0x0f, 0x1f, 0x0e, 0x01, 0x32, 0x11, 0x02, 0x01, 0xf0,
+	0xde, 0xf0, 0xf2, 0xef,
+	0x00, 0x01, 0xf2, 0xe0, 0xd0, 0x0f, 0x00, 0x00, 0xfd, 0xce, 0x00, 0x02,
+	0x13, 0x12, 0x12, 0x00,
+	0x0f, 0x0f, 0x00, 0x23, 0x00, 0x00, 0x10, 0x1e, 0x0f, 0x00, 0x12, 0x20,
+	0x12, 0x22, 0x10, 0x11,
+	0x01, 0x00, 0x02, 0x11, 0x23, 0x0e, 0x0f, 0xdf, 0x00, 0x12, 0x00, 0xeb,
+	0x00, 0xef, 0x0f, 0xf0,
+	0xf0, 0xee, 0xff, 0xef, 0xfe, 0xdc, 0xfe, 0xe0, 0x00, 0xf0, 0xee, 0x0e,
+	0xff, 0xee, 0x00, 0x0f,
+	0x0f, 0x00, 0x12, 0x10, 0x00, 0x01, 0x00, 0x0f, 0x0f, 0x00, 0x22, 0x1f,
+	0xfa, 0x00, 0x20, 0x00,
+	0xf0, 0x00, 0x00, 0x0e, 0x10, 0x00, 0xef, 0x10, 0x22, 0x00, 0x0e, 0x00,
+	0x10, 0x21, 0x03, 0x00,
+	0x00, 0xff, 0xee, 0xf0, 0x03, 0x13, 0xe0, 0xe0, 0xf1, 0x02, 0x01, 0x00,
+	0x00, 0xfe, 0x00, 0x12,
+	0x21, 0x11, 0x10, 0xfd, 0xff, 0x00, 0x31, 0x0f, 0x0f, 0x11, 0x11, 0x00,
+	0x00, 0x35, 0x10, 0x10,
+	0x00, 0x0f, 0xf0, 0x12, 0x22, 0x10, 0x02, 0x00, 0xdf, 0x02, 0x00, 0xc0,
+	0x01, 0x11, 0xf0, 0xe0,
+	0xed, 0xf0, 0xf0, 0xff, 0x00, 0x02, 0x00, 0x00, 0x0f, 0xff, 0x00, 0x21,
+	0x01, 0x00, 0x23, 0x33,
+	0x00, 0x10, 0x00, 0x20, 0x21, 0x33, 0x2f, 0x0e, 0x3f, 0x30, 0x12, 0x12,
+	0x31, 0x21, 0x21, 0x23,
+	0x22, 0x01, 0xf0, 0x02, 0x01, 0x0f, 0xfe, 0xee, 0xf0, 0x01, 0x12, 0x00,
+	0xfc, 0xf0, 0xfe, 0xef,
+	0xff, 0xff, 0x00, 0x12, 0x10, 0x0f, 0xdc, 0xfd, 0xff, 0x0e, 0x00, 0x1e,
+	0x00, 0x00, 0x10, 0x10,
+	0x00, 0xef, 0x00, 0x0f, 0x00, 0xf0, 0x05, 0x22, 0xe0, 0xff, 0x02, 0x02,
+	0xfe, 0xee, 0x00, 0x14,
+	0x01, 0xf0, 0xfe, 0xff, 0xdf, 0x01, 0x01, 0xf1, 0xe0, 0x00, 0xf0, 0x0f,
+	0xf0, 0x00, 0x10, 0x00,
+	0x01, 0x31, 0x0e, 0x0f, 0x0f, 0x10, 0x00, 0x0f, 0x0f, 0x22, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0xa6, 0xed, 0x88, 0xf3, 0x86, 0xed, 0x56, 0xf3, 0x00, 0x10, 0x11, 0xff,
+	0x0f, 0x0e, 0x0f, 0x20,
+	0x00, 0x00, 0xf0, 0x00, 0x0e, 0x10, 0x10, 0xff, 0x00, 0x00, 0x02, 0x01,
+	0x00, 0x00, 0x00, 0xf0,
+	0x00, 0x00, 0x0f, 0x10, 0xf1, 0xf2, 0xf0, 0xf0, 0x01, 0x02, 0x02, 0x0f,
+	0x0f, 0xf0, 0xee, 0x00,
+	0x10, 0x22, 0x10, 0x11, 0x20, 0x00, 0x0f, 0x00, 0x10, 0x0f, 0xfe, 0x00,
+	0x21, 0x13, 0x12, 0x01,
+	0x00, 0xdd, 0x0f, 0x0f, 0x10, 0x0e, 0xe0, 0x00, 0x01, 0x21, 0x10, 0x0f,
+	0xee, 0x10, 0x0f, 0xff,
+	0x02, 0x33, 0xf0, 0x0f, 0xf1, 0xe1, 0x02, 0xf1, 0xff, 0xfe, 0x01, 0x00,
+	0xf0, 0xff, 0x0f, 0x00,
+	0x01, 0x0e, 0x00, 0x20, 0x00, 0x20, 0x01, 0x01, 0x0f, 0x00, 0x20, 0x02,
+	0x00, 0xf1, 0x01, 0xe0,
+	0x00, 0xff, 0x00, 0x11, 0x00, 0xff, 0x00, 0x02, 0x10, 0xee, 0xfd, 0x01,
+	0x00, 0x00, 0x01, 0x0e,
+	0xfe, 0x01, 0xf0, 0xf0, 0x00, 0x10, 0x0c, 0x00, 0x01, 0x00, 0x01, 0x34,
+	0x10, 0xf0, 0x0f, 0x11,
+	0x01, 0xf4, 0x01, 0x02, 0xde, 0x0f, 0x02, 0x10, 0x01, 0x11, 0x21, 0x00,
+	0x02, 0x1f, 0x0d, 0x0e,
+	0x10, 0x13, 0x10, 0xe0, 0x0d, 0xff, 0xfe, 0xfd, 0xfe, 0x0d, 0x0f, 0x20,
+	0x0f, 0x1f, 0x1f, 0x1f,
+	0x10, 0x0e, 0x10, 0x0e, 0x10, 0x10, 0x01, 0xd0, 0x03, 0x10, 0xe0, 0xef,
+	0xf0, 0xf0, 0xdf, 0x00,
+	0x01, 0x12, 0xf0, 0x0f, 0x00, 0x03, 0x11, 0xf1, 0xf0, 0xff, 0x00, 0x00,
+	0xf0, 0x13, 0x11, 0x10,
+	0x10, 0x21, 0x12, 0x10, 0x00, 0x0f, 0x0f, 0x10, 0x12, 0x32, 0x20, 0x02,
+	0x11, 0x00, 0x22, 0x01,
+	0x00, 0x23, 0x10, 0x0f, 0x01, 0xf1, 0x02, 0x00, 0x01, 0x00, 0xfc, 0xdd,
+	0x00, 0x10, 0xef, 0xff,
+	0x00, 0xfe, 0xff, 0xee, 0xf0, 0x0e, 0x00, 0x00, 0xfe, 0xfd, 0x0f, 0xee,
+	0x0f, 0x00, 0xf0, 0xf0,
+	0xfe, 0x0f, 0xde, 0x00, 0x10, 0x00, 0xed, 0x00, 0x12, 0x00, 0xf0, 0x0f,
+	0x00, 0x00, 0x00, 0x0f,
+	0x00, 0x0f, 0x02, 0x00, 0xff, 0xff, 0x10, 0x00, 0x0f, 0x00, 0xee, 0x00,
+	0x11, 0x34, 0x12, 0xfe,
+	0xcc, 0x01, 0x01, 0xe0, 0x01, 0xf0, 0xf0, 0xf0, 0xf0, 0x00, 0x02, 0x00,
+	0x02, 0x20, 0xff, 0x0e,
+	0x00, 0x00, 0x20, 0x12, 0xfd, 0xfd, 0x01, 0x22, 0x01, 0x00, 0x20, 0x10,
+	0x13, 0x10, 0x21, 0x11,
+	0x03, 0x10, 0x01, 0x00, 0x00, 0x10, 0x0f, 0x00, 0xe0, 0xf0, 0x02, 0x00,
+	0xde, 0xf0, 0xf0, 0xdd,
+	0x00, 0xf0, 0x00, 0x00, 0x00, 0xf1, 0xf0, 0xf0, 0x0f, 0xff, 0x00, 0x00,
+	0xf0, 0x00, 0x21, 0x00,
+	0x1f, 0x0f, 0x20, 0x10, 0x35, 0x30, 0x10, 0x1f, 0x20, 0x34, 0x20, 0x1f,
+	0x0f, 0x01, 0x22, 0x23,
+	0x10, 0x10, 0x00, 0xf0, 0xfd, 0x01, 0x03, 0x12, 0x22, 0x00, 0xf0, 0xee,
+	0xff, 0xff, 0xdd, 0x0e,
+	0xff, 0x0e, 0xfe, 0xf0, 0x01, 0xfe, 0xfe, 0x0e, 0x00, 0x0d, 0x00, 0x02,
+	0x00, 0xff, 0x11, 0x01,
+	0x00, 0x01, 0xf0, 0xe0, 0x11, 0x22, 0x00, 0xed, 0xf1, 0x02, 0x32, 0x11,
+	0xf0, 0x00, 0xf1, 0x00,
+	0xf0, 0xff, 0xee, 0x00, 0x12, 0xf0, 0xe0, 0xef, 0x00, 0x00, 0xff, 0xf0,
+	0x00, 0x11, 0x33, 0x11,
+	0x10, 0x00, 0x00, 0x01, 0x30, 0x00, 0xf0, 0x1f, 0x1f, 0x11, 0x02, 0x22,
+	0x11, 0x00, 0x20, 0x01,
+	0x0f, 0x10, 0x12, 0x21, 0x0f, 0xda, 0x00, 0x31, 0x21, 0x00, 0xe0, 0x1f,
+	0x10, 0x0e, 0xf0, 0x00,
+	0x10, 0x13, 0x33, 0x02, 0xef, 0x00, 0x00, 0xe1, 0xf1, 0xd0, 0x0f, 0x01,
+	0x01, 0xf1, 0x01, 0x00,
+	0xff, 0x0f, 0x30, 0x01, 0x20, 0x10, 0x20, 0x24, 0x12, 0x11, 0x11, 0xed,
+	0xdd, 0x00, 0x00, 0x21,
+	0x01, 0x32, 0x0f, 0x0d, 0x00, 0x00, 0x00, 0x2e, 0x01, 0x00, 0x0d, 0x0f,
+	0x0f, 0x0f, 0xf0, 0xf0,
+	0xe0, 0x0f, 0x02, 0xf0, 0xf2, 0xd0, 0x0f, 0x00, 0x03, 0x0f, 0x00, 0x0e,
+	0xef, 0xef, 0xff, 0x0f,
+	0x2f, 0x0f, 0x0f, 0x21, 0x01, 0x11, 0x11, 0x11, 0x31, 0xfe, 0xff, 0x0f,
+	0x10, 0x10, 0x01, 0x01,
+	0x10, 0xff, 0xdc, 0xf0, 0xf1, 0xf1, 0x02, 0x00, 0xfc, 0xf1, 0xf1, 0xe0,
+	0xff, 0x00, 0x20, 0xf0,
+	0xef, 0x0e, 0x00, 0x00, 0x10, 0xee, 0x00, 0x0f, 0xf0, 0x00, 0x04, 0x10,
+	0x00, 0x01, 0x12, 0x10,
+	0x10, 0x00, 0x02, 0x22, 0x02, 0xe1, 0x00, 0xfd, 0x0f, 0x00, 0x22, 0x24,
+	0x01, 0x00, 0x03, 0x01,
+	0x21, 0xff, 0xee, 0x01, 0x21, 0xfd, 0xec, 0x0f, 0x0f, 0xfe, 0x0e, 0x1f,
+	0x0d, 0x20, 0x10, 0x0f,
+	0x0f, 0x10, 0x22, 0x10, 0x00, 0x1f, 0x11, 0x00, 0x00, 0xef, 0xf0, 0x00,
+	0xee, 0xfe, 0x00, 0xe0,
+	0xe0, 0x00, 0x02, 0xf0, 0xf0, 0xf0, 0xff, 0x0f, 0xe1, 0x14, 0x00, 0xef,
+	0xee, 0xff, 0xf0, 0xef,
+	0x00, 0xf0, 0x10, 0x21, 0x01, 0x11, 0x01, 0x00, 0x03, 0x21, 0x20, 0x00,
+	0x10, 0x00, 0x32, 0x11,
+	0x01, 0x10, 0x30, 0x02, 0x01, 0x13, 0x10, 0xf0, 0x00, 0x00, 0x21, 0x00,
+	0xee, 0xee, 0xf0, 0x00,
+	0xed, 0x0e, 0xee, 0xfe, 0xff, 0xfe, 0xef, 0xff, 0x0f, 0x00, 0xf0, 0xf0,
+	0xfe, 0xee, 0xfd, 0xff,
+	0x00, 0x10, 0xff, 0x00, 0xff, 0xf0, 0x00, 0x20, 0x0d, 0xcd, 0x00, 0x11,
+	0x02, 0x22, 0x0f, 0xff,
+	0x0f, 0x10, 0x1f, 0x1f, 0x10, 0x0e, 0x00, 0x1e, 0x0f, 0xff, 0x01, 0x33,
+	0x01, 0xf0, 0x00, 0x01,
+	0x02, 0x00, 0x00, 0x03, 0x00, 0xf0, 0xf0, 0x00, 0xe0, 0xed, 0x01, 0x01,
+	0x01, 0x00, 0x0f, 0x02,
+	0x12, 0x01, 0x00, 0xff, 0x00, 0x00, 0x1e, 0x0f, 0x0f, 0x21, 0x22, 0x00,
+	0x00, 0x1e, 0x21, 0x01,
+	0x11, 0x32, 0x13, 0x10, 0x00, 0xe0, 0x12, 0x00, 0xff, 0x00, 0x00, 0xf3,
+	0x00, 0xee, 0x00, 0xe0,
+	0xef, 0xf0, 0xff, 0x01, 0x01, 0x03, 0x00, 0x00, 0x00, 0xe0, 0xff, 0x00,
+	0x01, 0x11, 0x00, 0x00,
+	0x02, 0x20, 0x2e, 0x20, 0x20, 0x20, 0x25, 0x42, 0x12, 0x31, 0x10, 0x1e,
+	0x00, 0x0f, 0x10, 0x20,
+	0x11, 0x00, 0x21, 0x00, 0x01, 0x03, 0x01, 0xf0, 0x12, 0x22, 0x01, 0xff,
+	0xff, 0x00, 0x0e, 0xff,
+	0x22, 0x0e, 0xec, 0x00, 0xfe, 0xfe, 0xff, 0x00, 0x10, 0x10, 0x0e, 0xf0,
+	0xf0, 0x00, 0x00, 0x0f,
+	0x00, 0x0f, 0xec, 0x00, 0x33, 0xf0, 0xff, 0x01, 0x10, 0xef, 0xfd, 0x02,
+	0xf2, 0x01, 0x01, 0x03,
+	0x00, 0xe0, 0xf0, 0xf0, 0xee, 0xff, 0xf0, 0xf0, 0xff, 0xdd, 0xf0, 0x02,
+	0xf1, 0x00, 0x30, 0x13,
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+	0x20, 0x0e, 0x0e, 0x10,
+	0x14, 0x00, 0x00, 0x22, 0x12, 0x00, 0x0b, 0xf0, 0xfd, 0xfe, 0x1f, 0x32,
+	0x00, 0x10, 0x11, 0x0f,
+	0xef, 0xe0, 0x11, 0x13, 0x01, 0x01, 0xf0, 0x01, 0xf1, 0x00, 0xf1, 0xe0,
+	0xfd, 0x0f, 0x0f, 0x10,
+	0x00, 0x03, 0x10, 0x0e, 0x1f, 0x22, 0x03, 0x00, 0xf0, 0x00, 0xf0, 0x12,
+	0x01, 0xf0, 0x0e, 0xef,
+	0x0e, 0x00, 0x20, 0x10, 0x1f, 0x30, 0x0f, 0xf0, 0x10, 0x12, 0x10, 0x12,
+	0x01, 0xee, 0xfd, 0xef,
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+	0x00, 0xf0, 0x02, 0x11,
+	0x00, 0xee, 0x0f, 0x10, 0x0f, 0x20, 0x0f, 0xf0, 0x11, 0x21, 0x1f, 0x00,
+	0xf0, 0x01, 0x01, 0x02,
+	0x22, 0x01, 0xdf, 0x00, 0x10, 0xff, 0x00, 0xe0, 0x00, 0x02, 0x0f, 0x0e,
+	0xdf, 0x00, 0x10, 0x00,
+	0x10, 0x01, 0x01, 0x0f, 0xee, 0xf0, 0x0e, 0x0f, 0xfe, 0xef, 0x0e, 0x00,
+	0x02, 0x01, 0x01, 0x10,
+	0x00, 0x01, 0x34, 0x11, 0x0f, 0xff, 0x00, 0x02, 0xf0, 0x00, 0x02, 0x00,
+	0x03, 0x21, 0x00, 0xfe,
+	0x02, 0x02, 0x01, 0x00, 0x00, 0xf0, 0x0e, 0x0f, 0x0f, 0xec, 0x0f, 0x2f,
+	0x1f, 0x1e, 0x0d, 0x1e,
+	0x20, 0x01, 0x21, 0x00, 0x0d, 0x00, 0x0f, 0x0f, 0x20, 0x01, 0x11, 0x0f,
+	0xee, 0xef, 0x0e, 0x0f,
+	0xf0, 0x02, 0x01, 0xf0, 0xf0, 0x01, 0x01, 0x03, 0x01, 0xff, 0xee, 0x01,
+	0x02, 0x00, 0xc0, 0x0f,
+	0x0f, 0xf0, 0xf0, 0x00, 0x00, 0x00, 0x22, 0x11, 0x11, 0x11, 0x00, 0x33,
+	0x01, 0x20, 0x23, 0x20,
+	0x12, 0x01, 0x10, 0x1e, 0x00, 0xf0, 0x03, 0x22, 0x13, 0x11, 0x00, 0x00,
+	0xf0, 0xff, 0x00, 0x10,
+	0x0f, 0x0f, 0x10, 0x00, 0x0f, 0xef, 0x0f, 0x00, 0x0e, 0xee, 0x0f, 0xf0,
+	0xed, 0xf0, 0x00, 0xff,
+	0x0f, 0xed, 0xfe, 0xff, 0x00, 0x0f, 0xff, 0xfe, 0x00, 0x00, 0xf0, 0xfe,
+	0xef, 0x00, 0x00, 0xf2,
+	0xf0, 0x0f, 0x0d, 0x00, 0x01, 0x21, 0x00, 0x0e, 0x0f, 0x0e, 0xdb, 0x00,
+	0x00, 0x12, 0x24, 0x00,
+	0xf0, 0xff, 0x01, 0x02, 0x02, 0x00, 0x10, 0xef, 0xfe, 0xf1, 0xf0, 0x00,
+	0x00, 0xe0, 0xff, 0xfe,
+	0xf0, 0x01, 0x13, 0x12, 0x01, 0xfe, 0xff, 0x00, 0x00, 0x00, 0x20, 0x23,
+	0x1f, 0x0e, 0x10, 0x2e,
+	0xff, 0x00, 0x12, 0x22, 0x33, 0x10, 0x00, 0x02, 0x01, 0xf0, 0xf0, 0x11,
+	0x11, 0x00, 0xf0, 0xee,
+	0xff, 0xee, 0xe0, 0x00, 0xed, 0x00, 0x01, 0x01, 0xf1, 0xf1, 0x00, 0xde,
+	0x0f, 0x0f, 0xe0, 0x00,
+	0xf0, 0x0f, 0x00, 0x01, 0x31, 0x10, 0x1e, 0x21, 0x21, 0x21, 0x10, 0x21,
+	0x21, 0x11, 0x11, 0x10,
+	0x10, 0x12, 0x10, 0x01, 0x00, 0x00, 0xf1, 0x13, 0x22, 0x00, 0x21, 0x02,
+	0x11, 0x01, 0x00, 0x0e,
+	0x0d, 0xef, 0x1e, 0x10, 0x00, 0xf0, 0xff, 0xee, 0x0e, 0xff, 0xe0, 0x0e,
+	0x00, 0x00, 0x02, 0xff,
+	0x0e, 0x0f, 0x0f, 0x00, 0x0f, 0x20, 0x01, 0x00, 0x00, 0xf0, 0x10, 0x00,
+	0x01, 0xf0, 0xe0, 0x02,
+	0x02, 0x00, 0xf0, 0xef, 0x00, 0xf0, 0xf0, 0x0f, 0xe0, 0x0f, 0xff, 0xdf,
+	0xed, 0x00, 0x12, 0x00,
+	0x11, 0x21, 0x02, 0x12, 0x34, 0x00, 0x00, 0x1d, 0x00, 0x10, 0x12, 0x00,
+	0x13, 0x10, 0x00, 0xf0,
+	0x0f, 0x00, 0x02, 0x01, 0x31, 0x10, 0x00, 0x00, 0x20, 0x0e, 0x10, 0x20,
+	0x1f, 0x0e, 0x0f, 0x00,
+	0x11, 0x13, 0x00, 0x00, 0xff, 0xef, 0x12, 0x14, 0x00, 0xe0, 0x00, 0xf0,
+	0x00, 0x1f, 0x00, 0x00,
+	0x00, 0x10, 0x12, 0x02, 0x00, 0xe0, 0x0f, 0x02, 0x24, 0x01, 0x00, 0xff,
+	0x00, 0x00, 0x10, 0x1f,
+	0x00, 0x01, 0x11, 0x30, 0x0f, 0xee, 0x00, 0x31, 0x00, 0x10, 0x0f, 0x1e,
+	0x0f, 0x01, 0x33, 0x00,
+	0xee, 0xfb, 0x00, 0xf0, 0x00, 0xef, 0x00, 0xf0, 0xf0, 0xf1, 0xf0, 0xee,
+	0xe0, 0x00, 0x14, 0x0f,
+	0xf0, 0x1e, 0x10, 0x0f, 0x0e, 0x2f, 0x10, 0x21, 0x10, 0x10, 0x01, 0xf0,
+	0xf0, 0x00, 0x12, 0x12,
+	0x01, 0xf0, 0xef, 0x00, 0xf0, 0xf0, 0x1f, 0x30, 0xff, 0x0f, 0xf0, 0xff,
+	0x00, 0x00, 0xef, 0xf0,
+	0x01, 0x01, 0xff, 0xef, 0x02, 0x00, 0xee, 0x0e, 0xf0, 0x0e, 0xff, 0xfe,
+	0x0f, 0xe0, 0x23, 0x11,
+	0x0f, 0xff, 0xf0, 0x01, 0x34, 0x12, 0x22, 0x0f, 0xe0, 0x00, 0x00, 0x01,
+	0x21, 0x01, 0x00, 0x12,
+	0x02, 0xf0, 0xf0, 0x01, 0xf1, 0x01, 0x00, 0xef, 0x00, 0x0e, 0x00, 0x20,
+	0x2f, 0x10, 0x0d, 0x0f,
+	0x0d, 0x00, 0x30, 0x0f, 0x20, 0x10, 0x00, 0x00, 0x0e, 0xf0, 0x00, 0x01,
+	0x22, 0x0e, 0xdd, 0x00,
+	0x0f, 0x0f, 0x00, 0x00, 0x00, 0x02, 0x0f, 0xe0, 0xff, 0xf0, 0xef, 0x00,
+	0xe0, 0xee, 0x00, 0xe1,
+	0xe1, 0xe0, 0xff, 0xfe, 0xef, 0x00, 0x00, 0x00, 0xf0, 0x12, 0x10, 0x22,
+	0x0f, 0x00, 0x35, 0x21,
+	0xf0, 0x00, 0x32, 0x12, 0x00, 0x01, 0x00, 0x00, 0x0f, 0x01, 0x03, 0x10,
+	0x00, 0x33, 0x00, 0xef,
+	0x0f, 0x20, 0x22, 0x0e, 0xcc, 0x10, 0x00, 0xfd, 0x0f, 0x00, 0x0f, 0xed,
+	0xff, 0xe0, 0xed, 0xfe,
+	0xf0, 0x00, 0x00, 0x0d, 0xef, 0xf0, 0x0f, 0x00, 0xf0, 0x0f, 0xfe, 0xff,
+	0xf0, 0x01, 0x00, 0x0f,
+	0x00, 0xef, 0xfe, 0x0e, 0x10, 0x10, 0x20, 0x10, 0x10, 0x00, 0x0e, 0x10,
+	0x0f, 0xff, 0xef, 0x21,
+	0x03, 0x02, 0x21, 0x01, 0xf0, 0xf0, 0x11, 0x00, 0x00, 0x10, 0x03, 0x0f,
+	0x0f, 0xf0, 0xff, 0x00,
+	0xf0, 0xf0, 0x01, 0xf1, 0xf2, 0xf1, 0xf1, 0xff, 0xfe, 0x00, 0x12, 0x00,
+	0x20, 0x10, 0x1e, 0x0e,
+	0x20, 0x0f, 0x30, 0x23, 0x00, 0x0e, 0x11, 0x22, 0x01, 0x03, 0x22, 0x01,
+	0xf2, 0x0e, 0xff, 0x0f,
+	0x00, 0xf0, 0xf0, 0xff, 0x00, 0x01, 0xf0, 0xf1, 0xf1, 0x00, 0xe1, 0xf0,
+	0x00, 0xe0, 0x00, 0x00,
+	0x00, 0x02, 0x02, 0x20, 0x0f, 0x00, 0x00, 0x20, 0x40, 0x12, 0x21, 0x00,
+	0x1f, 0x1f, 0x11, 0x34,
+	0x11, 0x21, 0x00, 0x00, 0x10, 0x12, 0x00, 0x11, 0x00, 0x01, 0x10, 0x00,
+	0x11, 0x22, 0x02, 0x00,
+	0x01, 0x0f, 0x00, 0x0f, 0xfe, 0x00, 0x01, 0x01, 0x00, 0x1f, 0x0c, 0xde,
+	0x00, 0x20, 0x02, 0x00,
+	0xff, 0x00, 0x0e, 0x0d, 0xfe, 0x10, 0x00, 0x00, 0x31, 0x12, 0x00, 0xec,
+	0xff, 0x00, 0x13, 0x00,
+	0x0f, 0xf0, 0xc0, 0x00, 0xf1, 0xe0, 0xf1, 0x00, 0xe0, 0xeb, 0xf0, 0x01,
+	0x00, 0xdf, 0xfe, 0x00,
+	0x10, 0xf0, 0xee, 0x00, 0x12, 0x23, 0x00, 0x02, 0x00, 0xff, 0x00, 0x31,
+	0x02, 0x11, 0x00, 0x10,
+	0x0f, 0xfe, 0xff, 0x10, 0x20, 0x12, 0x12, 0x0e, 0xff, 0x10, 0x00, 0x11,
+	0x21, 0x21, 0xfd, 0xef,
+	0x10, 0x00, 0x01, 0x00, 0x10, 0x00, 0xed, 0x0f, 0x00, 0xf0, 0x01, 0x03,
+	0x01, 0x0f, 0x0f, 0x0f,
+	0xf1, 0x01, 0x02, 0x0f, 0x10, 0x00, 0x13, 0xff, 0xef, 0xfe, 0x00, 0x03,
+	0x12, 0x1f, 0x0d, 0xff,
+	0x00, 0x00, 0x00, 0x23, 0x10, 0x23, 0x00, 0xee, 0x0e, 0x0f, 0x10, 0x20,
+	0x00, 0x0e, 0xff, 0x00,
+	0x11, 0x12, 0x00, 0x00, 0x0e, 0xe0, 0x01, 0x03, 0xef, 0xee, 0x01, 0x01,
+	0x01, 0x00, 0xff, 0x00,
+	0x00, 0x10, 0x1f, 0x00, 0x0f, 0x21, 0x00, 0x00, 0x12, 0x10, 0x10, 0x1f,
+	0x00, 0xe1, 0x01, 0x10,
+	0x01, 0xf0, 0x00, 0x00, 0x00, 0x00, 0xe0, 0xff, 0x20, 0x35, 0x10, 0x00,
+	0xf0, 0xf1, 0x02, 0xf0,
+	0xf0, 0x0d, 0xe0, 0xfe, 0xff, 0xf0, 0xff, 0x00, 0x10, 0x0f, 0xdc, 0x00,
+	0x0f, 0xff, 0x00, 0x11,
+	0x00, 0xfe, 0x02, 0x22, 0x0e, 0xee, 0x01, 0x22, 0x00, 0x01, 0x33, 0x00,
+	0xff, 0x01, 0x12, 0x10,
+	0xf1, 0xe0, 0xf0, 0x01, 0xf1, 0x00, 0xe0, 0xf0, 0x00, 0xf0, 0x01, 0x11,
+	0x10, 0x0d, 0x2f, 0x1d,
+	0x0e, 0x20, 0x10, 0x22, 0x0f, 0x1e, 0x0d, 0x0e, 0xff, 0x00, 0x2f, 0x0f,
+	0x20, 0xf0, 0x00, 0x10,
+	0x10, 0x01, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x01, 0x02, 0xe0, 0x00, 0xdf,
+	0xf0, 0xff, 0xef, 0xfe,
+	0x01, 0xf1, 0xe0, 0x01, 0xf0, 0x00, 0x01, 0x01, 0x02, 0x00, 0x0e, 0xef,
+	0x00, 0x41, 0x01, 0x11,
+	0x22, 0x24, 0x10, 0x00, 0x12, 0x00, 0x10, 0x23, 0x01, 0x00, 0x00, 0x11,
+	0x01, 0x01, 0x01, 0x01,
+	0x0f, 0x10, 0x31, 0x00, 0x01, 0x21, 0x00, 0x00, 0xfe, 0xfe, 0x00, 0x1e,
+	0x00, 0x00, 0x00, 0xe0,
+	0x0f, 0xec, 0xff, 0xee, 0x00, 0x01, 0x11, 0x00, 0x0e, 0xef, 0x00, 0xfd,
+	0xed, 0xf0, 0xf0, 0x01,
+	0x00, 0x0d, 0xe0, 0xfc, 0xee, 0xf0, 0x00, 0x10, 0x11, 0x11, 0x0e, 0xff,
+	0xf0, 0x00, 0x00, 0x00,
+	0xef, 0xec, 0xf0, 0x12, 0x01, 0x10, 0x01, 0x00, 0xf0, 0x0e, 0x01, 0x12,
+	0x01, 0x11, 0x01, 0xf0,
+	0xf0, 0x0f, 0xef, 0xf0, 0x02, 0x01, 0xff, 0xee, 0xef, 0xf0, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0xab, 0x0e, 0xdf, 0x09, 0xca, 0x0f, 0x8c, 0x0a, 0x00, 0x10, 0x20, 0x10,
+	0x00, 0x20, 0x00, 0x0f,
+	0x11, 0x43, 0x11, 0x00, 0xde, 0x00, 0x31, 0x00, 0x00, 0x10, 0x0f, 0x10,
+	0x12, 0xff, 0xf0, 0x02,
+	0x00, 0x00, 0xef, 0xf0, 0xf0, 0xe0, 0xe0, 0xe0, 0x00, 0xff, 0xfd, 0xee,
+	0x0f, 0x00, 0x00, 0x00,
+	0x12, 0x30, 0x00, 0x00, 0x03, 0x10, 0x10, 0x0e, 0x00, 0x00, 0x11, 0xff,
+	0x0d, 0x12, 0x23, 0x22,
+	0x21, 0x01, 0x12, 0x22, 0x01, 0x10, 0x10, 0x01, 0x23, 0x00, 0x13, 0x10,
+	0x00, 0x00, 0x0f, 0xf0,
+	0x0e, 0x00, 0x10, 0x00, 0x10, 0x00, 0x00, 0x02, 0x10, 0xfc, 0xef, 0x0f,
+	0x0f, 0x0f, 0x10, 0x00,
+	0xff, 0xed, 0x0e, 0x00, 0x20, 0x00, 0x10, 0x00, 0x11, 0x01, 0x01, 0xe0,
+	0x10, 0x22, 0xe0, 0xd0,
+	0xfe, 0xdf, 0xee, 0xf0, 0x00, 0x00, 0x00, 0xfe, 0x00, 0xf0, 0x00, 0x00,
+	0xff, 0xfe, 0x00, 0xf1,
+	0x01, 0x02, 0x02, 0x22, 0xf0, 0xee, 0x02, 0x12, 0x10, 0x11, 0x21, 0x23,
+	0x00, 0x10, 0x00, 0x20,
+	0x01, 0x10, 0x00, 0xff, 0x00, 0x21, 0x12, 0x00, 0xff, 0x0f, 0x00, 0x21,
+	0x11, 0x21, 0x0f, 0x00,
+	0x00, 0x1f, 0x1f, 0x00, 0x10, 0x00, 0xf0, 0x00, 0x00, 0xf0, 0x00, 0x10,
+	0x00, 0x03, 0x10, 0x00,
+	0x04, 0x00, 0x00, 0xee, 0x00, 0x02, 0x12, 0xf0, 0xff, 0x00, 0x00, 0xf0,
+	0x03, 0x10, 0x00, 0x0e,
+	0x22, 0x20, 0x00, 0x11, 0x23, 0x21, 0x0e, 0xfe, 0x0d, 0x00, 0x20, 0x1f,
+	0x00, 0x00, 0xee, 0x00,
+	0x12, 0x10, 0x00, 0xf0, 0xf1, 0x0f, 0xe0, 0x0e, 0xde, 0xff, 0x0f, 0x00,
+	0x00, 0x00, 0xf0, 0x0e,
+	0x0f, 0xff, 0x10, 0x21, 0x11, 0x00, 0x00, 0x12, 0x00, 0x10, 0x02, 0x00,
+	0xff, 0x0e, 0xf0, 0x0f,
+	0xf0, 0x00, 0x32, 0x12, 0x10, 0xf0, 0xf0, 0x10, 0x21, 0x02, 0x10, 0x02,
+	0xf0, 0x0f, 0xfe, 0xcf,
+	0x0e, 0xe0, 0xee, 0x00, 0xff, 0x00, 0xf0, 0xff, 0x00, 0xef, 0x00, 0x10,
+	0x00, 0xf0, 0x0e, 0xfe,
+	0x0f, 0xf0, 0x10, 0x11, 0x00, 0x00, 0x00, 0x14, 0x10, 0x02, 0x13, 0x00,
+	0x00, 0xf0, 0x00, 0x02,
+	0x03, 0x13, 0x00, 0xef, 0xee, 0x0f, 0xf0, 0x00, 0x11, 0x22, 0x01, 0x0e,
+	0x00, 0x00, 0x2e, 0x10,
+	0x10, 0x02, 0x1f, 0x0e, 0x0d, 0x00, 0x0f, 0x10, 0x10, 0x0e, 0x0e, 0x00,
+	0x11, 0x00, 0x10, 0x31,
+	0x01, 0x01, 0x00, 0x0d, 0xe0, 0x00, 0xf0, 0x00, 0xff, 0xdd, 0xfd, 0xf0,
+	0xf0, 0xe0, 0x00, 0xe0,
+	0xf0, 0xee, 0x00, 0x01, 0xf0, 0x02, 0xf0, 0x02, 0x00, 0xdd, 0x00, 0xec,
+	0x00, 0x02, 0x0f, 0x21,
+	0x25, 0x00, 0x01, 0x00, 0x02, 0x03, 0x10, 0x13, 0x22, 0x11, 0xf0, 0x00,
+	0x00, 0x10, 0x10, 0x00,
+	0x2e, 0x00, 0x0f, 0x11, 0x00, 0x00, 0x0c, 0x00, 0xff, 0x01, 0x21, 0xff,
+	0xef, 0x0f, 0x00, 0xfd,
+	0xee, 0xec, 0x00, 0xe0, 0x11, 0x00, 0x0f, 0xdd, 0x0f, 0x11, 0x01, 0xfe,
+	0x0d, 0x01, 0xff, 0xee,
+	0x10, 0x01, 0xfe, 0xef, 0x10, 0x0f, 0x1f, 0x01, 0x00, 0xff, 0x00, 0x10,
+	0x00, 0xf0, 0x10, 0x00,
+	0x0e, 0x00, 0xf0, 0x1f, 0x21, 0x11, 0x02, 0x00, 0xe0, 0x11, 0x23, 0x00,
+	0xdf, 0x00, 0x00, 0x02,
+	0x00, 0xff, 0xf1, 0x02, 0xf0, 0xdd, 0xe0, 0x0f, 0xff, 0xf0, 0x00, 0x00,
+	0x11, 0x10, 0x44, 0x00,
+	0x10, 0x10, 0x2f, 0x10, 0x0e, 0x20, 0x21, 0x11, 0xff, 0x0e, 0x10, 0x00,
+	0x11, 0x03, 0x01, 0x01,
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+	0x01, 0x00, 0xf0, 0x02,
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+	0xf0, 0xee, 0xe0, 0xef,
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+	0x11, 0x01, 0x0f, 0x0f,
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+	0x21, 0x10, 0xff, 0xed,
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+	0xf0, 0x01, 0x02, 0x01,
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+	0x0f, 0x0f, 0x00, 0x01,
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+	0xee, 0x10, 0x11, 0x15,
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+	0xee, 0xf0, 0xfe, 0x00,
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+	0x00, 0x0e, 0x30, 0x11,
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+	0x12, 0x1f, 0x0f, 0x0f,
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+	0xf0, 0xef, 0xde, 0x00,
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+	0x12, 0x10, 0x00, 0xf1,
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+	0x00, 0x0e, 0xef, 0xff,
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+	0x00, 0x31, 0xf0, 0xee,
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+	0x00, 0x20, 0x00, 0x21,
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+	0x0f, 0x00, 0x0f, 0xf0,
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+	0xdf, 0x00, 0x00, 0x0f,
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+	0x00, 0xe0, 0x00, 0xf0,
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+	0x12, 0x11, 0xf0, 0xe1,
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+	0xf0, 0x10, 0x1f, 0x00,
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+	0x02, 0xef, 0x0f, 0x0f,
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+	0x11, 0x01, 0x11, 0x0f,
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+	0x32, 0x03, 0x02, 0x00,
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+	0x11, 0x00, 0x00, 0x00,
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+	0x22, 0x11, 0xff, 0xfd,
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+	0xef, 0xdf, 0x0f, 0xdf,
+	0xed, 0xf1, 0xe0, 0xf0, 0xff, 0xff, 0xff, 0xff, 0x00, 0x02, 0x00, 0x00,
+	0x0f, 0x00, 0x00, 0x24,
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+	0x00, 0xfd, 0x00, 0x00,
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+	0x00, 0x00, 0x02, 0x12,
+	0x00, 0xfd, 0xdf, 0x0f, 0x00, 0xf0, 0xf0, 0x01, 0x13, 0x10, 0x0e, 0xde,
+	0x0f, 0xff, 0x01, 0x22,
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+	0x00, 0x10, 0x11, 0x11,
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+	0x12, 0x12, 0x00, 0xf0,
+	0x10, 0x00, 0x00, 0x00, 0x10, 0x12, 0xf0, 0xfe, 0x0f, 0xed, 0xff, 0xfe,
+	0x00, 0x11, 0x12, 0x20,
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+	0x13, 0x24, 0x00, 0xde,
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+	0x01, 0x0f, 0xdb, 0xf0,
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+	0x20, 0x00, 0x00, 0xf0,
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+	0x0e, 0x01, 0x00, 0x0f,
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+	0x21, 0x0f, 0xff, 0x10,
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+	0x01, 0x10, 0xef, 0xf0,
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+	0x00, 0x00, 0x00, 0x12,
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+	0x01, 0x33, 0x01, 0x22,
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+	0x00, 0x00, 0x00, 0x20,
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+	0x0d, 0x00, 0x01, 0x0f,
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+	0x00, 0x00, 0xdd, 0xee,
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+	0x0f, 0x10, 0x00, 0xf1,
+	0x03, 0x00, 0xf0, 0xdf, 0x01, 0xe0, 0xf0, 0xf0, 0x01, 0xe1, 0x01, 0xf0,
+	0x0f, 0x00, 0x12, 0x11,
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+	0x0e, 0x00, 0x11, 0x01,
+	0x02, 0x20, 0x00, 0x00, 0xf1, 0x00, 0xfe, 0x0f, 0x0f, 0x1f, 0x00, 0xf0,
+	0x10, 0x00, 0x00, 0xf1,
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+	0xfe, 0x0e, 0x00, 0x20,
+	0x12, 0x1f, 0x0f, 0x0f, 0x00, 0x13, 0x22, 0x01, 0x01, 0x00, 0x01, 0x01,
+	0x01, 0x01, 0x13, 0x21,
+	0x01, 0x11, 0x01, 0x00, 0x01, 0x02, 0x31, 0x13, 0x0f, 0xef, 0x0f, 0x00,
+	0xee, 0x11, 0x23, 0x0f,
+	0xfc, 0x0f, 0x10, 0x1f, 0x21, 0x00, 0x21, 0x00, 0x0d, 0x0f, 0x11, 0x11,
+	0x0f, 0xf0, 0x10, 0x00,
+	0x0f, 0x0f, 0x00, 0x00, 0x20, 0x00, 0x00, 0x0f, 0xed, 0x00, 0x02, 0x11,
+	0x00, 0xee, 0xee, 0xe0,
+	0xf0, 0xef, 0xdf, 0x0f, 0xdf, 0xf0, 0xff, 0x00, 0x00, 0x02, 0x01, 0x10,
+	0x0f, 0x10, 0x01, 0x0f,
+	0x00, 0x00, 0x03, 0x01, 0x01, 0x13, 0x0e, 0xe1, 0x00, 0x00, 0x03, 0x12,
+	0x31, 0x02, 0x21, 0x00,
+	0xfe, 0x00, 0x20, 0x31, 0x0f, 0x0f, 0x10, 0x00, 0x10, 0x22, 0x00, 0xed,
+	0x12, 0x21, 0xf0, 0xff,
+	0x02, 0x32, 0x01, 0xfd, 0xff, 0x00, 0xf1, 0xff, 0xf0, 0x00, 0x10, 0x11,
+	0x01, 0x00, 0x00, 0x10,
+	0x13, 0xff, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x00, 0x34, 0x11,
+	0xf0, 0x0e, 0x0f, 0x0f,
+	0x1f, 0x21, 0x22, 0x00, 0x00, 0x0e, 0x02, 0x00, 0x10, 0x10, 0xf0, 0xff,
+	0xfe, 0xff, 0x00, 0x0e,
+	0xf0, 0x02, 0x0f, 0x00, 0xff, 0x00, 0x12, 0xff, 0x0f, 0x0e, 0xee, 0xff,
+	0x0f, 0x00, 0xf0, 0x0e,
+	0xff, 0x10, 0x11, 0x00, 0x10, 0x22, 0x23, 0x00, 0x12, 0x00, 0x21, 0xf1,
+	0xff, 0xff, 0x00, 0x01,
+	0x33, 0x00, 0xe0, 0x00, 0x13, 0x00, 0x00, 0x10, 0x12, 0x00, 0xf0, 0x11,
+	0x00, 0xdd, 0x11, 0x00,
+	0xdb, 0x00, 0xfe, 0xe0, 0x00, 0x0e, 0xdc, 0xf0, 0x0f, 0x00, 0x01, 0x00,
+	0x0e, 0x00, 0x22, 0x01,
+	0x00, 0x00, 0x11, 0xf0, 0xef, 0x00, 0x01, 0xf0, 0xe3, 0x01, 0xff, 0xf0,
+	0x02, 0x10, 0x11, 0x03,
+	0x00, 0x00, 0xf0, 0x00, 0x11, 0x0f, 0x0f, 0x10, 0x00, 0x01, 0x01, 0x01,
+	0x10, 0x00, 0x10, 0x00,
+	0x3b, 0x17, 0x17, 0x1a, 0x23, 0x17, 0x83, 0x19, 0x00, 0x1f, 0x10, 0x10,
+	0x10, 0x11, 0x21, 0x00,
+	0x0e, 0x10, 0x10, 0x00, 0x1f, 0x00, 0x23, 0x0f, 0xee, 0x0f, 0x00, 0x10,
+	0x00, 0xf1, 0xfe, 0xfe,
+	0xfe, 0x00, 0xff, 0xee, 0x0f, 0x0f, 0xff, 0xff, 0xf0, 0xf0, 0x02, 0x00,
+	0xfe, 0xdf, 0xff, 0x0f,
+	0xff, 0xef, 0x13, 0x04, 0xf0, 0xee, 0xf0, 0xf0, 0xf0, 0x00, 0x12, 0x01,
+	0x01, 0x02, 0x00, 0x02,
+	0x10, 0x12, 0x23, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x1f, 0x0e, 0x00,
+	0x00, 0x1f, 0x00, 0x00,
+	0x10, 0x01, 0x0f, 0x0f, 0x20, 0x0e, 0x00, 0x01, 0x22, 0x00, 0xfc, 0x00,
+	0x0f, 0x0d, 0xef, 0xff,
+	0x10, 0x12, 0x00, 0xff, 0xef, 0xfd, 0x0e, 0x02, 0x01, 0x00, 0xf0, 0xee,
+	0x0f, 0x0e, 0xf0, 0xf0,
+	0xf0, 0xfe, 0x0f, 0x0f, 0x0e, 0x0f, 0x01, 0x21, 0x12, 0x00, 0x00, 0x20,
+	0x01, 0xe0, 0x00, 0x0f,
+	0xf0, 0xf1, 0xef, 0x00, 0xe1, 0xf2, 0xf0, 0xe0, 0x01, 0xff, 0xf0, 0x10,
+	0x21, 0x12, 0x11, 0x00,
+	0x1e, 0x10, 0x1f, 0xfd, 0x0f, 0xff, 0x20, 0x13, 0x00, 0x11, 0x00, 0x00,
+	0x02, 0x00, 0x00, 0x20,
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+	0x03, 0x03, 0x00, 0xe1,
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+	0x0e, 0x00, 0x10, 0x13,
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+	0x11, 0x10, 0x01, 0x01,
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+	0xff, 0xee, 0xff, 0xe0,
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+	0x01, 0x00, 0xde, 0x01,
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+	0x21, 0x21, 0x00, 0x00,
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+	0x01, 0xf0, 0xfe, 0x01,
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+	0x02, 0x32, 0xfe, 0xff,
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+	0xf0, 0x0e, 0xfe, 0x10,
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+	0xfe, 0x01, 0x01, 0x24,
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+	0x00, 0x00, 0x1f, 0xff,
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+	0xff, 0xed, 0x0f, 0x10,
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+	0xdd, 0x00, 0xfe, 0xed,
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+	0x00, 0xf2, 0xf0, 0xe0,
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+	0x01, 0x01, 0x20, 0x10,
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+	0x12, 0x11, 0x00, 0x0e,
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+	0x12, 0x01, 0x00, 0xf0,
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+	0x0f, 0x1f, 0xfd, 0xfe,
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+	0x11, 0x10, 0x1f, 0x10,
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+	0xff, 0x0e, 0xe0, 0x00,
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+	0x11, 0x01, 0x11, 0x00,
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+	0x00, 0x11, 0x02, 0x23,
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+	0x22, 0x23, 0x1f, 0x20,
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+	0x01, 0x01, 0x00, 0xfe,
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+	0x11, 0x01, 0x10, 0x02,
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+	0x00, 0x1e, 0x00, 0xf1,
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+	0x00, 0x00, 0x00, 0xee,
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+	0x10, 0xe0, 0x01, 0x02,
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+	0x10, 0x00, 0xff, 0x0e,
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+	0x01, 0x00, 0xf0, 0xef,
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+	0xff, 0x01, 0x01, 0x01,
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+	0x01, 0x00, 0x0f, 0x0f,
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+	0xe0, 0x0f, 0x11, 0x00,
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+	0x0f, 0x01, 0x02, 0xf0,
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+	0xf0, 0x01, 0x01, 0x12,
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+	0x00, 0x02, 0x01, 0x00,
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+	0x00, 0x10, 0x11, 0x22,
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+	0x00, 0x22, 0x01, 0x0f,
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+	0x00, 0x02, 0x13, 0x00,
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+	0x10, 0x20, 0x11, 0x20,
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+	0x00, 0xfd, 0xfe, 0x10,
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+	0xee, 0xf0, 0xef, 0xf0,
+	0xf0, 0xf1, 0x01, 0x01, 0x00, 0xdf, 0xf0, 0x01, 0x00, 0xef, 0x00, 0x00,
+	0x23, 0x01, 0x01, 0xf1,
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+	0x0f, 0xf0, 0xf2, 0x00,
+	0x01, 0xef, 0x10, 0x10, 0xf0, 0xfd, 0x01, 0x01, 0x10, 0x10, 0x00, 0xff,
+	0xf0, 0x00, 0x0f, 0x00,
+	0x00, 0x00, 0x00, 0xf0, 0x00, 0xef, 0x0f, 0x00, 0x0c, 0xff, 0x00, 0x01,
+	0x32, 0x10, 0x0e, 0x20,
+	0x35, 0x00, 0x0f, 0x0f, 0x01, 0x02, 0x00, 0x0f, 0x00, 0x1f, 0x00, 0xf1,
+	0x01, 0x12, 0x00, 0xf0,
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+	0xfe, 0x0e, 0x00, 0x10,
+	0x12, 0x0f, 0x00, 0x0f, 0x0f, 0xfd, 0xee, 0x20, 0x11, 0x12, 0x02, 0x00,
+	0x0f, 0xf0, 0x00, 0x03,
+	0x21, 0x12, 0x0f, 0xff, 0x01, 0x31, 0x01, 0x00, 0x02, 0x00, 0x00, 0x03,
+	0x00, 0x00, 0x00, 0x01,
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+	0xfe, 0x0f, 0x00, 0x22,
+	0x00, 0x00, 0xed, 0x00, 0x00, 0x00, 0x01, 0xe2, 0x00, 0xf0, 0xfd, 0xe0,
+	0xfe, 0x00, 0xf1, 0xe0,
+	0x0f, 0x00, 0x00, 0xef, 0x00, 0x22, 0x00, 0x11, 0x01, 0x00, 0x0d, 0x00,
+	0x00, 0xf0, 0x11, 0x11,
+	0x0f, 0xff, 0x10, 0x00, 0x31, 0x00, 0x21, 0x21, 0x1f, 0x10, 0x22, 0x01,
+	0x10, 0x01, 0x10, 0x0f,
+	0xf0, 0x0e, 0x00, 0x00, 0x00, 0x22, 0x00, 0xfe, 0xfe, 0xff, 0x0f, 0x00,
+	0x03, 0x21, 0x00, 0xfe,
+	0x00, 0xf0, 0xfe, 0xef, 0x00, 0x00, 0x00, 0xf0, 0x00, 0x00, 0xef, 0x11,
+	0x11, 0x11, 0x00, 0xf0,
+	0x00, 0xf1, 0xf1, 0x02, 0x10, 0x0f, 0xe0, 0x01, 0x00, 0x01, 0xe1, 0x01,
+	0x21, 0x00, 0x00, 0x0f,
+	0x00, 0x14, 0x00, 0x33, 0x10, 0x00, 0xf0, 0x0f, 0x1f, 0x20, 0x12, 0x11,
+	0x00, 0xec, 0x00, 0x00,
+	0x12, 0x32, 0x00, 0x0f, 0xff, 0x00, 0x0f, 0x10, 0x01, 0x00, 0xf0, 0x00,
+	0x0d, 0xf0, 0x0e, 0x0f,
+	0x1f, 0x0d, 0xef, 0x1f, 0x1e, 0x00, 0x0f, 0xf0, 0x00, 0xef, 0xff, 0x0f,
+	0x01, 0xef, 0xee, 0xf0,
+	0x00, 0xff, 0xf0, 0xff, 0xf0, 0xfe, 0xe0, 0x00, 0x02, 0x00, 0x00, 0xf0,
+	0x01, 0xe1, 0x00, 0xff,
+	0xf1, 0xf0, 0x00, 0xf0, 0x00, 0x00, 0x01, 0x01, 0x1f, 0x1f, 0x00, 0x2f,
+	0x00, 0xf0, 0x12, 0x00,
+	0x0d, 0x00, 0xff, 0x0f, 0x01, 0x43, 0x00, 0xff, 0x0f, 0x0f, 0x1f, 0x0f,
+	0xff, 0x00, 0x11, 0x00,
+	0xf1, 0x02, 0x00, 0x00, 0xee, 0xff, 0x00, 0xf0, 0x00, 0x00, 0x0f, 0xff,
+	0x00, 0x11, 0x0f, 0x0f,
+	0x0f, 0x10, 0x00, 0x03, 0x21, 0xf0, 0xff, 0x00, 0x0f, 0x00, 0x10, 0x00,
+	0x03, 0x00, 0x10, 0x32,
+	0x02, 0xf0, 0x01, 0x33, 0x02, 0xe0, 0xf0, 0x02, 0x22, 0xf0, 0xef, 0x02,
+	0x01, 0x10, 0xff, 0x0f,
+	0x01, 0x00, 0xf1, 0x00, 0x2f, 0x11, 0x00, 0xf0, 0x0f, 0x0f, 0x00, 0x0f,
+	0x20, 0x0e, 0x00, 0x30,
+	0x22, 0x33, 0x0f, 0x0f, 0x10, 0x20, 0x0f, 0x11, 0x22, 0x00, 0xf0, 0x0e,
+	0xfe, 0x00, 0x00, 0x00,
+	0x0f, 0x00, 0x00, 0x0e, 0x00, 0xee, 0x0f, 0x01, 0x01, 0x0f, 0x0f, 0xef,
+	0xef, 0x00, 0xfe, 0xe0,
+	0xff, 0x00, 0xf0, 0xd0, 0x00, 0x02, 0x01, 0xf0, 0xf0, 0x00, 0x00, 0xf0,
+	0xf0, 0x00, 0x12, 0x02,
+	0x11, 0xf0, 0x0f, 0x00, 0x02, 0x32, 0x10, 0x11, 0x21, 0x01, 0x20, 0x01,
+	0x01, 0x20, 0x01, 0x0f,
+	0xff, 0x00, 0x01, 0x10, 0x12, 0x01, 0x10, 0x00, 0x10, 0x00, 0x0f, 0x00,
+	0x01, 0x12, 0x00, 0x00,
+	0xff, 0x00, 0x02, 0x0f, 0xcd, 0x00, 0x10, 0x00, 0xee, 0x0f, 0x00, 0x0e,
+	0x10, 0x24, 0x10, 0x21,
+	0x0f, 0x0e, 0x01, 0x31, 0x02, 0x21, 0x10, 0x01, 0xfd, 0xfe, 0x00, 0x00,
+	0x12, 0x10, 0x00, 0x21,
+	0x01, 0xe0, 0x00, 0x23, 0x00, 0x00, 0xd0, 0x0e, 0x00, 0x00, 0xee, 0xf0,
+	0x0f, 0x0e, 0xed, 0x00,
+	0x00, 0x0f, 0x10, 0xf0, 0x00, 0x30, 0x01, 0xef, 0x0f, 0x10, 0x00, 0xf2,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x02, 0x23, 0x20, 0x01, 0x00, 0x13, 0x00, 0x20, 0x01, 0x00,
+	0x11, 0x01, 0x01, 0xff,
+	0x00, 0x00, 0x10, 0x00, 0x0f, 0x00, 0xfd, 0xf0, 0x00, 0x00, 0xfd, 0x00,
+	0x0e, 0x0f, 0x00, 0xff,
+	0xfe, 0x00, 0x02, 0x00, 0xed, 0x01, 0x01, 0x00, 0x00, 0x03, 0x00, 0xed,
+	0xff, 0xf0, 0xe0, 0x00,
+	0x02, 0x01, 0xf0, 0xff, 0x01, 0xf0, 0xf0, 0x00, 0x01, 0x33, 0x10, 0x00,
+	0xf0, 0x0e, 0x00, 0x02,
+	0x0f, 0x0f, 0x00, 0x12, 0x23, 0x0f, 0xff, 0x01, 0x21, 0x10, 0x00, 0x10,
+	0x10, 0x22, 0x0f, 0xf0,
+	0x00, 0x30, 0x00, 0x0f, 0x00, 0x0f, 0x00, 0x00, 0x00, 0x0d, 0xff, 0xed,
+	0x00, 0x10, 0x00, 0x0f,
+	0x00, 0x0f, 0xee, 0xf0, 0x0e, 0xe0, 0x0f, 0xf0, 0x00, 0xde, 0xf0, 0xf0,
+	0xf0, 0xfe, 0x10, 0x01,
+	0x00, 0xed, 0x00, 0xe0, 0x03, 0x12, 0x00, 0xf2, 0xf0, 0xff, 0xe0, 0x01,
+	0xf2, 0x01, 0x02, 0x0f,
+	0x0f, 0x00, 0x11, 0x33, 0xfe, 0x00, 0x40, 0x02, 0x0e, 0x10, 0x30, 0xe0,
+	0x0f, 0x00, 0x11, 0x0f,
+	0xde, 0x10, 0x01, 0x11, 0x11, 0x11, 0x00, 0xff, 0x11, 0x11, 0x0f, 0x0c,
+	0x0f, 0xff, 0x00, 0x10,
+	0x0f, 0x0f, 0x0e, 0x01, 0x01, 0x0e, 0xfe, 0x00, 0x02, 0x00, 0x00, 0x00,
+	0x0f, 0x0e, 0xe0, 0x0f,
+	0xff, 0x00, 0xfe, 0xef, 0xff, 0x01, 0x01, 0xff, 0x00, 0xf2, 0xf0, 0x00,
+	0xf0, 0x00, 0xf0, 0x00,
+	0xff, 0xf0, 0x0f, 0xf0, 0xff, 0x01, 0x01, 0xef, 0x00, 0x33, 0x00, 0xfd,
+	0x00, 0x11, 0x32, 0x11,
+	0x00, 0x00, 0x2f, 0xfe, 0xfe, 0x10, 0x22, 0x10, 0x00, 0x01, 0x0f, 0x00,
+	0x20, 0x02, 0x00, 0x00,
+	0xf0, 0x01, 0xf0, 0x02, 0x10, 0x02, 0xff, 0xfe, 0xed, 0x00, 0x00, 0xf0,
+	0x00, 0x00, 0x02, 0x10,
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+	0x10, 0x11, 0x11, 0x00,
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+	0xf0, 0xe0, 0x02, 0x10,
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+	0x0f, 0xdd, 0x0f, 0x00,
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+	0x10, 0x11, 0x00, 0x1f,
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+	0x01, 0x23, 0x00, 0xf0,
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+	0xfe, 0xff, 0x00, 0x00,
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+	0x01, 0x10, 0x00, 0xe0,
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+	0x00, 0x02, 0x0f, 0x0d,
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+	0x1f, 0xff, 0x00, 0x1f,
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+	0xed, 0x10, 0x01, 0x01,
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+	0x01, 0xf0, 0xf0, 0xff,
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+	0x10, 0x11, 0x31, 0xff,
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+	0x02, 0x10, 0x11, 0x00,
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+	0xee, 0x0e, 0xf0, 0xef,
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+	0x0f, 0x1f, 0x0f, 0x1f,
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+	0x00, 0x01, 0x11, 0x11,
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+	0xef, 0x00, 0x00, 0xef,
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+	0xed, 0x00, 0xf1, 0x02,
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+	0x0f, 0x00, 0x20, 0x0f,
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+	0x00, 0x00, 0x01, 0x0f,
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+	0x12, 0x00, 0x10, 0x0f,
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+	0x01, 0x00, 0x10, 0x00,
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+	0x00, 0x0f, 0xe0, 0x0f,
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+	0x0e, 0xde, 0x0f, 0xf1,
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+	0x11, 0x00, 0x01, 0xf1,
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+	0x0e, 0x00, 0x00, 0x21,
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+	0x00, 0xfd, 0x0f, 0xff,
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+	0x00, 0x00, 0xe0, 0x00,
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+	0x22, 0x12, 0x0f, 0x0f,
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+	0x00, 0x01, 0xe2, 0x00,
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+	0x10, 0x10, 0x1e, 0x20,
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+	0xff, 0x00, 0x00, 0x0f,
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+	0x00, 0x00, 0xd1, 0x00,
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+	0x24, 0x21, 0x00, 0x10,
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+	0x02, 0xff, 0xdd, 0x00,
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+	0x00, 0x00, 0x10, 0x10,
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+	0x33, 0x01, 0x10, 0x00,
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+	0xe0, 0xff, 0x00, 0x10,
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+	0x11, 0x00, 0xff, 0xf0,
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+	0x00, 0xf1, 0x00, 0x12,
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+	0x0e, 0x01, 0x01, 0x0f,
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+	0xfe, 0x00, 0x00, 0xe0,
+	0x1f, 0x00, 0xef, 0x00, 0xf0, 0xef, 0x0f, 0xf0, 0x0f, 0xef, 0xf0, 0xfe,
+	0xf0, 0x00, 0xf1, 0xff,
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+	0x00, 0x01, 0x22, 0x0f,
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+	0x10, 0x0d, 0xf0, 0x00,
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+	0x0f, 0x00, 0xf0, 0x10,
+	0x13, 0x00, 0xed, 0xe0, 0xff, 0x01, 0x00, 0xee, 0x01, 0x01, 0xf0, 0xff,
+	0x00, 0xf0, 0x01, 0x33,
+	0xf0, 0xf0, 0x0f, 0xf0, 0x02, 0x12, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff,
+	0x10, 0x12, 0x12, 0xf1,
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+	0x00, 0x00, 0x00, 0x10,
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+	0xef, 0x0e, 0x00, 0x01,
+	0x0e, 0x0e, 0xee, 0x10, 0x02, 0x00, 0x00, 0xee, 0xd0, 0xfd, 0xe0, 0xf0,
+	0x02, 0xe0, 0xf2, 0xee,
+	0xee, 0xfe, 0xff, 0xf1, 0x00, 0xf1, 0x00, 0xee, 0x00, 0x00, 0xe0, 0x0f,
+	0x00, 0x00, 0x10, 0x10,
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+	0x00, 0x12, 0x00, 0x0d,
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+	0xf0, 0x0f, 0xe0, 0x00,
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+	0x10, 0x00, 0x20, 0x03,
+	0x12, 0x01, 0x12, 0x10, 0x00, 0xef, 0x01, 0x11, 0x12, 0x01, 0x00, 0xf1,
+	0x00, 0x00, 0x03, 0x02,
+	0x02, 0x0f, 0x00, 0xdf, 0x0f, 0x00, 0xf0, 0x00, 0x02, 0x00, 0x0d, 0x00,
+	0x01, 0x10, 0x00, 0x00,
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+	0x22, 0x00, 0x0e, 0x10,
+	0x32, 0x10, 0x10, 0x10, 0x0f, 0x10, 0x20, 0x00, 0xf0, 0xfe, 0x0f, 0x0f,
+	0x0f, 0xee, 0xef, 0x10,
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+	0xe0, 0x0f, 0xf0, 0xff,
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+	0x00, 0xe0, 0x00, 0x10,
+	0x10, 0x00, 0x10, 0x22, 0x22, 0x00, 0x10, 0x11, 0x02, 0x11, 0x12, 0x12,
+	0x0f, 0xf0, 0xf1, 0x00,
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+	0x00, 0x0f, 0x10, 0x00,
+	0x0e, 0xe0, 0x1f, 0x0f, 0x0e, 0x0f, 0x10, 0x10, 0x11, 0x11, 0x00, 0xfe,
+	0x0f, 0x0f, 0x21, 0x23,
+	0x12, 0x03, 0x10, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x01, 0x10, 0x0e,
+	0xfe, 0x12, 0x01, 0x11,
+	0x00, 0xfe, 0xff, 0x00, 0xf0, 0x0e, 0xf0, 0xff, 0x00, 0x0f, 0x0f, 0xef,
+	0x0f, 0x00, 0x01, 0x22,
+	0x00, 0x00, 0x00, 0x00, 0xf0, 0x00, 0xf1, 0x00, 0x00, 0xf1, 0x00, 0xf0,
+	0x0f, 0x21, 0x01, 0x02,
+	0x00, 0x10, 0x00, 0x21, 0x10, 0x02, 0x20, 0x10, 0x0e, 0xee, 0x00, 0x32,
+	0x00, 0x0f, 0xff, 0x00,
+	0x10, 0x10, 0x00, 0x00, 0x0f, 0x00, 0x0f, 0xf0, 0x10, 0x00, 0xf0, 0x03,
+	0x0f, 0xee, 0x00, 0xef,
+	0x00, 0xf0, 0x00, 0xff, 0xf0, 0x01, 0x01, 0x0e, 0xff, 0xdd, 0x00, 0x01,
+	0x01, 0xf0, 0x01, 0xf1,
+	0xf0, 0xed, 0xf0, 0x00, 0x01, 0x01, 0x11, 0x11, 0x10, 0x00, 0x11, 0x11,
+	0x10, 0x00, 0x12, 0x0f,
+	0x0f, 0xf0, 0x01, 0x22, 0x02, 0x10, 0x00, 0x01, 0xf0, 0x00, 0x10, 0x21,
+	0x12, 0x0f, 0xff, 0x0f,
+	0x20, 0x0f, 0x10, 0x0f, 0x0f, 0xff, 0x0d, 0x0f, 0xff, 0x0f, 0x10, 0x00,
+	0xff, 0x0f, 0xff, 0xfe,
+	0x00, 0xff, 0xf0, 0x0f, 0x00, 0xe0, 0xff, 0xff, 0xf0, 0xee, 0xf0, 0x00,
+	0x01, 0x00, 0x01, 0x01,
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+	0x01, 0x0f, 0x00, 0x11,
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+	0x00, 0x00, 0xff, 0x10,
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+	0x10, 0x00, 0x0e, 0xfe,
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+	0xe0, 0xf0, 0xee, 0x00,
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+	0x00, 0xf0, 0x00, 0xff,
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+	0x00, 0x0f, 0x00, 0x10,
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+	0x01, 0x00, 0x00, 0xff,
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+	0x12, 0x10, 0x00, 0x01,
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+	0x00, 0x00, 0x00, 0x12,
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+	0x00, 0xe0, 0xf0, 0xf0,
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+	0xf0, 0xe0, 0x00, 0x00,
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+	0x00, 0x11, 0x02, 0x11,
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+	0x01, 0xf1, 0x02, 0x10,
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+	0x00, 0xf0, 0xff, 0xee,
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+	0x00, 0x00, 0x02, 0x00,
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+	0x00, 0x0e, 0x0f, 0xff,
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+	0x01, 0x02, 0x01, 0x01,
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+	0xff, 0xe0, 0x00, 0x00,
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+	0x00, 0xff, 0x00, 0x00,
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+	0x0e, 0x0f, 0x10, 0x11,
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+	0x02, 0x00, 0xf1, 0x01,
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+	0x10, 0x00, 0xf0, 0x01,
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+	0x00, 0x00, 0x0e, 0xf0,
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+	0x01, 0x1f, 0x1f, 0x01,
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+	0x10, 0x0f, 0x00, 0x00,
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+	0x00, 0xe0, 0x0f, 0xff,
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+	0x10, 0xf1, 0xf0, 0x00,
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+	0x11, 0x00, 0x00, 0x1f,
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+	0xff, 0xfe, 0x01, 0x00,
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+	0xe0, 0xf0, 0x00, 0x00,
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+	0xf0, 0x00, 0x00, 0x01,
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+	0x01, 0x00, 0x01, 0x01,
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+	0x30, 0x20, 0x1f, 0x20,
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+	0x00, 0xfe, 0x0f, 0x01,
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+	0x0f, 0xf0, 0xfe, 0xe0,
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+	0x11, 0x00, 0x0f, 0x00,
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+	0x00, 0x00, 0x0f, 0x00,
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+	0x00, 0x00, 0x00, 0x00,
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+	0x00, 0x10, 0x02, 0x02,
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+	0xff, 0xf0, 0xf1, 0x01,
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+	0x00, 0x22, 0x01, 0x21,
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+	0x10, 0x1f, 0x0e, 0x20,
+	0x10, 0x00, 0x20, 0x01, 0x10, 0x00, 0x1f, 0xfe, 0xff, 0x00, 0xef, 0x10,
+	0x00, 0x00, 0xff, 0xf0,
+	0x00, 0x02, 0xf0, 0xf0, 0xff, 0x00, 0xff, 0x00, 0xe0, 0x0f, 0xf0, 0x00,
+	0xf0, 0x01, 0x00, 0x0f,
+	0x00, 0xf1, 0x00, 0x11, 0x10, 0x01, 0x01, 0xe0, 0x00, 0x00, 0xf2, 0x00,
+	0x0f, 0x21, 0x02, 0xf0,
+	0x01, 0x02, 0x11, 0x0f, 0xff, 0xef, 0x01, 0x12, 0x12, 0x00, 0x00, 0x00,
+	0x0f, 0x00, 0x00, 0x10,
+	0x00, 0x0f, 0xfe, 0x0f, 0x0f, 0xfd, 0x0f, 0x00, 0x1f, 0x00, 0x0e, 0x00,
+	0x12, 0x10, 0x00, 0x0e,
+	0x00, 0x10, 0x0f, 0x11, 0x11, 0x10, 0x01, 0x00, 0x10, 0x10, 0x00, 0x01,
+	0xf2, 0x00, 0x01, 0x12,
+	0x01, 0xe0, 0x00, 0x20, 0x00, 0x01, 0x00, 0x0f, 0x2f, 0x00, 0x21, 0x00,
+	0xe0, 0x0f, 0x00, 0x1f,
+	0x00, 0xef, 0x00, 0x0f, 0xf0, 0x01, 0xe0, 0x00, 0x00, 0x01, 0x01, 0x00,
+	0xff, 0xf0, 0x00, 0x00,
+	0x10, 0x00, 0x01, 0xf0, 0xf0, 0xff, 0x00, 0xff, 0x00, 0x22, 0x00, 0xff,
+	0x00, 0x10, 0x00, 0x10,
+	0x1f, 0x00, 0x10, 0x20, 0x00, 0x11, 0x10, 0x10, 0x0f, 0x00, 0x0f, 0x11,
+	0x10, 0x01, 0x10, 0x00,
+	0x10, 0x01, 0x00, 0x00, 0x00, 0x00, 0x01, 0xf0, 0xe0, 0x0f, 0xef, 0xf0,
+	0xff, 0xe0, 0xff, 0x00,
+	0xf0, 0x01, 0xf0, 0x00, 0x10, 0x0e, 0x00, 0x00, 0x01, 0x00, 0x00, 0xfe,
+	0xf0, 0xf0, 0x01, 0xf0,
+	0xff, 0x00, 0xf1, 0xf0, 0x00, 0x01, 0x12, 0x10, 0x02, 0x10, 0x00, 0x10,
+	0x12, 0x00, 0x11, 0x01,
+	0x00, 0x00, 0x10, 0x00, 0xf0, 0x00, 0x0f, 0x00, 0x21, 0x00, 0x00, 0x1f,
+	0x10, 0x00, 0x0f, 0x0f,
+	0x0f, 0x0e, 0x0f, 0xfe, 0x0f, 0x0f, 0x10, 0x00, 0x0f, 0x0f, 0xee, 0x00,
+	0xf0, 0x0f, 0xff, 0x00,
+	0x01, 0x00, 0xf0, 0xff, 0x00, 0x00, 0x01, 0x0f, 0xee, 0x10, 0x02, 0x10,
+	0x00, 0xf0, 0xf0, 0x0f,
+	0xf0, 0x0f, 0x01, 0x02, 0x01, 0x01, 0x00, 0xf0, 0xf0, 0xf1, 0x01, 0x01,
+	0x02, 0xf0, 0x01, 0x00,
+	0x01, 0x00, 0x00, 0x01, 0x01, 0x11, 0x00, 0x01, 0x10, 0x1f, 0x00, 0x0e,
+	0x10, 0x10, 0x10, 0x10,
+	0x10, 0x0f, 0x10, 0x1f, 0x10, 0x0f, 0x10, 0x1f, 0x00, 0xfe, 0x0f, 0x00,
+	0x00, 0x00, 0x10, 0xfe,
+	0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x00, 0xe0, 0xf0, 0xee, 0xff, 0xff,
+	0xef, 0xe0, 0x0f, 0xee,
+	0xf0, 0x00, 0x02, 0x01, 0x00, 0x0e, 0x00, 0x00, 0xf0, 0x00, 0x20, 0x0f,
+	0x11, 0x00, 0x01, 0xf0,
+	0x0f, 0x00, 0x00, 0x11, 0x21, 0x00, 0x00, 0x02, 0x21, 0x00, 0x0f, 0x00,
+	0x0f, 0x00, 0x01, 0x00,
+	0x00, 0x11, 0xf0, 0xf0, 0xff, 0xee, 0x0f, 0x00, 0x00, 0x00, 0x01, 0xff,
+	0x0f, 0x00, 0x00, 0x0f,
+	0xf0, 0x00, 0x01, 0x20, 0x01, 0x00, 0x10, 0x21, 0x13, 0x10, 0xf0, 0x00,
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+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00,
+};
+
+#endif	/* WAVEDATA_H */
diff --git a/usr/src/cmd/audio/mixerctl/Makefile b/usr/src/cmd/audio/mixerctl/Makefile
index 258b75f792..bc800eb4a6 100644
--- a/usr/src/cmd/audio/mixerctl/Makefile
+++ b/usr/src/cmd/audio/mixerctl/Makefile
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Makefile for cmd/audio/mixerctl
 
 PROG= mixerctl
diff --git a/usr/src/cmd/audio/mixerctl/mixerctl.c b/usr/src/cmd/audio/mixerctl/mixerctl.c
index 16509a5b71..2449003649 100644
--- a/usr/src/cmd/audio/mixerctl/mixerctl.c
+++ b/usr/src/cmd/audio/mixerctl/mixerctl.c
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,680 +19,1181 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
+#include <stdio.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <string.h>
+#include <strings.h>
+#include <locale.h>
+#include <libintl.h>
+#include <stdarg.h>
+#include <stddef.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <ctype.h>
+#include <sys/param.h>
+#include <sys/soundcard.h>
+
+#if !defined(TEXT_DOMAIN)	/* Should be defined by cc -D */
+#define	TEXT_DOMAIN "SYS_TEST"	/* Use this only if it weren't */
+#endif
+
+#define	_(s)	gettext(s)
+
+#ifndef PATH_MAX
+#define	PATH_MAX	1024
+#endif
+
+#define	AUDIO_CTRL_STEREO_LEFT(v)	((uint8_t)((v) & 0xff))
+#define	AUDIO_CTRL_STEREO_RIGHT(v)	((uint8_t)(((v) >> 8) & 0xff))
+#define	AUDIO_CTRL_STEREO_VAL(l, r)	(((l) & 0xff) | (((r) & 0xff) << 8))
+
 
 /*
- * mixerctl command:
- *	mixerctl [-a|-d dev] [-iv] [-e|-o]
- *
- *	NOTE: Option D is not documented, it is for debugging.
+ * Column display information
+ * All are related to the types enumerated in col_t and any change should be
+ * reflected in the corresponding indices and offsets for all the variables
+ * accordingly.  Most tweaks to the display can be done by adjusting the
+ * values here.
  */
 
-#include <stdio.h>
-#include <stdlib.h>		/* getopt() */
-#include <errno.h>		/* errno */
-#include <strings.h>		/* strrchr(), strcpy(), strcat(), strerror() */
-#include <locale.h>		/* setlocale() */
-#include <libintl.h>		/* textdomain(), gettext() */
-#include <sys/types.h>		/* open(), fstat() */
-#include <sys/stat.h>		/* open(), fstat() */
-#include <fcntl.h>		/* open() */
-#include <unistd.h>		/* close(), ioctl() */
-#include <stropts.h>		/* ioctl() */
-#include <ctype.h>		/* isdigit() */
-#include <dirent.h>		/* opendir(), readdir(), closedir() */
-#include <sys/audioio.h>
-#include <sys/mixer.h>
-
-#define	GETOPT		"ad:iveoD"
-
-#define	DEFAULT_DEVICE	"/dev/audioctl"
-#define	CONTROL		"ctl"
-
-#define	SOUND_DIR	"/dev/sound"
-#define	DOT		"."
-#define	DOTDOT		".."
-#define	SLASH		"/"
-
-#define	ENV		"AUDIODEV"
-
-#define	AUX1		"AUX1"
-#define	AUX2		"AUX2"
-#define	CD		"CD"
-#define	HDPHONE		"HDPHONE"
-#define	LINE		"LINE"
-#define	LOOPBACK	"CODEC LOOPBACK"
-#define	MIC		"MIC"
-#define	NONE		"NONE"
-#define	SPDIF		"SPDIF"
-#define	SPEAKER		"SPKR"
-#define	SUNVTSLB	"SunVTS LOOPBACK"
-#define	UNKNOWN		"UNKNOWN"
-#define	SEPARATOR	"|"
-
-#define	ALL_PLAY_PORTS	(AUDIO_SPEAKER|AUDIO_HEADPHONE|AUDIO_LINE_OUT|\
-				AUDIO_SPDIF_OUT|AUDIO_AUX1_OUT|AUDIO_AUX2_OUT)
-#define	ALL_REC_PORTS	(AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CD|\
-				AUDIO_SPDIF_IN|AUDIO_AUX1_IN|AUDIO_AUX2_IN|\
-				AUDIO_CODEC_LOOPB_IN|AUDIO_SUNVTS)
-
-#define	BUF_SIZE	128
-#define	PPORT_STR_SIZE	30
-
-#define	NOT_SUPPORTED	(-1)
-
-typedef struct info {
-	char	*pgm;
-	char	*opt;
-	char	*device;
-	int	all;
-	int	debug;
-	int	disable;
-	int	enable;
-	int	info;
-	int	verbose;
-} info_t;
-
-/* local functions */
-static int doit(info_t *, char *);
-static void pport2str(uint_t, char *);
-static void rport2str(uint_t, char *);
+/* types of columns displayed */
+typedef enum { COL_DV = 0, COL_NM, COL_VAL, COL_TYP, COL_SEL} col_t;
 
-int
-main(const int argc, char * const *argv)
+/* corresponding sizes of columns; does not include trailing null */
+#define	COL_DV_SZ	16
+#define	COL_NM_SZ	24
+#define	COL_VAL_SZ	10
+#define	COL_TYP_SZ	8
+#define	COL_SEL_SZ	20
+#define	COL_MAX_SZ	64
+
+/* corresponding sizes of columns, indexed by col_t value */
+static int col_sz[] = {
+	COL_DV_SZ, COL_NM_SZ, COL_VAL_SZ, COL_TYP_SZ, COL_SEL_SZ
+};
+
+/* used by callers of the printing function */
+typedef struct col_prt {
+	char *col_dv;
+	char *col_nm;
+	char *col_val;
+	char *col_typ;
+	char *col_sel;
+} col_prt_t;
+
+/* columns displayed in order with vopt = 0 */
+static int col_dpy[] = {COL_DV, COL_NM, COL_VAL, COL_SEL};
+static int col_dpy_len = sizeof (col_dpy) / sizeof (*col_dpy);
+
+/* tells the printing function what members to use; follows col_dpy[] */
+static size_t col_dpy_prt[] = {
+	offsetof(col_prt_t, col_dv),
+	offsetof(col_prt_t, col_nm),
+	offsetof(col_prt_t, col_val),
+	offsetof(col_prt_t, col_sel)
+};
+
+/* columns displayed in order with vopt = 1 */
+static int col_dpy_vopt[] = { COL_NM, COL_VAL, COL_TYP, COL_SEL};
+static int col_dpy_vopt_len = sizeof (col_dpy_vopt) / sizeof (*col_dpy_vopt);
+
+/* tells the printing function what members to use; follows col_dpy_vopt[] */
+static size_t col_dpy_prt_vopt[] = {
+	offsetof(col_prt_t, col_nm),
+	offsetof(col_prt_t, col_val),
+	offsetof(col_prt_t, col_typ),
+	offsetof(col_prt_t, col_sel)
+};
+
+/* columns displayed in order with tofile = 1 */
+static int col_dpy_tofile[] = { COL_NM, COL_VAL};
+static int col_dpy_tofile_len = sizeof (col_dpy_tofile) /
+    sizeof (*col_dpy_tofile);
+
+/* tells the printing function what members to use; follows col_dpy_tofile[] */
+static size_t col_dpy_prt_tofile[] = {
+	offsetof(col_prt_t, col_nm),
+	offsetof(col_prt_t, col_val)
+};
+
+
+/*
+ * mixer and control accounting
+ */
+
+typedef struct cinfo {
+	oss_mixext ci;
+	oss_mixer_enuminfo *enump;
+} cinfo_t;
+
+typedef struct mlist {
+	oss_card_info cdi;
+	oss_mixerinfo mi;
+
+	int cmax;
+	cinfo_t *controls;
+
+	int mfd;
+
+	struct mlist *nextp;
+} mlist_t;
+
+
+/*PRINTFLIKE1*/
+static void
+msg(char *fmt, ...)
 {
-	DIR		*dirp;
-	struct dirent	*dp;
-	info_t		info;
-	char		dev_buffer[BUF_SIZE];
-	int		c;
-	int		errflag = 0;
+	va_list ap;
 
-	(void) setlocale(LC_ALL, "");
+	va_start(ap, fmt);
+	(void) vprintf(fmt, ap);
+	va_end(ap);
+}
 
-#if !defined(TEXT_DOMAIN)	/* Should be defined by cc -D */
-#define	TEXT_DOMAIN "SYS_TEST"	/* Use this only if it weren't */
-#endif
-	(void) textdomain(TEXT_DOMAIN);
+/*PRINTFLIKE1*/
+static void
+warn(char *fmt, ...)
+{
+	va_list ap;
 
-	/* make sure info structure is zeroed out */
-	(void) memset(&info, 0, sizeof (info));
+	va_start(ap, fmt);
+	(void) vfprintf(stderr, fmt, ap);
+	va_end(ap);
+}
 
-	if (info.pgm = strrchr(argv[0], '/')) {
-		++info.pgm;
-	} else {
-		info.pgm = argv[0];
-	}
 
-	info.opt = GETOPT;
+static void
+mixers_free(mlist_t **mlistpp)
+{
+	mlist_t *mlp = *mlistpp;
+	mlist_t *nextp;
+	int i;
 
-	/* check for options, they're listed at the top */
-	while ((c = getopt(argc, argv, info.opt)) != EOF) {
-		switch (c) {
-		case 'a':	info.all++;		break;
-		case 'd':	(void) strcpy(dev_buffer, optarg);
-				info.device = dev_buffer;
-				break;
-		case 'e':	info.enable++;		break;
-		case 'o':	info.disable++;		break;
-		case 'i':	info.info++;		break;
-		case 'v':	info.verbose++;		break;
-		case 'D':	info.debug++;		break;
-		default:	errflag++;		break;
+	while (mlp != NULL) {
+		nextp = mlp->nextp;
+
+		for (i = 0; i < mlp->cmax; i++) {
+			if (mlp->controls[i].enump != NULL)
+				free(mlp->controls[i].enump);
 		}
+
+		if (mlp->mfd >= 0)
+			(void) close(mlp->mfd);
+
+		free(mlp);
+		mlp = nextp;
 	}
 
-	/* now we do some checks to make sure we haven't done anything bad */
-	if (errflag || (info.all && info.device) ||
-	    (info.enable && info.disable) || optind != argc) {
-		(void) fprintf(stderr,
-		    gettext("usage: %s [-a|-d dev] [-iv] [-e|-o]\n"),
-		    info.pgm);
-		return (1);
+	*mlistpp = NULL;
+}
+
+
+/*
+ * adds to the end of mlistpp and returns a pointer to the new entry
+ */
+static mlist_t *
+mlist_addnew(mlist_t **mlistpp)
+{
+	mlist_t *p;
+	mlist_t *mlp = calloc(1, sizeof (*mlp));
+
+	mlp->cdi.card = -1;
+	mlp->mi.dev = -1;
+	mlp->mfd = -1;
+
+	if (*mlistpp == NULL)
+		*mlistpp = mlp;
+	else {
+		for (p = *mlistpp; p->nextp != NULL; p = p->nextp) {}
+
+		p->nextp = mlp;
+	}
+	return (mlp);
+}
+
+
+/*
+ * cinfop->enump needs to be present
+ * idx should be: >= 0 to < cinfop->ci.maxvalue
+ */
+static char *
+get_enum_str(cinfo_t *cinfop, int idx)
+{
+	int sz = sizeof (*cinfop->ci.enum_present) * 8;
+
+	if (cinfop->ci.enum_present[idx / sz] & (1 << (idx % sz)))
+		return (cinfop->enump->strings + cinfop->enump->strindex[idx]);
+
+	return (NULL);
+}
+
+
+/*
+ * caller fills in mlistp->mi.devnode; func fills in the rest
+ */
+static int
+mlist_getinfo(mlist_t *mlistp)
+{
+	int mfd = -1;
+	int rv = -1;
+	int i;
+	char *mpath = mlistp->mi.devnode;
+	cinfo_t *cinfop;
+
+	mfd = open(mpath, O_RDWR);
+	if (mfd < 0) {
+		rv = errno;
+		perror(_("Error opening mixer"));
+		goto OUT;
 	}
+	mlistp->mfd = mfd;
 
-	/* make sure we have something to do */
-	if (!info.disable && !info.enable && !info.info && !info.verbose) {
-		info.info++;
+	if (ioctl(mfd, SNDCTL_CARDINFO, &mlistp->cdi) < 0) {
+		rv = errno;
+		perror(_("Error getting card info"));
+		goto OUT;
 	}
 
-	/* if "all" then we loop, otherwise we just take care of business */
-	if (info.all) {
-		/*
-		 * Devices in /dev/sound are in the form of:
-		 *	dev_number{function}
-		 *
-		 * Currently there are only entries like /dev/sound/0 (audio
-		 * device) and /dev/sound/0ctl (audio control device).
-		 */
-		if ((dirp = opendir(SOUND_DIR)) == NULL) {
-			(void) fprintf(stderr, gettext(
-			    "Couldn't open directory %s: %s\n"), SOUND_DIR,
-			    strerror(errno));
-			return (1);
+	mlistp->cmax = -1;
+	if (ioctl(mfd, SNDCTL_MIX_NREXT, &mlistp->cmax) < 0) {
+		rv = errno;
+		perror(_("Error getting control count"));
+		goto OUT;
+	}
+
+	mlistp->controls = calloc(mlistp->cmax, sizeof (*mlistp->controls));
+
+	for (i = 0; i < mlistp->cmax; i++) {
+		cinfop = &mlistp->controls[i];
+
+		cinfop->ci.dev = -1;
+		cinfop->ci.ctrl = i;
+
+		if (ioctl(mfd, SNDCTL_MIX_EXTINFO, &cinfop->ci) < 0) {
+			rv = errno;
+			perror(_("Error getting control info"));
+			goto OUT;
 		}
 
-		while ((dp = readdir(dirp)) != NULL) {
-			char	*cp = dp->d_name;
+		if (cinfop->ci.type == MIXT_ENUM) {
+			cinfop->enump = calloc(1, sizeof (*cinfop->enump));
+			cinfop->enump->dev = -1;
+			cinfop->enump->ctrl = cinfop->ci.ctrl;
 
-			/* skip . and .. */
-			if (strcmp(cp, DOT) == 0 || strcmp(cp, DOTDOT) == 0) {
-				if (info.debug) {
-					(void) printf(gettext("Skipping %s\n"),
-					    cp);
-				}
-				continue;
+			if (ioctl(mfd, SNDCTL_MIX_ENUMINFO,
+			    cinfop->enump) < 0) {
+				rv = errno;
+				perror(_("Error getting enum info"));
+				goto OUT;
 			}
+		}
+	}
 
-			/* go past the device number */
-			while (isdigit(*cp)) {
-				cp++;
-			}
+	rv = 0;
 
-			/* if next char is NULL then not an audio ctl device */
-			if (*cp == '\0') {
-				if (info.debug) {
-					(void) printf(gettext("Skipping %s\n"),
-					    dp->d_name);
-				}
-				continue;
-			}
+OUT:
+	if (rv != 0) {
+		(void) close(mfd);
+		mlistp->mfd = -1;
+	}
+	return (rv);
+}
 
-			if (strcmp(cp, CONTROL) == 0) {
-				/* we've got a control device, so call it */
-				(void) strcpy(dev_buffer, SOUND_DIR);
-				(void) strcat(dev_buffer, SLASH);
-				(void) strcat(dev_buffer, dp->d_name);
 
-				if (info.debug) {
-					(void) printf(gettext(
-					    "Found device: %s\n"), dev_buffer);
-				}
+static int
+mixers_getinfo(char *devname, mlist_t **mlistpp)
+{
+	int rv = -1;
+	int fd = -1;
+	int i;
+	oss_sysinfo si;
+	oss_mixerinfo mi;
+	mlist_t *mlp;
+	char *mdef;
 
-				/* we don't bother with the return code */
-				(void) doit(&info, dev_buffer);
-			}
+	mdef = devname ? devname : "/dev/mixer";
+
+	if ((fd = open(mdef, O_RDWR)) < 0) {
+		rv = errno;
+		warn(_("Error opening mixer\n"));
+		goto OUT;
+	}
+
+	if (devname == NULL) {
+		if (ioctl(fd, SNDCTL_SYSINFO, &si) < 0) {
+			rv = errno;
+			perror(_("Error getting system information"));
+			goto OUT;
 		}
 
-		(void) closedir(dirp);
+		for (i = 0; i < si.nummixers; i++) {
+			mi.dev = i;
+			if (ioctl(fd, SNDCTL_MIXERINFO, &mi) != 0)
+				continue;
+
+			mi.dev = -1;
+			mlp = mlist_addnew(mlistpp);
+			mlp->mi = mi;
+
+			rv = mlist_getinfo(mlp);
+			if (rv != 0)
+				goto OUT;
+		}
 	} else {
-		/* make sure we've got a good device */
-		if (info.device == NULL) {
-			if ((info.device = getenv(ENV)) == NULL) {
-				if (info.debug) {
-					(void) printf(gettext(
-					    "using default device: %s\n"),
-					    DEFAULT_DEVICE);
-				}
-				info.device = DEFAULT_DEVICE;
-			} else {
-				/* using ENV value, so get it */
-				(void) strcpy(dev_buffer, info.device);
-				info.device = dev_buffer;
-				if (info.debug) {
-					(void) printf(gettext(
-					    "using ENV device: %s\n"),
-					    info.device);
-				}
-			}
+		mi.dev = -1;
+		if (ioctl(fd, SNDCTL_MIXERINFO, &mi) != 0) {
+			rv = errno;
+			perror(_("Error getting mixer information"));
+			goto OUT;
 		}
-		/* we have a device, make sure it is a control device */
-		if (strspn(CONTROL, info.device) != strlen(CONTROL)) {
-			(void) strcpy(dev_buffer, info.device);
-			(void) strcat(dev_buffer, CONTROL);
-			info.device = dev_buffer;
+		mi.dev = -1;
+		mlp = mlist_addnew(mlistpp);
+		mlp->mi = mi;
+
+		rv = mlist_getinfo(mlp);
+		if (rv != 0)
+			goto OUT;
+	}
+
+	rv = 0;
+
+OUT:
+	if (fd >= 0)
+		(void) close(fd);
+	return (rv);
+}
+
+
+static void
+mixers_prt_mlist(FILE *sfp, int tofile, mlist_t *mlistp)
+{
+	oss_card_info *cdip = &mlistp->cdi;
+
+	if (tofile) {
+		(void) fprintf(sfp, "# Device: %s\n", mlistp->mi.devnode);
+		(void) fprintf(sfp, "# Name    = %s\n", cdip->shortname);
+		(void) fprintf(sfp, "# Config  = %s\n", cdip->longname);
+
+		if (strlen(cdip->hw_info)) {
+			(void) fprintf(sfp, "# HW Info = %s",
+			    cdip->hw_info);
 		}
 
-		if (info.debug) {
-			(void) printf(
-			    gettext("calling doit(%s)\n"), info.device);
+		(void) fprintf(sfp, "#\n");
+	} else {
+		msg(_("Device: %s\n"), mlistp->mi.devnode);
+		msg(_("  Name    = %s\n"), cdip->shortname);
+		msg(_("  Config  = %s\n"), cdip->longname);
+
+		if (strlen(cdip->hw_info)) {
+			msg(_("  HW Info = %s"), cdip->hw_info);
 		}
-		return (doit(&info, info.device));
+
+		msg("\n");
 	}
+}
 
-	return (0);
 
-}	/* main() */
+static char *
+ctype_str(int type)
+{
+	switch (type) {
+	case MIXT_DEVROOT:
+		return (_("root"));
+	case MIXT_ONOFF:
+		return (_("boolean"));
+	case MIXT_ENUM:
+		return (_("enum"));
+	case MIXT_MONOSLIDER:
+		return (_("mono"));
+	case MIXT_STEREOSLIDER:
+		return (_("stereo"));
+	case MIXT_MARKER:
+		return (_("marker"));
+	case MIXT_GROUP:
+		return (_("group"));
+	default:
+		return (_("unknown"));
+	}
+}
+
 
-/*
- * doit()
- *
- * Description:
- *	This routine does all the real work
- *
- * Arguments:
- *	info_t	*info	Program info
- *	char	*dev	Device
- *
- * Returns:
- *	1		Error
- *	0		Okay
- */
 static int
-doit(info_t *info, char *dev)
+ctype_valid(int type)
 {
-	audio_device_t	dev_info;
-	audio_info_t	audio_info;
-	struct stat	buf;
-	char		*pencoding;
-	char		*rencoding;
-	char		pport_buf[BUF_SIZE];
-	char		rport_buf[BUF_SIZE];
-	int		fd;
-	int		mode;
-	int		new_mode;
-	int		rc = 0;
-
-	/* first, make sure the device is there */
-	if ((fd = open(dev, O_RDONLY)) < 0) {
-		/* return a meaningful error message if possible */
-		switch (errno) {
-		case EPERM :
-		case EACCES :
-			(void) fprintf(stderr,
-			    gettext("%s: device %s permission denied\n"),
-			    info->pgm, dev);
-			break;
-		case ENOENT :
-		case ENXIO :
-		case ENODEV :
-			(void) fprintf(stderr,
-			    gettext("%s: device %s does not exist\n"),
-			    info->pgm, dev);
-			break;
-		default:
-			(void) fprintf(stderr,
-			    gettext("%s: device %s not available\n"),
-			    info->pgm, dev);
-			break;
-		}
+	switch (type) {
+	case MIXT_ONOFF:
+	case MIXT_ENUM:
+	case MIXT_MONOSLIDER:
+	case MIXT_STEREOSLIDER:
 		return (1);
+	default:
+		return (0);
 	}
+}
 
-	/* make sure it is a special device */
-	if (fstat(fd, &buf) < 0) {
-		(void) fprintf(stderr,
-		    gettext("%s: fstat() of %s failed\n"), info->pgm, dev);
-		(void) close(fd);
-		return (1);
+
+static void
+mixers_prt_ctl_line(FILE *sfp, int tofile, col_prt_t *colp, int vopt)
+{
+	int i;
+	size_t *col_prtp;
+	int *col_dpyp;
+	int col_cnt;
+	int col_type;
+	int width;
+	char *colstr;
+	char cbuf[COL_MAX_SZ + 1];
+	char line[128];
+	char *colsep =  " ";
+
+	if (tofile) {
+		col_prtp = col_dpy_prt_tofile;
+		col_dpyp = col_dpy_tofile;
+		col_cnt = col_dpy_tofile_len;
+	} else if (vopt) {
+		col_prtp = col_dpy_prt_vopt;
+		col_dpyp = col_dpy_vopt;
+		col_cnt = col_dpy_vopt_len;
+	} else {
+		col_prtp = col_dpy_prt;
+		col_dpyp = col_dpy;
+		col_cnt = col_dpy_len;
 	}
-	if (!S_ISCHR(buf.st_mode)) {
-		(void) fprintf(stderr,
-		    gettext("%s: %s is not a character special device\n"),
-		    info->pgm, dev);
-		(void) close(fd);
-		return (1);
+
+	line[0] = '\0';
+
+	for (i = 0; i < col_cnt; i++) {
+		col_type = col_dpyp[i];
+		width = col_sz[col_type];
+		colstr = *(char **)(((size_t)colp) + col_prtp[i]);
+
+		(void) snprintf(cbuf, sizeof (cbuf), "%- *s",
+		    width > 0 ? width : 1,
+		    (colstr == NULL) ? "" : colstr);
+
+		(void) strlcat(line, cbuf, sizeof (line));
+		if (i < col_cnt - 1)
+			(void) strlcat(line, colsep, sizeof (line));
 	}
 
-	/* if verbose or info wanted, we get the device info */
-	if (info->verbose || info->info) {
-		if (ioctl(fd, AUDIO_GETDEV, &dev_info) < 0) {
-			(void) fprintf(stderr, gettext(
-			    "%s: AUDIO_GETDEV ioctl() for device %s failed\n"),
-			    info->pgm, dev);
-			(void) close(fd);
-			return (1);
-		}
+	(void) fprintf(sfp, "%s\n", line);
+}
 
-		(void) printf(gettext("Device %s:\n"), dev);
-		(void) printf(gettext("  Name    = %s\n"), dev_info.name);
-		(void) printf(gettext("  Version = %s\n"), dev_info.version);
-		(void) printf(gettext("  Config  = %s\n"), dev_info.config);
-	}
-
-	/* get the mixer mode, if there is a mixer */
-	if (ioctl(fd, AUDIO_MIXERCTL_GET_MODE, &mode) < 0) {
-		(void) printf(
-		    gettext("\n%s doesn't support the audio mixer function\n"),
-		    dev);
-		mode = NOT_SUPPORTED;
-	} else if (!info->enable && !info->disable) {
-		(void) printf(gettext("\nAudio mixer for %s is %s\n"),
-		    dev, (mode == AM_MIXER_MODE ?
-		    gettext("enabled") : gettext("disabled")));
-	} else if (info->enable) {
-		new_mode = AM_MIXER_MODE;
-		if (mode != AM_MIXER_MODE) {
-			if (ioctl(fd, AUDIO_MIXERCTL_SET_MODE, &new_mode) < 0) {
-				(void) fprintf(stderr, gettext(
-				    "%s: couldn't enable audio mixer for %s\n"),
-				    info->pgm, dev);
-				rc = 1;
-			}
-		}
-		if (!rc) {
-			(void) printf(gettext(
-			    "Audio mixer for %s is enabled\n"), dev);
-		}
+
+static void
+mixers_prt_ctl_hdr(FILE *sfp, int tofile, int vopt)
+{
+	col_prt_t col;
+
+	if (tofile) {
+		col.col_nm = _("#CONTROL");
+		col.col_val = _("VALUE");
 	} else {
-		/* sanity check */
-		if (!info->disable) {
-			(void) fprintf(stderr,
-			    gettext("%s: program error #1\n"), info->pgm);
-			rc = 1;
-		} else if (mode != AM_COMPAT_MODE) {
-			new_mode = AM_COMPAT_MODE;
-			if (ioctl(fd, AUDIO_MIXERCTL_SET_MODE, &new_mode) < 0) {
-				(void) fprintf(stderr, gettext("%s: "
-				    "couldn't disable audio mixer for %s\n"),
-				    info->pgm, dev);
-				rc = 1;
-			}
-		}
-		if (!rc) {
-			(void) printf(gettext(
-			    "Audio mixer for %s is disabled\n"), dev);
-		}
+		col.col_dv = _("DEVICE");
+		col.col_nm = _("CONTROL");
+		col.col_val = _("VALUE");
+		col.col_typ = _("TYPE");
+		col.col_sel = _("POSSIBLE");
 	}
+	mixers_prt_ctl_line(sfp, tofile, &col, vopt);
+}
 
-	/* only for verbose do we get the device's info_t structure */
-	if (info->verbose) {
-		if (ioctl(fd, AUDIO_GETINFO, &audio_info) < 0) {
-			(void) fprintf(stderr, gettext(
-			    "%s: AUDIO_GETINFO ioctl() for device %s failed\n"),
-			    info->pgm, dev);
-			(void) close(fd);
-			return (1);
-		}
 
-		switch (audio_info.play.encoding) {
-		case AUDIO_ENCODING_ULAW:
-			pencoding = gettext("u-law");
-			break;
-		case AUDIO_ENCODING_ALAW:
-			pencoding = gettext("A-law");
-			break;
-		case AUDIO_ENCODING_LINEAR:
-			pencoding = gettext("linear");
-			break;
-		default:
-			pencoding = gettext("UNKNOWN");
-			break;
-		}
+static int
+mixers_prt_cinfo(FILE *sfp, int tofile, mlist_t *mlistp, cinfo_t *cinfop,
+    int vopt)
+{
+	int mfd = mlistp->mfd;
+	char *devnm = mlistp->cdi.shortname;
+	oss_mixer_value cval;
+	char *str;
+	int i;
+	int idx = -1;
+	int rv = -1;
+	char valbuf[COL_VAL_SZ + 1];
+	char selbuf[COL_SEL_SZ + 1];
+	col_prt_t col;
 
-		switch (audio_info.record.encoding) {
-		case AUDIO_ENCODING_ULAW:
-			rencoding = gettext("u-law");
-			break;
-		case AUDIO_ENCODING_ALAW:
-			rencoding = gettext("A-law");
-			break;
-		case AUDIO_ENCODING_LINEAR:
-			rencoding = gettext("linear");
-			break;
-		default:
-			rencoding = gettext("UNKNOWN");
-			break;
-		}
+	cval.dev = -1;
+	cval.ctrl = cinfop->ci.ctrl;
 
-		(void) printf(
-		    gettext("Sample Rate\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.sample_rate, audio_info.record.sample_rate);
-		(void) printf(gettext("Channels\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.channels, audio_info.record.channels);
-		(void) printf(
-		    gettext("Precision\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.precision, audio_info.record.precision);
-		(void) printf(gettext("Encoding\n  Play\t\t%u (%s)\n"),
-		    audio_info.play.encoding, pencoding);
-		(void) printf(gettext("  Record\t%u (%s)\n"),
-		    audio_info.record.encoding, rencoding);
-		(void) printf(gettext("Gain\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.gain, audio_info.record.gain);
-		(void) printf(gettext("Balance\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.balance, audio_info.record.balance);
-		pport2str(audio_info.play.port, pport_buf),
-		rport2str(audio_info.record.port, rport_buf),
-		(void) printf(
-		    gettext("Port\n  Play\t\t0x%08x %s\n  Record\t0x%08x %s\n"),
-		    audio_info.play.port, pport_buf,
-		    audio_info.record.port, rport_buf);
-		pport2str(audio_info.play.avail_ports, pport_buf),
-		rport2str(audio_info.record.avail_ports, rport_buf),
-		(void) printf(
-		    gettext("Avail Ports\n  Play\t\t0x%08x %s\n"
-		    "  Record\t0x%08x %s\n"),
-		    audio_info.play.avail_ports, pport_buf,
-		    audio_info.record.avail_ports, rport_buf);
-		pport2str(audio_info.play.mod_ports, pport_buf),
-		rport2str(audio_info.record.mod_ports, rport_buf),
-		(void) printf(
-		    gettext("Mod Ports\n  Play\t\t0x%08x %s\n"
-		    "  Record\t0x%08x %s\n"),
-		    audio_info.play.mod_ports, pport_buf,
-		    audio_info.record.mod_ports, rport_buf);
-		(void) printf(gettext("Samples\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.samples, audio_info.record.samples);
-		(void) printf(gettext("Active\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.active, audio_info.record.active);
-		(void) printf(gettext("Pause\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.pause, audio_info.record.pause);
-		(void) printf(gettext("Error\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.error, audio_info.record.error);
-		(void) printf(gettext("EOF Count\n  Play\t\t%u\n"),
-		    audio_info.play.eof);
-		(void) printf(gettext("Waiting\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.waiting, audio_info.record.waiting);
-		(void) printf(gettext("Open\n  Play\t\t%u\n  Record\t%u\n"),
-		    audio_info.play.open, audio_info.record.open);
-		(void) printf(gettext("HW Features\t\t0x%08x\n"),
-		    audio_info.hw_features);
-		if (audio_info.hw_features & AUDIO_HWFEATURE_PLAY) {
-			(void) printf(gettext("  PLAY\n"));
-		}
-		if (audio_info.hw_features & AUDIO_HWFEATURE_RECORD) {
-			(void) printf(gettext("  RECORD\n"));
-		}
-		if (audio_info.hw_features & AUDIO_HWFEATURE_DUPLEX) {
-			(void) printf(gettext("  DUPLEX\n"));
+	if (ctype_valid(cinfop->ci.type)) {
+		if (ioctl(mfd, SNDCTL_MIX_READ, &cval) < 0) {
+			rv = errno;
+			perror(_("Error reading control\n"));
+			return (rv);
 		}
-		if (audio_info.hw_features & AUDIO_HWFEATURE_MSCODEC) {
-			(void) printf(gettext("  MULTI-STREAM CODEC\n"));
-		}
-		if (audio_info.hw_features & AUDIO_HWFEATURE_IN2OUT) {
-			(void) printf(gettext("  INPUT TO OUTPUT LOOPBACK\n"));
+	} else {
+		/* if verbose, then continue to display the "pseudo" ctrls */
+		if (vopt < 2 || tofile)
+			return (0);
+	}
+
+	/*
+	 * convert the control value into a string
+	 */
+	switch (cinfop->ci.type) {
+	case MIXT_ONOFF:
+		(void) snprintf(valbuf, sizeof (valbuf), "%s",
+		    cval.value ? _("on") : _("off"));
+		break;
+
+	case MIXT_MONOSLIDER:
+		(void) snprintf(valbuf, sizeof (valbuf), "%d",
+		    cval.value & 0xff);
+		break;
+
+	case MIXT_STEREOSLIDER:
+		(void) snprintf(valbuf, sizeof (valbuf), "%d:%d",
+		    (int)AUDIO_CTRL_STEREO_LEFT(cval.value),
+		    (int)AUDIO_CTRL_STEREO_RIGHT(cval.value));
+		break;
+
+	case MIXT_ENUM:
+		str = get_enum_str(cinfop, cval.value);
+		if (str == NULL) {
+			warn(_("Bad enum index %d for control '%s'\n"),
+			    cval.value, cinfop->ci.extname);
+			return (EINVAL);
 		}
-		(void) printf(gettext("SW Features\t\t0x%08x\n"),
-		    audio_info.sw_features);
-		if (audio_info.sw_features & AUDIO_SWFEATURE_MIXER) {
-			(void) printf(gettext("  MIXER\n"));
+
+		(void) snprintf(valbuf, sizeof (valbuf), "%s", str);
+		break;
+
+	default:
+		(void) snprintf(valbuf, sizeof (valbuf), "-");
+	}
+
+	/*
+	 * possible control values (range/selection)
+	 */
+	switch (cinfop->ci.type) {
+	case MIXT_ONOFF:
+		(void) snprintf(selbuf, sizeof (selbuf), _("on,off"));
+		break;
+
+	case MIXT_MONOSLIDER:
+		(void) snprintf(selbuf, sizeof (selbuf), "%d-%d",
+		    cinfop->ci.minvalue, cinfop->ci.maxvalue);
+		break;
+	case MIXT_STEREOSLIDER:
+		(void) snprintf(selbuf, sizeof (selbuf), "%d-%d:%d-%d",
+		    cinfop->ci.minvalue, cinfop->ci.maxvalue,
+		    cinfop->ci.minvalue, cinfop->ci.maxvalue);
+		break;
+
+	case MIXT_ENUM:
+		/*
+		 * display the first choice on the same line, then display
+		 * the rest on multiple lines
+		 */
+		selbuf[0] = 0;
+		for (i = 0; i < cinfop->ci.maxvalue; i++) {
+			str = get_enum_str(cinfop, i);
+			if (str == NULL)
+				continue;
+
+			if ((strlen(str) + 1 + strlen(selbuf)) >=
+			    sizeof (selbuf)) {
+				break;
+			}
+			if (strlen(selbuf)) {
+				(void) strlcat(selbuf, ",", sizeof (selbuf));
+			}
+
+			(void) strlcat(selbuf, str, sizeof (selbuf));
 		}
-		(void) printf(gettext("SW Features Enabled\t0x%08x\n"),
-		    audio_info.sw_features_enabled);
-		if (audio_info.sw_features_enabled & AUDIO_SWFEATURE_MIXER) {
-			(void) printf(gettext("  MIXER\n"));
+		idx = i;
+		break;
+
+	default:
+		(void) snprintf(selbuf, sizeof (selbuf), "-");
+	}
+
+	col.col_dv = devnm;
+	col.col_nm = strlen(cinfop->ci.extname) ?
+	    cinfop->ci.extname : cinfop->ci.id;
+	while (strchr(col.col_nm, '_') != NULL) {
+		col.col_nm = strchr(col.col_nm, '_') + 1;
+	}
+	col.col_val = valbuf;
+	col.col_typ = ctype_str(cinfop->ci.type);
+	col.col_sel = selbuf;
+	mixers_prt_ctl_line(sfp, tofile, &col, vopt);
+
+	/* print leftover enum value selections */
+	while ((!tofile) && (idx >= 0) && (idx < cinfop->ci.maxvalue)) {
+		selbuf[0] = 0;
+		for (i = idx; i < cinfop->ci.maxvalue; i++) {
+			str = get_enum_str(cinfop, i);
+			if (str == NULL)
+				continue;
+
+			if ((strlen(str) + 1 + strlen(selbuf)) >=
+			    sizeof (selbuf)) {
+				break;
+			}
+			if (strlen(selbuf)) {
+				(void) strlcat(selbuf, ",", sizeof (selbuf));
+			}
+
+			(void) strlcat(selbuf, str, sizeof (selbuf));
 		}
+		idx = i;
+		col.col_dv = NULL;
+		col.col_nm = NULL;
+		col.col_val = NULL;
+		col.col_typ = NULL;
+		col.col_sel = selbuf;
+		mixers_prt_ctl_line(sfp, tofile, &col, vopt);
 	}
 
-	(void) close(fd);
+	return (0);
+}
 
-	(void) printf("\n");
 
-	return (rc);
+static int
+mixers_prt_controls(FILE *sfp, int tofile, mlist_t *mlistp, char *cname,
+    int vopt)
+{
+	cinfo_t *cinfop;
+	char *n;
+	int i;
+	int rv;
+	int rval = 0;
 
-}	/* doit() */
+	for (i = 0; i < mlistp->cmax; i++) {
+		cinfop = &mlistp->controls[i];
 
-/*
- * pport2str()
- *
- * Description:
- *	Convert the play port bit map into strings. This makes the port number
- *	a bit more readable. The returned string is in parenthesis and always
- *	has 30 valid characters for spacing.
- *
- * Arguments:
- *	uint	port		The port to convert
- *	char	*buf		The storage to put the string into
- *
- * Returns:
- *	void
- */
-static void
-pport2str(uint_t port, char *buf)
+		n = cinfop->ci.extname;
+		while (strchr(n, '_') != NULL) {
+			n = strchr(n, '_') + 1;
+		}
+		if (cname != NULL && strcmp(cname, n) != 0)
+			continue;
+
+		rv = mixers_prt_cinfo(sfp, tofile, mlistp, cinfop, vopt);
+		if (rv != 0)
+			rval = rv;
+	}
+
+	return (rval);
+}
+
+
+static int
+mixers_sav_prt(FILE *sfp, int tofile, mlist_t *mlistp, char *cname,
+    int eopt, int lopt, int vopt)
 {
-	int		sep = 0;
+	mlist_t *mlp;
+	int rv;
+	int rval = 0;
+
+	for (mlp = mlistp; mlp != NULL; mlp = mlp->nextp) {
+
+		if ((mlp->mi.enabled == 0) && (vopt == NULL))
+			continue;
 
-	(void) strcpy(buf, gettext("("));
+		if ((lopt) || (vopt) || (tofile))
+			mixers_prt_mlist(sfp, tofile, mlp);
 
-	if (port & AUDIO_SPEAKER) {
-		(void) strcat(buf, gettext(SPEAKER));
-		sep++;
+		if (eopt)
+			msg(_("Audio mixer for %s is enabled\n"),
+			    mlp->mi.devnode);
+
+		if ((lopt || eopt) && !tofile)
+			continue;
+
+		if ((mlistp->cmax > 0) && (vopt || (mlp == mlistp)))
+			mixers_prt_ctl_hdr(sfp, tofile, vopt);
+		rv = mixers_prt_controls(sfp, tofile, mlp, cname, vopt);
+		if (rv != 0)
+			rval = rv;
+
+		if (vopt)
+			msg("\n");
 	}
-	if (port & AUDIO_HEADPHONE) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
+
+	return (rval);
+}
+
+static int
+mixers_save(mlist_t *mlistp, char *sstr, int ovopt, int vopt)
+{
+	FILE		*sfp;
+	int		fp;
+	int		retval;
+	struct	stat	sbuf;
+
+	if (stat(sstr, &sbuf) == 0) {
+		if (!ovopt) {
+			warn(_("Would over write existing file [%s]\n"), sstr);
+			return (EINVAL);
 		} else {
-			sep++;
+			if (unlink(sstr)) {
+				retval = errno;
+				perror(_("Failed to unlink existing file"));
+				return (retval);
+			}
 		}
-		(void) strcat(buf, gettext(HDPHONE));
 	}
-	if (port & AUDIO_LINE_OUT) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
-		}
-		(void) strcat(buf, gettext(LINE));
+
+	if ((fp = open(sstr, O_WRONLY|O_CREAT|O_EXCL, 0666)) == 0) {
+		retval = errno;
+		perror(_("Failed to create file"));
+		return (retval);
 	}
-	if (port & AUDIO_SPDIF_OUT) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
+
+	sfp = fdopen(fp, "w");
+	if (sfp == NULL)
+		return (errno);
+	retval = mixers_sav_prt(sfp, 1, mlistp, NULL, 0, 0, vopt);
+	if (fclose(sfp))
+		return (errno);
+	return (retval);
+}
+
+static int
+mixers_set_cinfo(mlist_t *mlistp, cinfo_t *cinfop, char *wstr, int vopt)
+{
+	int mfd = mlistp->mfd;
+	oss_mixer_value cval;
+	int wlen = strlen(wstr);
+	int lval, rval;
+	char *lstr, *rstr;
+	char *str;
+	int i;
+	int rv = -1;
+
+	cval.dev = -1;
+	cval.ctrl = cinfop->ci.ctrl;
+	cval.value = 0;
+
+	switch (cinfop->ci.type) {
+	case MIXT_ONOFF:
+		cval.value = (strncmp(_("on"), wstr, wlen) == 0) ? 1 : 0;
+		break;
+
+	case MIXT_MONOSLIDER:
+		cval.value = atoi(wstr);
+		break;
+
+	case MIXT_STEREOSLIDER:
+		lstr = wstr;
+		rstr = strchr(wstr, ':');
+		if (rstr != NULL) {
+			*rstr = '\0';
+			rstr++;
+
+			rval = atoi(rstr);
+			lval = atoi(lstr);
+
+			rstr--;
+			*rstr = ':';
 		} else {
-			sep++;
+			lval = atoi(lstr);
+			rval = lval;
 		}
-		(void) strcat(buf, gettext(SPDIF));
-	}
-	if (port & AUDIO_AUX1_OUT) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+
+		cval.value = AUDIO_CTRL_STEREO_VAL(lval, rval);
+		break;
+
+	case MIXT_ENUM:
+		for (i = 0; i < cinfop->ci.maxvalue; i++) {
+			str = get_enum_str(cinfop, i);
+			if (str == NULL)
+				continue;
+
+			if (strncmp(wstr, str, wlen) == 0) {
+				cval.value = i;
+				break;
+			}
 		}
-		(void) strcat(buf, gettext(AUX1));
-	}
-	if (port & AUDIO_AUX2_OUT) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+
+		if (i >= cinfop->ci.maxvalue) {
+			warn(_("Invalid enumeration value"));
+			return (EINVAL);
 		}
-		(void) strcat(buf, gettext(AUX2));
+		break;
+
+	default:
+		warn(_("Unsupported control type\n"));
+		return (EINVAL);
 	}
-	if (port & ~ALL_PLAY_PORTS) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
-		}
-		(void) strcat(buf, gettext(UNKNOWN));
+
+	if (vopt) {
+		msg(_("%s: '%s' set to '%s'\n"), mlistp->cdi.shortname,
+		    cinfop->ci.extname, wstr);
 	}
 
-	if (sep == 0) {
-		(void) strcat(buf, gettext(NONE));
+	if (ioctl(mfd, SNDCTL_MIX_WRITE, &cval) < 0) {
+		rv = errno;
+		perror(_("Error writing control"));
+		return (rv);
 	}
 
-	(void) strcat(buf, gettext(")"));
+	rv = 0;
+	return (rv);
+}
 
-}	/* pport2str() */
 
-/*
- * rport2str()
- *
- * Description:
- *	Convert the record port bit map into strings. This makes the port
- *	number a bit more readable. The returned string in parenthesis.
- *
- * Arguments:
- *	uint	port		The port to convert
- *	char	*buf		The storage to put the string into
- *
- * Returns:
- *	void
- */
-static void
-rport2str(uint_t port, char *buf)
+static int
+mixers_set_controls(mlist_t *mlistp, char *cname, char *wstr, int vopt)
 {
-	int		sep = 0;
+	cinfo_t *cinfop;
+	char	*n;
+	int i;
+	int rv;
+	int rval = 0;
 
-	(void) strcpy(buf, gettext("("));
+	/*
+	 * Note errors that occur but continue on to the next control;
+	 * XXX
+	 * we allow multiple controls with the same name
+	 */
+	for (i = 0; i < mlistp->cmax; i++) {
+		cinfop = &mlistp->controls[i];
+		n = cinfop->ci.extname;
+		while (strchr(n, '_') != NULL) {
+			n = strchr(n, '_') + 1;
+		}
+		if (cname != NULL && strcmp(cname, n) != 0)
+			continue;
 
-	if (port & AUDIO_MICROPHONE) {
-		(void) strcat(buf, gettext(MIC));
-		sep++;
+		rv = mixers_set_cinfo(mlistp, cinfop, wstr, vopt);
+		if (rv != 0)
+			rval = rv;
 	}
-	if (port & AUDIO_LINE_IN) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
-		}
-		(void) strcat(buf, gettext(LINE));
+
+	return (rval);
+}
+
+
+static int
+mixers_set(mlist_t *mlistp, char *cname, char *wstr, int vopt)
+{
+	mlist_t *mlp;
+	int rv;
+	int rval = 0;
+
+	/*
+	 * Note errors that occur but continue on to the next mixer; this is
+	 * to allow us to specify a single control to set on all mixers,
+	 * if such a control is available on the mixer
+	 */
+	for (mlp = mlistp; mlp != NULL; mlp = mlp->nextp) {
+
+		rv = mixers_set_controls(mlp, cname, wstr, vopt);
+		if (rv != 0)
+			rval = rv;
 	}
-	if (port & AUDIO_CD) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+
+	return (rval);
+}
+
+static int
+mixers_restore(mlist_t *mlistp, char *sstr, int vopt)
+{
+	FILE	*sfp;
+	int	retval = 0;
+	int	rv;
+	int	lineno = 0;
+	char	linebuf[PATH_MAX];
+	char	*col_nm, *col_val;
+
+	sfp = fopen(sstr, "r");
+	if (sfp == NULL)
+		return (errno);
+
+	while (fgets(linebuf, sizeof (linebuf), sfp) != NULL) {
+		lineno++;
+		if (linebuf[strlen(linebuf) - 1] != '\n') {
+			warn(_("Warning: line too long at line %d\n"), lineno);
+			/* read in the rest of the line and discard it */
+			while (fgets(linebuf, sizeof (linebuf), sfp) != NULL &&
+			    (linebuf[strlen(linebuf) - 1] != '\n')) {
+				continue;
+			}
+			continue;
 		}
-		(void) strcat(buf, gettext(CD));
-	}
-	if (port & AUDIO_SPDIF_IN) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+		/* now we have a good line ... */
+		col_nm = strtok(linebuf, " \t\n");
+		/* skip comments and blank lines */
+		if ((col_nm == NULL) || (col_nm[0] == '#')) {
+			continue;
 		}
-		(void) strcat(buf, gettext(SPDIF));
-	}
-	if (port & AUDIO_AUX1_IN) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+		col_val = strtok(NULL, " \t\n");
+		if ((col_val == NULL) || (*col_val == 0)) {
+			warn(_("Warning: missing value at line %d\n"), lineno);
+			continue;
 		}
-		(void) strcat(buf, gettext(AUX1));
+
+		rv = mixers_set(mlistp, col_nm, col_val, vopt);
+		if (rv)
+			retval = rv;
 	}
-	if (port & AUDIO_AUX2_IN) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+
+	if (fclose(sfp))
+		return (errno);
+	return (retval);
+}
+
+
+static void
+help(void)
+{
+#define	HELP_STR	_(						\
+"mixerctl [ -d <device> | -a ] [ -f ] [ -s <file> | -r <file> ]\n"	\
+"mixerctl [ -d <device> | -a ] [ -Civ ]\n"				\
+"mixerctl [ -d <device> | -a ] [ -v ] -c <ctrl> [ -w <value> ]\n"	\
+"\n"									\
+"	-d	<device> = audio device path\n"				\
+"	-s	<file> = save control settings to file\n"		\
+"	-r	<file> = restore control settings from file\n"		\
+"	-f	force overwrite of existing control settings file\n"	\
+"	-a	process all audio devices\n"				\
+"	-C	dump all audio controls\n"				\
+"	-i	list audio device info\n"				\
+"	-v	verbose output; also affects output for -w;\n"		\
+"		multiple uses increase verbosity\n"			\
+"	-c	<ctrl> = control name\n"				\
+"	-w	<value> = control value\n"				\
+"		{integer} for mono; {integer:integer} for stereo;\n"	\
+"		{string} for enum; {on|off} for boolean\n"		\
+"\n"									\
+"	without arguments, device info for all devices will be listed\n")
+
+	(void) fprintf(stderr, HELP_STR);
+}
+
+
+#define	strstarts(s, prefix)	strncmp(s, prefix, strlen(prefix))
+
+int
+main(int argc, char **argv)
+{
+	int rv = -1;
+	int opt;
+	char *dstr = NULL;
+	char *cstr = NULL;
+	char *wstr = NULL;
+	char *sstr = NULL;
+	int aopt = 0;
+	int iopt = 0;
+	int vopt = 0;
+	int eopt = 0;
+	int oopt = 0;
+	int Copt = 0;
+	int sopt = 0;
+	int ropt = 0;
+	int ovopt = 0;
+	mlist_t *mlistp = NULL;
+
+	(void) setlocale(LC_ALL, "");
+	(void) textdomain(TEXT_DOMAIN);
+
+	while ((opt = getopt(argc, argv, "Cd:c:w:fs:r:ivhaeo")) != EOF) {
+		switch (opt) {
+		case 'C':
+			Copt++;
+			break;
+		case 'a':
+			aopt++;
+			break;
+		case 'e':
+			eopt++;
+			break;
+		case 'f':
+			ovopt++;
+			break;
+		case 'o':
+			oopt++;
+			break;
+		case 'd':
+			dstr = optarg;
+			break;
+
+		case 'c':
+			cstr = strdup(optarg);
+			break;
+		case 'w':
+			wstr = strdup(optarg);
+			break;
+		case 's':
+			sopt++;
+			sstr = strdup(optarg);
+			break;
+		case 'r':
+			ropt++;
+			sstr = strdup(optarg);
+			break;
+		case 'i':
+			iopt++;
+			break;
+		case 'v':
+			vopt++;
+			break;
+		case 'h':
+			help();
+			rv = 0;
+			goto OUT;
+		default:
+			help();
+			rv = EINVAL;
+			goto OUT;
 		}
-		(void) strcat(buf, gettext(AUX2));
 	}
-	if (port & AUDIO_CODEC_LOOPB_IN) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
-		} else {
-			sep++;
+
+	if (((eopt || oopt) && (Copt || sopt || ropt)) ||
+	    (Copt && cstr) ||
+	    ((Copt || cstr) && (sopt || ropt)) ||
+	    (aopt && dstr) ||
+	    (sopt && ropt)) {
+		warn(_("Illegal combination of options.\n"));
+		rv = EINVAL;
+		goto OUT;
+	}
+
+	if ((Copt) && (dstr == NULL)) {
+		/* asume -a when dumping controls with -C */
+		aopt++;
+	}
+
+	if ((!aopt) && (dstr == NULL)) {
+		/* if no mixer specified, assume a default */
+		dstr = getenv("AUDIODEV");
+		if (dstr == NULL) {
+			dstr = "/dev/audio";
 		}
-		(void) strcat(buf, gettext(LOOPBACK));
 	}
-	if (port & AUDIO_SUNVTS) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
+
+	if (oopt) {
+		/* legacy option, nuke it */
+		warn(_("Cannot disable audio mixer\n"));
+		rv = EINVAL;
+		goto OUT;
+	}
+
+	if (dstr != NULL) {
+		char	scratch[MAXPATHLEN + 1];
+		char	*s;
+		char *bn;
+		int num = -1;
+		/*
+		 * User may have specified:
+		 *
+		 * /dev/dsp[<num>]
+		 * /dev/mixer[<num>]
+		 * /dev/audio[<num>9]
+		 * /dev/audioctl[<num>]
+		 * /dev/sound/<num>{,ctl,dsp,mixer}
+		 * /dev/sound/<driver>:<num>{,ctl,dsp,mixer}
+		 *
+		 * We want to deal only with SADA names though.
+		 */
+
+		if (strcmp(dstr, "/dev/mixer") == 0) {
+			/* /dev/mixer node doesn't point to real hw */
+			dstr = "/dev/dsp";
+		}
+		if ((bn = strrchr(dstr, '/')) != NULL) {
+			bn++;
 		} else {
-			sep++;
+			bn = dstr;
 		}
-		(void) strcat(buf, gettext(SUNVTSLB));
-	}
-	if (port & ~ALL_REC_PORTS) {
-		if (sep) {
-			(void) strcat(buf, gettext(SEPARATOR));
+		if ((strstarts(dstr, "/dev/sound/") == 0) &&
+		    ((strrchr(dstr, ':')) != NULL)) {
+			char	*colon;
+			int	n;
+
+			(void) strlcpy(scratch, dstr, sizeof (scratch));
+			colon = strrchr(scratch, ':');
+			colon++;
+			n = atoi(colon);
+			*colon = '\0';
+			(void) snprintf(colon,
+			    sizeof (scratch) - strlen(scratch), "%dmixer", n);
+
+			dstr = strdup(scratch);
+
+		} else if ((strcmp(dstr, "/dev/audio") == 0) ||
+		    (strcmp(dstr, "/dev/audioctl") == 0) ||
+		    (strcmp(dstr, "/dev/dsp") == 0)) {
+			/* "default" device, read the link */
+			if (readlink(dstr, scratch, sizeof (scratch)) >= 0) {
+				if ((s = strchr(scratch, '/')) != NULL) {
+					num = atoi(s + 1);
+				}
+			}
+
+		} else if ((strstarts(dstr, "/dev/sound/") == 0) &&
+		    (isdigit(bn[0]))) {
+			num = atoi(bn);
+		}
+		if (num >= 0) {
+			/* Convert SADA name to OSS name */
+			(void) snprintf(scratch, sizeof (scratch),
+			    "/dev/mixer%d", num);
+			dstr = strdup(scratch);
 		} else {
-			sep++;
+			dstr = strdup(dstr);
 		}
-		(void) strcat(buf, gettext(UNKNOWN));
 	}
 
-	if (sep == 0) {
-		(void) strcat(buf, gettext(NONE));
+	if (wstr != NULL && cstr == NULL) {
+		warn(_("Control value specified without name\n"));
+		rv = EINVAL;
+		goto OUT;
 	}
 
-	(void) strcat(buf, gettext(")"));
+	if ((cstr == NULL) && (Copt == 0) && (eopt == 0) &&
+	    (ropt == 0) && (sopt == 0)) {
+		iopt = 1;
+	}
+
+	rv = mixers_getinfo(dstr, &mlistp);
+	if (rv != 0)
+		goto OUT;
+
+	if (wstr != NULL) {
+		rv = mixers_set(mlistp, cstr, wstr, vopt);
+		goto OUT;
+	}
+
+	if (ropt) {
+		rv = mixers_restore(mlistp, sstr, vopt);
+		goto OUT;
+	}
+
+	if (sopt) {
+		rv = mixers_save(mlistp, sstr, ovopt, vopt);
+		goto OUT;
+	}
+
+	rv = mixers_sav_prt(stdout, 0, mlistp, cstr, eopt, iopt, vopt);
+	if (rv != 0)
+		goto OUT;
+
+	rv = 0;
+
+OUT:
+	if (dstr != NULL)
+		free(dstr);
+	if (cstr != NULL)
+		free(cstr);
+	if (wstr != NULL)
+		free(wstr);
+	if (sstr != NULL)
+		free(sstr);
 
-}	/* rport2str() */
+	mixers_free(&mlistp);
+	return (rv);
+}
diff --git a/usr/src/cmd/devfsadm/audio_link.c b/usr/src/cmd/devfsadm/audio_link.c
index 6b3e08a66f..9552b0aa4e 100644
--- a/usr/src/cmd/devfsadm/audio_link.c
+++ b/usr/src/cmd/devfsadm/audio_link.c
@@ -19,19 +19,22 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 #include <regex.h>
 #include <devfsadm.h>
 #include <strings.h>
+#include <ctype.h>
 #include <stdlib.h>
 #include <limits.h>
 #include <stdio.h>
+#include <syslog.h>
 #include <bsm/devalloc.h>
+#include <sys/audio.h>
+#include <sys/soundcard.h>
+#include <unistd.h>
 
 #define	MAX_AUDIO_LINK 100
 #define	RE_SIZE 64
@@ -39,13 +42,18 @@
 extern int system_labeled;
 
 static void check_audio_link(char *secondary_link,
-				const char *primary_link_format);
+    const char *primary_link_format);
+
 static int audio_process(di_minor_t minor, di_node_t node);
+static int sndstat_process(di_minor_t minor, di_node_t node);
 
 static devfsadm_create_t audio_cbt[] = {
 	{ "audio", "ddi_audio", NULL,
-	TYPE_EXACT, ILEVEL_0, audio_process
-	}
+	    TYPE_EXACT, ILEVEL_0, audio_process
+	},
+	{ "pseudo", "ddi_pseudo", "audio",
+	    TYPE_EXACT|DRV_EXACT, ILEVEL_0, sndstat_process
+	},
 };
 
 DEVFSADM_CREATE_INIT_V0(audio_cbt);
@@ -55,54 +63,49 @@ DEVFSADM_CREATE_INIT_V0(audio_cbt);
  * because recurse_dev_re() would not work.
  */
 static devfsadm_remove_t audio_remove_cbt[] = {
-	{ "audio",
-	"^audio(ctl)?$",
-	RM_POST|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_link
+	/*
+	 * Secondary links.
+	 */
+
+	/* /dev/audio, /dev/audioctl, /dev/dsp */
+	{ "audio", "^(audio|audioctl|dsp)$",
+	    RM_POST|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_link
 	},
-	{ "audio",
-	"^sound/[0-9]+.*$",
-	RM_PRE|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_all
+	/* /dev/mixer0, /dev/dsp0 */
+	{ "audio", "^(mixer|dsp)[0-9]+$",
+	    RM_POST|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_link
 	},
-	{ "audio",
-	"^isdn/[0-9]+/mgt$",
-	RM_PRE, ILEVEL_0, devfsadm_rm_all
+	/* /dev/sound/0, 0ctl */
+	{ "audio", "^sound/[0-9]+(ctl)?$",
+	    RM_POST|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_link
 	},
-	{ "audio",
-	"^isdn/[0-9]+/aux/0(ctl)?$",
-	RM_PRE, ILEVEL_0, devfsadm_rm_all
+	/* /dev/mixer */
+	{ "pseudo", "^(mixer)$",
+	    RM_POST|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_link
 	},
-	{ "audio",
-	"^isdn/[0-9]+/(nt)|(te)/((dtrace)|(mgt)|(b1)|(b2)|(d))$",
-	RM_PRE, ILEVEL_0, devfsadm_rm_all
-	}
-};
 
-DEVFSADM_REMOVE_INIT_V0(audio_remove_cbt);
-
-static regex_t isdn_re;
+	/*
+	 * Primary links.
+	 */
 
-#define	ISDN_RE "((nt)|(te)|(aux))\\,((0)|(0ctl)|(d)|(b1)|(b2)|(mgt)|(dtrace))"
-#define	F 1
-#define	S 5
+	/* /dev/sndstat */
+	{ "pseudo", "^sndstat$",
+	    RM_PRE|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_all
+	},
+	/* /dev/sound/audio810:0, 0ctl, etc */
+	{ "audio", "^sound/.*:[0-9]+(ctl|dsp|mixer)?$",
+	    RM_PRE|RM_HOT|RM_ALWAYS, ILEVEL_0, devfsadm_rm_all
+	},
+};
 
-int
-minor_init(void)
-{
-	if (0 != regcomp(&isdn_re, ISDN_RE, REG_EXTENDED)) {
-		devfsadm_errprint("SUNW_audio_link: minor_init: regular "
-		    "expression bad: '%s'\n", ISDN_RE);
-		return (DEVFSADM_FAILURE);
-	} else {
-		return (DEVFSADM_SUCCESS);
-	}
-}
+DEVFSADM_REMOVE_INIT_V0(audio_remove_cbt);
 
 int
 minor_fini(void)
 {
-	regfree(&isdn_re);
 	check_audio_link("audio", "sound/%d");
 	check_audio_link("audioctl", "sound/%dctl");
+	check_audio_link("dsp", "dsp%d");
 	return (DEVFSADM_SUCCESS);
 }
 
@@ -111,6 +114,82 @@ minor_fini(void)
 		    pm[pos].rm_eo - pm[pos].rm_so); \
 		    to[pm[pos].rm_eo - pm[pos].rm_so] = 0;
 
+static void
+send_number(long num)
+{
+	char		buf[PATH_MAX+1];
+
+	/*
+	 * This is not safe with -r.
+	 */
+	if (strcmp(devfsadm_root_path(), "/") != 0)
+		return;
+
+	(void) snprintf(buf, sizeof (buf), "/dev/mixer%ld", num);
+	if (device_exists(buf)) {
+		int	fd;
+
+		if ((fd = open(buf, O_RDWR)) < 0)
+			return;
+
+		(void) ioctl(fd, SNDCTL_SUN_SEND_NUMBER, &num);
+		(void) close(fd);
+		devfsadm_print(CHATTY_MID,
+		    "sent devnum audio %ld to %s\n", num, buf);
+	}
+}
+
+static void
+send_all(void)
+{
+	/*
+	 * This module traverses /dev/sound to look for nodes that
+	 * have been previously created.
+	 */
+	finddevhdl_t	fh;
+	const char	*path;
+
+	/*
+	 * This is not safe with -r.
+	 */
+	if ((strcmp(devfsadm_root_path(), "/") == 0) &&
+	    (finddev_readdir("/dev", &fh) == 0)) {
+		static int	doneit = 0;
+
+		while ((!doneit) && ((path = finddev_next(fh)) != NULL)) {
+			long num;
+			char *ep;
+			if ((strncmp(path, "mixer", 5) != 0) ||
+			    (!isdigit(path[5]))) {
+				continue;
+			}
+			num = strtol(path + 5, &ep, 10);
+			send_number(num);
+		}
+		finddev_close(fh);
+		doneit = 1;
+	}
+}
+
+static int
+sndstat_process(di_minor_t minor, di_node_t node)
+{
+	char *mn;
+
+	mn = di_minor_name(minor);
+
+	/*
+	 * "Special" handling for /dev/sndstat and /dev/mixer.
+	 */
+	if (strcmp(mn, "sound,sndstat0") == 0) {
+		(void) devfsadm_mklink("sndstat", node, minor, 0);
+		(void) devfsadm_secondary_link("mixer", "sndstat", 0);
+		send_all();
+	}
+
+	return (DEVFSADM_CONTINUE);
+}
+
 /*
  * This function is called for every audio node.
  * Calls enumerate to assign a logical unit id, and then
@@ -120,113 +199,167 @@ static int
 audio_process(di_minor_t minor, di_node_t node)
 {
 	int flags = 0;
-	char path[PATH_MAX + 1];
+	char devpath[PATH_MAX + 1];
+	char newpath[PATH_MAX + 1];
 	char *buf;
 	char *mn;
-	char m1[10];
-	char m2[10];
-	char *devfspath;
+	char *tmp;
+	char *ep;
 	char re_string[RE_SIZE+1];
 	devfsadm_enumerate_t rules[1] = {NULL};
-	regmatch_t pmatch[12];
-	char *au_mn;
 
+	if (system_labeled)
+		flags = DA_ADD|DA_AUDIO;
 
 	mn = di_minor_name(minor);
 
-	if ((devfspath = di_devfs_path(node)) == NULL) {
+	if ((tmp = di_devfs_path(node)) == NULL) {
 		return (DEVFSADM_CONTINUE);
 	}
-	(void) strcpy(path, devfspath);
-	(void) strcat(path, ":");
-	(void) strcat(path, mn);
-	di_devfs_path_free(devfspath);
-
-	if (strstr(mn, "sound,") != NULL) {
-		(void) snprintf(re_string, RE_SIZE, "%s", "^sound$/^([0-9]+)");
-	} else {
-		(void) strcpy(re_string, "isdn/([0-9]+)");
-	}
+	(void) snprintf(devpath, sizeof (devpath), "%s:%s", tmp, mn);
+	di_devfs_path_free(tmp);
 
 	/*
-	 * We want a match against the physical path
-	 * without the minor name component.
+	 * "Special" handling for /dev/sndstat and /dev/mixer.
 	 */
-	rules[0].re = re_string;
-	rules[0].subexp = 1;
-	rules[0].flags = MATCH_ADDR;
+	if (strcmp(mn, "sound,sndstat0") == 0) {
+		(void) strlcpy(newpath, "sound/sndstat", sizeof (newpath));
 
-	/*
-	 * enumerate finds the logical audio id, and stuffs
-	 * it in buf
-	 */
-	if (devfsadm_enumerate_int(path, 0, &buf, rules, 1)) {
+		(void) devfsadm_mklink(newpath, node, minor, flags);
+		(void) devfsadm_secondary_link("sndstat", newpath, flags);
+		(void) devfsadm_secondary_link("mixer", newpath, flags);
+		send_all();
 		return (DEVFSADM_CONTINUE);
 	}
 
-	path[0] = '\0';
+	if (strncmp(mn, "sound,", 6) == 0) {
+		char base[PATH_MAX + 1];
+		char linksrc[PATH_MAX + 1];
+		char linkdst[PATH_MAX + 1];
+		long num;
+		long inst;
+		int i;
+		char *driver;
+
+		/* strlen("sound,") */
+		(void) strlcpy(base, mn + 6, sizeof (base));
+		mn = base;
+
+		driver = di_driver_name(node);
+
+		/* if driver name override in minor name */
+		if ((tmp = strchr(mn, ',')) != NULL) {
+			driver = mn;
+			*tmp = '\0';
+			mn = tmp + 1;
+		}
 
-	if (strstr(mn, "sound,") != NULL) {
-		(void) strcpy(path, "sound/");
-		(void) strcat(path, buf);
+		/* skip past "audio" portion of the minor name */
+		if (strncmp(mn, "audio", 5) == 0) {
+			mn += 5;
+		}
 
-		/* if this is a minor node, tack on the correct suffix */
-		au_mn = strchr(mn, ',');
-		if (strcmp(++au_mn, "audio") != 0) {
+		/* parse the instance number */
+		for (i = strlen(mn); i; i--) {
+			if (!isdigit(mn[i - 1]))
+				break;
+		}
+		inst = strtol(mn + i, &ep, 10);
+		mn[i] = 0;	/* lop off the instance number */
+
+		/*
+		 * First we create a node with the driver under /dev/sound.
+		 * Note that "instance numbers" used by the audio framework
+		 * are guaranteed to be unique for each driver.
+		 */
+		(void) snprintf(newpath, sizeof (newpath), "sound/%s:%d%s",
+		    driver, inst, mn);
+		(void) devfsadm_mklink(newpath, node, minor, flags);
+
+		/*
+		 * The rest of this logic is a gross simplification that
+		 * is made possible by the fact that each audio node will
+		 * have several different minors associated with it.  Rather
+		 * processing each node separately, we just create the links
+		 * all at once.
+		 *
+		 * This reduces the chances of the various links being out
+		 * of sync with each other.
+		 */
+		if (strcmp(mn, "mixer") != 0) {
+			return (DEVFSADM_CONTINUE);
+		}
 
-			/*
-			 * audioctl is a special case. It is handled
-			 * by stripping off the audio from the node name
-			 */
-			if (strcmp(au_mn, "audioctl") == 0)
-				au_mn = strstr(au_mn, "ctl");
-			(void) strcat(path, au_mn);
+		/*
+		 * Its the control node, so create the various
+		 * secondary links.
+		 */
+
+		/*
+		 * We want a match against the physical path
+		 * without the minor name component.
+		 */
+		(void) snprintf(re_string, RE_SIZE, "%s", "^mixer([0-9]+)");
+		rules[0].re = re_string;
+		rules[0].subexp = 1;
+		rules[0].flags = MATCH_ALL;
+
+		/*
+		 * enumerate finds the logical audio id, and stuffs
+		 * it in buf
+		 */
+		(void) strlcpy(devpath, newpath, sizeof (devpath));
+		if (devfsadm_enumerate_int(devpath, 0, &buf, rules, 1)) {
+			return (DEVFSADM_CONTINUE);
 		}
-	}
+		num = strtol(buf, &ep, 10);
+		free(buf);
 
-	if (regexec(&isdn_re, mn, sizeof (pmatch) / sizeof (pmatch[0]),
-	    pmatch, 0) == 0) {
-		COPYSUB(m1, mn, pmatch, F);
-		COPYSUB(m2, mn, pmatch, S);
-		(void) strcpy(path, "isdn/");
-		(void) strcat(path, buf);
-		(void) strcat(path, "/");
-		(void) strcat(path, m1);
-		(void) strcat(path, "/");
-		(void) strcat(path, m2);
-	}
+		/* /dev/sound/0 */
+		(void) snprintf(linksrc, sizeof (linksrc), "sound/%s:%ld",
+		    driver, inst);
+		(void) snprintf(linkdst, sizeof (linkdst), "sound/%ld", num);
+		(void) devfsadm_secondary_link(linkdst, linksrc, flags);
 
-	if (strstr("mgt,mgt", mn) != NULL) {
-		(void) strcpy(path, "isdn/");
-		(void) strcat(path, buf);
-		(void) strcat(path, "/mgt");
-	}
+		(void) snprintf(linksrc, sizeof (linksrc), "sound/%s:%ldctl",
+		    driver, inst);
+		(void) snprintf(linkdst, sizeof (linkdst), "sound/%ldctl", num);
+		(void) devfsadm_secondary_link(linkdst, linksrc, flags);
+
+		(void) snprintf(linksrc, sizeof (linksrc), "sound/%s:%lddsp",
+		    driver, inst);
+		(void) snprintf(linkdst, sizeof (linkdst), "dsp%ld", num);
+		(void) devfsadm_secondary_link(linkdst, linksrc, flags);
 
-	free(buf);
+		(void) snprintf(linksrc, sizeof (linksrc), "sound/%s:%ldmixer",
+		    driver, inst);
+		(void) snprintf(linkdst, sizeof (linkdst), "mixer%ld", num);
+		(void) devfsadm_secondary_link(linkdst, linksrc, flags);
+
+		/* Send control number */
+		send_number(num);
 
-	if (path[0] == '\0') {
-		devfsadm_errprint("SUNW_audio_link: audio_process: can't find"
-		    " match for'%s'\n", mn);
 		return (DEVFSADM_CONTINUE);
 	}
 
-	if (system_labeled)
-		flags = DA_ADD|DA_AUDIO;
+	if (strncmp(mn, "internal,", 9) == 0) {
+		/* we don't set up internal nodes in /dev */
+		return (DEVFSADM_CONTINUE);
+	}
 
-	(void) devfsadm_mklink(path, node, minor, flags);
+	devfsadm_errprint("SUNW_audio_link: can't find match for'%s'\n", mn);
 	return (DEVFSADM_CONTINUE);
 }
 
-
 static void
-check_audio_link(char *secondary_link, const char *primary_link_format)
+check_audio_link(char *secondary, const char *primary_format)
 {
-	char primary_link[PATH_MAX + 1];
+	char primary[PATH_MAX + 1];
 	int i;
 	int flags = 0;
 
 	/* if link is present, return */
-	if (devfsadm_link_valid(secondary_link) == DEVFSADM_TRUE) {
+	if (devfsadm_link_valid(secondary) == DEVFSADM_TRUE) {
 		return;
 	}
 
@@ -234,10 +367,11 @@ check_audio_link(char *secondary_link, const char *primary_link_format)
 		flags = DA_ADD|DA_AUDIO;
 
 	for (i = 0; i < MAX_AUDIO_LINK; i++) {
-		(void) sprintf(primary_link, primary_link_format, i);
-		if (devfsadm_link_valid(primary_link) == DEVFSADM_TRUE) {
-			(void) devfsadm_secondary_link(secondary_link,
-			    primary_link, flags);
+		(void) sprintf(primary, primary_format, i);
+		if (devfsadm_link_valid(primary) == DEVFSADM_TRUE) {
+			/* we read link to get it to the master "real" link */
+			(void) devfsadm_secondary_link(secondary,
+			    primary, flags);
 			break;
 		}
 	}
diff --git a/usr/src/cmd/mdb/Makefile.common b/usr/src/cmd/mdb/Makefile.common
index bb341fdc8f..109b110645 100644
--- a/usr/src/cmd/mdb/Makefile.common
+++ b/usr/src/cmd/mdb/Makefile.common
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # MDB modules used for debugging user processes that every ISA's build
@@ -52,7 +52,6 @@ COMMON_MODULES_PROC_32BIT = \
 #
 COMMON_MODULES_KVM = \
 	arp \
-	audiosup \
 	cpc \
 	crypto	\
 	dtrace	\
diff --git a/usr/src/cmd/mdb/common/modules/audiosup/audiosup.c b/usr/src/cmd/mdb/common/modules/audiosup/audiosup.c
deleted file mode 100644
index ab98383de7..0000000000
--- a/usr/src/cmd/mdb/common/modules/audiosup/audiosup.c
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#include <mdb/mdb_modapi.h>
-#include <sys/types.h>
-#include <sys/audio/audio_trace.h>
-
-typedef struct audiotrace_walk_data {
-	uintptr_t aw_bufstart;
-	uintptr_t aw_bufend;
-	int aw_cnt;
-} audiotrace_walk_data_t;
-
-int
-audiotrace_walk_init(mdb_walk_state_t *wsp)
-{
-	size_t size;
-	uint_t seq, pos;
-	GElf_Sym sym;
-	audio_trace_buf_t *buffer;
-	audio_trace_buf_t atb[2];
-	audiotrace_walk_data_t *aw;
-
-	if (mdb_readvar(&size, "audio_tb_siz") == -1) {
-		mdb_warn("failed to read 'audio_tb_siz'");
-		return (WALK_ERR);
-	}
-
-	if (size == 0)
-		return (WALK_DONE);
-
-	if (mdb_lookup_by_name("audio_trace_buffer", &sym) == -1) {
-		mdb_warn("failed to find 'audio_trace_buffer'");
-		return (WALK_ERR);
-	}
-
-	buffer = (audio_trace_buf_t *)(uintptr_t)sym.st_value;
-
-	if (mdb_readvar(&seq, "audio_tb_seq") == -1) {
-		mdb_warn("failed to read 'audio_tb_seq'");
-		return (WALK_ERR);
-	}
-
-	if (mdb_readvar(&pos, "audio_tb_pos") == -1) {
-		mdb_warn("failed to read 'audio_tb_pos'");
-		return (WALK_ERR);
-	}
-
-	/* by default start the walk with the first buffer entry */
-	wsp->walk_addr = (uintptr_t)buffer;
-
-	if (seq > size && pos > 0) {
-		/* start the walk with the oldest buffer entry */
-
-		if (mdb_vread(&atb, sizeof (atb),
-		    (uintptr_t)(buffer + pos - 1)) == -1) {
-			mdb_warn("failed to read audio_trace_buf_t at %p",
-			    buffer + pos - 1);
-			return (WALK_ERR);
-		}
-
-		if (atb[1].atb_seq < atb[0].atb_seq)
-			wsp->walk_addr = (uintptr_t)(buffer + pos);
-	}
-
-	aw = mdb_alloc(sizeof (audiotrace_walk_data_t), UM_SLEEP);
-	aw->aw_bufstart = (uintptr_t)buffer;
-	aw->aw_bufend = (uintptr_t)(buffer + size);
-	aw->aw_cnt = MIN(seq, size);
-	wsp->walk_data = aw;
-
-	return (WALK_NEXT);
-}
-
-int
-audiotrace_walk_step(mdb_walk_state_t *wsp)
-{
-	uintptr_t addr = wsp->walk_addr;
-	audiotrace_walk_data_t *aw = wsp->walk_data;
-	audio_trace_buf_t atb;
-
-	if (aw->aw_cnt == 0)
-		return (WALK_DONE);
-
-	aw->aw_cnt--;
-	wsp->walk_addr += sizeof (audio_trace_buf_t);
-
-	if (wsp->walk_addr >= aw->aw_bufend)
-		wsp->walk_addr = aw->aw_bufstart;
-
-	if (mdb_vread(&atb, sizeof (atb), addr) == -1) {
-		mdb_warn("failed to read audio_trace_buf_t at %p", addr);
-		return (WALK_DONE);
-	}
-
-	return (wsp->walk_callback(addr, &atb, wsp->walk_cbdata));
-}
-
-void
-audiotrace_walk_fini(mdb_walk_state_t *wsp)
-{
-	mdb_free(wsp->walk_data, sizeof (audiotrace_walk_data_t));
-}
-
-
-/*ARGSUSED*/
-int
-audiotrace(uintptr_t addr, uint_t flags, int argc, const mdb_arg_t *argv)
-{
-	int count;
-	uint_t seq;
-	const mdb_arg_t *argp = &argv[0];
-	audio_trace_buf_t atb;
-	char str[256];
-
-	if (!(flags & DCMD_ADDRSPEC)) {
-		if (mdb_walk_dcmd("audiotrace", "audiotrace", argc, argv)
-		    == -1) {
-			mdb_warn("can't walk audio trace buffer entries");
-			return (DCMD_ERR);
-		}
-		return (DCMD_OK);
-	}
-
-	if (argc > 1)
-		return (DCMD_USAGE);
-
-	/*
-	 * Specifying an address and a count at the same time doesn't make
-	 * sense.
-	 */
-	if (argc == 1 && !(flags & DCMD_LOOP))
-		return (DCMD_USAGE);
-
-	if (DCMD_HDRSPEC(flags))
-		mdb_printf("%<u>%6s %?s %-40s%</u>\n",
-		    "SEQNO", "DATA", "COMMENT");
-
-	if (mdb_vread(&atb, sizeof (atb), addr) == -1) {
-		mdb_warn("failed to read audiotrace_t at %p", addr);
-		return (DCMD_ERR);
-	}
-
-	if (argc == 1) {
-		/*
-		 * Display the last 'count' entries only. Skip any other
-		 * entries.
-		 */
-		if (mdb_readvar(&seq, "audio_tb_seq") == -1) {
-			mdb_warn("failed to read 'audio_tb_seq'");
-			return (DCMD_ERR);
-		}
-
-		if (argp->a_type == MDB_TYPE_IMMEDIATE)
-			count = argp->a_un.a_val;
-		else
-			count = mdb_strtoull(argp->a_un.a_str);
-
-		if (atb.atb_seq < seq - count)
-			return (DCMD_OK);
-	}
-
-	if (mdb_readstr(str, sizeof (str), (uintptr_t)atb.atb_comment) == -1) {
-		mdb_warn("failed to read string at %p", atb.atb_comment);
-		return (DCMD_ERR);
-	}
-
-	mdb_printf("%6d %?x %s\n", atb.atb_seq, atb.atb_data, str);
-
-	return (DCMD_OK);
-}
-
-void
-audiotrace_help(void)
-{
-	mdb_printf(
-	    "If count is specified ::audiotrace prints the specified\n"
-	    "number of entries from the logical end of the trace buffer.\n");
-}
-
-/*
- * MDB module linkage
- */
-static const mdb_dcmd_t dcmds[] = {
-	{ "audiotrace", "?[count]", "display audio trace buffer entries",
-	    audiotrace, audiotrace_help },
-	{ NULL }
-};
-
-static const mdb_walker_t walkers[] = {
-	{ "audiotrace", "walk audio trace buffer entries", audiotrace_walk_init,
-	    audiotrace_walk_step, audiotrace_walk_fini },
-	{ NULL }
-};
-
-static const mdb_modinfo_t modinfo = { MDB_API_VERSION, dcmds, walkers };
-
-const mdb_modinfo_t *
-_mdb_init(void)
-{
-	return (&modinfo);
-}
diff --git a/usr/src/cmd/svc/milestone/Makefile b/usr/src/cmd/svc/milestone/Makefile
index df0e542f76..d727a17ebb 100644
--- a/usr/src/cmd/svc/milestone/Makefile
+++ b/usr/src/cmd/svc/milestone/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -60,7 +60,8 @@ MAINMILESTONES= \
 MAINMANIFESTS= $(MAINMILESTONES:%=$(ROOTSVCMILESTONE)/%)
 
 SYSDEVSVCS= \
-	devices-local.xml
+	devices-local.xml \
+	devices-audio.xml
 
 SYSDEVMANIFESTS= $(SYSDEVSVCS:%=$(ROOTSVCSYSTEMDEVICE)/%)
 
@@ -94,6 +95,7 @@ MANIFEST= $(FSSVCS) $(NETSVCS) $(MAINMILESTONES) $(SYSTEMSVCS) \
 SVCMETHOD=\
 	boot-archive \
 	console-login \
+	devices-audio \
 	devices-local \
 	fs-local \
 	fs-minimal \
diff --git a/usr/src/cmd/svc/milestone/devices-audio b/usr/src/cmd/svc/milestone/devices-audio
new file mode 100644
index 0000000000..74c2457588
--- /dev/null
+++ b/usr/src/cmd/svc/milestone/devices-audio
@@ -0,0 +1,44 @@
+#!/sbin/sh
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+
+. /lib/svc/share/smf_include.sh
+
+smf_is_globalzone || exit $SMF_EXIT_OK
+
+#
+# In order to populate the initial startup mapping of devfs links to
+# the audio subsystem (necessary to properly report numbers in
+# /dev/sndstat and in the OSS sysinfo ioctls -- thank you Linux!) we
+# need to run some code at boot.  We key this off of the handler in
+# devfsadm for the audio node(s).
+#
+# Additionally, we need to make sure the streams audio module is
+# loaded so that the symbolic links are properly created for STREAMs
+# devices and point to a reasonable device node.
+#
+/usr/sbin/devfsadm -i audio
+/usr/sbin/devfsadm -i austr
+exit $SMF_EXIT_OK
diff --git a/usr/src/cmd/svc/milestone/devices-audio.xml b/usr/src/cmd/svc/milestone/devices-audio.xml
new file mode 100644
index 0000000000..48dee7299e
--- /dev/null
+++ b/usr/src/cmd/svc/milestone/devices-audio.xml
@@ -0,0 +1,91 @@
+<?xml version='1.0'?>
+<!DOCTYPE service_bundle SYSTEM '/usr/share/lib/xml/dtd/service_bundle.dtd.1'>
+<!--
+ Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ Use is subject to license terms.
+
+ CDDL HEADER START
+
+ The contents of this file are subject to the terms of the
+ Common Development and Distribution License (the "License").
+ You may not use this file except in compliance with the License.
+
+ You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ or http://www.opensolaris.org/os/licensing.
+ See the License for the specific language governing permissions
+ and limitations under the License.
+
+ When distributing Covered Code, include this CDDL HEADER in each
+ file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ If applicable, add the following below this CDDL HEADER, with the
+ fields enclosed by brackets "[]" replaced with your own identifying
+ information: Portions Copyright [yyyy] [name of copyright owner]
+
+ CDDL HEADER END
+
+	NOTE:  This service manifest is not editable; its contents will
+	be overwritten by package or patch operations, including
+	operating system upgrade.  Make customizations in a different
+	file.
+
+	Service manifest for audio.
+-->
+
+<service_bundle type='manifest' name='SUNWcsr:devices-audio'>
+
+<service
+	name='system/device/audio'
+	type='service'
+	version='1'>
+
+	<create_default_instance enabled='true' />
+
+	<single_instance/>
+
+	<!--
+	  We don't really ndeed devices, just /usr.  However this
+	  avoids thrashing in devfsadm.
+	-->
+	<dependency
+		name='devices'
+		grouping='require_all'
+		restart_on='none'
+		type='service'>
+		<service_fmri value='svc:/milestone/devices' />
+	</dependency>
+
+	<dependent
+		name='devices_audio_multi-user'
+		grouping='require_all'
+		restart_on='none'>
+		<service_fmri value='svc:/milestone/multi-user' />
+	</dependent>
+
+	<exec_method
+		type='method'
+		name='start'
+		exec='/lib/svc/method/devices-audio'
+		timeout_seconds='30' />
+
+	<exec_method
+		type='method'
+		name='stop'
+		exec=':true'
+		timeout_seconds='1' />
+
+	<property_group name='startd' type='framework'>
+		<propval name='duration' type='astring' value='transient' />
+	</property_group>
+
+	<stability value='Evolving' />
+
+	<template>
+		<common_name>
+			<loctext xml:lang='C'>
+			  Solaris audio device configuration.
+			</loctext>
+		</common_name>
+	</template>
+</service>
+
+</service_bundle>
diff --git a/usr/src/pkgdefs/SUNWad810/Makefile b/usr/src/pkgdefs/SUNWad810/Makefile
index 057f909322..9811a569cb 100644
--- a/usr/src/pkgdefs/SUNWad810/Makefile
+++ b/usr/src/pkgdefs/SUNWad810/Makefile
@@ -19,19 +19,19 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 
 include ../Makefile.com
 
+TMPLFILES += postinstall preremove
 DATAFILES += i.preserve
 
 .KEEP_STATE:
 
-all: $(FILES) depend postinstall preremove
+all: $(FILES) depend
 install: all pkg
 
 include ../Makefile.targ
+include ../Makefile.prtarg
diff --git a/usr/src/pkgdefs/SUNWad810/postinstall b/usr/src/pkgdefs/SUNWad810/postinstall.tmpl
index 27f0fcd2e2..acee80b819 100644
--- a/usr/src/pkgdefs/SUNWad810/postinstall
+++ b/usr/src/pkgdefs/SUNWad810/postinstall.tmpl
@@ -20,52 +20,31 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # SUNWad810 postinstall script
 
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
-
-not_installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	return $?
-}
-
-
-#
-# Audio810(7d) driver had been released before solaris had
-# update_drv(1M) command so that we have to manually add
-# aliases via echo(1).
-#
-update_driver() {
-	driver=$1
-	aliases=$2
-	for alias in ${aliases}
-	do
-		egrep "^${driver}[ 	]+${alias}" ${BASEDIR}/etc/driver_aliases>/dev/null 2>&1
-		if [ $? -ne 0 ]
-		then
-			 echo "${driver} ${alias}" >> ${BASEDIR}/etc/driver_aliases
-		fi
-	done
-}
-
-EXIT=0
+include drv_utils
 
 I810_ALIASES="\
 	\"pci1022,7445\" \
 	\"pci1022,746d\" \
+	\"pci1039,7012\" \
+	\"pci8086,2415\" \
+	\"pci8086,2425\" \
+	\"pci8086,2445\" \
 	\"pci8086,2485\" \
 	\"pci8086,24c5\" \
 	\"pci8086,24d5\" \
 	\"pci8086,266e\" \
+	\"pci8086,25a6\" \
+	\"pci8086,2698\" \
 	\"pci8086,27de\" \
+	\"pci8086,7195\" \
+	\"pci10de,1b1\" \
 	\"pci10de,6a\" \
+	\"pci10de,8a\" \
 	\"pci10de,da\" \
 	\"pci10de,ea\" \
 	\"pci10de,59\" \
@@ -73,16 +52,4 @@ I810_ALIASES="\
 	\"pci10de,26b\" \
 	"
 	
-case "${ARCH}" in
-	i386)
-		not_installed audio810
-		if [ $? -ne 0 ] 
-		then 
-			add_drv -b "${BASEDIR}" -i "${I810_ALIASES}" -n audio810 || EXIT=1
-		else
-			update_driver  audio810 "${I810_ALIASES}" || EXIT=1
-		fi
-	;;
-esac
-
-exit ${EXIT}
+pkg_drvadd -i "${I810_ALIASES}" audio810 || exit 1
diff --git a/usr/src/pkgdefs/SUNWad810/preremove b/usr/src/pkgdefs/SUNWad810/preremove
deleted file mode 100644
index 6a075b1a8c..0000000000
--- a/usr/src/pkgdefs/SUNWad810/preremove
+++ /dev/null
@@ -1,52 +0,0 @@
-#! /bin/sh
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# SUNWad810 preremove script
-
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
-
-EXIT=0
-
-installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	if [ "$?" -eq 0 ]; then
-		return 1
-	else
-		return 0
-	fi
-}
-
-case "${ARCH}" in
-	i386)
-		installed audio810 || rem_drv -b ${BASEDIR} audio810 || EXIT=1
-	;;
-esac
-
-exit ${EXIT}
diff --git a/usr/src/cmd/mdb/intel/ia32/audiosup/Makefile b/usr/src/pkgdefs/SUNWad810/preremove.tmpl
index 6e7a229eaa..4b1e20cc4e 100644
--- a/usr/src/cmd/mdb/intel/ia32/audiosup/Makefile
+++ b/usr/src/pkgdefs/SUNWad810/preremove.tmpl
@@ -1,10 +1,10 @@
+#! /bin/sh
 #
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,16 +20,11 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
+# SUNWad810 preremove script
 
-MODULE = audiosup.so
-MDBTGT = kvm
+include drv_utils
 
-MODSRCS = audiosup.c
-
-include ../../../../Makefile.cmd
-include ../../Makefile.ia32
-include ../../../Makefile.module
+pkg_drvrem audio810 || exit 1
diff --git a/usr/src/pkgdefs/SUNWadixp/Makefile b/usr/src/pkgdefs/SUNWadixp/Makefile
index 10ffbabe4d..c131ff1b31 100644
--- a/usr/src/pkgdefs/SUNWadixp/Makefile
+++ b/usr/src/pkgdefs/SUNWadixp/Makefile
@@ -18,20 +18,19 @@
 #
 # CDDL HEADER END
 #
-
 #
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 
 include ../Makefile.com
 
+TMPLFILES += postinstall preremove
 
 .KEEP_STATE:
 
-all: $(FILES) depend postinstall preremove
+all: $(FILES) depend
 install: all pkg
 
 include ../Makefile.targ
+include ../Makefile.prtarg
diff --git a/usr/src/pkgdefs/SUNWadixp/postinstall b/usr/src/pkgdefs/SUNWadixp/postinstall.tmpl
index 5e82d6ebfd..018ca2b1a7 100644
--- a/usr/src/pkgdefs/SUNWadixp/postinstall
+++ b/usr/src/pkgdefs/SUNWadixp/postinstall.tmpl
@@ -21,33 +21,21 @@
 #
 
 #
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # SUNWadixp postinstall script
 
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
-
-not_installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	return $?
-}
-
-EXIT=0
-
-case "${ARCH}" in
-	i386)
-		not_installed audioixp|| \
-		add_drv -b "${BASEDIR}" \
-			-i '"pci1002,4370" "pci1002,4341"' \
-			-n audioixp || \
-		EXIT=1
+include drv_utils
 
-	;;
-esac
-
-exit ${EXIT}
+#
+# We could probably also do SB600, pci1002,4382, but we don't have any
+# hardware to confirm with.
+#
+IXP_ALIASES="\
+	\"pci1002,4370\" \
+	\"pci1002,4361\" \
+	\"pci1002,4341\" \
+	"
+	
+pkg_drvadd -i "${IXP_ALIASES}" audioixp || exit 1
diff --git a/usr/src/cmd/mdb/intel/amd64/audiosup/Makefile b/usr/src/pkgdefs/SUNWadixp/preremove.tmpl
index d31cbe9363..d5e761cbad 100644
--- a/usr/src/cmd/mdb/intel/amd64/audiosup/Makefile
+++ b/usr/src/pkgdefs/SUNWadixp/preremove.tmpl
@@ -1,10 +1,10 @@
+#! /bin/sh
 #
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,17 +20,11 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
+# SUNWadixp preremove script
 
-MODULE = audiosup.so
-MDBTGT = kvm
+include drv_utils
 
-MODSRCS = audiosup.c
-
-include ../../../../Makefile.cmd
-include ../../../../Makefile.cmd.64
-include ../../Makefile.amd64
-include ../../../Makefile.module
+pkg_drvrem audioixp || exit 1
diff --git a/usr/src/pkgdefs/SUNWauda/prototype_com b/usr/src/pkgdefs/SUNWauda/prototype_com
index 582e28748c..0006160282 100644
--- a/usr/src/pkgdefs/SUNWauda/prototype_com
+++ b/usr/src/pkgdefs/SUNWauda/prototype_com
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,10 +19,8 @@
 # CDDL HEADER END
 #
 #
-# Copyright (c) 1999-2001 by Sun Microsystems, Inc.
-# All rights reserved.
-#
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
 #
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
@@ -49,5 +46,6 @@ d none usr/bin 755 root bin
 f none usr/bin/audioplay 555 root bin
 f none usr/bin/audiorecord 555 root bin
 f none usr/bin/audioconvert 555 root bin
+f none usr/bin/audiotest 555 root bin
 d none usr/sbin 755 root bin
 f none usr/sbin/mixerctl 555 root bin
diff --git a/usr/src/pkgdefs/SUNWaudd/Makefile b/usr/src/pkgdefs/SUNWaudd/Makefile
index 2f359c8e57..c6aee2f3cc 100644
--- a/usr/src/pkgdefs/SUNWaudd/Makefile
+++ b/usr/src/pkgdefs/SUNWaudd/Makefile
@@ -19,16 +19,14 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 
 include ../Makefile.com
 
+TMPLFILES += postinstall preremove
 DATAFILES += depend
-LICENSEFILES += $(OSBL)
 
 .KEEP_STATE:
 
@@ -36,3 +34,4 @@ all: $(FILES)
 install: all pkg
 
 include ../Makefile.targ
+include ../Makefile.prtarg
diff --git a/usr/src/pkgdefs/SUNWaudd/pkginfo.tmpl b/usr/src/pkgdefs/SUNWaudd/pkginfo.tmpl
index f265f0f084..b7d9bb4c46 100644
--- a/usr/src/pkgdefs/SUNWaudd/pkginfo.tmpl
+++ b/usr/src/pkgdefs/SUNWaudd/pkginfo.tmpl
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 
 #
 # This required package information file describes characteristics of the
@@ -36,7 +34,7 @@ NAME="Audio Drivers"
 ARCH="ISA"
 VERSION="ONVERS,REV=0.0.0"
 CATEGORY="system"
-DESC="SunOS audio device drivers using the new audio driver architecture"
+DESC="SunOS audio device drivers"
 SUNW_PRODNAME="SunOS"
 SUNW_PRODVERS="RELEASE/VERSION"
 SUNW_PKGTYPE="root"
diff --git a/usr/src/pkgdefs/SUNWaudd/postinstall b/usr/src/pkgdefs/SUNWaudd/postinstall
deleted file mode 100644
index 93f2039cdb..0000000000
--- a/usr/src/pkgdefs/SUNWaudd/postinstall
+++ /dev/null
@@ -1,69 +0,0 @@
-#! /bin/sh
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-
-#
-# Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# SUNWaudd postinstall script
-
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
-
-not_installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	return $?
-}
-
-EXIT=0
-
-case "${ARCH}" in
-	sparc)
-		not_installed audiocs || \
-		add_drv -b ${BASEDIR} -i '"SUNW,CS4231"' -n audiocs || EXIT=1
-
-		not_installed audioens || \
-		add_drv -b "${BASEDIR}" \
-			-i '"pci1274,1371" "pci1274,5880"' \
-			-n audioens || \
-		EXIT=1
-
-		not_installed audiots || \
-		add_drv -b ${BASEDIR} -i '"pci10b9,5451"' -n audiots || EXIT=1
-
-		not_installed audio1575 || \
-		add_drv -b ${BASEDIR} -i '"pci10b9,5455"' -n audio1575 || EXIT=1
-	;;
-	i386)
-		not_installed audioens || \
-		add_drv -b "${BASEDIR}" \
-			-i '"pci1274,1371" "pci1274,5880"' \
-			-n audioens || \
-		EXIT=1
-			
-	;;
-esac
-
-exit ${EXIT}
diff --git a/usr/src/pkgdefs/SUNWaudd/preremove b/usr/src/pkgdefs/SUNWaudd/postinstall.tmpl
index f646f7ecfa..1077780acb 100644
--- a/usr/src/pkgdefs/SUNWaudd/preremove
+++ b/usr/src/pkgdefs/SUNWaudd/postinstall.tmpl
@@ -21,37 +21,28 @@
 #
 
 #
-# Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# SUNWaudd preremove script
-
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
+# SUNWaudd postinstall script
 
+include drv_utils
 EXIT=0
 
-installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	if [ "$?" -eq 0 ]; then
-		return 1
-	else
-		return 0
-	fi
-}
+# common drivers first
+pkg_drvadd -n -m '* 0666 root sys' audio || EXIT=1
+pkg_drvadd -n austr || EXIT=1
+pkg_drvadd -n -i '"pci1274,1371" "pci1274,5880"' audioens || EXIT=1
+pkg_drvadd -n -i '"pci1274,5000"' audiopci || EXIT=1
 
 case "${ARCH}" in
-	sparc)
-		installed audiocs || rem_drv -b ${BASEDIR} audiocs || EXIT=1
-		installed audioens || rem_drv -b ${BASEDIR} audioens || EXIT=1
-		installed audiots || rem_drv -b ${BASEDIR} audiots || EXIT=1
-		installed audio1575 || rem_drv -b ${BASEDIR} audio1575 || EXIT=1
+    sparc)
+
+	pkg_drvadd -n -i '"SUNW,CS4231"' audiocs || EXIT=1
+	pkg_drvadd -n -i '"pci10b9,5451"' audiots || EXIT=1
+	pkg_drvadd -n -i '"pci10b9,5455"' audio1575 || EXIT=1
 	;;
-	i386)
-		installed audioens || rem_drv -b ${BASEDIR} audioens || EXIT=1
+    i386)
 	;;
 esac
 
diff --git a/usr/src/pkgdefs/SUNWadixp/preremove b/usr/src/pkgdefs/SUNWaudd/preremove.tmpl
index a81ab22365..670212c5de 100644
--- a/usr/src/pkgdefs/SUNWadixp/preremove
+++ b/usr/src/pkgdefs/SUNWaudd/preremove.tmpl
@@ -21,32 +21,27 @@
 #
 
 #
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# SUNWadixp preremove script
-
-PATH=/usr/bin:/usr/sbin:${PATH}
-export PATH
+# SUNWaudd preremove script
 
+include drv_utils
 EXIT=0
 
-installed() {
-	driver=$1
-	grep "^${driver} " $BASEDIR/etc/name_to_major > /dev/null 2>&1
-	if [ "$?" -eq 0 ]; then
-		return 1
-	else
-		return 0
-	fi
-}
-
 case "${ARCH}" in
-	i386)
-		installed audioixp|| rem_drv -b ${BASEDIR} audioixp || EXIT=1
+    sparc)
+	pkg_drvrem audiocs || EXIT=1
+	pkg_drvrem audiots || EXIT=1
+	pkg_drvrem audio1575 || EXIT=1
+	;;
+    i386)
 	;;
 esac
 
+pkg_drvrem audioens || EXIT=1
+pkg_drvrem audiopci || EXIT=1
+pkg_drvrem austr || EXIT=1
+pkg_drvrem audio || EXIT=1
+
 exit ${EXIT}
diff --git a/usr/src/pkgdefs/SUNWaudd/prototype_com b/usr/src/pkgdefs/SUNWaudd/prototype_com
index 8f3468eb41..628b244651 100644
--- a/usr/src/pkgdefs/SUNWaudd/prototype_com
+++ b/usr/src/pkgdefs/SUNWaudd/prototype_com
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 1999-2003 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -49,4 +46,7 @@ i preremove
 d none kernel 755 root sys
 d none kernel/misc 755 root sys
 d none kernel/drv 755 root sys
+f none kernel/drv/audio.conf 644 root sys
 f none kernel/drv/audioens.conf 644 root sys
+f none kernel/drv/audiopci.conf 644 root sys
+f none kernel/drv/austr.conf 644 root sys
diff --git a/usr/src/pkgdefs/SUNWaudd/prototype_i386 b/usr/src/pkgdefs/SUNWaudd/prototype_i386
index 11b7f9c956..2e8ec83faa 100644
--- a/usr/src/pkgdefs/SUNWaudd/prototype_i386
+++ b/usr/src/pkgdefs/SUNWaudd/prototype_i386
@@ -18,12 +18,11 @@
 #
 # CDDL HEADER END
 #
+
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -47,13 +46,13 @@
 #
 d none kernel/misc/amd64 755 root sys
 d none kernel/drv/amd64 755 root sys
-f none kernel/misc/amsrc1 755 root sys
-f none kernel/misc/amsrc2 755 root sys
-f none kernel/misc/audiosup 755 root sys
-f none kernel/misc/mixer 755 root sys
+f none kernel/misc/ac97 755 root sys
+f none kernel/drv/audio 755 root sys
 f none kernel/drv/audioens 755 root sys
-f none kernel/misc/amd64/amsrc1 755 root sys
-f none kernel/misc/amd64/amsrc2 755 root sys
-f none kernel/misc/amd64/audiosup 755 root sys
-f none kernel/misc/amd64/mixer 755 root sys
+f none kernel/drv/audiopci 755 root sys
+f none kernel/drv/austr 755 root sys
+f none kernel/misc/amd64/ac97 755 root sys
+f none kernel/drv/amd64/audio 755 root sys
 f none kernel/drv/amd64/audioens 755 root sys
+f none kernel/drv/amd64/audiopci 755 root sys
+f none kernel/drv/amd64/austr 755 root sys
diff --git a/usr/src/pkgdefs/SUNWaudd/prototype_sparc b/usr/src/pkgdefs/SUNWaudd/prototype_sparc
index b74cce5a43..d47f7d6d53 100644
--- a/usr/src/pkgdefs/SUNWaudd/prototype_sparc
+++ b/usr/src/pkgdefs/SUNWaudd/prototype_sparc
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -48,12 +46,12 @@ f none kernel/drv/audiocs.conf 644 root sys
 f none kernel/drv/audiots.conf 644 root sys
 f none kernel/drv/audio1575.conf 644 root sys
 d none kernel/drv/sparcv9 755 root sys
+f none kernel/drv/sparcv9/audio 755 root sys
 f none kernel/drv/sparcv9/audiocs 755 root sys
 f none kernel/drv/sparcv9/audioens 755 root sys
+f none kernel/drv/sparcv9/audiopci 755 root sys
 f none kernel/drv/sparcv9/audiots 755 root sys
 f none kernel/drv/sparcv9/audio1575 755 root sys
+f none kernel/drv/sparcv9/austr 755 root sys
 d none kernel/misc/sparcv9 755 root sys
-f none kernel/misc/sparcv9/amsrc1 755 root sys
-f none kernel/misc/sparcv9/amsrc2 755 root sys
-f none kernel/misc/sparcv9/audiosup 755 root sys
-f none kernel/misc/sparcv9/mixer 755 root sys
+f none kernel/misc/sparcv9/ac97 755 root sys
diff --git a/usr/src/pkgdefs/SUNWaudh/prototype_com b/usr/src/pkgdefs/SUNWaudh/prototype_com
index f8b1084510..992b63d102 100644
--- a/usr/src/pkgdefs/SUNWaudh/prototype_com
+++ b/usr/src/pkgdefs/SUNWaudh/prototype_com
@@ -19,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -45,20 +43,16 @@ i depend
 #
 d none usr 755 root sys
 d none usr/include 755 root bin
+d none usr/include/audio 755 root bin
+f none usr/include/audio/au.h 644 root bin
 d none usr/include/sys 755 root bin
 f none usr/include/sys/audio.h 644 root bin
 f none usr/include/sys/audioio.h 644 root bin
-f none usr/include/sys/audiovar.h 644 root bin
 f none usr/include/sys/mixer.h 644 root bin
+f none usr/include/sys/soundcard.h 644 root bin
 d none usr/include/sys/audio 755 root bin
-d none usr/include/audio 755 root bin
-f none usr/include/audio/au.h 644 root bin
 f none usr/include/sys/audio/ac97.h 644 root bin
-f none usr/include/sys/audio/am_src1.h 644 root bin
-f none usr/include/sys/audio/audioens.h 644 root bin
-f none usr/include/sys/audio/audio_apm.h 644 root bin
-f none usr/include/sys/audio/audio_mixer.h 644 root bin
-f none usr/include/sys/audio/audio_src.h 644 root bin
-f none usr/include/sys/audio/audio_support.h 644 root bin
-f none usr/include/sys/audio/audio_trace.h 644 root bin
+f none usr/include/sys/audio/audio_common.h 644 root bin
+f none usr/include/sys/audio/audio_driver.h 644 root bin
+f none usr/include/sys/audio/audio_oss.h 644 root bin
 f none usr/include/sys/audio/g711.h 644 root bin
diff --git a/usr/src/pkgdefs/SUNWaudh/prototype_i386 b/usr/src/pkgdefs/SUNWaudh/prototype_i386
index 7050a4f45d..08f9663c82 100644
--- a/usr/src/pkgdefs/SUNWaudh/prototype_i386
+++ b/usr/src/pkgdefs/SUNWaudh/prototype_i386
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -46,7 +44,3 @@
 #
 # SUNWaudh
 #
-f none usr/include/sys/audio/audio810.h 644 root bin
-f none usr/include/sys/audio/audiohd.h 644 root bin
-f none usr/include/sys/audio/audioixp.h 644 root bin
-f none usr/include/sys/audio/audiovia823x.h 644 root bin
diff --git a/usr/src/pkgdefs/SUNWaudh/prototype_sparc b/usr/src/pkgdefs/SUNWaudh/prototype_sparc
index 5c59192742..0e3250562d 100644
--- a/usr/src/pkgdefs/SUNWaudh/prototype_sparc
+++ b/usr/src/pkgdefs/SUNWaudh/prototype_sparc
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -44,7 +42,3 @@
 #
 # SUNWaudh
 #
-f none usr/include/sys/audio/audio_4231.h 644 root bin
-f none usr/include/sys/audio/audiots.h 644 root bin
-f none usr/include/sys/audio/audio1575.h 644 root bin
-f none usr/include/sys/audio/am_src2.h 644 root bin
diff --git a/usr/src/pkgdefs/SUNWaudiohd/postinstall b/usr/src/pkgdefs/SUNWaudiohd/postinstall
index e02d59e4f5..77af8bf454 100644
--- a/usr/src/pkgdefs/SUNWaudiohd/postinstall
+++ b/usr/src/pkgdefs/SUNWaudiohd/postinstall
@@ -20,10 +20,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
 #
 # SUNWaudiohd postinstall script
 
@@ -55,6 +54,7 @@ AUDIOHD_ALIASES="\
 	\"pci8086,2668\" \
 	\"pci10de,371\" \
 	\"pciclass,0403\" \
+	\"pciexclass,0403\"\
 	"
 	
 case "${ARCH}" in
diff --git a/usr/src/pkgdefs/SUNWcsr/prototype_com b/usr/src/pkgdefs/SUNWcsr/prototype_com
index d3bbcae315..1d3ca9cf52 100644
--- a/usr/src/pkgdefs/SUNWcsr/prototype_com
+++ b/usr/src/pkgdefs/SUNWcsr/prototype_com
@@ -349,6 +349,7 @@ d none lib/svc/method 0755 root bin
 f none lib/svc/method/boot-archive 0555 root bin
 f none lib/svc/method/boot-archive-update 0555 root bin
 f none lib/svc/method/console-login 0555 root bin
+f none lib/svc/method/devices-audio 0555 root bin
 f none lib/svc/method/devices-local 0555 root bin
 f none lib/svc/method/fs-local 0555 root bin
 f none lib/svc/method/fs-minimal 0555 root bin
@@ -531,6 +532,7 @@ d none var/svc/manifest/system 755 root sys
 f manifest var/svc/manifest/system/boot-archive.xml 0444 root sys
 f manifest var/svc/manifest/system/boot-archive-update.xml 0444 root sys
 d none var/svc/manifest/system/device 755 root sys
+f manifest var/svc/manifest/system/device/devices-audio.xml 0444 root sys
 f manifest var/svc/manifest/system/device/devices-local.xml 0444 root sys
 f manifest var/svc/manifest/system/device/mpxio-upgrade.xml 0444 root sys
 d none var/svc/manifest/system/filesystem 0755 root sys
diff --git a/usr/src/pkgdefs/SUNWmdb/prototype_i386 b/usr/src/pkgdefs/SUNWmdb/prototype_i386
index fb1a898f13..09e244c11c 100644
--- a/usr/src/pkgdefs/SUNWmdb/prototype_i386
+++ b/usr/src/pkgdefs/SUNWmdb/prototype_i386
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -57,7 +57,6 @@ f none usr/platform/i86xpv/lib/mdb/kvm/xpv_psm.so 555 root sys
 f none usr/platform/i86xpv/lib/mdb/kvm/xpv_uppc.so 555 root sys
 d none usr/lib/mdb/kvm/amd64 755 root sys
 f none usr/lib/mdb/kvm/amd64/arp.so 555 root sys
-f none usr/lib/mdb/kvm/amd64/audiosup.so 555 root sys
 f none usr/lib/mdb/kvm/amd64/cpc.so 555 root sys
 f none usr/lib/mdb/kvm/amd64/cpu.generic.so 555 root sys
 f none usr/lib/mdb/kvm/amd64/cpu_ms.AuthenticAMD.15.so 555 root sys
@@ -91,7 +90,6 @@ f none usr/lib/mdb/kvm/amd64/uhci.so 555 root sys
 f none usr/lib/mdb/kvm/amd64/usba.so 555 root sys
 f none usr/lib/mdb/kvm/amd64/sockfs.so 555 root sys
 f none usr/lib/mdb/kvm/arp.so 555 root sys
-f none usr/lib/mdb/kvm/audiosup.so 555 root sys
 f none usr/lib/mdb/kvm/cpc.so 555 root sys
 f none usr/lib/mdb/kvm/cpu.generic.so 555 root sys
 f none usr/lib/mdb/kvm/cpu_ms.AuthenticAMD.15.so 555 root sys
diff --git a/usr/src/pkgdefs/SUNWmdb/prototype_sparc b/usr/src/pkgdefs/SUNWmdb/prototype_sparc
index eae343b703..8724dadf21 100644
--- a/usr/src/pkgdefs/SUNWmdb/prototype_sparc
+++ b/usr/src/pkgdefs/SUNWmdb/prototype_sparc
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -38,7 +38,6 @@ f none usr/bin/sparcv9/mdb 555 root bin
 l none usr/bin/sparcv9/adb=../../../usr/bin/sparcv9/mdb
 d none usr/lib/mdb/kvm/sparcv9 755 root sys
 f none usr/lib/mdb/kvm/sparcv9/arp.so 555 root sys
-f none usr/lib/mdb/kvm/sparcv9/audiosup.so 555 root sys
 f none usr/lib/mdb/kvm/sparcv9/cpc.so 555 root sys
 f none usr/lib/mdb/kvm/sparcv9/crypto.so 555 root sys
 f none usr/lib/mdb/kvm/sparcv9/genunix.so 555 root sys
diff --git a/usr/src/pkgdefs/SUNWmdbr/prototype_i386 b/usr/src/pkgdefs/SUNWmdbr/prototype_i386
index 662f4cb1e3..bc503a6419 100644
--- a/usr/src/pkgdefs/SUNWmdbr/prototype_i386
+++ b/usr/src/pkgdefs/SUNWmdbr/prototype_i386
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -27,7 +27,6 @@
 
 d none kernel/kmdb/amd64 755 root sys
 f none kernel/kmdb/amd64/arp 555 root sys
-f none kernel/kmdb/amd64/audiosup 555 root sys
 f none kernel/kmdb/amd64/cpc 555 root sys
 f none kernel/kmdb/amd64/cpu.generic 555 root sys
 f none kernel/kmdb/amd64/cpu_ms.AuthenticAMD.15 555 root sys
@@ -60,7 +59,6 @@ f none kernel/kmdb/amd64/ufs 555 root sys
 f none kernel/kmdb/amd64/uhci 555 root sys
 f none kernel/kmdb/amd64/usba 555 root sys
 f none kernel/kmdb/arp 555 root sys
-f none kernel/kmdb/audiosup 555 root sys
 f none kernel/kmdb/cpc 555 root sys
 f none kernel/kmdb/cpu.generic 555 root sys
 f none kernel/kmdb/cpu_ms.AuthenticAMD.15 555 root sys
diff --git a/usr/src/pkgdefs/SUNWmdbr/prototype_sparc b/usr/src/pkgdefs/SUNWmdbr/prototype_sparc
index 0e3e805552..d290db813d 100644
--- a/usr/src/pkgdefs/SUNWmdbr/prototype_sparc
+++ b/usr/src/pkgdefs/SUNWmdbr/prototype_sparc
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -27,7 +27,6 @@
 #
 d none kernel/kmdb/sparcv9 755 root sys
 f none kernel/kmdb/sparcv9/arp 555 root sys
-f none kernel/kmdb/sparcv9/audiosup 555 root sys
 f none kernel/kmdb/sparcv9/cpc 555 root sys
 f none kernel/kmdb/sparcv9/crypto 555 root sys
 f none kernel/kmdb/sparcv9/genunix 555 root sys
diff --git a/usr/src/pkgdefs/SUNWtsg/prototype_com b/usr/src/pkgdefs/SUNWtsg/prototype_com
index ce19bf8603..c1c0897e0f 100644
--- a/usr/src/pkgdefs/SUNWtsg/prototype_com
+++ b/usr/src/pkgdefs/SUNWtsg/prototype_com
@@ -19,11 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
 # and their location on the development machine when building the package.
@@ -76,14 +74,17 @@ e devallocdefs etc/security/tsol/devalloc_defaults 0644 root sys
 # Device clean scripts for the window system.
 #
 d none etc/security/lib 755 root sys
-f none etc/security/lib/audio_clean_wrapper 555 root sys
-s none etc/security/lib/audio_clean_wrapper.windowing=./audio_clean_wrapper
+s none etc/security/lib/audio_clean.windowing=./audio_clean
 f none etc/security/lib/disk_clean 555 root sys
 s none etc/security/lib/disk_clean.windowing=./disk_clean
 s none etc/security/lib/st_clean.windowing=./st_clean
 f none etc/security/lib/wdwmsg 555 root sys
 f none etc/security/lib/wdwwrapper 555 root sys
 #
+# Directory for saved audio settings
+#
+d none etc/security/audio 755 root sys
+#
 # smf glue
 d none lib 755 root bin
 d none lib/svc 755 root bin
diff --git a/usr/src/pkgdefs/SUNWusb/prototype_i386 b/usr/src/pkgdefs/SUNWusb/prototype_i386
index 3f11cdcf2c..f838cda4e1 100644
--- a/usr/src/pkgdefs/SUNWusb/prototype_i386
+++ b/usr/src/pkgdefs/SUNWusb/prototype_i386
@@ -23,7 +23,6 @@
 # Use is subject to license terms.
 #
 
-
 #
 # This required package information file contains a list of package contents.
 # The 'pkgmk' command uses this file to identify the contents of a package
@@ -48,7 +47,6 @@
 #
 # SUNWusb
 #
-f none kernel/dacf/usb_ac_dacf 0755 root sys
 f none kernel/drv/uhci 0755 root sys
 f none kernel/drv/ohci 0755 root sys
 f none kernel/drv/ehci 0755 root sys
@@ -72,7 +70,6 @@ f none kernel/strmod/usbkbm 0755 root sys
 f none kernel/strmod/usbms 0755 root sys
 f none kernel/strmod/usb_ah 0755 root sys
 d none kernel/dacf/amd64 0755 root sys
-f none kernel/dacf/amd64/usb_ac_dacf 0755 root sys
 d none kernel/drv/amd64 0755 root sys
 f none kernel/drv/amd64/uhci 0755 root sys
 f none kernel/drv/amd64/ohci 0755 root sys
diff --git a/usr/src/pkgdefs/SUNWusb/prototype_sparc b/usr/src/pkgdefs/SUNWusb/prototype_sparc
index 032085ff5a..afe299a46b 100644
--- a/usr/src/pkgdefs/SUNWusb/prototype_sparc
+++ b/usr/src/pkgdefs/SUNWusb/prototype_sparc
@@ -48,7 +48,6 @@
 # SUNWusb
 #
 d none kernel/dacf/sparcv9 0755 root sys
-f none kernel/dacf/sparcv9/usb_ac_dacf 0755 root sys
 d none kernel/drv/sparcv9 0755 root sys
 f none kernel/drv/sparcv9/ohci 0755 root sys
 f none kernel/drv/sparcv9/ehci 0755 root sys
diff --git a/usr/src/pkgdefs/etc/exception_list_sparc b/usr/src/pkgdefs/etc/exception_list_sparc
index ac70d0ee47..b78baab717 100644
--- a/usr/src/pkgdefs/etc/exception_list_sparc
+++ b/usr/src/pkgdefs/etc/exception_list_sparc
@@ -822,21 +822,6 @@ kernel/drv/usbskel.conf			sparc
 #
 usr/include/sys/libdevid.h		sparc
 #
-# amd 8111 audio driver doesn't ship in sparc platform.
-#
-kernel/drv/sparcv9/audio810		sparc
-kernel/drv/audio810.conf		sparc
-#
-# High definition audio driver doesn't ship in sparc platform.
-#
-kernel/drv/sparcv9/audiohd		sparc
-kernel/drv/audiohd.conf			sparc
-#
-# VIA VT823x audio driver doesn't ship in sparc platform.
-#
-kernel/drv/sparcv9/audiovia823x		sparc
-kernel/drv/audiovia823x.conf		sparc
-#
 # The following files are installed in the proto area by the build of
 # libprtdiag.  libprtdiag contains interfaces which are all private.
 # Only the shared object is shipped.
diff --git a/usr/src/tools/findunref/exception_list.open b/usr/src/tools/findunref/exception_list.open
index 8d70459071..1d0c7a43b5 100644
--- a/usr/src/tools/findunref/exception_list.open
+++ b/usr/src/tools/findunref/exception_list.open
@@ -20,7 +20,7 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 
@@ -117,11 +117,6 @@
 ./src/*.ref
 
 #
-# Ignore warlock-related goop; not used during a nightly.
-#
-./src/uts/sparc/amsrc*/main.c
-
-#
 # Ignore internal test directories and test programs.
 #
 */tests
@@ -250,7 +245,6 @@
 ./src/uts/sparc/si3124/Makefile
 ./src/uts/sparc/nv_sata/Makefile
 ./src/uts/sparc/ahci/Makefile
-./src/uts/sparc/audioixp/Makefile
 
 #
 # Ignore uttrack.c.  It is provided as part of the standard
diff --git a/usr/src/tools/scripts/bfu.sh b/usr/src/tools/scripts/bfu.sh
index ee62f02801..58614feb6d 100644
--- a/usr/src/tools/scripts/bfu.sh
+++ b/usr/src/tools/scripts/bfu.sh
@@ -302,6 +302,7 @@ superfluous_nonglobal_zone_files="
 	kernel
 	lib/libmeta.so
 	lib/libmeta.so.1
+	lib/svc/method/devices-audio
 	lib/svc/method/fc-fabric
 	lib/svc/method/iscsi-initiator
 	lib/svc/method/npivconfig
@@ -379,6 +380,7 @@ superfluous_nonglobal_zone_files="
 	var/svc/manifest/platform/sun4u/sf880drd.xml
 	var/svc/manifest/platform/sun4v
 	var/svc/manifest/system/cvc.xml
+	var/svc/manifest/system/device/devices-audio.xml
 	var/svc/manifest/system/device/devices-fc-fabric.xml
 	var/svc/manifest/system/dumpadm.xml
 	var/svc/manifest/system/filesystem/rmvolmgr.xml
@@ -963,6 +965,33 @@ nfsmapid_cfg() {
 }
 
 #
+# Detect Boomer audio framework; used to emit a notice at the end of BFU
+# telling the user to run update_audio to complete the upgrade.
+#
+BOOMER_PRESENT_SYS=false
+BOOMER_PRESENT_BFU=false
+BOOMER_DRIVERS="audio austr"
+
+check_boomer_sys() {
+	typeset root=$1
+	typeset n2m=$root/etc/name_to_major
+	typeset drv
+
+	for drv in $BOOMER_DRIVERS; do
+		if ! grep -w $drv $n2m > /dev/null 2>&1; then
+			return 1
+		fi
+	done
+
+	return 0
+}
+
+check_boomer_bfu() {
+	$ZCAT $cpiodir/generic.root$ZFIX | cpio -it 2>/dev/null |
+	    grep devices-audio.xml > /dev/null 2>&1
+}
+
+#
 # Define global variables
 #
 CALL_DEVID_DESTROY=""
@@ -1646,6 +1675,16 @@ smf_cleanup_vt() {
 	)
 }
 
+smf_cleanup_boomer() {
+	(
+		smf_delete_manifest var/src/manifest/system/devices-audio.xml
+		cd $root
+		rm -f lib/svc/method/devices-audio
+
+		/tmp/bfubin/svccfg delete -f svc:/system/device/audio
+	)
+}
+
 old_mfst_dir="var/svc/manifest.orig"
 new_mfst_dir="var/svc/manifest"
 
@@ -1980,6 +2019,13 @@ smf_apply_conf () {
 		smf_cleanup_vt
 	fi
 
+	#
+	# Remove the Boomer audio service when BFUing to legacy audio bits
+	#
+	if ! check_boomer_bfu && [ $zone = global ]; then
+		smf_cleanup_boomer
+	fi
+
 	print "Disabling unneeded inetd.conf entries ..."
 	smf_inetd_disable
 	smf_tftp_reinstall
@@ -6679,7 +6725,6 @@ mondo_loop() {
 		$root/kernel/misc/consconfig \
 		$root/kernel/misc/dada \
 		$root/kernel/misc/des \
-		$root/kernel/misc/diaudio \
 		$root/kernel/misc/fctl \
 		$root/kernel/misc/fssnap_if \
 		$root/kernel/misc/gld \
@@ -7035,6 +7080,64 @@ mondo_loop() {
 	fi
 
 	#
+	# Remove obsolete audio bits. The SADA modules are removed as
+	# no driver uses them any more.  We don't ship certain device
+	# private headers anymore, as well as headers for the now
+	# obsolete (and never documented) legacy audio driver API.
+	#
+	rm -f $usr/include/sys/audiovar.h
+	rm -f $usr/include/sys/audio/am_src1.h
+	rm -f $usr/include/sys/audio/am_src2.h
+	rm -f $usr/include/sys/audio/audio1575.h
+	rm -f $usr/include/sys/audio/audio810.h
+	rm -f $usr/include/sys/audio/audioens.h
+	rm -f $usr/include/sys/audio/audiohd.h
+	rm -f $usr/include/sys/audio/audioixp.h
+	rm -f $usr/include/sys/audio/audiots.h
+	rm -f $usr/include/sys/audio/audiovia823x.h
+	rm -f $usr/include/sys/audio/audio_4231.h
+	rm -f $usr/include/sys/audio/audio_apm.h
+	rm -f $usr/include/sys/audio/audio_mixer.h
+	rm -f $usr/include/sys/audio/audio_src.h
+	rm -f $usr/include/sys/audio/audio_support.h
+	rm -f $usr/include/sys/audio/audio_trace.h
+	rm -f $root/kernel/misc/amsrc1
+	rm -f $root/kernel/misc/amsrc2
+	rm -f $root/kernel/misc/audiosup
+	rm -f $root/kernel/misc/diaudio
+	rm -f $root/kernel/misc/mixer
+	rm -f $root/kernel/misc/amd64/amsrc1
+	rm -f $root/kernel/misc/amd64/amsrc2
+	rm -f $root/kernel/misc/amd64/audiosup
+	rm -f $root/kernel/misc/amd64/diaudio
+	rm -f $root/kernel/misc/amd64/mixer
+	rm -f $root/kernel/misc/sparcv9/amsrc1
+	rm -f $root/kernel/misc/sparcv9/amsrc2
+	rm -f $root/kernel/misc/sparcv9/audiosup
+	rm -f $root/kernel/misc/sparcv9/diaudio
+	rm -f $root/kernel/misc/sparcv9/mixer
+
+	#
+	# Determine whether to emit update_audio notice or not
+	#
+	check_boomer_sys $root && BOOMER_PRESENT_SYS=true
+	check_boomer_bfu && BOOMER_PRESENT_BFU=true
+
+	# Cleanup audio devlinks when bfu'ing back to legacy audio bits.
+	# We also cleanup devlinks the first time when upgrading from legacy
+	# to Boomer bits in the update_audio script
+	#
+	if $BOOMER_PRESENT_SYS && ! $BOOMER_PRESENT_BFU; then
+		rm -f $root/dev/mixer*
+		rm -f $root/dev/sndstat*
+		rm -f $root/dev/audio*
+		rm -f $root/dev/dsp*
+		rm -f $root/dev/sound/*
+
+		touch $root/reconfigure
+	fi
+
+	#
 	# Diskless clients have already extracted /usr so don't delete this
 	# Non-global zones never extracts /usr so don't delete these
 	#
@@ -8220,6 +8323,17 @@ mondo_loop() {
 		$epilogue || print "WARNING: $epilogue failed with code $?"
 	fi
 
+	#
+	# Emit notice about upgrading the audio sub-system
+	#
+	if $BOOMER_PRESENT_BFU && ! $BOOMER_PRESENT_SYS; then
+		print
+		print "NOTICE: you must run the 'update_audio' script (from"
+		print "${GATE}/public/bin) to complete the upgrade of "
+		print "the audio sub-system"
+		print
+	fi
+
 	((seconds = SECONDS))
 	((min = seconds / 60))
 	((sec = seconds % 60))
diff --git a/usr/src/uts/common/Makefile.files b/usr/src/uts/common/Makefile.files
index e39f3c4273..c94490f6e5 100644
--- a/usr/src/uts/common/Makefile.files
+++ b/usr/src/uts/common/Makefile.files
@@ -418,16 +418,31 @@ DCPC_OBJS += dcpc.o
 #
 IPP_OBJS +=	ippctl.o
 
+AUDIO_OBJS += audio_client.o audio_ddi.o audio_engine.o \
+	   audio_fltdata.o audio_format.o audio_ctrl.o \
+	   audio_grc3.o audio_output.o audio_input.o \
+	   audio_oss.o audio_sun.o
+
+AUSTR_OBJS += austr.o
+
+AUDIOENS_OBJS += audioens.o
+
 AUDIOVIA823X_OBJS += audiovia823x.o
 
+AUDIO1575_OBJS += audio1575.o
+
 AUDIO810_OBJS += audio810.o
 
 AUDIOHD_OBJS +=	audiohd.o
 
 AUDIOIXP_OBJS += audioixp.o
 
+AUDIOPCI_OBJS += audiopci.o
+
 AUDIOTS_OBJS +=	audiots.o
 
+AC97_OBJS += ac97.o ac97_ad.o ac97_alc.o ac97_cmi.o
+
 CARDBUS_OBJS += cardbus.o cardbus_hp.o cardbus_cfg.o
 
 CONSKBD_OBJS += conskbd.o
@@ -669,9 +684,7 @@ USBVC_OBJS += usbvc.o usbvc_v4l2.o
 
 HIDPARSER_OBJS += hidparser.o
 
-USB_AC_OBJS += usb_ac.o
-
-USB_AC_DACF_OBJS += usb_ac_dacf.o
+USB_AC_OBJS += usb_ac.o audio_shim.o
 
 USB_AS_OBJS += usb_as.o
 
@@ -1446,12 +1459,6 @@ KSSL_OBJS +=	kssl.o ksslioctl.o ksslapi.o ksslrec.o
 #			misc. modules
 #
 
-AMSRC2_OBJS +=	am_src2.o
-
-AUDIO_SUP_OBJS +=	audio_support.o
-
-MIXER_OBJS +=	am_main.o am_ad.o am_ioctl.o
-
 C2AUDIT_OBJS +=	adr.o audit.o audit_event.o audit_io.o \
 		audit_path.o audit_start.o audit_syscalls.o audit_token.o \
 		audit_mem.o audit_zone.o
diff --git a/usr/src/uts/common/Makefile.rules b/usr/src/uts/common/Makefile.rules
index c78620958b..0aa7d11769 100644
--- a/usr/src/uts/common/Makefile.rules
+++ b/usr/src/uts/common/Makefile.rules
@@ -532,31 +532,43 @@ $(OBJS_DIR)/%.o:               $(UTSBASE)/common/io/atu/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/amsrc2/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/impl/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/drv/audio810/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/ac97/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/drv/audiohd/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audioens/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/drv/audioixp/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audio1575/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/drv/audiots/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audio810/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/framework/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audiohd/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/sada/mixer/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audioixp/%.c
+	$(COMPILE.c) -o $@ $<
+	$(CTFCONVERT_O)
+
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audiopci/%.c
+	$(COMPILE.c) -o $@ $<
+	$(CTFCONVERT_O)
+
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audiots/%.c
+	$(COMPILE.c) -o $@ $<
+	$(CTFCONVERT_O)
+
+$(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/audio/drv/audiovia823x/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
@@ -1695,25 +1707,34 @@ $(LINTS_DIR)/%.ln:              $(UTSBASE)/common/io/ath/%.c
 $(LINTS_DIR)/%.ln:              $(UTSBASE)/common/io/atu/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/amsrc2/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/impl/%.c
+	@($(LHEAD) $(LINT.c) $< $(LTAIL))
+
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/ac97/%.c
+	@($(LHEAD) $(LINT.c) $< $(LTAIL))
+
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audio1575/%.c
+	@($(LHEAD) $(LINT.c) $< $(LTAIL))
+
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audio810/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/mixer/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audioens/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/framework/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audiohd/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/drv/audio810/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audioixp/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/drv/audiohd/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audiopci/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/drv/audioixp/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audiots/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/sada/drv/audiots/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/audio/drv/audiovia823x/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
 $(LINTS_DIR)/%.ln:		$(UTSBASE)/common/io/bge/%.c
diff --git a/usr/src/uts/common/brand/lx/io/lx_audio.c b/usr/src/uts/common/brand/lx/io/lx_audio.c
index 8f80c61853..957e3de108 100644
--- a/usr/src/uts/common/brand/lx/io/lx_audio.c
+++ b/usr/src/uts/common/brand/lx/io/lx_audio.c
@@ -1086,30 +1086,14 @@ lxa_audio_getinfo(lxa_state_t *lxa_state, intptr_t arg, int mode)
 static int
 lxa_mixer_ai_from_lh(ldi_handle_t lh, audio_info_t *ai)
 {
-	am_control_t	*actl;
-	int		rv, ch_count, junk;
+	int		rv, junk;
 
 	ASSERT((lh != NULL) && (ai != NULL));
 
-	/* get the number of channels for the underlying device */
-	if ((rv = ldi_ioctl(lh, AUDIO_GET_NUM_CHS,
-	    (intptr_t)&ch_count, FKIOCTL, kcred, &junk)) != 0)
-		return (rv);
-
-	/* allocate the am_control_t structure */
-	actl = kmem_alloc(AUDIO_MIXER_CTL_STRUCT_SIZE(ch_count), KM_SLEEP);
-
 	/* get the device state and channel state */
-	if ((rv = ldi_ioctl(lh, AUDIO_MIXERCTL_GETINFO,
-	    (intptr_t)actl, FKIOCTL, kcred, &junk)) != 0) {
-		kmem_free(actl, AUDIO_MIXER_CTL_STRUCT_SIZE(ch_count));
-		return (rv);
-	}
+	rv = ldi_ioctl(lh, AUDIO_GETINFO, (intptr_t)ai, FKIOCTL, kcred, &junk);
 
-	/* return the audio_info structure */
-	*ai = actl->dev_info;
-	kmem_free(actl, AUDIO_MIXER_CTL_STRUCT_SIZE(ch_count));
-	return (0);
+	return (rv);
 }
 
 static int
@@ -1199,18 +1183,8 @@ lxa_mixer_set_common(lxa_state_t *lxa_state, int cmd, intptr_t arg, int mode)
 		break;
 	}
 
-	/*
-	 * we're going to cheat here.  normally the
-	 * MIXERCTL_SETINFO ioctl take am_control_t and the
-	 * AUDIO_SETINFO takes an audio_info_t.  as it turns
-	 * out the first element in a am_control_t is an
-	 * audio_info_t.  also, the rest of the am_control_t
-	 * structure is normally ignored for a MIXERCTL_SETINFO
-	 * ioctl.  so here we'll try to fall back to the code
-	 * that handles AUDIO_SETINFO ioctls.
-	 */
-	return (lxa_audio_setinfo(lxa_state, AUDIO_MIXERCTL_SETINFO,
-	    (intptr_t)&ai, FKIOCTL));
+	return (lxa_audio_setinfo(lxa_state, AUDIO_SETINFO, (intptr_t)&ai,
+	    FKIOCTL));
 }
 
 static int
@@ -1665,6 +1639,7 @@ lxa_read(dev_t dev, struct uio *uiop, cred_t *credp)
 {
 	lxa_state_t	*lxa_state;
 	minor_t		minor = getminor(dev);
+	int		rv;
 
 	/* get the handle for this device */
 	if (mod_hash_find(lxa_state_hash, (mod_hash_key_t)(uintptr_t)minor,
@@ -1683,7 +1658,13 @@ lxa_read(dev_t dev, struct uio *uiop, cred_t *credp)
 		return (EBADF);
 
 	/* pass the request on */
-	return (ldi_read(lxa_state->lxas_idev_lh, uiop, kcred));
+	while (uiop->uio_resid != 0) {
+		rv = ldi_read(lxa_state->lxas_idev_lh, uiop, kcred);
+		if ((rv != 0) || (uiop->uio_fmode & (FNONBLOCK|FNDELAY))) {
+			break;
+		}
+	}
+	return (rv);
 }
 
 static int
@@ -1692,6 +1673,7 @@ lxa_write(dev_t dev, struct uio *uiop, cred_t *credp)
 {
 	lxa_state_t	*lxa_state;
 	minor_t		minor = getminor(dev);
+	int		rv;
 
 	/* get the handle for this device */
 	if (mod_hash_find(lxa_state_hash, (mod_hash_key_t)(uintptr_t)minor,
@@ -1710,7 +1692,13 @@ lxa_write(dev_t dev, struct uio *uiop, cred_t *credp)
 		return (EBADF);
 
 	/* pass the request on */
-	return (ldi_write(lxa_state->lxas_odev_lh, uiop, kcred));
+	while (uiop->uio_resid != 0) {
+		rv = ldi_write(lxa_state->lxas_odev_lh, uiop, kcred);
+		if ((rv != 0) || (uiop->uio_fmode & (FNONBLOCK|FNDELAY))) {
+			break;
+		}
+	}
+	return (rv);
 }
 
 static int
diff --git a/usr/src/uts/common/io/audio/ac97/ac97.c b/usr/src/uts/common/io/audio/ac97/ac97.c
new file mode 100644
index 0000000000..783e1d1efe
--- /dev/null
+++ b/usr/src/uts/common/io/audio/ac97/ac97.c
@@ -0,0 +1,1974 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/list.h>
+#include <sys/sysmacros.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include <sys/note.h>
+#include "ac97_impl.h"
+
+/*
+ * This is the initial value for many controls. This is
+ * a 75% level.
+ */
+#define	INIT_VAL_MAIN	((75 << 8) | 75)
+#define	INIT_VAL_ST	((75 << 8) | 75)
+#define	INIT_VAL_MN	75
+#define	INIT_IGAIN_ST	((50 << 8) | 50)
+#define	INIT_IGAIN_MN	50
+
+/*
+ * In AC'97 v2.3, the registers are carved up as follows:
+ *
+ * Audio Base Registers: 	0x00 - 0x26
+ * Audio Extended Registers:	0x28 - 0x3A
+ * Modem Extended Registers:	0x3C - 0x58
+ * Vendor Reserved Registers:	0x5A - 0x5F
+ * Page Registers:		0x60 - 0x6F
+ * Vendor Reserved Registers:	0x70 - 0x7A
+ * Vendor ID Registers:		0x7C - 0x7F
+ *
+ * We only need to shadow the normal audio registers by default.
+ * TBD: Handling of codec-specific registers in vendor reserved space.
+ * We cannot necessarily meaningfully shadow them.
+ */
+#define	LAST_SHADOW_REG	0x3A
+#define	NUM_SHADOW	((LAST_SHADOW_REG / sizeof (uint16_t)) + 1)
+#define	SHADOW(ac, reg)	((ac)->shadow[((reg) / sizeof (uint16_t))])
+
+/*
+ * Record source selection.
+ */
+#define	INPUT_MIC		0
+#define	INPUT_CD		1
+#define	INPUT_VIDEO		2
+#define	INPUT_AUXIN		3
+#define	INPUT_LINEIN		4
+#define	INPUT_STEREOMIX		5
+#define	INPUT_MONOMIX		6
+#define	INPUT_PHONE		7
+
+static const char *ac97_insrcs[] = {
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_CD,
+	AUDIO_PORT_VIDEO,
+	AUDIO_PORT_AUX1IN,
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_STEREOMIX,
+	AUDIO_PORT_MONOMIX,
+	AUDIO_PORT_PHONE,
+	NULL,
+};
+
+/*
+ * Per audio device state structure
+ */
+struct ac97 {
+	dev_info_t	*dip;	/* DDI device instance */
+	audio_dev_t	*d;
+	void		*private;  /* drivers devc */
+	ac97_rd_t	rd;	/* drivers port read routine */
+	ac97_wr_t	wr;	/* drivers port write routine */
+	char		name[128]; /* driver instance name */
+
+	uint16_t	shadow[NUM_SHADOW];
+
+	boolean_t	suspended;		/* true if suspended */
+	kt_did_t	resumer;		/* resumer if suspended */
+	uint32_t	flags;
+#define	AC97_FLAG_AMPLIFIER	(1 << 0)	/* ext. amp on by default */
+#define	AC97_FLAG_MICBOOST	(1 << 1)	/* micboost on by default */
+#define	AC97_FLAG_SPEAKER	(1 << 2)	/* mono out on by default */
+
+#define	AC97_FLAG_AUX_HP	(1 << 4)	/* possible uses for AUX_OUT */
+#define	AC97_FLAG_AUX_4CH	(1 << 5)
+#define	AC97_FLAG_AUX_LVL	(1 << 6)
+#define	AC97_FLAG_SPEAKER_OK	(1 << 7)	/* expose mono out */
+#define	AC97_FLAG_NO_HEADPHONE	(1 << 8)	/* do not expose headphone */
+#define	AC97_FLAG_NO_CDROM	(1 << 9)	/* do not expose CDROM */
+#define	AC97_FLAG_NO_PHONE	(1 << 10)	/* do not expose phone in */
+#define	AC97_FLAG_NO_VIDEO	(1 << 11)	/* do not expose video in */
+#define	AC97_FLAG_NO_AUXIN	(1 << 12)	/* do not expose aux in */
+#define	AC97_FLAG_NO_AUXOUT	(1 << 13)	/* do not expose aux out */
+#define	AC97_FLAG_NO_LINEIN	(1 << 14)	/* do not expose linein */
+#define	AC97_FLAG_NO_MIC	(1 << 15)	/* do not expose mic */
+
+	uint32_t	vid;			/* Vendor ID for CODEC */
+	uint16_t	caps;
+
+	void		(*codec_init)(ac97_t *);
+	void		(*codec_reset)(ac97_t *);
+
+	kmutex_t	ac_lock;
+	list_t		ctrls;
+
+	uint64_t	inputs;
+};
+
+struct modlmisc ac97_modlmisc = {
+	&mod_miscops,
+	"Audio Codec '97 Support"
+};
+
+struct modlinkage ac97_modlinkage = {
+	MODREV_1,
+	{ &ac97_modlmisc, NULL }
+};
+
+int
+_init(void)
+{
+	return (mod_install(&ac97_modlinkage));
+}
+
+int
+_fini(void)
+{
+	return (mod_install(&ac97_modlinkage));
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&ac97_modlinkage, modinfop));
+}
+
+
+#if 0
+/*
+ * The following table, and the code to scale it, works in percentages.
+ * This may be convenient for humans, but it would be faster if the table
+ * entries were rescaled to 256.  (Division by 100 is painful.  Divison by
+ * 256 is trivial.)
+ */
+static const char ac97_val_cvt[101] = {
+	0, 0, 3, 7, 10, 13, 16, 19,
+	21, 23, 26, 28, 30, 32, 34, 35,
+	37, 39, 40, 42,	43, 45, 46, 47,
+	49, 50, 51, 52, 53, 55, 56, 57,
+	58, 59, 60, 61, 62, 63, 64, 65,
+	65, 66, 67, 68, 69, 70, 70, 71,
+	72, 73, 73, 74, 75, 75, 76, 77,
+	77, 78, 79, 79, 80, 81, 81, 82,
+	82, 83, 84, 84, 85, 85, 86, 86,
+	87, 87, 88, 88, 89, 89, 90, 90,
+	91, 91, 92, 92, 93, 93, 94, 94,
+	95, 95, 96, 96, 96, 97, 97, 98,
+	98, 98, 99, 99, 100
+};
+#endif
+
+/*
+ * This code has three main functions. All related to converting
+ * a standard controls value to hardware specific values. All
+ * Standard passed in values are 0-100 as in percent.
+ *
+ * First it takes a value passed in as volume or gain and
+ * converts to attenuation or gain correspondingly. Since this is
+ * what the hardware needs.
+ *
+ * Second it adjusts the value passed in to compensate for the none
+ * linear nature of human hearing, sound loudness, sensitivity. It
+ * converts the linear value to a logarithmic value. This gives users
+ * the perception that the controls are linear.
+ *
+ * Third it converts the value to the number of bits that a hardware
+ * register needs to be.
+ *
+ * On input the following are supplied:
+ * left           - The gain or volume in percent for left channel.
+ * right          - The gain or volume in percent for right channel.
+ * bits           - The number of bits the hardware needs. If this value
+ *                  is negetive then right and left are gain else they
+ *                  are volume.
+ *
+ * On return the following is returned:
+ *
+ * bit: 15             8 7             0
+ *     ----------------------------------
+ *     | left channel   | right channel |
+ *     ----------------------------------
+ *      ( each channel is "bits" wide )
+ */
+uint16_t
+ac97_val_scale(int left, int right, int bits)
+{
+	ASSERT(left <= 100);
+	ASSERT(right <= 100);
+
+	if (bits < 0) {		/* This is gain not ATTN */
+		left = 100 - left;
+		right = 100 - right;
+		bits = -bits;
+	}
+
+#if 0
+	/*
+	 * 4Front's code used a table to smooth the transitions
+	 * somewhat.  Without this change, the volume levels adjusted
+	 * near the top of the table seem to have less effect.  Its
+	 * hard to notice a volume change from 100 to 95, without the
+	 * val_cvt table, for example.  However, the scaling has an
+	 * ugly side effect, which is at the default volumes (75%), we
+	 * wind up having the level set too high for some
+	 * codec/amplifier combinations.
+	 *
+	 * Legacy Sun code didn't have this table, and some
+	 * qualitative testing shows that it isn't really necessary.
+	 */
+	left = 100 - ac97_val_cvt[left];
+	right = 100 - ac97_val_cvt[right];
+#else
+	left = 100 - left;
+	right = 100 - right;
+#endif
+	return (((left * ((1 << bits) - 1) / 100) << 8) |
+	    (right * ((1 << bits) - 1) / 100));
+}
+
+uint16_t
+ac97_mono_scale(int val, int bits)
+{
+	ASSERT(val <= 100);
+
+	if (bits < 0) {		/* This is gain not ATTN */
+		bits = -bits;
+	} else {
+		val = 100 - val;	/* convert to attenuation */
+	}
+	return (val * ((1 << bits) - 1) / 100);
+}
+
+
+audio_dev_t *
+ac97_get_dev(ac97_t *ac)
+{
+	return (ac->d);
+}
+
+int
+ac97_get_prop(ac97_t *ac, char *prop, int defval)
+{
+	int	rv;
+
+	rv = ddi_prop_get_int(DDI_DEV_T_ANY, ac->dip, DDI_PROP_DONTPASS,
+	    prop, defval);
+	return (rv);
+}
+
+/*
+ * This calls the Hardware drivers access write routine
+ * to write to a device register.
+ */
+#define	WR(r, v)	(ac)->wr((ac)->private, (r), (v))
+#define	RD(r)		(ac)->rd((ac)->private, (r))
+
+/*
+ * Probe routines for optional controls
+ *
+ * These routines each probe one aspect of hardware
+ * for controls presents.
+ * If the control is present these routines should
+ * return none zero.
+ */
+
+/*
+ * Is the named register implemented?  This routine saves and
+ * restores the original value, and relies on the fact that the
+ * registers (if implemented) will have at least one bit that acts
+ * as a mute (0x8000, 0x8080), so we can probe "silently".
+ *
+ * The probe logic is suggested by the AC'97 2.3 spec.  (Unimplemented
+ * registers are required to return zero to facilitate this sort of
+ * detection.)
+ */
+static int
+ac97_probe_reg(ac97_t *ac, uint8_t reg)
+{
+	uint16_t	val;
+	int		rv = 0;
+
+	/* get the original value */
+	val = RD(reg);
+	WR(reg, 0xffff);
+	if (RD(reg) != 0) {
+		rv = 1;
+	}
+	/* restore the original value */
+	WR(reg, val);
+	return (rv);
+}
+
+/*
+ * Does this device have bass/treble controls?
+ */
+static int
+ac97_probe_tone(ac97_t *ac)
+{
+	/* Bass/Treble contols  present */
+	if (ac->caps & RR_BASS_TREBLE)
+		return (1);
+	else
+		return (0);
+}
+
+/*
+ * If there is a loudness switch?
+ */
+static int
+ac97_probe_loud(ac97_t *ac)
+{
+	/* loudness contol present */
+	if (ac->caps & RR_LOUDNESS_SUPPORT)
+		return (1);
+	else
+		return (0);
+}
+
+/*
+ * Does this device have a mono-mic input volume control?
+ */
+static int
+ac97_probe_mmic(ac97_t *ac)
+{
+	/* mono mic present */
+	if (ac->caps & RR_DEDICATED_MIC)
+		return (1);
+	else
+		return (0);
+}
+
+/*
+ * Does this device have a simulated stereo switch?
+ */
+static int
+ac97_probe_stsim(ac97_t *ac)
+{
+	/* simulated stereocontol present */
+	if (ac->caps & RR_PSEUDO_STEREO)
+		return (1);
+	else
+		return (0);
+}
+
+/*
+ * Does this device have a PC beeper input volume control?
+ */
+static int
+ac97_probe_pcbeep(ac97_t *ac)
+{
+	return (ac97_probe_reg(ac, AC97_PC_BEEP_REGISTER));
+}
+
+/*
+ * Does this device have AUX output port volume control?
+ */
+static int
+ac97_probe_rear(ac97_t *ac)
+{
+	if (ac->flags & AC97_FLAG_AUX_4CH)
+		return (1);
+	else
+		return (0);
+
+}
+
+/*
+ * Does this device have a mic?
+ */
+static int
+ac97_probe_mic(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_MIC)) &&
+	    (ac97_probe_reg(ac, AC97_MIC_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_MIC);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * If this device has an AUX output port is it used for headphones?
+ */
+static int
+ac97_probe_headphone(ac97_t *ac)
+{
+	/* headphone control present */
+	if ((ac->flags & AC97_FLAG_AUX_HP) &&
+	    !(ac->flags & AC97_FLAG_NO_HEADPHONE)) {
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have AUX output port volume control?
+ */
+static int
+ac97_probe_auxout(ac97_t *ac)
+{
+	/* ALT PCM control present */
+	if ((ac->flags & AC97_FLAG_AUX_LVL) &&
+	    !(ac->flags & AC97_FLAG_NO_AUXOUT)) {
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have an AUX input port volume control?
+ */
+static int
+ac97_probe_auxin(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_AUXIN)) &&
+	    (ac97_probe_reg(ac, AC97_AUX_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_AUXIN);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a phone input port with a volume control?
+ */
+static int
+ac97_probe_phone(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_PHONE)) &&
+	    (ac97_probe_reg(ac, AC97_PHONE_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_PHONE);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a mono output port with volume control?
+ */
+static int
+ac97_probe_mono(ac97_t *ac)
+{
+	if (!(ac->flags & AC97_FLAG_SPEAKER_OK)) {
+		return (0);
+	}
+	if (ac97_probe_reg(ac, AC97_MONO_MASTER_VOLUME_REGISTER)) {
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a line input port with volume control?
+ */
+static int
+ac97_probe_linein(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_LINEIN)) &&
+	    (ac97_probe_reg(ac, AC97_LINE_IN_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_LINEIN);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a cdrom input port with volume control?
+ */
+static int
+ac97_probe_cdrom(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_CDROM)) &&
+	    (ac97_probe_reg(ac, AC97_CD_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_CD);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a video input port with volume control?
+ */
+static int
+ac97_probe_video(ac97_t *ac)
+{
+	if ((!(ac->flags & AC97_FLAG_NO_VIDEO)) &&
+	    (ac97_probe_reg(ac, AC97_VIDEO_VOLUME_REGISTER))) {
+		ac->inputs |= (1U << INPUT_VIDEO);
+		return (1);
+	}
+	return (0);
+}
+
+/*
+ * Does this device have a 3D sound enhancement?
+ */
+static int
+ac97_probe_3d(ac97_t *ac)
+{
+	/* 3D control present */
+	if (ac->caps & RR_3D_STEREO_ENHANCE_MASK)
+		return (1);
+	else
+		return (0);
+}
+
+static int
+ac97_probe_3d_impl(ac97_t *ac, uint16_t mask)
+{
+	int	rv = 0;
+	uint16_t val;
+
+	if ((ac->caps & RR_3D_STEREO_ENHANCE_MASK) == 0)
+		return (0);
+
+	/* get the original value */
+	val = RD(AC97_THREE_D_CONTROL_REGISTER);
+	WR(AC97_THREE_D_CONTROL_REGISTER, mask);
+	if ((RD(AC97_THREE_D_CONTROL_REGISTER) & mask) != 0) {
+		rv = 1;
+	}
+	/* restore the original value */
+	WR(AC97_THREE_D_CONTROL_REGISTER, val);
+	return (rv);
+}
+
+static int
+ac97_probe_3d_depth(ac97_t *ac)
+{
+	return (ac97_probe_3d_impl(ac, TDCR_DEPTH_MASK));
+}
+
+static int
+ac97_probe_3d_center(ac97_t *ac)
+{
+	return (ac97_probe_3d_impl(ac, TDCR_CENTER_MASK));
+}
+
+/*
+ * Does this device have a center output port with volume control?
+ */
+static int
+ac97_probe_center(ac97_t *ac)
+{
+	uint16_t val;
+
+	val = RD(AC97_EXTENDED_AUDIO_REGISTER);
+
+	/* center volume present */
+	if (val & EAR_CDAC)
+		return (1);
+	else
+		return (0);
+}
+
+/*
+ * Does this device have a LFE (Sub-woofer) output port with
+ * a volume control?
+ */
+static int
+ac97_probe_lfe(ac97_t *ac)
+{
+	uint16_t val;
+
+	val = RD(AC97_EXTENDED_AUDIO_REGISTER);
+
+	/* We have LFE control */
+	if (val & EAR_LDAC)
+		return (1);
+	else
+		return (0);
+
+}
+
+/*
+ * Are we a multichannel codec?
+ */
+static int
+ac97_probe_front(ac97_t *ac)
+{
+	uint16_t val;
+
+	val = RD(AC97_EXTENDED_AUDIO_REGISTER);
+
+	/* Are any of the Surround, Center, or LFE dacs present? */
+	if (val & (EAR_SDAC | EAR_CDAC | EAR_LDAC))
+		return (1);
+	else
+		return (0);
+}
+
+static int
+ac97_probe_lineout(ac97_t *ac)
+{
+	/* if not multichannel, then use "lineout" instead of "front" label */
+	return (!ac97_probe_front(ac));
+}
+
+static const char *ac97_mics[] = {
+	AUDIO_PORT_MIC1,
+	AUDIO_PORT_MIC2,
+	NULL,
+};
+
+static const char *ac97_monos[] = {
+	AUDIO_PORT_MONOMIX,
+	AUDIO_PORT_MIC,
+	NULL
+};
+
+/*
+ * This calls the Hardware drivers access write routine
+ * to write to a device register.
+ */
+void
+ac97_wr(ac97_t *ac, uint8_t reg, uint16_t val)
+{
+	if ((reg < LAST_SHADOW_REG) && (reg > 0)) {
+		SHADOW(ac, reg) = val;
+	}
+
+	/*
+	 * Don't touch hardware _unless_ if we are suspended, unless we
+	 * are in the process of resuming.
+	 */
+	if ((!ac->suspended) || (ac->resumer == ddi_get_kt_did())) {
+		ac->wr(ac->private, reg, val);
+	}
+}
+
+/*
+ * This obtains the shadowed value of a register.  If the register is
+ * out of range, zero is returned.
+ *
+ * To read a hardware register, use the RD() macro above.
+ */
+uint16_t
+ac97_rd(ac97_t *ac, uint8_t reg)
+{
+	if ((reg < LAST_SHADOW_REG) && (reg > 0)) {
+		return (SHADOW(ac, reg));
+	}
+	if ((!ac->suspended) || (ac->resumer == ddi_get_kt_did())) {
+		return (ac->rd(ac->private, reg));
+	}
+	return (0);
+}
+
+/*
+ * This calls the hardware driver's access read/write routine
+ * to set bits in a device register.
+ */
+void
+ac97_set(ac97_t *ac, uint8_t reg, uint16_t val)
+{
+	ac97_wr(ac, reg, ac->rd(ac->private, reg) | val);
+}
+
+/*
+ * This calls the hardware driver's access read/write routine
+ * to clear bits in a device register.
+ */
+void
+ac97_clr(ac97_t *ac, uint8_t reg, uint16_t val)
+{
+	ac97_wr(ac, reg, ac->rd(ac->private, reg) & ~val);
+}
+
+/*
+ * Look for a control attached to this device based
+ * on its control number.
+ *
+ * If this control number is found the per controls state
+ * structure is returned.
+ */
+ac97_ctrl_t *
+ac97_control_find(ac97_t *ac, const char *name)
+{
+	ac97_ctrl_t *ctrl;
+	list_t *l = &ac->ctrls;
+
+	/* Validate that ctrlnum is real and usable */
+	mutex_enter(&ac->ac_lock);
+	for (ctrl = list_head(l); ctrl; ctrl = list_next(l, ctrl)) {
+		if (strcmp(ctrl->actrl_name, name) == 0) {
+			mutex_exit(&ac->ac_lock);
+			return (ctrl);
+		}
+	}
+	mutex_exit(&ac->ac_lock);
+	return (NULL);
+}
+
+/*
+ * This will update all the codec registers from the shadow table.
+ */
+static void
+ac97_restore(ac97_t *ac)
+{
+	/*
+	 * If we are restoring previous settings, just reload from the
+	 * shadowed settings.
+	 */
+	for (int i = 2; i < LAST_SHADOW_REG; i += sizeof (uint16_t)) {
+		ac->wr(ac->private, i, SHADOW(ac, i));
+	}
+
+	/*
+	 * Then go and do the controls.  This is important because some of
+	 * the controls might use registers that aren't shadowed.  Doing it
+	 * a second time also may help guarantee that it all works.
+	 */
+	for (ac97_ctrl_t *ctrl = list_head(&ac->ctrls); ctrl;
+	    ctrl = list_next(&ac->ctrls, ctrl)) {
+		ctrl->actrl_write_fn(ctrl, ctrl->actrl_value);
+	}
+}
+
+/*
+ * This will update all the hardware controls to the initial values at
+ * start of day.
+ */
+static void
+ac97_init_values(ac97_t *ac)
+{
+	ac97_ctrl_t	*ctrl;
+
+	mutex_enter(&ac->ac_lock);
+	for (ctrl = list_head(&ac->ctrls); ctrl;
+	    ctrl = list_next(&ac->ctrls, ctrl)) {
+		ctrl->actrl_value = ctrl->actrl_initval;
+		ctrl->actrl_write_fn(ctrl, ctrl->actrl_initval);
+	}
+	mutex_exit(&ac->ac_lock);
+}
+
+/*
+ * Select the input source for recording. This is the set routine
+ * for the control AUDIO_CONTROL_INPUTS.
+ */
+static void
+ac97_insrc_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_t		*ac = ctrl->actrl_ac97;
+	uint16_t	set_val;
+
+	set_val = ddi_ffs(value & 0xffff);
+	if ((set_val > 0) && (set_val <= 8)) {
+		set_val--;
+		ac97_wr(ac, AC97_RECORD_SELECT_CTRL_REGISTER,
+		    set_val | (set_val << 8));
+	}
+}
+
+static void
+ac97_gpr_toggle(ac97_ctrl_t *ctrl, int bit, uint64_t onoff)
+{
+	ac97_t			*ac = ctrl->actrl_ac97;
+	uint16_t		v;
+
+	v = SHADOW(ac, AC97_GENERAL_PURPOSE_REGISTER);
+	if (onoff) {
+		v |= bit;
+	} else {
+		v &= ~bit;
+	}
+	ac97_wr(ac, AC97_GENERAL_PURPOSE_REGISTER, v);
+}
+
+static void
+ac97_3donoff_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_3D_STEREO_ENHANCE, value);
+}
+
+static void
+ac97_loudness_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_BASS_BOOST, value);
+}
+
+static void
+ac97_loopback_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_LPBK, value);
+}
+
+/*
+ * This will set simulated stereo control to on or off.
+ */
+static void
+ac97_stsim_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_ST, value);
+}
+
+/*
+ * This will set mic select control to mic1=0 or mic2=1.
+ */
+static void
+ac97_selmic_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_MS_MIC2, value & 2);
+}
+
+/*
+ * This will set mono source select control to mix=0 or mic=1.
+ */
+static void
+ac97_monosrc_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_gpr_toggle(ctrl, GPR_MONO_MIC_IN, value & 2);
+}
+
+static void
+ac97_stereo_set(ac97_ctrl_t *ctrl, uint64_t value, uint8_t reg)
+{
+	ac97_t			*ac = ctrl->actrl_ac97;
+	uint8_t			left, right;
+	uint16_t		mute;
+
+	left = (value >> 8) & 0xff;
+	right = value & 0xff;
+	mute = value ? 0 : ctrl->actrl_muteable;
+
+	ac97_wr(ac, reg, ac97_val_scale(left, right, ctrl->actrl_bits) | mute);
+}
+
+static void
+ac97_mono_set(ac97_ctrl_t *ctrl, uint64_t value, uint8_t reg, int shift)
+{
+	ac97_t			*ac = ctrl->actrl_ac97;
+	uint8_t			val;
+	uint16_t		mute, v;
+	uint16_t		mask;
+
+	val = value & 0xff;
+	mute = val ? 0 : ctrl->actrl_muteable;
+
+	mask = ctrl->actrl_muteable |
+	    (((1 << ABS(ctrl->actrl_bits)) - 1) << shift);
+
+	v = SHADOW(ac, reg);
+	v &= ~mask;	/* clear all of our bits, preserve others */
+
+	/* now set the mute bit, and volume bits */
+	v |= mute;
+	v |= (ac97_mono_scale(val, ctrl->actrl_bits) << shift);
+
+	ac97_wr(ac, reg, v);
+}
+
+static void
+ac97_master_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	value = value | (value << 8);
+	ac97_stereo_set(ctrl, value, AC97_PCM_OUT_VOLUME_REGISTER);
+}
+
+static void
+ac97_lineout_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_MASTER_VOLUME_REGISTER);
+}
+
+static void
+ac97_surround_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_EXTENDED_LRS_VOLUME_REGISTER);
+}
+
+static void
+ac97_aux1out_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_HEADPHONE_VOLUME_REGISTER);
+}
+
+static void
+ac97_headphone_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_HEADPHONE_VOLUME_REGISTER);
+}
+
+static void
+ac97_cd_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_CD_VOLUME_REGISTER);
+}
+
+static void
+ac97_video_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_VIDEO_VOLUME_REGISTER);
+}
+
+static void
+ac97_auxin_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_AUX_VOLUME_REGISTER);
+}
+
+static void
+ac97_linein_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_LINE_IN_VOLUME_REGISTER);
+}
+
+/*
+ * This will set mono mic gain control.
+ */
+static void
+ac97_monomic_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_RECORD_GAIN_MIC_REGISTER, 0);
+}
+
+static void
+ac97_phone_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_PHONE_VOLUME_REGISTER, 0);
+}
+
+static void
+ac97_mic_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_MIC_VOLUME_REGISTER, 0);
+}
+
+static void
+ac97_speaker_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_MONO_MASTER_VOLUME_REGISTER, 0);
+}
+
+static void
+ac97_pcbeep_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_PC_BEEP_REGISTER, 1);
+}
+
+static void
+ac97_recgain_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_stereo_set(ctrl, value, AC97_RECORD_GAIN_REGISTER);
+}
+
+static void
+ac97_center_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_EXTENDED_C_LFE_VOLUME_REGISTER, 0);
+}
+
+static void
+ac97_lfe_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_EXTENDED_C_LFE_VOLUME_REGISTER, 8);
+}
+
+static void
+ac97_bass_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_MASTER_TONE_CONTROL_REGISTER, 8);
+}
+
+static void
+ac97_treble_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_mono_set(ctrl, value, AC97_MASTER_TONE_CONTROL_REGISTER, 0);
+}
+
+static void
+ac97_3ddepth_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	/*
+	 * XXX: This is all wrong... 3D depth/center cannot necessarily
+	 * be scaled, because the technology in use may vary.  We
+	 * need more information about each of the options available
+	 * to do the right thing.
+	 */
+	ac97_mono_set(ctrl, value, AC97_THREE_D_CONTROL_REGISTER, 0);
+}
+
+static void
+ac97_3dcent_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	/*
+	 * XXX: This is all wrong... 3D depth/center cannot necessarily
+	 * be scaled, because the technology in use may vary.  We
+	 * need more information about each of the options available
+	 * to do the right thing.
+	 */
+	ac97_mono_set(ctrl, value, AC97_THREE_D_CONTROL_REGISTER, 8);
+}
+
+static void
+ac97_micboost_set(ac97_ctrl_t *ctrl, uint64_t value)
+{
+	ac97_t		*ac = ctrl->actrl_ac97;
+	uint16_t	v;
+
+	v = SHADOW(ac, AC97_MIC_VOLUME_REGISTER);
+	if (value) {
+		v |= MICVR_20dB_BOOST;
+	} else {
+		v &= ~MICVR_20dB_BOOST;
+	}
+	ac97_wr(ac, AC97_MIC_VOLUME_REGISTER, v);
+}
+
+/*
+ * This will return the stored value for any control that has been set.
+ * Note this does not return the actual hardware value from a port. But
+ * instead returns the cached value from the last write to the hardware
+ * port.
+ *
+ * arg            - This control structure for this control.
+ * value          - This is a pointer to the location to put the
+ *                  controls value.
+ *
+ * On success zero is returned.
+ */
+static int
+ac97_control_get(void *arg, uint64_t *value)
+{
+	ac97_ctrl_t	*ctrl = arg;
+	ac97_t		*ac = ctrl->actrl_ac97;
+
+	mutex_enter(&ac->ac_lock);
+	*value = ctrl->actrl_value;
+	mutex_exit(&ac->ac_lock);
+
+	return (0);
+}
+
+static int
+ac97_control_set(void *arg, uint64_t value)
+{
+	ac97_ctrl_t		*ctrl = arg;
+	ac97_t			*ac = ctrl->actrl_ac97;
+	uint8_t			v1, v2;
+
+	/* a bit of quick checking */
+	switch (ctrl->actrl_type) {
+	case AUDIO_CTRL_TYPE_STEREO:
+		v1 = (value >> 8) & 0xff;
+		v2 = value & 0xff;
+		if ((v1 < ctrl->actrl_minval) || (v1 > ctrl->actrl_maxval) ||
+		    (v2 < ctrl->actrl_minval) || (v2 > ctrl->actrl_maxval) ||
+		    (value > 0xffff)) {
+			return (EINVAL);
+		}
+		break;
+
+	case AUDIO_CTRL_TYPE_ENUM:
+		if ((value & ~ctrl->actrl_minval) !=
+		    (ctrl->actrl_maxval & ~ctrl->actrl_minval)) {
+			return (EINVAL);
+		}
+		break;
+
+	case AUDIO_CTRL_TYPE_MONO:
+	case AUDIO_CTRL_TYPE_BOOLEAN:
+		if ((value < ctrl->actrl_minval) ||
+		    (value > ctrl->actrl_maxval)) {
+			return (EINVAL);
+		}
+		break;
+	}
+
+	mutex_enter(&ac->ac_lock);
+	ctrl->actrl_value = value;
+	ctrl->actrl_write_fn(ctrl, value);
+	mutex_exit(&ac->ac_lock);
+
+	return (0);
+}
+
+/*
+ * This simply sets a flag to block calls to the underlying
+ * hardware driver to get or set hardware controls. This is usually
+ * called just before a power down of devices. Once this gets called any
+ * calls to set controls will not touch the real hardware. But
+ * since all control updates are always saved in soft registers it
+ * is a simple mater to update the hardware with the latest values
+ * on resume which also unblocks calls to the hardware controls.
+ */
+void
+ac97_suspend(ac97_t *ac)
+{
+	mutex_enter(&ac->ac_lock);
+
+	/* This will prevent any new operations from starting! */
+	ac->suspended = B_TRUE;
+	ac->resumer = 0;
+
+	/* XXX - should we powerdown codec's here?? */
+	mutex_exit(&ac->ac_lock);
+}
+
+/*
+ * Reset the analog codec hardware
+ *
+ * Reset all analog AC97 hardware, input ADC's, output DAC's and MIXER.
+ * Wait a resonable amount of time for hardware to become ready.
+ */
+static void
+ac97_analog_reset(ac97_t *ac)
+{
+	uint16_t	tmp;
+	int		wait = 1000; /* delay for up to 1s */
+
+	/* Clear stale data and resync register accesses */
+	tmp = RD(AC97_POWERDOWN_CTRL_STAT_REGISTER);
+
+	/* reset the codec */
+	WR(AC97_RESET_REGISTER, 0);
+	tmp = RD(AC97_RESET_REGISTER);
+
+	/* power up */
+	WR(AC97_POWERDOWN_CTRL_STAT_REGISTER, 0);
+
+	/* Wait for ADC/DAC/MIXER to become ready */
+	while (wait--) {
+		/* 1 msec delay */
+		drv_usecwait(1000);
+
+		/* If all ready - end delay */
+		tmp = RD(AC97_POWERDOWN_CTRL_STAT_REGISTER);
+		SHADOW(ac, AC97_POWERDOWN_CTRL_STAT_REGISTER) = tmp;
+		if ((tmp & PCSR_POWERD_UP) == PCSR_POWERD_UP) {
+			return;
+		}
+	}
+
+	audio_dev_warn(ac->d, "AC'97 analog powerup timed out");
+}
+
+/*
+ * This is the internal hardware reset routine.
+ * It has no locking and we must be locked before it is
+ * called!
+ *
+ * This will reset and re-initialize the device.
+ * It has two modes of operation that affect how it handles
+ * all controls.
+ *
+ * It re-initializes the device and reloads values with
+ * last updated versions.
+ */
+static void
+ac97_hw_reset(ac97_t *ac)
+{
+	/*
+	 * Fully Power up the device
+	 */
+	if (ac->flags & AC97_FLAG_AMPLIFIER) {
+		/* power up - external amp powerd up */
+		ac97_wr(ac, AC97_POWERDOWN_CTRL_STAT_REGISTER, 0);
+	} else {
+		/* power up - external amp powered down */
+		ac97_wr(ac, AC97_POWERDOWN_CTRL_STAT_REGISTER, PCSR_EAPD);
+	}
+
+	ac97_wr(ac, AC97_GENERAL_PURPOSE_REGISTER, 0);
+
+	switch (ac->vid) {
+	case AC97_CODEC_STAC9708:
+#if 0
+		/* non-inverted phase */
+		/* ac97_rd(ac, AC97_VENDOR_REGISTER_11) & ~0x8); */
+#endif
+		WR(AC97_VENDOR_REGISTER_11, 8);
+		break;
+
+	case AC97_CODEC_EM28028:
+		ac97_wr(ac, AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER,
+		    (ac97_rd(ac, AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER) &
+		    ~3800) | 0xE0);
+		break;
+
+	case AC97_CODEC_AD1886:
+		/* jack sense */
+		WR(AC97_VENDOR_REGISTER_13,
+		    (RD(AC97_VENDOR_REGISTER_13) & ~0xEF) | 0x10);
+		break;
+
+	case AC97_CODEC_AD1888:
+		WR(AC97_VENDOR_REGISTER_15, 0xC420);
+#if 0
+		/* GED: This looks fishy to me, so I'm nuking it for now */
+		/* headphone/aux volume (?) */
+		ac97_wr(ac, AC97_HEADPHONE_VOLUME_REGISTER,  0x0808);
+#endif
+		break;
+
+	case AC97_CODEC_AD1980:
+#if 0
+		/* set jacksense to mute line if headphone is plugged */
+		WR(AC97_VENDOR_REGISTER_13,
+		    (RD(AC97_VENDOR_REGISTER_13) & ~0xe00) | 0x400);
+#endif
+		WR(AC97_VENDOR_REGISTER_15, 0xC420);
+		break;
+
+	case AC97_CODEC_AD1985:
+		WR(AC97_VENDOR_REGISTER_15, 0xC420);
+		break;
+
+	case AC97_CODEC_WM9704:
+		/* enable I2S */
+		WR(AC97_VENDOR_REGISTER_01, RD(AC97_VENDOR_REGISTER_01) | 0x80);
+		break;
+
+	case AC97_CODEC_VT1612A:
+	case AC97_CODEC_VT1617A:
+	case AC97_CODEC_VT1616:
+		/* Turn off Center, Surround, and LFE DACs */
+		ac97_clr(ac, AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER,
+		    EASCR_PRI | EASCR_PRJ | EASCR_PRK);
+		WR(AC97_VENDOR_REGISTER_01, 0x0230);
+		break;
+
+	case AC97_CODEC_YMF753:
+		/* set TX8 + 3AWE */
+		WR(AC97_VENDOR_REGISTER_07, RD(AC97_VENDOR_REGISTER_07) | 0x9);
+		break;
+
+	default:
+		break;
+	}
+
+	/* call codec specific reset hook */
+	if (ac->codec_reset != NULL) {
+		ac->codec_reset(ac);
+	}
+
+	/* Turn off variable sampling rate support */
+	ac97_clr(ac, AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER, EASCR_VRA);
+}
+
+/*
+ * This will reset and re-initialize the device.
+ * It has two modes of operation that affect how it handles
+ * all controls.
+ *
+ * It re-initializes the device and then can either reset
+ * all controls back to their initial values or it can
+ * re-load all controls with their last updated values.
+ *
+ * initval         - If this is none zero then all controls will
+ *                   be restored to their initial values.
+ */
+void
+ac97_reset(ac97_t *ac)
+{
+	/* If we are about to suspend so no point in going on */
+	mutex_enter(&ac->ac_lock);
+	if (ac->suspended) {
+		mutex_exit(&ac->ac_lock);
+		return;
+	}
+	ac97_analog_reset(ac);
+	ac97_hw_reset(ac);
+	ac97_restore(ac);
+
+	mutex_exit(&ac->ac_lock);
+}
+
+/*
+ * Given the need to resume the hardware this reloads the base hardware
+ * and then takes the stored values for each control and sends them
+ * to the hardware again.
+ */
+void
+ac97_resume(ac97_t *ac)
+{
+
+	/*
+	 * This should only be called when already suspended.
+	 * this takes us out of suspend state after it brings the
+	 * controls back to life.
+	 */
+	ASSERT(ac->suspended);
+	mutex_enter(&ac->ac_lock);
+	ac->resumer = ddi_get_kt_did();
+
+	/* We simply call reset since the operation is the same */
+	ac97_analog_reset(ac);
+	ac97_hw_reset(ac);
+	ac97_restore(ac);
+
+	ac->resumer = 0;
+	ac->suspended = B_FALSE;
+	mutex_exit(&ac->ac_lock);
+}
+
+
+/*
+ * Register a control -- if it fails, it will generate a message to
+ * syslog, but the driver muddles on.  (Failure to register a control
+ * should never occur, and is generally benign if it happens.)
+ */
+void
+ac97_alloc_control(ac97_t *ac, ac97_ctrl_probe_t *cpt)
+{
+	ac97_ctrl_t		*ctrl;
+	audio_ctrl_desc_t	ctrl_des;
+
+	ASSERT(ac);
+	ASSERT(ac->d);
+
+	ctrl = kmem_zalloc(sizeof (ac97_ctrl_t), KM_SLEEP);
+
+	bzero(&ctrl_des, sizeof (ctrl_des));
+	ctrl_des.acd_name = cpt->cp_name;
+	ctrl_des.acd_minvalue = cpt->cp_minval;
+	ctrl_des.acd_maxvalue = cpt->cp_maxval;
+	ctrl_des.acd_type = cpt->cp_type;
+	ctrl_des.acd_flags = cpt->cp_flags;
+	if (cpt->cp_enum) {
+		for (int e = 0; e < 64; e++) {
+			if (cpt->cp_enum[e] == NULL)
+				break;
+			ctrl_des.acd_enum[e] = cpt->cp_enum[e];
+		}
+	}
+
+	ctrl->actrl_ac97 = ac;
+	/*
+	 * Warning for extended controls this field gets changed
+	 * by audio_dev_add_control() to be a unique value.
+	 */
+	ctrl->actrl_minval = cpt->cp_minval;
+	ctrl->actrl_maxval = cpt->cp_maxval;
+	ctrl->actrl_initval = cpt->cp_initval;
+	ctrl->actrl_muteable = cpt->cp_muteable;
+	ctrl->actrl_write_fn = cpt->cp_write_fn;
+	ctrl->actrl_bits = cpt->cp_bits;
+	ctrl->actrl_type = cpt->cp_type;
+	ctrl->actrl_name = cpt->cp_name;
+
+	/* Register control with framework */
+	ctrl->actrl_ctrl = audio_dev_add_control(ac->d, &ctrl_des,
+	    ac97_control_get, ac97_control_set, ctrl);
+	if (!ctrl->actrl_ctrl) {
+		audio_dev_warn(ac->d, "AC97 %s alloc failed", cpt->cp_name);
+		kmem_free(ctrl, sizeof (ac97_ctrl_t));
+		return;
+	}
+
+	/*
+	 * Not that it can not be referenced until in is in the
+	 * list. So again by adding to the list last we avoid the need
+	 * for control locks.
+	 */
+	mutex_enter(&ac->ac_lock);
+	list_insert_tail(&ac->ctrls, ctrl);
+	mutex_exit(&ac->ac_lock);
+}
+
+/*
+ * De-Register and free up a control
+ *
+ * Note ctrl_lock read write must be held for writing when calling
+ * this function
+ */
+void
+ac97_free_control(ac97_ctrl_t *ctrl)
+{
+	ac97_t	*ac = ctrl->actrl_ac97;
+
+	mutex_enter(&ac->ac_lock);
+	list_remove(&ac->ctrls, ctrl);
+	mutex_exit(&ac->ac_lock);
+
+	audio_dev_del_control(ctrl->actrl_ctrl);
+	kmem_free(ctrl, sizeof (ac97_ctrl_t));
+}
+
+/*
+ * This is the master list of all controls known and handled by
+ * the AC97 framework. This is the list used to probe, allocate
+ * and configure controls. If a control is not in this list it
+ * will not be handled. If a control is in this list but does not
+ * have a probe routine then it will always be included. If a
+ * control in list has a probe routine then it must return true
+ * for that control to be included.
+ */
+
+#define	MONCTL	(AC97_FLAGS | AUDIO_CTRL_FLAG_MONITOR)
+#define	PLAYCTL	(AC97_FLAGS | AUDIO_CTRL_FLAG_PLAY)
+#define	RECCTL	(AC97_FLAGS | AUDIO_CTRL_FLAG_REC)
+#define	T3DCTL	(AC97_FLAGS | AUDIO_CTRL_FLAG_3D)
+#define	TONECTL	(AC97_FLAGS | AUDIO_CTRL_FLAG_TONE)
+#define	MAINVOL	(PLAYCTL | AUDIO_CTRL_FLAG_MAINVOL)
+#define	PCMVOL	(PLAYCTL | AUDIO_CTRL_FLAG_PCMVOL)
+#define	RECVOL	(RECCTL | AUDIO_CTRL_FLAG_RECVOL)
+#define	MONVOL	(MONCTL | AUDIO_CTRL_FLAG_MONVOL)
+
+ac97_ctrl_probe_t	ctrl_probe_tbl[] = {
+
+	/* Master PCM Volume */
+	{AUDIO_CTRL_ID_VOLUME, INIT_VAL_MAIN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	PCMVOL, PCMOVR_MUTE, ac97_master_set, NULL, 5},
+
+	/* LINE out volume */
+	{AUDIO_CTRL_ID_LINEOUT, INIT_VAL_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MAINVOL, 0x8080, ac97_lineout_set, ac97_probe_lineout, 6},
+
+	/* Front volume */
+	{AUDIO_CTRL_ID_FRONT, INIT_VAL_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MAINVOL, 0x8080, ac97_lineout_set, ac97_probe_front, 6},
+
+	/* 4CH out volume (has one of three possible uses, first use) */
+	{AUDIO_CTRL_ID_SURROUND, INIT_VAL_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MAINVOL, 0x8080, ac97_surround_set, ac97_probe_rear, 6},
+
+	/* ALT out volume (has one of three possible uses, second use) */
+	{AUDIO_CTRL_ID_HEADPHONE, INIT_VAL_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MAINVOL, 0x8080, ac97_headphone_set, ac97_probe_headphone, 6},
+
+	/* ALT out volume (has one of three possible uses, third use) */
+	{AUDIO_CTRL_ID_AUX1OUT, INIT_VAL_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MAINVOL, 0x8080, ac97_aux1out_set, ac97_probe_auxout, 6},
+
+	/* center out volume */
+	{AUDIO_CTRL_ID_CENTER, INIT_VAL_MN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	MAINVOL, EXLFEVR_CENTER_MUTE, ac97_center_set, ac97_probe_center, 6},
+
+	/* LFE out volume (sub-woofer) */
+	{AUDIO_CTRL_ID_LFE, INIT_VAL_MN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	MAINVOL, EXLFEVR_LFE_MUTE, ac97_lfe_set, ac97_probe_lfe, 6},
+
+	/* MONO out volume */
+	{AUDIO_CTRL_ID_SPEAKER, INIT_VAL_MN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	MAINVOL, MMVR_MUTE, ac97_speaker_set, ac97_probe_mono, 6},
+
+	/* Record in GAIN */
+	{AUDIO_CTRL_ID_RECGAIN, INIT_IGAIN_ST, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	RECVOL, RGR_MUTE, ac97_recgain_set, NULL, -4},
+
+	/* MIC in volume */
+	{AUDIO_CTRL_ID_MIC, 0, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MONVOL, MICVR_MUTE, ac97_mic_set, ac97_probe_mic, 5},
+
+	/* LINE in volume */
+	{AUDIO_CTRL_ID_LINEIN, 0, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MONVOL, LIVR_MUTE, ac97_linein_set, ac97_probe_linein, 5},
+
+	/* CD in volume */
+	{AUDIO_CTRL_ID_CD, 0, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MONVOL, CDVR_MUTE, ac97_cd_set, ac97_probe_cdrom, 5},
+
+	/* VIDEO in volume */
+	{AUDIO_CTRL_ID_VIDEO, 0, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MONVOL, VIDVR_MUTE, ac97_video_set, ac97_probe_video, 5},
+
+	/* AUX in volume */
+	{AUDIO_CTRL_ID_AUX1IN, 0, 0, 100, AUDIO_CTRL_TYPE_STEREO,
+	MONVOL, AUXVR_MUTE, ac97_auxin_set, ac97_probe_auxin, 5},
+
+	/* PHONE in volume */
+	{AUDIO_CTRL_ID_PHONE, 0, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	MONVOL, PVR_MUTE, ac97_phone_set, ac97_probe_phone, 5},
+
+	/* PC BEEPER in volume (motherboard speaker pins) */
+	{AUDIO_CTRL_ID_BEEP, INIT_VAL_MN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	AC97_RW, PCBR_MUTE, ac97_pcbeep_set, ac97_probe_pcbeep, 4},
+
+	/* BASS out level (note, zero is hardware bypass) */
+	{AUDIO_CTRL_ID_BASS, 0, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	TONECTL, 0, ac97_bass_set, ac97_probe_tone, 4},
+
+	/* TREBLE out level (note, zero is hardware bypass) */
+	{AUDIO_CTRL_ID_TREBLE, 0, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	TONECTL, 0, ac97_treble_set, ac97_probe_tone, 4},
+
+	/* Loudness on/off switch */
+	{AUDIO_CTRL_ID_LOUDNESS, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+	TONECTL, 0, ac97_loudness_set, ac97_probe_loud, 0},
+
+	/* 3D depth out level */
+	{AUDIO_CTRL_ID_3DDEPTH, 0, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	T3DCTL, 0, ac97_3ddepth_set, ac97_probe_3d_depth, 4},
+
+	/* 3D center out level */
+	{AUDIO_CTRL_ID_3DCENT, 0, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	T3DCTL, 0, ac97_3dcent_set, ac97_probe_3d_center, 4},
+
+	/* 3D enhance on/off switch */
+	{AUDIO_CTRL_ID_3DENHANCE, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+	T3DCTL, 0, ac97_3donoff_set, ac97_probe_3d, 0},
+
+	/* MIC BOOST switch */
+	{AUDIO_CTRL_ID_MICBOOST, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+	RECCTL, 0, ac97_micboost_set, ac97_probe_mic, 0},
+
+	/* Loopback on/off switch */
+	{AUDIO_CTRL_ID_LOOPBACK, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+	AC97_RW, 0, ac97_loopback_set, NULL, 0},
+
+	/*
+	 * The following selectors *must* come after the others, as they rely
+	 * on the probe results of other controls.
+	 */
+	/* record src select  (only one port at a time) */
+	{AUDIO_CTRL_ID_RECSRC, (1U << INPUT_MIC), 0, 0, AUDIO_CTRL_TYPE_ENUM,
+	RECCTL, 0, ac97_insrc_set, NULL, 0, ac97_insrcs},
+
+	/* Start of non-standard private controls */
+
+	/* Simulated stereo on/off switch */
+	{AUDIO_CTRL_ID_STEREOSIM, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+	AC97_RW, 0, ac97_stsim_set, ac97_probe_stsim, 0},
+
+	/* mono MIC GAIN */
+	{AUDIO_CTRL_ID_MICGAIN, INIT_IGAIN_MN, 0, 100, AUDIO_CTRL_TYPE_MONO,
+	RECCTL, RGMR_MUTE, ac97_monomic_set, ac97_probe_mmic, -4},
+
+	/* MIC select switch 0=mic1 1=mic2 */
+	{AUDIO_CTRL_ID_MICSRC, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM,
+	RECCTL, 0, ac97_selmic_set, ac97_probe_mic, 0, ac97_mics},
+
+	/* MONO out src select 0=mix 1=mic */
+	{AUDIO_CTRL_ID_SPKSRC, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM,
+	AC97_RW, 0, ac97_monosrc_set, ac97_probe_mono, 0, ac97_monos},
+
+	{NULL}
+};
+
+/*
+ * Probe all possible controls and register existing
+ * ones and set initial values
+ *
+ * Returns zero on success
+ */
+static int
+ac97_probeinit_ctrls(ac97_t *ac, int vol_bits, int enh_bits)
+{
+	ac97_ctrl_probe_t	*cpt;
+	ac97_ctrl_probe_t	my_cpt;
+
+	ASSERT(ac);
+
+	/*
+	 * Set some ports which are always present.
+	 */
+	ac->inputs = (1U << INPUT_STEREOMIX) | (1U << INPUT_MONOMIX);
+	for (cpt = &ctrl_probe_tbl[0]; cpt->cp_name != NULL; cpt++) {
+		bcopy(cpt, &my_cpt, sizeof (my_cpt));
+
+		if (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_RECSRC) == 0) {
+			my_cpt.cp_minval |= ac->inputs;
+			my_cpt.cp_maxval |= ac->inputs;
+		}
+
+		if (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_MICBOOST) == 0) {
+			if (ac->flags & AC97_FLAG_MICBOOST)
+				my_cpt.cp_initval = 1;
+		}
+
+		if ((strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_FRONT) == 0) ||
+		    (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_HEADPHONE) == 0) ||
+		    (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_SURROUND) == 0) ||
+		    (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_SPEAKER) == 0)) {
+			my_cpt.cp_bits = vol_bits;
+		}
+
+		if ((strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_3DDEPTH) == 0) ||
+		    (strcmp(my_cpt.cp_name, AUDIO_CTRL_ID_3DCENT) == 0)) {
+			my_cpt.cp_bits = enh_bits;
+		}
+
+		if (!my_cpt.cp_probe || my_cpt.cp_probe(ac)) {
+			ac97_alloc_control(ac, &my_cpt);
+		}
+	}
+
+	if (ac->codec_init != NULL) {
+		ac->codec_init(ac);
+	}
+
+	return (0);
+}
+
+/*
+ * Allocate an AC97 instance for use by a hardware driver.
+ *
+ * returns an allocated and initialize ac97 structure.
+ */
+ac97_t *
+ac97_alloc(dev_info_t *dip, ac97_rd_t rd, ac97_wr_t wr, void *priv)
+{
+	ac97_t	*ac;
+
+	ac = kmem_zalloc(sizeof (ac97_t), KM_SLEEP);
+	ac->dip = dip;
+	ac->rd = rd;
+	ac->wr = wr;
+	ac->private = priv;
+
+	list_create(&ac->ctrls, sizeof (struct ac97_ctrl),
+	    offsetof(struct ac97_ctrl, actrl_linkage));
+
+	mutex_init(&ac->ac_lock, NULL, MUTEX_DRIVER, NULL);
+
+#define	PROP_FLAG(prop, flag, def)				    \
+	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS, \
+	    (prop), (def))) {					    \
+		ac->flags |= (flag);				    \
+	} else {						    \
+		ac->flags &= ~(flag);				    \
+	}
+
+	/*
+	 * Engage the external amplifier by default, suppress with
+	 * a property of the form "ac97-amplifier=0".
+	 */
+	PROP_FLAG(AC97_PROP_AMPLIFIER, AC97_FLAG_AMPLIFIER, 1);
+
+	/*
+	 * We cannot necessarily know if the headphone jack is present
+	 * or not.  There's a technique to probe the codec for
+	 * headphone support, but many vendors seem to simply hang the
+	 * headphone jack on the line out circuit, and have some kind
+	 * of jack sense detection to enable or disable it by default.
+	 * None of this is visible in the AC'97 registers.
+	 *
+	 * We cannot do much about it, but what we can do is offer users
+	 * a way to suppress the option for a headphone port.  Users and
+	 * administrators can then set a flag in the driver.conf to suppress
+	 * the option from display.
+	 *
+	 * It turns out that this problem exists for other signals as
+	 * well.
+	 */
+	PROP_FLAG(AC97_PROP_NO_HEADPHONE, AC97_FLAG_NO_HEADPHONE, 0);
+	PROP_FLAG(AC97_PROP_NO_AUXOUT, AC97_FLAG_NO_AUXOUT, 0);
+	PROP_FLAG(AC97_PROP_NO_CDROM, AC97_FLAG_NO_CDROM, 0);
+	PROP_FLAG(AC97_PROP_NO_AUXIN, AC97_FLAG_NO_AUXIN, 0);
+	PROP_FLAG(AC97_PROP_NO_VIDEO, AC97_FLAG_NO_VIDEO, 0);
+	PROP_FLAG(AC97_PROP_NO_LINEIN, AC97_FLAG_NO_LINEIN, 0);
+	PROP_FLAG(AC97_PROP_NO_MIC, AC97_FLAG_NO_MIC, 0);
+
+	/*
+	 * Most SPARC systems use the AC97 monoaural output for the
+	 * built-in speaker.  On these systems, we want to expose and
+	 * enable the built-in speaker by default.
+	 *
+	 * On most x86 systems, the mono output is not connected to
+	 * anything -- the AC'97 spec makes it pretty clear that the
+	 * output was actually intended for use with speaker phones.
+	 * So on those systems, we really don't want to activate the
+	 * speaker -- we don't even want to expose it's presence
+	 * normally.
+	 *
+	 * However, there could be an exception to the rule here.  To
+	 * facilitate this, we allow for the presence of the property
+	 * to indicate that the speaker should be exposed.  Whether it
+	 * is enabled by default or not depends on the value of the
+	 * property.  (Generally on SPARC, we enable by default.  On
+	 * other systems we do not.)
+	 */
+#ifdef	__sparc
+	ac->flags |= AC97_FLAG_SPEAKER_OK;
+	PROP_FLAG(AC97_PROP_SPEAKER, AC97_FLAG_SPEAKER, 1);
+#else
+	if (ddi_prop_exists(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
+	    AC97_PROP_SPEAKER)) {
+		ac->flags |= AC97_FLAG_SPEAKER_OK;
+		PROP_FLAG(AC97_PROP_SPEAKER, AC97_FLAG_SPEAKER, 0);
+	}
+#endif
+
+	/*
+	 * Enable microphone boost (20dB normally) by default?
+	 */
+	PROP_FLAG(AC97_PROP_MICBOOST, AC97_FLAG_MICBOOST, 0);
+
+	return (ac);
+}
+
+/*
+ * Free an AC97 instance.
+ */
+void
+ac97_free(ac97_t *ac)
+{
+	ac97_ctrl_t *ctrl;
+
+	/* Clear out any controls that are still attached */
+	while ((ctrl = list_head(&ac->ctrls)) != NULL) {
+		ac97_free_control(ctrl);
+	}
+
+	list_destroy(&ac->ctrls);
+	mutex_destroy(&ac->ac_lock);
+	kmem_free(ac, sizeof (ac97_t));
+}
+
+static struct vendor {
+	unsigned	id;
+	const char	*name;
+} vendors[] = {
+	{ AC97_VENDOR_ADS,	"Analog Devices" },
+	{ AC97_VENDOR_AKM,	"Asahi Kasei" },
+	{ AC97_VENDOR_ALC,	"Realtek" },
+	{ AC97_VENDOR_ALG,	"Avance Logic" },
+	{ AC97_VENDOR_CMI,	"C-Media" },
+	{ AC97_VENDOR_CRY,	"Cirrus Logic" },
+	{ AC97_VENDOR_CXT,	"Conexant" },
+	{ AC97_VENDOR_ESS,	"ESS Technology" },
+	{ AC97_VENDOR_EV,	"Ectiva" },
+	{ AC97_VENDOR_ICE,	"ICEnsemble" },
+	{ AC97_VENDOR_ST,	"SigmaTel" },
+	{ AC97_VENDOR_TRA,	"TriTech", },
+	{ AC97_VENDOR_VIA,	"VIA Technologies" },
+	{ AC97_VENDOR_WML,	"Wolfson" },
+	{ AC97_VENDOR_YMH,	"Yamaha" },
+	{ 0, NULL },
+};
+
+static struct codec {
+	unsigned	id;
+	const char	*name;
+	int		enh_bits;
+	void		(*init)(ac97_t *ac);
+	void		(*reset)(ac97_t *ac);
+} codecs[] = {
+	{ AC97_CODEC_AK4540,	"AK4540" },
+	{ AC97_CODEC_STAC9700,	"STAC9700" },
+	{ AC97_CODEC_STAC9701,  "STAC9701" },
+	{ AC97_CODEC_STAC9701_2,	"STAC9701" },
+	{ AC97_CODEC_STAC9704,	"STAC9704" },
+	{ AC97_CODEC_STAC9705,	"STAC9705" },
+	{ AC97_CODEC_STAC9721,	"STAC9721" },
+	{ AC97_CODEC_STAC9708,	"STAC9708", 2 },
+	{ AC97_CODEC_STAC9744,	"STAC9744" },
+	{ AC97_CODEC_STAC9750,	"STAC9750", 3 },
+	{ AC97_CODEC_STAC9752,	"STAC9752", 3 },
+	{ AC97_CODEC_STAC9756,	"STAC9756", 3 },
+	{ AC97_CODEC_STAC9758,	"STAC9758", 3 },
+	{ AC97_CODEC_STAC9766,	"STAC9766", 3 },
+	{ AC97_CODEC_TR28028,	"TR28028" },
+	{ AC97_CODEC_TR28028_2,	"TR28028" },
+	{ AC97_CODEC_TR28023,	"TR28023" },
+	{ AC97_CODEC_TR28023_2,	"TR28023" },
+	{ AC97_CODEC_EM28028,	"EM28028" },
+	{ AC97_CODEC_CX20468,	"CX20468" },
+	{ AC97_CODEC_CX20468_2,	"CX20468" },
+	{ AC97_CODEC_CX20468_21,	"CX20468-21" },
+	{ AC97_CODEC_CS4297,	"CS4297" },
+	{ AC97_CODEC_CS4297A,	"CS4297A" },
+	{ AC97_CODEC_CS4294,	"CS4294" },
+	{ AC97_CODEC_CS4299,	"CS4299" },
+	{ AC97_CODEC_CS4202,	"CS4202" },
+	{ AC97_CODEC_CS4205,	"CS4205" },
+	{ AC97_CODEC_AD1819B,	"AD1819B" },
+	{ AC97_CODEC_AD1881,	"AD1881" },
+	{ AC97_CODEC_AD1881A,	"AD1881A" },
+	{ AC97_CODEC_AD1885,	"AD1885" },
+	{ AC97_CODEC_AD1886,	"AD1886" },
+	{ AC97_CODEC_AD1887,	"AD1887" },
+	{ AC97_CODEC_AD1888,	"AD1888" },
+	{ AC97_CODEC_AD1980,	"AD1980" },
+	{ AC97_CODEC_AD1981,	"AD1981" },	/* no data sheet */
+	{ AC97_CODEC_AD1981A,	"AD1981A", 0, ad1981a_init },
+	{ AC97_CODEC_AD1981B,	"AD1981B", 0, ad1981b_init },
+	{ AC97_CODEC_AD1985,	"AD1985" },
+	{ AC97_CODEC_WM9701A,	"WM9701A" },
+	{ AC97_CODEC_WM9703,	"WM9703" },
+	{ AC97_CODEC_WM9704,	"WM9704" },
+	{ AC97_CODEC_ES1921,	"ES1921" },
+	{ AC97_CODEC_ICE1232,	"ICE1232/VT1611A" },
+	{ AC97_CODEC_VT1612A,	"VT1612A" },
+	{ AC97_CODEC_VT1616,	"VT1616" },
+	{ AC97_CODEC_VT1616A,	"VT1616A" },
+	{ AC97_CODEC_VT1617A,	"VT1617A" },
+	{ AC97_CODEC_VT1618,	"VT1618" },
+	{ AC97_CODEC_ALC100,	"ALC100", 2 },
+	{ AC97_CODEC_ALC200P,	"ALC200P", 2 },
+	{ AC97_CODEC_ALC202,	"ALC202", 2 },
+	{ AC97_CODEC_ALC203,	"ALC203", 2 },
+	{ AC97_CODEC_ALC250,	"ALC250", 2 },
+	{ AC97_CODEC_ALC250_2,	"ALC250", 2 },
+	{ AC97_CODEC_ALC650,	"ALC650", 2, alc650_init },
+	{ AC97_CODEC_ALC655,	"ALC655", 2, alc650_init },
+	{ AC97_CODEC_ALC658,	"ALC658", 2, alc650_init },
+	{ AC97_CODEC_ALC850,	"ALC850", 2, alc850_init },
+	{ AC97_CODEC_EV1938,	"EV1938" },
+	{ AC97_CODEC_CMI9738,	"CMI9738", 0, cmi9738_init },
+	{ AC97_CODEC_CMI9739,	"CMI9739", 0, cmi9739_init },
+	{ AC97_CODEC_CMI9780,	"CMI9780" },
+	{ AC97_CODEC_CMI9761,	"CMI9761A", 0, cmi9761_init },
+	{ AC97_CODEC_CMI9761_2,	"CMI9761B", 0, cmi9761_init },
+	{ AC97_CODEC_CMI9761_3,	"CMI9761A+", 0, cmi9761_init },
+	{ AC97_CODEC_YMF743,	"YMF743" },
+	{ AC97_CODEC_YMF753,	"YMF753" },
+	{ 0, NULL }
+};
+
+/*
+ * Init the actual hardware related to a previously
+ * allocated instance of an AC97 device.
+ *
+ * Return zero on success.
+ */
+int
+ac97_init(ac97_t *ac, struct audio_dev *d)
+{
+	uint32_t		vid1, vid2;
+	const char		*name = NULL;
+	const char		*vendor = NULL;
+	int			enh_bits;
+	int			vol_bits;
+	uint32_t		flags;
+	char			nmbuf[128];
+	char			buf[128];
+
+	/* Save audio framework instance structure */
+	ac->d = d;
+
+	ac97_analog_reset(ac);
+
+	vid1 = RD(AC97_VENDOR_ID1_REGISTER);
+	vid2 = RD(AC97_VENDOR_ID2_REGISTER);
+
+	if (vid1 == 0xffff) {
+		audio_dev_warn(d, "AC'97 codec unresponsive");
+		return (-1);
+	}
+
+	ac->vid = (vid1 << 16) | vid2;
+
+	/*
+	 * Find out kind of codec we have and set any special case
+	 * settings needed.
+	 */
+	for (int i = 0; codecs[i].id; i++) {
+		if (ac->vid == codecs[i].id) {
+			name = codecs[i].name;
+			enh_bits = codecs[i].enh_bits;
+			ac->codec_init = codecs[i].init;
+			break;
+		}
+	}
+	for (int i = 0; vendors[i].id; i++) {
+		if ((ac->vid & 0xffffff00) == vendors[i].id) {
+			vendor = vendors[i].name;
+			break;
+		}
+	}
+	if (name == NULL) {
+		(void) snprintf(nmbuf, sizeof (nmbuf), "0x%04x%04x",
+		    vid1, vid2);
+		name = nmbuf;
+	}
+	if (vendor == NULL) {
+		vendor = "Unknown";
+	}
+
+	/*
+	 * Populate the initial shadow table.
+	 */
+	for (int i = 0; i < LAST_SHADOW_REG; i += sizeof (uint16_t)) {
+		SHADOW(ac, i) = RD(i);
+	}
+
+	ac->caps = RD(AC97_RESET_REGISTER);
+
+	enh_bits = 4;
+	vol_bits = 6;
+	flags = 0;
+
+	/* detect the bit width of the master volume controls */
+	WR(AC97_MASTER_VOLUME_REGISTER, 0x20);
+	if ((RD(AC97_MASTER_VOLUME_REGISTER) & 0x1f) == 0x1f) {
+		vol_bits = 5;
+	}
+
+	/*
+	 * AC'97 2.3 spec indicates three possible uses for AUX_OUT
+	 * (aka LNLVL_OUT aka HP_OUT).  We have to figure out which one
+	 * is in use.
+	 */
+	if (ac->caps & RR_HEADPHONE_SUPPORT) {
+		/* it looks like it is probably headphones */
+		if (ac97_probe_reg(ac, AC97_HEADPHONE_VOLUME_REGISTER)) {
+			/* it is implemented */
+			ac->flags |= AC97_FLAG_AUX_HP;
+		}
+	}
+
+	/*
+	 * If not a headphone, is it 4CH_OUT (surround?)
+	 */
+	if ((!(ac->flags & AC97_FLAG_AUX_HP)) &&
+	    (RD(AC97_EXTENDED_AUDIO_REGISTER) & EAR_SDAC)) {
+		if (ac97_probe_reg(ac, AC97_EXTENDED_LRS_VOLUME_REGISTER)) {
+			ac->flags |= AC97_FLAG_AUX_4CH;
+		}
+	}
+
+	/*
+	 * If neither, then maybe its an auxiliary line level output?
+	 */
+	if (!(ac->flags & (AC97_FLAG_AUX_HP | AC97_FLAG_AUX_4CH))) {
+		if (ac97_probe_reg(ac, AC97_HEADPHONE_VOLUME_REGISTER)) {
+			ac->flags |= AC97_FLAG_AUX_LVL;
+		}
+	}
+
+	ac->flags |= flags;
+	(void) snprintf(ac->name, sizeof (ac->name), "%s %s", vendor, name);
+
+	(void) snprintf(buf, sizeof (buf), "AC'97 codec: %s", ac->name);
+	audio_dev_add_info(d, buf);
+
+	cmn_err(CE_CONT, "?%s#%d: AC'97 codec id %s (%x, caps %x)\n",
+	    ddi_driver_name(ac->dip), ddi_get_instance(ac->dip),
+	    ac->name, ac->vid, ac->caps);
+
+	/*
+	 * Probe and register all known controls with framework
+	 */
+	if (ac97_probeinit_ctrls(ac, vol_bits, enh_bits)) {
+		audio_dev_warn(d, "AC97 controls init failed");
+
+		/* XXX - need to free all controls registered? */
+		return (-1);
+	}
+
+	ac97_hw_reset(ac);
+	ac97_init_values(ac);
+
+	return (0);
+}
diff --git a/usr/src/uts/common/io/audio/ac97/ac97_ad.c b/usr/src/uts/common/io/audio/ac97/ac97_ad.c
new file mode 100644
index 0000000000..1faddcb6fa
--- /dev/null
+++ b/usr/src/uts/common/io/audio/ac97/ac97_ad.c
@@ -0,0 +1,257 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * ADS (Analog Devices) codec extensions.
+ */
+
+/*
+ * TODO:
+ *
+ * Most vendors connect the surr-out of ad1980/ad1985 codecs to the
+ * line-out jack. So far we haven't found which vendors don't
+ * do that. So we assume that all vendors swap the surr-out
+ * and the line-out outputs. So we need swap the two outputs.
+ *
+ * Historically we internally processed the "ad198x-swap-output"
+ * property. If someday some vendors do not swap the outputs, we would
+ * set "ad198x-swap-output = 0" in the driver.conf file, and unload
+ * and reload the driver (or reboot).
+ *
+ * TODO:
+ *
+ * Since we don't have access (at present) to any such systems, we have
+ * not implemented this swapping property.  Once we can test it, we will
+ * add it.  This is noted as CR 6819556.
+ *
+ * The old code did this:
+ *
+ *	if (ddi_prop_get_int(DDI_DEV_T_ANY, statep->dip,
+ *	    DDI_PROP_DONTPASS, "ad198x-swap-output", 1) == 1) {
+ *		statep->swap_out = B_TRUE;
+ *		(void) audioixp_read_ac97(statep, CODEC_AD_REG_MISC, &tmp);
+ *		(void) audioixp_write_ac97(statep,
+ *		    CODEC_AD_REG_MISC,
+ *		    tmp | AD1980_MISC_LOSEL | AD1980_MISC_HPSEL);
+ *
+ */
+
+#include <sys/types.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include <sys/note.h>
+#include "ac97_impl.h"
+
+#define	ADS_EQ_CTRL_REGISTER		0x60
+#define	AECR_EQM			0x8000	/* disable EQ */
+#define	AECR_SYM			0x0080
+
+#define	ADS_EQ_DATA_REGISTER		0x62
+
+#define	ADS_MIXER_ADC_IGAIN_REGISTER	0x64
+#define	AMADIR_LEFT_MASK		0x0f00
+#define	AMADIR_RIGHT_MASK		0x000f
+#define	AMADIR_MXM			0x8000
+
+#define	ADS_JS_INTS_STATUS_REGISTER	0x72
+#define	AJISR_JS0INT			0x0001
+#define	AJISR_JS1INT			0x0002
+#define	AJISR_JS0ST			0x0004
+#define	AJISR_JS1ST			0x0008
+#define	AJISR_JS0MD			0x0010
+#define	AJISR_JS1MD			0x0020
+#define	AJISR_JS0TMR			0x0040
+#define	AJISR_JS1TMR			0x0080
+#define	AJISR_JS0EQB			0x0100
+#define	AJISR_JS1EQB			0x0200
+#define	AJISR_JSMT_MASK			0x1c00
+#define	AJISR_JSMT_NONE			0x0000
+#define	AJISR_JSMT_HP_LNOUT		0x0400	/* hp mutes line out */
+#define	AJISR_JSMT_HP_BOTH		0x0800	/* hp mutes both mono & line */
+#define	AJISR_JSMT_LNOUT_MONO		0x1000	/* lineout mutes mono */
+#define	AJISR_JSMT_ALL			0x1800	/* all JS muting enabled */
+
+#define	ADS_SERIAL_CFG_REGISTER		0x74
+#define	ASCR_SPLNK			0x0001
+#define	ASCR_SPDZ			0x0002
+#define	ASCR_SPAL			0x0004
+#define	ASCR_INTS			0x0010
+#define	ASCR_CHEN			0x0100
+#define	ASCR_REGM0			0x1000
+#define	ASCR_REGM1			0x2000
+#define	ASCR_REGM2			0x4000
+#define	ASCR_SLOT16			0x8000
+
+#define	ADS_MISC_CFG_REGISTER		0x76
+#define	AMCR_MBG_MASK			0x0003
+#define	AMCR_MBG_20dB			0x0000
+#define	AMCR_MBG_10dB			0x0001
+#define	AMCR_MBG_30dB			0x0002
+#define	AMCR_VREFD			0x0004
+#define	AMCR_VREFH			0x0008
+#define	AMCR_MADST			0x0010	/* AD1981B */
+#define	AMCR_SRU			0x0010	/* AD1980 */
+#define	AMCR_LOSEL			0x0020	/* AD1980 */
+#define	AMCR_2CMIC			0x0040
+#define	AMCR_MADPD			0x0080	/* AD1981B */
+#define	AMCR_SPRD			0x0080	/* AD1980 */
+#define	AMCR_DMIX_6TO2			0x0100	/* AD1980 */
+#define	AMCR_DMIX_FORCE			0x0200	/* AD1980 */
+#define	AMCR_FMXE			0x0200	/* AD1981B */
+#define	AMCR_HPSEL			0x0400	/* AD1980 */
+#define	AMCR_CLDIS			0x0800	/* AD1980 */
+#define	AMCR_LODIS			0x1000	/* AD1980 */
+#define	AMCR_DAM			0x0800	/* AD1981B */
+#define	AMCR_MSPLT			0x2000
+#define	AMCR_AC97NC			0x4000	/* AD1980 */
+#define	AMCR_DACZ			0x8000
+
+static void
+ads_set_micboost(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t	*ac = actrl->actrl_ac97;
+	uint16_t	v;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	switch (value) {
+	case 0x1:
+		/* 0db */
+		ac97_clr(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		break;
+	case 0x2:
+		/* 10dB */
+		ac97_set(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		v = ac97_rd(ac, ADS_MISC_CFG_REGISTER);
+		v &= ~AMCR_MBG_MASK;
+		v |= AMCR_MBG_10dB;
+		ac97_wr(ac, ADS_MISC_CFG_REGISTER, v);
+		break;
+	case 0x4:
+		/* 20dB */
+		ac97_set(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		v = ac97_rd(ac, ADS_MISC_CFG_REGISTER);
+		v &= ~AMCR_MBG_MASK;
+		v |= AMCR_MBG_20dB;
+		ac97_wr(ac, ADS_MISC_CFG_REGISTER, v);
+		break;
+	case 0x8:
+		/* 30dB */
+		ac97_set(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		v = ac97_rd(ac, ADS_MISC_CFG_REGISTER);
+		v &= ~AMCR_MBG_MASK;
+		v |= AMCR_MBG_30dB;
+		ac97_wr(ac, ADS_MISC_CFG_REGISTER, v);
+		break;
+	}
+}
+
+static void
+ads_set_micsrc(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t	*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	switch (value) {
+	case 0x1:	/* mic1 */
+		ac97_clr(ac, ADS_MISC_CFG_REGISTER, AMCR_2CMIC);
+		ac97_clr(ac, AC97_GENERAL_PURPOSE_REGISTER, GPR_MS_MIC2);
+		break;
+	case 0x2:	/* mic2 */
+		ac97_clr(ac, ADS_MISC_CFG_REGISTER, AMCR_2CMIC);
+		ac97_set(ac, AC97_GENERAL_PURPOSE_REGISTER, GPR_MS_MIC2);
+		break;
+	case 0x4:	/* stereo - ms bit clear to allow MIC1 to be mixed */
+		ac97_set(ac, ADS_MISC_CFG_REGISTER, AMCR_2CMIC);
+		ac97_clr(ac, AC97_GENERAL_PURPOSE_REGISTER, GPR_MS_MIC2);
+		break;
+	}
+}
+
+static void
+ads_setup_micsrc(ac97_t *ac)
+{
+	ac97_ctrl_t		*ctrl;
+	static const char	*values[] = {
+		AUDIO_PORT_MIC1,
+		AUDIO_PORT_MIC2,
+		AUDIO_PORT_STEREO,
+		NULL
+	};
+	ac97_ctrl_probe_t cpt = {
+		AUDIO_CTRL_ID_MICSRC, 1, 0x7, 0x7, AUDIO_CTRL_TYPE_ENUM,
+		AC97_FLAGS | AUDIO_CTRL_FLAG_REC, 0, ads_set_micsrc,
+		NULL, 0, values };
+
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_MICSRC);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+
+	ac97_alloc_control(ac, &cpt);
+}
+
+static void
+ads_setup_micboost(ac97_t *ac)
+{
+	ac97_ctrl_t		*ctrl;
+
+	static const char	*values[] = {
+		AUDIO_VALUE_OFF,	/* 0dB */
+		AUDIO_VALUE_LOW,	/* 10dB */
+		AUDIO_VALUE_MEDIUM,	/* 20dB */
+		AUDIO_VALUE_HIGH,	/* 30dB */
+		NULL
+	};
+	ac97_ctrl_probe_t cpt = {
+		AUDIO_CTRL_ID_MICBOOST, 1, 0xf, 0xf, AUDIO_CTRL_TYPE_ENUM,
+		AC97_FLAGS | AUDIO_CTRL_FLAG_REC, 0, ads_set_micboost,
+		NULL, 0, values };
+
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_MICBOOST);
+	if (ctrl) {
+		if (ctrl->actrl_initval) {
+			/* 20dB by default */
+			cpt.cp_initval = 2;
+		}
+		ac97_free_control(ctrl);
+	}
+
+	ac97_alloc_control(ac, &cpt);
+}
+
+void
+ad1981a_init(ac97_t *ac)
+{
+	ads_setup_micboost(ac);
+}
+
+void
+ad1981b_init(ac97_t *ac)
+{
+	ads_setup_micboost(ac);
+	ads_setup_micsrc(ac);	/* this part can use a mic array */
+}
diff --git a/usr/src/uts/common/io/audio/ac97/ac97_alc.c b/usr/src/uts/common/io/audio/ac97/ac97_alc.c
new file mode 100644
index 0000000000..1c621c5094
--- /dev/null
+++ b/usr/src/uts/common/io/audio/ac97/ac97_alc.c
@@ -0,0 +1,219 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * ALC (Realtek/Advance Logic) codec extensions.
+ */
+
+#include <sys/types.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include <sys/note.h>
+#include "ac97_impl.h"
+
+#define	ALC_DATA_FLOW_CTRL_REGISTER	0x6a
+#define	ADFC_SPDIFIN_EN			0x8000
+#define	ADFC_SPDIFIN_MON_EN		0x4000
+#define	ADFC_SPDIF_OUT_MASK		0x3000
+#define	ADFC_SPDIF_OUT_ACLINK		0x0000
+#define	ADFC_SPDIF_OUT_ADC		0x1000
+#define	ADFC_SPDIF_OUT_BYPASS		0x2000
+#define	ADFC_PCM_SPDIFIN		0x0800
+#define	ADFC_BACK_SURROUND		0x0400	/* ALC850 only */
+#define	ADFC_CENTER_LFE			0x0400	/* ALC650 series */
+#define	ADFC_MIC			0x0000
+#define	ADFC_SURROUND			0x0200
+#define	ADFC_LINEIN			0x0000
+#define	ADFC_FRONT_MIC_MONO_OUT		0x0100	/* ALC850 */
+#define	ADFC_ANALOG_INPUT_PASS_CLFE	0x0020
+#define	ADFC_ANALOG_INPUT_PASS_SURROUND	0x0010
+#define	ADFC_SURROUND_MIRROR		0x0001
+
+#define	ALC_SURROUND_DAC_REGISTER	0x64
+#define	ASD_SURROUND_MUTE		0x8000
+#define	ASD_SURR_LEFT_VOL		0x1f00
+#define	ASD_SURR_RIGHT_VOL		0x001f
+
+#define	ALC_CEN_LFE_DAC_REGISTER	0x66
+#define	ACLD_CEN_LFE_MUTE		0x8000
+#define	ACLD_LFE_VOL			0x1f00
+#define	ACLD_CEN_VOL			0x001f
+
+#define	ALC_MISC_CTRL_REGISTER		0x7a
+#define	AMC_XTLSEL			0x8000
+#define	AMC_VREFOUT_DIS			0x1000
+#define	AMC_INDEP_MUTE_CTRL		0x0800
+#define	AMC_JD2_SURR_CEN_LFE		0x0008
+#define	AMC_JD1_SURR_CEN_LFE		0x0004
+#define	AMC_PIN47_SPDIF			0x0002
+#define	AMC_PIN47_EAPD			0x0000
+#define	AMC_JD0_SURR_CEN_LFE		0x0001
+
+static void
+alc650_set_linein_func(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	if (value & 2) {
+		ac97_set(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_SURROUND);
+	} else {
+		ac97_clr(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_SURROUND);
+	}
+}
+
+static void
+alc650_set_mic_func(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	if (value & 2) {
+		ac97_set(ac, ALC_MISC_CTRL_REGISTER, AMC_VREFOUT_DIS);
+		ac97_set(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_CENTER_LFE);
+	} else {
+		ac97_clr(ac, ALC_MISC_CTRL_REGISTER, AMC_VREFOUT_DIS);
+		ac97_clr(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_CENTER_LFE);
+	}
+}
+
+#if 0
+static void
+alc850_set_auxin_func(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	if (value & 2) {
+		ac97_set(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_BACK_SURROUND);
+	} else {
+		ac97_clr(ac, ALC_DATA_FLOW_CTRL_REGISTER, ADFC_BACK_SURROUND);
+	}
+}
+#endif
+
+static void
+alc650_set_pcm(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+	uint16_t	adj_value;
+	uint16_t	mute;
+	uint8_t		vol;
+
+	/* limit input values to 16 bits and split to right and left */
+	vol = value & 0xff;
+
+	/* If this control is mute-able than set as muted if needed */
+	mute = vol ? 0 : ASD_SURROUND_MUTE;
+	adj_value = ac97_val_scale(vol, vol, 5) | mute;
+
+	/* select page 0 */
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);
+	/* adjust all three PCM volumes */
+	ac97_wr(ac, AC97_PCM_OUT_VOLUME_REGISTER, adj_value);
+	ac97_wr(ac, ALC_SURROUND_DAC_REGISTER, adj_value);
+	ac97_wr(ac, ALC_CEN_LFE_DAC_REGISTER, adj_value);
+}
+
+static const char *alc_linein_funcs[] = {
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_SURROUND,
+	NULL
+};
+
+static const char *alc_mic_funcs[] = {
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_CENLFE,
+	NULL
+};
+
+static ac97_ctrl_probe_t alc650_linein_func_cpt = {
+	AUDIO_CTRL_ID_JACK1, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM, AC97_FLAGS,
+	0, alc650_set_linein_func, NULL, 0, alc_linein_funcs
+};
+static ac97_ctrl_probe_t alc650_mic_func_cpt = {
+	AUDIO_CTRL_ID_JACK2, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM, AC97_FLAGS,
+	0, alc650_set_mic_func, NULL, 0, alc_mic_funcs
+};
+
+static void
+alc_pcm_override(ac97_t *ac)
+{
+	ac97_ctrl_t	*ctrl;
+
+	/* override master PCM volume function */
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_VOLUME);
+	if (ctrl != NULL) {
+		ctrl->actrl_write_fn = alc650_set_pcm;
+	}
+}
+
+void
+alc650_init(ac97_t *ac)
+{
+	ac97_ctrl_probe_t	cp;
+	int			ival;
+
+	bcopy(&alc650_linein_func_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_LINEIN_FUNC, 0);
+	if ((ival >= 1) && (ival <= 2)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+
+	bcopy(&alc650_mic_func_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_MIC_FUNC, 0);
+	if ((ival >= 1) && (ival <= 2)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+
+	alc_pcm_override(ac);
+}
+
+void
+alc850_init(ac97_t *ac)
+{
+	/*
+	 * NB: We could probably enable 7.1 here using the AUXIN source,
+	 * but there are a few details still missing from the data sheet.
+	 * (Such as, how is volume from the back-surround DAC managed?,
+	 * and what SDATA slots are the back surround delivered on?)
+	 *
+	 * Also, the AC'97 controllers themselves don't necessarily support
+	 * 7.1, so we'd have to figure out how to coordinate detection
+	 * with the controller.  5.1 should be good enough for now.
+	 *
+	 * Unlike other products, ALC850 has separate pins for 5.1 data,
+	 * so jack retasking isn't needed.  However, it can retask
+	 * some jacks, but we don't have full details for that right
+	 * now.  We've not seen it on any systems (yet) where this was
+	 * necessary, though.
+	 */
+
+	alc_pcm_override(ac);
+}
diff --git a/usr/src/uts/common/io/audio/ac97/ac97_cmi.c b/usr/src/uts/common/io/audio/ac97/ac97_cmi.c
new file mode 100644
index 0000000000..50351cb08c
--- /dev/null
+++ b/usr/src/uts/common/io/audio/ac97/ac97_cmi.c
@@ -0,0 +1,311 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * CMI (C-Media) codec extensions.
+ */
+
+#include <sys/types.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include <sys/note.h>
+#include "ac97_impl.h"
+
+/*
+ * C-Media 9739 part is weird.  Instead of having independent volume
+ * controls for each of the channels, it uses a single master volume
+ * and just provides mute support for the other bits.  It does this
+ * for PCM volume as well, so we can't use it either.  Ugh.  It also
+ * has an optional 30 dB mic boost.  Apparently the 9761 behaves in
+ * much the same fashion as the 9739.
+ *
+ * C-Media 9738 is a more or less typical 4CH device according to the
+ * datasheet.  It however supports jack retasking allowing the line in
+ * jack to function as a surround output.  Google suggests that the
+ * volume controls on this part are about as busted as on the other
+ * parts.  So, we just use synthetic volume for it.
+ *
+ * C-Media 9780 is largely a mystery (ENODATASHEET).
+ */
+
+
+#define	CMI_TASK_REGISTER	0x5A	/* 9738 jack retasking */
+#define	CTR_F2R			0x2000	/* front routed to rear */
+#define	CTR_S2LNI		0x0400	/* surround to line in */
+
+#define	CMI_MULTICH_REGISTER	0x64	/* 9739 and 9761a */
+#define	CMR_PCBSW		0x8000	/* PC Beep volume bypass */
+#define	CMR_P47			0x4000	/* configure P47 function */
+#define	CMR_REFCTL		0x2000	/* enable vref output */
+#define	CMR_CLCTL		0x1000	/* center/lfe output enable */
+#define	CMR_S2LNI		0x0400	/* surround to line in */
+#define	CMR_MIX2S		0x0200	/* analog input pass to surround */
+#define	CMR_BSTSEL		0x0001	/* micboost use 30dB */
+
+static void
+cmi_set_micboost(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t	*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	switch (value) {
+	case 0x1:
+		/* 0db */
+		ac97_clr(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		ac97_clr(ac, CMI_MULTICH_REGISTER, CMR_BSTSEL);
+		break;
+	case 0x2:
+		/* 20dB */
+		ac97_set(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		ac97_clr(ac, CMI_MULTICH_REGISTER, CMR_BSTSEL);
+		break;
+	case 0x4:
+		/* 30dB */
+		ac97_set(ac, AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
+		ac97_set(ac, CMI_MULTICH_REGISTER, CMR_BSTSEL);
+		break;
+	}
+}
+
+static void
+cmi_set_linein_func(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	if (value & 2) {
+		ac97_set(ac, CMI_MULTICH_REGISTER, CMR_S2LNI);
+	} else {
+		ac97_clr(ac, CMI_MULTICH_REGISTER, CMR_S2LNI);
+	}
+}
+
+static void
+cmi_set_mic_func(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	if (value & 2) {
+		ac97_set(ac, CMI_MULTICH_REGISTER, CMR_CLCTL);
+	} else {
+		ac97_clr(ac, CMI_MULTICH_REGISTER, CMR_CLCTL);
+	}
+}
+
+static void
+cmi_setup_micboost(ac97_t *ac)
+{
+	ac97_ctrl_t		*ctrl;
+
+	static const char	*values[] = {
+		AUDIO_VALUE_OFF,	/* 0dB */
+		AUDIO_VALUE_MEDIUM,	/* 20dB */
+		AUDIO_VALUE_HIGH,	/* 30dB */
+		NULL
+	};
+	ac97_ctrl_probe_t cpt = {
+		AUDIO_CTRL_ID_MICBOOST, 1, 0xf, 0xf, AUDIO_CTRL_TYPE_ENUM,
+		AC97_FLAGS | AUDIO_CTRL_FLAG_REC, 0, cmi_set_micboost,
+		NULL, 0, values };
+
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_MICBOOST);
+	if (ctrl) {
+		if (ctrl->actrl_initval) {
+			/* 20dB by default */
+			cpt.cp_initval = 1;
+		}
+		ac97_free_control(ctrl);
+	}
+
+	ac97_alloc_control(ac, &cpt);
+}
+
+static const char *cmi_linein_funcs[] = {
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_SURROUND,
+	NULL
+};
+
+static const char *cmi_mic_funcs[] = {
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_CENLFE,
+	NULL
+};
+
+static void
+cmi_setup_jack_funcs(ac97_t *ac)
+{
+	ac97_ctrl_probe_t	cp;
+	int			ival;
+
+	ac97_ctrl_probe_t linein_cpt = {
+		AUDIO_CTRL_ID_JACK1, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM, AC97_FLAGS,
+		0, cmi_set_linein_func, NULL, 0, cmi_linein_funcs
+	};
+	ac97_ctrl_probe_t mic_cpt = {
+		AUDIO_CTRL_ID_JACK2, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM, AC97_FLAGS,
+		0, cmi_set_mic_func, NULL, 0, cmi_mic_funcs
+	};
+
+	bcopy(&linein_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_LINEIN_FUNC, 0);
+	if ((ival >= 1) && (ival <= 2)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+
+	bcopy(&mic_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_MIC_FUNC, 0);
+	if ((ival >= 1) && (ival <= 2)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+}
+
+static void
+cmi_set_linein_func_9738(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	if (value & 2) {
+		ac97_set(ac, CMI_TASK_REGISTER, CTR_S2LNI);
+	} else {
+		ac97_clr(ac, CMI_TASK_REGISTER, CTR_S2LNI);
+	}
+}
+
+static void
+cmi_set_spread_9738(ac97_ctrl_t *actrl, uint64_t value)
+{
+	ac97_t		*ac = actrl->actrl_ac97;
+
+	if (value) {
+		ac97_set(ac, CMI_TASK_REGISTER, CTR_F2R);
+	} else {
+		ac97_clr(ac, CMI_TASK_REGISTER, CTR_F2R);
+	}
+}
+
+static void
+cmi_setup_jack_func_9738(ac97_t *ac)
+{
+	ac97_ctrl_probe_t	cp;
+	int			ival;
+
+	ac97_ctrl_probe_t linein_cpt = {
+		AUDIO_CTRL_ID_JACK1, 1, 3, 3, AUDIO_CTRL_TYPE_ENUM, AC97_FLAGS,
+		0, cmi_set_linein_func_9738, NULL, 0, cmi_linein_funcs
+	};
+	ac97_ctrl_probe_t spread_cpt = {
+		AUDIO_CTRL_ID_SPREAD, 0, 0, 1, AUDIO_CTRL_TYPE_BOOLEAN,
+		AC97_FLAGS, 0, cmi_set_spread_9738,
+	};
+
+	bcopy(&linein_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_LINEIN_FUNC, 0);
+	if ((ival >= 1) && (ival <= 2)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+
+	bcopy(&spread_cpt, &cp, sizeof (cp));
+	ival = ac97_get_prop(ac, AC97_PROP_SPREAD, -1);
+	if ((ival >= 0) && (ival <= 1)) {
+		cp.cp_initval = ival;
+	}
+	ac97_alloc_control(ac, &cp);
+}
+
+
+static void
+cmi_setup_volume(ac97_t *ac)
+{
+	ac97_ctrl_t	*ctrl;
+
+	/*
+	 * These CMI parts seem to be really weird.  They don't have
+	 * *any* functioning volume controls on them (mute only) apart
+	 * from the record and monitor sources (excluding PCM).  I
+	 * don't understand why not.  We just eliminate all of the
+	 * volume controls and replace with a soft volume control.
+	 * Its not an ideal situation, but I don't know what else I
+	 * can do about it.
+	 */
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_VOLUME);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_FRONT);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_SURROUND);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_CENTER);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+	ctrl = ac97_control_find(ac, AUDIO_CTRL_ID_LFE);
+	if (ctrl) {
+		ac97_free_control(ctrl);
+	}
+
+	/* make sure we have disabled mute and attenuation on physical ctrls */
+	ac97_wr(ac, AC97_INTERRUPT_PAGING_REGISTER, 0);	/* select page 0 */
+	ac97_wr(ac, AC97_PCM_OUT_VOLUME_REGISTER, 0);
+	ac97_wr(ac, AC97_MASTER_VOLUME_REGISTER, 0);
+	ac97_wr(ac, AC97_EXTENDED_C_LFE_VOLUME_REGISTER, 0);
+	ac97_wr(ac, AC97_EXTENDED_LRS_VOLUME_REGISTER, 0);
+
+	(void) audio_dev_add_soft_volume(ac97_get_dev(ac));
+}
+
+void
+cmi9739_init(ac97_t *ac)
+{
+	cmi_setup_volume(ac);
+	cmi_setup_micboost(ac);
+	cmi_setup_jack_funcs(ac);
+}
+
+void
+cmi9761_init(ac97_t *ac)
+{
+	cmi_setup_volume(ac);
+	cmi_setup_micboost(ac);
+	cmi_setup_jack_funcs(ac);
+}
+
+void
+cmi9738_init(ac97_t *ac)
+{
+	cmi_setup_volume(ac);
+	cmi_setup_jack_func_9738(ac);
+}
diff --git a/usr/src/uts/common/io/audio/ac97/ac97_impl.h b/usr/src/uts/common/io/audio/ac97/ac97_impl.h
new file mode 100644
index 0000000000..073707c0d9
--- /dev/null
+++ b/usr/src/uts/common/io/audio/ac97/ac97_impl.h
@@ -0,0 +1,113 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_SYS_AC97_IMPL_H
+#define	_SYS_AC97_IMPL_H
+
+typedef struct ac97_ctrl ac97_ctrl_t;
+typedef void (*ac97_set_t)(ac97_ctrl_t *, uint64_t);
+
+/*
+ * Per control state
+ */
+struct ac97_ctrl {
+	list_node_t	actrl_linkage;  /* For private cntrls list */
+	struct ac97	*actrl_ac97;
+	int		actrl_bits;	/* Port width */
+	uint32_t	actrl_type;	/* control type */
+	const char	*actrl_name;	/* control's name */
+	audio_ctrl_t	*actrl_ctrl;    /* control framework handle */
+	ac97_set_t	actrl_write_fn; /* control write function */
+	uint64_t	actrl_minval;   /* MIN value for control */
+	uint64_t	actrl_maxval;   /* MAX value for control */
+	uint64_t	actrl_value;    /* current value in port */
+	uint64_t	actrl_initval;  /* initial value in port */
+	uint16_t	actrl_muteable; /* if muteable, bits to do it */
+};
+
+/*
+ * Function Type used on controls that are optional
+ * This will return non-zero if the control should be
+ * installed and configured.
+ */
+typedef int (*cp_probe_t)(ac97_t *ac);
+
+
+/*
+ * This is used to enumerate and probe all controls that are
+ * supported and configurable on AC97 hardware
+ */
+typedef struct ac97_ctrl_probe {
+	const char	*cp_name;	/* name of control */
+	uint64_t	cp_initval;	/* Initial value for control */
+	uint64_t	cp_minval;	/* MIN value for control */
+	uint64_t	cp_maxval;	/* MAX value for control */
+	uint32_t	cp_type;	/* Type of control */
+	uint32_t	cp_flags;	/* control type flags */
+	uint16_t	cp_muteable;	/* Mute bit mask */
+	ac97_set_t	cp_write_fn;	/* control write function */
+	cp_probe_t	cp_probe;	/* Probe if control exists */
+	int		cp_bits;	/* Bits for volume controls */
+	const char	**cp_enum;	/* Enumeration value */
+} ac97_ctrl_probe_t;
+
+typedef struct ac97_hooks {
+	void		(*h_detach)(ac97_t *);
+	void		(*h_reset)(ac97_t *);
+	void		(*h_restore)(ac97_t *);
+} ac97_hooks_t;
+
+/*
+ * These are the flags for most of our controls
+ */
+#define	AC97_RW		(AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE)
+#define	AC97_FLAGS	(AC97_RW | AUDIO_CTRL_FLAG_POLL)
+
+void ac97_wr(ac97_t *, uint8_t, uint16_t);
+uint16_t ac97_rd(ac97_t *, uint8_t);
+void ac97_clr(ac97_t *, uint8_t, uint16_t);
+void ac97_set(ac97_t *, uint8_t, uint16_t);
+void ac97_alloc_control(ac97_t *, ac97_ctrl_probe_t *);
+void ac97_free_control(ac97_ctrl_t *);
+ac97_ctrl_t *ac97_control_find(ac97_t *, const char *);
+uint16_t ac97_val_scale(int left, int right, int bits);
+uint16_t ac97_mono_scale(int val, int bits);
+audio_dev_t *ac97_get_dev(ac97_t *);
+int ac97_get_prop(ac97_t *, char *, int);
+
+/* Codec specific initializations */
+
+void ad1981a_init(ac97_t *);
+void ad1981b_init(ac97_t *);
+
+void alc650_init(ac97_t *);
+void alc850_init(ac97_t *);
+
+void cmi9738_init(ac97_t *);
+void cmi9739_init(ac97_t *);
+void cmi9761_init(ac97_t *);
+
+#endif	/* _SYS_AC97_IMPL_H */
diff --git a/usr/src/uts/common/io/audio/drv/audio1575/audio1575.c b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.c
new file mode 100644
index 0000000000..6820d7fd57
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.c
@@ -0,0 +1,1952 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+/*
+ * audio1575 Audio Driver
+ *
+ * The driver is primarily targeted at providing audio support for
+ * those systems which use the Uli M1575 audio core.
+ *
+ * The M1575 audio core, in AC'97 controller mode, has independent
+ * channels for PCM in, PCM out, mic in, modem in, and modem out.
+ *
+ * The AC'97 controller is a PCI bus master with scatter/gather
+ * support. Each channel has a DMA engine. Currently, we use only
+ * the PCM in and PCM out channels. Each DMA engine uses one buffer
+ * descriptor list. And the buffer descriptor list is an array of up
+ * to 32 entries, each of which describes a data buffer. Each entry
+ * contains a pointer to a data buffer, control bits, and the length
+ * of the buffer being pointed to, where the length is expressed as
+ * the number of samples. This, combined with the 16-bit sample size,
+ * gives the actual physical length of the buffer.
+ *
+ * TODO: System power management is not yet supported by the driver.
+ *
+ * 	NOTE:
+ * 	This driver depends on the drv/audio, misc/ac97
+ * 	modules being loaded first.
+ */
+
+#include <sys/types.h>
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/conf.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/pci.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include "audio1575.h"
+
+/*
+ * Module linkage routines for the kernel
+ */
+static int audio1575_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audio1575_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+
+/*
+ * Entry point routine prototypes
+ */
+static int audio1575_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void audio1575_close(void *);
+static int audio1575_start(void *);
+static void audio1575_stop(void *);
+static int audio1575_format(void *);
+static int audio1575_channels(void *);
+static int audio1575_rate(void *);
+static uint64_t audio1575_count(void *);
+static void audio1575_sync(void *, unsigned);
+static size_t audio1575_qlen(void *);
+
+static audio_engine_ops_t audio1575_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	audio1575_open,
+	audio1575_close,
+	audio1575_start,
+	audio1575_stop,
+	audio1575_count,
+	audio1575_format,
+	audio1575_channels,
+	audio1575_rate,
+	audio1575_sync,
+	audio1575_qlen
+};
+
+/*
+ * interrupt handler
+ */
+static uint_t	audio1575_intr(caddr_t, caddr_t);
+
+/*
+ * Local Routine Prototypes
+ */
+static int audio1575_attach(dev_info_t *);
+static int audio1575_resume(dev_info_t *);
+static int audio1575_detach(dev_info_t *);
+static int audio1575_suspend(dev_info_t *);
+
+static int audio1575_alloc_port(audio1575_state_t *, int);
+static void audio1575_free_port(audio1575_port_t *);
+static void audio1575_start_port(audio1575_port_t *);
+static void audio1575_stop_port(audio1575_port_t *);
+static void audio1575_reset_port(audio1575_port_t *);
+static void audio1575_update_port(audio1575_port_t *);
+
+static int audio1575_setup_intr(audio1575_state_t *);
+static int audio1575_codec_sync(audio1575_state_t *);
+static void audio1575_write_ac97(void *, uint8_t, uint16_t);
+static uint16_t audio1575_read_ac97(void *, uint8_t);
+static int audio1575_chip_init(audio1575_state_t *);
+static int audio1575_map_regs(audio1575_state_t *);
+static void audio1575_unmap_regs(audio1575_state_t *);
+static void audio1575_dma_stop(audio1575_state_t *);
+static void audio1575_pci_enable(audio1575_state_t *);
+static void audio1575_pci_disable(audio1575_state_t *);
+
+static void audio1575_destroy(audio1575_state_t *);
+
+/*
+ * Global variables, but used only by this file.
+ */
+
+/*
+ * DDI Structures
+ */
+
+
+/* Device operations structure */
+static struct dev_ops audio1575_dev_ops = {
+	DEVO_REV,		/* devo_rev */
+	0,			/* devo_refcnt */
+	NULL,			/* devo_getinfo */
+	nulldev,		/* devo_identify - obsolete */
+	nulldev,		/* devo_probe */
+	audio1575_ddi_attach,	/* devo_attach */
+	audio1575_ddi_detach,	/* devo_detach */
+	nodev,			/* devo_reset */
+	NULL,			/* devi_cb_ops */
+	NULL,			/* devo_bus_ops */
+	NULL,			/* devo_power */
+	ddi_quiesce_not_supported,	/* devo_quiesce */
+};
+
+/* Linkage structure for loadable drivers */
+static struct modldrv audio1575_modldrv = {
+	&mod_driverops,		/* drv_modops */
+	M1575_MOD_NAME,		/* drv_linkinfo */
+	&audio1575_dev_ops,	/* drv_dev_ops */
+};
+
+/* Module linkage structure */
+static struct modlinkage audio1575_modlinkage = {
+	MODREV_1,			/* ml_rev */
+	(void *)&audio1575_modldrv,	/* ml_linkage */
+	NULL				/* NULL terminates the list */
+};
+
+
+/*
+ * device access attributes for register mapping
+ */
+static struct ddi_device_acc_attr dev_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static struct ddi_device_acc_attr buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * DMA attributes of buffer descriptor list
+ */
+static ddi_dma_attr_t bdlist_dma_attr = {
+	DMA_ATTR_V0,	/* version */
+	0x0000000000000000LL,		/* dlim_addr_lo */
+	0x00000000ffffffffLL,		/* dlim_addr_hi */
+	0x000000000000ffffLL,		/* DMA counter register - 64 bits */
+	0x0000000000000008LL,		/* DMA address align must be 8-bytes */
+	0x0000003c,			/* 1 through 64 byte burst sizes */
+	0x00000008,			/* min xfer DMA size BDList entry */
+	0x00000000000ffffLL,		/* max xfer size, 64K */
+	0x000000000001fffLL,		/* seg, set to PAGESIZE */
+	0x00000001,			/* s/g list length, no s/g */
+	0x00000008,			/* granularity of device minxfer */
+	0				/* DMA flags use virtual address */
+};
+
+/*
+ * DMA attributes of buffers to be used to receive/send audio data
+ */
+static ddi_dma_attr_t	sample_buf_dma_attr = {
+	DMA_ATTR_V0,
+	0x0000000000000000LL,		/* dlim_addr_lo */
+	0x00000000ffffffffLL,		/* dlim_addr_hi */
+	0x000000000001fffeLL,		/* DMA counter register - 16 bits */
+	0x0000000000000004LL,		/* DMA address align 2-byte boundary */
+	0x0000003c,			/* 1 through 60 byte burst sizes */
+	0x00000004,			/* min xfer DMA size BDList entry */
+	0x000000000001ffffLL,		/* max xfer size, 64K */
+	0x000000000001ffffLL,		/* seg, set to 64K */
+	0x00000001,			/* s/g list length, no s/g */
+	0x00000004,			/* granularity of device minxfer */
+	0				/* DMA flags use virtual address */
+};
+
+/*
+ * _init()
+ *
+ * Description:
+ *	Driver initialization, called when driver is first loaded.
+ *	This is how access is initially given to all the static structures.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_install() status, see mod_install(9f)
+ */
+int
+_init(void)
+{
+	int	error;
+
+	audio_init_ops(&audio1575_dev_ops, M1575_NAME);
+
+	if ((error = mod_install(&audio1575_modlinkage)) != 0) {
+		audio_fini_ops(&audio1575_dev_ops);
+	}
+
+	return (error);
+}
+
+/*
+ * _fini()
+ *
+ * Description:
+ *	Module de-initialization, called when the driver is to be unloaded.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_remove() status, see mod_remove(9f)
+ */
+int
+_fini(void)
+{
+	int		error;
+
+	if ((error = mod_remove(&audio1575_modlinkage)) != 0) {
+		return (error);
+	}
+
+	/* clean up ops */
+	audio_fini_ops(&audio1575_dev_ops);
+
+	return (0);
+}
+
+/*
+ * _info()
+ *
+ * Description:
+ *	Module information, returns information about the driver.
+ *
+ * Arguments:
+ *	modinfo		*modinfop	Pointer to the opaque modinfo structure
+ *
+ * Returns:
+ *	mod_info() status, see mod_info(9f)
+ */
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&audio1575_modlinkage, modinfop));
+}
+
+
+/* ******************* Driver Entry Points ********************************* */
+
+/*
+ * audio1575_ddi_attach()
+ *
+ * Description:
+ *	Implements the DDI attach(9e) entry point.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audio1575_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audio1575_attach(dip));
+
+	case DDI_RESUME:
+		return (audio1575_resume(dip));
+	}
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio1575_ddi_detach()
+ *
+ * Description:
+ *	Implements the detach(9e) entry point.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *	ddi_detach_cmd_t	cmd	Detach command
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audio1575_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audio1575_detach(dip));
+
+	case DDI_SUSPEND:
+		return (audio1575_suspend(dip));
+	}
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio1575_intr()
+ *
+ * Description:
+ *	Interrupt service routine for both play and record. For play we
+ *	get the next buffers worth of audio. For record we send it on to
+ *	the mixer.
+ *
+ *	Each of buffer descriptor has a field IOC(interrupt on completion)
+ *	When both this and the IOC bit of correspondent dma control register
+ *	is set, it means that the controller should issue an interrupt upon
+ *	completion of this buffer. Note that in the clearing of the interrupts
+ *	below that the PCM IN and PCM out interrupts ar cleared by their
+ *	respective control registers and not by writing a '1' to the INTRSR
+ *	the interrupt status register. Only CPRINTR,SPINTR,and GPIOINTR
+ *	require a '1' written to the INTRSR register to clear those
+ *	interrupts. See comments below.
+ *
+ * Arguments:
+ *	caddr_t		arg	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+audio1575_intr(caddr_t arg, caddr_t dontcare)
+{
+	audio1575_state_t	*statep = (void *)arg;
+	uint32_t		intrsr;
+	uint8_t			index;
+	audio1575_port_t	*consume = NULL;
+	audio1575_port_t	*produce = NULL;
+
+	_NOTE(ARGUNUSED(dontcare));
+
+	mutex_enter(&statep->lock);
+
+	intrsr = GET32(M1575_INTRSR_REG);
+
+	/* check if device is interrupting */
+	if (intrsr == 0) {
+		if (statep->ksp) {
+			/* increment the spurious ino5 interrupt cnt */
+			M1575_KIOP(statep)->intrs[KSTAT_INTR_SPURIOUS]++;
+		}
+
+		mutex_exit(&statep->lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	/* update the kernel interrupt statistics */
+	if (statep->ksp) {
+		M1575_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	/*
+	 * The Uli M1575 generates an interrupt for each interrupt
+	 * type. therefore we only process one interrupt type
+	 * per invocation of the audio1575_intr() routine.
+	 * WARNING: DO NOT attempt to optimize this by looping
+	 * until the INTRSR register is clear as this will
+	 * generate spurious ino5 interrupts.
+	 */
+	if (GET16(M1575_PCMISR_REG) & M1575_PCMISR_BCIS) {
+		/* Clear PCM IN interrupt */
+		PUT16(M1575_PCMISR_REG, M1575_SR_CLR);
+		/*
+		 * Note: This interrupt is not cleared by writing a '1'
+		 * to the M1575_INTRSR_REG according to the M1575 Super I/O
+		 * data sheet on page 189.
+		 */
+
+		/* update the LVI -- we just set it to the current value - 1 */
+		index = GET8(M1575_PCMICIV_REG);
+		index = (index - 1) % M1575_BD_NUMS;
+		PUT8(M1575_PCMILVIV_REG, index);
+		produce = statep->ports[M1575_REC];
+
+	} else if (GET16(M1575_PCMOSR_REG) & M1575_PCMOSR_BCIS) {
+		/* Clear PCM OUT interrupt */
+		PUT16(M1575_PCMOSR_REG, M1575_SR_CLR);
+		/*
+		 * Note: This interrupt is not cleared by writing a '1'
+		 * to the M1575_INTRSR_REG according to the M1575 Super I/O
+		 * data sheet on page 189.
+		 */
+
+		/* update the LVI -- we just set it to the current value - 1 */
+		index = GET8(M1575_PCMOCIV_REG);
+		index = (index - 1) % M1575_BD_NUMS;
+		PUT8(M1575_PCMOLVIV_REG, index);
+		consume = statep->ports[M1575_PLAY];
+
+	} else {
+		/* Clear other interrupts (there should not be any) */
+		PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
+	}
+
+	mutex_exit(&statep->lock);
+
+	if (produce) {
+		audio_engine_produce(produce->engine);
+	}
+	if (consume) {
+		audio_engine_consume(consume->engine);
+	}
+
+	return (DDI_INTR_CLAIMED);
+}
+
+/*
+ * audio1575_open()
+ *
+ * Description:
+ *	Opens a DMA engine for use.
+ *
+ * Arguments:
+ *	void		*arg		The DMA engine to set up
+ *	int		flag		Open flags
+ *	unsigned	*fragfrp	Receives number of frames per fragment
+ *	unsigned	*nfragsp	Receives number of fragments
+ *	caddr_t		*bufp		Receives kernel data buffer
+ *
+ * Returns:
+ *	0	on success
+ *	errno	on failure
+ */
+static int
+audio1575_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	audio1575_port_t	*port = arg;
+
+	_NOTE(ARGUNUSED(flag));
+
+	port->started = B_FALSE;
+	port->count = 0;
+	*fragfrp = port->fragfr;
+	*nfragsp = M1575_BD_NUMS;
+	*bufp = port->samp_kaddr;
+
+	mutex_enter(&port->statep->lock);
+	audio1575_reset_port(port);
+	mutex_exit(&port->statep->lock);
+
+	return (0);
+}
+
+
+/*
+ * audio1575_close()
+ *
+ * Description:
+ *	Closes an audio DMA engine that was previously opened.  Since
+ *	nobody is using it, we take this opportunity to possibly power
+ *	down the entire device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to shut down
+ */
+static void
+audio1575_close(void *arg)
+{
+	audio1575_port_t	*port = arg;
+	audio1575_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->lock);
+	audio1575_stop_port(port);
+	port->started = B_FALSE;
+	mutex_exit(&statep->lock);
+}
+
+/*
+ * audio1575_stop()
+ *
+ * Description:
+ *	This is called by the framework to stop a port that is
+ *	transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to stop
+ */
+static void
+audio1575_stop(void *arg)
+{
+	audio1575_port_t	*port = arg;
+	audio1575_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->lock);
+	if (port->started) {
+		audio1575_stop_port(port);
+	}
+	port->started = B_FALSE;
+	mutex_exit(&statep->lock);
+}
+
+/*
+ * audio1575_start()
+ *
+ * Description:
+ *	This is called by the framework to start a port transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to start
+ *
+ * Returns:
+ *	0 	on success (never fails, errno if it did)
+ */
+static int
+audio1575_start(void *arg)
+{
+	audio1575_port_t	*port = arg;
+	audio1575_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->lock);
+	if (!port->started) {
+		audio1575_start_port(port);
+		port->started = B_TRUE;
+	}
+	mutex_exit(&statep->lock);
+	return (0);
+}
+
+/*
+ * audio1575_format()
+ *
+ * Description:
+ *	Called by the framework to query the format for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	AUDIO_FORMAT_S16_LE
+ */
+static int
+audio1575_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+/*
+ * audio1575_channels()
+ *
+ * Description:
+ *	Called by the framework to query the channels for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	Number of channels for the device
+ */
+static int
+audio1575_channels(void *arg)
+{
+	audio1575_port_t *port = arg;
+
+	return (port->nchan);
+}
+
+/*
+ * audio1575_rate()
+ *
+ * Description:
+ *	Called by the framework to query the sample rate for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	48000
+ */
+static int
+audio1575_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
+}
+
+/*
+ * audio1575_count()
+ *
+ * Description:
+ *	This is called by the framework to get the engine's frame counter
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	frame count for current engine
+ */
+static uint64_t
+audio1575_count(void *arg)
+{
+	audio1575_port_t	*port = arg;
+	audio1575_state_t	*statep = port->statep;
+	uint64_t		val;
+
+	mutex_enter(&statep->lock);
+	audio1575_update_port(port);
+	val = port->count + port->picb;
+	mutex_exit(&statep->lock);
+
+	return (val);
+}
+
+/*
+ * audio1575_sync()
+ *
+ * Description:
+ *	This is called by the framework to synchronize DMA caches.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to sync
+ */
+static void
+audio1575_sync(void *arg, unsigned nframes)
+{
+	audio1575_port_t *port = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(port->samp_dmah, 0, 0, port->sync_dir);
+}
+
+/*
+ * audio1575_qlen()
+ *
+ * Description:
+ *	This is called by the framework to determine on-device queue length.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	hardware queue length not reported by count (0 for this device)
+ */
+static size_t
+audio1575_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (0);
+}
+
+
+/*
+ * audio1575_start_port()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ * Arguments:
+ *	audio1575_port_t	*port	Port of DMA engine to start.
+ */
+static void
+audio1575_start_port(audio1575_port_t *port)
+{
+	audio1575_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+
+	if (port->num == M1575_REC) {
+		/* ULi says do fifo resets here */
+		SET32(M1575_FIFOCR1_REG, M1575_FIFOCR1_PCMIRST);
+		CLR32(M1575_DMACR_REG, M1575_DMACR_PCMIPAUSE);
+		PUT8(M1575_PCMICR_REG, M1575_PCMICR_IOCE);
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMISTART);
+	} else {
+		CLR32(M1575_DMACR_REG, M1575_DMACR_PCMOPAUSE);
+		PUT8(M1575_PCMOCR_REG, M1575_PCMOCR_IOCE);
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMOSTART);
+	}
+}
+
+/*
+ * audio1575_stop_port()
+ *
+ * Description:
+ *	This routine stops the DMA engine.
+ *
+ * Arguments:
+ *	audio1575_port_t	*port	Port of DMA engine to stop.
+ */
+static void
+audio1575_stop_port(audio1575_port_t *port)
+{
+	audio1575_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+
+	if (port->num == M1575_REC) {
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMIPAUSE);
+	} else {
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMOPAUSE);
+	}
+}
+
+/*
+ * audio1575_reset_port()
+ *
+ * Description:
+ *	This routine resets the DMA engine pareparing it for work.
+ *
+ * Arguments:
+ *	audio1575_port_t	*port	Port of DMA engine to reset.
+ */
+static void
+audio1575_reset_port(audio1575_port_t *port)
+{
+	audio1575_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->lock));
+
+	port->civ = 0;
+	port->picb = 0;
+
+	if (statep->suspended)
+		return;
+
+	if (port->num == M1575_REC) {
+		/* Uli FIFO madness ... */
+		SET32(M1575_FIFOCR1_REG, M1575_FIFOCR1_PCMIRST);
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMIPAUSE);
+
+		PUT8(M1575_PCMICR_REG, 0);
+		PUT8(M1575_PCMICR_REG, M1575_CR_RR | M1575_CR_IOCE);
+
+		PUT32(M1575_PCMIBDBAR_REG, port->bdl_paddr);
+		PUT8(M1575_PCMILVIV_REG, M1575_BD_NUMS - 1);
+
+		CLR32(M1575_DMACR_REG, M1575_DMACR_PCMIPAUSE);
+
+	} else {
+
+		/* Uli FIFO madness ... */
+		SET32(M1575_FIFOCR1_REG, M1575_FIFOCR1_PCMORST);
+		SET32(M1575_DMACR_REG, M1575_DMACR_PCMOPAUSE);
+
+		PUT8(M1575_PCMOCR_REG, 0);
+		PUT8(M1575_PCMOCR_REG, M1575_CR_RR | M1575_CR_IOCE);
+
+		PUT32(M1575_PCMOBDBAR_REG, port->bdl_paddr);
+		PUT8(M1575_PCMOLVIV_REG, M1575_BD_NUMS - 1);
+
+		CLR32(M1575_DMACR_REG, M1575_DMACR_PCMOPAUSE);
+	}
+}
+
+/*
+ * audio1575_update_port()
+ *
+ * Description:
+ *	This routine updates the ports frame counter from hardware, and
+ *	gracefully handles wraps.
+ *
+ * Arguments:
+ *	audio1575_port_t	*port		The port to update.
+ */
+static void
+audio1575_update_port(audio1575_port_t *port)
+{
+	audio1575_state_t	*statep = port->statep;
+	uint8_t			civ;
+	uint16_t		picb;
+	unsigned		n;
+	int			civoff;
+	int			picoff;
+
+	if (port->num == M1575_REC) {
+		civoff = M1575_PCMICIV_REG;
+		picoff = M1575_PCMIPICB_REG;
+	} else {
+		civoff = M1575_PCMOCIV_REG;
+		picoff = M1575_PCMOPICB_REG;
+	}
+
+	if (statep->suspended) {
+		civ = 0;
+		picb = 0;
+	} else {
+		/*
+		 * We read the position counters, but we're careful to avoid
+		 * the situation where the position counter resets at the end
+		 * of a buffer.
+		 */
+		for (int i = 0; i < 2; i++) {
+			civ = GET8(civoff);
+			picb = GET16(picoff);
+			if (GET8(civoff) == civ) {
+				/*
+				 * Chip did not start a new index, so
+				 * the picb is valid.
+				 */
+				break;
+			}
+		}
+		if (civ >= port->civ) {
+			n = civ - port->civ;
+		} else {
+			n = civ + (M1575_BD_NUMS - port->civ);
+		}
+		port->count += (n * port->fragfr);
+	}
+	port->civ = civ;
+	port->picb = picb;
+}
+
+/*
+ * audio1575_attach()
+ *
+ * Description:
+ *	Attach an instance of the audio1575 driver. This routine does the
+ * 	device dependent attach tasks. When it is completed, it registers
+ *	with the audio framework.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audio1575_attach(dev_info_t *dip)
+{
+	audio1575_state_t	*statep;
+	audio_dev_t		*adev;
+	uint32_t		devid;
+	const char		*name;
+	const char		*rev;
+
+	/* allocate the soft state structure */
+	statep = kmem_zalloc(sizeof (*statep), KM_SLEEP);
+	ddi_set_driver_private(dip, statep);
+	statep->dip = dip;
+
+	/*
+	 * We want the micboost enabled by default as well.
+	 */
+	ddi_prop_update_int(DDI_DEV_T_NONE, dip, AC97_PROP_MICBOOST, 1);
+
+	/* allocate common audio dev structure */
+	adev = audio_dev_alloc(dip, 0);
+	if (adev == NULL) {
+		audio_dev_warn(NULL, "unable to allocate audio dev");
+		goto error;
+	}
+	statep->adev = adev;
+
+	/* map in the audio registers */
+	if (audio1575_map_regs(statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "couldn't map registers");
+		goto error;
+	}
+
+	if (audio1575_setup_intr(statep) != DDI_SUCCESS) {
+		/* message already noted */
+		goto error;
+	}
+
+	/* Enable PCI I/O and Memory Spaces */
+	audio1575_pci_enable(statep);
+
+	devid = (pci_config_get16(statep->pcih, PCI_CONF_VENID) << 16) |
+	    pci_config_get16(statep->pcih, PCI_CONF_DEVID);
+	switch (devid) {
+	case 0x10b95455:
+		name = "Uli M1575 AC'97";
+		rev = "M5455";
+		break;
+	default:
+		name = "Uli AC'97";
+		rev = "Unknown";
+		break;
+	}
+	/* set device information -- this should check PCI config space */
+	audio_dev_set_description(adev, name);
+	audio_dev_set_version(adev, rev);
+
+
+	/* allocate port structures */
+	if ((audio1575_alloc_port(statep, M1575_PLAY) != DDI_SUCCESS) ||
+	    (audio1575_alloc_port(statep, M1575_REC) != DDI_SUCCESS)) {
+		goto error;
+	}
+
+	statep->ac97 = ac97_alloc(dip, audio1575_read_ac97,
+	    audio1575_write_ac97, statep);
+	ASSERT(statep->ac97 != NULL);
+
+	if (audio1575_chip_init(statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to init chip");
+		goto error;
+	}
+
+	if (ac97_init(statep->ac97, adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ac'97 initialization failed");
+		goto error;
+	}
+
+	/* set up kernel statistics */
+	if ((statep->ksp = kstat_create(M1575_NAME,
+	    ddi_get_instance(dip), M1575_NAME, "controller",
+	    KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT)) != NULL) {
+		kstat_install(statep->ksp);
+	}
+
+	/* Enable PCI Interrupts */
+	pci_config_put8(statep->pcih, M1575_PCIMISC_REG, M1575_PCIMISC_INTENB);
+
+	/* enable audio interrupts */
+	if (ddi_intr_enable(statep->ih) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_intr_enable() failure");
+		goto error;
+	}
+
+	/* register with the framework */
+	if (audio_dev_register(adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "unable to register with framework");
+		goto error;
+	}
+
+	/* everything worked out, so report the device */
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	audio1575_destroy(statep);
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio1575_detach()
+ *
+ * Description:
+ *	Detach an instance of the audio1575 driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audio1575_detach(dev_info_t *dip)
+{
+	audio1575_state_t	*statep;
+
+	statep = ddi_get_driver_private(dip);
+
+	if (audio_dev_unregister(statep->adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	audio1575_destroy(statep);
+	return (DDI_SUCCESS);
+}
+
+/* *********************** Local Routines *************************** */
+
+/*
+ * audio1575_setup_intr()
+ *
+ * Description:
+ *	This routine initializes the audio driver's interrupt handle and
+ *	mutex.
+ *
+ * Arguments:
+ *	audio1575_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Interrupt handle & mutex initialized
+ *	DDI_FAILURE		Interrupt handle & mutex not initialized
+ */
+int
+audio1575_setup_intr(audio1575_state_t *statep)
+{
+	audio_dev_t		*adev;
+	dev_info_t		*dip;
+	uint_t			ipri;
+	int			actual;
+	int			rv;
+	int			itype;
+	int			count;
+	ddi_intr_handle_t	ih = NULL;
+
+	dip = statep->dip;
+	adev = statep->adev;
+
+	/* get supported interrupt types */
+	rv = ddi_intr_get_supported_types(dip, &itype);
+	if ((rv != DDI_SUCCESS) || (!(itype & DDI_INTR_TYPE_FIXED))) {
+		audio_dev_warn(adev, "Fixed type interrupts not supported");
+		return (DDI_FAILURE);
+	}
+
+	/* make sure we only have one fixed type interrupt */
+	rv = ddi_intr_get_nintrs(dip, DDI_INTR_TYPE_FIXED, &count);
+	if ((rv != DDI_SUCCESS) || (count != 1)) {
+		audio_dev_warn(adev, "No fixed interrupts");
+		return (DDI_FAILURE);
+	}
+
+	rv = ddi_intr_alloc(statep->dip, &ih, DDI_INTR_TYPE_FIXED,
+	    0, 1, &actual, DDI_INTR_ALLOC_STRICT);
+	if ((rv != DDI_SUCCESS) || (actual != 1)) {
+		audio_dev_warn(adev, "Can't alloc interrupt handle");
+		return (DDI_FAILURE);
+	}
+
+	/* test for a high level interrupt */
+	if (ddi_intr_get_pri(ih, &ipri) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "Can't get interrupt priority");
+		(void) ddi_intr_free(ih);
+		return (DDI_FAILURE);
+	}
+	if (ipri >= ddi_intr_get_hilevel_pri()) {
+		audio_dev_warn(adev, "Unsupported high level interrupt");
+		(void) ddi_intr_free(ih);
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_add_handler(ih, audio1575_intr, statep, NULL) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(adev, "Can't add interrupt handler");
+		(void) ddi_intr_free(ih);
+		return (DDI_FAILURE);
+	}
+
+	statep->ih = ih;
+	mutex_init(&statep->lock, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+	mutex_init(&statep->ac_lock, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio1575_alloc_port()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use.  It also configures the BDL lists properly
+ *	for use.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audio1575_alloc_port(audio1575_state_t *statep, int num)
+{
+	ddi_dma_cookie_t	cookie;
+	uint_t			count;
+	int			dir;
+	unsigned		caps;
+	char			*prop;
+	audio_dev_t		*adev;
+	audio1575_port_t	*port;
+	uint32_t		*kaddr;
+	uint32_t		paddr;
+	int			rc;
+	dev_info_t		*dip;
+
+	adev = statep->adev;
+	dip = statep->dip;
+
+	port = kmem_zalloc(sizeof (*port), KM_SLEEP);
+	statep->ports[num] = port;
+	port->num = num;
+	port->statep = statep;
+	port->started = B_FALSE;
+
+	if (num == M1575_REC) {
+		prop = "record-interrupts";
+		dir = DDI_DMA_READ;
+		caps = ENGINE_INPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORKERNEL;
+		port->nchan = 2;
+	} else {
+		prop = "play-interrupts";
+		dir = DDI_DMA_WRITE;
+		caps = ENGINE_OUTPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORDEV;
+		/* XXX: possibly support multichannel! */
+		port->nchan = 2;
+	}
+
+	port->intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, prop, M1575_INTS);
+
+	/* make sure the values are good */
+	if (port->intrs < M1575_MIN_INTS) {
+		audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+		    prop, port->intrs, M1575_INTS);
+		port->intrs = M1575_INTS;
+	} else if (port->intrs > M1575_MAX_INTS) {
+		audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+		    prop, port->intrs, M1575_INTS);
+		port->intrs = M1575_INTS;
+	}
+
+	/*
+	 * Figure out how much space we need.  Sample rate is 48kHz, and
+	 * we need to store 32 chunks.  (Note that this means that low
+	 * interrupt frequencies will require more RAM.  We could probably
+	 * do some cleverness to use a shorter BD list.)
+	 */
+	port->fragfr = 48000 / port->intrs;
+	port->fragfr = M1575_ROUNDUP(port->fragfr, M1575_MOD_SIZE);
+	port->samp_size = port->fragfr * port->nchan * 2;
+	port->samp_size *= M1575_BD_NUMS;
+
+	/* allocate dma handle */
+	rc = ddi_dma_alloc_handle(dip, &sample_buf_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->samp_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	/* allocate DMA buffer */
+	rc = ddi_dma_mem_alloc(port->samp_dmah, port->samp_size, &buf_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->samp_kaddr,
+	    &port->samp_size, &port->samp_acch);
+	if (rc == DDI_FAILURE) {
+		audio_dev_warn(adev, "dma_mem_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	/* bind DMA buffer */
+	rc = ddi_dma_addr_bind_handle(port->samp_dmah, NULL,
+	    port->samp_kaddr, port->samp_size, dir|DDI_DMA_CONSISTENT,
+	    DDI_DMA_SLEEP, NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev,
+		    "ddi_dma_addr_bind_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	port->samp_paddr = cookie.dmac_address;
+
+	/*
+	 * now, from here we allocate DMA memory for buffer descriptor list.
+	 * we allocate adjacent DMA memory for all DMA engines.
+	 */
+	rc = ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->bdl_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * we allocate all buffer descriptors lists in continuous dma memory.
+	 */
+	port->bdl_size = sizeof (m1575_bd_entry_t) * M1575_BD_NUMS;
+	rc = ddi_dma_mem_alloc(port->bdl_dmah, port->bdl_size,
+	    &dev_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
+	    &port->bdl_kaddr, &port->bdl_size, &port->bdl_acch);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_mem_alloc(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * Wire up the BD list.  We do this *before* binding the BD list
+	 * so that we don't have to do an extra ddi_dma_sync.
+	 */
+	paddr = port->samp_paddr;
+	kaddr = (void *)port->bdl_kaddr;
+	for (int i = 0; i < M1575_BD_NUMS; i++) {
+
+		/* set base address of buffer */
+		ddi_put32(port->bdl_acch, kaddr, paddr);
+		kaddr++;
+
+		/* set size in frames, and enable IOC interrupt */
+		ddi_put32(port->bdl_acch, kaddr,
+		    ((port->fragfr * port->nchan) | (1U << 31)));
+		kaddr++;
+
+		paddr += (port->fragfr * port->nchan * 2);
+	}
+
+	rc = ddi_dma_addr_bind_handle(port->bdl_dmah, NULL, port->bdl_kaddr,
+	    port->bdl_size, DDI_DMA_WRITE|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP,
+	    NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev, "addr_bind_handle failed");
+		return (DDI_FAILURE);
+	}
+	port->bdl_paddr = cookie.dmac_address;
+
+	port->engine = audio_engine_alloc(&audio1575_engine_ops, caps);
+	if (port->engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(port->engine, port);
+	audio_dev_add_engine(adev, port->engine);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio1575_free_port()
+ *
+ * Description:
+ *	This routine unbinds the DMA cookies, frees the DMA buffers,
+ *	deallocates the DMA handles.
+ *
+ * Arguments:
+ *	audio810_port_t	*port	The port structure for a DMA engine.
+ */
+static void
+audio1575_free_port(audio1575_port_t *port)
+{
+	if (port == NULL)
+		return;
+
+	if (port->engine) {
+		audio_dev_remove_engine(port->statep->adev, port->engine);
+		audio_engine_free(port->engine);
+	}
+	if (port->bdl_paddr) {
+		(void) ddi_dma_unbind_handle(port->bdl_dmah);
+	}
+	if (port->bdl_acch) {
+		ddi_dma_mem_free(&port->bdl_acch);
+	}
+	if (port->bdl_dmah) {
+		ddi_dma_free_handle(&port->bdl_dmah);
+	}
+	if (port->samp_paddr) {
+		(void) ddi_dma_unbind_handle(port->samp_dmah);
+	}
+	if (port->samp_acch) {
+		ddi_dma_mem_free(&port->samp_acch);
+	}
+	if (port->samp_dmah) {
+		ddi_dma_free_handle(&port->samp_dmah);
+	}
+	kmem_free(port, sizeof (*port));
+}
+
+/*
+ * audio1575_map_regs()
+ *
+ * Description:
+ *	The registers are mapped in.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audio1575_map_regs(audio1575_state_t *statep)
+{
+	dev_info_t		*dip = statep->dip;
+
+	/* Check for fault management capabilities */
+	if (DDI_FM_ACC_ERR_CAP(ddi_fm_capable(dip))) {
+		dev_attr.devacc_attr_access = DDI_FLAGERR_ACC;
+	}
+
+	/* map the M1575 Audio PCI Cfg Space */
+	if (pci_config_setup(dip, &statep->pcih) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "PCI config map failure");
+		goto error;
+	}
+
+	/* map the M1575 Audio registers in PCI IO Space */
+	if ((ddi_regs_map_setup(dip, M1575_AUDIO_IO_SPACE, &statep->regsp,
+	    0, 0, &dev_attr, &statep->regsh)) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "Audio IO mapping failure");
+		goto error;
+	}
+	return (DDI_SUCCESS);
+
+error:
+	audio1575_unmap_regs(statep);
+
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio1575_unmap_regs()
+ *
+ * Description:
+ *	This routine unmaps control registers.
+ *
+ * Arguments:
+ *	audio1575_state_t	*state	The device's state structure
+ */
+static void
+audio1575_unmap_regs(audio1575_state_t *statep)
+{
+	if (statep->regsh) {
+		ddi_regs_map_free(&statep->regsh);
+	}
+
+	if (statep->pcih) {
+		pci_config_teardown(&statep->pcih);
+	}
+}
+
+/*
+ * audio1575_chip_init()
+ *
+ * Description:
+ *	This routine initializes the M1575 AC97 audio controller and the AC97
+ *	codec.	The AC97 codec registers are programmed from codec_shadow[].
+ *	If we are not doing a restore, we initialize codec_shadow[], otherwise
+ *	we use the current values of shadow.	This routine expects that the
+ *	PCI IO and Memory spaces have been mapped and enabled already.
+ * Arguments:
+ *	audio1575_state_t	*state		The device's state structure
+ *						restore	from codec_shadow[]
+ * Returns:
+ *	DDI_SUCCESS	The hardware was initialized properly
+ *	DDI_FAILURE	The hardware couldn't be initialized properly
+ */
+static int
+audio1575_chip_init(audio1575_state_t *statep)
+{
+	uint32_t		ssr;
+	uint32_t		rtsr;
+	uint32_t		intrsr;
+	int 			i;
+	int			j;
+	uint8_t			clk_detect;
+	clock_t			ticks;
+	ddi_acc_handle_t	pcih;
+
+	/*
+	 * clear the interrupt control and status register
+	 * READ/WRITE/READ workaround required
+	 * for buggy hardware
+	 */
+
+	PUT32(M1575_INTRCR_REG, 0);
+	(void) GET32(M1575_INTRCR_REG);
+
+	intrsr = GET32(M1575_INTRSR_REG);
+	PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
+	(void) GET32(M1575_INTRSR_REG);
+
+	ticks = drv_usectohz(M1575_LOOP_CTR);
+
+	/*
+	 * SADA only supports stereo, so we set the channel bits
+	 * to "00" to select 2 channels.
+	 * will also set the following:
+	 *
+	 * Disable double rate enable
+	 * no SPDIF output selected
+	 * 16 bit audio record mode
+	 * 16 bit pcm out mode
+	 * PCM Out 6 chan mode FL FR CEN BL BR LFE
+	 * PCM Out 2 channel mode (00)
+	 */
+	for (i = 0; i < M1575_LOOP_CTR; i++) {
+		/* Reset the AC97 Codec	and default to 2 channel 16 bit mode */
+		PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
+		delay(ticks<<1);
+
+		/* Read the System Status Reg */
+		ssr = GET32(M1575_SSR_REG);
+
+		/* make sure and release the blocked reset bit */
+		if (ssr & M1575_SSR_RSTBLK) {
+			SET32(M1575_INTFCR_REG, M1575_INTFCR_RSTREL);
+			delay(ticks);
+
+			/* Read the System Status Reg */
+			ssr = GET32(M1575_SSR_REG);
+
+			/* make sure and release the blocked reset bit */
+			if (ssr & M1575_SSR_RSTBLK) {
+				return (DDI_FAILURE);
+			}
+
+			/* Reset the controller */
+			PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
+			delay(ticks);
+		}
+
+		/* according AC'97 spec, wait for codec reset */
+		for (j = 0; j < M1575_LOOP_CTR; j++) {
+			if ((GET32(M1575_SCR_REG) & M1575_SCR_COLDRST) == 0) {
+				break;
+			}
+			delay(ticks);
+		}
+
+		/* codec reset failed */
+		if (j >= M1575_LOOP_CTR) {
+			audio_dev_warn(statep->adev,
+			    "failure to reset codec");
+			return (DDI_FAILURE);
+		}
+
+		/*
+		 * Wait for FACRDY First codec ready. The hardware can
+		 * provide the state of
+		 * codec ready bit on SDATA_IN[0] and as reflected in
+		 * the Recv Tag Slot Reg.
+		 */
+		rtsr = GET32(M1575_RTSR_REG);
+		if (rtsr & M1575_RTSR_FACRDY) {
+			break;
+		} else { /* reset the status and wait for new status to set */
+			rtsr |= M1575_RTSR_FACRDY;
+			PUT32(M1575_RTSR_REG, rtsr);
+			drv_usecwait(10);
+		}
+	}
+
+	/* if we could not reset the AC97 codec then report failure */
+	if (i >= M1575_LOOP_CTR) {
+		audio_dev_warn(statep->adev,
+		    "no codec ready signal received");
+		return (DDI_FAILURE);
+	}
+
+	/* Magic code from ULi to Turn on the AC_LINK clock */
+	pcih = statep->pcih;
+	pci_config_put8(pcih, M1575_PCIACD_REG, 0);
+	pci_config_put8(pcih, M1575_PCIACD_REG, 4);
+	pci_config_put8(pcih, M1575_PCIACD_REG, 0);
+	(void) pci_config_get8(pcih, M1575_PCIACD_REG);
+	pci_config_put8(pcih, M1575_PCIACD_REG, 2);
+	pci_config_put8(pcih, M1575_PCIACD_REG, 0);
+	clk_detect = pci_config_get8(pcih, M1575_PCIACD_REG);
+
+	if (clk_detect != 1) {
+		audio_dev_warn(statep->adev, "No AC97 Clock Detected");
+		return (DDI_FAILURE);
+	}
+
+	/* Magic code from Uli to Init FIFO1 and FIFO2 */
+	PUT32(M1575_FIFOCR1_REG, 0x81818181);
+	PUT32(M1575_FIFOCR2_REG, 0x81818181);
+	PUT32(M1575_FIFOCR3_REG, 0x81818181);
+
+	/* Make sure that PCM in and PCM out are enabled */
+	SET32(M1575_INTFCR_REG, (M1575_INTFCR_PCMIENB | M1575_INTFCR_PCMOENB));
+
+	audio1575_dma_stop(statep);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio1575_dma_stop()
+ *
+ * Description:
+ *	This routine is used to put each DMA engine into the quiet state.
+ *
+ * Arguments:
+ *	audio1575_state_t *statep	The device's state structure
+ */
+static void
+audio1575_dma_stop(audio1575_state_t *statep)
+{
+	uint32_t	intrsr;
+	int		i;
+
+	if (statep->regsh == NULL) {
+		return;
+	}
+
+	/* pause bus master (needed for the following reset register) */
+	for (i = 0; i < M1575_LOOP_CTR; i++) {
+
+		SET32(M1575_DMACR_REG, M1575_DMACR_PAUSE_ALL);
+		if (GET32(M1575_DMACR_REG) & M1575_DMACR_PAUSE_ALL) {
+			break;
+		}
+		drv_usecwait(10);
+	}
+
+	if (i >= M1575_LOOP_CTR) {
+		audio_dev_warn(statep->adev, "failed to stop DMA");
+		return;
+	}
+
+	/* Pause bus master (needed for the following reset register) */
+	PUT8(M1575_PCMICR_REG, 0);
+	PUT8(M1575_PCMOCR_REG, 0);
+	PUT8(M1575_MICICR_REG, 0);
+	PUT8(M1575_CSPOCR_REG, 0);
+	PUT8(M1575_PCMI2CR_RR, 0);
+	PUT8(M1575_MICI2CR_RR, 0);
+
+	/* Reset the bus master registers for all DMA engines */
+	PUT8(M1575_PCMICR_REG, M1575_PCMICR_RR);
+	PUT8(M1575_PCMOCR_REG, M1575_PCMOCR_RR);
+	PUT8(M1575_MICICR_REG, M1575_MICICR_RR);
+	PUT8(M1575_CSPOCR_REG, M1575_CSPOCR_RR);
+	PUT8(M1575_PCMI2CR_REG, M1575_PCMI2CR_RR);
+	PUT8(M1575_MICI2CR_REG, M1575_MICI2CR_RR);
+
+	/* Reset FIFOS */
+	PUT32(M1575_FIFOCR1_REG, 0x81818181);
+	PUT32(M1575_FIFOCR2_REG, 0x81818181);
+	PUT32(M1575_FIFOCR3_REG, 0x81818181);
+
+	/* Clear Interrupts */
+	SET16(M1575_PCMISR_REG, M1575_SR_CLR);
+	SET16(M1575_PCMOSR_REG, M1575_SR_CLR);
+	SET16(M1575_MICISR_REG, M1575_SR_CLR);
+	SET16(M1575_CSPOSR_REG, M1575_SR_CLR);
+	SET16(M1575_PCMI2SR_REG, M1575_SR_CLR);
+	SET16(M1575_MICI2SR_REG, M1575_SR_CLR);
+
+	/*
+	 * clear the interrupt control and status register
+	 * READ/WRITE/READ workaround required
+	 * for buggy hardware
+	 */
+
+	PUT32(M1575_INTRCR_REG, 0);
+	(void) GET32(M1575_INTRCR_REG);
+
+	intrsr = GET32(M1575_INTRSR_REG);
+	PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
+	(void) GET32(M1575_INTRSR_REG);
+}
+
+/*
+ * audio1575_codec_sync()
+ *
+ * Description:
+ *	Serialize access to the AC97 audio mixer registers.
+ *
+ * Arguments:
+ *	audio1575_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Ready for an I/O access to the codec
+ *	DDI_FAILURE		An I/O access is currently in progress, can't
+ *				perform another I/O access.
+ */
+static int
+audio1575_codec_sync(audio1575_state_t *statep)
+{
+	/* do the Uli Shuffle ... */
+	for (int i = 0; i < M1575_LOOP_CTR; i++) {
+		/* Read the semaphore, and loop till we own it */
+		if ((GET32(M1575_CASR_REG) & 1) == 0) {
+			for (int j = 0; j < M1575_LOOP_CTR; j++) {
+				/* Wait for CWRSUCC 0x8 */
+				if (GET32(M1575_CSPSR_REG) &
+				    M1575_CSPSR_SUCC) {
+					return (DDI_SUCCESS);
+				}
+				drv_usecwait(1);
+			}
+		}
+		drv_usecwait(10);
+	}
+
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio1575_write_ac97()
+ *
+ * Description:
+ *	Set the specific AC97 Codec register.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *	uint16_t	data		The data want to be set
+ */
+static void
+audio1575_write_ac97(void *arg, uint8_t reg, uint16_t data)
+{
+	audio1575_state_t	*statep = arg;
+	int			i;
+
+	mutex_enter(&statep->ac_lock);
+
+	if (audio1575_codec_sync(statep) != DDI_SUCCESS) {
+		mutex_exit(&statep->ac_lock);
+		return;
+	}
+
+	/* write the data to WRITE to the lo word of the CPR register */
+	PUT16(M1575_CPR_REG, data);
+
+	/* write the address to WRITE to the hi word of the CPR register */
+	PUT16(M1575_CPR_REG+2, reg);
+
+	/* wait until command is completed sucessfully */
+	for (i = 0; i < M1575_LOOP_CTR; i++) {
+		/* Wait for Write Ready	0x01 */
+		if (GET32(M1575_CSPSR_REG) & M1575_CSPSR_WRRDY) {
+			break;
+		}
+		drv_usecwait(1);
+	}
+
+	mutex_exit(&statep->ac_lock);
+
+	if (i < M1575_LOOP_CTR) {
+		(void) audio1575_read_ac97(statep, reg);
+	}
+}
+
+/*
+ * audio1575_read_ac97()
+ *
+ * Description:
+ *	Get the specific AC97 Codec register. It also updates codec_shadow[]
+ *	with the register value.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *
+ * Returns:
+ *	Value of AC97 register.  (0xffff in failure situations).
+ */
+static uint16_t
+audio1575_read_ac97(void *arg, uint8_t reg)
+{
+	audio1575_state_t	*statep = arg;
+	uint16_t		addr = 0;
+	uint16_t		data = 0xffff;
+	int			i;
+
+	mutex_enter(&statep->ac_lock);
+	if ((audio1575_codec_sync(statep)) != DDI_SUCCESS) {
+		mutex_exit(&statep->ac_lock);
+		return (data);
+	}
+
+	/*
+	 * at this point we have the CASR semaphore
+	 * and the codec is r/w ready
+	 * OR in the READ opcode into the address field
+	 */
+
+	addr = (reg | M1575_CPR_READ);
+
+	/* write the address to READ to the hi word of the CPR register */
+	PUT16(M1575_CPR_REG+2, addr);
+
+	/* wait until command is completed sucessfully */
+	for (i = 0; i < M1575_LOOP_CTR; i++) {
+		/* Wait for Read Ready	0x02 */
+		if (GET32(M1575_CSPSR_REG) & M1575_CSPSR_RDRDY) {
+			break;
+		}
+		drv_usecwait(1);
+	}
+
+	if (i < M1575_LOOP_CTR) {
+		/* read back the data and address */
+		data = GET16(M1575_SPR_REG);
+		addr = GET16(M1575_SPR_REG+2);
+		if (addr != reg) {
+			data = 0xffff;
+		}
+	}
+
+	mutex_exit(&statep->ac_lock);
+	return (data);
+}
+
+/*
+ * audio1575_pci_enable()
+ *
+ * Description:
+ *	This routine Enables all PCI IO and MEMORY accesses
+ *
+ * Arguments:
+ *	audio1575_state_t *statep	 The device's state structure
+ */
+static void
+audio1575_pci_enable(audio1575_state_t *statep)
+{
+	uint16_t pcics_reg;
+
+	pcics_reg = pci_config_get16(statep->pcih, PCI_CONF_COMM);
+	pcics_reg |= (PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
+	pci_config_put16(statep->pcih, PCI_CONF_COMM, pcics_reg);
+}
+
+/*
+ * audio1575_pci_disable()
+ *
+ * Description:
+ *	This routine Disables all PCI IO and MEMORY accesses
+ *
+ * Arguments:
+ *	audio1575_state_t *statep	The device's state structure
+ */
+static void
+audio1575_pci_disable(audio1575_state_t *statep)
+{
+	uint16_t pcics_reg;
+
+	if (statep->pcih == NULL)
+		return;
+	pcics_reg = pci_config_get16(statep->pcih, PCI_CONF_COMM);
+	pcics_reg &= ~(PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
+	pci_config_put16(statep->pcih, PCI_CONF_COMM, pcics_reg);
+}
+
+/*
+ * audio1575_resume()
+ *
+ * Description:
+ *	Resume operation of the device after sleeping or hibernating.
+ *	Note that this should never fail, even if hardware goes wonky,
+ *	because the current PM framework will panic if it does.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was resumed
+ */
+static int
+audio1575_resume(dev_info_t *dip)
+{
+	audio1575_state_t	*statep;
+	audio_dev_t		*adev;
+
+	/* we've already allocated the state structure so get ptr */
+	statep = ddi_get_driver_private(dip);
+	adev = statep->adev;
+	ASSERT(!mutex_owned(&statep->lock));
+
+	if (audio1575_chip_init(statep) != DDI_SUCCESS) {
+		/*
+		 * Note that PM gurus say we should return
+		 * success here.  Failure of audio shouldn't
+		 * be considered FATAL to the system.  The
+		 * upshot is that audio will not progress.
+		 */
+		audio_dev_warn(adev, "DDI_RESUME failed to init chip");
+		return (DDI_SUCCESS);
+	}
+
+	/* allow ac97 operations again */
+	ac97_resume(statep->ac97);
+
+	mutex_enter(&statep->lock);
+
+	ASSERT(statep->suspended);
+	statep->suspended = B_FALSE;
+
+	for (int i = 0; i < M1575_NUM_PORTS; i++) {
+
+		audio1575_port_t *port = statep->ports[i];
+
+		if (port != NULL) {
+			/* reset framework DMA engine buffer */
+			if (port->engine != NULL) {
+				audio_engine_reset(port->engine);
+			}
+
+			/* reset and initialize hardware ports */
+			audio1575_reset_port(port);
+			if (port->started) {
+				audio1575_start_port(port);
+			} else {
+				audio1575_stop_port(port);
+			}
+		}
+	}
+	mutex_exit(&statep->lock);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio1575_suspend()
+ *
+ * Description:
+ *	Suspend an instance of the audio1575 driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was suspended
+ */
+static int
+audio1575_suspend(dev_info_t *dip)
+{
+	audio1575_state_t	*statep;
+
+	statep = ddi_get_driver_private(dip);
+
+	ac97_suspend(statep->ac97);
+
+	mutex_enter(&statep->lock);
+
+	statep->suspended = B_TRUE;
+
+	/*
+	 * stop all DMA operations
+	 */
+	audio1575_dma_stop(statep);
+
+	mutex_exit(&statep->lock);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio1575_destroy()
+ *
+ * Description:
+ *	This routine releases all resources held by the device instance,
+ *	as part of either detach or a failure in attach.
+ *
+ * Arguments:
+ *	audio1575_state_t	*state	The device soft state.
+ *
+ * Returns:
+ *	None
+ */
+void
+audio1575_destroy(audio1575_state_t *statep)
+{
+	ddi_acc_handle_t	pcih;
+
+	/* stop DMA engines */
+	audio1575_dma_stop(statep);
+
+	if (statep->regsh != NULL) {
+		/* reset the codec */
+		PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
+	}
+
+	if ((pcih = statep->pcih) != NULL) {
+		/* turn off the AC_LINK clock */
+		pci_config_put8(pcih, M1575_PCIACD_REG, 0);
+		pci_config_put8(pcih, M1575_PCIACD_REG, 4);
+		pci_config_put8(pcih, M1575_PCIACD_REG, 0);
+	}
+
+	/* Disable PCI I/O and Memory Spaces */
+	audio1575_pci_disable(statep);
+
+	if (statep->ih != NULL) {
+		(void) ddi_intr_disable(statep->ih);
+		(void) ddi_intr_remove_handler(statep->ih);
+		(void) ddi_intr_free(statep->ih);
+		mutex_destroy(&statep->lock);
+		mutex_destroy(&statep->ac_lock);
+	}
+
+	if (statep->ksp != NULL) {
+		kstat_delete(statep->ksp);
+	}
+
+	audio1575_free_port(statep->ports[M1575_PLAY]);
+	audio1575_free_port(statep->ports[M1575_REC]);
+
+	audio1575_unmap_regs(statep);
+
+	if (statep->ac97 != NULL) {
+		ac97_free(statep->ac97);
+	}
+
+	if (statep->adev != NULL) {
+		audio_dev_free(statep->adev);
+	}
+
+	kmem_free(statep, sizeof (*statep));
+}
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audio1575/audio1575.conf b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.conf
index 78b15ccbeb..cb3986c128 100644
--- a/usr/src/uts/sun/io/audio/sada/drv/audio1575/audio1575.conf
+++ b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.conf
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,25 +19,14 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Configuration file for the audio1575 audio driver.
 #
 # WARNING: This is an UNSTABLE configuration file. Its contents
 #		may change at any time.
 #
-# mixer-mode determines the mode the mixer audio personality module
-# sets for this driver when it is loaded. The mixerctl(1) command may
-# be used to change modes on the fly.
-#
-#	mixer-mode=1;	<-- mixer enabled
-#	mixer-mode=0;	<-- mixer disabled, old audio(7I) behavior
-#
-
-mixer-mode=1;
 
 #
 # play-interrupts sets the number of interrupts per second when playing.
@@ -58,17 +46,3 @@ mixer-mode=1;
 
 play-interrupts=175;
 record-interrupts=175;
-
-#
-# uncomment the cdrom=1 to enable the CDROM analog input in the
-# audio1575 driver .
-#
-#cdrom=1;
-#
-#
-# Uncomment reset-configuration to cause the audio1575 driver's state to
-# be reset to the default when the driver is loaded. Otherwise this state
-# is retained across driver unload/reload cycles, but not across reboots.
-#
-#reset-configuration=1;
-
diff --git a/usr/src/uts/common/io/audio/drv/audio1575/audio1575.h b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.h
new file mode 100644
index 0000000000..efdfb46cf6
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audio1575/audio1575.h
@@ -0,0 +1,468 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	AUDIO1575_H
+#define	AUDIO1575_H
+
+/*
+ * Header file for the audio1575 device driver
+ */
+
+/*
+ * Values returned by the AUDIO_GETDEV ioctl()
+ */
+#define	M1575_DEV_NAME			"SUNW,audio1575"
+#define	M1575_DEV_CONFIG		"onboard1"
+#define	M1575_DEV_VERSION		"a"
+
+/*
+ * Driver supported configuration information
+ */
+#define	M1575_NAME			"audio1575"
+#define	M1575_MOD_NAME			"M1575 audio driver"
+
+#define	M1575_INTS			(175)	/* default interrupt rate */
+#define	M1575_MIN_INTS			(25)	/* minimum interrupt rate */
+#define	M1575_MAX_INTS			(5000)	/* maximum interrupt rate */
+
+/*
+ * Implementation specific header file for the audio1575 device driver.
+ */
+
+/* Misc. defines */
+#define	M1575_CONFIG_DEVICE_ID			M1575_IDNUM
+#define	M1575_CONFIG_VENDOR_ID			(0x10b9)
+#define	M1575_CONFIG_SUBSYSTEM_ID		M1575_IDNUM
+#define	M1575_CONFIG_SUBSYSTEM_VENDOR_ID	M1575_CONFIG_VENDOR_ID
+#define	M1575_AUDIO_PCICFG_SPACE	(0)
+#define	M1575_AUDIO_IO_SPACE		(1)
+#define	M1575_AUDIO_MEM_SPACE		(2)
+
+#define	M1575_LOOP_CTR			(100)
+
+/* Gain and attenuation shift values */
+#define	M1575_GAIN_SHIFT3		(3)
+#define	M1575_GAIN_SHIFT4		(4)
+#define	M1575_BYTE_SHIFT		(8)
+
+/* audio direction */
+#define	M1575_PLAY			(0)
+#define	M1575_REC			(1)
+
+#define	M1575_DMA_PCM_IN		(1)
+#define	M1575_DMA_PCM_OUT		(2)
+
+/* last AC97 saved register */
+#define	M1575_LAST_AC_REG		(0x3a)
+
+/* Restore audio flags */
+#define	M1575_INIT_RESTORE		(0)
+#define	M1575_INIT_NO_RESTORE		~M1575_INIT_RESTORE
+
+/* AC97 codec shadow reg to index macro */
+#define	M1575_CODEC_REG(r)		((r) >> 1)
+
+/* play and record sample buffer counts */
+#define	M1575_PLAY_BUFS			(2)
+#define	M1575_PLAY_BUF_MSK		(M1575_PLAY_BUFS - 1)
+#define	M1575_REC_BUFS			(4)
+#define	M1575_REC_BUF_MSK		(M1575_REC_BUFS - 1)
+
+/* Buffer Descriptor  List defines */
+#define	M1575_BD_NUMS			(32)
+#define	M1575_NUM_PORTS			(2)
+#define	M1575_MOD_SIZE			(16)
+
+/* default buffer size */
+#define	M1575_BSIZE			(8*1024)
+#define	M1575_MOD_SIZE			(16)
+#define	M1575_PLAY_BUF_SZ		(1024)
+#define	M1575_RECORD_BUF_SZ		(1024)
+#define	M1575_BUF_MIN			(512)
+#define	M1575_BUF_MAX			(8192)
+
+/* kstat interrupt counter define */
+#define	M1575_KIOP(X)			((kstat_intr_t *)(X->ksp->ks_data))
+#define	M1575_ROUNDUP(x, algn)		(((x) + ((algn) - 1)) & ~((algn) - 1))
+
+/* PCI CFG SPACE REGISTERS for Audio (Device 29, Function 0) */
+#define	M1575_PCIPMR_REG	0x42	/* Power Capabilities 16 */
+#define	M1575_PCIPMCSR_REG	0x44	/* Power Cmd & Status 16 */
+#define	M1575_PCISCCR_REG	0x48	/* System Cfg Cntrl 16 */
+#define	M1575_PCIAPMUCR1_REG	0x54	/* Add. PMU Cntrl Reg  8 */
+#define	M1575_PCISCRR_REG	0x57	/* Scratch Reg  8 */
+#define	M1575_PCIMISC_REG	0x58	/* Misc Reg  8 */
+#define	M1575_PCIGCC_REG	0x59	/* Global Clk Control 16 */
+#define	M1575_PCIACD_REG	0x5C	/* AC97 Codec Detect  8 */
+#define	M1575_PCIMISC_REG	0x58	/* Misc Reg  8 */
+#define	M1575_PCIGCLK_REG	0x59	/* Misc Reg  8 */
+#define	M1575_PCIMSICTRL_REG	0x62	/* MSI Control Reg 16 */
+#define	M1575_PCIMSIADDR_REG	0x64	/* MSI Address Reg 32 */
+#define	M1575_PCIMSIDATA_REG	0x68	/* MSI Data Reg 16 */
+#define	M1575_PCIMSIMASK_REG	0x6C	/* MSI Data Reg 32 */
+#define	M1575_PCIMSIPEND_REG	0x70	/* MSI Pend Reg 32 */
+
+/* Bit definitions for PCI AC97 Clk detect Reg */
+#define	M1575_PCIACD_CLKDET	0x01
+#define	M1575_PCIMISC_INTENB	0x40
+#define	M1575_PCIINT_LINE	0x05
+
+/* Base Line Audio I/O Memory Registers */
+#define	M1575_SCR_REG		0x00	/* System Control Reg 32 */
+#define	M1575_SSR_REG		0x04	/* System System Reg 32 */
+#define	M1575_DMACR_REG		0x08	/* DMA Control Reg 32 */
+#define	M1575_FIFOCR1_REG	0x0C	/* FIFO 1 Control Reg 32 */
+#define	M1575_INTFCR_REG	0x10	/* Interface Ctrl Reg 32 */
+#define	M1575_INTRCR_REG	0x14	/* Interrupt Ctrl Reg 32 */
+#define	M1575_INTRSR_REG	0x18	/* Interrupt Status Reg 32 */
+#define	M1575_FIFOCR2_REG	0x1C	/* FIFO 2 Control Reg 32 */
+#define	M1575_CPR_REG		0x20	/* Cmd Port Reg 32 */
+#define	M1575_SPR_REG		0x24	/* Status Port Reg 32 */
+#define	M1575_FIFOCR3_REG	0x2C	/* FIFO 3 Control Reg 32 */
+#define	M1575_TTSR_REG		0x30	/* Tx Tag Slot Reg 32 */
+#define	M1575_RTSR_REG		0x34	/* Rx Tag Slot Reg 32 */
+#define	M1575_CSPSR_REG		0x38	/* CSP Status Reg 32 */
+#define	M1575_CASR_REG		0x3C	/* Codec Access Sem Reg 32 */
+
+/* PCM IN Registers */
+#define	M1575_PCMIBDBAR_REG	0x40 	/* 32 */
+#define	M1575_PCMICIV_REG	0x44 	/* 8 */
+#define	M1575_PCMILVIV_REG	0x45 	/* 8 */
+#define	M1575_PCMISR_REG	0x46 	/* 16 */
+#define	M1575_PCMIPICB_REG	0x48 	/* 16 */
+#define	M1575_PCMICR_REG	0x4B 	/* 8 */
+
+/* PCM OUT Registers */
+#define	M1575_PCMOBDBAR_REG	0x50 	/* 32 */
+#define	M1575_PCMOCIV_REG	0x54 	/* 8 */
+#define	M1575_PCMOLVIV_REG	0x55 	/* 8 */
+#define	M1575_PCMOSR_REG	0x56 	/* 16 */
+#define	M1575_PCMOPICB_REG	0x58 	/* 16 */
+#define	M1575_PCMOCR_REG	0x5B 	/* 8 */
+
+/* MIC In Registers */
+#define	M1575_MICIBDBAR_REG	0x60 	/* 32 */
+#define	M1575_MICICIV_REG	0x64 	/* 8 */
+#define	M1575_MICILVIV_REG	0x65 	/* 8 */
+#define	M1575_MICISR_REG	0x66 	/* 16 */
+#define	M1575_MICIPICB_REG	0x68 	/* 16 */
+#define	M1575_MICICR_REG	0x6B 	/* 8 */
+
+/* SPIDOF Registers */
+#define	M1575_CSPOBDBAR_REG	0x70 	/* 32 */
+#define	M1575_CSPOCIV_REG	0x74 	/* 8 */
+#define	M1575_CSPOLVIV_REG	0x75 	/* 8 */
+#define	M1575_CSPOSR_REG	0x76 	/* 16 */
+#define	M1575_CSPOPICB_REG	0x78 	/* 16 */
+#define	M1575_CSPOCR_REG	0x7B 	/* 8 */
+
+/* PCM IN2 Registers */
+#define	M1575_PCMI2BDBAR_REG	0xd0 	/* 32 */
+#define	M1575_PCMI2CIV_REG	0xd4 	/* 8 */
+#define	M1575_PCMI2LVIV_REG	0xd5 	/* 8 */
+#define	M1575_PCMI2SR_REG	0xd6 	/* 16 */
+#define	M1575_PCMI2PICB_REG	0xd8 	/* 16 */
+#define	M1575_PCMI2CR_REG	0xdB 	/* 8 */
+
+/* MIC2 IN2 Registers */
+#define	M1575_MICI2BDBAR_REG	0xe0 	/* 32 */
+#define	M1575_MICI2CIV_REG	0xe4 	/* 8 */
+#define	M1575_MICI2LVIV_REG	0xe5 	/* 8 */
+#define	M1575_MICI2SR_REG	0xe6 	/* 16 */
+#define	M1575_MICI2PICB_REG	0xe8 	/* 16 */
+#define	M1575_MICI2CR_REG	0xeB 	/* 8 */
+
+/* Bits of FIFO Control Register1 */
+#define	M1575_FIFOCR1_CSPORST	0x80000000 /* SPDIF Out Reset */
+#define	M1575_FIFOCR1_MICIRST	0x00800000 /* MIC In Reset */
+#define	M1575_FIFOCR1_PCMORST	0x00008000 /* PCM Out Reset */
+#define	M1575_FIFOCR1_PCMIRST	0x00000080 /* PCM In Reset */
+
+/* Bits of FIFO Control Register2 */
+#define	M1575_FIFOCR2_SPORST	0x80000000 /* SPDIF Out FIFO Reset */
+#define	M1575_FIFOCR2_SPIRST	0x00800000 /* SPDIF In  FIFO Reset */
+#define	M1575_FIFOCR2_LFEORST	0x00008000 /* LFE Out FIFO Reset */
+#define	M1575_FIFOCR2_CENORST	0x00000080 /* CENTER Out Reset */
+
+/* Bits of FIFO Control Register3 */
+#define	M1575_FIFOCR3_PCMI2RST	0x00800000 /* PCM In2 FIFO  Reset */
+#define	M1575_FIFOCR3_MICI2RST	0x00008000 /* MIC In2 FIFO Reset */
+#define	M1575_FIFOCR3_I2SIRST	0x00000080 /* I2S In FIFO Reset */
+
+/* Bits of DMA Control Register */
+#define	M1575_DMACR_PCMISTART	0x00000001
+#define	M1575_DMACR_PCMOSTART	0x00000002
+#define	M1575_DMACR_MICISTART	0x00000004
+#define	M1575_DMACR_CSPOSTART	0x00000008
+#define	M1575_DMACR_CENOSTART	0x00000010
+#define	M1575_DMACR_LFEOSTART	0x00000020
+#define	M1575_DMACR_SPISTART	0x00000040
+#define	M1575_DMACR_SPOSTART	0x00000080
+#define	M1575_DMACR_I2SISTART	0x00000100
+#define	M1575_DMACR_PCMI2START	0x00000200
+#define	M1575_DMACR_MICI2START	0x00000400
+#define	M1575_DMACR_PCMIPAUSE	0x00010000
+#define	M1575_DMACR_PCMOPAUSE	0x00020000
+#define	M1575_DMACR_MICIPAUSE	0x00040000
+#define	M1575_DMACR_CSPOPAUSE	0x00080000
+#define	M1575_DMACR_CENOPAUSE	0x00100000
+#define	M1575_DMACR_LFEOPAUSE	0x00200000
+#define	M1575_DMACR_SPIPAUSE	0x00400000
+#define	M1575_DMACR_SPOPAUSE	0x00800000
+#define	M1575_DMACR_I2SIPAUSE	0x01000000
+#define	M1575_DMACR_PCMI2PAUSE	0x02000000
+#define	M1575_DMACR_MICI2PAUSE	0x04000000
+
+#define	M1575_DMACR_PAUSE_ALL	0x07ff0000
+
+/* Bits of INTRSR Interrupt Status Register */
+#define	M1575_INTRSR_GPIOINTR	0x0000002
+#define	M1575_INTRSR_SPRINTR	0x0000020
+#define	M1575_INTRSR_CPRINTR	0x0000080
+#define	M1575_INTRSR_PCMIINTR   0x0010000
+#define	M1575_INTRSR_PCMOINTR 	0x0020000
+#define	M1575_INTRSR_MICIINTR  	0x0040000
+#define	M1575_INTRSR_CSPOINTR  	0x0080000
+#define	M1575_INTRSR_CENOINTR  	0x0100000
+#define	M1575_INTRSR_LFEOINTR  	0x0200000
+#define	M1575_INTRSR_SPIINTR	0x0400000
+#define	M1575_INTRSR_SPOINTR	0x0800000
+#define	M1575_INTRSR_I2SIINTR	0x1000000
+#define	M1575_INTRSR_PCMI2INTR	0x2000000
+#define	M1575_INTRSR_MICI2INTR	0x4000000
+
+#define	M1575_INTR_MASK (M1575_INTRSR_GPIOINTR |\
+	M1575_INTRSR_SPRINTR  |\
+	M1575_INTRSR_CPRINTR  |\
+	M1575_INTRSR_PCMIINTR |\
+	M1575_INTRSR_PCMOINTR |\
+	M1575_INTRSR_MICIINTR |\
+	M1575_INTRSR_CSPOINTR |\
+	M1575_INTRSR_CENOINTR |\
+	M1575_INTRSR_LFEOINTR |\
+	M1575_INTRSR_SPIINTR  |\
+	M1575_INTRSR_SPOINTR  |\
+	M1575_INTRSR_I2SIINTR |\
+	M1575_INTRSR_PCMI2INTR|\
+	M1575_INTRSR_MICI2INTR)
+
+#define	M1575_UNUSED_INTR_MASK (M1575_INTRSR_GPIOINTR |\
+	M1575_INTRSR_SPRINTR  |\
+	M1575_INTRSR_CPRINTR  |\
+	M1575_INTRSR_MICIINTR |\
+	M1575_INTRSR_CSPOINTR |\
+	M1575_INTRSR_CENOINTR |\
+	M1575_INTRSR_LFEOINTR |\
+	M1575_INTRSR_SPIINTR  |\
+	M1575_INTRSR_SPOINTR  |\
+	M1575_INTRSR_I2SIINTR |\
+	M1575_INTRSR_PCMI2INTR|\
+	M1575_INTRSR_MICI2INTR)
+
+/* Defines a generic clear for all MIC and PCM Status Registers */
+#define	M1575_SR_CLR		0x001e
+#define	M1575_SR_DMACS		0x0001
+
+/* Defines a generic RESET for all MIC and PCM Control Registers */
+#define	M1575_CR_IOCE		0x10
+#define	M1575_CR_RR		0x02
+
+/* Bits of PCM In Status Register */
+#define	M1575_PCMISR_DMACS	0x01 	/* DMACS=0 if DMA Engine is IDLE */
+#define	M1575_PCMISR_CELV	0x02
+#define	M1575_PCMISR_LVBCI	0x04
+#define	M1575_PCMISR_BCIS	0x08
+#define	M1575_PCMISR_FIFOE	0x10
+
+/* Bits in PCM In Control Register */
+#define	M1575_PCMICR_RR		0x02	/* Reset */
+#define	M1575_PCMICR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
+#define	M1575_PCMICR_IOCE	0x10	/* Intr On Completion Enable */
+
+/* Bits of PCM Out Status Register */
+#define	M1575_PCMOSR_DMACS	0x01	/* DMACS=0 if DMA Engine is IDLE */
+#define	M1575_PCMOSR_CELV	0x02
+#define	M1575_PCMOSR_LVBCI	0x04
+#define	M1575_PCMOSR_BCIS	0x08
+#define	M1575_PCMOSR_FIFOE	0x10
+
+/* Bits in PCM Out Control Register */
+#define	M1575_PCMOCR_RR		0x02	/* Reset */
+#define	M1575_PCMOCR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
+#define	M1575_PCMOCR_IOCE	0x10	/* Intr On Completion Enable */
+
+/* Bits of MIC In Status Register */
+#define	M1575_MICISR_DMACS	0x01	/* DMACS=0 if DMA Engine is IDLE */
+#define	M1575_MICISR_CELV	0x02
+#define	M1575_MICISR_LVBCI	0x04
+#define	M1575_MICISR_BCIS	0x08
+#define	M1575_MICISR_FIFOE	0x10
+
+/* Bits in PCM In Control Register */
+#define	M1575_MICICR_RR		0x02	/* Reset */
+#define	M1575_MICICR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
+#define	M1575_MICICR_IOCE	0x10	/* Intr On Completion Enable */
+
+/* Bits in System Control  Register */
+#define	M1575_SCR_WARMRST	0x00000001
+#define	M1575_SCR_COLDRST	0x00000002
+#define	M1575_SCR_DRENT		0x40000000
+#define	M1575_SCR_MSTRST	0x80000000
+
+/* Bits in System Status Register */
+#define	M1575_SSR_RSTBLK	0x00000002
+#define	M1575_SSR_FACCS_MSK	0x00000018
+#define	M1575_SSR_SCID		0x00000040
+
+/* Bits in Command Port Register */
+#define	M1575_CPR_ACSCS		0x0100 /* Audio Codec for cmd 1=codec 2 */
+#define	M1575_CPR_READ		0x0080
+
+/* Bits in Cmd Status Port Register */
+#define	M1575_CSPSR_SUCC	0x08	/* cmd successful */
+#define	M1575_CSPSR_RDRDY	0x02	/* ready for read cmd */
+#define	M1575_CSPSR_WRRDY	0x01	/* ready for write cmd */
+#define	M1575_PCMI2CR_RR	0x02	/* Reset */
+#define	M1575_MICI2CR_RR	0x02	/* Reset */
+#define	M1575_CSPOCR_RR		0x02	/* Reset */
+
+/* Bits in  Interface  Control  Register */
+#define	M1575_INTFCR_RSTREL		0x02000000
+#define	M1575_INTFCR_RSTBLK		0x00200000
+#define	M1575_INTFCR_MICENB		0x00100000
+#define	M1575_INTFCR_PCMIENB		0x00080000
+#define	M1575_INTFCR_MICI2ENB		0x00040000
+#define	M1575_INTFCR_PCMI2ENB		0x00020000
+#define	M1575_INTFCR_MICI2SEL		0x00008000
+#define	M1575_INTFCR_MICISEL		0x00004000
+#define	M1575_INTFCR_PCMOENB		0x00000002
+
+#define	M1575_INTRCR_CPRINTR		0x00000080
+#define	M1575_INTRCR_SPRINTR		0x00000020
+#define	M1575_INTRCR_GPIOINTR		0x00000002
+
+/* Bits of Recv Tag Slot Register */
+#define	M1575_RTSR_SACRDY	0x20u	/* 2nd Audio Codec Rdy */
+#define	M1575_RTSR_FACRDY	0x80u	/* 1st Audio Codec Rdy */
+
+/* Semaphore busy */
+#define	M1575_CASR_SEMBSY	0x80000000
+
+/*
+ * buffer descripter list entry, see M1575 datasheet
+ */
+#define	IOC 0x8000
+#define	BUP 0x4000
+
+struct m1575_bd_entry {
+	uint32_t buf_base;	/* the address of the buffer */
+	uint16_t buf_len;	/* the number of samples */
+	uint16_t buf_cmd;
+};
+typedef struct m1575_bd_entry m1575_bd_entry_t;
+
+struct audio1575_port {
+	struct audio1575_state	*statep;
+	ddi_dma_handle_t	samp_dmah;
+	ddi_acc_handle_t	samp_acch;
+	size_t			samp_size;
+	caddr_t			samp_kaddr;
+	uint32_t		samp_paddr;
+
+	ddi_dma_handle_t	bdl_dmah;
+	ddi_acc_handle_t	bdl_acch;
+	size_t			bdl_size;
+	caddr_t			bdl_kaddr;
+	uint32_t		bdl_paddr;
+
+	int			num;
+	unsigned		intrs;
+	unsigned		fragfr;
+	uint64_t		count;
+	uint8_t			nchan;
+
+	uint8_t			civ;
+	uint16_t		picb;
+	unsigned		sync_dir;
+
+	boolean_t		started;
+
+	audio_engine_t		*engine;
+};
+typedef struct audio1575_port audio1575_port_t;
+
+
+/*
+ * audio1575_state_t per instance state and operation data
+ */
+struct audio1575_state	{
+	kmutex_t		lock;			/* intr mutex */
+	kmutex_t		ac_lock;		/* ac'97 mutex */
+	dev_info_t		*dip;			/* dev instance ptr */
+	audio_dev_t		*adev;			/* audio handle */
+	ac97_t			*ac97;			/* ac'97 handle */
+	audio1575_port_t	*ports[2];		/* DMA engines */
+
+	ddi_intr_handle_t	ih;			/* intr handle */
+
+	ddi_acc_handle_t	pcih;			/* pci config space */
+
+	ddi_acc_handle_t	regsh;			/* audio i/o regs */
+	caddr_t			regsp;			/* base of i/o regs */
+
+	kstat_t			*ksp;			/* kernel statistics */
+
+	boolean_t		suspended;		/* if DDI_SUSPENDed */
+};
+typedef struct audio1575_state audio1575_state_t;
+
+/* audio i/o register macros */
+#define	GET8(reg)						\
+	ddi_get8(statep->regsh, (void *)(statep->regsp + (reg)))
+
+#define	GET16(reg)						\
+	ddi_get16(statep->regsh, (void *)(statep->regsp + (reg)))
+
+#define	GET32(reg)						\
+	ddi_get32(statep->regsh, (void *)(statep->regsp + (reg)))
+
+#define	PUT8(reg, val)						\
+	ddi_put8(statep->regsh,	(void *)(statep->regsp + (reg)), (val))
+
+#define	PUT16(reg, val)						\
+	ddi_put16(statep->regsh, (void *)(statep->regsp + (reg)), (val))
+
+#define	PUT32(reg, val)						\
+	ddi_put32(statep->regsh, (void *)(statep->regsp + (reg)), (val))
+
+#define	SET8(reg, bit)		PUT8(reg, GET8(reg) | (bit))
+#define	SET16(reg, bit)		PUT16(reg, GET16(reg) | (bit))
+#define	SET32(reg, bit)		PUT32(reg, GET32(reg) | (bit))
+#define	CLR8(reg, bit)		PUT8(reg, GET8(reg) & ~(bit))
+#define	CLR16(reg, bit)		PUT16(reg, GET16(reg) & ~(bit))
+#define	CLR32(reg, bit)		PUT32(reg, GET32(reg) & ~(bit))
+
+
+#endif	/* AUDIO1575_H */
diff --git a/usr/src/uts/common/io/audio/drv/audio810/audio810.c b/usr/src/uts/common/io/audio/drv/audio810/audio810.c
new file mode 100644
index 0000000000..9d186df9e0
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audio810/audio810.c
@@ -0,0 +1,1971 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+/*
+ * audio810 Audio Driver
+ *
+ * The driver is primarily targeted at providing audio support for the
+ * W1100z and W2100z systems, which use the AMD 8111 audio core and
+ * the Realtek ALC 655 codec. The ALC 655 chip supports only fixed 48k
+ * sample rate. However, the audio core of AMD 8111 is completely
+ * compatible to the Intel ICHx chips (Intel 8x0 chipsets), so the
+ * driver can work for the ICHx.  We only support the 48k maximum
+ * rate, since we only have a single PCM out channel.
+ *
+ * The AMD 8111 audio core, as an AC'97 controller, has independent
+ * channels for PCM in, PCM out, mic in, modem in, and modem out.
+ * The AC'97 controller is a PCI bus master with scatter/gather
+ * support. Each channel has a DMA engine. Currently, we use only
+ * the PCM in and PCM out channels. Each DMA engine uses one buffer
+ * descriptor list. And the buffer descriptor list is an array of up
+ * to 32 entries, each of which describes a data buffer. Each entry
+ * contains a pointer to a data buffer, control bits, and the length
+ * of the buffer being pointed to, where the length is expressed as
+ * the number of samples. This, combined with the 16-bit sample size,
+ * gives the actual physical length of the buffer.
+ *
+ * A workaround for the AD1980 and AD1985 codec:
+ *	Most vendors connect the surr-out of the codecs to the line-out jack.
+ *	So far we haven't found which vendors don't do that. So we assume that
+ *	all vendors swap the surr-out and the line-out outputs. So we need swap
+ *	the two outputs. But we still internally process the
+ *	"ad198x-swap-output" property. If someday some vendors do not swap the
+ *	outputs, we would set "ad198x-swap-output = 0" in the
+ *	/kernel/drv/audio810.conf file, and unload and reload the audio810
+ *	driver (or reboot).
+ *
+ * 	NOTE:
+ * 	This driver depends on the drv/audio and misc/ac97
+ * 	modules being loaded first.
+ */
+#include <sys/types.h>
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/conf.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/pci.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include "audio810.h"
+
+/*
+ * Module linkage routines for the kernel
+ */
+static int audio810_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audio810_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+static int audio810_ddi_quiesce(dev_info_t *);
+
+/*
+ * Entry point routine prototypes
+ */
+static int audio810_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void audio810_close(void *);
+static int audio810_start(void *);
+static void audio810_stop(void *);
+static int audio810_format(void *);
+static int audio810_channels(void *);
+static int audio810_rate(void *);
+static uint64_t audio810_count(void *);
+static void audio810_sync(void *, unsigned);
+static size_t audio810_qlen(void *);
+
+static audio_engine_ops_t audio810_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	audio810_open,
+	audio810_close,
+	audio810_start,
+	audio810_stop,
+	audio810_count,
+	audio810_format,
+	audio810_channels,
+	audio810_rate,
+	audio810_sync,
+	audio810_qlen
+};
+
+/*
+ * interrupt handler
+ */
+static uint_t	audio810_intr(caddr_t);
+
+/*
+ * Local Routine Prototypes
+ */
+static int audio810_attach(dev_info_t *);
+static int audio810_resume(dev_info_t *);
+static int audio810_detach(dev_info_t *);
+static int audio810_suspend(dev_info_t *);
+
+static int audio810_alloc_port(audio810_state_t *, int, uint8_t);
+static void audio810_start_port(audio810_port_t *);
+static void audio810_stop_port(audio810_port_t *);
+static void audio810_reset_port(audio810_port_t *);
+static void audio810_update_port(audio810_port_t *);
+static int audio810_codec_sync(audio810_state_t *);
+static void audio810_write_ac97(void *, uint8_t, uint16_t);
+static uint16_t audio810_read_ac97(void *, uint8_t);
+static int audio810_map_regs(dev_info_t *, audio810_state_t *);
+static void audio810_unmap_regs(audio810_state_t *);
+static void audio810_stop_dma(audio810_state_t *);
+static int audio810_chip_init(audio810_state_t *);
+static void audio810_destroy(audio810_state_t *);
+
+/*
+ * Global variables, but used only by this file.
+ */
+
+/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
+static char	*audio810_name = I810_NAME;
+
+
+/*
+ * DDI Structures
+ */
+
+/* Device operations structure */
+static struct dev_ops audio810_dev_ops = {
+	DEVO_REV,		/* devo_rev */
+	0,			/* devo_refcnt */
+	NULL,			/* devo_getinfo */
+	nulldev,		/* devo_identify - obsolete */
+	nulldev,		/* devo_probe */
+	audio810_ddi_attach,	/* devo_attach */
+	audio810_ddi_detach,	/* devo_detach */
+	nodev,			/* devo_reset */
+	NULL,			/* devi_cb_ops */
+	NULL,			/* devo_bus_ops */
+	NULL,			/* devo_power */
+	audio810_ddi_quiesce,	/* devo_quiesce */
+};
+
+/* Linkage structure for loadable drivers */
+static struct modldrv audio810_modldrv = {
+	&mod_driverops,		/* drv_modops */
+	I810_MOD_NAME,		/* drv_linkinfo */
+	&audio810_dev_ops,	/* drv_dev_ops */
+};
+
+/* Module linkage structure */
+static struct modlinkage audio810_modlinkage = {
+	MODREV_1,			/* ml_rev */
+	(void *)&audio810_modldrv,	/* ml_linkage */
+	NULL				/* NULL terminates the list */
+};
+
+/*
+ * device access attributes for register mapping
+ */
+static struct ddi_device_acc_attr dev_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static struct ddi_device_acc_attr buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * DMA attributes of buffer descriptor list
+ */
+static ddi_dma_attr_t	bdlist_dma_attr = {
+	DMA_ATTR_V0,	/* version */
+	0,		/* addr_lo */
+	0xffffffff,	/* addr_hi */
+	0x0000ffff,	/* count_max */
+	8,		/* align, BDL must be aligned on a 8-byte boundary */
+	0x3c,		/* burstsize */
+	8,		/* minxfer, set to the size of a BDlist entry */
+	0x0000ffff,	/* maxxfer */
+	0x00000fff,	/* seg, set to the RAM pagesize of intel platform */
+	1,		/* sgllen, there's no scatter-gather list */
+	8,		/* granular, set to the value of minxfer */
+	0		/* flags, use virtual address */
+};
+
+/*
+ * DMA attributes of buffers to be used to receive/send audio data
+ */
+static ddi_dma_attr_t	sample_buf_dma_attr = {
+	DMA_ATTR_V0,
+	0,		/* addr_lo */
+	0xffffffff,	/* addr_hi */
+	0x0001ffff,	/* count_max */
+	4,		/* align, data buffer is aligned on a 4-byte boundary */
+	0x3c,		/* burstsize */
+	4,		/* minxfer, set to the size of a sample data */
+	0x0001ffff,	/* maxxfer */
+	0x0001ffff,	/* seg */
+	1,		/* sgllen, no scatter-gather */
+	4,		/* granular, set to the value of minxfer */
+	0,		/* flags, use virtual address */
+};
+
+/*
+ * _init()
+ *
+ * Description:
+ *	Driver initialization, called when driver is first loaded.
+ *	This is how access is initially given to all the static structures.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_install() status, see mod_install(9f)
+ */
+int
+_init(void)
+{
+	int	error;
+
+	audio_init_ops(&audio810_dev_ops, I810_NAME);
+
+	if ((error = mod_install(&audio810_modlinkage)) != 0) {
+		audio_fini_ops(&audio810_dev_ops);
+	}
+
+	return (error);
+}
+
+/*
+ * _fini()
+ *
+ * Description:
+ *	Module de-initialization, called when the driver is to be unloaded.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_remove() status, see mod_remove(9f)
+ */
+int
+_fini(void)
+{
+	int		error;
+
+	if ((error = mod_remove(&audio810_modlinkage)) != 0) {
+		return (error);
+	}
+
+	/* clean up ops */
+	audio_fini_ops(&audio810_dev_ops);
+
+	return (0);
+}
+
+/*
+ * _info()
+ *
+ * Description:
+ *	Module information, returns information about the driver.
+ *
+ * Arguments:
+ *	modinfo		*modinfop	Pointer to the opaque modinfo structure
+ *
+ * Returns:
+ *	mod_info() status, see mod_info(9f)
+ */
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&audio810_modlinkage, modinfop));
+}
+
+
+/* ******************* Driver Entry Points ********************************* */
+
+/*
+ * audio810_ddi_attach()
+ *
+ * Description:
+ *	Implements the DDI attach(9e) entry point.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audio810_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audio810_attach(dip));
+
+	case DDI_RESUME:
+		return (audio810_resume(dip));
+	}
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio810_ddi_detach()
+ *
+ * Description:
+ *	Implements the detach(9e) entry point.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *	ddi_detach_cmd_t	cmd	Detach command
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audio810_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audio810_detach(dip));
+
+	case DDI_SUSPEND:
+		return (audio810_suspend(dip));
+	}
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio810_ddi_quiesce()
+ *
+ * Description:
+ *	Implements the quiesce(9e) entry point.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was quiesced
+ *	DDI_FAILURE	The driver couldn't be quiesced
+ */
+static int
+audio810_ddi_quiesce(dev_info_t *dip)
+{
+	audio810_state_t	*statep;
+
+	if ((statep = ddi_get_driver_private(dip)) == NULL)
+		return (DDI_FAILURE);
+
+	audio810_stop_dma(statep);
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_intr()
+ *
+ * Description:
+ *	Interrupt service routine for both play and record. For play we
+ *	get the next buffers worth of audio. For record we send it on to
+ *	the mixer.
+ *
+ *	Each of buffer descriptor has a field IOC(interrupt on completion)
+ *	When both this and the IOC bit of correspondent dma control register
+ *	is set, it means that the controller should issue an interrupt upon
+ *	completion of this buffer.
+ *	(AMD 8111 hypertransport I/O hub data sheet. 3.8.3 page 71)
+ *
+ * Arguments:
+ *	caddr_t		arg	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+audio810_intr(caddr_t arg)
+{
+	audio810_state_t	*statep;
+	uint16_t		gsr;
+
+	statep = (void *)arg;
+	mutex_enter(&statep->inst_lock);
+
+	if (statep->suspended) {
+		mutex_exit(&statep->inst_lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	gsr = I810_BM_GET32(I810_REG_GSR);
+
+	/* check if device is interrupting */
+	if ((gsr & I810_GSR_USE_INTR) == 0) {
+		mutex_exit(&statep->inst_lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	for (int pnum = 0; pnum < I810_NUM_PORTS; pnum++) {
+		audio810_port_t	*port;
+		uint8_t		regoff, index;
+
+		port = statep->ports[pnum];
+		if (port == NULL) {
+			continue;
+		}
+		regoff = port->regoff;
+
+		if (!(I810_BM_GET8(port->stsoff) & I810_BM_SR_BCIS))
+			continue;
+
+		/* update the LVI -- we just set it to the current value - 1 */
+		index = I810_BM_GET8(regoff + I810_OFFSET_CIV);
+		index = (index - 1) % I810_BD_NUMS;
+
+		I810_BM_PUT8(regoff + I810_OFFSET_LVI, index);
+
+		/* clear any interrupt */
+		I810_BM_PUT8(port->stsoff,
+		    I810_BM_SR_LVBCI | I810_BM_SR_BCIS | I810_BM_SR_FIFOE);
+	}
+
+	/* update the kernel interrupt statistics */
+	if (statep->ksp) {
+		I810_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	mutex_exit(&statep->inst_lock);
+
+	/* notify the framework */
+	if (gsr & I810_GSR_INTR_PIN) {
+		audio_engine_produce(statep->ports[I810_PCM_IN]->engine);
+	}
+	if (gsr & I810_GSR_INTR_POUT) {
+		audio_engine_consume(statep->ports[I810_PCM_OUT]->engine);
+	}
+
+	return (DDI_INTR_CLAIMED);
+}
+
+/*
+ * audio810_open()
+ *
+ * Description:
+ *	Opens a DMA engine for use.
+ *
+ * Arguments:
+ *	void		*arg		The DMA engine to set up
+ *	int		flag		Open flags
+ *	unsigned	*fragfrp	Receives number of frames per fragment
+ *	unsigned	*nfragsp	Receives number of fragments
+ *	caddr_t		*bufp		Receives kernel data buffer
+ *
+ * Returns:
+ *	0	on success
+ *	errno	on failure
+ */
+static int
+audio810_open(void *arg, int flag, unsigned *fragfrp, unsigned *nfragsp,
+    caddr_t *bufp)
+{
+	audio810_port_t	*port = arg;
+
+	_NOTE(ARGUNUSED(flag));
+
+	port->started = B_FALSE;
+	port->count = 0;
+	*fragfrp = port->fragfr;
+	*nfragsp = port->nfrag;
+	*bufp = port->samp_kaddr;
+
+	mutex_enter(&port->statep->inst_lock);
+	audio810_reset_port(port);
+	mutex_exit(&port->statep->inst_lock);
+
+	return (0);
+}
+
+/*
+ * audio810_close()
+ *
+ * Description:
+ *	Closes an audio DMA engine that was previously opened.  Since
+ *	nobody is using it, we take this opportunity to possibly power
+ *	down the entire device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to shut down
+ */
+static void
+audio810_close(void *arg)
+{
+	audio810_port_t		*port = arg;
+	audio810_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	audio810_stop_port(port);
+	port->started = B_FALSE;
+	mutex_exit(&statep->inst_lock);
+}
+
+/*
+ * audio810_stop()
+ *
+ * Description:
+ *	This is called by the framework to stop a port that is
+ *	transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to stop
+ */
+static void
+audio810_stop(void *arg)
+{
+	audio810_port_t		*port = arg;
+	audio810_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	if (port->started) {
+		audio810_stop_port(port);
+	}
+	port->started = B_FALSE;
+	mutex_exit(&statep->inst_lock);
+}
+
+/*
+ * audio810_start()
+ *
+ * Description:
+ *	This is called by the framework to start a port transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to start
+ *
+ * Returns:
+ *	0 	on success (never fails, errno if it did)
+ */
+static int
+audio810_start(void *arg)
+{
+	audio810_port_t		*port = arg;
+	audio810_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	if (!port->started) {
+		audio810_start_port(port);
+		port->started = B_TRUE;
+	}
+	mutex_exit(&statep->inst_lock);
+	return (0);
+}
+
+/*
+ * audio810_format()
+ *
+ * Description:
+ *	This is called by the framework to query the format of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	Format of the device (fixed at AUDIO_FORMAT_S16_LE)
+ */
+static int
+audio810_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+/*
+ * audio810_channels()
+ *
+ * Description:
+ *	This is called by the framework to query the num channels of
+ *	the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	0 number of channels for device
+ */
+static int
+audio810_channels(void *arg)
+{
+	audio810_port_t	*port = arg;
+
+	return (port->nchan);
+}
+
+/*
+ * audio810_rate()
+ *
+ * Description:
+ *	This is called by the framework to query the rate of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	Rate of device (fixed at 48000 Hz)
+ */
+static int
+audio810_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
+}
+
+/*
+ * audio810_count()
+ *
+ * Description:
+ *	This is called by the framework to get the engine's frame counter
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	frame count for current engine
+ */
+static uint64_t
+audio810_count(void *arg)
+{
+	audio810_port_t		*port = arg;
+	audio810_state_t	*statep = port->statep;
+	uint64_t		val;
+	uint16_t		picb;
+	uint64_t		count;
+	uint8_t			nchan;
+
+	mutex_enter(&statep->inst_lock);
+	audio810_update_port(port);
+	count = port->count;
+	picb = port->picb;
+	nchan = port->nchan;
+	mutex_exit(&statep->inst_lock);
+
+	if (statep->quirk == QUIRK_SIS7012) {
+		val = count + picb / (2 * nchan);
+	} else {
+		val = count + (picb / nchan);
+	}
+
+	return (val);
+}
+
+/*
+ * audio810_sync()
+ *
+ * Description:
+ *	This is called by the framework to synchronize DMA caches.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to sync
+ */
+static void
+audio810_sync(void *arg, unsigned nframes)
+{
+	audio810_port_t *port = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(port->samp_dmah, 0, 0, port->sync_dir);
+}
+
+/*
+ * audio810_qlen()
+ *
+ * Description:
+ *	This is called by the framework to determine on-device queue length.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	hardware queue length not reported by count (0 for this device)
+ */
+static size_t
+audio810_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (0);
+}
+
+/* *********************** Local Routines *************************** */
+
+/*
+ * audio810_start_port()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ * Arguments:
+ *	audio810_port_t	*port		Port of DMA engine to start.
+ */
+static void
+audio810_start_port(audio810_port_t *port)
+{
+	audio810_state_t	*statep = port->statep;
+	uint8_t			cr;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+
+	cr = I810_BM_GET8(port->regoff + I810_OFFSET_CR);
+	cr |= I810_BM_CR_IOCE;
+	I810_BM_PUT8(port->regoff + I810_OFFSET_CR, cr);
+	cr |= I810_BM_CR_RUN;
+	I810_BM_PUT8(port->regoff + I810_OFFSET_CR, cr);
+}
+
+/*
+ * audio810_stop_port()
+ *
+ * Description:
+ *	This routine stops the DMA engine.
+ *
+ * Arguments:
+ *	audio810_port_t	*port		Port of DMA engine to stop.
+ */
+static void
+audio810_stop_port(audio810_port_t *port)
+{
+	audio810_state_t	*statep = port->statep;
+	uint8_t			cr;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+
+	cr = I810_BM_GET8(port->regoff + I810_OFFSET_CR);
+	cr &= ~I810_BM_CR_RUN;
+	I810_BM_PUT8(port->regoff + I810_OFFSET_CR, cr);
+}
+
+/*
+ * audio810_reset_port()
+ *
+ * Description:
+ *	This routine resets the DMA engine pareparing it for work.
+ *
+ * Arguments:
+ *	audio810_port_t	*port		Port of DMA engine to reset.
+ */
+static void
+audio810_reset_port(audio810_port_t *port)
+{
+	audio810_state_t	*statep = port->statep;
+	uint32_t		gcr;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	port->civ = 0;
+	port->picb = 0;
+
+	if (statep->suspended)
+		return;
+
+	/*
+	 * Make sure we put once in stereo, to ensure we always start from
+	 * front left.
+	 */
+	if (port->num == I810_PCM_OUT) {
+
+		if (statep->quirk == QUIRK_SIS7012) {
+			/*
+			 * SiS 7012 needs its own special multichannel config.
+			 */
+			gcr = I810_BM_GET32(I810_REG_GCR);
+			gcr &= ~I810_GCR_SIS_CHANNELS_MASK;
+			I810_BM_PUT32(I810_REG_GCR, gcr);
+			delay(drv_usectohz(50000));	/* 50 msec */
+
+			switch (statep->maxch) {
+			case 2:
+				gcr |= I810_GCR_SIS_2_CHANNELS;
+				break;
+			case 4:
+				gcr |= I810_GCR_SIS_4_CHANNELS;
+				break;
+			case 6:
+				gcr |= I810_GCR_SIS_6_CHANNELS;
+				break;
+			}
+			I810_BM_PUT32(I810_REG_GCR, gcr);
+			delay(drv_usectohz(50000));	/* 50 msec */
+
+			/*
+			 * SiS 7012 has special unmute bit.
+			 */
+			I810_BM_PUT8(I810_REG_SISCTL, I810_SISCTL_UNMUTE);
+
+		} else {
+
+			/*
+			 * All other devices work the same.
+			 */
+			gcr = I810_BM_GET32(I810_REG_GCR);
+			gcr &= ~I810_GCR_CHANNELS_MASK;
+
+			I810_BM_PUT32(I810_REG_GCR, gcr);
+			delay(drv_usectohz(50000));	/* 50 msec */
+
+			switch (statep->maxch) {
+			case 2:
+				gcr |= I810_GCR_2_CHANNELS;
+				break;
+			case 4:
+				gcr |= I810_GCR_4_CHANNELS;
+				break;
+			case 6:
+				gcr |= I810_GCR_6_CHANNELS;
+				break;
+			}
+			I810_BM_PUT32(I810_REG_GCR, gcr);
+			delay(drv_usectohz(50000));	/* 50 msec */
+		}
+	}
+
+	/*
+	 * Perform full reset of the engine, but leave it turned off.
+	 */
+	I810_BM_PUT8(port->regoff + I810_OFFSET_CR, 0);
+	I810_BM_PUT8(port->regoff + I810_OFFSET_CR, I810_BM_CR_RST);
+
+	/* program the offset of the BD list */
+	I810_BM_PUT32(port->regoff + I810_OFFSET_BD_BASE, port->bdl_paddr);
+
+	/* we set the last index to the full count -- all buffers are valid */
+	I810_BM_PUT8(port->regoff + I810_OFFSET_LVI, I810_BD_NUMS - 1);
+}
+
+/*
+ * audio810_update_port()
+ *
+ * Description:
+ *	This routine updates the ports frame counter from hardware, and
+ *	gracefully handles wraps.
+ *
+ * Arguments:
+ *	audio810_port_t	*port		The port to update.
+ */
+static void
+audio810_update_port(audio810_port_t *port)
+{
+	audio810_state_t	*statep = port->statep;
+	uint8_t			regoff = port->regoff;
+	uint8_t			civ;
+	uint16_t		picb;
+	unsigned		n;
+
+	if (statep->suspended) {
+		civ = 0;
+		picb = 0;
+	} else {
+		/*
+		 * We read the position counters, but we're careful to avoid
+		 * the situation where the position counter resets at the end
+		 * of a buffer.
+		 */
+		for (int i = 0; i < 2; i++) {
+			civ = I810_BM_GET8(regoff + I810_OFFSET_CIV);
+			picb = I810_BM_GET16(port->picboff);
+			if (I810_BM_GET8(regoff + I810_OFFSET_CIV) == civ) {
+				/*
+				 * Chip did not start a new index,
+				 * so the picb is valid.
+				 */
+				break;
+			}
+		}
+
+		if (civ >= port->civ) {
+			n = civ - port->civ;
+		} else {
+			n = civ + (I810_BD_NUMS - port->civ);
+		}
+		port->count += (n * port->fragfr);
+	}
+	port->civ = civ;
+	port->picb = picb;
+}
+
+/*
+ * audio810_attach()
+ *
+ * Description:
+ *	Attach an instance of the audio810 driver. This routine does the
+ * 	device dependent attach tasks, and registers with the audio framework.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audio810_attach(dev_info_t *dip)
+{
+	uint16_t		cmdreg;
+	audio810_state_t	*statep;
+	audio_dev_t		*adev;
+	ddi_acc_handle_t	pcih;
+	uint32_t		devid;
+	uint32_t		gsr;
+	const char		*name;
+	const char		*vers;
+	uint8_t			nch;
+	int			maxch;
+
+	/* we don't support high level interrupts in the driver */
+	if (ddi_intr_hilevel(dip, 0) != 0) {
+		cmn_err(CE_WARN, "!%s: unsupported high level interrupt",
+		    audio810_name);
+		return (DDI_FAILURE);
+	}
+
+	/* allocate the soft state structure */
+	statep = kmem_zalloc(sizeof (*statep), KM_SLEEP);
+	ddi_set_driver_private(dip, statep);
+
+	/* get iblock cookie information */
+	if (ddi_get_iblock_cookie(dip, 0, &statep->iblock) != DDI_SUCCESS) {
+		cmn_err(CE_WARN, "!%s: cannot get iblock cookie",
+		    audio810_name);
+		kmem_free(statep, sizeof (*statep));
+		return (DDI_FAILURE);
+	}
+	mutex_init(&statep->inst_lock, NULL, MUTEX_DRIVER, statep->iblock);
+	mutex_init(&statep->ac_lock, NULL, MUTEX_DRIVER, statep->iblock);
+
+	if ((adev = audio_dev_alloc(dip, 0)) == NULL) {
+		cmn_err(CE_WARN, "!%s: unable to allocate audio dev",
+		    audio810_name);
+		goto error;
+	}
+	statep->adev = adev;
+	statep->dip = dip;
+
+	/* map in the registers, allocate DMA buffers, etc. */
+	if (audio810_map_regs(dip, statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "couldn't map registers");
+		goto error;
+	}
+
+	/* set PCI command register */
+	if (pci_config_setup(dip, &pcih) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "pci conf mapping failed");
+		goto error;
+	}
+	cmdreg = pci_config_get16(pcih, PCI_CONF_COMM);
+	pci_config_put16(pcih, PCI_CONF_COMM,
+	    cmdreg | PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
+	devid = pci_config_get16(pcih, PCI_CONF_VENID);
+	devid <<= 16;
+	devid |= pci_config_get16(pcih, PCI_CONF_DEVID);
+	pci_config_teardown(&pcih);
+
+	/*
+	 * Override "max-channels" property to prevent configuration
+	 * of 4 or 6 (or possibly even 8!) channel audio.  The default
+	 * is to support as many channels as the hardware can do.
+	 *
+	 * (Hmmm... perhaps this should be driven in the common
+	 * framework.  The framework could even offer simplistic upmix
+	 * and downmix for various standard configs.)
+	 */
+	maxch = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, "max-channels", 8);
+	if (maxch < 2) {
+		maxch = 2;
+	}
+
+	gsr = I810_BM_GET32(I810_REG_GSR);
+	if (gsr & I810_GSR_CAP6CH) {
+		nch = 6;
+	} else if (gsr & I810_GSR_CAP4CH) {
+		nch = 4;
+	} else {
+		nch = 2;
+	}
+
+	statep->maxch = (uint8_t)min(nch, maxch);
+	statep->maxch &= ~1;
+	name = "Unknown AC'97";
+	vers = "";
+
+	statep->quirk = QUIRK_NONE;
+	switch (devid) {
+	case 0x80862415:
+		name = "Intel AC'97";
+		vers = "ICH";
+		break;
+	case 0x80862425:
+		name = "Intel AC'97";
+		vers = "ICH0";
+		break;
+	case 0x80867195:
+		name = "Intel AC'97";
+		vers = "440MX";
+		break;
+	case 0x80862445:
+		name = "Intel AC'97";
+		vers = "ICH2";
+		break;
+	case 0x80862485:
+		name = "Intel AC'97";
+		vers = "ICH3";
+		break;
+	case 0x808624C5:
+		name = "Intel AC'97";
+		vers = "ICH4";
+		break;
+	case 0x808624D5:
+		name = "Intel AC'97";
+		vers = "ICH5";
+		break;
+	case 0x8086266E:
+		name = "Intel AC'97";
+		vers = "ICH6";
+		break;
+	case 0x808627DE:
+		name = "Intel AC'97";
+		vers = "ICH7";
+		break;
+	case 0x808625A6:
+		name = "Intel AC'97";
+		vers = "6300ESB";
+		break;
+	case 0x80862698:
+		name = "Intel AC'97";
+		vers = "ESB2";
+		break;
+	case 0x10397012:
+		name = "SiS AC'97";
+		vers = "7012";
+		statep->quirk = QUIRK_SIS7012;
+		break;
+	case 0x10de01b1:	/* nForce */
+		name = "NVIDIA AC'97";
+		vers = "MCP1";
+		break;
+	case 0x10de006a:	/* nForce 2 */
+		name = "NVIDIA AC'97";
+		vers = "MCP2";
+		break;
+	case 0x10de00da:	/* nForce 3 */
+		name = "NVIDIA AC'97";
+		vers = "MCP3";
+		break;
+	case 0x10de00ea:
+		name = "NVIDIA AC'97";
+		vers = "CK8S";
+		break;
+	case 0x10de0059:
+		name = "NVIDIA AC'97";
+		vers = "CK804";
+		break;
+	case 0x10de008a:
+		name = "NVIDIA AC'97";
+		vers = "CK8";
+		break;
+	case 0x10de003a:	/* nForce 4 */
+		name = "NVIDIA AC'97";
+		vers = "MCP4";
+		break;
+	case 0x10de026b:
+		name = "NVIDIA AC'97";
+		vers = "MCP51";
+		break;
+	case 0x1022746d:
+		name = "AMD AC'97";
+		vers = "8111";
+		break;
+	case 0x10227445:
+		name = "AMD AC'97";
+		vers = "AMD768";
+		break;
+	}
+	/* set device information */
+	audio_dev_set_description(adev, name);
+	audio_dev_set_version(adev, vers);
+
+	/* allocate port structures */
+	if ((audio810_alloc_port(statep, I810_PCM_OUT, statep->maxch) !=
+	    DDI_SUCCESS) ||
+	    (audio810_alloc_port(statep, I810_PCM_IN, 2) != DDI_SUCCESS)) {
+		goto error;
+	}
+
+	/* allocate ac97 handle */
+	statep->ac97 = ac97_alloc(dip, audio810_read_ac97, audio810_write_ac97,
+	    statep);
+	if (statep->ac97 == NULL) {
+		audio_dev_warn(adev, "failed to allocate ac97 handle");
+		goto error;
+	}
+
+	/* initialize audio controller and AC97 codec */
+	if (audio810_chip_init(statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to init chip");
+		goto error;
+	}
+
+	/* initialize the AC'97 part */
+	if (ac97_init(statep->ac97, adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ac'97 initialization failed");
+		goto error;
+	}
+
+	/* set up kernel statistics */
+	if ((statep->ksp = kstat_create(I810_NAME, ddi_get_instance(dip),
+	    I810_NAME, "controller", KSTAT_TYPE_INTR, 1,
+	    KSTAT_FLAG_PERSISTENT)) != NULL) {
+		kstat_install(statep->ksp);
+	}
+
+	/* set up the interrupt handler */
+	if (ddi_add_intr(dip, 0, &statep->iblock,
+	    NULL, audio810_intr, (caddr_t)statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "bad interrupt specification");
+		goto error;
+	}
+	statep->intr_added = B_TRUE;
+
+	if (audio_dev_register(adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "unable to register with framework");
+		goto error;
+	}
+
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	audio810_destroy(statep);
+
+	return (DDI_FAILURE);
+}
+
+
+/*
+ * audio810_resume()
+ *
+ * Description:
+ *	Resume operation of the device after sleeping or hibernating.
+ *	Note that this should never fail, even if hardware goes wonky,
+ *	because the current PM framework will panic if it does.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was resumed.
+ */
+static int
+audio810_resume(dev_info_t *dip)
+{
+	audio810_state_t	*statep;
+	audio_dev_t		*adev;
+
+	/* this should always be valid */
+	statep = ddi_get_driver_private(dip);
+	adev = statep->adev;
+
+	ASSERT(statep != NULL);
+	ASSERT(dip == statep->dip);
+
+	/* Restore the audio810 chip's state */
+	if (audio810_chip_init(statep) != DDI_SUCCESS) {
+		/*
+		 * Note that PM gurus say we should return
+		 * success here.  Failure of audio shouldn't
+		 * be considered FATAL to the system.  The
+		 * upshot is that audio will not progress.
+		 */
+		audio_dev_warn(adev, "DDI_RESUME failed to init chip");
+		return (DDI_SUCCESS);
+	}
+
+	/* allow ac97 operations again */
+	ac97_resume(statep->ac97);
+
+	mutex_enter(&statep->inst_lock);
+
+	ASSERT(statep->suspended);
+	statep->suspended = B_FALSE;
+
+	for (int i = 0; i < I810_NUM_PORTS; i++) {
+
+		audio810_port_t *port = statep->ports[i];
+
+		if (port != NULL) {
+			/* reset framework DMA engine buffer */
+			if (port->engine != NULL) {
+				audio_engine_reset(port->engine);
+			}
+
+			/* reset and initialize hardware ports */
+			audio810_reset_port(port);
+			if (port->started) {
+				audio810_start_port(port);
+			} else {
+				audio810_stop_port(port);
+			}
+		}
+	}
+	mutex_exit(&statep->inst_lock);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_detach()
+ *
+ * Description:
+ *	Detach an instance of the audio810 driver.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audio810_detach(dev_info_t *dip)
+{
+	audio810_state_t	*statep;
+
+	statep = ddi_get_driver_private(dip);
+	ASSERT(statep != NULL);
+
+	/* don't detach us if we are still in use */
+	if (audio_dev_unregister(statep->adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	audio810_destroy(statep);
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_suspend()
+ *
+ * Description:
+ *	Suspend an instance of the audio810 driver, in preparation for
+ *	sleep or hibernation.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was suspended
+ */
+static int
+audio810_suspend(dev_info_t *dip)
+{
+	audio810_state_t	*statep;
+
+	statep = ddi_get_driver_private(dip);
+	ASSERT(statep != NULL);
+
+	ac97_suspend(statep->ac97);
+
+	mutex_enter(&statep->inst_lock);
+
+	ASSERT(statep->suspended == B_FALSE);
+
+	statep->suspended = B_TRUE; /* stop new ops */
+
+	/* stop DMA engines */
+	audio810_stop_dma(statep);
+
+	mutex_exit(&statep->inst_lock);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_alloc_port()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use.  It also configures the BDL lists properly
+ *	for use.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audio810_alloc_port(audio810_state_t *statep, int num, uint8_t nchan)
+{
+	ddi_dma_cookie_t	cookie;
+	uint_t			count;
+	int			dir;
+	unsigned		caps;
+	char			*prop;
+	char			*nfprop;
+	audio_dev_t		*adev;
+	audio810_port_t		*port;
+	uint32_t		paddr;
+	int			rc;
+	dev_info_t		*dip;
+	i810_bd_entry_t		*bdentry;
+
+	adev = statep->adev;
+	dip = statep->dip;
+
+	port = kmem_zalloc(sizeof (*port), KM_SLEEP);
+	statep->ports[num] = port;
+	port->statep = statep;
+	port->started = B_FALSE;
+	port->nchan = nchan;
+	port->num = num;
+
+	switch (num) {
+	case I810_PCM_IN:
+		prop = "record-interrupts";
+		nfprop = "record-fragments";
+		dir = DDI_DMA_READ;
+		caps = ENGINE_INPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORKERNEL;
+		port->regoff = I810_BASE_PCM_IN;
+		break;
+	case I810_PCM_OUT:
+		prop = "play-interrupts";
+		nfprop = "play-fragments";
+		dir = DDI_DMA_WRITE;
+		caps = ENGINE_OUTPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORDEV;
+		port->regoff = I810_BASE_PCM_OUT;
+		break;
+	default:
+		audio_dev_warn(adev, "bad port number (%d)!", num);
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * SiS 7012 swaps status and picb registers.
+	 */
+	if (statep->quirk == QUIRK_SIS7012) {
+		port->stsoff = port->regoff + I810_OFFSET_PICB;
+		port->picboff = port->regoff + I810_OFFSET_SR;
+	} else {
+		port->stsoff = port->regoff + I810_OFFSET_SR;
+		port->picboff = port->regoff + I810_OFFSET_PICB;
+	}
+
+	port->intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, prop, I810_INTS);
+
+	/* make sure the values are good */
+	if (port->intrs < I810_MIN_INTS) {
+		audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+		    prop, port->intrs, I810_INTS);
+		port->intrs = I810_INTS;
+	} else if (port->intrs > I810_MAX_INTS) {
+		audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+		    prop, port->intrs, I810_INTS);
+		port->intrs = I810_INTS;
+	}
+
+	port->nfrag = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, nfprop, I810_NFRAGS);
+
+	/*
+	 * Note that fragments must divide evenly into I810_BD_NUMS (32).
+	 */
+	if (port->nfrag <= 4) {
+		port->nfrag = 4;
+	} else if (port->nfrag <= 8) {
+		port->nfrag = 8;
+	} else if (port->nfrag <= 16) {
+		port->nfrag = 16;
+	} else {
+		port->nfrag = I810_BD_NUMS;
+	}
+
+	/*
+	 * Figure out how much space we need.  Sample rate is 48kHz, and
+	 * we need to store 32 chunks.  (Note that this means that low
+	 * interrupt frequencies will require more RAM.  We could probably
+	 * do some cleverness to use a shorter BD list.)
+	 */
+	port->fragfr = 48000 / port->intrs;
+	port->fragfr = I810_ROUNDUP(port->fragfr, I810_MOD_SIZE);
+	port->fragsz = port->fragfr * port->nchan * 2;
+	port->samp_size = port->fragsz * port->nfrag;
+
+	/* allocate dma handle */
+	rc = ddi_dma_alloc_handle(dip, &sample_buf_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->samp_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	/* allocate DMA buffer */
+	rc = ddi_dma_mem_alloc(port->samp_dmah, port->samp_size, &buf_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->samp_kaddr,
+	    &port->samp_size, &port->samp_acch);
+	if (rc == DDI_FAILURE) {
+		audio_dev_warn(adev, "dma_mem_alloc (%d) failed: %d",
+		    port->samp_size, rc);
+		return (DDI_FAILURE);
+	}
+
+	/* bind DMA buffer */
+	rc = ddi_dma_addr_bind_handle(port->samp_dmah, NULL,
+	    port->samp_kaddr, port->samp_size, dir|DDI_DMA_CONSISTENT,
+	    DDI_DMA_SLEEP, NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev,
+		    "ddi_dma_addr_bind_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	port->samp_paddr = cookie.dmac_address;
+
+	/*
+	 * now, from here we allocate DMA memory for buffer descriptor list.
+	 * we allocate adjacent DMA memory for all DMA engines.
+	 */
+	rc = ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->bdl_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * we allocate all buffer descriptors lists in continuous dma memory.
+	 */
+	port->bdl_size = sizeof (i810_bd_entry_t) * I810_BD_NUMS;
+	rc = ddi_dma_mem_alloc(port->bdl_dmah, port->bdl_size,
+	    &dev_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
+	    &port->bdl_kaddr, &port->bdl_size, &port->bdl_acch);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_mem_alloc(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	rc = ddi_dma_addr_bind_handle(port->bdl_dmah, NULL, port->bdl_kaddr,
+	    port->bdl_size, DDI_DMA_WRITE|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP,
+	    NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev, "addr_bind_handle failed");
+		return (DDI_FAILURE);
+	}
+	port->bdl_paddr = cookie.dmac_address;
+
+	/*
+	 * Wire up the BD list.
+	 */
+	paddr = port->samp_paddr;
+	bdentry = (void *)port->bdl_kaddr;
+	for (int i = 0; i < I810_BD_NUMS; i++) {
+
+		/* set base address of buffer */
+		ddi_put32(port->bdl_acch, &bdentry->buf_base, paddr);
+		/*
+		 * SiS 7012 counts samples in bytes, all other count
+		 * in words.
+		 */
+		ddi_put16(port->bdl_acch, &bdentry->buf_len,
+		    statep->quirk == QUIRK_SIS7012 ? port->fragsz :
+		    port->fragsz / 2);
+		ddi_put16(port->bdl_acch, &bdentry->buf_cmd,
+		    BUF_CMD_IOC | BUF_CMD_BUP);
+		paddr += port->fragsz;
+		if ((i % port->nfrag) == (port->nfrag - 1)) {
+			/* handle wrap */
+			paddr = port->samp_paddr;
+		}
+		bdentry++;
+	}
+	(void) ddi_dma_sync(port->bdl_dmah, 0, 0, DDI_DMA_SYNC_FORDEV);
+
+	port->engine = audio_engine_alloc(&audio810_engine_ops, caps);
+	if (port->engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(port->engine, port);
+	audio_dev_add_engine(adev, port->engine);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_free_port()
+ *
+ * Description:
+ *	This routine unbinds the DMA cookies, frees the DMA buffers,
+ *	deallocates the DMA handles.
+ *
+ * Arguments:
+ *	audio810_port_t	*port	The port structure for a DMA engine.
+ */
+static void
+audio810_free_port(audio810_port_t *port)
+{
+	if (port == NULL)
+		return;
+
+	if (port->engine) {
+		audio_dev_remove_engine(port->statep->adev, port->engine);
+		audio_engine_free(port->engine);
+	}
+	if (port->bdl_paddr) {
+		(void) ddi_dma_unbind_handle(port->bdl_dmah);
+	}
+	if (port->bdl_acch) {
+		ddi_dma_mem_free(&port->bdl_acch);
+	}
+	if (port->bdl_dmah) {
+		ddi_dma_free_handle(&port->bdl_dmah);
+	}
+	if (port->samp_paddr) {
+		(void) ddi_dma_unbind_handle(port->samp_dmah);
+	}
+	if (port->samp_acch) {
+		ddi_dma_mem_free(&port->samp_acch);
+	}
+	if (port->samp_dmah) {
+		ddi_dma_free_handle(&port->samp_dmah);
+	}
+	kmem_free(port, sizeof (*port));
+}
+
+/*
+ * audio810_map_regs()
+ *
+ * Description:
+ *	The registers are mapped in.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audio810_map_regs(dev_info_t *dip, audio810_state_t *statep)
+{
+	uint_t			nregs = 0;
+	int			*regs_list;
+	int			i;
+	int			pciBar1 = 0;
+	int			pciBar2 = 0;
+	int			pciBar3 = 0;
+	int			pciBar4 = 0;
+
+	/* check the "reg" property to get the length of memory-mapped I/O */
+	if (ddi_prop_lookup_int_array(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
+	    "reg", (int **)&regs_list, &nregs) != DDI_PROP_SUCCESS) {
+		audio_dev_warn(statep->adev, "inquire regs property failed");
+		goto error;
+	}
+	/*
+	 * Some hardwares, such as Intel ICH0/ICH and AMD 8111, use PCI 0x10
+	 * and 0x14 BAR separately for native audio mixer BAR and native bus
+	 * mastering BAR. More advanced hardwares, such as Intel ICH4 and ICH5,
+	 * support PCI memory BAR, via PCI 0x18 and 0x1C BAR, that allows for
+	 * higher performance access to the controller register. All features
+	 * can be accessed via this BAR making the I/O BAR (PCI 0x10 and 0x14
+	 * BAR) capabilities obsolete. However, these controller maintain the
+	 * I/O BAR capability to allow for the reuse of legacy code maintaining
+	 * backward compatibility. The I/O BAR is disabled unless system BIOS
+	 * enables the simultaneous backward compatible capability on the 0x41
+	 * register.
+	 *
+	 * When I/O BAR is enabled, the value of "reg" property should be like
+	 * this,
+	 *	phys_hi   phys_mid  phys_lo   size_hi   size_lo
+	 * --------------------------------------------------------
+	 *	0000fd00  00000000  00000000  00000000  00000000
+	 *	0100fd10  00000000  00000000  00000000  00000100
+	 *	0100fd14  00000000  00000000  00000000  00000040
+	 *	0200fd18  00000000  00000000  00000000  00000200
+	 *	0200fd1c  00000000  00000000  00000000  00000100
+	 *
+	 * When I/O BAR is disabled, the "reg" property of the device node does
+	 * not consist of the description for the I/O BAR. The following example
+	 * illustrates the vaule of "reg" property,
+	 *
+	 *	phys_hi   phys_mid  phys_lo   size_hi   size_lo
+	 * --------------------------------------------------------
+	 *	0000fd00  00000000  00000000  00000000  00000000
+	 *	0200fd18  00000000  00000000  00000000  00000200
+	 *	0200fd1c  00000000  00000000  00000000  00000100
+	 *
+	 * If the hardware has memory-mapped I/O access, first try to use
+	 * this facility, otherwise we will try I/O access.
+	 */
+	for (i = 1; i < nregs/I810_INTS_PER_REG_PROP; i++) {
+		switch (regs_list[I810_INTS_PER_REG_PROP * i] & 0x000000ff) {
+			case 0x10:
+				pciBar1 = i;
+				break;
+			case 0x14:
+				pciBar2 = i;
+				break;
+			case 0x18:
+				pciBar3 = i;
+				break;
+			case 0x1c:
+				pciBar4 = i;
+				break;
+			default:	/* we don't care others */
+				break;
+		}
+	}
+
+	if ((pciBar3 != 0) && (pciBar4 != 0)) {
+		/* map audio mixer registers */
+		if ((ddi_regs_map_setup(dip, pciBar3, &statep->am_regs_base, 0,
+		    0, &dev_attr, &statep->am_regs_handle)) != DDI_SUCCESS) {
+			audio_dev_warn(statep->adev,
+			    "memory am mapping failed");
+			goto error;
+		}
+
+		/* map bus master register */
+		if ((ddi_regs_map_setup(dip, pciBar4, &statep->bm_regs_base, 0,
+		    0, &dev_attr, &statep->bm_regs_handle)) != DDI_SUCCESS) {
+			audio_dev_warn(statep->adev,
+			    "memory bm mapping failed");
+			goto error;
+		}
+
+	} else if ((pciBar1 != 0) && (pciBar2 != 0)) {
+		/* map audio mixer registers */
+		if ((ddi_regs_map_setup(dip, pciBar1, &statep->am_regs_base, 0,
+		    0, &dev_attr, &statep->am_regs_handle)) != DDI_SUCCESS) {
+			audio_dev_warn(statep->adev, "I/O am mapping failed");
+			goto error;
+		}
+
+		/* map bus master register */
+		if ((ddi_regs_map_setup(dip, pciBar2, &statep->bm_regs_base, 0,
+		    0, &dev_attr, &statep->bm_regs_handle)) != DDI_SUCCESS) {
+			audio_dev_warn(statep->adev, "I/O bm mapping failed");
+			goto error;
+		}
+	} else {
+		audio_dev_warn(statep->adev, "map_regs() pci BAR error");
+		goto error;
+	}
+
+	ddi_prop_free(regs_list);
+
+	return (DDI_SUCCESS);
+
+error:
+	if (nregs > 0) {
+		ddi_prop_free(regs_list);
+	}
+	audio810_unmap_regs(statep);
+
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio810_unmap_regs()
+ *
+ * Description:
+ *	This routine unmaps control registers.
+ *
+ * Arguments:
+ *	audio810_state_t	*state	The device's state structure
+ */
+static void
+audio810_unmap_regs(audio810_state_t *statep)
+{
+	if (statep->bm_regs_handle) {
+		ddi_regs_map_free(&statep->bm_regs_handle);
+	}
+
+	if (statep->am_regs_handle) {
+		ddi_regs_map_free(&statep->am_regs_handle);
+	}
+}
+
+/*
+ * audio810_chip_init()
+ *
+ * Description:
+ *	This routine initializes the AMD 8111 audio controller.
+ *	codec.
+ *
+ * Arguments:
+ *	audio810_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS	The hardware was initialized properly
+ *	DDI_FAILURE	The hardware couldn't be initialized properly
+ */
+static int
+audio810_chip_init(audio810_state_t *statep)
+{
+	uint32_t	gcr;
+	uint32_t	gsr;
+	uint32_t	codec_ready;
+	int 		loop;
+	clock_t		ticks;
+
+	gcr = I810_BM_GET32(I810_REG_GCR);
+	ticks = drv_usectohz(100);
+
+	/*
+	 * Clear the channels bits for now.  We'll set them later in
+	 * reset port.
+	 */
+	if (statep->quirk == QUIRK_SIS7012) {
+		gcr &= ~(I810_GCR_ACLINK_OFF | I810_GCR_SIS_CHANNELS_MASK);
+	} else {
+		gcr &= ~(I810_GCR_ACLINK_OFF | I810_GCR_CHANNELS_MASK);
+	}
+
+	/*
+	 * Datasheet(ICH5, document number of Intel: 252751-001):
+	 * 3.6.5.5(page 37)
+	 * 	if reset bit(bit1) is "0", driver must set it
+	 * 	to "1" to de-assert the AC_RESET# signal in AC
+	 * 	link, thus completing a cold reset. But if the
+	 * 	bit is "1", then a warm reset is required.
+	 */
+	gcr |= (gcr & I810_GCR_COLD_RST) == 0 ?
+	    I810_GCR_COLD_RST:I810_GCR_WARM_RST;
+	I810_BM_PUT32(I810_REG_GCR, gcr);
+
+	/* according AC'97 spec, wait for codec reset */
+	for (loop = 6000; --loop >= 0; ) {
+		delay(ticks);
+		gcr = I810_BM_GET32(I810_REG_GCR);
+		if ((gcr & I810_GCR_WARM_RST) == 0) {
+			break;
+		}
+	}
+
+	/* codec reset failed */
+	if (loop < 0) {
+		audio_dev_warn(statep->adev, "Failed to reset codec");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * Wait for codec ready. The hardware can provide the state of
+	 * codec ready bit on SDATA_IN[0], SDATA_IN[1] or SDATA_IN[2]
+	 */
+	codec_ready =
+	    I810_GSR_PRI_READY | I810_GSR_SEC_READY | I810_GSR_TRI_READY;
+	for (loop = 7000; --loop >= 0; ) {
+		delay(ticks);
+		gsr = I810_BM_GET32(I810_REG_GSR);
+		if ((gsr & codec_ready) != 0) {
+			break;
+		}
+	}
+	if (loop < 0) {
+		audio_dev_warn(statep->adev, "No codec ready signal received");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * put the audio controller into quiet state, everything off
+	 */
+	audio810_stop_dma(statep);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audio810_stop_dma()
+ *
+ * Description:
+ *	This routine is used to put each DMA engine into the quiet state.
+ *
+ * Arguments:
+ *	audio810_state_t	*state		The device's state structure
+ */
+static void
+audio810_stop_dma(audio810_state_t *statep)
+{
+	if (statep->bm_regs_handle == NULL) {
+		return;
+	}
+	/* pause bus master (needed for the following reset register) */
+	I810_BM_PUT8(I810_BASE_PCM_IN + I810_OFFSET_CR, 0x0);
+	I810_BM_PUT8(I810_BASE_PCM_OUT + I810_OFFSET_CR, 0x0);
+	I810_BM_PUT8(I810_BASE_MIC + I810_OFFSET_CR, 0x0);
+
+	/* and then reset the bus master registers for a three DMA engines */
+	I810_BM_PUT8(I810_BASE_PCM_IN + I810_OFFSET_CR, I810_BM_CR_RST);
+	I810_BM_PUT8(I810_BASE_PCM_OUT + I810_OFFSET_CR, I810_BM_CR_RST);
+	I810_BM_PUT8(I810_BASE_MIC + I810_OFFSET_CR, I810_BM_CR_RST);
+}
+
+
+/*
+ * audio810_codec_sync()
+ *
+ * Description:
+ *	Serialize access to the AC97 audio mixer registers.
+ *
+ * Arguments:
+ *	audio810_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Ready for an I/O access to the codec
+ *	DDI_FAILURE		An I/O access is currently in progress, can't
+ *				perform another I/O access.
+ */
+static int
+audio810_codec_sync(audio810_state_t *statep)
+{
+	int 		i;
+	uint16_t	casr;
+
+	for (i = 0; i < 300; i++) {
+		casr = I810_BM_GET8(I810_REG_CASR);
+		if ((casr & 1) == 0) {
+			return (DDI_SUCCESS);
+		}
+		drv_usecwait(10);
+	}
+
+	return (DDI_FAILURE);
+}
+
+/*
+ * audio810_write_ac97()
+ *
+ * Description:
+ *	Set the specific AC97 Codec register.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *	uint16_t	data		The data want to be set
+ */
+static void
+audio810_write_ac97(void *arg, uint8_t reg, uint16_t data)
+{
+	audio810_state_t	*statep = arg;
+
+	mutex_enter(&statep->ac_lock);
+	if (audio810_codec_sync(statep) == DDI_SUCCESS) {
+		I810_AM_PUT16(reg, data);
+	}
+	mutex_exit(&statep->ac_lock);
+
+	(void) audio810_read_ac97(statep, reg);
+}
+
+/*
+ * audio810_read_ac97()
+ *
+ * Description:
+ *	Get the specific AC97 Codec register.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *
+ * Returns:
+ *	The register value.
+ */
+static uint16_t
+audio810_read_ac97(void *arg, uint8_t reg)
+{
+	audio810_state_t	*statep = arg;
+	uint16_t		val = 0xffff;
+
+	mutex_enter(&statep->ac_lock);
+	if (audio810_codec_sync(statep) == DDI_SUCCESS) {
+		val = I810_AM_GET16(reg);
+	}
+	mutex_exit(&statep->ac_lock);
+	return (val);
+}
+
+/*
+ * audio810_destroy()
+ *
+ * Description:
+ *	This routine releases all resources held by the device instance,
+ *	as part of either detach or a failure in attach.
+ *
+ * Arguments:
+ *	audio810_state_t	*state	The device soft state.
+ */
+void
+audio810_destroy(audio810_state_t *statep)
+{
+	/* stop DMA engines */
+	audio810_stop_dma(statep);
+
+	if (statep->intr_added) {
+		ddi_remove_intr(statep->dip, 0, NULL);
+	}
+
+	if (statep->ksp) {
+		kstat_delete(statep->ksp);
+	}
+
+	for (int i = 0; i < I810_NUM_PORTS; i++) {
+		audio810_free_port(statep->ports[i]);
+	}
+
+	audio810_unmap_regs(statep);
+
+	if (statep->ac97)
+		ac97_free(statep->ac97);
+
+	if (statep->adev)
+		audio_dev_free(statep->adev);
+
+	mutex_destroy(&statep->inst_lock);
+	mutex_destroy(&statep->ac_lock);
+	kmem_free(statep, sizeof (*statep));
+}
diff --git a/usr/src/uts/common/io/audio/drv/audio810/audio810.conf b/usr/src/uts/common/io/audio/drv/audio810/audio810.conf
new file mode 100644
index 0000000000..1041a2a8d1
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audio810/audio810.conf
@@ -0,0 +1,97 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Configuration file for the audio810 audio driver.
+#
+#	WARNING: This is an UNSTABLE configuration file. Its contents
+#		may change at any time.
+
+#
+# play-interrupts sets the number of interrupts per second when playing.
+# This affects the resolution of various things, such as sample counts.
+# record-interrupts does the same for record interrupts.
+#
+# These may be tuned to get more accurate information by increasing the
+# count. However, the larger the interrupts per second the larger the
+# load on the system. So use this capability cautiously. The audio810
+# driver enforces a maximum and minimum count.
+#
+# It should also be understood that not all interrupt rates are legal.
+# The hardware is restricted to DMA buffers being allocated on certain
+# boundaries. If those boundaries are violated the driver will not be
+# loaded and an error message is entered into the messages log
+#
+# play-interrupts=250;
+# record-interrupts=250;
+
+#
+# The presence of the ac97-speaker property enables the use of a monoaural
+# output, normally intended for use with a speaker phone.  Most systems
+# do not connect this to anything.  The value of the property indicates
+# whether the speaker will be enabled by default or not.
+#
+# ac97-speaker=0;
+
+#
+# Uncomment ac97-micboost below to enable the use of a 20dB microphone boost.
+#
+# ac97-micboost=1;
+
+#
+# Uncomment properties below to suppress the AC'97 outputs that are
+# supported by the codec, but not connected to anything.  Note
+# that the headphone feature in particular is commonly implemented as
+# an alternate output for the line out, and uses that with jack detection
+# instead of the codec's headphone feature.
+#
+# We don't allow diabling the lineout -- that's such a fundamental
+# part of ac97 that we doubt any vendor isn't using it.
+#
+# ac97-no-headphone=1;
+# ac97-no-auxout=1;
+
+#
+# Uncomment properties below to suppress the AC'97 inputs that are
+# supported by the codec, but not connected to anything.  Note that
+# CDROM input support is mandatory in AC'97, so all codecs implement
+# it, even though many systems may not even have a CDROM installed.
+#
+# Generally, you can count on a linein and mic jack being present,
+# although some Toshiba models have been seen that lack a line-level
+# input jack.
+#
+# ac97-no-cdrom=1;
+# ac97-no-auxin=1;
+# ac97-no-video=1;
+# ac97-no-phone=1;
+# ac97-no-linein=1;
+# ac97-no-mic=1;
+
+#
+# Uncomment ac97-amplifier below to power down the external
+# amplifier. Most hardware doesn't need this property. But some
+# devices, such as some Sony Vaio laptops, need to set this property to
+# 0 to correct the inverted polarity of the external amplifier device.
+#
+# ac97-amplifier=0;
diff --git a/usr/src/uts/common/io/audio/drv/audio810/audio810.h b/usr/src/uts/common/io/audio/drv/audio810/audio810.h
new file mode 100644
index 0000000000..65917f0d02
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audio810/audio810.h
@@ -0,0 +1,267 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_AUDIO810_H_
+#define	_AUDIO810_H_
+
+/*
+ * Header file for the audio810 device driver
+ */
+
+/*
+ * Driver supported configuration information
+ */
+#define	I810_NAME			"audio810"
+#define	I810_MOD_NAME			"audio810 audio driver"
+
+#define	I810_INTS			(175)	/* default interrupt rate */
+#define	I810_MIN_INTS			(24)	/* minimum interrupt rate */
+#define	I810_MAX_INTS			(5000)	/* maximum interrupt rate */
+#define	I810_NFRAGS			(8)	/* default # fragments */
+
+/*
+ * Misc. defines
+ */
+
+#define	I810_BD_NUMS			(32)
+#define	I810_NUM_PORTS			(2)
+#define	I810_MOD_SIZE			(16)
+
+#define	I810_ROUNDUP(x, algn)		(((x) + ((algn) - 1)) & ~((algn) - 1))
+#define	I810_KIOP(X)			((kstat_intr_t *)(X->ksp->ks_data))
+
+/* The size of each entry of "reg" property is 5 integers */
+#define	I810_INTS_PER_REG_PROP		5
+
+/* offset from the base of specified DMA engine */
+#define	I810_OFFSET_BD_BASE		(0x00)
+#define	I810_OFFSET_CIV			(0x04)
+#define	I810_OFFSET_LVI			(0x05)
+#define	I810_OFFSET_SR			(0x06)
+#define	I810_OFFSET_PICB		(0x08)
+#define	I810_OFFSET_PIV			(0x0A)
+#define	I810_OFFSET_CR			(0x0B)
+
+/* DMA engine offset from base */
+#define	I810_BASE_PCM_IN		(0x00)
+#define	I810_BASE_PCM_OUT		(0x10)
+#define	I810_BASE_MIC			(0x20)
+
+#define	I810_REG_GCR			0x2C
+#define	I810_REG_GSR			0x30
+#define	I810_REG_CASR			0x34
+#define	I810_REG_SISCTL			0x4C	/* SiS 7012 control */
+
+/* bits of bus master status register */
+#define	I810_BM_SR_DCH			0x01
+#define	I810_BM_SR_CELV			0x02
+#define	I810_BM_SR_LVBCI		0x04
+#define	I810_BM_SR_BCIS			0x08
+#define	I810_BM_SR_FIFOE		0x10
+
+/* bits of bus master control register */
+#define	I810_BM_CR_RUN			0x01
+#define	I810_BM_CR_RST			0x02
+#define	I810_BM_CR_LVBIE		0x04
+#define	I810_BM_CR_FEIE			0x08
+#define	I810_BM_CR_IOCE			0x10
+
+#define	I810_BM_CR_PAUSE		0x00
+
+/*
+ * Global Control Register
+ */
+#define	I810_GCR_GPIE			0x00000001
+#define	I810_GCR_COLD_RST		0x00000002
+#define	I810_GCR_WARM_RST		0x00000004
+#define	I810_GCR_ACLINK_OFF		0x00000008
+#define	I810_GCR_PRI_INTR_ENABLE	0x00000010
+#define	I810_GCR_SEC_INTR_ENABLE	0x00000020
+
+/* For ICH2 or more, bit21:20 is the PCM 4/6-channel enable bits */
+#define	I810_GCR_2_CHANNELS		(0 << 20)
+#define	I810_GCR_4_CHANNELS		(1 << 20)
+#define	I810_GCR_6_CHANNELS		(2 << 20)
+#define	I810_GCR_CHANNELS_MASK		(3 << 20)
+/* SiS 7012 has its own flags here */
+#define	I810_GCR_SIS_2_CHANNELS		(0 << 6)
+#define	I810_GCR_SIS_4_CHANNELS		(1 << 6)
+#define	I810_GCR_SIS_6_CHANNELS		(2 << 6)
+#define	I810_GCR_SIS_CHANNELS_MASK	(3 << 6)
+
+
+/*
+ * Global Status Register
+ */
+#define	I810_GSR_TRI_READY		0x10000000	/* for ICH4/5 */
+#define	I810_GSR_CAP8CH			0x00400000
+#define	I810_GSR_CAP6CH			0x00200000
+#define	I810_GSR_CAP4CH			0x00100000
+#define	I810_GSR_READ_COMPL		0x00008000
+#define	I810_GSR_INTR_SEC_RESUME	0x00000800
+#define	I810_GSR_INTR_PRI_RESUME	0x00000400
+#define	I810_GSR_SEC_READY		0x00000200
+#define	I810_GSR_PRI_READY		0x00000100
+#define	I810_GSR_INTR_MIC		0x00000080
+#define	I810_GSR_INTR_POUT		0x00000040
+#define	I810_GSR_INTR_PIN		0x00000020
+#define	I810_GSR_INTR_MO		0x00000004
+#define	I810_GSR_INTR_MI		0x00000002
+#define	I810_GSR_INTR_GSI		0x00000001
+#define	I810_GSR_USE_INTR		0x00000060	/* PCM-IN ,PCM-OUT */
+
+/*
+ * SiS Control Register
+ */
+#define	I810_SISCTL_UNMUTE		0x01
+
+/*
+ * Macro for AD1980 codec
+ */
+#define	AD1980_VID1		0x4144
+#define	AD1980_VID2		0x5370
+#define	AD1985_VID2		0x5375
+#define	CODEC_AD_REG_MISC	0x76	/* offset of ad1980 misc control reg */
+#define	AD1980_MISC_LOSEL	0x0020	/* Line-out amplifier output selector */
+#define	AD1980_MISC_HPSEL	0x0400	/* HP-out amplifier output selector */
+#define	AD1980_SURR_MUTE	0x8080	/* Mute for surround volume register */
+
+#define	I810_PCM_IN		0
+#define	I810_PCM_OUT		1
+
+struct audio810_port {
+	struct audio810_state	*statep;
+	int			num;
+	ddi_dma_handle_t	samp_dmah;
+	ddi_acc_handle_t	samp_acch;
+	size_t			samp_size;
+	caddr_t			samp_kaddr;
+	uint32_t		samp_paddr;
+
+	ddi_dma_handle_t	bdl_dmah;
+	ddi_acc_handle_t	bdl_acch;
+	size_t			bdl_size;
+	caddr_t			bdl_kaddr;
+	uint32_t		bdl_paddr;
+
+	unsigned		intrs;
+	unsigned		fragfr;
+	unsigned		fragsz;
+	uint64_t		count;
+	uint8_t			nfrag;
+	uint8_t			nchan;
+
+	uint8_t			regoff;
+	uint8_t			stsoff;		/* status offset */
+	uint8_t			picboff;	/* picb offset */
+	uint8_t			civ;
+	uint16_t		picb;
+	unsigned		sync_dir;
+
+	boolean_t		started;
+
+	audio_engine_t		*engine;
+};
+typedef struct audio810_port audio810_port_t;
+
+/*
+ * buffer descripter list entry, sees datasheet
+ */
+struct i810_bd_entry {
+	uint32_t	buf_base;	/* the address of the buffer */
+	uint16_t	buf_len;	/* the number of samples */
+	uint16_t	buf_cmd;
+};
+typedef struct	i810_bd_entry	i810_bd_entry_t;
+#define	BUF_CMD_BUP	0x4000
+#define	BUF_CMD_IOC	0x8000
+
+typedef enum i810_quirk {
+	QUIRK_NONE = 0,
+	QUIRK_SIS7012,		/* weird registers and such */
+} i810_quirk_t;
+
+/*
+ * audio810_state_t	-per instance state and operation data
+ */
+struct audio810_state {
+	kmutex_t		inst_lock;	/* state protection lock */
+	kmutex_t		ac_lock;
+	ddi_iblock_cookie_t	iblock;
+	dev_info_t		*dip;	/* used by audio810_getinfo() */
+	audio_dev_t		*adev;
+	ac97_t			*ac97;
+	audio810_port_t		*ports[2];
+
+	ddi_acc_handle_t	am_regs_handle;	/* for audio mixer register */
+	ddi_acc_handle_t	bm_regs_handle;	/* for bus master register */
+	caddr_t			am_regs_base;	/* base of audio mixer regs */
+	caddr_t			bm_regs_base;	/* base of bus master regs */
+
+	kstat_t			*ksp;		/* kernel statistics */
+
+	boolean_t		intr_added;
+	boolean_t		suspended;	/* suspend/resume state */
+	uint8_t			maxch;
+	i810_quirk_t		quirk;
+};
+typedef struct audio810_state	 audio810_state_t;
+
+/*
+ * Useful bit twiddlers
+ */
+#define	I810_BM_GET8(reg)	\
+	ddi_get8(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)))
+
+#define	I810_BM_GET16(reg)	\
+	ddi_get16(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)))
+
+#define	I810_BM_GET32(reg)	\
+	ddi_get32(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)))
+
+#define	I810_BM_PUT8(reg, val)	\
+	ddi_put8(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)), (val))
+
+#define	I810_BM_PUT16(reg, val)	\
+	ddi_put16(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)), (val))
+
+#define	I810_BM_PUT32(reg, val)	\
+	ddi_put32(statep->bm_regs_handle, \
+	(void *)((char *)statep->bm_regs_base + (reg)), (val))
+
+#define	I810_AM_GET16(reg)	\
+	ddi_get16(statep->am_regs_handle, \
+	(void *)((char *)statep->am_regs_base + (reg)))
+
+#define	I810_AM_PUT16(reg, val)	\
+	ddi_put16(statep->am_regs_handle, \
+	(void *)((char *)statep->am_regs_base + (reg)), (val))
+
+#endif /* _AUDIO810_H_ */
diff --git a/usr/src/uts/common/io/audio/drv/audioens/audioens.c b/usr/src/uts/common/io/audio/drv/audioens/audioens.c
new file mode 100644
index 0000000000..f5220f5229
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audioens/audioens.c
@@ -0,0 +1,1458 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * Purpose: Creative/Ensoniq AudioPCI97  driver (ES1371/ES1373)
+ *
+ * This driver is used with the original Ensoniq AudioPCI97 card and many
+ * PCI based Sound Blaster cards by Creative Technologies. For example
+ * Sound Blaster PCI128 and Creative/Ectiva EV1938.
+ */
+
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include <sys/note.h>
+#include <sys/pci.h>
+#include "audioens.h"
+
+/*
+ * The original OSS driver used a single duplex engine and a separate
+ * playback only engine.  Instead, we expose three engines, one for input
+ * and two for output.
+ */
+
+/*
+ * Set the latency to 32, 64, 96, 128 clocks - some APCI97 devices exhibit
+ * garbled audio in some cases and setting the latency to higer values fixes it
+ * Values: 32, 64, 96, 128 - Default: 64 (or defined by bios)
+ */
+int audioens_latency = 0;
+
+/*
+ * Enable SPDIF port on SoundBlaster 128D or Sound Blaster Digital-4.1 models
+ * Values: 1=Enable 0=Disable Default: 0
+ */
+int audioens_spdif = 0;
+
+/*
+ * Note: Latest devices can support SPDIF with AC3 pass thru.
+ * However, in order to do this, one of the two DMA engines must be
+ * dedicated to this, which would prevent the card from supporting 4
+ * channel audio.  For now we don't bother with the AC3 pass through
+ * mode, and instead just focus on 4 channel support.  In the future,
+ * this could be selectable via a property.
+ */
+
+#define	ENSONIQ_VENDOR_ID	0x1274
+#define	CREATIVE_VENDOR_ID	0x1102
+#define	ECTIVA_VENDOR_ID	0x1102
+#define	ENSONIQ_ES1371		0x1371
+#define	ENSONIQ_ES5880		0x8001
+#define	ENSONIQ_ES5880A		0x8002
+#define	ENSONIQ_ES5880B		0x5880
+#define	ECTIVA_ES1938		0x8938
+
+#define	DEFRATE			48000
+#define	DEFINTS			75
+#define	DRVNAME			"audioens"
+
+typedef struct audioens_port
+{
+	/* Audio parameters */
+	boolean_t		trigger;
+	boolean_t		suspended;
+
+	int			speed;
+
+	int			num;
+#define	PORT_DAC		0
+#define	PORT_ADC		1
+#define	PORT_MAX		PORT_ADC
+
+	caddr_t			kaddr;
+	uint32_t		paddr;
+	ddi_acc_handle_t	acch;
+	ddi_dma_handle_t	dmah;
+	int			nchan;
+	unsigned		fragfr;
+	unsigned		nfrags;
+	unsigned		nframes;
+	unsigned		frameno;
+	uint64_t		count;
+
+	struct audioens_dev	*dev;
+	audio_engine_t	*engine;
+} audioens_port_t;
+
+typedef struct audioens_dev
+{
+	audio_dev_t		*osdev;
+	kmutex_t		mutex;
+	uint16_t		devid;
+	uint8_t			revision;
+	dev_info_t		*dip;
+	boolean_t		enabled;
+
+
+	int			pintrs;
+	int			rintrs;
+
+	kstat_t			*ksp;
+
+	audioens_port_t		port[PORT_MAX + 1];
+
+	ac97_t			*ac97;
+
+	caddr_t			regs;
+	ddi_acc_handle_t	acch;
+	ddi_intr_handle_t	ihandle[1];
+} audioens_dev_t;
+
+static ddi_device_acc_attr_t acc_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static ddi_device_acc_attr_t buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * The hardware appears to be able to address up to 16-bits worth of longwords,
+ * giving a total address space of 256K.  Note, however, that we will restrict
+ * this further when we do fragment and memory allocation.  At its very highest
+ * clock rate (48 kHz) and sample size (16-bit stereo), and lowest interrupt
+ * rate (32 Hz), we only need 6000 bytes per fragment.
+ *
+ * So with an allocated buffer size of 64K, we can support at least 10 frags,
+ * which is more than enough.  (The legacy Sun driver used only 2 fragments.)
+ */
+#define	AUDIOENS_BUF_LEN	(65536)
+
+static ddi_dma_attr_t dma_attr = {
+	DMA_ATTR_VERSION,	/* dma_attr_version */
+	0x0,			/* dma_attr_addr_lo */
+	0xffffffffU,		/* dma_attr_addr_hi */
+	0x3ffff,		/* dma_attr_count_max */
+	0x8,			/* dma_attr_align */
+	0x7f,			/* dma_attr_burstsizes */
+	0x1,			/* dma_attr_minxfer */
+	0x3ffff,		/* dma_attr_maxxfer */
+	0x3ffff,		/* dma_attr_seg */
+	0x1,			/* dma_attr_sgllen */
+	0x1,			/* dma_attr_granular */
+	0			/* dma_attr_flags */
+};
+
+#define	GET8(dev, offset)	\
+	ddi_get8(dev->acch, (uint8_t *)(dev->regs + (offset)))
+#define	GET16(dev, offset)	\
+	ddi_get16(dev->acch, (uint16_t *)(void *)(dev->regs + (offset)))
+#define	GET32(dev, offset)	\
+	ddi_get32(dev->acch, (uint32_t *)(void *)(dev->regs + (offset)))
+#define	PUT8(dev, offset, v)	\
+	ddi_put8(dev->acch, (uint8_t *)(dev->regs + (offset)), v)
+#define	PUT16(dev, offset, v)	\
+	ddi_put16(dev->acch, (uint16_t *)(void *)(dev->regs + (offset)), v)
+#define	PUT32(dev, offset, v)	\
+	ddi_put32(dev->acch, (uint32_t *)(void *)(dev->regs + (offset)), v)
+
+#define	CLR8(dev, offset, v)	PUT8(dev, offset, GET8(dev, offset) & ~(v))
+#define	SET8(dev, offset, v)	PUT8(dev, offset, GET8(dev, offset) | (v))
+#define	CLR32(dev, offset, v)	PUT32(dev, offset, GET32(dev, offset) & ~(v))
+#define	SET32(dev, offset, v)	PUT32(dev, offset, GET32(dev, offset) | (v))
+
+#define	KSINTR(dev)	((kstat_intr_t *)((dev)->ksp->ks_data))
+
+static void audioens_init_hw(audioens_dev_t *);
+static void audioens_init_port(audioens_port_t *);
+static void audioens_start_port(audioens_port_t *);
+static void audioens_stop_port(audioens_port_t *);
+static void audioens_update_port(audioens_port_t *);
+
+static uint16_t
+audioens_rd97(void *dev_, uint8_t wAddr)
+{
+	audioens_dev_t *dev = dev_;
+	int i, dtemp;
+
+	mutex_enter(&dev->mutex);
+	dtemp = GET32(dev, CONC_dCODECCTL_OFF);
+	/* wait for WIP to go away saving the current state for later */
+	for (i = 0; i < 0x100UL; ++i) {
+		dtemp = GET32(dev, CONC_dCODECCTL_OFF);
+		if ((dtemp & (1UL << 30)) == 0)
+			break;
+	}
+
+	/* write addr w/data=0 and assert read request ... */
+	PUT32(dev, CONC_dCODECCTL_OFF, ((int)wAddr << 16) | (1UL << 23));
+
+	/* now wait for the data (RDY) */
+	for (i = 0; i < 0x100UL; ++i) {
+		dtemp = GET32(dev, CONC_dCODECCTL_OFF);
+		if (dtemp & (1UL << 31))
+			break;
+	}
+	dtemp = GET32(dev, CONC_dCODECCTL_OFF);
+	mutex_exit(&dev->mutex);
+
+	return (dtemp & 0xffff);
+}
+
+static void
+audioens_wr97(void *dev_, uint8_t wAddr, uint16_t wData)
+{
+	audioens_dev_t *dev = dev_;
+	int i, dtemp;
+
+	mutex_enter(&dev->mutex);
+	/* wait for WIP to go away */
+	for (i = 0; i < 0x100UL; ++i) {
+		dtemp = GET32(dev, CONC_dCODECCTL_OFF);
+		if ((dtemp & (1UL << 30)) == 0)
+			break;
+	}
+
+	PUT32(dev, CONC_dCODECCTL_OFF, ((int)wAddr << 16) | wData);
+
+	mutex_exit(&dev->mutex);
+}
+
+static unsigned short
+SRCRegRead(audioens_dev_t *dev, unsigned short reg)
+{
+	int i, dtemp;
+
+	dtemp = GET32(dev, CONC_dSRCIO_OFF);
+	/* wait for ready */
+	for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+		dtemp = GET32(dev, CONC_dSRCIO_OFF);
+		if ((dtemp & SRC_BUSY) == 0)
+			break;
+	}
+
+	/* assert a read request */
+	PUT32(dev, CONC_dSRCIO_OFF, (dtemp & SRC_CTLMASK) | ((int)reg << 25));
+
+	/* now wait for the data */
+	for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+		dtemp = GET32(dev, CONC_dSRCIO_OFF);
+		if ((dtemp & SRC_BUSY) == 0)
+			break;
+	}
+
+	return ((unsigned short) dtemp);
+}
+
+static void
+SRCRegWrite(audioens_dev_t *dev, unsigned short reg, unsigned short val)
+{
+	int i, dtemp;
+	int writeval;
+
+	dtemp = GET32(dev, CONC_dSRCIO_OFF);
+	/* wait for ready */
+	for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+		dtemp = GET32(dev, CONC_dSRCIO_OFF);
+		if ((dtemp & SRC_BUSY) == 0)
+			break;
+	}
+
+	/* assert the write request */
+	writeval = (dtemp & SRC_CTLMASK) | SRC_WENABLE |
+	    ((int)reg << 25) | val;
+	PUT32(dev, CONC_dSRCIO_OFF, writeval);
+}
+
+static void
+SRCSetRate(audioens_dev_t *dev, unsigned char base, unsigned short rate)
+{
+	int i, freq, dtemp;
+	unsigned short N, truncM, truncStart;
+
+	if (base != SRC_ADC_BASE) {
+		/* freeze the channel */
+		dtemp = (base == SRC_DAC1_BASE) ?
+		    SRC_DAC1FREEZE : SRC_DAC2FREEZE;
+		for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+			if (!(GET32(dev, CONC_dSRCIO_OFF) & SRC_BUSY))
+				break;
+		}
+		PUT32(dev, CONC_dSRCIO_OFF,
+		    (GET32(dev, CONC_dSRCIO_OFF) & SRC_CTLMASK) | dtemp);
+
+		/* calculate new frequency and write it - preserve accum */
+		freq = ((int)rate << 16) / 3000U;
+		SRCRegWrite(dev, (unsigned short) base + SRC_INT_REGS_OFF,
+		    (SRCRegRead(dev, (unsigned short) base + SRC_INT_REGS_OFF)
+		    & 0x00ffU) | ((unsigned short) (freq >> 6) & 0xfc00));
+		SRCRegWrite(dev, (unsigned short) base + SRC_VFREQ_FRAC_OFF,
+		    (unsigned short) freq >> 1);
+
+		/* un-freeze the channel */
+		for (i = 0; i < SRC_IOPOLL_COUNT; ++i)
+			if (!(GET32(dev, CONC_dSRCIO_OFF) & SRC_BUSY))
+				break;
+		PUT32(dev, CONC_dSRCIO_OFF,
+		    (GET32(dev, CONC_dSRCIO_OFF) & SRC_CTLMASK) & ~dtemp);
+	} else {
+		/* derive oversample ratio */
+		N = rate / 3000U;
+		if (N == 15 || N == 13 || N == 11 || N == 9)
+			--N;
+
+		/* truncate the filter and write n/trunc_start */
+		truncM = (21 * N - 1) | 1;
+		if (rate >= 24000U) {
+			if (truncM > 239)
+				truncM = 239;
+			truncStart = (239 - truncM) >> 1;
+
+			SRCRegWrite(dev, base + SRC_TRUNC_N_OFF,
+			    (truncStart << 9) | (N << 4));
+		} else {
+			if (truncM > 119)
+				truncM = 119;
+			truncStart = (119 - truncM) >> 1;
+
+			SRCRegWrite(dev, base + SRC_TRUNC_N_OFF,
+			    0x8000U | (truncStart << 9) | (N << 4));
+		}
+
+		/* calculate new frequency and write it - preserve accum */
+		freq = ((48000UL << 16) / rate) * N;
+		SRCRegWrite(dev, base + SRC_INT_REGS_OFF,
+		    (SRCRegRead(dev, (unsigned short) base + SRC_INT_REGS_OFF)
+		    & 0x00ff) | ((unsigned short) (freq >> 6) & 0xfc00));
+		SRCRegWrite(dev, base + SRC_VFREQ_FRAC_OFF,
+		    (unsigned short) freq >> 1);
+
+		SRCRegWrite(dev, SRC_ADC_VOL_L, N << 8);
+		SRCRegWrite(dev, SRC_ADC_VOL_R, N << 8);
+	}
+}
+
+static void
+SRCInit(audioens_dev_t *dev)
+{
+	int i;
+
+	/* Clear all SRC RAM then init - keep SRC disabled until done */
+	for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+		if (!(GET32(dev, CONC_dSRCIO_OFF) & SRC_BUSY))
+			break;
+	}
+	PUT32(dev, CONC_dSRCIO_OFF, SRC_DISABLE);
+
+	for (i = 0; i < 0x80; ++i)
+		SRCRegWrite(dev, (unsigned short) i, 0U);
+
+	SRCRegWrite(dev, SRC_DAC1_BASE + SRC_TRUNC_N_OFF, 16 << 4);
+	SRCRegWrite(dev, SRC_DAC1_BASE + SRC_INT_REGS_OFF, 16 << 10);
+	SRCRegWrite(dev, SRC_DAC2_BASE + SRC_TRUNC_N_OFF, 16 << 4);
+	SRCRegWrite(dev, SRC_DAC2_BASE + SRC_INT_REGS_OFF, 16 << 10);
+	SRCRegWrite(dev, SRC_DAC1_VOL_L, 1 << 12);
+	SRCRegWrite(dev, SRC_DAC1_VOL_R, 1 << 12);
+	SRCRegWrite(dev, SRC_DAC2_VOL_L, 1 << 12);
+	SRCRegWrite(dev, SRC_DAC2_VOL_R, 1 << 12);
+	SRCRegWrite(dev, SRC_ADC_VOL_L, 1 << 12);
+	SRCRegWrite(dev, SRC_ADC_VOL_R, 1 << 12);
+
+	/* default some rates */
+	SRCSetRate(dev, SRC_DAC1_BASE, 48000);
+	SRCSetRate(dev, SRC_DAC2_BASE, 48000);
+	SRCSetRate(dev, SRC_ADC_BASE, 48000);
+
+	/* now enable the whole deal */
+	for (i = 0; i < SRC_IOPOLL_COUNT; ++i) {
+		if (!(GET32(dev, CONC_dSRCIO_OFF) & SRC_BUSY))
+			break;
+	}
+	PUT32(dev, CONC_dSRCIO_OFF, 0);
+}
+
+static void
+audioens_writemem(audioens_dev_t *dev, uint32_t page, uint32_t offs,
+    uint32_t data)
+{
+	/* Select memory page */
+	PUT32(dev, CONC_bMEMPAGE_OFF, page);
+	PUT32(dev, offs, data);
+}
+
+static uint32_t
+audioens_readmem(audioens_dev_t *dev, uint32_t page, uint32_t offs)
+{
+	PUT32(dev, CONC_bMEMPAGE_OFF, page);	/* Select memory page */
+	return (GET32(dev, offs));
+}
+
+static uint_t
+audioens_intr(caddr_t arg1, caddr_t arg2)
+{
+	audioens_dev_t *dev = (void *)arg1;
+	int stats;
+	int tmp;
+	unsigned char ackbits = 0;
+	audioens_port_t *port;
+	audio_engine_t *do_dac, *do_adc;
+
+	_NOTE(ARGUNUSED(arg2));
+
+	/*
+	 * NB: The old audioens didn't report spurious interrupts.  On
+	 * a system with shared interrupts (typical!) there will
+	 * normally be lots of these (each time the "other" device
+	 * interrupts).
+	 *
+	 * Also, because of the way the interrupt chain handling
+	 * works, reporting of spurious interrupts is probably not
+	 * terribly useful.
+	 *
+	 * However, we can count interrupts where the master interrupt
+	 * bit is set but none of the ackbits that we are prepared to
+	 * process is set.  That is probably useful.
+	 */
+	mutex_enter(&dev->mutex);
+	if (!dev->enabled) {
+
+		mutex_exit(&dev->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	stats = GET32(dev, CONC_dSTATUS_OFF);
+
+	if (!(stats & CONC_STATUS_PENDING)) {	/* No interrupt pending */
+		mutex_exit(&dev->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	do_dac = do_adc = NULL;
+
+	/* DAC1 (synth) interrupt */
+	if (stats & CONC_STATUS_DAC1INT) {
+
+		ackbits |= CONC_SERCTL_DAC1IE;
+		port = &dev->port[PORT_DAC];
+		if (port->trigger) {
+			do_dac = port->engine;
+		}
+	}
+
+	/* DAC2 interrupt */
+	if (stats & CONC_STATUS_DAC2INT) {
+
+		ackbits |= CONC_SERCTL_DAC2IE;
+	}
+
+	/* ADC interrupt */
+	if (stats & CONC_STATUS_ADCINT) {
+
+		ackbits |= CONC_SERCTL_ADCIE;
+		port = &dev->port[PORT_ADC];
+
+		if (port->trigger) {
+			do_adc = port->engine;
+		}
+	}
+
+	/* UART interrupt - we shouldn't get this! */
+	if (stats & CONC_STATUS_UARTINT) {
+		uint8_t uart_stat = GET8(dev, CONC_bUARTCSTAT_OFF);
+		while (uart_stat & CONC_UART_RXRDY)
+			uart_stat = GET8(dev, CONC_bUARTCSTAT_OFF);
+	}
+
+	/* Ack the interrupt */
+	tmp = GET8(dev, CONC_bSERCTL_OFF);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp & (~ackbits));	/* Clear bits */
+	PUT8(dev, CONC_bSERCTL_OFF, tmp | ackbits);	/* Turn them back on */
+
+	if (dev->ksp) {
+		if (ackbits == 0) {
+			KSINTR(dev)->intrs[KSTAT_INTR_SPURIOUS]++;
+		} else {
+			KSINTR(dev)->intrs[KSTAT_INTR_HARD]++;
+		}
+	}
+
+	mutex_exit(&dev->mutex);
+
+	if (do_dac)
+		audio_engine_consume(do_dac);
+	if (do_adc)
+		audio_engine_produce(do_adc);
+
+	return (DDI_INTR_CLAIMED);
+}
+
+/*
+ * Audio routines
+ */
+static int
+audioens_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	/* hardware can also do AUDIO_FORMAT_U8, but no need for it */
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+static int
+audioens_channels(void *arg)
+{
+	audioens_port_t *port = arg;
+
+	return (port->nchan);
+}
+
+static int
+audioens_rate(void *arg)
+{
+	audioens_port_t *port = arg;
+
+	return (port->speed);
+}
+
+static void
+audioens_init_port(audioens_port_t *port)
+{
+	audioens_dev_t	*dev = port->dev;
+	unsigned tmp;
+
+	if (port->suspended)
+		return;
+
+	switch (port->num) {
+	case PORT_DAC:
+		/* Set physical address of the DMA buffer */
+		audioens_writemem(dev, CONC_DAC1CTL_PAGE, CONC_dDAC1PADDR_OFF,
+		    port->paddr);
+		audioens_writemem(dev, CONC_DAC2CTL_PAGE, CONC_dDAC2PADDR_OFF,
+		    port->paddr + (port->nframes * sizeof (int16_t) * 2));
+
+		/* Set DAC rate */
+		SRCSetRate(dev, SRC_DAC1_BASE, port->speed);
+		SRCSetRate(dev, SRC_DAC2_BASE, port->speed);
+
+		/* Configure the channel setup - SPDIF only uses front */
+		tmp = GET32(dev, CONC_dSTATUS_OFF);
+		tmp &= ~(CONC_STATUS_SPKR_MASK | CONC_STATUS_SPDIF_MASK);
+		tmp |= CONC_STATUS_SPKR_4CH | CONC_STATUS_SPDIF_P1;
+		PUT32(dev, CONC_dSTATUS_OFF, tmp);
+
+		/* Set format */
+		PUT8(dev, CONC_bSKIPC_OFF, 0x10);
+		SET8(dev, CONC_bSERFMT_OFF,
+		    CONC_PCM_DAC1_16BIT | CONC_PCM_DAC2_16BIT |
+		    CONC_PCM_DAC1_STEREO | CONC_PCM_DAC2_STEREO);
+
+		/* Set the frame count */
+		audioens_writemem(dev, CONC_DAC1CTL_PAGE, CONC_wDAC1FC_OFF,
+		    port->nframes - 1);
+		audioens_writemem(dev, CONC_DAC2CTL_PAGE, CONC_wDAC2FC_OFF,
+		    port->nframes - 1);
+
+		/* Set # of frames between interrupts */
+		PUT16(dev, CONC_wDAC1IC_OFF, port->fragfr - 1);
+		PUT16(dev, CONC_wDAC2IC_OFF, port->fragfr - 1);
+		break;
+
+	case PORT_ADC:
+		/* Set physical address of the DMA buffer */
+		audioens_writemem(dev, CONC_ADCCTL_PAGE, CONC_dADCPADDR_OFF,
+		    port->paddr);
+
+		/* Set ADC rate */
+		SRCSetRate(dev, SRC_ADC_BASE, port->speed);
+
+		/* Set format - for input we only support 16 bit input */
+		tmp = GET8(dev, CONC_bSERFMT_OFF);
+		tmp |= CONC_PCM_ADC_16BIT;
+		tmp |= CONC_PCM_ADC_STEREO;
+
+		PUT8(dev, CONC_bSKIPC_OFF, 0x10);
+
+		PUT8(dev, CONC_bSERFMT_OFF, tmp);
+
+		/* Set the frame count */
+		audioens_writemem(dev, CONC_ADCCTL_PAGE, CONC_wADCFC_OFF,
+		    port->nframes - 1);
+
+		/* Set # of frames between interrupts */
+		PUT16(dev, CONC_wADCIC_OFF, port->fragfr - 1);
+
+		break;
+	}
+
+	port->frameno = 0;
+}
+
+static int
+audioens_open(void *arg, int flag, unsigned *fragfrp, unsigned *nfragsp,
+    caddr_t *bufp)
+{
+	audioens_port_t	*port = arg;
+	audioens_dev_t	*dev = port->dev;
+	int intrs;
+
+	_NOTE(ARGUNUSED(flag));
+
+	mutex_enter(&dev->mutex);
+
+	if (port->num == PORT_ADC) {
+		intrs = dev->rintrs;
+	} else {
+		intrs = dev->pintrs;
+	}
+
+	/* interrupt at least at 25 Hz, and not more than 250 Hz */
+	intrs = min(250, max(25, intrs));
+
+	port->fragfr = (port->speed / intrs);
+	port->nfrags = AUDIOENS_BUF_LEN /
+	    (port->fragfr * port->nchan * sizeof (int16_t));
+	port->nfrags = max(4, min(port->nfrags, 1024));
+	port->nframes = port->nfrags * port->fragfr;
+	port->trigger = B_FALSE;
+	port->count = 0;
+
+	audioens_init_port(port);
+
+	*fragfrp = port->fragfr;
+	*nfragsp = port->nfrags;
+	*bufp = port->kaddr;
+	mutex_exit(&dev->mutex);
+
+	return (0);
+}
+
+static void
+audioens_start_port(audioens_port_t *port)
+{
+	audioens_dev_t *dev = port->dev;
+
+	if (!port->suspended) {
+		switch (port->num) {
+		case PORT_DAC:
+			SET8(dev, CONC_bDEVCTL_OFF,
+			    CONC_DEVCTL_DAC2_EN | CONC_DEVCTL_DAC1_EN);
+			SET8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_DAC1IE);
+			break;
+		case PORT_ADC:
+			SET8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_ADC_EN);
+			SET8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_ADCIE);
+			break;
+		}
+	}
+}
+
+static void
+audioens_stop_port(audioens_port_t *port)
+{
+	audioens_dev_t *dev = port->dev;
+
+	if (!port->suspended) {
+		switch (port->num) {
+		case PORT_DAC:
+			CLR8(dev, CONC_bDEVCTL_OFF,
+			    CONC_DEVCTL_DAC2_EN | CONC_DEVCTL_DAC1_EN);
+			CLR8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_DAC1IE);
+			break;
+		case PORT_ADC:
+			CLR8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_ADC_EN);
+			CLR8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_ADCIE);
+			break;
+		}
+	}
+}
+
+static int
+audioens_start(void *arg)
+{
+	audioens_port_t *port = arg;
+	audioens_dev_t *dev = port->dev;
+
+	mutex_enter(&dev->mutex);
+	if (!port->trigger) {
+		port->trigger = B_TRUE;
+		audioens_start_port(port);
+	}
+	mutex_exit(&dev->mutex);
+
+	return (0);
+}
+
+static void
+audioens_stop(void *arg)
+{
+	audioens_port_t *port = arg;
+	audioens_dev_t *dev = port->dev;
+
+	mutex_enter(&dev->mutex);
+	if (port->trigger) {
+		port->trigger = B_FALSE;
+		audioens_stop_port(port);
+	}
+	mutex_exit(&dev->mutex);
+}
+
+static void
+audioens_update_port(audioens_port_t *port)
+{
+	uint32_t page, offs;
+	int frameno, n;
+
+	switch (port->num) {
+	case PORT_DAC:
+		page = CONC_DAC1CTL_PAGE;
+		offs = CONC_wDAC1FC_OFF;
+		break;
+
+	case PORT_ADC:
+		page = CONC_ADCCTL_PAGE;
+		offs = CONC_wADCFC_OFF;
+		break;
+	}
+
+	/*
+	 * Note that the current frame counter is in the high nybble.
+	 */
+	frameno = audioens_readmem(port->dev, page, offs) >> 16;
+	n = frameno >= port->frameno ?
+	    frameno - port->frameno :
+	    frameno + port->nframes - port->frameno;
+	port->frameno = frameno;
+	port->count += n;
+}
+
+static uint64_t
+audioens_count(void *arg)
+{
+	audioens_port_t *port = arg;
+	audioens_dev_t *dev = port->dev;
+	uint64_t val;
+
+	mutex_enter(&dev->mutex);
+	if (!port->suspended) {
+		audioens_update_port(port);
+	}
+	val = port->count;
+	mutex_exit(&dev->mutex);
+	return (val);
+}
+
+static void
+audioens_close(void *arg)
+{
+	audioens_port_t *port = arg;
+
+	audioens_stop(port);
+}
+
+static void
+audioens_sync(void *arg, unsigned nframes)
+{
+	audioens_port_t *port = arg;
+
+	_NOTE(ARGUNUSED(nframes));
+
+	if (port->num == PORT_ADC) {
+		(void) ddi_dma_sync(port->dmah, 0, 0, DDI_DMA_SYNC_FORCPU);
+	} else {
+		(void) ddi_dma_sync(port->dmah, 0, 0, DDI_DMA_SYNC_FORDEV);
+	}
+}
+
+static size_t
+audioens_qlen(void *arg)
+{
+	audioens_port_t *port = arg;
+
+	return (port->fragfr);
+}
+
+static void
+audioens_chinfo(void *arg, int chan, unsigned *offset, unsigned *incr)
+{
+	audioens_port_t *port = arg;
+
+	if ((port->num == PORT_DAC) && (chan >= 2)) {
+		*offset = (port->nframes * 2) + (chan % 2);
+		*incr = 2;
+	} else {
+		*offset = chan;
+		*incr = 2;
+	}
+}
+
+audio_engine_ops_t audioens_engine_ops = {
+	AUDIO_ENGINE_VERSION,		/* version number */
+	audioens_open,
+	audioens_close,
+	audioens_start,
+	audioens_stop,
+	audioens_count,
+	audioens_format,
+	audioens_channels,
+	audioens_rate,
+	audioens_sync,
+	audioens_qlen,
+	audioens_chinfo
+};
+
+void
+audioens_init_hw(audioens_dev_t *dev)
+{
+	int tmp;
+
+	if ((dev->devid == ENSONIQ_ES5880) ||
+	    (dev->devid == ENSONIQ_ES5880A) ||
+	    (dev->devid == ENSONIQ_ES5880B) ||
+	    (dev->devid == 0x1371 && dev->revision == 7) ||
+	    (dev->devid == 0x1371 && dev->revision >= 9)) {
+
+		/* Have a ES5880 so enable the codec manually */
+		tmp = GET8(dev, CONC_bINTSUMM_OFF) & 0xff;
+		tmp |= 0x20;
+		PUT8(dev, CONC_bINTSUMM_OFF, tmp);
+		for (int i = 0; i < 2000; i++)
+			drv_usecwait(10);
+	}
+
+	SRCInit(dev);
+
+#if 0
+	PUT8(dev, CONC_bSERCTL_OFF, 0x00);
+	PUT8(dev, CONC_bNMIENA_OFF, 0x00); /* NMI off */
+	PUT8(dev, CONC_wNMISTAT_OFF, 0x00); /* PUT8? */
+#endif
+
+	/*
+	 * Turn on CODEC (UART and joystick left disabled)
+	 */
+	tmp = GET32(dev, CONC_bDEVCTL_OFF) & 0xff;
+	tmp &= ~(CONC_DEVCTL_PCICLK_DS | CONC_DEVCTL_XTALCLK_DS);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bUARTCSTAT_OFF, 0x00);
+
+	/* Perform AC97 codec warm reset */
+	tmp = GET8(dev, CONC_bMISCCTL_OFF) & 0xff;
+	PUT8(dev, CONC_bMISCCTL_OFF, tmp | CONC_MISCCTL_SYNC_RES);
+	drv_usecwait(200);
+	PUT8(dev, CONC_bMISCCTL_OFF, tmp);
+	drv_usecwait(200);
+
+	if (dev->revision >= 4) {
+		/* XXX: enable SPDIF - PCM only for now */
+		if (audioens_spdif) {
+			/* enable SPDIF */
+			PUT32(dev, 0x04, GET32(dev, 0x04) | (1 << 18));
+			/* SPDIF out = data from DAC */
+			PUT32(dev, 0x00, GET32(dev, 0x00) | (1 << 26));
+			CLR32(dev, CONC_dSPDIF_OFF, CONC_SPDIF_AC3);
+
+		} else {
+			/* disable spdif out */
+			PUT32(dev, 0x04, GET32(dev, 0x04) & ~(1 << 18));
+			PUT32(dev, 0x00, GET32(dev, 0x00) & ~(1 << 26));
+		}
+
+		/* we want to run each channel independently */
+		CLR32(dev, CONC_dSTATUS_OFF, CONC_STATUS_ECHO);
+	}
+
+	dev->enabled = B_TRUE;
+}
+
+static int
+audioens_init(audioens_dev_t *dev)
+{
+
+	audioens_init_hw(dev);
+
+	/*
+	 * On this hardware, we want to disable the internal speaker by
+	 * default, if it exists.  (We don't have a speakerphone on any
+	 * of these cards, and no SPARC hardware uses it either!)
+	 */
+	ddi_prop_update_int(DDI_DEV_T_NONE, dev->dip, AC97_PROP_SPEAKER, 0);
+
+	/*
+	 * Init mixer
+	 */
+
+	dev->ac97 = ac97_alloc(dev->dip, audioens_rd97, audioens_wr97, dev);
+	if (dev->ac97 == NULL)
+		return (DDI_FAILURE);
+
+	if (ac97_init(dev->ac97, dev->osdev) != 0) {
+		return (DDI_FAILURE);
+	}
+
+	dev->pintrs = ddi_prop_get_int(DDI_DEV_T_ANY, dev->dip,
+	    DDI_PROP_DONTPASS, "play-interrupts", DEFINTS);
+
+	dev->rintrs = ddi_prop_get_int(DDI_DEV_T_ANY, dev->dip,
+	    DDI_PROP_DONTPASS, "record-interrupts", DEFINTS);
+
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audioens_port_t *port;
+		unsigned caps;
+		unsigned dmaflags;
+		size_t rlen;
+		ddi_dma_cookie_t c;
+		unsigned ccnt;
+
+		port = &dev->port[i];
+		port->dev = dev;
+
+		switch (i) {
+		case PORT_DAC:
+			port->nchan = 4;
+			port->speed = 48000;
+			caps = ENGINE_OUTPUT_CAP;
+			dmaflags = DDI_DMA_WRITE | DDI_DMA_CONSISTENT;
+			break;
+
+		case PORT_ADC:
+			port->nchan = 2;
+			port->speed = 48000;
+			caps = ENGINE_INPUT_CAP;
+			dmaflags = DDI_DMA_READ | DDI_DMA_CONSISTENT;
+			break;
+		}
+
+		port->num = i;
+
+		/*
+		 * Allocate DMA resources.
+		 */
+
+		if (ddi_dma_alloc_handle(dev->dip, &dma_attr, DDI_DMA_SLEEP,
+		    NULL, &port->dmah) != DDI_SUCCESS) {
+			audio_dev_warn(dev->osdev,
+			    "port %d: dma handle allocation failed", i);
+			return (DDI_FAILURE);
+		}
+		if (ddi_dma_mem_alloc(port->dmah, AUDIOENS_BUF_LEN, &buf_attr,
+		    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->kaddr,
+		    &rlen, &port->acch) != DDI_SUCCESS) {
+			audio_dev_warn(dev->osdev,
+			    "port %d: dma memory allocation failed", i);
+			return (DDI_FAILURE);
+		}
+		/* ensure that the buffer is zeroed out properly */
+		bzero(port->kaddr, rlen);
+		if (ddi_dma_addr_bind_handle(port->dmah, NULL, port->kaddr,
+		    AUDIOENS_BUF_LEN, dmaflags, DDI_DMA_SLEEP, NULL,
+		    &c, &ccnt) != DDI_DMA_MAPPED) {
+			audio_dev_warn(dev->osdev,
+			    "port %d: dma binding failed", i);
+			return (DDI_FAILURE);
+		}
+		port->paddr = c.dmac_address;
+
+		/*
+		 * Allocate and configure audio engine.
+		 */
+		port->engine = audio_engine_alloc(&audioens_engine_ops, caps);
+		if (port->engine == NULL) {
+			audio_dev_warn(dev->osdev,
+			    "port %d: audio_engine_alloc failed", i);
+			return (DDI_FAILURE);
+		}
+
+		audio_engine_set_private(port->engine, port);
+		audio_dev_add_engine(dev->osdev, port->engine);
+	}
+
+	/*
+	 * Set up kstats for interrupt reporting.
+	 */
+	dev->ksp = kstat_create(ddi_driver_name(dev->dip),
+	    ddi_get_instance(dev->dip), ddi_driver_name(dev->dip),
+	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT);
+	if (dev->ksp != NULL) {
+		kstat_install(dev->ksp);
+	}
+
+	if (audio_dev_register(dev->osdev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev,
+		    "unable to register with audio framework");
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+}
+
+int
+audioens_setup_interrupts(audioens_dev_t *dev)
+{
+	int actual;
+	uint_t ipri;
+
+	if ((ddi_intr_alloc(dev->dip, dev->ihandle, DDI_INTR_TYPE_FIXED,
+	    0, 1, &actual, DDI_INTR_ALLOC_NORMAL) != DDI_SUCCESS) ||
+	    (actual != 1)) {
+		audio_dev_warn(dev->osdev, "can't alloc intr handle");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_get_pri(dev->ihandle[0], &ipri) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev,  "can't determine intr priority");
+		(void) ddi_intr_free(dev->ihandle[0]);
+		dev->ihandle[0] = NULL;
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_add_handler(dev->ihandle[0], audioens_intr, dev,
+	    NULL) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev, "can't add intr handler");
+		(void) ddi_intr_free(dev->ihandle[0]);
+		dev->ihandle[0] = NULL;
+		return (DDI_FAILURE);
+	}
+
+	mutex_init(&dev->mutex, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+
+	return (DDI_SUCCESS);
+}
+
+void
+audioens_destroy(audioens_dev_t *dev)
+{
+	int	i;
+
+	if (dev->ihandle[0] != NULL) {
+		(void) ddi_intr_disable(dev->ihandle[0]);
+		(void) ddi_intr_remove_handler(dev->ihandle[0]);
+		(void) ddi_intr_free(dev->ihandle[0]);
+		mutex_destroy(&dev->mutex);
+	}
+
+	if (dev->ksp != NULL) {
+		kstat_delete(dev->ksp);
+	}
+
+	/* free up ports, including DMA resources for ports */
+	for (i = 0; i <= PORT_MAX; i++) {
+		audioens_port_t	*port = &dev->port[i];
+
+		if (port->paddr != 0)
+			(void) ddi_dma_unbind_handle(port->dmah);
+		if (port->acch != NULL)
+			ddi_dma_mem_free(&port->acch);
+		if (port->dmah != NULL)
+			ddi_dma_free_handle(&port->dmah);
+
+		if (port->engine != NULL) {
+			audio_dev_remove_engine(dev->osdev, port->engine);
+			audio_engine_free(port->engine);
+		}
+	}
+
+	if (dev->acch != NULL) {
+		ddi_regs_map_free(&dev->acch);
+	}
+
+	if (dev->ac97) {
+		ac97_free(dev->ac97);
+	}
+
+	if (dev->osdev != NULL) {
+		audio_dev_free(dev->osdev);
+	}
+
+	kmem_free(dev, sizeof (*dev));
+}
+
+int
+audioens_attach(dev_info_t *dip)
+{
+	uint16_t pci_command, vendor, device;
+	uint8_t revision;
+	audioens_dev_t *dev;
+	ddi_acc_handle_t pcih;
+	const char *chip_name;
+	const char *chip_vers;
+
+	dev = kmem_zalloc(sizeof (*dev), KM_SLEEP);
+	dev->dip = dip;
+	ddi_set_driver_private(dip, dev);
+
+	if (pci_config_setup(dip, &pcih) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev, "pci_config_setup failed");
+		kmem_free(dev, sizeof (*dev));
+		return (DDI_FAILURE);
+	}
+
+	vendor = pci_config_get16(pcih, PCI_CONF_VENID);
+	device = pci_config_get16(pcih, PCI_CONF_DEVID);
+	revision = pci_config_get8(pcih, PCI_CONF_REVID);
+
+	if ((vendor != ENSONIQ_VENDOR_ID && vendor != CREATIVE_VENDOR_ID) ||
+	    (device != ENSONIQ_ES1371 && device != ENSONIQ_ES5880 &&
+	    device != ENSONIQ_ES5880A && device != ECTIVA_ES1938 &&
+	    device != ENSONIQ_ES5880B))
+		goto err_exit;
+
+	chip_name = "AudioPCI97";
+	chip_vers = "unknown";
+
+	switch (device) {
+	case ENSONIQ_ES1371:
+		chip_name = "AudioPCI97";
+		switch (revision) {
+		case 0x02:
+		case 0x09:
+		default:
+			chip_vers = "ES1371";
+			break;
+		case 0x04:
+		case 0x06:
+		case 0x08:
+			chip_vers = "ES1373";
+			break;
+		case 0x07:
+			chip_vers = "ES5880";
+			break;
+		}
+		break;
+
+	case ENSONIQ_ES5880:
+		chip_name = "SB PCI128";
+		chip_vers = "ES5880";
+		break;
+	case ENSONIQ_ES5880A:
+		chip_name = "SB PCI128";
+		chip_vers = "ES5880A";
+		break;
+	case ENSONIQ_ES5880B:
+		chip_name = "SB PCI128";
+		chip_vers = "ES5880B";
+		break;
+
+	case ECTIVA_ES1938:
+		chip_name = "AudioPCI";
+		chip_vers = "ES1938";
+		break;
+	}
+
+	dev->revision = revision;
+	dev->devid = device;
+
+	dev->osdev = audio_dev_alloc(dip, 0);
+	if (dev->osdev == NULL) {
+		goto err_exit;
+	}
+
+	audio_dev_set_description(dev->osdev, chip_name);
+	audio_dev_set_version(dev->osdev, chip_vers);
+
+	/* set the PCI latency */
+	if ((audioens_latency == 32) || (audioens_latency == 64) ||
+	    (audioens_latency == 96))
+		pci_config_put8(pcih, PCI_CONF_LATENCY_TIMER,
+		    audioens_latency);
+
+	/* activate the device */
+	pci_command = pci_config_get16(pcih, PCI_CONF_COMM);
+	pci_command |= PCI_COMM_ME | PCI_COMM_IO;
+	pci_config_put16(pcih, PCI_CONF_COMM, pci_command);
+
+	/* map registers */
+	if (ddi_regs_map_setup(dip, 1, &dev->regs, 0, 0, &acc_attr,
+	    &dev->acch) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev, "can't map registers");
+		goto err_exit;
+	}
+
+	if (audioens_setup_interrupts(dev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev, "can't register interrupts");
+		goto err_exit;
+	}
+
+	/* This allocates and configures the engines */
+	if (audioens_init(dev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->osdev, "can't init device");
+		goto err_exit;
+	}
+
+	(void) ddi_intr_enable(dev->ihandle[0]);
+
+	pci_config_teardown(&pcih);
+
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+err_exit:
+	pci_config_teardown(&pcih);
+
+	audioens_destroy(dev);
+
+	return (DDI_FAILURE);
+}
+
+int
+audioens_detach(audioens_dev_t *dev)
+{
+	int tmp;
+
+	/* first unregister us from the DDI framework, might be busy */
+	if (audio_dev_unregister(dev->osdev) != DDI_SUCCESS)
+		return (DDI_FAILURE);
+
+	mutex_enter(&dev->mutex);
+
+	tmp = GET8(dev, CONC_bSERCTL_OFF) &
+	    ~(CONC_SERCTL_DAC2IE | CONC_SERCTL_DAC1IE | CONC_SERCTL_ADCIE);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+
+	tmp = GET8(dev, CONC_bDEVCTL_OFF) &
+	    ~(CONC_DEVCTL_DAC2_EN | CONC_DEVCTL_ADC_EN | CONC_DEVCTL_DAC1_EN);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+
+	dev->enabled = B_FALSE;
+
+	mutex_exit(&dev->mutex);
+
+	audioens_destroy(dev);
+
+	return (DDI_SUCCESS);
+}
+
+/*ARGSUSED*/
+static int
+audioens_resume(audioens_dev_t *dev)
+{
+	/* ask framework to reset/relocate engine data */
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audio_engine_reset(dev->port[i].engine);
+	}
+
+	/* reinitialize hardware */
+	audioens_init_hw(dev);
+
+	/* restore AC97 state */
+	ac97_resume(dev->ac97);
+
+	/* restart ports */
+	mutex_enter(&dev->mutex);
+	for (int i = 0; i < PORT_MAX; i++) {
+		audioens_port_t	*port = &dev->port[i];
+		port->suspended = B_FALSE;
+		audioens_init_port(port);
+		/* possibly start it up if was going when we suspended */
+		if (port->trigger) {
+			audioens_start_port(port);
+
+		}
+	}
+	mutex_exit(&dev->mutex);
+	for (int i = 0; i < PORT_MAX; i++) {
+		audioens_port_t	*port = &dev->port[i];
+		/* signal callbacks on resume */
+		if (!port->trigger)
+			continue;
+		if (port->num == PORT_ADC) {
+			audio_engine_produce(port->engine);
+		} else {
+			audio_engine_consume(port->engine);
+		}
+	}
+	return (DDI_SUCCESS);
+}
+
+/*ARGSUSED*/
+static int
+audioens_suspend(audioens_dev_t *dev)
+{
+	/*
+	 * Stop all engines/DMA data.
+	 */
+	mutex_enter(&dev->mutex);
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audioens_stop_port(&dev->port[i]);
+		audioens_update_port(&dev->port[i]);
+		dev->port[i].suspended = B_TRUE;
+	}
+	dev->enabled = B_FALSE;
+	mutex_exit(&dev->mutex);
+
+	/*
+	 * Framework needs to save off AC'97 state.
+	 */
+	ac97_suspend(dev->ac97);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audioens_quiesce(dev_info_t *dip)
+{
+	audioens_dev_t	*dev;
+	uint8_t		tmp;
+
+	if ((dev = ddi_get_driver_private(dip)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	/* This disables all DMA engines and interrupts */
+	tmp = GET8(dev, CONC_bSERCTL_OFF) &
+	    ~(CONC_SERCTL_DAC2IE | CONC_SERCTL_DAC1IE | CONC_SERCTL_ADCIE);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+
+	tmp = GET8(dev, CONC_bDEVCTL_OFF) &
+	    ~(CONC_DEVCTL_DAC2_EN | CONC_DEVCTL_ADC_EN | CONC_DEVCTL_DAC1_EN);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+
+	return (DDI_SUCCESS);
+}
+
+
+static int
+audioens_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	audioens_dev_t *dev;
+
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audioens_attach(dip));
+
+	case DDI_RESUME:
+		if ((dev = ddi_get_driver_private(dip)) == NULL) {
+			return (DDI_FAILURE);
+		}
+		return (audioens_resume(dev));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+static int
+audioens_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	audioens_dev_t *dev;
+
+	if ((dev = ddi_get_driver_private(dip)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audioens_detach(dev));
+
+	case DDI_SUSPEND:
+		return (audioens_suspend(dev));
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+static int audioens_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audioens_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+
+static struct dev_ops audioens_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	NULL,			/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	audioens_ddi_attach,	/* attach */
+	audioens_ddi_detach,	/* detach */
+	nodev,			/* reset */
+	NULL,			/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+	audioens_quiesce,	/* quiesce */
+};
+
+static struct modldrv audioens_modldrv = {
+	&mod_driverops,			/* drv_modops */
+	"Ensoniq 1371/1373 Audio",	/* linkinfo */
+	&audioens_dev_ops,		/* dev_ops */
+};
+
+static struct modlinkage modlinkage = {
+	MODREV_1,
+	{ &audioens_modldrv, NULL }
+};
+
+int
+_init(void)
+{
+	int	rv;
+
+	audio_init_ops(&audioens_dev_ops, DRVNAME);
+	if ((rv = mod_install(&modlinkage)) != 0) {
+		audio_fini_ops(&audioens_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_fini(void)
+{
+	int	rv;
+
+	if ((rv = mod_remove(&modlinkage)) == 0) {
+		audio_fini_ops(&audioens_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&modlinkage, modinfop));
+}
diff --git a/usr/src/uts/common/io/audio/sada/drv/audio810/audio810.conf b/usr/src/uts/common/io/audio/drv/audioens/audioens.conf
index 3ec17d2cbc..7e116fffb6 100644
--- a/usr/src/uts/common/io/audio/sada/drv/audio810/audio810.conf
+++ b/usr/src/uts/common/io/audio/drv/audioens/audioens.conf
@@ -19,51 +19,35 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# Configuration file for the audio810 audio driver.
+# Configuration file for the audioens audio driver.
 #
 #	WARNING: This is an UNSTABLE configuration file. Its contents
 #		may change at any time.
 
 #
-# Uncomment cdrom to enable the audio810 driver to use the CDROM analog
-# input. 
-#
-# cdrom=1;
-
-#
 # play-interrupts sets the number of interrupts per second when playing.
 # This affects the resolution of various things, such as sample counts.
 # record-interrupts does the same for record interrupts.
 #
 # These may be tuned to get more accurate information by increasing the
 # count. However, the larger the interrupts per second the larger the
-# load on the system. So use this capability cautiously. The audio810
+# load on the system. So use this capability cautiously. The audioens
 # driver enforces a maximum and minimum count.
 #
 # It should also be understood that not all interrupt rates are legal.
 # The hardware is restricted to DMA buffers being allocated on certain
-# boundaries. If those boundaries are violated the driver will not be
-# loaded and an error message is entered into the messages log
+# boundaries. If those boundaries are violated then the value specified
+# will be ignored.
 #
-play-interrupts=175;
-record-interrupts=175;
+play-interrupts=75;
+record-interrupts=75;
 
 #
-# Uncomment reset-configuration to cause the audio810 driver's state to
+# Uncomment reset-configuration to cause the audioens driver's state to
 # be reset to the default when the driver is loaded. Otherwise this state
 # is retained across driver unload/reload cycles, but not across reboots.
 #
 #reset-configuration=1;
-
-#
-# Uncomment ac97-invert-amp to power down/up external amplifier. Most
-# hardware does't need this property. But some devices, such as some
-# Sony Vaio laptops need to set this property to 1 to correct the
-# inverted polarity of the external amplifier device.
-#
-# ac97-invert-amp=1;
diff --git a/usr/src/uts/common/io/audio/drv/audioens/audioens.h b/usr/src/uts/common/io/audio/drv/audioens/audioens.h
new file mode 100644
index 0000000000..6097e076db
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audioens/audioens.h
@@ -0,0 +1,210 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * Purpose: Definitions for the Creative/Ensoniq AudioPCI97 driver.
+ */
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#ifndef	_AUDIOENS_H
+#define	_AUDIOENS_H
+
+/* CONCERT PCI-SIG defines */
+#define	CONC_PCI_VENDID		0x1274U
+#define	CONC_PCI_DEVID		0x1371U
+
+/* Concert97 direct register offset defines */
+#define	CONC_bDEVCTL_OFF	0x00	/* Device control/enable */
+#define	CONC_bMISCCTL_OFF	0x01	/* Miscellaneous control */
+#define	CONC_bGPIO_OFF		0x02	/* General purpose I/O control */
+#define	CONC_bJOYCTL_OFF	0x03	/* Joystick control (decode) */
+#define	CONC_dSTATUS_OFF	0x04	/* long status register */
+#define	CONC_bINTSUMM_OFF	0x07	/* Interrupt summary status */
+#define	CONC_bUARTDATA_OFF	0x08	/* UART data R/W - read clears RX int */
+#define	CONC_bUARTCSTAT_OFF	0x09	/* UART control and status */
+#define	CONC_bUARTTEST_OFF	0x0a	/* UART test control reg */
+#define	CONC_bMEMPAGE_OFF	0x0c	/* Memory page select */
+#define	CONC_dSRCIO_OFF		0x10	/* I/O ctl/stat/data for SRC RAM */
+#define	CONC_dCODECCTL_OFF	0x14	/* CODEC control - dword read/write */
+#define	CONC_wNMISTAT_OFF	0x18	/* Legacy NMI status */
+#define	CONC_bNMIENA_OFF	0x1a	/* Legacy NMI enable */
+#define	CONC_bNMICTL_OFF	0x1b	/* Legacy control */
+#define	CONC_dSPDIF_OFF		0x1c	/* SPDIF status control */
+#define	CONC_bSERFMT_OFF	0x20	/* Serial device control */
+#define	CONC_bSERCTL_OFF	0x21	/* Serial device format */
+#define	CONC_bSKIPC_OFF		0x22	/* DAC skip count reg */
+#define	CONC_wDAC1IC_OFF	0x24	/* Synth int count in sample frames */
+#define	CONC_wDAC1CIC_OFF	0x26	/* Synth current int count */
+#define	CONC_wDAC2IC_OFF	0x28	/* DAC int count in sample frames */
+#define	CONC_wDAC2CIC_OFF	0x2a	/* DAC current int count */
+#define	CONC_wADCIC_OFF		0x2c	/* ADC int count in sample frames */
+#define	CONC_wADCCIC_OFF	0x2e	/* ADC current int count */
+#define	CONC_MEMBASE_OFF	0x30 /* Memory window base - 16 byte window */
+
+/* Concert memory page-banked register offset defines */
+#define	CONC_dDAC1PADDR_OFF	0x30	/* Synth host frame PCI phys addr */
+#define	CONC_wDAC1FC_OFF	0x34	/* Synth host frame count in DWORDS */
+#define	CONC_wDAC1CFC_OFF	0x36	/* Synth host current frame count */
+#define	CONC_dDAC2PADDR_OFF	0x38	/* DAC host frame PCI phys addr */
+#define	CONC_wDAC2FC_OFF	0x3c	/* DAC host frame count in DWORDS */
+#define	CONC_wDAC2CFC_OFF	0x3e	/* DAC host current frame count */
+#define	CONC_dADCPADDR_OFF	0x30	/* ADC host frame PCI phys addr */
+#define	CONC_wADCFC_OFF		0x34	/* ADC host frame count in DWORDS */
+#define	CONC_wADCCFC_OFF	0x36	/* ADC host current frame count */
+
+/* Concert memory page number defines */
+#define	CONC_DAC1RAM_PAGE	0x00	/* Synth host/serial I/F RAM */
+#define	CONC_DAC2RAM_PAGE	0x04	/* DAC host/serial I/F RAM */
+#define	CONC_ADCRAM_PAGE	0x08	/* ADC host/serial I/F RAM */
+#define	CONC_DAC1CTL_PAGE	0x0c	/* Page bank for synth host control */
+#define	CONC_DAC2CTL_PAGE	0x0c	/* Page bank for DAC host control */
+#define	CONC_ADCCTL_PAGE	0x0d	/* Page bank for ADC host control */
+#define	CONC_FIFO0_PAGE		0x0e	/* page 0 of UART "FIFO" (rx stash) */
+#define	CONC_FIFO1_PAGE		0x0f	/* page 1 of UART "FIFO" (rx stash) */
+
+/* SPDIF defines - only newer chips */
+#define	CONC_SPDIF_CLKACCURACY	0x00000000U	/* normal mode */
+#define	CONC_SPDIF_SR48KHZ	0x02000000U	/* 48KHZ clock, must be set */
+#define	CONC_SPDIF_CHNO_MASK	0x00f00000U	/* channel number */
+#define	CONC_SPDIF_SRCNO_MASK	0x000f0000U	/* source number */
+#define	CONC_SPDIF_L		0x00008000U	/* 0 = commercial original */
+#define	CONC_SPDIF_CATCODE	0x00007f00U	/* category code */
+#define	CONC_SPDIF_EMPHASIS	0x00000008U	/* 2 ch, 50/15 usec preemph */
+#define	CONC_SPDIF_COPY		0x00000004U	/* copy permitted */
+#define	CONC_SPDIF_AC3		0x00000002U	/* data is not pcm (AC3) */
+
+/* PCM format defines */
+#define	CONC_PCM_DAC1_STEREO	0x01
+#define	CONC_PCM_DAC1_16BIT	0x02
+#define	CONC_PCM_DAC2_STEREO	0x04
+#define	CONC_PCM_DAC2_16BIT	0x08
+#define	CONC_PCM_ADC_STEREO	0x10
+#define	CONC_PCM_ADC_16BIT	0x20
+
+/* Device Control defines */
+#define	CONC_DEVCTL_PCICLK_DS	0x01	/* PCI Clock Disable */
+#define	CONC_DEVCTL_XTALCLK_DS	0x02	/* Crystal Clock Disable */
+#define	CONC_DEVCTL_JSTICK_EN	0x04	/* Joystick Enable */
+#define	CONC_DEVCTL_UART_EN	0x08	/* UART Enable  */
+#define	CONC_DEVCTL_ADC_EN	0x10	/* ADC Enable (record) */
+#define	CONC_DEVCTL_DAC2_EN	0x20	/* DAC2 Enable (playback) */
+#define	CONC_DEVCTL_DAC1_EN	0x40	/* DAC1 Enabale (synth) */
+
+/* Misc Control defines */
+#define	CONC_MISCCTL_PDLEV_D0	0x00	/* These bits reflect the */
+#define	CONC_MISCCTL_PDLEV_D1	0x01	/* power down state of  */
+#define	CONC_MISCCTL_PDLEV_D2	0x02	/* the part */
+#define	CONC_MISCCTL_PDLEV_D3	0x03	/* */
+#define	CONC_MISCCTL_CCBINTRM_EN	0x04	/* CCB module interrupt mask */
+
+#define	CONC_MISCCTL_SYNC_RES	0x40	/* for AC97 warm reset */
+
+/* Serial Control defines */
+#define	CONC_SERCTL_DAC1IE	0x01 /* playback interrupt enable P1_INT_EN */
+#define	CONC_SERCTL_DAC2IE	0x02 /* playback interrupt enable P2_INT_EN */
+#define	CONC_SERCTL_ADCIE	0x04	/* record interrupt enable R1_INT_EN */
+#define	CONC_SERCTL_DAC1PAUSE	0x08	/* playback pause */
+#define	CONC_SERCTL_DAC2PAUSE	0x10	/* playback pause */
+#define	CONC_SERCTL_ADCLOOP	0x80
+#define	CONC_SERCTL_DAC2LOOP	0x40
+#define	CONC_SERCTL_DAC1LOOP	0x20
+
+/* Interrupt Status defines */
+#define	CONC_STATUS_ADCINT	0x00000001	/* A/D interrupt pending */
+#define	CONC_STATUS_DAC2INT	0x00000002	/* DAC2 interrupt pending */
+#define	CONC_STATUS_DAC1INT	0x00000004	/* DAC1 interrupt pending */
+#define	CONC_STATUS_UARTINT	0x00000008	/* UART interrupt pending */
+#define	CONC_STATUS_PENDING	0x80000000	/* any interrupt pending */
+#define	CONC_STATUS_SPDIF_MASK	0x18000000
+#define	CONC_STATUS_SPDIF_P1P2	0x00000000
+#define	CONC_STATUS_SPDIF_P1	0x08000000
+#define	CONC_STATUS_SPDIF_P2	0x10000000
+#define	CONC_STATUS_SPDIF_REC	0x18000000
+#define	CONC_STATUS_ECHO	0x04000000
+#define	CONC_STATUS_SPKR_MASK	0x03000000
+#define	CONC_STATUS_SPKR_2CH	0x00000000
+#define	CONC_STATUS_SPKR_4CH	0x01000000
+#define	CONC_STATUS_SPKR_P1	0x02000000
+#define	CONC_STATUS_SPKR_P2	0x03000000
+#define	CONC_STATUS_EN_SPDIF	0x00040000
+
+/* JOYCTL register defines */
+#define	CONC_JOYCTL_200		0x00
+#define	CONC_JOYCTL_208		0x01
+#define	CONC_JOYCTL_210		0x02
+#define	CONC_JOYCTL_218		0x03
+#define	CONC_JOYCTL_SPDIFEN_B	0x04
+#define	CONC_JOYCTL_RECEN_B	0x08
+
+/* UARTCSTAT register masks */
+#define	CONC_UART_RXRDY		0x01
+#define	CONC_UART_TXRDY		0x02
+#define	CONC_UART_TXINT		0x04
+#define	CONC_UART_RXINT		0x80
+
+#define	CONC_UART_CTL		0x03
+#define	CONC_UART_TXINTEN	0x20
+#define	CONC_UART_RXINTEN	0x80
+
+/* defines for the CONCERT97 Sample Rate Converters */
+
+/* register/base equates for the SRC RAM */
+#define	SRC_DAC1_FIFO		0x00
+#define	SRC_DAC2_FIFO		0x20
+#define	SRC_ADC_FIFO		0x40
+#define	SRC_ADC_VOL_L		0x6c
+#define	SRC_ADC_VOL_R		0x6d
+#define	SRC_DAC1_BASE		0x70
+#define	SRC_DAC2_BASE		0x74
+#define	SRC_ADC_BASE		0x78
+#define	SRC_DAC1_VOL_L		0x7c
+#define	SRC_DAC1_VOL_R		0x7d
+#define	SRC_DAC2_VOL_L		0x7e
+#define	SRC_DAC2_VOL_R		0x7f
+
+#define	SRC_TRUNC_N_OFF		0x00
+#define	SRC_INT_REGS_OFF	0x01
+#define	SRC_ACCUM_FRAC_OFF	0x02
+#define	SRC_VFREQ_FRAC_OFF	0x03
+
+
+/* miscellaneous control defines */
+#define	SRC_IOPOLL_COUNT	0x20000UL
+#define	SRC_WENABLE		(1UL << 24)
+#define	SRC_BUSY		(1UL << 23)
+#define	SRC_DISABLE		(1UL << 22)
+#define	SRC_DAC1FREEZE		(1UL << 21)
+#define	SRC_DAC2FREEZE		(1UL << 20)
+#define	SRC_ADCFREEZE		(1UL << 19)
+#define	SRC_CTLMASK		0x00780000UL
+
+#endif /* _AUDIOENS_H */
diff --git a/usr/src/uts/common/io/audio/sada/drv/audiohd/audiohd.c b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.c
index 7ea73cd696..0b3b5dfee8 100644
--- a/usr/src/uts/common/io/audio/sada/drv/audiohd/audiohd.c
+++ b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.c
@@ -23,66 +23,23 @@
  * Use is subject to license terms.
  */
 
-/*
- * The audiohd driver provides functionality for playing audio on
- * the nVidia MCP55 high-definition audio controller with Realtek
- * ALC88x codec. This enables basic audio functionality for Sun's
- * 2006 line of new workstations and PCs, which are MCP55-based
- * with Realtek ALC88x codec. Due to short time constraints on the
- * production of this software kernel module, we used a minimalistic
- * approach to provide audio functionality for just the nVidia HD
- * audio controller and the Realtek ALC880, ALC883 and ALC885
- * codecs.  Certainly, the driver may work and attach to Intel
- * High-Definition audio devices which have same the Realtek Codec.
- *
- * HD audio supports multiple streams, each of which can act as an
- * independent device. However, we just support two streams: the
- * first input stream for recording, and the first output stream
- * for playback. And because ALC880 doesn't support sample rates
- * below 48K (except 44.1K), this driver just supports 48K sample
- * rate in compatible mode.
- *
- */
-#include <sys/types.h>
-#include <sys/modctl.h>
-#include <sys/conf.h>
-#include <sys/devops.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
+#include <sys/audio/audio_driver.h>
 #include <sys/note.h>
 #include <sys/pci.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/audiohd.h>
-#include <sys/audio/impl/audiohd_impl.h>
+#include "audiohd.h"
 
+#define	DEFINTS			175
+#define	DRVNAME			"audiohd"
 /*
  * Module linkage routines for the kernel
  */
-static int audiohd_getinfo(dev_info_t *, ddi_info_cmd_t, void *, void **);
+
 static int audiohd_attach(dev_info_t *, ddi_attach_cmd_t);
 static int audiohd_detach(dev_info_t *, ddi_detach_cmd_t);
 static int audiohd_quiesce(dev_info_t *);
 static int audiohd_resume(audiohd_state_t *);
 static int audiohd_suspend(audiohd_state_t *);
 
-/*
- * Entry point routine prototypes
- */
-static int audiohd_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int audiohd_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int audiohd_ad_start_play(audiohdl_t, int);
-static int audiohd_ad_start_record(audiohdl_t, int);
-static void audiohd_ad_pause_play(audiohdl_t, int);
-static void audiohd_ad_stop_play(audiohdl_t, int);
-static void audiohd_ad_stop_record(audiohdl_t, int);
-
 /* interrupt handler */
 static uint_t audiohd_intr(caddr_t);
 
@@ -99,68 +56,26 @@ static void audiohd_stop_dma(audiohd_state_t *);
 static void audiohd_disable_intr(audiohd_state_t *);
 static int audiohd_create_codec(audiohd_state_t *);
 static void audiohd_build_path(audiohd_state_t *);
-static int audiohd_init_ports(audiohd_state_t *);
 static void audiohd_destroy_codec(audiohd_state_t *);
-static int audiohd_reset_stream(audiohd_state_t	*, int);
-static int audiohd_fill_pbuf(audiohd_state_t *);
-static void audiohd_refill_pbuf(audiohd_state_t *);
-static void audiohd_preset_rbuf(audiohd_state_t *);
-static void audiohd_get_rbuf(audiohd_state_t *);
-static int audiohd_set_gain(audiohd_state_t *, int, int, int);
-static int audiohd_set_port(audiohd_state_t *, int, int);
-static void audiohd_mute_outputs(audiohd_state_t *, boolean_t);
-static int audiohd_set_monitor_gain(audiohd_state_t *, int);
 static int audiohd_alloc_dma_mem(audiohd_state_t *, audiohd_dma_t *,
     size_t, ddi_dma_attr_t *, uint_t);
 static void audiohd_finish_output_path(hda_codec_t *codec);
-static void audiohd_finish_input_path(hda_codec_t *codec);
-static void audiohd_finish_monitor_path(hda_codec_t *codec);
-
 static uint32_t audioha_codec_verb_get(void *, uint8_t,
     uint8_t, uint16_t, uint8_t);
 static uint32_t audioha_codec_4bit_verb_get(void *, uint8_t,
     uint8_t, uint16_t, uint16_t);
 static int audiohd_reinit_hda(audiohd_state_t *);
-static void audiohd_set_busy(audiohd_state_t *);
-static void audiohd_set_idle(audiohd_state_t *);
 static int audiohd_response_from_codec(audiohd_state_t *statep,
     uint32_t *resp, uint32_t *respex);
 static void audiohd_restore_codec_gpio(audiohd_state_t *statep);
 static void audiohd_change_speaker_state(audiohd_state_t *statep, int on);
-/* anchor for soft state structures */
-static void *audiohd_statep;
-
-/* driver name */
-static char *audiohd_name = AUDIOHD_NAME;
-
-static uint_t audiohd_mixer_srs[] = {
-	AUDIOHD_SAMPR5510, AUDIOHD_SAMPR48000, 0
-};
-
-static uint_t audiohd_comp_srs[] = {
-	AUDIOHD_SAMPR48000,
-	0
-};
-
-static am_ad_sample_rates_t audiohd_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	audiohd_mixer_srs
-};
-
-static am_ad_sample_rates_t audiohd_comp_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audiohd_comp_srs
-};
-
-static uint_t audiohd_channels[] = {
-	AUDIO_CHANNELS_STEREO,
-	0
-};
-
-static am_ad_cap_comb_t audiohd_combinations[] = {
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
+static int audiohd_allocate_port(audiohd_state_t *statep);
+static void audiohd_free_port(audiohd_state_t *statep);
+static void audiohd_restore_path(audiohd_state_t *statep);
+static int audiohd_add_controls(audiohd_state_t *statep);
+static void audiohd_get_channels(audiohd_state_t *statep);
+static void audiohd_init_path(audiohd_state_t *statep);
+static void audiohd_del_controls(audiohd_state_t *statep);
 
 static ddi_device_acc_attr_t hda_dev_accattr = {
 	DDI_DEVICE_ATTR_V0,
@@ -168,222 +83,120 @@ static ddi_device_acc_attr_t hda_dev_accattr = {
 	DDI_STRICTORDER_ACC
 };
 
-/* STREAMS driver id and limit value struct */
-static struct module_info audiohd_modinfo = {
-	AUDIOHD_IDNUM,
-	AUDIOHD_NAME,
-	AUDIOHD_MINPACKET,
-	AUDIOHD_MAXPACKET,
-	AUDIOHD_HIWATER,
-	AUDIOHD_LOWATER,
-};
-
-/* STREAMS queue processing procedures structures for read queue */
-static struct qinit audiohd_rqueue = {
-	audio_sup_rput,
-	audio_sup_rsvc,
-	audio_sup_open,
-	audio_sup_close,
-	NULL,
-	&audiohd_modinfo,
+static const char *audiohd_dtypes[] = {
+	AUDIO_PORT_LINEOUT,
+	AUDIO_PORT_SPEAKER,
+	AUDIO_PORT_HEADPHONES,
+	AUDIO_PORT_CD,
+	AUDIO_PORT_SPDIFOUT,
+	AUDIO_PORT_DIGOUT,
+	AUDIO_PORT_MODEM,
+	AUDIO_PORT_HANDSET,
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_AUX1IN,
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_PHONE,
+	AUDIO_PORT_SPDIFIN,
+	AUDIO_PORT_DIGIN,
+	AUDIO_PORT_NONE,	/* reserved port, don't use */
+	AUDIO_PORT_OTHER,
 	NULL,
 };
 
-/* STREAMS queue processing procedures structures for write queue */
-static struct qinit audiohd_wqueue = {
-	audio_sup_wput,
-	audio_sup_wsvc,
-	NULL,
-	NULL,
-	NULL,
-	&audiohd_modinfo,
-	NULL
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab audiohd_streamtab = {
-	&audiohd_rqueue,
-	&audiohd_wqueue,
-	NULL,
-	NULL,
-};
-
-/* Entry points structure */
-static struct cb_ops audiohd_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&audiohd_streamtab,		/* cb_str */
-	D_NEW | D_MP | D_64BIT,	/* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev			/* cb_awrite */
-};
-
-/* Device operations structure */
-static struct dev_ops audiohd_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	audiohd_getinfo,	/* devo_getinfo */
-	nulldev,		/* devo_identify */
-	nulldev,		/* devo_probe */
-	audiohd_attach,		/* devo_attach */
-	audiohd_detach,		/* devo_detach */
-	nodev,			/* devo_reset */
-	&audiohd_cb_ops,		/* devo_cb_ops */
-	NULL,			/* devo_bus_ops */
-	NULL,			/* devo_power */
-	audiohd_quiesce,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv audiohd_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	AUDIOHD_MOD_NAME,	/* drv_linkinfo */
-	&audiohd_dev_ops,		/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage audiohd_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&audiohd_modldrv,	/* ml_linkage */
-	NULL				/* NULL */
-};
-
-
-/*
- * Audio driver ops vector for mixer
- */
-static am_ad_entry_t audiohd_entry = {
-	NULL,				/* ad_setup() */
-	NULL,				/* ad_teardown() */
-	audiohd_ad_set_config,		/* ad_set_config() */
-	audiohd_ad_set_format,		/* ad_set_format() */
-	audiohd_ad_start_play,		/* ad_start_play() */
-	audiohd_ad_pause_play,		/* ad_pause_play() */
-	audiohd_ad_stop_play,		/* ad_stop_play() */
-	audiohd_ad_start_record,	/* ad_start_record() */
-	audiohd_ad_stop_record,		/* ad_stop_record() */
-	NULL,				/* ad_ioctl() */
-	NULL				/* ad_iocdata() */
-};
-
-static const uint_t g_outport[] = {
-	AUDIO_LINE_OUT,
-	AUDIO_SPEAKER,
-	AUDIO_HEADPHONE,
-	AUDIO_AUX1_OUT,
-	AUDIO_AUX2_OUT
+enum {
+	CTL_VOLUME = 0,
+	CTL_FRONT,
+	CTL_SPEAKER,
+	CTL_HEADPHONE,
+	CTL_REAR,
+	CTL_CENTER,
+	CTL_SURROUND,
+	CTL_LFE,
+	CTL_IGAIN,
+	CTL_LINEIN,
+	CTL_MIC,
+	CTL_CD,
+	CTL_MONGAIN,
+	CTL_MONSRC,
+	CTL_RECSRC
 };
 
-
-int
-_init(void)
-{
-	int error;
-
-	if ((error = ddi_soft_state_init(&audiohd_statep,
-	    sizeof (audiohd_state_t), 1)) != 0) {
-		return (error);
-	}
-
-	if ((error = mod_install(&audiohd_modlinkage)) != 0) {
-		ddi_soft_state_fini(&audiohd_statep);
-	}
-
-	return (error);
-
-}	/* _init() */
-
-
-int
-_fini(void)
-{
-	int error;
-
-	if ((error = mod_remove(&audiohd_modlinkage)) != 0) {
-		return (error);
-	}
-
-	ddi_soft_state_fini(&audiohd_statep);
-
-	return (0);
-
-}	/* _fini() */
-
-
-int
-_info(struct modinfo *modinfop)
+static void
+audiohd_set_chipset_info(audiohd_state_t *statep)
 {
-	int error;
+	uint32_t		devid;
+	const char		*name;
+	const char		*vers;
 
-	error = mod_info(&audiohd_modlinkage, modinfop);
+	devid = pci_config_get16(statep->hda_pci_handle, PCI_CONF_VENID);
+	devid <<= 16;
+	devid |= pci_config_get16(statep->hda_pci_handle, PCI_CONF_DEVID);
 
-	return (error);
-
-}	/* _info() */
-
-
-int
-audiohd_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd, void *arg, void **result)
-{
-	audiohd_state_t	*statep;
-	int		instance;
-	int		error;
+	name = AUDIOHD_DEV_CONFIG;
+	vers = AUDIOHD_DEV_VERSION;
 
-	ATRACE("in audiohd_getinfo()", dip);
-	error = DDI_FAILURE;
-	switch (cmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((statep = ddi_get_soft_state(audiohd_statep,
-		    instance)) != NULL) {
-			*result = statep->hda_dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
+	switch (devid) {
+	case 0x80862668:
+		name = "Intel HD Audio";
+		vers = "ICH6";
 		break;
-
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void *)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
+	case 0x808627d8:
+		name = "Intel HD Audio";
+		vers = "ICH7";
 		break;
-
-	default:
+	case 0x8086284b:
+		name = "Intel HD Audio";
+		vers = "ICH8";
+		break;
+	case 0x8086293e:
+		name = "Intel HD Audio";
+		vers = "ICH9";
+		break;
+	case 0x10de0371:
+		name = "NVIDIA HD Audio";
+		vers = "MCP55";
+		break;
+	case 0x10de03f0:
+		name = "NVIDIA HD Audio";
+		vers = "MCP61A";
+		break;
+	case 0x10de026c:
+		name = "NVIDIA HD Audio";
+		vers = "6151";
+		break;
+	case 0x10de03e4:
+		name = "NVIDIA HD Audio";
+		vers = "MCP61";
+		break;
+	case 0x10de044a:
+		name = "NVIDIA HD Audio";
+		vers = "MCP65";
+		break;
+	case 0x10de055c:
+		name = "NVIDIA HD Audio";
+		vers = "MCP67";
+		break;
+	case 0x1002437b:
+		name = "ATI HD Audio";
+		vers = "SB450";
+		break;
+	case 0x10024383:
+		name = "ATI HD Audio";
+		vers = "SB600";
+		break;
+	case 0x11063288:
+		name = "VIA HD Audio";
+		vers = "HDA";
 		break;
 	}
+	/* set device information */
+	audio_dev_set_description(statep->adev, name);
+	audio_dev_set_version(statep->adev, vers);
+}
 
-	return (error);
-
-}	/* audiohd_getinfo() */
-
-/*
- * audiohd_attach()
- *
- * Arguments:
- * 	dev_info_t  *dip    Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd    Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-int
+static int
 audiohd_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 {
-	audio_sup_reg_data_t		data;
 	audiohd_state_t		*statep;
 	int			instance;
 
@@ -393,8 +206,7 @@ audiohd_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 		break;
 
 	case DDI_RESUME:
-		ATRACE("audiohd_attach() DDI_RESUME", NULL);
-		statep = ddi_get_soft_state(audiohd_statep, instance);
+		statep = ddi_get_driver_private(dip);
 		ASSERT(statep != NULL);
 		return (audiohd_resume(statep));
 
@@ -404,93 +216,54 @@ audiohd_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 
 	/* High-level interrupt isn't supported by this driver */
 	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audiohd_attach() unsupported high level interrupt",
-		    audiohd_name, instance);
+		cmn_err(CE_WARN,
+		    "unsupported high level interrupt");
 		return (DDI_FAILURE);
 	}
 
-	if (ddi_soft_state_zalloc(audiohd_statep, instance) != DDI_SUCCESS) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audiohd_attach() softstate alloc failed",
-		    audiohd_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	if ((statep = ddi_get_soft_state(audiohd_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audiohd_attach() no softstate",
-		    audiohd_name, instance);
-		goto err_attach_exit1;
-	}
-
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-	if ((statep->hda_ahandle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audiohd_attach() audio_sup_register() failed",
-		    audiohd_name, instance);
-		goto err_attach_exit2;
-	}
-
-	/* save private state */
-	audio_sup_set_private(statep->hda_ahandle, statep);
+	/* allocate the soft state structure */
+	statep = kmem_zalloc(sizeof (*statep), KM_SLEEP);
+	ddi_set_driver_private(dip, statep);
 
 	/* interrupt cookie and initialize mutex */
 	if (audiohd_init_state(statep, dip) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_attach() audiohd_init_state failed");
+		cmn_err(CE_NOTE,
+		    "audiohd_init_state failed");
 		goto err_attach_exit3;
 	}
 
 	/* Set PCI command register to enable bus master and memeory I/O */
 	if (audiohd_init_pci(statep, &hda_dev_accattr) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_attach() couldn't init pci regs");
+		audio_dev_warn(statep->adev,
+		    "couldn't init pci regs");
 		goto err_attach_exit4;
 	}
 
+	audiohd_set_chipset_info(statep);
+
 	if (audiohd_init_controller(statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_attach() couldn't init controller");
+		audio_dev_warn(statep->adev,
+		    "couldn't init controller");
 		goto err_attach_exit5;
 	}
 
 	if (audiohd_create_codec(statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_attach() couldn't create codec");
+		audio_dev_warn(statep->adev,
+		    "couldn't create codec");
 		goto err_attach_exit6;
 	}
 
 	audiohd_build_path(statep);
 
-	if (audiohd_init_ports(statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_attach() couldn't init ports");
-		goto err_attach_exit7;
-	}
-
-	/*
-	 * This is a workaround. ALC880 doesn't support 8k sample rate,
-	 * but solaris requires hardware to be set to 8K by default. This
-	 * checking is performed in am_attach(). To overcome this flaw,
-	 * we have to set default sample to 48K before calling am_attach().
-	 * After we fix bug 6363625 in near future, we will change this
-	 * and set sample rate to 8K by default.
-	 */
-	statep->hda_info_defaults.play.sample_rate = audiohd_comp_srs[0];
-	statep->hda_info_defaults.record.sample_rate = audiohd_comp_srs[0];
-
-	if (am_attach(statep->hda_ahandle, cmd, &statep->hda_ad_info) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!attach() am_attach() failed");
+	audiohd_get_channels(statep);
+	if (audiohd_allocate_port(statep) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "allocate port failure");
 		goto err_attach_exit7;
 	}
-
+	audiohd_init_path(statep);
 	/* set up kernel statistics */
-	if ((statep->hda_ksp = kstat_create(AUDIOHD_NAME, instance,
-	    AUDIOHD_NAME, "controller", KSTAT_TYPE_INTR, 1,
+	if ((statep->hda_ksp = kstat_create(DRVNAME, instance,
+	    DRVNAME, "controller", KSTAT_TYPE_INTR, 1,
 	    KSTAT_FLAG_PERSISTENT)) != NULL) {
 		kstat_install(statep->hda_ksp);
 	}
@@ -502,23 +275,38 @@ audiohd_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 	if (ddi_add_intr(dip, 0, &statep->hda_intr_cookie,
 	    (ddi_idevice_cookie_t *)NULL, audiohd_intr, (caddr_t)statep) !=
 	    DDI_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!attach() bad interrupt specification ");
+		audio_dev_warn(statep->adev,
+		    "bad interrupt specification ");
 		goto err_attach_exit8;
 	}
+
+	/*
+	 * Register audio controls.
+	 */
+	if (audiohd_add_controls(statep) == DDI_FAILURE) {
+		audio_dev_warn(statep->adev,
+		    "unable to allocate controls");
+		goto err_attach_exit9;
+	}
+	if (audio_dev_register(statep->adev) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev,
+		    "unable to register with framework");
+		goto err_attach_exit9;
+	}
 	ddi_report_dev(dip);
 
 	/* enable interrupt */
 	AUDIOHD_REG_SET32(AUDIOHD_REG_INTCTL,
 	    AUDIOHD_INTCTL_BIT_GIE |
-	    AUDIOHD_INTCTL_BIT_CIE |
 	    AUDIOHD_INTCTL_BIT_SIE);
 	return (DDI_SUCCESS);
+err_attach_exit9:
+	audiohd_del_controls(statep);
 
 err_attach_exit8:
 	if (statep->hda_ksp)
 		kstat_delete(statep->hda_ksp);
-	(void) am_detach(statep->hda_ahandle, DDI_DETACH);
+	audiohd_free_port(statep);
 
 err_attach_exit7:
 	audiohd_destroy_codec(statep);
@@ -531,40 +319,23 @@ err_attach_exit5:
 
 err_attach_exit4:
 	mutex_destroy(&statep->hda_mutex);
-	cv_destroy(&statep->hda_cv);
 
 err_attach_exit3:
-	(void) audio_sup_unregister(statep->hda_ahandle);
+	(void) audio_dev_unregister(statep->adev);
 
 err_attach_exit2:
 err_attach_exit1:
-	ddi_soft_state_free(audiohd_statep, instance);
 
 	return (DDI_FAILURE);
+}
 
-}	/* audiohd_attach() */
-
-/*
- * audiohd_detach()
- * Arguments:
- *	dev_info_t		*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t	cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was detached
- *	DDI_FAILURE		The driver couldn't be detached
- */
-int
+static int
 audiohd_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
 {
 	audiohd_state_t		*statep;
-	int			instance;
 
-	instance = ddi_get_instance(dip);
-
-	if ((statep = ddi_get_soft_state(audiohd_statep, instance)) == NULL) {
-		return (DDI_FAILURE);
-	}
+	statep = ddi_get_driver_private(dip);
+	ASSERT(statep != NULL);
 
 	switch (cmd) {
 	case DDI_DETACH:
@@ -576,6 +347,8 @@ audiohd_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
 	default:
 		return (DDI_FAILURE);
 	}
+	if (audio_dev_unregister(statep->adev) != DDI_SUCCESS)
+		return (DDI_FAILURE);
 
 	mutex_enter(&statep->hda_mutex);
 	audiohd_stop_dma(statep);
@@ -584,1143 +357,1488 @@ audiohd_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
 	ddi_remove_intr(dip, 0, statep->hda_intr_cookie);
 	if (statep->hda_ksp)
 		kstat_delete(statep->hda_ksp);
-	(void) am_detach(statep->hda_ahandle, DDI_DETACH);
+	audiohd_free_port(statep);
 	audiohd_destroy_codec(statep);
+	audiohd_del_controls(statep);
 	audiohd_fini_controller(statep);
 	audiohd_fini_pci(statep);
 	mutex_destroy(&statep->hda_mutex);
-	cv_destroy(&statep->hda_cv);
-	(void) audio_sup_unregister(statep->hda_ahandle);
-	ddi_soft_state_free(audiohd_statep, instance);
-
+	if (statep->adev)
+		audio_dev_free(statep->adev);
+	kmem_free(statep, sizeof (*statep));
 	return (DDI_SUCCESS);
+}
+
+static struct dev_ops audiohd_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	NULL,			/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	audiohd_attach,		/* attach */
+	audiohd_detach,		/* detach */
+	nodev,			/* reset */
+	NULL,			/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+	audiohd_quiesce,	/* quiesce */
+};
 
-}	/* audiohd_detach() */
+static struct modldrv audiohd_modldrv = {
+	&mod_driverops,			/* drv_modops */
+	"AudioHD",			/* linkinfo */
+	&audiohd_dev_ops,		/* dev_ops */
+};
+
+static struct modlinkage modlinkage = {
+	MODREV_1,
+	{ &audiohd_modldrv, NULL }
+};
+
+int
+_init(void)
+{
+	int	rv;
+
+	audio_init_ops(&audiohd_dev_ops, DRVNAME);
+	if ((rv = mod_install(&modlinkage)) != 0) {
+		audio_fini_ops(&audiohd_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_fini(void)
+{
+	int	rv;
+
+	if ((rv = mod_remove(&modlinkage)) == 0) {
+		audio_fini_ops(&audiohd_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&modlinkage, modinfop));
+}
 
 /*
- * quiesce(9E) entry point.
- *
- * This function is called when the system is single-threaded at high
- * PIL with preemption disabled. Therefore, this function must not be
- * blocked.
- *
- * This function returns DDI_SUCCESS on success, or DDI_FAILURE on failure.
- * DDI_FAILURE indicates an error condition and should almost never happen.
+ * Audio routines
  */
+
 static int
-audiohd_quiesce(dev_info_t *dip)
+audiohd_engine_format(void *arg)
 {
-	audiohd_state_t		*statep;
-	int			instance;
+	_NOTE(ARGUNUSED(arg));
 
-	instance = ddi_get_instance(dip);
+	return (AUDIO_FORMAT_S16_LE);
+}
 
-	if ((statep = ddi_get_soft_state(audiohd_statep, instance)) == NULL) {
-		return (DDI_SUCCESS);
-	}
+static int
+audiohd_engine_channels(void *arg)
+{
+	audiohd_port_t *port = arg;
 
-	audiohd_stop_dma(statep);
-	audiohd_disable_intr(statep);
+	return (port->nchan);
+}
 
-	return (DDI_SUCCESS);
+static int
+audiohd_engine_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
 }
+
 /*
- * audiohd_change_widget_power_state(audiohd_state_t *statep, int off)
- * Description:
- * 	This routine is used to change the widget power betwen D0 and D2.
- * 	D0 is fully on; D2 allows the lowest possible power consuming state
- * 	from which it can return to the fully on state: D0.
+ * get the max channels the hardware supported
  */
 static void
-audiohd_change_widget_power_state(audiohd_state_t *statep, int off)
+audiohd_get_channels(audiohd_state_t *statep)
+{
+	int		i;
+	uint8_t		maxp, assoc;
+
+	maxp = 2;
+	for (i = 0; i < AUDIOHD_MAX_ASSOC; i++) {
+		if (maxp < statep->chann[i]) {
+			maxp = statep->chann[i];
+			assoc = i;
+		}
+	}
+	statep->pchan = maxp;
+	statep->assoc = assoc;
+	/* for record, support stereo so far */
+	statep->rchan = 2;
+}
+static void
+audiohd_init_play_path(audiohd_path_t *path)
 {
+	int				i;
+	uint32_t			ctrl;
+	uint8_t				ctrl8;
+	uint8_t				nchann;
+	audiohd_widget_t		*widget;
+	audiohd_pin_t			*pin;
+	wid_t				wid;
+	audiohd_pin_color_t		color;
+
+	audiohd_state_t		*statep = path->statep;
+	hda_codec_t		*codec = path->codec;
+
+	/* enable SPDIF output */
+	for (i = 0; i < path->pin_nums; i++) {
+		wid = path->pin_wid[i];
+		widget = codec->widget[wid];
+		pin = (audiohd_pin_t *)widget->priv;
+		if (pin->device == DTYPE_SPDIF_OUT) {
+			ctrl = audioha_codec_verb_get(
+			    statep,
+			    codec->index,
+			    path->adda_wid,
+			    AUDIOHDC_VERB_GET_SPDIF_CTL,
+			    0);
+			ctrl |= AUDIOHD_SPDIF_ON;
+			ctrl8 = ctrl &
+			    AUDIOHD_SPDIF_MASK;
+			(void) audioha_codec_verb_get(
+			    statep,
+			    codec->index,
+			    path->adda_wid,
+			    AUDIOHDC_VERB_SET_SPDIF_LCL,
+			    ctrl8);
+		}
+	}
+	wid = path->pin_wid[0];
+	widget = codec->widget[wid];
+	pin = (audiohd_pin_t *)widget->priv;
+
+	/* two channels supported */
+	if (pin->device == DTYPE_SPEAKER ||
+	    pin->assoc != statep->assoc) {
+		(void) audioha_codec_verb_get(
+		    statep,
+		    codec->index,
+		    path->adda_wid,
+		    AUDIOHDC_VERB_SET_STREAM_CHANN,
+		    statep->port[PORT_DAC]->index <<
+		    AUDIOHD_PLAY_TAG_OFF);
+		(void) audioha_codec_4bit_verb_get(
+		    statep,
+		    codec->index,
+		    path->adda_wid,
+		    AUDIOHDC_VERB_SET_CONV_FMT,
+		    AUDIOHD_FMT_PCM << 4 |
+		    statep->pchan - 1);
+	/* multichannel supported */
+	} else {
+		color = (pin->config >> AUDIOHD_PIN_CLR_OFF) &
+		    AUDIOHD_PIN_CLR_MASK;
+		switch (color) {
+		case AUDIOHD_PIN_BLACK:
+			nchann = statep->pchan - 2;
+			break;
+		case AUDIOHD_PIN_ORANGE:
+			nchann = 2;
+			break;
+		case AUDIOHD_PIN_GREY:
+			nchann = 4;
+			break;
+		case AUDIOHD_PIN_GREEN:
+			nchann = 0;
+			break;
+		default:
+			nchann = 0;
+			break;
+		}
+		(void) audioha_codec_verb_get(statep,
+		    codec->index,
+		    path->adda_wid,
+		    AUDIOHDC_VERB_SET_STREAM_CHANN,
+		    statep->port[PORT_DAC]->index <<
+		    AUDIOHD_PLAY_TAG_OFF |
+		    nchann);
+		(void) audioha_codec_4bit_verb_get(
+		    statep,
+		    codec->index,
+		    path->adda_wid,
+		    AUDIOHDC_VERB_SET_CONV_FMT,
+		    AUDIOHD_FMT_PCM << 4 |
+		    statep->pchan - 1);
+	}
+}
+static void
+audiohd_init_record_path(audiohd_path_t *path)
+{
+	audiohd_state_t		*statep = path->statep;
+	hda_codec_t		*codec = path->codec;
 	int			i;
 	wid_t			wid;
-	hda_codec_t		*codec;
+	audiohd_pin_t		*pin;
 	audiohd_widget_t	*widget;
 
-	/* Change power to D2 */
-	if (off) {
-		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-			codec = statep->codec[i];
-			if (!codec)
-				continue;
-			for (wid = codec->first_wid; wid <= codec->last_wid;
-			    wid++) {
-				widget = codec->widget[wid];
-				if (widget->widget_cap &
-				    AUDIOHD_WIDCAP_PWRCTRL) {
-					(void) audioha_codec_verb_get(statep,
-					    codec->index, wid,
-					    AUDIOHDC_VERB_SET_POWER_STATE,
-					    AUDIOHD_PW_D2);
-				}
-			}
+	for (i = 0; i < path->pin_nums; i++) {
+		wid = path->pin_wid[i];
+		widget = codec->widget[wid];
+		pin = (audiohd_pin_t *)widget->priv;
+	/*
+	 * Since there is no SPDIF input device available for test,
+	 * we will use this code in the future to support SPDIF input
+	 */
+#if 0
+		if (pin->device == DTYPE_SPDIF_IN) {
+			ctrl = audioha_codec_verb_get(
+			    statep,
+			    codec->index,
+			    path->adda_wid,
+			    AUDIOHDC_VERB_GET_SPDIF_CTL,
+			    0);
+			ctrl |= AUDIOHD_SPDIF_ON;
+			ctrl8 = ctrl &
+			    AUDIOHD_SPDIF_MASK;
+			(void) audioha_codec_verb_get(
+			    statep,
+			    codec->index,
+			    path->adda_wid,
+			    AUDIOHDC_VERB_SET_SPDIF_LCL,
+			    ctrl8);
+			statep->inmask |= (1U << DTYPE_SPDIF_IN);
 		}
-	/* Change power to D0 */
-	} else {
-		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-			codec = statep->codec[i];
-			if (!codec)
-				continue;
-			for (wid = codec->first_wid; wid <= codec->last_wid;
-			    wid++) {
-				widget = codec->widget[wid];
-				if (widget->widget_cap &
-				    AUDIOHD_WIDCAP_PWRCTRL) {
-					(void) audioha_codec_verb_get(statep,
-					    codec->index, wid,
-					    AUDIOHDC_VERB_SET_POWER_STATE,
-					    AUDIOHD_PW_D0);
-				}
-			}
+#endif
+		if (pin->device == DTYPE_MIC_IN) {
+			if (((pin->config >>
+			    AUDIOHD_PIN_CONTP_OFF) &
+			    AUDIOHD_PIN_CONTP_MASK) ==
+			    AUDIOHD_PIN_CON_FIXED)
+				statep->port[PORT_ADC]->index = path->tag;
+		}
+		if ((pin->device == DTYPE_LINE_IN) ||
+		    (pin->device == DTYPE_CD) ||
+		    (pin->device == DTYPE_MIC_IN)) {
+			statep->inmask |= (1U << pin->device);
 		}
 	}
+	(void) audioha_codec_verb_get(statep,
+	    codec->index,
+	    path->adda_wid,
+	    AUDIOHDC_VERB_SET_STREAM_CHANN,
+	    path->tag <<
+	    AUDIOHD_REC_TAG_OFF);
+	(void) audioha_codec_4bit_verb_get(statep,
+	    codec->index,
+	    path->adda_wid,
+	    AUDIOHDC_VERB_SET_CONV_FMT,
+	    AUDIOHD_FMT_PCM << 4 | statep->rchan - 1);
+
 }
-/*
- * audiohd_restore_path()
- * Description:
- * 	This routine is used to restore the path on the codec.
- */
 static void
-audiohd_restore_path(audiohd_state_t *statep)
+audiohd_init_path(audiohd_state_t *statep)
 {
-	int			i;
-	hda_codec_t		*codec;
+	int				i;
+	audiohd_path_t			*path;
 
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		codec = statep->codec[i];
-		if (!codec)
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path)
 			continue;
-		audiohd_finish_output_path(statep->codec[i]);
-		audiohd_finish_input_path(statep->codec[i]);
-		audiohd_finish_monitor_path(statep->codec[i]);
+		switch (path->path_type) {
+			case PLAY:
+				audiohd_init_play_path(path);
+				break;
+			case RECORD:
+				audiohd_init_record_path(path);
+				break;
+			default:
+				break;
+		}
 	}
+	statep->in_port = 0;
 }
 
-/*
- * restore_play_and_record()
- */
-static void
-audiohd_restore_play_and_record(audiohd_state_t *statep)
+static int
+audiohd_reset_port(audiohd_port_t *port)
 {
-	uint64_t	sbd_phys_addr;
-	uint_t		regbase;
+	uint16_t		regbase;
+	audiohd_state_t		*statep;
+	uint8_t			bTmp;
+	int			i;
+
+	regbase = port->regoff;
+	statep = port->statep;
+
+	bTmp = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_CTL);
+	/* stop stream */
+	bTmp &= ~AUDIOHD_REG_RIRBSIZE;
+	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
+
+	/* wait 40us for stream to stop as HD spec */
+	drv_usecwait(40);
+
+	/* reset stream */
+	bTmp |= AUDIOHDR_SD_CTL_SRST;
+	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
+
+	for (i = 0; i < AUDIOHD_RETRY_TIMES; i++) {
+		/* Empirical testing time, which works well */
+		drv_usecwait(50);
+		bTmp = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_CTL);
+		bTmp &= AUDIOHDR_SD_CTL_SRST;
+		if (bTmp)
+			break;
+	}
+
+	if (!bTmp) {
+		audio_dev_warn(statep->adev, "Failed to reset stream %d",
+		    port->index);
+		return (AUDIO_FAILURE);
+	}
+
+	/* Empirical testing time, which works well */
+	drv_usecwait(300);
+
+	/* exit reset stream */
+	bTmp &= ~AUDIOHDR_SD_CTL_SRST;
+	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
+
+	for (i = 0; i < AUDIOHD_RETRY_TIMES; i++) {
+		/* Empircal testing time */
+		drv_usecwait(50);
+		bTmp = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_CTL);
+		bTmp &= AUDIOHDR_SD_CTL_SRST;
+		if (!bTmp)
+			break;
+	}
+
+	if (bTmp) {
+		audio_dev_warn(statep->adev,
+		    "Failed to exit reset state for"
+		    " stream %d, bTmp=0x%02x", port->index, bTmp);
+		return (AUDIO_FAILURE);
+	}
 
-	regbase = statep->hda_play_regbase;
-	sbd_phys_addr = statep->hda_dma_play_bd.ad_paddr;
 	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPL,
-	    (uint32_t)sbd_phys_addr);
+	    (uint32_t)port->bdl_paddr);
 	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPU,
-	    (uint32_t)(sbd_phys_addr >> 32));
+	    (uint32_t)(port->bdl_paddr >> 32));
 	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_LVI,
 	    AUDIOHD_BDLE_NUMS - 1);
 	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_CBL,
-	    statep->hda_pbuf_size * AUDIOHD_BDLE_NUMS);
+	    port->samp_size * AUDIOHD_BDLE_NUMS);
 
 	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_FORMAT,
-	    statep->hda_play_format);
+	    port->format << 4 | port->nchan - 1);
 
 	/* clear status */
 	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS,
 	    AUDIOHDR_SD_STS_BCIS | AUDIOHDR_SD_STS_FIFOE |
 	    AUDIOHDR_SD_STS_DESE);
 
-	/* set playback stream tag */
+	/* set stream tag */
 	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL +
 	    AUDIOHD_PLAY_CTL_OFF,
-	    (statep->hda_play_stag) << AUDIOHD_PLAY_TAG_OFF);
-
-	if (statep->hda_flags & AUDIOHD_PLAY_STARTED) {
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-		    AUDIOHDR_SD_CTL_SRUN);
-	}
+	    (port->index) << AUDIOHD_PLAY_TAG_OFF);
 
-	regbase = statep->hda_record_regbase;
-	sbd_phys_addr = statep->hda_dma_record_bd.ad_paddr;
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPL,
-	    (uint32_t)sbd_phys_addr);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPU,
-	    (uint32_t)(sbd_phys_addr >> 32));
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_FORMAT,
-	    statep->hda_record_format);
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_LVI,
-	    AUDIOHD_BDLE_NUMS - 1);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_CBL,
-	    statep->hda_rbuf_size * AUDIOHD_BDLE_NUMS);
+	return (AUDIO_SUCCESS);
+}
+static int
+audiohd_engine_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	audiohd_port_t	*port = arg;
+	audiohd_state_t	*statep = port->statep;
 
-	/* clear status */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS,
-	    AUDIOHDR_SD_STS_INTRS);
+	_NOTE(ARGUNUSED(flag));
 
-	/* set stream tag to 1 */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL +
-	    AUDIOHD_REC_CTL_OFF,
-	    statep->hda_record_stag << AUDIOHD_REC_TAG_OFF);
+	mutex_enter(&statep->hda_mutex);
+	(void) audiohd_reset_port(port);
+	mutex_exit(&statep->hda_mutex);
 
-	if (statep->hda_flags & AUDIOHD_RECORD_STARTED) {
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-		    AUDIOHDR_SD_CTL_SRUN);
-	}
+	port->started = B_FALSE;
+	port->count = 0;
+	port->curpos = 0;
+	*fragfrp = port->fragfr;
+	*nfragsp = AUDIOHD_BDLE_NUMS;
+	*bufp = port->samp_kaddr;
 
+	return (0);
 }
-/*
- * audiohd_reset_pins_ur_cap()
- * Description:
- * 	Enable the unsolicited response of the pins which have the unsolicited
- * 	response capability
- */
+
 static void
-audiohd_reset_pins_ur_cap(audiohd_state_t *statep)
+audiohd_start_port(audiohd_port_t *port)
 {
-	hda_codec_t		*codec;
-	audiohd_pin_t		*pin;
-	audiohd_widget_t	*widget;
-	uint32_t		urctrl;
-	int			i;
+	audiohd_state_t	*statep = port->statep;
 
-	for (i = 0; i <= AUDIOHD_CODEC_MAX; i++) {
-		codec = statep->codec[i];
-		if (!codec)
-			continue;
-		pin = codec->first_pin;
-		while (pin) {
-			/* enable the unsolicited response of the pin */
-			widget = codec->widget[pin->wid];
-			if ((widget->widget_cap &
-			    (AUDIOHD_URCAP_MASK) &&
-			    (pin->cap & AUDIOHD_DTCCAP_MASK)) &&
-			    ((pin->device == DTYPE_LINEOUT) ||
-			    (pin->device == DTYPE_SPDIF_OUT) ||
-			    (pin->device == DTYPE_HP_OUT) ||
-			    (pin->device == DTYPE_MIC_IN))) {
-				urctrl = (uint8_t)(1 <<
-				    (AUDIOHD_UR_ENABLE_OFF - 1));
-				urctrl |= (pin->wid & AUDIOHD_UR_TAG_MASK);
-				(void) audioha_codec_verb_get(statep,
-				    codec->index,
-				    pin->wid,
-				    AUDIOHDC_VERB_SET_URCTRL, urctrl);
-			}
-			pin = pin->next;
-		}
+	ASSERT(mutex_owned(&statep->hda_mutex));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
 	}
+
+	/* Enable interrupt and start DMA */
+	AUDIOHD_REG_SET8(port->regoff + AUDIOHD_SDREG_OFFSET_CTL,
+	    AUDIOHDR_SD_CTL_INTS | AUDIOHDR_SD_CTL_SRUN);
 }
+
 static void
-audiohd_restore_codec_gpio(audiohd_state_t *statep)
+audiohd_stop_port(audiohd_port_t *port)
 {
-	int		i;
-	wid_t		wid;
-	hda_codec_t	*codec;
-
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		codec = statep->codec[i];
-		if (codec == NULL)
-			continue;
-		wid = codec->wid_afg;
-
-		/*
-		 * GPIO controls which are laptop specific workarounds and
-		 * might be changed. Some laptops use GPIO, so we need to
-		 * enable and set the GPIO correctly.
-		 */
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_MASK, AUDIOHDC_GPIO_ENABLE);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_DIREC, AUDIOHDC_GPIO_DIRECT);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_STCK, AUDIOHDC_GPIO_DATA_CTRL);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_DATA, AUDIOHDC_GPIO_STCK_CTRL);
-
-		/* power-up audio function group */
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_POWER_STATE, 0);
+	audiohd_state_t	*statep = port->statep;
 
+	ASSERT(mutex_owned(&statep->hda_mutex));
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
 	}
+	AUDIOHD_REG_SET8(port->regoff + AUDIOHD_SDREG_OFFSET_CTL, 0);
 }
-/*
- * audiohd_resume()
- */
+
 static int
-audiohd_resume(audiohd_state_t *statep)
+audiohd_engine_start(void *arg)
 {
-	uint_t		regbase;
-	uint8_t		rirbsts;
+	audiohd_port_t		*port = arg;
+	audiohd_state_t		*statep = port->statep;
 
 	mutex_enter(&statep->hda_mutex);
-	statep->suspended = B_FALSE;
-	/* Restore the hda state */
-	if (audiohd_reinit_hda(statep) == AUDIO_FAILURE) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_resume() hda reinit failed");
-		mutex_exit(&statep->hda_mutex);
-		return (DDI_SUCCESS);
+	if (!port->started) {
+		audiohd_start_port(port);
+		port->started = B_TRUE;
+		port->triggered = B_TRUE;
 	}
-	/* reset to enable the capability of unsolicited response for pin */
-	audiohd_reset_pins_ur_cap(statep);
-	/* Enable interrupt */
-	AUDIOHD_REG_SET32(AUDIOHD_REG_INTCTL,
-	    AUDIOHD_INTCTL_BIT_GIE |
-	    AUDIOHD_INTCTL_BIT_SIE);
-	/* clear the unsolicited response interrupt */
-	rirbsts = AUDIOHD_REG_GET8(AUDIOHD_REG_RIRBSTS);
-	AUDIOHD_REG_SET8(AUDIOHD_REG_RIRBSTS, rirbsts);
 	mutex_exit(&statep->hda_mutex);
+	return (0);
+}
 
-	/* Resume playing and recording */
-	if (audio_sup_restore_state(statep->hda_ahandle,
-	    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!audiohd_resume() audioh restore failed");
-		audiohd_disable_intr(statep);
-		audiohd_stop_dma(statep);
-	}
-	audiohd_restore_play_and_record(statep);
-	/* Enable interrupt and start DMA for play */
-	if (statep->hda_flags & AUDIOHD_PLAY_STARTED) {
-		regbase = statep->hda_play_regbase;
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-		    AUDIOHDR_SD_CTL_INTS | AUDIOHDR_SD_CTL_SRUN);
-	}
+static void
+audiohd_engine_stop(void *arg)
+{
+	audiohd_port_t		*port = arg;
+	audiohd_state_t		*statep = port->statep;
 
-	/* Enable interrupt and start DMA for record */
-	if (statep->hda_flags & AUDIOHD_RECORD_STARTED) {
-		regbase = statep->hda_record_regbase;
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-		    AUDIOHDR_SD_CTL_INTS | AUDIOHDR_SD_CTL_SRUN);
-	}
-	/* set widget power to D0 */
-	audiohd_change_widget_power_state(statep, AUDIOHD_PW_ON);
 	mutex_enter(&statep->hda_mutex);
-	cv_broadcast(&statep->hda_cv); /* wake up entry points */
+	if (port->started) {
+		audiohd_stop_port(port);
+	}
+	port->started = B_FALSE;
 	mutex_exit(&statep->hda_mutex);
+}
 
-	return (DDI_SUCCESS);
-}	/* audiohd_resume */
+static void
+audiohd_update_port(audiohd_port_t *port)
+{
+	int			pos;
+	uint32_t		len;
+	audiohd_state_t		*statep = port->statep;
 
-/*
- * audiohd_suspend()
- */
-static int
-audiohd_suspend(audiohd_state_t *statep)
+	pos = AUDIOHD_REG_GET32(port->regoff + AUDIOHD_SDREG_OFFSET_LPIB);
+	pos &= AUDIOHD_POS_MASK;
+	if (pos > port->curpos)
+		len = (pos - port->curpos) & AUDIOHD_POS_MASK;
+	else {
+		len = pos + port->samp_size * AUDIOHD_BDLE_NUMS - port->curpos;
+		len &= AUDIOHD_POS_MASK;
+	}
+	port->curpos += len;
+	if (port->curpos >= port->samp_size * AUDIOHD_BDLE_NUMS)
+		port->curpos -= port->samp_size * AUDIOHD_BDLE_NUMS;
+
+	port->len = len;
+	port->count += len / (port->nchan * 2);
+
+
+}
+
+static uint64_t
+audiohd_engine_count(void *arg)
 {
+	audiohd_port_t	*port = arg;
+	audiohd_state_t	*statep = port->statep;
+	uint64_t	val;
+
 	mutex_enter(&statep->hda_mutex);
-	statep->suspended = B_TRUE;
-	/* wait for current operations to complete */
-	while (statep->hda_busy_cnt != 0)
-		cv_wait(&statep->hda_cv, &statep->hda_mutex);
-
-	if (audio_sup_save_state(statep->hda_ahandle,
-	    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "audiohd_suspend() audio save failed");
-		statep->suspended = B_FALSE;
-		cv_broadcast(&statep->hda_cv);
-		mutex_exit(&statep->hda_mutex);
-		return (DDI_FAILURE);
-	}
+	audiohd_update_port(port);
+	val = port->count;
+	mutex_exit(&statep->hda_mutex);
+	return (val);
+}
 
-	/* set widget power to D2 */
-	audiohd_change_widget_power_state(statep, AUDIOHD_PW_OFF);
-	/* Disable h/w */
-	audiohd_disable_intr(statep);
-	audiohd_stop_dma(statep);
+static void
+audiohd_engine_close(void *arg)
+{
+	audiohd_port_t		*port = arg;
+	audiohd_state_t		*statep = port->statep;
+
+	mutex_enter(&statep->hda_mutex);
+	audiohd_stop_port(port);
+	port->started = B_FALSE;
+	port->triggered = B_FALSE;
 	mutex_exit(&statep->hda_mutex);
+}
 
-	return (DDI_SUCCESS);
-}	/* audiohd_suspend */
+static void
+audiohd_engine_sync(void *arg, unsigned nframes)
+{
+	audiohd_port_t *port = arg;
 
-/*
- * audiohd_disable_pin()
- */
-static int
-audiohd_disable_pin(audiohd_state_t *statep, int caddr, wid_t wid)
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(port->samp_dmah, port->curpos,
+	    port->len, port->sync_dir);
+
+}
+
+static size_t
+audiohd_engine_qlen(void *arg)
 {
-	AUDIOHD_DISABLE_PIN_OUT(statep, caddr, wid);
-	return (AUDIO_SUCCESS);
+	audiohd_port_t *port = arg;
+
+	return (port->fragfr);
 }
 
-/*
- * audiohd_enable_pin()
- */
+audio_engine_ops_t audiohd_engine_ops = {
+	AUDIO_ENGINE_VERSION,		/* version number */
+	audiohd_engine_open,
+	audiohd_engine_close,
+	audiohd_engine_start,
+	audiohd_engine_stop,
+	audiohd_engine_count,
+	audiohd_engine_format,
+	audiohd_engine_channels,
+	audiohd_engine_rate,
+	audiohd_engine_sync,
+	audiohd_engine_qlen,
+};
+
 static int
-audiohd_enable_pin(audiohd_state_t *statep, int caddr, wid_t wid)
+audiohd_get_value(void *arg, uint64_t *val)
 {
-	AUDIOHD_ENABLE_PIN_OUT(statep, caddr, wid);
-	return (AUDIO_SUCCESS);
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+
+	mutex_enter(&statep->hda_mutex);
+	*val = pc->val;
+	mutex_exit(&statep->hda_mutex);
+	return (0);
 }
-/*
- * audiohd_change_speaker_state()
- */
+
 static void
-audiohd_change_speaker_state(audiohd_state_t *statep, int on)
+audiohd_set_output_gain(audiohd_state_t *statep)
 {
-	hda_codec_t		*codec;
-	audiohd_ostream_t	*ostream;
-	audiohd_widget_t	*widget;
-	audiohd_pin_t		*pin;
-	int			i, j;
-
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		if (statep->codec[i]) {
-			codec = statep->codec[i];
-			break;
+	int		i, j;
+	uint64_t	val = statep->controls[CTL_VOLUME]->val;
+	audiohd_path_t	*path;
+	uint_t		tmp;
+	uint8_t		lgain = val;
+	uint8_t		rgain = val;
+
+	statep->hda_play_lgain = lgain;
+	statep->hda_play_rgain = rgain;
+
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->path_type == RECORD)
+			continue;
+		if (path->gain_wid) {
+			tmp = lgain * path->gain_bits / 100;
+			(void) audioha_codec_4bit_verb_get(statep,
+			    path->codec->index,
+			    path->gain_wid,
+			    AUDIOHDC_VERB_SET_AMP_MUTE,
+			    AUDIOHDC_AMP_SET_LEFT | path->gain_dir | tmp);
+			tmp = rgain * path->gain_bits / 100;
+			(void) audioha_codec_4bit_verb_get(statep,
+			    path->codec->index,
+			    path->gain_wid,
+			    AUDIOHDC_VERB_SET_AMP_MUTE,
+			    AUDIOHDC_AMP_SET_RIGHT | path->gain_dir | tmp);
+			continue;
 		}
-	}
-	if (!codec)
-		return;
-	if (on) {
-		ostream = codec->ostream;
-		while (ostream) {
-			for (j = 0; j < ostream->pin_nums; j++) {
-				widget = codec->widget[ostream->pin_wid[j]];
-				pin = (audiohd_pin_t *)widget->priv;
-				if (pin->device == DTYPE_SPEAKER) {
-					(void) audiohd_enable_pin(statep,
-					    codec->index, pin->wid);
-				}
+		for (j = 0; j < path->pin_nums; j++) {
+			audiohd_widget_t	*w;
+			audiohd_pin_t		*pin;
+			wid_t			wid;
+
+			wid = path->pin_wid[j];
+			w = path->codec->widget[wid];
+			pin = (audiohd_pin_t *)w->priv;
+			if (pin->gain_wid) {
+				tmp = lgain * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_LEFT | path->gain_dir |
+				    tmp);
+				tmp = rgain * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_RIGHT | path->gain_dir |
+				    tmp);
 			}
-			ostream = ostream->next_stream;
 		}
+	}
+}
 
-	} else {
-		ostream = codec->ostream;
-		while (ostream) {
-			for (j = 0; j < ostream->pin_nums; j++) {
-				widget = codec->widget[ostream->pin_wid[j]];
-				pin = (audiohd_pin_t *)widget->priv;
-				if (pin->device == DTYPE_SPEAKER) {
-					(void) audiohd_disable_pin(statep,
-					    codec->index, pin->wid);
-				}
+static void
+audiohd_set_pin_volume(audiohd_state_t *statep, audiohda_device_type_t type)
+{
+	int				i, j;
+	audiohd_path_t			*path;
+	audiohd_widget_t		*widget;
+	wid_t				wid;
+	audiohd_pin_t			*pin;
+	hda_codec_t			*codec;
+	uint8_t				l, r;
+	uint64_t			val;
+	uint_t				tmp;
+	audiohd_ctrl_t			*control;
+
+	switch (type) {
+		case DTYPE_LINEOUT:
+			control = statep->controls[CTL_FRONT];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+		case DTYPE_SPEAKER:
+			control = statep->controls[CTL_SPEAKER];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+		case DTYPE_HP_OUT:
+			control = statep->controls[CTL_HEADPHONE];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+		case DTYPE_CD:
+			control = statep->controls[CTL_CD];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+		case DTYPE_LINE_IN:
+			control = statep->controls[CTL_LINEIN];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+		case DTYPE_MIC_IN:
+			control = statep->controls[CTL_MIC];
+			if (control == NULL)
+				return;
+			val = control->val;
+			break;
+	}
+
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path)
+			continue;
+		codec = path->codec;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			widget = codec->widget[wid];
+			pin = (audiohd_pin_t *)widget->priv;
+			if ((pin->device == type) && pin->gain_wid) {
+				tmp = l * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_LEFT | path->gain_dir |
+				    tmp);
+				tmp = r * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_RIGHT | path->gain_dir |
+				    tmp);
 			}
-			ostream = ostream->next_stream;
 		}
 	}
 }
-/*
- * audiohd_select_mic()
- *
- * Description:
- *	This funciton is used for the recording path which has a selector
- *	as the sumwidget. We select the external MIC if it is plugged into the
- *	MIC jack, otherwise the internal integrated MIC is selected.
- */
+
+
 static void
-audiohd_select_mic(audiohd_state_t *statep, uint8_t index,
-uint8_t id, int select)
+audiohd_set_pin_volume_by_color(audiohd_state_t *statep,
+    audiohd_pin_color_t color)
 {
+	int			i, j;
+	audiohd_path_t		*path;
+	audiohd_widget_t	*widget;
+	wid_t			wid;
+	audiohd_pin_t		*pin;
 	hda_codec_t		*codec;
-	audiohd_istream_t	*istream;
-	audiohd_widget_t	*widget, *sumwgt;
+	uint8_t			l, r;
+	uint64_t		val;
+	uint_t			tmp;
+	audiohd_pin_color_t	clr;
+	audiohd_ctrl_t		*control;
+
+	switch (color) {
+		case AUDIOHD_PIN_BLACK:
+			control = statep->controls[CTL_REAR];
+			if (control == NULL)
+				return;
+			val = control->val;
+			l = (val & 0xff00) >> 8;
+			r = (val & 0xff);
+			break;
+		case AUDIOHD_PIN_ORANGE:
+			control = statep->controls[CTL_CENTER];
+			if (control == NULL)
+				return;
+			l = control->val;
+			control = statep->controls[CTL_LFE];
+			if (control == NULL)
+				return;
+			r = control->val;
+			break;
+		case AUDIOHD_PIN_GREY:
+			control = statep->controls[CTL_SURROUND];
+			if (control == NULL)
+				return;
+			val = control->val;
+			l = (val & 0xff00) >> 8;
+			r = (val & 0xff);
+			break;
+	}
+
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path)
+			continue;
+		codec = path->codec;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			widget = codec->widget[wid];
+			pin = (audiohd_pin_t *)widget->priv;
+			clr = (pin->config >> AUDIOHD_PIN_CLR_OFF) &
+			    AUDIOHD_PIN_CLR_MASK;
+			if ((clr == color) && pin->gain_wid) {
+				tmp = l * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_LEFT | path->gain_dir |
+				    tmp);
+				tmp = r * path->gain_bits / 100;
+				(void) audioha_codec_4bit_verb_get(statep,
+				    path->codec->index,
+				    path->gain_wid,
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_RIGHT | path->gain_dir |
+				    tmp);
+			}
+		}
+	}
+}
+
+static int
+audiohd_set_input_pin(audiohd_state_t *statep)
+{
+	uint64_t		val;
 	audiohd_pin_t		*pin;
-	int			i;
+	audiohd_path_t		*path;
+	audiohd_widget_t	*widget;
+	int			i, j;
 	wid_t			wid;
 
-	codec = statep->codec[index];
-	if (codec == NULL)
-		return;
-	istream = codec->istream;
-	sumwgt = codec->widget[istream->sum_wid];
-	while (istream && sumwgt && (sumwgt->type == WTYPE_AUDIO_SEL)) {
-		if (select) {
-			for (i = 0; i < istream->pin_nums; i++) {
-				wid = istream->pin_wid[i];
-				widget = codec->widget[wid];
-				if (widget == NULL)
-					return;
-				pin = (audiohd_pin_t *)widget->priv;
-				if (pin->device == DTYPE_MIC_IN &&
-				    pin->wid == id &&
-				    (((pin->config >> AUDIOHD_PIN_CONTP_OFF) &
-				    AUDIOHD_PIN_CONTP_MASK) ==
-				    AUDIOHD_PIN_CON_JACK)) {
-					(void) audioha_codec_verb_get(statep,
-					    index,
-					    istream->sum_wid,
-					    AUDIOHDC_VERB_SET_CONN_SEL,
-					    istream->sum_selconn[i]);
-					statep->hda_record_stag = istream->rtag;
-					return;
-				}
-			}
-		} else {
-			for (i = 0; i < istream->pin_nums; i++) {
-				wid = istream->pin_wid[i];
-				widget = codec->widget[wid];
-				if (widget == NULL)
-					return;
+	val = statep->controls[CTL_RECSRC]->val;
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->path_type != RECORD)
+			continue;
+		switch ((ddi_ffs(val & 0xffff)) - 1) {
+		case DTYPE_LINE_IN:
+		case DTYPE_MIC_IN:
+		case DTYPE_CD:
+			for (j = 0; j < path->pin_nums; j++) {
+				wid = path->pin_wid[j];
+				widget = path->codec->widget[wid];
 				pin = (audiohd_pin_t *)widget->priv;
-				if (pin->device == DTYPE_MIC_IN &&
-				    (((pin->config >> AUDIOHD_PIN_CONTP_OFF) &
-				    AUDIOHD_PIN_CONTP_MASK) ==
-				    AUDIOHD_PIN_CON_FIXED)) {
-					(void) audioha_codec_verb_get(statep,
-					    index,
-					    istream->sum_wid,
-					    AUDIOHDC_VERB_SET_CONN_SEL,
-					    istream->sum_selconn[i]);
-					statep->hda_record_stag = istream->rtag;
-					return;
+				if ((1U << pin->device) == val) {
+					AUDIOHD_ENABLE_PIN_IN(statep,
+					    path->codec->index,
+					    pin->wid);
+					statep->in_port = pin->device;
+				} else if (statep->in_port == pin->device) {
+					AUDIOHD_DISABLE_PIN_IN(statep,
+					    path->codec->index,
+					    pin->wid);
 				}
 			}
-		}
-		istream = istream->next_stream;
-		if (istream == NULL)
 			break;
-		sumwgt = codec->widget[istream->sum_wid];
+		default:
+			break;
+		}
+		break;
 	}
-	/*
-	 * If the input istream > 1, we should set the the record stream tag
-	 * repectively. All the input streams sharing one tag may make the
-	 * record sound distorted.
-	 */
-	if (codec->nistream > 1) {
-		istream = codec->istream;
-		while (istream) {
-			if (select) {
-				for (i = 0; i < istream->pin_nums; i++) {
-					wid = istream->pin_wid[i];
-					widget = codec->widget[wid];
-					if (widget == NULL)
-						return;
-					pin = (audiohd_pin_t *)widget->priv;
-					if (pin->device == DTYPE_MIC_IN &&
-					    pin->wid == id &&
-					    (((pin->config >>
-					    AUDIOHD_PIN_CONTP_OFF) &
-					    AUDIOHD_PIN_CONTP_MASK) ==
-					    AUDIOHD_PIN_CON_JACK)) {
-						statep->hda_record_stag =
-						    istream->rtag;
-						return;
-					}
-				}
-			} else {
-				for (i = 0; i < istream->pin_nums; i++) {
-					wid = istream->pin_wid[i];
-					widget = codec->widget[wid];
-					if (widget == NULL)
-						return;
-					pin = (audiohd_pin_t *)widget->priv;
-					if (pin->device == DTYPE_MIC_IN &&
-					    (((pin->config >>
-					    AUDIOHD_PIN_CONTP_OFF) &
-					    AUDIOHD_PIN_CONTP_MASK) ==
-					    AUDIOHD_PIN_CON_FIXED)) {
-						statep->hda_record_stag =
-						    istream->rtag;
-						return;
-					}
-				}
+	return (DDI_SUCCESS);
+}
+
+static void
+audiohd_set_pin_monitor_gain(hda_codec_t *codec, audiohd_state_t *statep,
+    uint_t caddr, audiohd_pin_t *pin, uint64_t gain)
+{
+	int 			i, k;
+	uint_t			ltmp, rtmp;
+	audiohd_widget_t	*widget;
+	uint8_t		l, r;
+
+	l = (gain & 0xff00) >> 8;
+	r = (gain & 0xff);
+
+	for (k = 0; k < pin->num; k++) {
+		ltmp = l * pin->mg_gain[k] / 100;
+		rtmp = r * pin->mg_gain[k] / 100;
+		widget = codec->widget[pin->mg_wid[k]];
+		if (pin->mg_dir[k] == AUDIOHDC_AMP_SET_OUTPUT) {
+			(void) audioha_codec_4bit_verb_get(
+			    statep,
+			    caddr,
+			    pin->mg_wid[k],
+			    AUDIOHDC_VERB_SET_AMP_MUTE,
+			    AUDIOHDC_AMP_SET_LEFT|
+			    pin->mg_dir[k] | ltmp);
+			(void) audioha_codec_4bit_verb_get(
+			    statep,
+			    caddr,
+			    pin->mg_wid[k],
+			    AUDIOHDC_VERB_SET_AMP_MUTE,
+			    AUDIOHDC_AMP_SET_RIGHT|
+			    pin->mg_dir[k] | rtmp);
+		} else if (pin->mg_dir[k] == AUDIOHDC_AMP_SET_INPUT) {
+			for (i = 0; i < widget->used; i++) {
+				(void) audioha_codec_4bit_verb_get(
+				    statep,
+				    caddr,
+				    pin->mg_wid[k],
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_RIGHT|
+				    widget->selmon[i]<<
+				    AUDIOHDC_AMP_SET_INDEX_OFFSET |
+				    pin->mg_dir[k] | rtmp);
+				(void) audioha_codec_4bit_verb_get(
+				    statep,
+				    caddr,
+				    pin->mg_wid[k],
+				    AUDIOHDC_VERB_SET_AMP_MUTE,
+				    AUDIOHDC_AMP_SET_LEFT|
+				    widget->selmon[i]<<
+				    AUDIOHDC_AMP_SET_INDEX_OFFSET |
+				    pin->mg_dir[k] | ltmp);
 			}
-			istream = istream->next_stream;
 		}
 	}
 }
-/*
- * audiohd_pin_sense()
- *
- * Description
- *
- * 	When the earphone is plugged into the jack associtated with the pin
- * 	complex, we disable the built in speaker. When the earphone is plugged
- * 	out of the jack, we enable the built in speaker.
- */
+
 static void
-audiohd_pin_sense(audiohd_state_t *statep, uint32_t resp, uint32_t respex)
+audiohd_set_monitor_gain(audiohd_state_t *statep)
 {
-	uint8_t			index;
-	uint8_t			id;
-	uint32_t		rs;
-	audiohd_widget_t	*widget;
+	int			i, j;
+	audiohd_path_t		*path;
+	uint_t			caddr;
+	audiohd_widget_t	*w;
+	wid_t			wid;
 	audiohd_pin_t		*pin;
-	hda_codec_t		*codec;
+	audiohd_ctrl_t		*ctrl;
+	uint64_t		val;
 
-	index = respex & AUDIOHD_RIRB_CODEC_MASK;
-	id = resp >> (AUDIOHD_RIRB_WID_OFF - 1);
-
-	codec = statep->codec[index];
-	if (codec == NULL)
-		return;
-	widget = codec->widget[id];
-	if (widget == NULL)
+	ctrl = statep->controls[CTL_MONGAIN];
+	if (ctrl == NULL)
 		return;
+	val = ctrl->val;
 
-	rs = audioha_codec_verb_get(statep, index, id,
-	    AUDIOHDC_VERB_GET_PIN_SENSE, 0);
-	if (rs >> (AUDIOHD_PIN_PRES_OFF - 1) & 1) {
-		/* A MIC is plugged in, we select the MIC as input */
-		if ((widget->type == WTYPE_PIN) &&
-		    (pin = (audiohd_pin_t *)widget->priv) &&
-		    (pin->device == DTYPE_MIC_IN)) {
-			audiohd_select_mic(statep, index, id, 1);
-			return;
-		}
-		/* output pin is plugged */
-		audiohd_change_speaker_state(statep, AUDIOHD_SP_OFF);
-	} else {
-		/*
-		 * A MIC is unplugged, we select the built in MIC
-		 * as input.
-		 */
-		if ((widget->type == WTYPE_PIN) &&
-		    (pin = (audiohd_pin_t *)widget->priv) &&
-		    (pin->device == DTYPE_MIC_IN)) {
-			audiohd_select_mic(statep, index, id, 0);
-			return;
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (path == NULL || path->path_type != PLAY)
+			continue;
+		caddr = path->codec->index;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			w = path->codec->widget[wid];
+			pin = (audiohd_pin_t *)w->priv;
+			audiohd_set_pin_monitor_gain(path->codec, statep,
+			    caddr, pin, val);
 		}
-		/* output pin is unplugged */
-		audiohd_change_speaker_state(statep, AUDIOHD_SP_ON);
 	}
 
 }
-/*
- * audiohd_intr()
- *
- * Description
- *
- *
- * Arguments:
- *	caddr_t     arg Pointer to the interrupting device's state
- *	            structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED    Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED  Interrupt not claimed, and thus ignored
- */
-static uint_t
-audiohd_intr(caddr_t arg)
+
+static void
+audiohd_restore_volume(audiohd_state_t *statep)
 {
-	audiohd_state_t	*statep = (audiohd_state_t *)arg;
-	uint32_t	status;
-	uint32_t	regbase;
-	uint32_t	resp, respex;
-	uint8_t		sdstatus, rirbsts;
-	int		i, ret;
+	audiohd_set_pin_volume(statep, DTYPE_LINEOUT);
+	audiohd_set_pin_volume(statep, DTYPE_SPEAKER);
+	audiohd_set_pin_volume(statep, DTYPE_HP_OUT);
 
-	mutex_enter(&statep->hda_mutex);
-	if (statep->suspended) {
-		mutex_exit(&statep->hda_mutex);
-		return (DDI_INTR_UNCLAIMED);
-	}
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_BLACK);
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_GREY);
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_ORANGE);
+}
+static void
+audiohd_configure_output(audiohd_state_t *statep)
+{
+	audiohd_set_output_gain(statep);
+}
+static void
+audiohd_configure_input(audiohd_state_t *statep)
+{
+	(void) audiohd_set_input_pin(statep);
+	audiohd_set_monitor_gain(statep);
+	audiohd_set_pin_volume(statep, DTYPE_LINE_IN);
+	audiohd_set_pin_volume(statep, DTYPE_CD);
+	audiohd_set_pin_volume(statep, DTYPE_MIC_IN);
+}
+static int
+audiohd_set_volume(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
 
-	status = AUDIOHD_REG_GET32(AUDIOHD_REG_INTSTS);
-	if (status == 0) {
-		mutex_exit(&statep->hda_mutex);
-		return (DDI_INTR_UNCLAIMED);
-	}
-	AUDIOHD_REG_SET32(AUDIOHD_REG_INTSTS, status);
+	val &= 0xff;
+	if (val > 100)
+		return (EINVAL);
 
-	/*
-	 * unsolicited response from pins, maybe something plugged in or out
-	 * of the jack.
-	 */
-	if (status & AUDIOHD_CIS_MASK) {
-		/* clear the unsolicited response interrupt */
-		rirbsts = AUDIOHD_REG_GET8(AUDIOHD_REG_RIRBSTS);
-		AUDIOHD_REG_SET8(AUDIOHD_REG_RIRBSTS, rirbsts);
-		/*
-		 * We have to wait and try several times to make sure the
-		 * unsolicited response is generated by our pins.
-		 * we need to make it work for audiohd spec 0.9, which is
-		 * just a draft version and requires more time to wait.
-		 */
-		for (i = 0; i < AUDIOHD_TEST_TIMES; i++) {
-			ret = audiohd_response_from_codec(statep, &resp,
-			    &respex);
-			if ((ret == AUDIO_SUCCESS) &&
-			    (respex & AUDIOHD_RIRB_UR_MASK)) {
-				/*
-				 * A pin may generate more than one ur rirb,
-				 * we only need handle one of them, and clear
-				 * the other ones
-				 */
-				statep->hda_rirb_rp =
-				    AUDIOHD_REG_GET16(AUDIOHD_REG_RIRBWP) &
-				    AUDIOHD_RIRB_WPMASK;
-				break;
-			}
-			drv_usecwait(30);
-		}
-		if ((ret == AUDIO_SUCCESS) &&
-		    (respex & AUDIOHD_RIRB_UR_MASK))
-			audiohd_pin_sense(statep, resp, respex);
-	}
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_configure_output(statep);
+	mutex_exit(&statep->hda_mutex);
 
-	/* stream intr */
-	for (i = 0; i < statep->hda_streams_nums; i++) {
-		if ((status & (1<<i)) == 0)
-			continue;
+	return (0);
+}
 
-		regbase = AUDIOHD_REG_SD_BASE + AUDIOHD_REG_SD_LEN * i;
-		sdstatus = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_STS);
+static int
+audiohd_set_recsrc(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t *statep = pc->statep;
 
-		/* clear intrs */
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS, sdstatus);
+	if (val & ~(statep->inmask))
+		return (EINVAL);
 
-		if ((sdstatus & AUDIOHDR_SD_STS_DESE) != 0) {
-			/*
-			 * fatal error. Hardware will clear run bit. So,
-			 * we need to shutdown the stream
-			 */
-			audio_sup_log(statep->hda_ahandle, CE_WARN,
-			    "!audiohd_intr() fatal error, shutdown "
-			    "stream %d", i);
-
-			if (i == statep->hda_input_streams) {
-				/* playback */
-				mutex_exit(&statep->hda_mutex);
-				am_play_shutdown(statep->hda_ahandle, NULL);
-				mutex_enter(&statep->hda_mutex);
-				statep->hda_flags &= ~(AUDIOHD_PLAY_EMPTY |
-				    AUDIOHD_PLAY_STARTED);
-				continue;
-			} else if (i == 0) { /* recording */
-				statep->hda_flags &= ~(AUDIOHD_RECORD_STARTED);
-				continue;
-			}
-		}
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_configure_input(statep);
+	mutex_exit(&statep->hda_mutex);
+	return (0);
+}
 
-		/* Buffer Complete Intr */
-		if (sdstatus & AUDIOHDR_SD_STS_BCIS)
-			if (i < statep->hda_input_streams) {
-				if (statep->hda_flags & AUDIOHD_RECORD_STARTED)
-					audiohd_get_rbuf(statep);
-			} else if (statep->hda_flags & AUDIOHD_PLAY_STARTED)
-				audiohd_refill_pbuf(statep);
-	}
+static int
+audiohd_set_rear(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-	/* update the kernel interrupt statistics */
-	if (statep->hda_ksp) {
-		((kstat_intr_t *)
-		    (statep->hda_ksp->ks_data))->intrs[KSTAT_INTR_HARD]++;
-	}
+	if (val & ~0xffff)
+		return (EINVAL);
 
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
+
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_BLACK);
 	mutex_exit(&statep->hda_mutex);
-	return (DDI_INTR_CLAIMED);
 
-}	/* audiohd_intr() */
+	return (0);
+}
 
-/*
- * audiohd_ad_start_play()
- *
- * Description:
- *	This routine starts the first output stream of hardware
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs
- *
- * Returns:
- *	AUDIO_SUCCESS   Playing started/restarted
- *	AUDIO_FAILURE   Play not started/restarted, no audio to play
- */
 static int
-audiohd_ad_start_play(audiohdl_t ahandle, int stream)
+audiohd_set_center(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	uint64_t	sbd_phys_addr;
-	uint8_t		cTmp;
-	uint_t	regbase;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+
+	val &= 0xff;
 
-	ATRACE_32("audiohd_ad_start_play() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	if (val > 100)
+		return (EINVAL);
 
-	audiohd_set_busy(statep);
 	mutex_enter(&statep->hda_mutex);
-	if (statep->hda_flags & AUDIOHD_PLAY_STARTED) {
-		mutex_exit(&statep->hda_mutex);
-		audiohd_set_idle(statep);
-		return (AUDIO_SUCCESS);
-	}
+	pc->val = val;
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_ORANGE);
+	mutex_exit(&statep->hda_mutex);
 
-	regbase = statep->hda_play_regbase;
-	if (statep->hda_flags & AUDIOHD_PLAY_PAUSED) {
-		cTmp = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_CTL);
-		statep->hda_flags |= AUDIOHD_PLAY_STARTED;
-		statep->hda_flags &= ~AUDIOHD_PLAY_PAUSED;
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-		    cTmp | AUDIOHDR_SD_CTL_SRUN);
-		mutex_exit(&statep->hda_mutex);
-		audiohd_set_idle(statep);
-		return (AUDIO_SUCCESS);
-	}
+	return (0);
+}
 
-	if (audiohd_reset_stream(statep, statep->hda_input_streams)
-	    != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!start_play() failed to reset play stream");
-		mutex_exit(&statep->hda_mutex);
-		audiohd_set_idle(statep);
-		return (AUDIO_FAILURE);
-	}
+static int
+audiohd_set_surround(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-	statep->hda_flags |= AUDIOHD_PLAY_STARTED;
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	if (audiohd_fill_pbuf(statep) != AUDIO_SUCCESS) {
-		mutex_exit(&statep->hda_mutex);
-		am_play_shutdown(statep->hda_ahandle, NULL);
-		audiohd_set_idle(statep);
-		return (AUDIO_FAILURE);
-	}
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	sbd_phys_addr = statep->hda_dma_play_bd.ad_paddr;
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPL,
-	    (uint32_t)sbd_phys_addr);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPU,
-	    (uint32_t)(sbd_phys_addr >> 32));
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_LVI,
-	    AUDIOHD_BDLE_NUMS - 1);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_CBL,
-	    statep->hda_pbuf_size * AUDIOHD_BDLE_NUMS);
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_GREY);
+	mutex_exit(&statep->hda_mutex);
 
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_FORMAT,
-	    statep->hda_play_format);
+	return (0);
+}
 
-	/* clear status */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS,
-	    AUDIOHDR_SD_STS_BCIS | AUDIOHDR_SD_STS_FIFOE |
-	    AUDIOHDR_SD_STS_DESE);
+static int
+audiohd_set_lfe(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
 
-	/* set playback stream tag */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL +
-	    AUDIOHD_PLAY_CTL_OFF,
-	    (statep->hda_play_stag) << AUDIOHD_PLAY_TAG_OFF);
+	val &= 0xff;
 
-	/* Enable interrupt and start DMA */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-	    AUDIOHDR_SD_CTL_INTS | AUDIOHDR_SD_CTL_SRUN);
+	if (val > 100)
+		return (EINVAL);
 
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume_by_color(statep, AUDIOHD_PIN_ORANGE);
 	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
-	return (AUDIO_SUCCESS);
 
-}	/* audiohd_ad_start_play() */
-
-
-/*
- * audiohd_ad_set_config()
- */
+	return (0);
+}
 static int
-audiohd_ad_set_config(audiohdl_t ahandle, int stream, int command,
-    int dir, int arg1, int arg2)
+audiohd_set_speaker(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	int 	rc = AUDIO_SUCCESS;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-	ATRACE_32("audiohd_ad_set_config() stream", stream);
-	ATRACE_32("audiohd_ad_set_config() command", command);
-	ATRACE_32("audiohd_ad_set_config() dir", dir);
-	ATRACE_32("audiohd_ad_set_config() arg1", arg1);
-	ATRACE_32("audiohd_ad_set_config() arg2", arg2);
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	audiohd_set_busy(statep);
 	mutex_enter(&statep->hda_mutex);
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, of balance on gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	arg1 --> gain
-		 * 	arg2 --> channel #, 0 == left, 1 == right
-		 */
-		rc = audiohd_set_gain(statep, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of in ports, as well as which ports
-		 * are modifiable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 * 	arg1 --> port bit pattern
-		 * 	arg2 --> not used
-		 */
-		rc = audiohd_set_port(statep, dir, arg1);
-		break;
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_SPEAKER);
+	mutex_exit(&statep->hda_mutex);
 
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	dir ---> N/A
-		 *	arg1 --> gain
-		 * 	arg2 --> not used
-		 */
-		rc = audiohd_set_monitor_gain(statep, arg1);
-		break;
+	return (0);
+}
+static int
+audiohd_set_front(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == mute, 0 == enable
-		 * 	arg2 --> not used
-		 */
-		audiohd_mute_outputs(statep, arg1);
-		break;
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	case AM_MIC_BOOST:
-		break;
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 *
-		 *
-		 * AM_SET_GAIN_BAL
-		 * AM_SET_MONO_MIC
-		 * AM_BASS_BOOST
-		 * AM_MID_BOOST
-		 * AM_TREBLE_BOOST
-		 * AM_LOUDNESS
-		 */
-		rc = AUDIO_FAILURE;
-		ATRACE_32("audiohd_ad_set_config() unsupported command",
-		    command);
-		break;
-	}
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_LINEOUT);
 	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
-
-	return (rc);
 
-}	/* audiohd_ad_set_config */
-
-/*
- * audiohd_ad_set_format()
- *
- * Description
- *	currently, only 48k sample rate, 16-bit precision,
- *	2-channel format is supported.
- */
-/*ARGSUSED*/
+	return (0);
+}
 static int
-audiohd_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-    int sample_rate, int channels, int precision, int encoding)
+audiohd_set_headphone(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	audiohd_ostream_t	*ostream;
-	audiohd_istream_t	*istream;
-	hda_codec_t		*codec;
-	uint_t		caddr;
-	int			i;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
+
+	if (val & ~0xffff)
+		return (EINVAL);
+
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	/*
-	 * we will support other format later. For the time being,
-	 * AUDIOHD_FMT_PCMOUT is used for 48k-16bit-2channel.
-	 */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
 	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_HP_OUT);
+	mutex_exit(&statep->hda_mutex);
 
-	if (dir == AUDIO_PLAY) {
-		statep->hda_psample_rate = sample_rate;
-		statep->hda_pchannels = channels;
-		statep->hda_pprecision = precision;
-		statep->hda_play_format = AUDIOHD_FMT_PCMOUT;
+	return (0);
+}
+static int
+audiohd_set_linein(void *arg, uint64_t val)
+{
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-			if (statep->codec[i] == NULL)
-				continue;
-			codec = statep->codec[i];
-			caddr = codec->index;
-			ostream = codec->ostream;
-			while (ostream) {
-				if (ostream->in_use == 0)
-					break;
-				(void) audioha_codec_4bit_verb_get(statep,
-				    caddr, ostream->dac_wid,
-				    AUDIOHDC_VERB_SET_CONV_FMT,
-				    AUDIOHD_FMT_PCMOUT);
-				ostream = ostream->next_stream;
-			}
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		statep->hda_csample_rate = sample_rate;
-		statep->hda_cchannels = channels;
-		statep->hda_cprecision = precision;
-		statep->hda_record_format = AUDIOHD_FMT_PCMIN;
-		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-			if (statep->codec[i] == NULL)
-				continue;
-			codec = statep->codec[i];
-			caddr = codec->index;
-			istream = codec->istream;
-			while (istream) {
-				if (istream->in_use == 0)
-					break;
-				(void) audioha_codec_4bit_verb_get(statep,
-				    caddr, istream->adc_wid,
-				    AUDIOHDC_VERB_SET_CONV_FMT,
-				    AUDIOHD_FMT_PCMOUT);
-				istream = istream->next_stream;
-			}
-		}
-	}
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	mutex_exit(&statep->hda_mutex);
-	return (AUDIO_SUCCESS);
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-}	/* audiohd_ad_set_format() */
+	mutex_enter(&statep->hda_mutex);
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_LINE_IN);
+	mutex_exit(&statep->hda_mutex);
 
+	return (0);
+}
 
-/*
- * audiohd_ad_pause_play()
- */
-/*ARGSUSED*/
-static void
-audiohd_ad_pause_play(audiohdl_t ahandle, int stream)
+static int
+audiohd_set_mic(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	uint32_t	regbase;
-	uint8_t		cTmp;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
+
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	audiohd_set_busy(statep);
 	mutex_enter(&statep->hda_mutex);
-	regbase = statep->hda_play_regbase;
-	cTmp = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_CTL);
-	cTmp &= ~AUDIOHDR_SD_CTL_SRUN;
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, cTmp);
-	statep->hda_flags &= ~AUDIOHD_PLAY_STARTED;
-	statep->hda_flags |= AUDIOHD_PLAY_PAUSED;
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_MIC_IN);
 	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
 
-}	/* audiohd_ad_pause_play() */
+	return (0);
+}
 
-/*
- * audiohd_ad_stop_play()
- */
-/*ARGSUSED*/
-static void
-audiohd_ad_stop_play(audiohdl_t ahandle, int stream)
+static int
+audiohd_set_cd(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	uint32_t	regbase;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
 
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	if (val & ~0xffff)
+		return (EINVAL);
+
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	audiohd_set_busy(statep);
 	mutex_enter(&statep->hda_mutex);
-	regbase = statep->hda_play_regbase;
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, 0);
-	statep->hda_flags &=
-	    ~(AUDIOHD_PLAY_EMPTY | AUDIOHD_PLAY_STARTED);
+	pc->val = val;
+	audiohd_set_pin_volume(statep, DTYPE_CD);
 	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
 
-}	/* audiohd_ad_stop_play() */
+	return (0);
+}
 
-/*
- * audiohd_ad_start_record()
- */
-/*ARGSUSED*/
 static int
-audiohd_ad_start_record(audiohdl_t ahandle, int stream)
+audiohd_set_mongain(void *arg, uint64_t val)
 {
-	audiohd_state_t	*statep;
-	uint64_t	sbd_phys_addr;
-	uint_t		regbase;
+	audiohd_ctrl_t	*pc = arg;
+	audiohd_state_t	*statep = pc->statep;
+	uint8_t		l, r;
+
+	if (val & ~0xffff)
+		return (EINVAL);
 
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	l = (val & 0xff00) >> 8;
+	r = (val & 0xff);
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
 
-	audiohd_set_busy(statep);
 	mutex_enter(&statep->hda_mutex);
-	if (statep->hda_flags & AUDIOHD_RECORD_STARTED) {
-		mutex_exit(&statep->hda_mutex);
-		audiohd_set_idle(statep);
-		return (AUDIO_SUCCESS);
+	pc->val = val;
+	audiohd_configure_input(statep);
+	mutex_exit(&statep->hda_mutex);
+
+	return (0);
+}
+
+#define	PLAYCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_PLAY)
+#define	RECCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_REC)
+#define	MONCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_MONITOR)
+#define	PCMVOL	(PLAYCTL | AUDIO_CTRL_FLAG_PCMVOL)
+#define	MONVOL	(MONCTL | AUDIO_CTRL_FLAG_MONVOL)
+#define	MAINVOL	(PLAYCTL | AUDIO_CTRL_FLAG_MAINVOL)
+#define	RECVOL	(RECCTL | AUDIO_CTRL_FLAG_RECVOL)
+
+static audiohd_ctrl_t *
+audiohd_alloc_ctrl(audiohd_state_t *statep, uint32_t num, uint64_t val)
+{
+	audio_ctrl_desc_t	desc;
+	audio_ctrl_wr_t		fn;
+	audiohd_ctrl_t		*pc;
+
+	pc = kmem_zalloc(sizeof (*pc), KM_SLEEP);
+	pc->statep = statep;
+	pc->num = num;
+
+	bzero(&desc, sizeof (desc));
+
+	switch (num) {
+	case CTL_VOLUME:
+		desc.acd_name = AUDIO_CTRL_ID_VOLUME;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = PCMVOL;
+		fn = audiohd_set_volume;
+		break;
+
+	case CTL_FRONT:
+		desc.acd_name = AUDIO_CTRL_ID_FRONT;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_front;
+		break;
+
+	case CTL_SPEAKER:
+		desc.acd_name = AUDIO_CTRL_ID_SPEAKER;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_speaker;
+		break;
+
+	case CTL_HEADPHONE:
+		desc.acd_name = AUDIO_CTRL_ID_HEADPHONE;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_headphone;
+		break;
+
+	case CTL_REAR:
+		desc.acd_name = AUDIO_CTRL_ID_REAR;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_rear;
+		break;
+
+	case CTL_CENTER:
+		desc.acd_name = AUDIO_CTRL_ID_CENTER;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_center;
+		break;
+
+	case CTL_SURROUND:
+		desc.acd_name = AUDIO_CTRL_ID_SURROUND;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_surround;
+		break;
+
+	case CTL_LFE:
+		desc.acd_name = AUDIO_CTRL_ID_LFE;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiohd_set_lfe;
+		break;
+
+	case CTL_LINEIN:
+		desc.acd_name = AUDIO_CTRL_ID_LINEIN;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiohd_set_linein;
+		break;
+
+	case CTL_MIC:
+		desc.acd_name = AUDIO_CTRL_ID_MIC;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiohd_set_mic;
+		break;
+
+	case CTL_CD:
+		desc.acd_name = AUDIO_CTRL_ID_CD;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiohd_set_cd;
+		break;
+
+	case CTL_MONGAIN:
+		desc.acd_name = AUDIO_CTRL_ID_MONGAIN;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MONVOL;
+		fn = audiohd_set_mongain;
+		break;
+
+	case CTL_RECSRC:
+		desc.acd_name = AUDIO_CTRL_ID_RECSRC;
+		desc.acd_type = AUDIO_CTRL_TYPE_ENUM;
+		desc.acd_minvalue = statep->inmask;
+		desc.acd_maxvalue = statep->inmask;
+		desc.acd_flags = RECCTL;
+		for (int i = 0; audiohd_dtypes[i]; i++) {
+			desc.acd_enum[i] = audiohd_dtypes[i];
+		}
+		fn = audiohd_set_recsrc;
+		break;
 	}
 
-	if (audiohd_reset_stream(statep, 0) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!start_record() failed to reset record stream");
-		mutex_exit(&statep->hda_mutex);
-		audiohd_set_idle(statep);
-		return (AUDIO_FAILURE);
+	pc->val = val;
+	pc->ctrl = audio_dev_add_control(statep->adev, &desc,
+	    audiohd_get_value, fn, pc);
+
+	return (pc);
+}
+
+static void
+audiohd_free_ctrl(audiohd_ctrl_t *pc)
+{
+	if (pc == NULL)
+		return;
+	if (pc->ctrl)
+		audio_dev_del_control(pc->ctrl);
+	kmem_free(pc, sizeof (*pc));
+}
+
+static void
+audiohd_del_controls(audiohd_state_t *statep)
+{
+	int		i;
+	for (i = 0; i < CTRL_NUM; i++) {
+		if (statep->controls[i])
+			audiohd_free_ctrl(statep->controls[i]);
 	}
+}
 
-	audiohd_preset_rbuf(statep);
-	statep->hda_rbuf_pos = 0;
-	regbase = statep->hda_record_regbase;
-	sbd_phys_addr = statep->hda_dma_record_bd.ad_paddr;
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPL,
-	    (uint32_t)sbd_phys_addr);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_BDLPU,
-	    (uint32_t)(sbd_phys_addr >> 32));
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_FORMAT,
-	    statep->hda_record_format);
-	AUDIOHD_REG_SET16(regbase + AUDIOHD_SDREG_OFFSET_LVI,
-	    AUDIOHD_BDLE_NUMS - 1);
-	AUDIOHD_REG_SET32(regbase + AUDIOHD_SDREG_OFFSET_CBL,
-	    statep->hda_rbuf_size * AUDIOHD_BDLE_NUMS);
+static int
+audiohd_add_controls(audiohd_state_t *statep)
+{
+	int			i, j;
+	audiohd_path_t		*path;
+	wid_t			wid;
+	audiohd_pin_t		*pin;
+	audiohd_widget_t	*widget;
+	hda_codec_t		*codec;
+	audiohd_pin_color_t	clr;
 
-	/* clear status */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS,
-	    AUDIOHDR_SD_STS_INTRS);
+#define	ADD_CTRL(ID, VAL)	\
+	if (statep->controls[ID] == NULL) \
+		statep->controls[ID] = audiohd_alloc_ctrl(statep, ID, VAL);\
+	if (statep->controls[ID] == NULL) {				\
+		audio_dev_warn(statep->adev,				\
+		    "Unable to allocate %s control", #ID);		\
+		return (DDI_FAILURE);					\
+	}
 
-	/* set stream tag to 1 */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL +
-	    AUDIOHD_REC_CTL_OFF,
-	    statep->hda_record_stag << AUDIOHD_REC_TAG_OFF);
-	statep->hda_flags |= AUDIOHD_RECORD_STARTED;
+	ADD_CTRL(CTL_VOLUME, 0x4b);
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path)
+			continue;
+		codec = path->codec;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			widget = codec->widget[wid];
+			pin = (audiohd_pin_t *)widget->priv;
+			if (pin->device == DTYPE_LINEOUT) {
+				ADD_CTRL(CTL_FRONT, 0x4b4b);
+			} else if (pin->device == DTYPE_SPEAKER) {
+				ADD_CTRL(CTL_SPEAKER, 0x4b4b);
+			} else if (pin->device == DTYPE_HP_OUT) {
+				ADD_CTRL(CTL_HEADPHONE, 0x4b4b);
+			} else if (pin->device == DTYPE_LINE_IN) {
+				ADD_CTRL(CTL_LINEIN, 0x3232);
+			} else if (pin->device == DTYPE_MIC_IN) {
+				ADD_CTRL(CTL_MIC, 0x3232);
+			} else if (pin->device == DTYPE_CD) {
+				ADD_CTRL(CTL_CD, 0x3232);
+			}
+			clr = (pin->config >> AUDIOHD_PIN_CLR_OFF) &
+			    AUDIOHD_PIN_CLR_MASK;
+			if (clr == AUDIOHD_PIN_BLACK &&
+			    pin->device != DTYPE_HP_OUT &&
+			    pin->device != DTYPE_MIC_IN) {
+				ADD_CTRL(CTL_REAR, 0x4b4b);
+			} else if (clr == AUDIOHD_PIN_ORANGE) {
+				ADD_CTRL(CTL_CENTER, 0x4b);
+				ADD_CTRL(CTL_LFE, 0x4b);
+			} else if (clr == AUDIOHD_PIN_GREY) {
+				ADD_CTRL(CTL_SURROUND, 0x4b4b);
+			}
+		}
+	}
 
-	/* start DMA */
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL,
-	    AUDIOHDR_SD_CTL_INTS | AUDIOHDR_SD_CTL_SRUN);
+	if (!statep->monitor_unsupported) {
+		ADD_CTRL(CTL_MONGAIN, 0);
+	}
 
-	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
-	return (AUDIO_SUCCESS);
+	ADD_CTRL(CTL_RECSRC, (1U << DTYPE_MIC_IN));
 
-}	/* audiohd_ad_start_record() */
+	audiohd_configure_output(statep);
+	audiohd_configure_input(statep);
 
+	return (DDI_SUCCESS);
+}
 
 /*
- * audiohd_ad_stop_record()
+ * quiesce(9E) entry point.
+ *
+ * This function is called when the system is single-threaded at high
+ * PIL with preemption disabled. Therefore, this function must not be
+ * blocked.
+ *
+ * This function returns DDI_SUCCESS on success, or DDI_FAILURE on failure.
+ * DDI_FAILURE indicates an error condition and should almost never happen.
  */
-/*ARGSUSED*/
-static void
-audiohd_ad_stop_record(audiohdl_t ahandle, int stream)
+static int
+audiohd_quiesce(dev_info_t *dip)
 {
-	audiohd_state_t	*statep;
-	uint32_t	regbase;
+	audiohd_state_t		*statep;
 
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
+	statep = ddi_get_driver_private(dip);
 
-	audiohd_set_busy(statep);
-	mutex_enter(&statep->hda_mutex);
-	regbase = statep->hda_record_regbase;
-	AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, 0);
-	statep->hda_flags &= ~(AUDIOHD_RECORD_STARTED);
-	mutex_exit(&statep->hda_mutex);
-	audiohd_set_idle(statep);
-
-}	/* audiohd_ad_stop_play */
+	audiohd_stop_dma(statep);
+	audiohd_disable_intr(statep);
 
+	return (DDI_SUCCESS);
+}
 /*
  * audiohd_init_state()
  *
@@ -1733,158 +1851,36 @@ audiohd_ad_stop_record(audiohdl_t ahandle, int stream)
 static int
 audiohd_init_state(audiohd_state_t *statep, dev_info_t *dip)
 {
-	int	pints, rints, mode;
+	audio_dev_t			*adev;
 
 	statep->hda_dip = dip;
 
-	/* get the mode from the .conf file */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "mixer-mode", 1)) {
-		mode = AM_MIXER_MODE;
-	} else {
-		mode = AM_COMPAT_MODE;
-	}
-
-	pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "play-interrupts", AUDIOHD_INTS);
-	if (pints > AUDIOHD_MAX_INTS) {
-		audio_sup_log(statep->hda_ahandle, CE_NOTE, "init_state() "
-		    "play interrupt rate set too high, %d, resetting to %d",
-		    pints, AUDIOHD_INTS);
-		pints = AUDIOHD_INTS;
-	} else if (pints < AUDIOHD_MIN_INTS) {
-		audio_sup_log(statep->hda_ahandle, CE_NOTE, "init_state() "
-		    "play interrupt rate set too low, %d, resetting to %d",
-		    pints, AUDIOHD_INTS);
-		pints = AUDIOHD_INTS;
-	}
-	rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "record-interrupts", AUDIOHD_INTS);
-	if (rints > AUDIOHD_MAX_INTS) {
-		audio_sup_log(statep->hda_ahandle, CE_NOTE, "init_state() "
-		    "record interrupt rate set too high, %d, resetting to %d",
-		    rints, AUDIOHD_INTS);
-		rints = AUDIOHD_INTS;
-	} else if (rints < AUDIOHD_MIN_INTS) {
-		audio_sup_log(statep->hda_ahandle, CE_NOTE, "init_state() "
-		    "record interrupt rate set too low, %d, resetting to %d",
-		    rints, AUDIOHD_INTS);
-		rints = AUDIOHD_INTS;
+	if ((adev = audio_dev_alloc(dip, 0)) == NULL) {
+		cmn_err(CE_WARN, "unable to allocate audio dev");
+		return (AUDIO_FAILURE);
 	}
+	statep->adev = adev;
 
-	statep->hda_pint_freq = pints;
-	statep->hda_rint_freq = rints;
-	statep->hda_pbuf_size = (AUDIOHD_SAMPLER_MAX * AUDIOHD_MAX_CHANNELS *
-	    AUDIOHD_MAX_PRECISION / 8) / pints;
-	statep->hda_pbuf_size = (statep->hda_pbuf_size +
-	    AUDIOHD_BDLE_BUF_ALIGN - 1) & ~(AUDIOHD_BDLE_BUF_ALIGN - 1);
-	statep->hda_rbuf_size = (AUDIOHD_SAMPLER_MAX * AUDIOHD_MAX_CHANNELS *
-	    AUDIOHD_MAX_PRECISION / 8) / rints;
-	statep->hda_rbuf_size = (statep->hda_rbuf_size +
-	    AUDIOHD_BDLE_BUF_ALIGN - 1) & ~(AUDIOHD_BDLE_BUF_ALIGN - 1);
-
-	/* fill in the device default state */
-	statep->hda_info_defaults.play.sample_rate = AUDIOHD_DEFAULT_SR;
-	statep->hda_info_defaults.play.channels = AUDIOHD_DEFAULT_CH;
-	statep->hda_info_defaults.play.precision = AUDIOHD_DEFAULT_PREC;
-	statep->hda_info_defaults.play.encoding = AUDIOHD_DEFAULT_ENC;
-	statep->hda_info_defaults.play.gain = AUDIOHD_DEFAULT_PGAIN;
-	statep->hda_info_defaults.play.buffer_size = AUDIOHD_BSIZE;
-	statep->hda_info_defaults.play.balance = AUDIOHD_DEFAULT_BAL;
-
-	statep->hda_info_defaults.record.sample_rate = AUDIOHD_DEFAULT_SR;
-	statep->hda_info_defaults.record.channels = AUDIOHD_DEFAULT_CH;
-	statep->hda_info_defaults.record.precision = AUDIOHD_DEFAULT_PREC;
-	statep->hda_info_defaults.record.encoding = AUDIOHD_DEFAULT_ENC;
-	statep->hda_info_defaults.record.gain = AUDIOHD_DEFAULT_RGAIN;
-	statep->hda_info_defaults.record.buffer_size = AUDIOHD_BSIZE;
-	statep->hda_info_defaults.record.balance = AUDIOHD_DEFAULT_BAL;
-
-	statep->hda_info_defaults.monitor_gain = AUDIOHD_DEFAULT_MONITOR_GAIN;
-	statep->hda_info_defaults.output_muted = B_FALSE;
-	statep->hda_info_defaults.ref_cnt = B_FALSE;
-	statep->hda_info_defaults.hw_features =
-	    AUDIO_HWFEATURE_DUPLEX | AUDIO_HWFEATURE_PLAY |
-	    AUDIO_HWFEATURE_IN2OUT | AUDIO_HWFEATURE_RECORD;
-	statep->hda_info_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	statep->hda_psample_rate =
-	    statep->hda_info_defaults.play.sample_rate;
-	statep->hda_pchannels =
-	    statep->hda_info_defaults.play.channels;
-	statep->hda_pprecision =
-	    statep->hda_info_defaults.play.precision;
-	statep->hda_play_format = AUDIOHD_FMT_PCMOUT;
-
-	statep->hda_csample_rate =
-	    statep->hda_info_defaults.record.sample_rate;
-	statep->hda_cchannels =
-	    statep->hda_info_defaults.record.channels;
-	statep->hda_cprecision =
-	    statep->hda_info_defaults.record.precision;
-	statep->hda_record_format = AUDIOHD_FMT_PCMIN;
-
-
-	/*
-	 * fill in the ad_info structure
-	 */
-	statep->hda_ad_info.ad_mode = mode;
-	statep->hda_ad_info.ad_int_vers = AM_VERSION;
-	statep->hda_ad_info.ad_add_mode = NULL;
-	statep->hda_ad_info.ad_codec_type = AM_TRAD_CODEC;
-	statep->hda_ad_info.ad_defaults = &statep->hda_info_defaults;
-	statep->hda_ad_info.ad_play_comb = audiohd_combinations;
-	statep->hda_ad_info.ad_rec_comb = audiohd_combinations;
-	statep->hda_ad_info.ad_entry = &audiohd_entry;
-	statep->hda_ad_info.ad_dev_info = &statep->hda_dev_info;
-	statep->hda_ad_info.ad_diag_flags = AM_DIAG_INTERNAL_LOOP;
-	statep->hda_ad_info.ad_diff_flags =
-	    AM_DIFF_SR | AM_DIFF_CH | AM_DIFF_PREC | AM_DIFF_ENC;
-	statep->hda_ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	statep->hda_ad_info.ad_misc_flags = AM_MISC_RP_EXCL | AM_MISC_MONO_DUP;
-	statep->hda_ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	statep->hda_ad_info.ad_play.ad_mixer_srs = audiohd_mixer_sample_rates;
-	statep->hda_ad_info.ad_play.ad_compat_srs = audiohd_comp_sample_rates;
-	statep->hda_ad_info.ad_play.ad_conv = &am_src2;
-	statep->hda_ad_info.ad_play.ad_sr_info = NULL;
-	statep->hda_ad_info.ad_play.ad_chs = audiohd_channels;
-	statep->hda_ad_info.ad_play.ad_int_rate = pints;
-	statep->hda_ad_info.ad_play.ad_max_chs = AUDIOHD_MAX_OUT_CHANNELS;
-	statep->hda_ad_info.ad_play.ad_bsize = AUDIOHD_BSIZE;
-
-	/* record capabilities */
-	statep->hda_ad_info.ad_record.ad_mixer_srs = audiohd_mixer_sample_rates;
-	statep->hda_ad_info.ad_record.ad_compat_srs = audiohd_comp_sample_rates;
-	statep->hda_ad_info.ad_record.ad_conv = &am_src2;
-	statep->hda_ad_info.ad_record.ad_sr_info = NULL;
-	statep->hda_ad_info.ad_record.ad_chs = audiohd_channels;
-	statep->hda_ad_info.ad_record.ad_int_rate = rints;
-	statep->hda_ad_info.ad_record.ad_max_chs = AUDIOHD_MAX_IN_CHANNELS;
-	statep->hda_ad_info.ad_record.ad_bsize = AUDIOHD_BSIZE;
-
-	/* fill in device info strings */
-	(void) strcpy(statep->hda_dev_info.name, AUDIOHD_DEV_NAME);
-	(void) strcpy(statep->hda_dev_info.config, AUDIOHD_DEV_CONFIG);
-	(void) strcpy(statep->hda_dev_info.version, AUDIOHD_DEV_VERSION);
+	/* set device information */
+	audio_dev_set_description(adev, AUDIOHD_DEV_CONFIG);
+	audio_dev_set_version(adev, AUDIOHD_DEV_VERSION);
 
 	if (ddi_get_iblock_cookie(dip, (uint_t)0, &statep->hda_intr_cookie) !=
 	    DDI_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_state() cannot get iblock cookie");
-		return (AUDIO_FAILURE);
+		audio_dev_warn(statep->adev,
+		    "cannot get iblock cookie");
+		goto error;
 	}
-	statep->hda_busy_cnt = 0;
 	mutex_init(&statep->hda_mutex, NULL,
 	    MUTEX_DRIVER, statep->hda_intr_cookie);
-	cv_init(&statep->hda_cv, NULL, CV_DRIVER, NULL);
 
-	statep->hda_outputs_muted = B_FALSE;
 	statep->hda_rirb_rp = 0;
 
 	return (AUDIO_SUCCESS);
-
+error:
+	if (statep->adev != NULL)
+		audio_dev_free(statep->adev);
+	return (AUDIO_FAILURE);
 }	/* audiohd_init_state() */
 
 /*
@@ -1901,18 +1897,18 @@ audiohd_init_pci(audiohd_state_t *statep, ddi_device_acc_attr_t *acc_attr)
 	uint16_t	vid;
 	uint8_t		cTmp;
 	dev_info_t	*dip = statep->hda_dip;
-	audiohdl_t	ahandle = statep->hda_ahandle;
+	audio_dev_t	*ahandle = statep->adev;
 
 	if (pci_config_setup(dip, &statep->hda_pci_handle) == DDI_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() pci config mapping failed");
+		audio_dev_warn(ahandle,
+		    "pci config mapping failed");
 		goto err_init_pci_exit1;
 	}
 
 	if (ddi_regs_map_setup(dip, 1, &statep->hda_reg_base, 0,
 	    0, acc_attr, &statep->hda_reg_handle) != DDI_SUCCESS) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() memory I/O mapping failed");
+		audio_dev_warn(ahandle,
+		    "memory I/O mapping failed");
 		goto err_init_pci_exit2;
 	}
 
@@ -2068,8 +2064,8 @@ audiohd_reset_controller(audiohd_state_t *statep)
 	}
 
 	if ((gctl & AUDIOHDR_GCTL_CRST) != 0) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!reset_controller() failed to enter reset state");
+		audio_dev_warn(statep->adev,
+		    "failed to enter reset state");
 		return (AUDIO_FAILURE);
 	}
 
@@ -2088,17 +2084,14 @@ audiohd_reset_controller(audiohd_state_t *statep)
 	}
 
 	if ((gctl & AUDIOHDR_GCTL_CRST) == 0) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!reset_controller() failed to exit reset state");
+		audio_dev_warn(statep->adev,
+		    "failed to exit reset state");
 		return (AUDIO_FAILURE);
 	}
 
 	/* HD spec requires to wait 250us at least. we use 500us */
 	drv_usecwait(500);
 
-	ATRACE_32("in reset_controller: mask is",
-	    AUDIOHD_REG_GET16(AUDIOHD_REG_STATESTS));
-
 	/* enable unsolicited response */
 	AUDIOHD_REG_SET32(AUDIOHD_REG_GCTL,
 	    gctl |  AUDIOHDR_GCTL_URESPE);
@@ -2121,12 +2114,12 @@ audiohd_alloc_dma_mem(audiohd_state_t *statep, audiohd_dma_t *pdma,
 	ddi_dma_cookie_t	cookie;
 	uint_t			count;
 	dev_info_t		*dip = statep->hda_dip;
-	audiohdl_t		ahandle = statep->hda_ahandle;
+	audio_dev_t		*ahandle = statep->adev;
 
 	if (ddi_dma_alloc_handle(dip, dma_attr_p, DDI_DMA_SLEEP,
-	    (caddr_t)0, &pdma->ad_dmahdl) != DDI_SUCCESS) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_alloc_hanlde failed");
+	    NULL, &pdma->ad_dmahdl) != DDI_SUCCESS) {
+		audio_dev_warn(ahandle,
+		    "ddi_dma_alloc_handle failed");
 		goto error_alloc_dma_exit1;
 	}
 
@@ -2135,29 +2128,19 @@ audiohd_alloc_dma_mem(audiohd_state_t *statep, audiohd_dma_t *pdma,
 	    DDI_DMA_SLEEP, NULL,
 	    (caddr_t *)&pdma->ad_vaddr, &pdma->ad_real_sz,
 	    &pdma->ad_acchdl) != DDI_SUCCESS) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_mem_alloc failed");
+		audio_dev_warn(ahandle,
+		    "ddi_dma_mem_alloc failed");
 		goto error_alloc_dma_exit2;
 	}
 
 	if (ddi_dma_addr_bind_handle(pdma->ad_dmahdl, NULL,
 	    (caddr_t)pdma->ad_vaddr, pdma->ad_real_sz, dma_flags,
 	    DDI_DMA_SLEEP, NULL, &cookie, &count) != DDI_DMA_MAPPED) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_addr_bind_handle failed");
+		audio_dev_warn(ahandle,
+		    "ddi_dma_addr_bind_handle failed");
 		goto error_alloc_dma_exit3;
 	}
 
-	/*
-	 * there some bugs in the DDI framework and it is possible to
-	 * get multiple cookies
-	 */
-	if (count != 1) {
-		(void) ddi_dma_unbind_handle(pdma->ad_dmahdl);
-		audio_sup_log(ahandle, CE_WARN,
-		    "!map_regs() addr_bind_handle failed, cookies > 1");
-		goto error_alloc_dma_exit3;
-	}
 	pdma->ad_paddr = (uint64_t)(cookie.dmac_laddress);
 	pdma->ad_req_sz = memsize;
 
@@ -2242,7 +2225,9 @@ audiohd_reinit_hda(audiohd_state_t *statep)
 
 	audiohd_restore_codec_gpio(statep);
 	audiohd_restore_path(statep);
-	return (audiohd_init_ports(statep));
+	audiohd_init_path(statep);
+
+	return (AUDIO_SUCCESS);
 }	/* audiohd_reinit_hda */
 
 /*
@@ -2292,13 +2277,11 @@ audiohd_init_controller(audiohd_state_t *statep)
 	statep->hda_streams_nums = statep->hda_input_streams +
 	    statep->hda_output_streams;
 
-	statep->hda_record_stag = 1;
-	statep->hda_play_stag = statep->hda_input_streams + 1;
+
 	statep->hda_record_regbase = AUDIOHD_REG_SD_BASE;
 	statep->hda_play_regbase = AUDIOHD_REG_SD_BASE + AUDIOHD_REG_SD_LEN *
 	    statep->hda_input_streams;
 
-	ATRACE_16("GCAP = ", gcap);
 
 	/* stop all dma before starting to reset controller */
 	audiohd_stop_dma(statep);
@@ -2309,8 +2292,8 @@ audiohd_init_controller(audiohd_state_t *statep)
 	/* check codec */
 	statep->hda_codec_mask = AUDIOHD_REG_GET16(AUDIOHD_REG_STATESTS);
 	if (! statep->hda_codec_mask) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() no codec exists");
+		audio_dev_warn(statep->adev,
+		    "no codec exists");
 		goto err_init_ctlr_exit1;
 	}
 
@@ -2319,8 +2302,8 @@ audiohd_init_controller(audiohd_state_t *statep)
 	    AUDIOHD_CDBIO_CORB_LEN, &dma_attr,
 	    DDI_DMA_WRITE | DDI_DMA_STREAMING);
 	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for CORB");
+		audio_dev_warn(statep->adev,
+		    "failed to alloc DMA for CORB");
 		goto err_init_ctlr_exit1;
 	}
 
@@ -2329,50 +2312,11 @@ audiohd_init_controller(audiohd_state_t *statep)
 	    AUDIOHD_CDBIO_RIRB_LEN, &dma_attr,
 	    DDI_DMA_READ | DDI_DMA_STREAMING);
 	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for RIRB");
+		audio_dev_warn(statep->adev,
+		    "failed to alloc DMA for RIRB");
 		goto err_init_ctlr_exit2;
 	}
 
-	/* allocate DMA for data buffer of playback stream */
-	retval = audiohd_alloc_dma_mem(statep, &statep->hda_dma_play_buf,
-	    statep->hda_pbuf_size * AUDIOHD_BDLE_NUMS, &dma_attr,
-	    DDI_DMA_WRITE | DDI_DMA_STREAMING);
-	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for playback buf");
-		goto err_init_ctlr_exit3;
-	}
-
-	/* allocate DMA for data buffer of recording stream */
-	retval = audiohd_alloc_dma_mem(statep, &statep->hda_dma_record_buf,
-	    statep->hda_rbuf_size * AUDIOHD_BDLE_NUMS, &dma_attr,
-	    DDI_DMA_READ | DDI_DMA_STREAMING);
-	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for recording buf");
-		goto err_init_ctlr_exit4;
-	}
-
-	/* allocate DMA for buffer descriptor list of playback stream */
-	retval = audiohd_alloc_dma_mem(statep, &statep->hda_dma_play_bd,
-	    sizeof (sd_bdle_t) * AUDIOHD_BDLE_NUMS, &dma_attr,
-	    DDI_DMA_WRITE | DDI_DMA_STREAMING);
-	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for playback BDL");
-		goto err_init_ctlr_exit5;
-	}
-
-	/* allocate DMA for buffer descriptor list of recording stream */
-	retval = audiohd_alloc_dma_mem(statep, &statep->hda_dma_record_bd,
-	    sizeof (sd_bdle_t) * AUDIOHD_BDLE_NUMS, &dma_attr,
-	    DDI_DMA_WRITE | DDI_DMA_STREAMING);
-	if (retval != AUDIO_SUCCESS) {
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!init_controller() failed to alloc DMA for record BDL");
-		goto err_init_ctlr_exit6;
-	}
 
 	AUDIOHD_REG_SET32(AUDIOHD_REG_SYNC, 0); /* needn't sync stream */
 
@@ -2399,15 +2343,6 @@ audiohd_init_controller(audiohd_state_t *statep)
 
 	return (AUDIO_SUCCESS);
 
-err_init_ctlr_exit6:
-	audiohd_release_dma_mem(&(statep->hda_dma_play_bd));
-
-err_init_ctlr_exit5:
-	audiohd_release_dma_mem(&(statep->hda_dma_record_buf));
-
-err_init_ctlr_exit4:
-	audiohd_release_dma_mem(&(statep->hda_dma_play_buf));
-
 err_init_ctlr_exit3:
 	audiohd_release_dma_mem(&(statep->hda_dma_rirb));
 
@@ -2431,10 +2366,6 @@ audiohd_fini_controller(audiohd_state_t *statep)
 	audiohd_stop_dma(statep);
 	audiohd_release_dma_mem(&statep->hda_dma_rirb);
 	audiohd_release_dma_mem(&statep->hda_dma_corb);
-	audiohd_release_dma_mem(&statep->hda_dma_play_buf);
-	audiohd_release_dma_mem(&statep->hda_dma_record_buf);
-	audiohd_release_dma_mem(&statep->hda_dma_record_bd);
-	audiohd_release_dma_mem(&statep->hda_dma_play_bd);
 
 }	/* audiohd_fini_controller() */
 
@@ -2460,9 +2391,6 @@ audiohd_get_conns_from_entry(hda_codec_t *codec, audiohd_widget_t *widget,
 			if (widget->nconns == 0) {
 				if (input_wid < codec->first_wid ||
 				    (input_wid > codec->last_wid)) {
-					ATRACE_32(
-					    "collection list out of range",
-					    widget->wid_wid);
 					break;
 				}
 				widget->avail_conn[widget->nconns++] =
@@ -2474,9 +2402,6 @@ audiohd_get_conns_from_entry(hda_codec_t *codec, audiohd_widget_t *widget,
 					    AUDIOHD_MAX_CONN);
 					if (k < codec->first_wid ||
 					    (k > codec->last_wid)) {
-						ATRACE_32(
-						    "collection out of range",
-						    widget->wid_wid);
 						break;
 					} else {
 						num = widget->nconns;
@@ -2545,9 +2470,6 @@ audiohd_get_conns(hda_codec_t *codec, wid_t wid)
 		input_wid = entry & prop.mask_wid;
 		if ((input_wid < codec->first_wid) ||
 		    (input_wid > codec->last_wid)) {
-			ATRACE("node input out of range:", NULL);
-			ATRACE_32("node:", wid);
-			ATRACE_32("input:", input_wid);
 			return;
 		}
 		widget->avail_conn[0] = input_wid;
@@ -2602,10 +2524,7 @@ audiohd_get_pin_config(audiohd_widget_t *widget)
 	pin->config = config;
 	pin->num = 0;
 	pin->finish = 0;
-	/*
-	 * get the voltage reference state supported by the pin
-	 * from high level to low level
-	 */
+
 	vrefbits = (cap >> AUDIOHD_PIN_VREF_OFF) & AUDIOHD_PIN_VREF_MASK;
 	if (vrefbits & AUDIOHD_PIN_VREF_L1)
 		pin->vrefvalue = 0x5;
@@ -2761,10 +2680,8 @@ audiohd_create_widgets(hda_codec_t *codec)
 			audiohd_get_pin_config(widget);
 			break;
 		case WTYPE_BEEP:
-			ATRACE("Get a beep widget", NULL);
 			break;
 		default:
-			ATRACE("Unknown widget", NULL);
 			break;
 		}
 	}
@@ -2788,6 +2705,231 @@ audiohd_destroy_widgets(hda_codec_t *codec)
 
 }	/* audiohd_destroy_widgets() */
 
+static void
+audiohd_set_codec_info(hda_codec_t *codec)
+{
+	char buf[256];
+
+	switch (codec->vid) {
+	case 0x10ec0260:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC260");
+		break;
+	case 0x10ec0262:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC262");
+		break;
+	case 0x10ec0268:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC268");
+		break;
+	case 0x10ec0662:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC662");
+		break;
+	case 0x10ec861:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC861");
+		break;
+	case 0x10ec0862:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC862");
+		break;
+	case 0x10ec0880:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC880");
+		break;
+	case 0x10ec0882:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC882");
+		break;
+	case 0x10ec0883:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC883");
+		break;
+	case 0x10ec0885:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC885");
+		break;
+	case 0x10ec0888:
+		(void) snprintf(buf, sizeof (buf), "Realtek HD codec: ALC888");
+		break;
+	case 0x13f69880:
+		(void) snprintf(buf, sizeof (buf), "CMedia HD codec: CMI19880");
+		break;
+	case 0x434d4980:
+		(void) snprintf(buf, sizeof (buf), "CMedia HD codec: CMI19880");
+		break;
+	case 0x11d41981:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1981");
+		break;
+	case 0x11d41983:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1983");
+		break;
+	case 0x11d41984:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1984");
+		break;
+	case 0x11d41986:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1986A");
+		break;
+	case 0x11d41988:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1988A");
+		break;
+	case 0x11d4198b:
+		(void) snprintf(buf, sizeof (buf),
+		    "Analog Devices HD codec: AD1988B");
+		break;
+	case 0x83847690:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9200");
+		break;
+	case 0x838476a0:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9205");
+		break;
+	case 0x838476a1:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9205D");
+		break;
+	case 0x838476a2:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9204");
+		break;
+	case 0x838476a3:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9204D");
+		break;
+	case 0x83847880:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9220 A1");
+		break;
+	case 0x83847882:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9220 A2");
+		break;
+	case 0x83847680:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9221 A1");
+		break;
+	case 0x83847681:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9220 D");
+		break;
+	case 0x83847682:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9221");
+		break;
+	case 0x83847683:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9221D");
+		break;
+	case 0x83847610:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9230XN");
+		break;
+	case 0x83847611:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9230DN");
+		break;
+	case 0x83847612:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9230XT");
+		break;
+	case 0x83847613:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9230DT");
+		break;
+	case 0x83847614:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9229X");
+		break;
+	case 0x83847615:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9229D");
+		break;
+	case 0x83847616:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9228X");
+		break;
+	case 0x83847617:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9228D");
+		break;
+	case 0x83847618:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9227X");
+		break;
+	case 0x83847619:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9227D");
+		break;
+	case 0x838476a4:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9255");
+		break;
+	case 0x838476a5:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9255D");
+		break;
+	case 0x838476a6:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9254");
+		break;
+	case 0x838476a7:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9254D");
+		break;
+	case 0x83847620:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9274");
+		break;
+	case 0x83847621:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9274D");
+		break;
+	case 0x83847622:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9273X");
+		break;
+	case 0x83847623:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9273D");
+		break;
+	case 0x83847624:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9272X");
+		break;
+	case 0x83847625:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9272D");
+		break;
+	case 0x83847626:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9271X");
+		break;
+	case 0x83847627:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9271D");
+		break;
+	case 0x83847628:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9274X5NH");
+		break;
+	case 0x83847629:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9274D5NH");
+		break;
+	case 0x83847662:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9872AK");
+		break;
+	case 0x83847664:
+		(void) snprintf(buf, sizeof (buf),
+		    "Sigmatel HD codec: STAC9872K");
+		break;
+	default:
+		(void) snprintf(buf, sizeof (buf),
+		    "Unkown HD codec");
+		break;
+
+	}
+	audio_dev_add_info(codec->soft_statep->adev, buf);
+}
 /*
  * audiohd_create_codec()
  *
@@ -2851,19 +2993,6 @@ audiohd_create_codec(audiohd_state_t *statep)
 		}
 
 		ASSERT(codec->wid_afg == wid);
-		/*
-		 * GPIO controls which are laptop specific workarounds and
-		 * might be changed. Some laptops use GPIO, so we need to
-		 * enable and set the GPIO correctly.
-		 */
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_MASK, AUDIOHDC_GPIO_ENABLE);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_DIREC, AUDIOHDC_GPIO_DIRECT);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_STCK, AUDIOHDC_GPIO_DATA_CTRL);
-		(void) audioha_codec_verb_get(statep, i, wid,
-		    AUDIOHDC_VERB_SET_GPIO_DATA, AUDIOHDC_GPIO_STCK_CTRL);
 
 		/* power-up audio function group */
 		(void) audioha_codec_verb_get(statep, i, wid,
@@ -2891,13 +3020,31 @@ audiohd_create_codec(audiohd_state_t *statep)
 		/*
 		 * We output the codec information to syslog
 		 */
-		ATRACE_32("codec = ", codec->index);
-		ATRACE_32("vid = ", codec->vid);
-		ATRACE_32("rev = ", codec->revid);
-
 		statep->codec[i] = codec;
 		codec->soft_statep = statep;
+		audiohd_set_codec_info(codec);
 		(void) audiohd_create_widgets(codec);
+
+		/* work around for Sony VAIO laptop with specific codec */
+		if ((codec->vid == AUDIOHD_CODECID_SONY1) ||
+		    (codec->vid == AUDIOHD_CODECID_SONY2))
+			continue;
+		else
+			statep->gpio_direct = AUDIOHDC_GPIO_DIRECT;
+		/*
+		 * GPIO controls which are laptop specific workarounds and
+		 * might be changed. Some laptops use GPIO, so we need to
+		 * enable and set the GPIO correctly.
+		 */
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_MASK, AUDIOHDC_GPIO_ENABLE);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_DIREC, statep->gpio_direct);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_STCK, AUDIOHDC_GPIO_DATA_CTRL);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_DATA, AUDIOHDC_GPIO_STCK_CTRL);
+
 	}
 
 	return (AUDIO_SUCCESS);
@@ -2913,34 +3060,12 @@ audiohd_create_codec(audiohd_state_t *statep)
 static void
 audiohd_destroy_codec(audiohd_state_t *statep)
 {
-	audiohd_ostream_t    *ostream, *nostream;
-	audiohd_istream_t    *istream, *nistream;
+	int			i;
 	audiohd_pin_t		*pin, *npin;
 
-	for (int i = 0; i < AUDIOHD_CODEC_MAX; i++) {
+	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
 		if (statep->codec[i]) {
 			audiohd_destroy_widgets(statep->codec[i]);
-
-			/*
-			 * free output streams
-			 */
-			ostream = statep->codec[i]->ostream;
-			while (ostream) {
-				nostream = ostream;
-				ostream = ostream->next_stream;
-				kmem_free(nostream, sizeof (audiohd_ostream_t));
-			}
-
-			/*
-			 * free input streams
-			 */
-			istream = statep->codec[i]->istream;
-			while (istream) {
-				nistream = istream;
-				istream = istream->next_stream;
-				kmem_free(nistream, sizeof (audiohd_istream_t));
-			}
-
 			/*
 			 * free pins
 			 */
@@ -2991,6 +3116,7 @@ audiohd_find_dac(hda_codec_t *codec, wid_t wid,
 
 	if (widget == NULL)
 		return (uint32_t)(AUDIO_FAILURE);
+
 	/*
 	 * If exclusive is true, we try to find a path which doesn't
 	 * share any widget with other paths.
@@ -3011,7 +3137,6 @@ audiohd_find_dac(hda_codec_t *codec, wid_t wid,
 		widget->path_flags |= AUDIOHD_PATH_DAC;
 		widget->out_weight++;
 		wdac = widget->wid_wid;
-		ATRACE_32("(DAC:", widget->wid_wid);
 		break;
 
 	case WTYPE_AUDIO_MIX:
@@ -3028,8 +3153,6 @@ audiohd_find_dac(hda_codec_t *codec, wid_t wid,
 					widget->selconn = i;
 					wdac = retval;
 				}
-				ATRACE_32("widget:", widget->wid_wid);
-				ATRACE_32("type:", widget->type);
 				widget->path_flags |= AUDIOHD_PATH_DAC;
 				widget->out_weight++;
 
@@ -3062,10 +3185,13 @@ audiohd_do_build_output_path(hda_codec_t *codec, int mixer, int *mnum,
 {
 	audiohd_pin_t		*pin;
 	audiohd_widget_t	*widget, *wdac;
-	audiohd_ostream_t	*ostream;
+	audiohd_path_t	*path;
 	wid_t			wid;
+	audiohd_state_t	*statep;
 	int			i;
 
+	statep = codec->soft_statep;
+
 	for (pin = codec->first_pin; pin; pin = pin->next) {
 		if ((pin->cap & AUDIOHD_PIN_CAP_MASK) == 0)
 			continue;
@@ -3096,29 +3222,25 @@ audiohd_do_build_output_path(hda_codec_t *codec, int mixer, int *mnum,
 			 */
 			if (wid == (wid_t)AUDIO_FAILURE)
 				continue;
-			ostream = (audiohd_ostream_t *)
-			    kmem_zalloc(sizeof (audiohd_ostream_t),
+			if (pin->device != DTYPE_SPEAKER)
+				statep->chann[pin->assoc] += 2;
+			path = (audiohd_path_t *)
+			    kmem_zalloc(sizeof (audiohd_path_t),
 			    KM_SLEEP);
-			ostream->dac_wid = wid;
-			ostream->pin_wid[0] = widget->wid_wid;
-			ostream->pin_nums = 1;
+			path->adda_wid = wid;
+			path->pin_wid[0] = widget->wid_wid;
+			path->pin_nums = 1;
+			path->path_type = PLAY;
+			path->codec = codec;
+			path->statep = statep;
 			wdac = codec->widget[wid];
-			wdac->priv = ostream;
+			wdac->priv = path;
 			pin->adc_dac_wid = wid;
 			pin->finish = 1;
-			ATRACE_32("widget:", widget->wid_wid);
-			ATRACE_32("type:", widget->type);
 			widget->path_flags |= AUDIOHD_PATH_DAC;
 			widget->out_weight++;
 			widget->selconn = i;
-			if (codec->ostream == NULL)
-				codec->ostream = ostream;
-			else {
-				audiohd_ostream_t *p = codec->ostream;
-				while (p->next_stream)
-					p = p->next_stream;
-				p->next_stream = ostream;
-			}
+			statep->path[statep->pathnum++] = path;
 			break;
 		}
 	}
@@ -3138,14 +3260,17 @@ audiohd_do_build_output_path(hda_codec_t *codec, int mixer, int *mnum,
 static void
 audiohd_build_output_path(hda_codec_t *codec)
 {
-	audiohd_pin_t		*pin;
 	int 			mnum = 0;
+	uint8_t			mixer_allow = 1;
 
+	/* work around for hp mini 1000 laptop */
+	if (codec->vid == AUDIOHD_CODECID_HP)
+		mixer_allow = 0;
 	/* search an exclusive mixer widget path. This is preferred */
-	audiohd_do_build_output_path(codec, 1, &mnum, 1, 0);
+	audiohd_do_build_output_path(codec, mixer_allow, &mnum, 1, 0);
 
 	/* search a shared mixer widget path for the remained pins */
-	audiohd_do_build_output_path(codec, 1, &mnum, 0, 0);
+	audiohd_do_build_output_path(codec, mixer_allow, &mnum, 0, 0);
 
 	/* search an exclusive widget path without mixer for the remained pin */
 	audiohd_do_build_output_path(codec, 0, &mnum, 1, 0);
@@ -3153,22 +3278,6 @@ audiohd_build_output_path(hda_codec_t *codec)
 	/* search a shared widget path without mixer for the remained pin */
 	audiohd_do_build_output_path(codec, 0, &mnum, 0, 0);
 
-	for (pin = codec->first_pin; pin; pin = pin->next) {
-		if ((pin->cap & AUDIOHD_PIN_CAP_MASK) == 0) {
-			continue;
-		}
-		if ((pin->config & AUDIOHD_PIN_CONF_MASK) ==
-		    AUDIOHD_PIN_NO_CONN)
-			continue;
-		if ((pin->device != DTYPE_LINEOUT) &&
-		    (pin->device != DTYPE_SPEAKER) &&
-		    (pin->device != DTYPE_SPDIF_OUT) &&
-		    (pin->device != DTYPE_HP_OUT))
-			continue;
-		if (!pin->finish) {
-			ATRACE_32("pin has no output path:", pin->wid);
-		}
-	}
 }	/* audiohd_build_output_path */
 
 /*
@@ -3180,18 +3289,21 @@ audiohd_build_output_path(hda_codec_t *codec)
 static void
 audiohd_build_output_amp(hda_codec_t *codec)
 {
-	audiohd_ostream_t	*ostream;
+	audiohd_path_t		*path;
 	audiohd_widget_t	*w, *widget, *wpin, *wdac;
 	audiohd_pin_t		*pin;
 	wid_t		wid;
 	int		weight;
-	int		i;
+	int		i, j;
 	uint32_t	gain;
 
-	ostream = codec->ostream;
-	while (ostream) {
-		for (i = 0; i < ostream->pin_nums; i++) {
-			wid = ostream->pin_wid[i];
+	for (i = 0; i < codec->soft_statep->pathnum; i++) {
+		path = codec->soft_statep->path[i];
+		if (path == NULL || path->path_type == RECORD ||
+		    path->codec != codec)
+			continue;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
 			wpin = codec->widget[wid];
 			pin = (audiohd_pin_t *)wpin->priv;
 			weight = wpin->out_weight;
@@ -3262,16 +3374,16 @@ audiohd_build_output_amp(hda_codec_t *codec)
 		 * all output pins of this stream to be used for the
 		 * whole stream
 		 */
-		if (ostream->pin_nums == 1) {
-			ostream->mute_wid = pin->mute_wid;
-			ostream->mute_dir = pin->mute_dir;
-			ostream->gain_wid = pin->gain_wid;
-			ostream->gain_dir = pin->gain_dir;
-			ostream->gain_bits = pin->gain_bits;
+		if (path->pin_nums == 1) {
+			path->mute_wid = pin->mute_wid;
+			path->mute_dir = pin->mute_dir;
+			path->gain_wid = pin->gain_wid;
+			path->gain_dir = pin->gain_dir;
+			path->gain_bits = pin->gain_bits;
 		} else {
-			wdac = codec->widget[ostream->dac_wid];
+			wdac = codec->widget[path->adda_wid];
 			weight = wdac->out_weight;
-			wid = ostream->pin_wid[0];
+			wid = path->pin_wid[0];
 			w = codec->widget[wid];
 			while (w && w->out_weight != weight) {
 				wid = w->avail_conn[w->selconn];
@@ -3283,15 +3395,15 @@ audiohd_build_output_amp(hda_codec_t *codec)
 			while (widget) {
 				if (widget->outamp_cap &
 				    AUDIOHDC_AMP_CAP_MUTE_CAP) {
-					ostream->mute_wid = widget->wid_wid;
-					ostream->mute_dir =
+					path->mute_wid = widget->wid_wid;
+					path->mute_dir =
 					    AUDIOHDC_AMP_SET_OUTPUT;
 					break;
 				}
 				if (widget->inamp_cap &
 				    AUDIOHDC_AMP_CAP_MUTE_CAP) {
-					ostream->mute_wid = widget->wid_wid;
-					ostream->mute_dir =
+					path->mute_wid = widget->wid_wid;
+					path->mute_dir =
 					    AUDIOHDC_AMP_SET_INPUT;
 					break;
 				}
@@ -3308,29 +3420,28 @@ audiohd_build_output_amp(hda_codec_t *codec)
 				gain = (widget->outamp_cap &
 				    AUDIOHDC_AMP_CAP_STEP_NUMS);
 				if (gain && gain > pin->gain_bits) {
-					ostream->gain_dir =
+					path->gain_dir =
 					    AUDIOHDC_AMP_SET_OUTPUT;
-					ostream->gain_bits = gain;
-					ostream->gain_wid = widget->wid_wid;
+					path->gain_bits = gain;
+					path->gain_wid = widget->wid_wid;
 				}
 				gain = widget->inamp_cap &
 				    AUDIOHDC_AMP_CAP_STEP_NUMS;
 				if (gain && (gain > pin->gain_bits) &&
 				    (widget->type != WTYPE_AUDIO_MIX)) {
-					ostream->gain_dir =
+					path->gain_dir =
 					    AUDIOHDC_AMP_SET_INPUT;
-					ostream->gain_bits = gain;
-					ostream->gain_wid = widget->wid_wid;
+					path->gain_bits = gain;
+					path->gain_wid = widget->wid_wid;
 				}
 				if (widget->selconn == AUDIOHD_NULL_CONN)
 					break;
 				wid = widget->avail_conn[widget->selconn];
 				widget = codec->widget[wid];
 			}
-			ostream->gain_bits >>= AUDIOHD_GAIN_OFF;
+			path->gain_bits >>= AUDIOHD_GAIN_OFF;
 		}
 
-		ostream = ostream->next_stream;
 	}
 
 }	/* audiohd_build_output_amp */
@@ -3345,17 +3456,19 @@ static void
 audiohd_finish_output_path(hda_codec_t *codec)
 {
 	audiohd_state_t		*statep = codec->soft_statep;
-	audiohd_ostream_t	*ostream;
+	audiohd_path_t		*path;
 	audiohd_widget_t	*widget;
 	audiohd_pin_t		*pin;
 	uint_t			caddr = codec->index;
 	wid_t			wid;
-	int			i;
+	int			i, j;
 
-	ostream = codec->ostream;
-	while (ostream) {
-		for (i = 0; i < ostream->pin_nums; i++) {
-			wid = ostream->pin_wid[i];
+	for (i = 0; i < codec->soft_statep->pathnum; i++) {
+		path = codec->soft_statep->path[i];
+		if (!path || path->path_type != PLAY || path->codec != codec)
+			continue;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
 			widget = codec->widget[wid];
 			pin = (audiohd_pin_t *)widget->priv;
 			{
@@ -3432,7 +3545,6 @@ audiohd_finish_output_path(hda_codec_t *codec)
 				widget = codec->widget[wid];
 			}
 		}
-		ostream = ostream->next_stream;
 	}
 }	/* audiohd_finish_output_path() */
 
@@ -3449,7 +3561,7 @@ audiohd_finish_output_path(hda_codec_t *codec)
  */
 static int
 audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
-    int depth, audiohd_istream_t *istream)
+    int depth, audiohd_path_t *path)
 {
 	audiohd_widget_t	*widget = codec->widget[wid];
 	audiohd_pin_t		*pin;
@@ -3491,9 +3603,8 @@ audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
 		case DTYPE_AUX:
 			widget->path_flags |= AUDIOHD_PATH_ADC;
 			widget->in_weight++;
-			istream->pin_wid[istream->pin_nums++] = wid;
-			pin->adc_dac_wid = istream->adc_wid;
-			ATRACE_32("Pin in:", wid);
+			path->pin_wid[path->pin_nums++] = wid;
+			pin->adc_dac_wid = path->adda_wid;
 			return (AUDIO_SUCCESS);
 		}
 		break;
@@ -3507,7 +3618,7 @@ audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
 			widget->selconn = 0;
 			retval = audiohd_find_input_pins(codec,
 			    widget->avail_conn[0],
-			    allowmixer, depth + 1, istream);
+			    allowmixer, depth + 1, path);
 			if (retval != AUDIO_FAILURE) {
 				widget->path_flags |= AUDIOHD_PATH_ADC;
 				widget->in_weight++;
@@ -3524,12 +3635,12 @@ audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
 			for (int i = 0; i < widget->nconns; i++) {
 				retval = audiohd_find_input_pins(codec,
 				    widget->avail_conn[i], 0, depth + 1,
-				    istream);
+				    path);
 				if (retval != AUDIO_FAILURE) {
 					widget->in_weight++;
-					num = istream->pin_nums - 1;
-					istream->sum_selconn[num] = i;
-					istream->sum_wid = wid;
+					num = path->pin_nums - 1;
+					path->sum_selconn[num] = i;
+					path->sum_wid = wid;
 					widget->path_flags |=
 					    AUDIOHD_PATH_ADC;
 					if (widget->selconn ==
@@ -3540,7 +3651,7 @@ audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
 			}
 
 			/* return SUCCESS if we found at least one input path */
-			if (istream->pin_nums > 0)
+			if (path->pin_nums > 0)
 				retval = AUDIO_SUCCESS;
 		} else {
 			/*
@@ -3550,7 +3661,7 @@ audiohd_find_input_pins(hda_codec_t *codec, wid_t wid, int allowmixer,
 			for (i = 0; i < widget->nconns; i++) {
 				retval = audiohd_find_input_pins(codec,
 				    widget->avail_conn[i], 0, depth + 1,
-				    istream);
+				    path);
 				if (retval != AUDIO_FAILURE) {
 					widget->selconn = i;
 					widget->path_flags |= AUDIOHD_PATH_ADC;
@@ -3577,11 +3688,12 @@ static void
 audiohd_build_input_path(hda_codec_t *codec)
 {
 	audiohd_widget_t	*widget;
-	audiohd_istream_t	*istream = NULL;
+	audiohd_path_t		*path = NULL;
 	wid_t			wid;
 	int			i;
 	int			retval;
 	uint8_t			rtag = 0;
+	audiohd_state_t		*statep = codec->soft_statep;
 
 	for (wid = codec->first_wid; wid <= codec->last_wid; wid++) {
 
@@ -3591,13 +3703,13 @@ audiohd_build_input_path(hda_codec_t *codec)
 		if (!widget || widget->type != WTYPE_AUDIO_IN)
 			continue;
 
-		if (istream == NULL)
-			istream = kmem_zalloc(sizeof (audiohd_istream_t),
+		if (path == NULL)
+			path = kmem_zalloc(sizeof (audiohd_path_t),
 			    KM_SLEEP);
 		else
-			bzero(istream, sizeof (audiohd_istream_t));
+			bzero(path, sizeof (audiohd_port_t));
 
-		istream->adc_wid = wid;
+		path->adda_wid = wid;
 
 		/*
 		 * Is there any ADC widget which has more than one input ??
@@ -3611,28 +3723,23 @@ audiohd_build_input_path(hda_codec_t *codec)
 		 */
 		for (i = 0; i < widget->nconns; i++) {
 			retval = audiohd_find_input_pins(codec,
-			    widget->avail_conn[i], 1, 0, istream);
+			    widget->avail_conn[i], 1, 0, path);
 			if (retval == AUDIO_SUCCESS) {
-				istream->rtag = ++rtag;
+				path->codec = codec;
+				path->statep = statep;
+				path->path_type = RECORD;
+				path->tag = ++rtag;
 				codec->nistream++;
-				if (codec->istream) {
-					audiohd_istream_t	*p =
-					    codec->istream;
-					while (p->next_stream)
-						p = p->next_stream;
-					p->next_stream = istream;
-				} else {
-					codec->istream = istream;
-				}
+				statep->path[statep->pathnum++] = path;
 				widget->selconn = i;
-				widget->priv = istream;
-				istream = NULL;
+				widget->priv = path;
+				path = NULL;
 				break;
 			}
 		}
 	}
-	if (istream)
-		kmem_free(istream, sizeof (audiohd_istream_t));
+	if (path)
+		kmem_free(path, sizeof (audiohd_path_t));
 }	/* audiohd_build_input_path */
 
 /*
@@ -3644,18 +3751,22 @@ audiohd_build_input_path(hda_codec_t *codec)
 static void
 audiohd_build_input_amp(hda_codec_t *codec)
 {
-	audiohd_istream_t	*istream;
+	audiohd_path_t		*path;
 	audiohd_widget_t	*wsum, *wadc, *w;
 	audiohd_pin_t		*pin;
 	uint_t			gain;
 	wid_t			wid;
-	int			i;
+	int			i, j;
 	int			weight;
 
-	istream = codec->istream;
-	while (istream) {
-		wid = istream->adc_wid;
-		wadc = codec->widget[wid];
+	for (i = 0; i < codec->soft_statep->pathnum; i++) {
+		path = codec->soft_statep->path[i];
+		if (path == NULL || path->path_type == PLAY ||
+		    path->codec != codec)
+			continue;
+
+		wid = path->adda_wid;
+		wadc = path->codec->widget[wid];
 		weight = wadc->in_weight;
 
 		/*
@@ -3665,21 +3776,21 @@ audiohd_build_input_amp(hda_codec_t *codec)
 		w = wadc;
 		while (w) {
 			if (w->outamp_cap & AUDIOHDC_AMP_CAP_MUTE_CAP) {
-				istream->mute_wid = w->wid_wid;
-				istream->mute_dir = AUDIOHDC_AMP_SET_OUTPUT;
+				path->mute_wid = w->wid_wid;
+				path->mute_dir = AUDIOHDC_AMP_SET_OUTPUT;
 				break;
 			}
 			if ((w->inamp_cap & AUDIOHDC_AMP_CAP_MUTE_CAP) &&
-			    (w->wid_wid != istream->sum_wid)) {
-				istream->mute_wid = w->wid_wid;
-				istream->mute_dir = AUDIOHDC_AMP_SET_INPUT;
+			    (w->wid_wid != path->sum_wid)) {
+				path->mute_wid = w->wid_wid;
+				path->mute_dir = AUDIOHDC_AMP_SET_INPUT;
 				break;
 			}
 
 			if (w->selconn == AUDIOHD_NULL_CONN)
 				break;
 			wid = w->avail_conn[w->selconn];
-			w = codec->widget[wid];
+			w = path->codec->widget[wid];
 			if (w && w->in_weight != weight)
 				break;
 		}
@@ -3692,39 +3803,38 @@ audiohd_build_input_amp(hda_codec_t *codec)
 		gain = 0;
 		while (w) {
 			gain = (w->outamp_cap & AUDIOHDC_AMP_CAP_STEP_NUMS);
-			if (gain && gain > istream->gain_bits) {
-				istream->gain_dir = AUDIOHDC_AMP_SET_OUTPUT;
-				istream->gain_bits = gain;
-				istream->gain_wid = w->wid_wid;
+			if (gain && gain > path->gain_bits) {
+				path->gain_dir = AUDIOHDC_AMP_SET_OUTPUT;
+				path->gain_bits = gain;
+				path->gain_wid = w->wid_wid;
 			}
 			gain = w->inamp_cap & AUDIOHDC_AMP_CAP_STEP_NUMS;
-			if (gain && (gain > istream->gain_bits) &&
-			    (w->wid_wid != istream->sum_wid)) {
-				istream->gain_dir = AUDIOHDC_AMP_SET_INPUT;
-				istream->gain_bits = gain;
-				istream->gain_wid = w->wid_wid;
+			if (gain && (gain > path->gain_bits) &&
+			    (w->wid_wid != path->sum_wid)) {
+				path->gain_dir = AUDIOHDC_AMP_SET_INPUT;
+				path->gain_bits = gain;
+				path->gain_wid = w->wid_wid;
 			}
 			if (w->selconn == AUDIOHD_NULL_CONN)
 				break;
 			wid = w->avail_conn[w->selconn];
-			w = codec->widget[wid];
+			w = path->codec->widget[wid];
 		}
-		istream->gain_bits >>= AUDIOHD_GAIN_OFF;
+		path->gain_bits >>= AUDIOHD_GAIN_OFF;
 
 		/*
 		 * If the input path has one pin only, the mute/amp
 		 * controlling is shared by the whole path and pin
 		 */
-		if (istream->pin_nums == 1) {
-			wid = istream->pin_wid[0];
-			w = codec->widget[wid];
+		if (path->pin_nums == 1) {
+			wid = path->pin_wid[0];
+			w = path->codec->widget[wid];
 			pin = (audiohd_pin_t *)w->priv;
-			pin->gain_dir = istream->gain_dir;
-			pin->gain_bits = istream->gain_bits;
-			pin->gain_wid = istream->gain_wid;
-			pin->mute_wid = istream->mute_wid;
-			pin->mute_dir = istream->mute_dir;
-			istream = istream->next_stream;
+			pin->gain_dir = path->gain_dir;
+			pin->gain_bits = path->gain_bits;
+			pin->gain_wid = path->gain_wid;
+			pin->mute_wid = path->mute_wid;
+			pin->mute_dir = path->mute_dir;
 			continue;
 		}
 
@@ -3733,16 +3843,12 @@ audiohd_build_input_amp(hda_codec_t *codec)
 		 * or mixer along the input path, and the sum_wid
 		 * is the widget's node id.
 		 */
-		wid = istream->sum_wid;
-		wsum = codec->widget[wid]; /* sum widget */
-		if (wsum == NULL) {
-			istream = istream->next_stream;
-			continue;
-		}
+		wid = path->sum_wid;
+		wsum = path->codec->widget[wid]; /* sum widget */
 
-		for (i = 0; i < istream->pin_nums; i++) {
-			wid = istream->pin_wid[i];
-			w = codec->widget[wid];
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			w = path->codec->widget[wid];
 			pin = (audiohd_pin_t *)w->priv;
 
 			/* find node for mute */
@@ -3750,8 +3856,8 @@ audiohd_build_input_amp(hda_codec_t *codec)
 				pin->mute_wid = wsum->wid_wid;
 				pin->mute_dir = AUDIOHDC_AMP_SET_INPUT;
 			} else {
-				wid = wsum->avail_conn[istream->sum_selconn[i]];
-				w = codec->widget[wid];
+				wid = wsum->avail_conn[path->sum_selconn[i]];
+				w = path->codec->widget[wid];
 				while (w) {
 					if (w->outamp_cap &
 					    AUDIOHDC_AMP_CAP_MUTE_CAP) {
@@ -3771,14 +3877,14 @@ audiohd_build_input_amp(hda_codec_t *codec)
 					if (w->selconn == AUDIOHD_NULL_CONN)
 						break;
 					wid = w->avail_conn[w->selconn];
-					w = codec->widget[wid];
+					w = path->codec->widget[wid];
 				}
 			}
 
 			/* find node for amp controlling */
 			gain = (wsum->inamp_cap & AUDIOHDC_AMP_CAP_STEP_NUMS);
-			wid = wsum->avail_conn[istream->sum_selconn[i]];
-			w = codec->widget[wid];
+			wid = wsum->avail_conn[path->sum_selconn[i]];
+			w = path->codec->widget[wid];
 			while (w) {
 				gain = (w->outamp_cap &
 				    AUDIOHDC_AMP_CAP_STEP_NUMS);
@@ -3797,11 +3903,10 @@ audiohd_build_input_amp(hda_codec_t *codec)
 				if (w->selconn == AUDIOHD_NULL_CONN)
 					break;
 				wid = w->avail_conn[w->selconn];
-				w = codec->widget[wid];
+				w = path->codec->widget[wid];
 			}
 			pin->gain_bits >>= AUDIOHD_GAIN_OFF;
 		}
-		istream = istream->next_stream;
 	}
 }	/* audiohd_build_input_amp() */
 
@@ -3815,17 +3920,20 @@ static void
 audiohd_finish_input_path(hda_codec_t *codec)
 {
 	audiohd_state_t		*statep = codec->soft_statep;
-	audiohd_istream_t	*istream;
+	audiohd_path_t	*path;
 	audiohd_widget_t	*w, *wsum;
 	uint_t			caddr = codec->index;
 	wid_t			wid;
-	int			i;
+	int			i, j;
 
-	for (istream = codec->istream; istream;
-	    istream = istream->next_stream) {
-		wid = istream->adc_wid;
-		w = codec->widget[wid];
-		while (w && (w->wid_wid != istream->sum_wid) &&
+	for (i = 0; i < codec->soft_statep->pathnum; i++) {
+		path = codec->soft_statep->path[i];
+		if (path == NULL || path->path_type == PLAY ||
+		    path->codec != codec)
+			continue;
+		wid = path->adda_wid;
+		w = path->codec->widget[wid];
+		while (w && (w->wid_wid != path->sum_wid) &&
 		    (w->type != WTYPE_PIN)) {
 			if ((w->type == WTYPE_AUDIO_SEL) && (w->nconns > 1))
 				(void) audioha_codec_verb_get(statep, caddr,
@@ -3851,7 +3959,7 @@ audiohd_finish_input_path(hda_codec_t *codec)
 			}
 
 			wid = w->avail_conn[w->selconn];
-			w = codec->widget[wid];
+			w = path->codec->widget[wid];
 		}
 
 		/*
@@ -3866,8 +3974,8 @@ audiohd_finish_input_path(hda_codec_t *codec)
 		/*
 		 * deal with multi-pin input devices.
 		 */
-		wid = istream->sum_wid;
-		wsum = codec->widget[wid];
+		wid = path->sum_wid;
+		wsum = path->codec->widget[wid];
 		if (wsum == NULL)
 			continue;
 		if (wsum->outamp_cap) {
@@ -3878,25 +3986,25 @@ audiohd_finish_input_path(hda_codec_t *codec)
 			    AUDIOHDC_GAIN_MAX);
 		}
 
-		for (i = 0; i < istream->pin_nums; i++) {
+		for (j = 0; j < path->pin_nums; j++) {
 			if (wsum->inamp_cap) {
 				(void) audioha_codec_4bit_verb_get(statep,
 				    caddr,
 				    wsum->wid_wid, AUDIOHDC_VERB_SET_AMP_MUTE,
 				    AUDIOHDC_AMP_SET_LR_INPUT |
 				    AUDIOHDC_GAIN_MAX |
-				    (istream->sum_selconn[i] <<
+				    (path->sum_selconn[j] <<
 				    AUDIOHDC_AMP_SET_INDEX_OFFSET));
 			}
 			if (wsum->type == WTYPE_AUDIO_SEL) {
 				(void) audioha_codec_verb_get(statep, caddr,
 				    wsum->wid_wid,
 				    AUDIOHDC_VERB_SET_CONN_SEL,
-				    istream->sum_selconn[i]);
+				    path->sum_selconn[j]);
 			}
 
-			wid = wsum->avail_conn[istream->sum_selconn[i]];
-			w = codec->widget[wid];
+			wid = wsum->avail_conn[path->sum_selconn[j]];
+			w = path->codec->widget[wid];
 			while (w && w->type != WTYPE_PIN) {
 				if ((w->type != WTYPE_AUDIO_MIX) &&
 				    (w->nconns > 1))
@@ -3927,7 +4035,7 @@ audiohd_finish_input_path(hda_codec_t *codec)
 					    AUDIOHDC_AMP_SET_INDEX_OFFSET));
 				}
 				wid = w->avail_conn[w->selconn];
-				w = codec->widget[wid];
+				w = path->codec->widget[wid];
 			}
 		}
 	}	/* end of istream loop */
@@ -3947,7 +4055,7 @@ audiohd_finish_input_path(hda_codec_t *codec)
  */
 static int
 audiohd_find_inpin_for_monitor(hda_codec_t *codec,
-    audiohd_ostream_t *ostream, wid_t id, int mixer)
+    audiohd_path_t *path, wid_t id, int mixer)
 {
 	wid_t wid;
 	audiohd_widget_t	*widget;
@@ -3965,24 +4073,19 @@ audiohd_find_inpin_for_monitor(hda_codec_t *codec,
 			return (uint32_t)(AUDIO_FAILURE);
 		switch (pin->device) {
 			case DTYPE_SPDIF_IN:
-				ATRACE("Monitor SPDIF found!", NULL);
 				widget->path_flags |= AUDIOHD_PATH_MON;
 				return (AUDIO_SUCCESS);
 			case DTYPE_CD:
-				ATRACE("Monitor CD found!", NULL);
 				widget->path_flags |= AUDIOHD_PATH_MON;
 				return (AUDIO_SUCCESS);
 			case DTYPE_LINE_IN:
-				ATRACE("Monitor Line in found!", NULL);
 				widget->path_flags |= AUDIOHD_PATH_MON;
 				return (AUDIO_SUCCESS);
 			case DTYPE_MIC_IN:
-				ATRACE("Monitor Mic found!", NULL);
 				widget->path_flags |= AUDIOHD_PATH_MON;
 				return (AUDIO_SUCCESS);
 			case DTYPE_AUX:
 				widget->path_flags |= AUDIOHD_PATH_MON;
-				ATRACE("Monitor AUX found!", NULL);
 				return (AUDIO_SUCCESS);
 			default:
 				return (uint32_t)(AUDIO_FAILURE);
@@ -4006,7 +4109,7 @@ audiohd_find_inpin_for_monitor(hda_codec_t *codec,
 				    AUDIOHD_PATH_DAC)
 					continue;
 				if (audiohd_find_inpin_for_monitor(codec,
-				    ostream,
+				    path,
 				    widget->avail_conn[i], mixer) ==
 				    AUDIO_SUCCESS) {
 					widget->selmon[widget->used++] = i;
@@ -4021,7 +4124,7 @@ audiohd_find_inpin_for_monitor(hda_codec_t *codec,
 				    AUDIOHD_PATH_DAC)
 					continue;
 				if (audiohd_find_inpin_for_monitor(codec,
-				    ostream,
+				    path,
 				    widget->avail_conn[i],
 				    mixer) ==
 				    AUDIO_SUCCESS) {
@@ -4055,99 +4158,106 @@ audiohd_find_inpin_for_monitor(hda_codec_t *codec,
 static void
 audiohd_build_monitor_path(hda_codec_t *codec)
 {
-	audiohd_ostream_t		*ostream;
-	audiohd_widget_t		*widget;
-	wid_t				wid;
-	int				i, j, k, find;
-	int				mixernum = 0;
-
-	ostream = codec->ostream;
-	while (ostream) {
-		for (i = 0; i < ostream->pin_nums; i++) {
-		wid = ostream->pin_wid[i];
-		widget = codec->widget[wid];
-		k = 0;
-
-		while (widget) {
-			while (widget && ((widget->type != WTYPE_AUDIO_MIX) ||
-			    (widget->nconns < 2))) {
-				if (widget->selconn == AUDIOHD_NULL_CONN)
-					break;
-				wid = widget->avail_conn[widget->selconn];
-				widget = codec->widget[wid];
-			}
+	audiohd_path_t		*path;
+	audiohd_widget_t	*widget;
+	audiohd_state_t		*statep = codec->soft_statep;
+	wid_t			wid;
+	int			i, j, k, l, find;
+	int			mixernum = 0;
 
-			/*
-			 * No mixer in this output path, we cannot build
-			 * mixer path for this path, skip it, and continue
-			 * for next output path.
-			 */
-			if (widget == NULL || widget->selconn ==
-			    AUDIOHD_NULL_CONN) {
-				break;
-			}
-			mixernum++;
-			for (j = 0; j < widget->nconns; j++) {
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->codec != codec ||path->path_type != PLAY)
+			continue;
+		for (j = 0; j < path->pin_nums; j++) {
+			wid = path->pin_wid[j];
+			widget = codec->widget[wid];
+			l = 0;
+			while (widget) {
+				while (widget &&
+				    ((widget->type != WTYPE_AUDIO_MIX) ||
+				    (widget->nconns < 2))) {
+					if (widget->selconn ==
+					    AUDIOHD_NULL_CONN)
+						break;
+					wid =
+					    widget->avail_conn[widget->selconn];
+					widget = codec->widget[wid];
+				}
 
 				/*
-				 * this connection must be routined to DAC
-				 * instead
-				 * of an input pin widget, we needn't waste
-				 * time for
-				 * it
+				 * No mixer in this output path, we cannot build
+				 * mixer path for this path, skip it,
+				 * and continue
+				 * for next output path.
 				 */
-				if (widget->selconn == j)
-					continue;
-				find = 0;
-				if (audiohd_find_inpin_for_monitor(codec,
-				    ostream,
-				    widget->avail_conn[j], 0) ==
-				    AUDIO_SUCCESS) {
-					ostream->mon_wid[i][k] = wid;
-					widget->selmon[widget->used++] = j;
-					widget->path_flags |= AUDIOHD_PATH_MON;
-					find = 1;
-					ATRACE("Exclusive monitor found",
-					    NULL);
-				} else if (
-				    audiohd_find_inpin_for_monitor(codec,
-				    ostream,
-				    widget->avail_conn[j], 1) ==
-				    AUDIO_SUCCESS) {
-					ostream->mon_wid[i][k] = wid;
-					widget->selmon[widget->used++] = j;
-					widget->path_flags |= AUDIOHD_PATH_MON;
+				if (widget == NULL || widget->selconn ==
+				    AUDIOHD_NULL_CONN) {
+					break;
+				}
+				mixernum++;
+				for (k = 0; k < widget->nconns; k++) {
+
+					/*
+					 * this connection must be routined
+					 * to DAC instead of an input pin
+					 * widget, we needn't waste time for
+					 * it
+					 */
+					if (widget->selconn == k)
+						continue;
+					find = 0;
+					if (audiohd_find_inpin_for_monitor(
+					    codec,
+					    path,
+					    widget->avail_conn[k], 0) ==
+					    AUDIO_SUCCESS) {
+						path->mon_wid[j][l] = wid;
+						widget->selmon[widget->used++] =
+						    k;
+						widget->path_flags |=
+						    AUDIOHD_PATH_MON;
+						find = 1;
+					} else if (
+					    audiohd_find_inpin_for_monitor(
+					    codec,
+					    path,
+					    widget->avail_conn[k], 1) ==
+					    AUDIO_SUCCESS) {
+						path->mon_wid[j][l] = wid;
+						widget->selmon[widget->used++] =
+						    k;
+						widget->path_flags |=
+						    AUDIOHD_PATH_MON;
+						find = 1;
 
-					find = 1;
-					ATRACE("Share monitor found", NULL);
+					}
 
 				}
 
+				/*
+				 * we needn't check widget->selconn here
+				 * since this
+				 * widget is a selector or mixer, it cannot
+				 * be NULL connection.
+				 */
+				if (!find) {
+					path->mon_wid[i][l] = 0;
+					widget->path_flags |=
+					    AUDIOHD_PATH_NOMON;
+				}
+				wid = widget->avail_conn[widget->selconn];
+				widget = codec->widget[wid];
+				l++;
 			}
-
-			/*
-			 * we needn't check widget->selconn here since this
-			 * widget
-			 * is a selector or mixer, it cannot be NULL connection.
-			 */
-			if (!find) {
-				ATRACE_32(
-				    "No input pin found on this mixer:",
-				    widget->wid_wid);
-				ostream->mon_wid[i][k] = 0;
-				widget->path_flags |= AUDIOHD_PATH_NOMON;
-			}
-			wid = widget->avail_conn[widget->selconn];
-			widget = codec->widget[wid];
-			k++;
-		}
-		ostream->maxmixer[i] = k;
+			path->maxmixer[j] = l;
 		}
 
-		ostream = ostream->next_stream;
 	}
 	if (mixernum == 0)
-		ATRACE("Monitor unsupported", NULL);
+		statep->monitor_unsupported = B_TRUE;
+	else
+		statep->monitor_unsupported = B_FALSE;
 }	/* audiohd_build_monitor_path */
 
 /*
@@ -4192,7 +4302,6 @@ audiohd_do_finish_monitor_path(hda_codec_t *codec, audiohd_widget_t *wgt)
 		(void) audioha_codec_verb_get(statep, caddr,
 		    widget->wid_wid,
 		    AUDIOHDC_VERB_SET_CONN_SEL, widget->selmon[0]);
-		ATRACE("Monitor selector exist!", NULL);
 	}
 	widget->finish = 1;
 	if (widget->used == 0)
@@ -4215,16 +4324,19 @@ audiohd_do_finish_monitor_path(hda_codec_t *codec, audiohd_widget_t *wgt)
 static void
 audiohd_finish_monitor_path(hda_codec_t *codec)
 {
-	audiohd_ostream_t	*ostream;
+	audiohd_path_t		*path;
 	audiohd_widget_t	*widget;
+	audiohd_state_t		*statep = codec->soft_statep;
 	wid_t			wid;
-	int 			i, j;
+	int 			i, j, k;
 
-	ostream = codec->ostream;
-	while (ostream) {
-		for (i = 0; i < ostream->pin_nums; i++) {
-			for (j = 0; j < ostream->maxmixer[j]; j++) {
-				wid = ostream->mon_wid[i][j];
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->codec != codec || path->path_type != PLAY)
+			continue;
+		for (j = 0; j < path->pin_nums; j++) {
+			for (k = 0; k < path->maxmixer[j]; k++) {
+				wid = path->mon_wid[j][k];
 				if (wid == 0) {
 					continue;
 				}
@@ -4232,7 +4344,6 @@ audiohd_finish_monitor_path(hda_codec_t *codec)
 				audiohd_do_finish_monitor_path(codec, widget);
 			}
 		}
-	ostream = ostream->next_stream;
 	}
 }	/* audiohd_finish_monitor_path */
 
@@ -4294,20 +4405,23 @@ audiohd_do_build_monitor_amp(hda_codec_t *codec, audiohd_pin_t *pin,
 static void
 audiohd_build_monitor_amp(hda_codec_t *codec)
 {
-	audiohd_ostream_t	*ostream;
+	audiohd_path_t		*path;
 	audiohd_widget_t	*widget, *w;
+	audiohd_state_t		*statep = codec->soft_statep;
 	audiohd_pin_t		*pin;
 	wid_t			wid, id;
-	int			i, j;
+	int			i, j, k;
 
-	ostream = codec->ostream;
-	while (ostream) {
-		for (i = 0; i < ostream->pin_nums; i++) {
-			id = ostream->pin_wid[i];
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->codec != codec || path->path_type != PLAY)
+			continue;
+		for (j = 0; j < path->pin_nums; j++) {
+			id = path->pin_wid[j];
 			w = codec->widget[id];
 			pin = (audiohd_pin_t *)(w->priv);
-			for (j = 0; j < ostream->maxmixer[i]; j++) {
-				wid = ostream->mon_wid[i][j];
+			for (k = 0; k < path->maxmixer[j]; k++) {
+				wid = path->mon_wid[j][k];
 				if (!wid)
 					continue;
 				widget = codec->widget[wid];
@@ -4315,10 +4429,10 @@ audiohd_build_monitor_amp(hda_codec_t *codec)
 				    widget);
 			}
 		}
-		ostream = ostream->next_stream;
 	}
 }
 
+
 /*
  * audiohd_build_path()
  *
@@ -4350,984 +4464,854 @@ audiohd_build_path(audiohd_state_t *statep)
 }	/* audiohd_build_path */
 
 /*
- * audiohd_init_ports()
- *
- * Description:
- * 	Since the Solaris framework only supports 5 types of output port,
- * 	and each type can have only one instance. So we use a global
- * 	array (g_outport) to hold  all output ports, and when the framework
- * 	has this restriction lifted, we need to reimplement it.
+ * audiohd_allocate_port()
  */
 static int
-audiohd_init_ports(audiohd_state_t *statep)
+audiohd_allocate_port(audiohd_state_t *statep)
 {
-	hda_codec_t			*codec;
-	audiohd_ostream_t		*ostream;
-	audiohd_istream_t		*istream;
-	audiohd_widget_t		*widget;
-	audiohd_pin_t			*pin;
-	uint_t				inputs, outputs;
-	int				i;
-	uint32_t			ctrl;
-	uint8_t				ctrl8;
+	int			i, j;
+	audiohd_port_t		*port;
+	int			dir;
+	unsigned		caps;
+	char			*prop;
+	int			rc;
+	audio_dev_t		*adev;
+	dev_info_t		*dip;
+	ddi_dma_cookie_t	cookie;
+	uint_t			count;
+	uint64_t		buf_phys_addr;
+	sd_bdle_t		*entry;
+	uint16_t		gcap;
+	size_t			real_size;
 
-	codec = NULL;
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++)
-		if (statep->codec[i]) {
-			codec = statep->codec[i];
+	adev = statep->adev;
+	dip = statep->hda_dip;
+
+	ddi_dma_attr_t	dma_attr = {
+		DMA_ATTR_V0,		/* version */
+		0,			/* addr_lo */
+		0xffffffffffffffffULL,	/* addr_hi */
+		0x00000000ffffffffULL,	/* count_max */
+		128,			/* 128-byte alignment as HD spec */
+		0xfff,			/* burstsize */
+		1,			/* minxfer */
+		0xffffffff,		/* maxxfer */
+		0xffffffff,		/* seg */
+		1,			/* sgllen */
+		1,			/* granular */
+		0			/* flags */
+	};
+
+	gcap = AUDIOHD_REG_GET16(AUDIOHD_REG_GCAP);
+	if ((gcap & AUDIOHDR_GCAP_64OK) == 0)
+		dma_attr.dma_attr_addr_hi = 0xffffffffUL;
+
+	for (i = 0; i < PORT_MAX; i++) {
+		port = kmem_zalloc(sizeof (*port), KM_SLEEP);
+		port->started = B_FALSE;
+		port->triggered = B_FALSE;
+		statep->port[i] = port;
+		port->statep = statep;
+		switch (i) {
+		case PORT_ADC:
+			prop = "record-interrupts";
+			dir = DDI_DMA_READ | DDI_DMA_CONSISTENT;
+			caps = ENGINE_INPUT_CAP;
+			port->sync_dir = DDI_DMA_SYNC_FORKERNEL;
+			port->nchan = statep->rchan;
+			port->index = 1;
+			port->regoff = AUDIOHD_REG_SD_BASE;
 			break;
+		case PORT_DAC:
+			prop = "play-interrupts";
+			dir = DDI_DMA_WRITE | DDI_DMA_CONSISTENT;
+			caps = ENGINE_OUTPUT_CAP;
+			port->sync_dir = DDI_DMA_SYNC_FORDEV;
+			port->nchan = statep->pchan;
+			port->index = statep->hda_input_streams + 1;
+			port->regoff = AUDIOHD_REG_SD_BASE +
+			    AUDIOHD_REG_SD_LEN *
+			    statep->hda_input_streams;
+			break;
+		default:
+			return (DDI_FAILURE);
 		}
 
-	if (codec == NULL)
-		return (DDI_FAILURE);
+		port->intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+		    DDI_PROP_DONTPASS, prop, AUDIOHD_INTS);
+
+		/* make sure the values are good */
+		if (port->intrs < AUDIOHD_MIN_INTS) {
+			audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+			    prop, port->intrs, AUDIOHD_INTS);
+			port->intrs = AUDIOHD_INTS;
+		} else if (port->intrs > AUDIOHD_MAX_INTS) {
+			audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+			    prop, port->intrs, AUDIOHD_INTS);
+			port->intrs = AUDIOHD_INTS;
+		}
 
-	if (codec->ostream) {
-		outputs = 0;
-		ostream = codec->ostream;
-		while (ostream) {
-			ostream->in_use = 1;
-			for (i = 0; i < ostream->pin_nums; i++) {
-				widget = codec->widget[ostream->pin_wid[i]];
-				pin = (audiohd_pin_t *)widget->priv;
-				if (pin->device == DTYPE_SPDIF_OUT) {
-					ATRACE("SPDIFOUT found!", NULL);
-					pin->sada_porttype = AUDIO_SPDIF_OUT |
-					    AUDIOHD_SADA_OUTPUT;
-					outputs |= AUDIO_SPDIF_OUT;
-					ctrl = audioha_codec_verb_get(statep,
-					    codec->index, ostream->dac_wid,
-					    AUDIOHDC_VERB_GET_SPDIF_CONTROL, 0);
-					ctrl |= AUDIOHD_SPDIF_ON;
-					ctrl8 = ctrl & AUDIOHD_SPDIF_MASK;
-					(void) audioha_codec_verb_get(statep,
-					    codec->index,
-					    ostream->dac_wid,
-					    AUDIOHDC_VERB_SET_SPDIF_LCONTROL,
-					    ctrl8);
-				} else if (pin->device == DTYPE_LINEOUT) {
-					pin->sada_porttype = AUDIO_LINE_OUT |
-					    AUDIOHD_SADA_OUTPUT;
-					outputs |= AUDIO_LINE_OUT;
-				} else if (pin->device == DTYPE_SPEAKER) {
-					pin->sada_porttype = AUDIO_SPEAKER |
-					    AUDIOHD_SADA_OUTPUT;
-					outputs |= AUDIO_SPEAKER;
-
-				} else if (pin->device == DTYPE_HP_OUT) {
-					pin->sada_porttype = AUDIO_HEADPHONE |
-					    AUDIOHD_SADA_OUTPUT;
-					outputs |= AUDIO_HEADPHONE;
-				} else {
-					pin->sada_porttype = AUDIO_AUX1_OUT |
-					    AUDIOHD_SADA_OUTPUT;
-					outputs |= AUDIO_AUX1_OUT;
-				}
-			}
-			ostream = ostream->next_stream;
+		port->format = AUDIOHD_FMT_PCM;
+		port->fragfr = 48000 / port->intrs;
+		port->fragfr = (port->fragfr + AUDIOHD_FRAGFR_ALIGN - 1) & ~
+		    (AUDIOHD_FRAGFR_ALIGN - 1);
+		port->samp_size = port->fragfr * port->nchan * 2;
+		port->samp_size = (port->samp_size +
+		    AUDIOHD_BDLE_BUF_ALIGN - 1) & ~
+		    (AUDIOHD_BDLE_BUF_ALIGN - 1);
+
+		/* allocate dma handle */
+		rc = ddi_dma_alloc_handle(dip, &dma_attr, DDI_DMA_SLEEP,
+		    NULL, &port->samp_dmah);
+		if (rc != DDI_SUCCESS) {
+			audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d",
+			    rc);
+			goto error_alloc_dma_exit1;
 		}
-		statep->hda_info_defaults.play.port = outputs;
-		statep->hda_info_defaults.play.avail_ports = outputs;
-		statep->hda_info_defaults.play.mod_ports = outputs;
-		statep->hda_out_ports = outputs;
-		ostream = codec->ostream;
-		while (ostream) {
-			if (ostream->in_use == 0)
-				break;
-			(void) audioha_codec_verb_get(statep, codec->index,
-			    ostream->dac_wid, AUDIOHDC_VERB_SET_STREAM_CHANN,
-			    statep->hda_play_stag << AUDIOHD_PLAY_TAG_OFF);
-			ostream = ostream->next_stream;
+		/* allocate DMA buffer */
+		rc = ddi_dma_mem_alloc(port->samp_dmah, port->samp_size *
+		    AUDIOHD_BDLE_NUMS,
+		    &hda_dev_accattr,
+		    DDI_DMA_CONSISTENT,
+		    DDI_DMA_SLEEP, NULL, &port->samp_kaddr,
+		    &real_size, &port->samp_acch);
+		if (rc == DDI_FAILURE) {
+			audio_dev_warn(adev, "dma_mem_alloc failed");
+			goto error_alloc_dma_exit2;
 		}
-	}
 
-	/*
-	 * Init input ports. Because solaris supports just one MIC, so we
-	 * will enable all MICs if applications select MIC as input sources.
-	 * For others, we do the same.
-	 */
-	inputs = 0;
-	for (istream = codec->istream; istream;
-	    istream = istream->next_stream) {
-		istream->in_use = 1;
-		for (i = 0; i < istream->pin_nums; i++) {
-				widget = codec->widget[istream->pin_wid[i]];
-				pin = (audiohd_pin_t *)widget->priv;
-				switch (pin->device) {
-				case DTYPE_SPDIF_IN:
-					break;
-				case DTYPE_MIC_IN:
-					inputs |= AUDIO_MICROPHONE;
-					pin->sada_porttype = AUDIO_MICROPHONE |
-					    AUDIOHD_SADA_INPUT;
-					if (((pin->config >>
-					    AUDIOHD_PIN_CONTP_OFF) &
-					    AUDIOHD_PIN_CONTP_MASK) ==
-					    AUDIOHD_PIN_CON_FIXED)
-						statep->hda_record_stag =
-						    istream->rtag;
-					break;
-				case DTYPE_LINE_IN:
-					inputs |= AUDIO_LINE_IN;
-					pin->sada_porttype = AUDIO_LINE_IN |
-					    AUDIOHD_SADA_INPUT;
-					break;
-				case DTYPE_CD:
-					inputs |= AUDIO_CD;
-					pin->sada_porttype = AUDIO_CD |
-					    AUDIOHD_SADA_INPUT;
-					break;
-				case DTYPE_AUX:
-					inputs |= AUDIO_AUX1_IN |
-					    AUDIOHD_SADA_INPUT;
-				}
+		/* bind DMA buffer */
+		rc = ddi_dma_addr_bind_handle(port->samp_dmah, NULL,
+		    port->samp_kaddr, real_size, dir,
+		    DDI_DMA_SLEEP, NULL, &cookie, &count);
+		if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+			audio_dev_warn(adev,
+			    "ddi_dma_addr_bind_handle failed: %d", rc);
+			goto error_alloc_dma_exit3;
 		}
-		if (inputs & AUDIO_MICROPHONE)
-			statep->hda_info_defaults.record.port =
-			    AUDIO_MICROPHONE;
-		else if (inputs & AUDIO_LINE_IN)
-			statep->hda_info_defaults.record.port = AUDIO_LINE_IN;
-		else if (inputs & AUDIO_CD)
-			statep->hda_info_defaults.record.port = AUDIO_CD;
-		else if (inputs & AUDIO_AUX1_IN)
-			statep->hda_info_defaults.record.port = AUDIO_AUX1_IN;
-		else if (inputs & AUDIO_SPDIF_IN)
-			statep->hda_info_defaults.record.port = AUDIO_SPDIF_IN;
-		else
-			statep->hda_info_defaults.record.port = 0;
+		port->samp_paddr = (uint64_t)cookie.dmac_laddress;
 
-		statep->hda_info_defaults.record.avail_ports = inputs;
-		statep->hda_info_defaults.record.mod_ports = inputs;
-		statep->hda_in_port = 0;
+		/*
+		 * now, from here we allocate DMA
+		 * memory for buffer descriptor list.
+		 * we allocate adjacent DMA memory for all DMA engines.
+		 */
+		rc = ddi_dma_alloc_handle(dip, &dma_attr, DDI_DMA_SLEEP,
+		    NULL, &port->bdl_dmah);
+		if (rc != DDI_SUCCESS) {
+			audio_dev_warn(adev,
+			    "ddi_dma_alloc_handle(bdlist) failed");
+			goto error_alloc_dma_exit3;
+		}
 
-		(void) audioha_codec_verb_get(statep, codec->index,
-		    istream->adc_wid, AUDIOHDC_VERB_SET_STREAM_CHANN,
-		    istream->rtag << AUDIOHD_REC_TAG_OFF);
-	}
+		/*
+		 * we allocate all buffer descriptors lists in continuous
+		 * dma memory.
+		 */
+		port->bdl_size = sizeof (sd_bdle_t) * AUDIOHD_BDLE_NUMS;
+		rc = ddi_dma_mem_alloc(port->bdl_dmah, port->bdl_size,
+		    &hda_dev_accattr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
+		    &port->bdl_kaddr, &real_size, &port->bdl_acch);
+		if (rc != DDI_SUCCESS) {
+			audio_dev_warn(adev,
+			    "ddi_dma_mem_alloc(bdlist) failed");
+			goto error_alloc_dma_exit4;
+		}
 
-	return (DDI_SUCCESS);
-}	/* audiohd_init_ports() */
+		rc = ddi_dma_addr_bind_handle(port->bdl_dmah, NULL,
+		    port->bdl_kaddr,
+		    real_size, DDI_DMA_WRITE | DDI_DMA_CONSISTENT,
+		    DDI_DMA_SLEEP,
+		    NULL, &cookie, &count);
+		if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+			audio_dev_warn(adev, "addr_bind_handle failed");
+			goto error_alloc_dma_exit5;
+		}
+		port->bdl_paddr = (uint64_t)cookie.dmac_laddress;
 
-/*
- * audiohd_disable_intr()
- *
- * Description:
- *	Disable all possible interrupts.
- */
-static void
-audiohd_disable_intr(audiohd_state_t *statep)
-{
-	int		i;
-	uint32_t	base;
+		entry = (sd_bdle_t *)port->bdl_kaddr;
+		buf_phys_addr = port->samp_paddr;
 
-	AUDIOHD_REG_SET32(AUDIOHD_REG_INTCTL, 0);
-	base = AUDIOHD_REG_SD_BASE;
-	for (i = 0; i < statep->hda_streams_nums; i++) {
-		AUDIOHD_REG_SET8(base + AUDIOHD_SDREG_OFFSET_STS,
-		    AUDIOHDR_SD_STS_INTRS);
-		base += AUDIOHD_REG_SD_LEN;
+		for (j = 0; j < AUDIOHD_BDLE_NUMS; j++) {
+			entry->sbde_addr = buf_phys_addr;
+			entry->sbde_len = port->samp_size;
+			entry->sbde_ioc = 1;
+			buf_phys_addr += port->samp_size;
+			entry++;
+		}
+		(void) ddi_dma_sync(port->bdl_dmah, 0, sizeof (sd_bdle_t) *
+		    AUDIOHD_BDLE_NUMS, DDI_DMA_SYNC_FORDEV);
+		port->curpos = 0;
+
+		port->engine = audio_engine_alloc(&audiohd_engine_ops, caps);
+		if (port->engine == NULL) {
+			audio_dev_warn(adev, "audio_engine_alloc failed");
+			goto error_alloc_dma_exit5;
+		}
+
+		audio_engine_set_private(port->engine, port);
+		audio_dev_add_engine(adev, port->engine);
 	}
-	AUDIOHD_REG_SET32(AUDIOHD_REG_INTSTS, (uint32_t)(-1));
 
-}	/* audiohd_disable_intr() */
+	return (DDI_SUCCESS);
+error_alloc_dma_exit5:
+	ddi_dma_mem_free(&port->bdl_acch);
 
+error_alloc_dma_exit4:
+	ddi_dma_free_handle(&port->bdl_dmah);
 
-/*
- * audiohd_reset_stream()
- *
- * Description:
- *	Reset specified stream
- */
-static int
-audiohd_reset_stream(audiohd_state_t *statep, int stream)
-{
-	uint32_t	base;
-	uint8_t		bTmp;
-	int		i;
+error_alloc_dma_exit3:
+	ddi_dma_mem_free(&port->samp_acch);
 
-	base = AUDIOHD_REG_SD_BASE + AUDIOHD_REG_SD_LEN * stream;
-	bTmp = AUDIOHD_REG_GET8(base + AUDIOHD_SDREG_OFFSET_CTL);
+error_alloc_dma_exit2:
+	ddi_dma_free_handle(&port->samp_dmah);
 
-	/* stop stream */
-	bTmp &= ~AUDIOHD_REG_RIRBSIZE;
-	AUDIOHD_REG_SET8(base + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
+error_alloc_dma_exit1:
+	return (AUDIO_FAILURE);
 
-	/* wait 40us for stream to stop as HD spec */
-	drv_usecwait(40);
+}
 
-	/* reset stream */
-	bTmp |= AUDIOHDR_SD_CTL_SRST;
-	AUDIOHD_REG_SET8(base + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
+static void
+audiohd_free_port(audiohd_state_t *statep)
+{
+	int			i;
+	audiohd_port_t		*port;
 
-	for (i = 0; i < AUDIOHD_RETRY_TIMES; i++) {
-		/* Empirical testing time, which works well */
-		drv_usecwait(50);
-		bTmp = AUDIOHD_REG_GET8(base + AUDIOHD_SDREG_OFFSET_CTL);
-		bTmp &= AUDIOHDR_SD_CTL_SRST;
-		if (bTmp)
-			break;
+	if (statep == NULL) {
+		return;
 	}
+	for (i = 0; i < PORT_MAX; i++) {
+		port = statep->port[i];
+		if (port == NULL)
+			continue;
+		if (port->engine) {
+			audio_dev_remove_engine(statep->adev,
+			    port->engine);
+			audio_engine_free(port->engine);
+		}
+		if (port->samp_dmah) {
+			(void) ddi_dma_unbind_handle(port->samp_dmah);
+		}
+		if (port->samp_acch) {
+			ddi_dma_mem_free(&port->samp_acch);
+		}
+		if (port->samp_dmah) {
+			ddi_dma_free_handle(&port->samp_dmah);
+		}
+		if (port->bdl_dmah) {
+			(void) ddi_dma_unbind_handle(port->bdl_dmah);
+		}
+		if (port->bdl_acch) {
+			ddi_dma_mem_free(&port->bdl_acch);
+		}
+		if (port->bdl_dmah) {
+			ddi_dma_free_handle(&port->bdl_dmah);
+		}
 
-	if (!bTmp) {
-		audio_sup_log(NULL, CE_WARN, "!Failed to reset stream %d",
-		    stream);
-		return (AUDIO_FAILURE);
+		kmem_free(port, sizeof (audiohd_port_t));
 	}
+}
 
-	/* Empirical testing time, which works well */
-	drv_usecwait(300);
-
-	/* exit reset stream */
-	bTmp &= ~AUDIOHDR_SD_CTL_SRST;
-	AUDIOHD_REG_SET8(base + AUDIOHD_SDREG_OFFSET_CTL, bTmp);
-
-	for (i = 0; i < AUDIOHD_RETRY_TIMES; i++) {
-		/* Empircal testing time */
-		drv_usecwait(50);
-		bTmp = AUDIOHD_REG_GET8(base + AUDIOHD_SDREG_OFFSET_CTL);
-		bTmp &= AUDIOHDR_SD_CTL_SRST;
-		if (!bTmp)
-			break;
-	}
+/*
+ * audiohd_change_widget_power_state(audiohd_state_t *statep, int off)
+ * Description:
+ * 	This routine is used to change the widget power betwen D0 and D2.
+ * 	D0 is fully on; D2 allows the lowest possible power consuming state
+ * 	from which it can return to the fully on state: D0.
+ */
+static void
+audiohd_change_widget_power_state(audiohd_state_t *statep, int off)
+{
+	int			i;
+	wid_t			wid;
+	hda_codec_t		*codec;
+	audiohd_widget_t	*widget;
 
-	if (bTmp) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!Failed to exit reset state for"
-		    " stream %d, bTmp=0x%02x", stream, bTmp);
-		return (AUDIO_FAILURE);
+	/* Change power to D2 */
+	if (off) {
+		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
+			codec = statep->codec[i];
+			if (!codec)
+				continue;
+			for (wid = codec->first_wid; wid <= codec->last_wid;
+			    wid++) {
+				widget = codec->widget[wid];
+				if (widget->widget_cap &
+				    AUDIOHD_WIDCAP_PWRCTRL) {
+					(void) audioha_codec_verb_get(statep,
+					    codec->index, wid,
+					    AUDIOHDC_VERB_SET_POWER_STATE,
+					    AUDIOHD_PW_D2);
+				}
+			}
+		}
+	/* Change power to D0 */
+	} else {
+		for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
+			codec = statep->codec[i];
+			if (!codec)
+				continue;
+			for (wid = codec->first_wid; wid <= codec->last_wid;
+			    wid++) {
+				widget = codec->widget[wid];
+				if (widget->widget_cap &
+				    AUDIOHD_WIDCAP_PWRCTRL) {
+					(void) audioha_codec_verb_get(statep,
+					    codec->index, wid,
+					    AUDIOHDC_VERB_SET_POWER_STATE,
+					    AUDIOHD_PW_D0);
+				}
+			}
+		}
 	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audiohd_reset_stream() */
-
+}
 /*
- * audiohd_preset_rbuf()
- *
+ * audiohd_restore_path()
  * Description:
- *	Fill out entries of stream descriptor list for
- *	recording.
+ * 	This routine is used to restore the path on the codec.
  */
 static void
-audiohd_preset_rbuf(audiohd_state_t *statep)
+audiohd_restore_path(audiohd_state_t *statep)
 {
-	sd_bdle_t	*entry;
-	uint64_t	buf_phys_addr;
-	int		i;
-
-	entry = (sd_bdle_t *)(statep->hda_dma_record_bd.ad_vaddr);
-	buf_phys_addr = statep->hda_dma_record_buf.ad_paddr;
+	int			i;
+	hda_codec_t		*codec;
 
-	for (i = 0; i < AUDIOHD_BDLE_NUMS; i++) {
-		entry->sbde_addr = buf_phys_addr;
-		entry->sbde_len = statep->hda_rbuf_size;
-		entry->sbde_ioc = 1;
-		buf_phys_addr += statep->hda_rbuf_size;
-		entry++;
+	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
+		codec = statep->codec[i];
+		if (!codec)
+			continue;
+		audiohd_finish_output_path(statep->codec[i]);
+		audiohd_finish_input_path(statep->codec[i]);
+		audiohd_finish_monitor_path(statep->codec[i]);
 	}
-
-	(void) ddi_dma_sync(statep->hda_dma_record_bd.ad_dmahdl, 0,
-	    sizeof (sd_bdle_t) * AUDIOHD_BDLE_NUMS, DDI_DMA_SYNC_FORDEV);
-
-}	/* audiohd_preset_rbuf() */
+}
 
 /*
- * audiohd_fill_pbuf()
- *
- * Description:
- *	Get pending audio data, and fill out entries of stream	descriptor
- *	list for playback. This routine returns	AUDIO_FAILURE if it doesn't
- *	fill any audio samples to DMA buffers. This can happen:
- *		1) when mixer cannot provide any samples;
- *		2) playback has been stopped during fetching samples;
+ * restore_play_and_record()
  */
-static int
-audiohd_fill_pbuf(audiohd_state_t *statep)
+static void
+audiohd_restore_play_and_record(audiohd_state_t *statep)
 {
-	uint64_t	 buf_phys_addr;
-	sd_bdle_t	*entry;
-	char		*buf;
-	int		samples;
-	int		rs;
 	int		i;
+	audiohd_port_t	*port;
 
-	entry = (sd_bdle_t *)(statep->hda_dma_play_bd.ad_vaddr);
-	buf = (char *)(statep->hda_dma_play_buf.ad_vaddr);
-	buf_phys_addr = statep->hda_dma_play_buf.ad_paddr;
-
-	/* assume that 2-channel, 16-bit */
-	samples = statep->hda_pbuf_size * AUDIOHD_BDLE_NUMS / 2;
-
-	mutex_exit(&statep->hda_mutex);
-	rs = am_get_audio(statep->hda_ahandle, buf,
-	    AUDIO_NO_CHANNEL, samples);
 	mutex_enter(&statep->hda_mutex);
+	for (i = 0; i < PORT_MAX; i++) {
+		port = statep->port[i];
+		if (port == NULL)
+			continue;
+		if (port != NULL)
+			audio_engine_reset(port->engine);
+		if (port->triggered) {
+			(void) audiohd_reset_port(port);
+			audiohd_start_port(port);
+		} else {
+			audiohd_stop_port(port);
 
-	if (rs == 0) {
-		/*
-		 * If system is busy so that no sample is available, an
-		 * AUDIO_PLAY_EMPTY flag is set. And we will try to get
-		 * sample to refill play buffer when we receive an interrupt.
-		 * Only if we still cannot get sample then, we could stop
-		 * the playback engine.
-		 */
-		statep->hda_flags |= AUDIOHD_PLAY_EMPTY;
-
-	} else if (rs < 0) {
-		/*
-		 * Unknown error occurs, we have to stop playback.
-		 */
-		audio_sup_log(statep->hda_ahandle, CE_WARN,
-		    "!fill_pbuf() failed to get play sample");
-		statep->hda_flags &= ~AUDIOHD_PLAY_STARTED;
-		return (AUDIO_FAILURE);
-	} else {
-
-		(void) ddi_dma_sync(statep->hda_dma_play_buf.ad_dmahdl,
-		    0, rs << 2, DDI_DMA_SYNC_FORDEV);
-	}
-
-	/*
-	 * Because users can quickly start and stop audio streams, it
-	 * is possible that playback already stopped before we re-grab
-	 * mutex. In this case, we dispose fetched samples, and return
-	 * AUDIO_FAILURE as we didn't get samples.
-	 */
-	if ((statep->hda_flags & AUDIOHD_PLAY_STARTED) == 0) {
-		return (AUDIO_FAILURE);
-	}
-
-	for (i = 0; i < AUDIOHD_BDLE_NUMS; i++) {
-		entry->sbde_addr = buf_phys_addr;
-		entry->sbde_len = statep->hda_pbuf_size;
-		entry->sbde_ioc = 1;
-		buf_phys_addr += statep->hda_pbuf_size;
-		entry++;
+		}
 	}
-
-	(void) ddi_dma_sync(statep->hda_dma_play_bd.ad_dmahdl, 0,
-	    sizeof (sd_bdle_t) * AUDIOHD_BDLE_NUMS, DDI_DMA_SYNC_FORDEV);
-
-	if (rs == samples)
-		statep->hda_pbuf_pos = 0;
-	else
-		statep->hda_pbuf_pos = (rs << 1);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audiohd_fill_pbuf() */
-
+	mutex_exit(&statep->hda_mutex);
+}
 /*
- * audiohd_refill_pbuf()
- *
+ * audiohd_reset_pins_ur_cap()
  * Description:
- *	Called by interrupt handler.
+ * 	Enable the unsolicited response of the pins which have the unsolicited
+ * 	response capability
  */
 static void
-audiohd_refill_pbuf(audiohd_state_t *statep)
+audiohd_reset_pins_ur_cap(audiohd_state_t *statep)
 {
-	int		rs;
-	int		pos;
-	char		*buf;
-	uint32_t	len;
-	uint32_t	regbase = statep->hda_play_regbase;
-
-	buf = (char *)(statep->hda_dma_play_buf.ad_vaddr);
-	buf += statep->hda_pbuf_pos;
-	pos = AUDIOHD_REG_GET32(regbase + AUDIOHD_SDREG_OFFSET_LPIB);
-	pos &= AUDIOHD_POS_MASK;
+	hda_codec_t		*codec;
+	audiohd_pin_t		*pin;
+	audiohd_widget_t	*widget;
+	uint32_t		urctrl;
+	int			i;
 
-	if (pos > statep->hda_pbuf_pos) {
-		len = (pos - statep->hda_pbuf_pos) & AUDIOHD_POS_MASK;
-	} else {
-		len = statep->hda_pbuf_size * AUDIOHD_BDLE_NUMS -
-		    statep->hda_pbuf_pos;
-		len &= AUDIOHD_POS_MASK;
+	for (i = 0; i <= AUDIOHD_CODEC_MAX; i++) {
+		codec = statep->codec[i];
+		if (!codec)
+			continue;
+		pin = codec->first_pin;
+		while (pin) {
+			/* enable the unsolicited response of the pin */
+			widget = codec->widget[pin->wid];
+			if ((widget->widget_cap &
+			    (AUDIOHD_URCAP_MASK) &&
+			    (pin->cap & AUDIOHD_DTCCAP_MASK)) &&
+			    ((pin->device == DTYPE_LINEOUT) ||
+			    (pin->device == DTYPE_SPDIF_OUT) ||
+			    (pin->device == DTYPE_HP_OUT) ||
+			    (pin->device == DTYPE_MIC_IN))) {
+				urctrl = (uint8_t)(1 <<
+				    (AUDIOHD_UR_ENABLE_OFF - 1));
+				urctrl |= (pin->wid & AUDIOHD_UR_TAG_MASK);
+				(void) audioha_codec_verb_get(statep,
+				    codec->index,
+				    pin->wid,
+				    AUDIOHDC_VERB_SET_URCTRL, urctrl);
+			}
+			pin = pin->next;
+		}
 	}
-	mutex_exit(&statep->hda_mutex);
-	rs = am_get_audio(statep->hda_ahandle, buf,
-	    AUDIO_NO_CHANNEL, len / 2);
-	mutex_enter(&statep->hda_mutex);
+}
+static void
+audiohd_restore_codec_gpio(audiohd_state_t *statep)
+{
+	int		i;
+	wid_t		wid;
+	hda_codec_t	*codec;
+
+	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
+		codec = statep->codec[i];
+		if (codec == NULL)
+			continue;
+		wid = codec->wid_afg;
 
-	(void) ddi_dma_sync(statep->hda_dma_play_buf.ad_dmahdl,
-	    statep->hda_pbuf_pos, rs << 2, DDI_DMA_SYNC_FORDEV);
+		/* power-up audio function group */
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_POWER_STATE, 0);
 
-	if (rs > 0) {
-		statep->hda_flags &= ~AUDIOHD_PLAY_EMPTY;
-		statep->hda_pbuf_pos += (rs << 1);
-		if (statep->hda_pbuf_pos >= statep->hda_pbuf_size *
-		    AUDIOHD_BDLE_NUMS)
-			statep->hda_pbuf_pos = 0;
-		return;
+		/* work around for Sony VAIO laptop with specific codec */
+		if ((codec->vid == AUDIOHD_CODECID_SONY1) ||
+		    (codec->vid == AUDIOHD_CODECID_SONY2))
+			continue;
+		/*
+		 * GPIO controls which are laptop specific workarounds and
+		 * might be changed. Some laptops use GPIO, so we need to
+		 * enable and set the GPIO correctly.
+		 */
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_MASK, AUDIOHDC_GPIO_ENABLE);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_DIREC, statep->gpio_direct);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_STCK, AUDIOHDC_GPIO_DATA_CTRL);
+		(void) audioha_codec_verb_get(statep, i, wid,
+		    AUDIOHDC_VERB_SET_GPIO_DATA, AUDIOHDC_GPIO_STCK_CTRL);
 	}
+}
+/*
+ * audiohd_resume()
+ */
+static int
+audiohd_resume(audiohd_state_t *statep)
+{
+	uint8_t		rirbsts;
 
-	/* We didn't get any sample */
-	if ((statep->hda_flags & AUDIOHD_PLAY_EMPTY) == 0)
-		statep->hda_flags |= AUDIOHD_PLAY_EMPTY;
-	else {
-		/* once again, we don't get samples, stop it */
-		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_CTL, 0);
-		statep->hda_flags &= ~(AUDIOHD_PLAY_EMPTY |
-		    AUDIOHD_PLAY_STARTED);
-
-		/* shutdown the mixer */
+	mutex_enter(&statep->hda_mutex);
+	statep->suspended = B_FALSE;
+	/* Restore the hda state */
+	if (audiohd_reinit_hda(statep) == AUDIO_FAILURE) {
+		audio_dev_warn(statep->adev,
+		    "hda reinit failed");
 		mutex_exit(&statep->hda_mutex);
-		am_play_shutdown(statep->hda_ahandle, NULL);
-		mutex_enter(&statep->hda_mutex);
+		return (DDI_SUCCESS);
 	}
+	/* reset to enable the capability of unsolicited response for pin */
+	audiohd_reset_pins_ur_cap(statep);
+	/* Enable interrupt */
+	AUDIOHD_REG_SET32(AUDIOHD_REG_INTCTL,
+	    AUDIOHD_INTCTL_BIT_GIE |
+	    AUDIOHD_INTCTL_BIT_SIE);
+	/* clear the unsolicited response interrupt */
+	rirbsts = AUDIOHD_REG_GET8(AUDIOHD_REG_RIRBSTS);
+	AUDIOHD_REG_SET8(AUDIOHD_REG_RIRBSTS, rirbsts);
+	mutex_exit(&statep->hda_mutex);
 
-}	/* audiohd_refill_pbuf() */
+	audiohd_restore_play_and_record(statep);
+	audiohd_configure_output(statep);
+	audiohd_restore_volume(statep);
+	audiohd_configure_input(statep);
 
+	/* set widget power to D0 */
+	audiohd_change_widget_power_state(statep, AUDIOHD_PW_ON);
+
+	return (DDI_SUCCESS);
+}	/* audiohd_resume */
 
 /*
- * audiohd_get_rbuf()
- *
- * Description:
- *	Called by interrupt handler.
+ * audiohd_suspend()
  */
-static void
-audiohd_get_rbuf(audiohd_state_t *statep)
+static int
+audiohd_suspend(audiohd_state_t *statep)
 {
-	uint32_t	regbase;
-	uint32_t	len;
-	char		*buf;
-	int		pos;
+	mutex_enter(&statep->hda_mutex);
+	statep->suspended = B_TRUE;
 
-	regbase = statep->hda_record_regbase;
-	pos = AUDIOHD_REG_GET32(regbase + AUDIOHD_SDREG_OFFSET_LPIB);
-	pos &= AUDIOHD_POS_MASK;
-	buf = (char *)statep->hda_dma_record_buf.ad_vaddr;
-	buf += statep->hda_rbuf_pos;
+	/* set widget power to D2 */
+	audiohd_change_widget_power_state(statep, AUDIOHD_PW_OFF);
+	/* Disable h/w */
+	audiohd_disable_intr(statep);
+	audiohd_stop_dma(statep);
+	mutex_exit(&statep->hda_mutex);
 
-	if (pos > statep->hda_rbuf_pos) {
-		len = (pos - statep->hda_rbuf_pos) & AUDIOHD_POS_MASK;
-		mutex_exit(&statep->hda_mutex);
-		am_send_audio(statep->hda_ahandle, buf, AUDIO_NO_CHANNEL,
-		    len / 2);
-		mutex_enter(&statep->hda_mutex);
-		statep->hda_rbuf_pos += len;
-		if (statep->hda_rbuf_pos >= statep->hda_rbuf_size *
-		    AUDIOHD_BDLE_NUMS)
-			statep->hda_rbuf_pos = 0;
-	} else {
-		len = statep->hda_rbuf_size * AUDIOHD_BDLE_NUMS -
-		    statep->hda_rbuf_pos;
-		mutex_exit(&statep->hda_mutex);
-		am_send_audio(statep->hda_ahandle, buf, AUDIO_NO_CHANNEL,
-		    len / 2);
-		mutex_enter(&statep->hda_mutex);
-		statep->hda_rbuf_pos = 0;
-	}
+	return (DDI_SUCCESS);
+}	/* audiohd_suspend */
 
-}	/* audiohd_get_rbuf() */
+/*
+ * audiohd_disable_pin()
+ */
+static int
+audiohd_disable_pin(audiohd_state_t *statep, int caddr, wid_t wid)
+{
+	AUDIOHD_DISABLE_PIN_OUT(statep, caddr, wid);
+	return (AUDIO_SUCCESS);
+}
 
 /*
- * auidohd_set_gain()
- *
- * Description:
- *	Set gain value for the path
+ * audiohd_enable_pin()
  */
 static int
-audiohd_set_gain(audiohd_state_t *statep, int dir, int gain, int channel)
+audiohd_enable_pin(audiohd_state_t *statep, int caddr, wid_t wid)
 {
-	audiohd_ostream_t	*ostream;
-	audiohd_istream_t	*istream;
+	AUDIOHD_ENABLE_PIN_OUT(statep, caddr, wid);
+	return (AUDIO_SUCCESS);
+}
+/*
+ * audiohd_change_speaker_state()
+ */
+static void
+audiohd_change_speaker_state(audiohd_state_t *statep, int on)
+{
+	audiohd_path_t		*path;
+	audiohd_widget_t	*widget;
 	audiohd_pin_t		*pin;
-	hda_codec_t		*codec;
-	uint_t			tmp;
-	uint_t			caddr;
-	uint_t			val;
-	int			i;
-
-	if (gain > AUDIO_MAX_GAIN) {
-		gain = AUDIO_MAX_GAIN;
-	} else if (gain < AUDIO_MIN_GAIN) {
-		gain = AUDIO_MIN_GAIN;
-	}
-
-	codec = NULL;
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		if (statep->codec[i]) {
-			codec = statep->codec[i];
-			break;
-		}
-	}
-	if (codec == NULL)
-		return (DDI_FAILURE);
-
-	caddr = codec->index;
-	if (dir == AUDIO_PLAY) {
-		if (channel == 0) { /* left channel */
-			val = AUDIOHDC_AMP_SET_LEFT;
-			statep->hda_play_lgain = gain;
-		} else { /* right channel */
-			val = AUDIOHDC_AMP_SET_RIGHT;
-			statep->hda_play_rgain = gain;
-		}
-
-		for (ostream = codec->ostream; ostream;
-		    ostream = ostream->next_stream) {
-			if (ostream->in_use == 0)
-				break;
-
-			if (ostream->gain_wid) {
-				tmp = gain * ostream->gain_bits /
-				    AUDIO_MAX_GAIN;
-				(void) audioha_codec_4bit_verb_get(statep,
-				    caddr,
-				    ostream->gain_wid,
-				    AUDIOHDC_VERB_SET_AMP_MUTE,
-				    val | ostream->gain_dir | tmp);
-				continue;
-			}
-			/* We have to set each pin one by one */
-			for (i = 0; i < ostream->pin_nums; i++) {
-				audiohd_widget_t	*w;
-				wid_t		wid;
+	int			i, j;
+	wid_t			wid;
 
-				wid = ostream->pin_wid[i];
-				w = codec->widget[wid];
-				pin = (audiohd_pin_t *)w->priv;
-				if (pin->gain_wid) {
-					tmp = gain * pin->gain_bits /
-					    AUDIO_MAX_GAIN;
-					(void) audioha_codec_4bit_verb_get(
-					    statep, caddr,
-					    pin->gain_wid,
-					    AUDIOHDC_VERB_SET_AMP_MUTE,
-					    val | pin->gain_dir | tmp);
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (!path || path->path_type != PLAY)
+			continue;
+		if (on) {
+			for (j = 0; j < path->pin_nums; j++) {
+				wid = path->pin_wid[j];
+				widget = path->codec->widget[wid];
+				pin = (audiohd_pin_t *)widget->priv;
+				if (pin->device == DTYPE_SPEAKER) {
+					(void) audiohd_enable_pin(
+					    statep,
+					    path->codec->index,
+					    pin->wid);
 				}
-
 			}
-		}
-		return (DDI_SUCCESS);
-	}
-
-	if (dir != AUDIO_RECORD)
-		return (DDI_FAILURE);
 
-	for (istream = codec->istream; istream;
-	    istream = istream->next_stream) {
-		if (istream->in_use == 0)
-			break;
-		if (istream->gain_wid) {
-			tmp = gain * istream->gain_bits / AUDIO_MAX_GAIN;
-			(void) audioha_codec_4bit_verb_get(statep, caddr,
-			    istream->gain_wid, AUDIOHDC_VERB_SET_AMP_MUTE,
-			    val | istream->gain_dir | tmp);
 		} else {
-			for (pin = codec->first_pin; pin; pin = pin->next) {
-				if ((statep->hda_in_port |
-				    AUDIOHD_SADA_OUTPUT) ==
-				    pin->sada_porttype) {
-					tmp = gain * pin->gain_bits /
-					    AUDIO_MAX_GAIN;
-					(void) audioha_codec_4bit_verb_get(
+			for (j = 0; j < path->pin_nums; j++) {
+				wid = path->pin_wid[j];
+				widget = path->codec->widget[wid];
+				pin = (audiohd_pin_t *)widget->priv;
+				if (pin->device == DTYPE_SPEAKER) {
+					(void) audiohd_disable_pin(
 					    statep,
-					    caddr,
-					    pin->gain_wid,
-					    AUDIOHDC_VERB_SET_AMP_MUTE,
-					    val | pin->gain_dir | tmp);
+					    path->codec->index,
+					    pin->wid);
 				}
 			}
 		}
 	}
-
-	return (DDI_SUCCESS);
-
-}	/* audiohd_set_gain() */
-
+}
 /*
- * audiohd_set_port()
+ * audiohd_select_mic()
  *
  * Description:
- *
+ *	This function is used for the recording path which has a selector
+ *	as the sumwidget. We select the external MIC if it is plugged into the
+ *	MIC jack, otherwise the internal integrated MIC is selected.
  */
-static int
-audiohd_set_port(audiohd_state_t *statep, int dir, int port)
+static void
+audiohd_select_mic(audiohd_state_t *statep, uint8_t index,
+uint8_t id, int select)
 {
-	int			i;
-	uint_t			tmp_port = 0;
-	wid_t			pin_wid = 0;
-	int			caddr;
-	audiohd_istream_t	*istream;
-	audiohd_widget_t	*w;
-	audiohd_pin_t		*pin;
 	hda_codec_t		*codec;
+	audiohd_path_t		*path;
+	audiohd_widget_t	*widget, *sumwgt;
+	audiohd_pin_t		*pin;
+	int			i, j;
+	wid_t			wid;
 
-	codec =  NULL;
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		if (statep->codec[i])
-			codec = statep->codec[i];
-	}
+	codec = statep->codec[index];
 	if (codec == NULL)
-		return (DDI_FAILURE);
-
-	caddr = codec->index;
-	if (dir == AUDIO_PLAY) {
-		for (i = 0; i < AUDIOHD_PIN_NUMS-1; i++) {
-			if (port & g_outport[i]) {
-				tmp_port |= g_outport[i];
-				pin = codec->first_pin;
-				while (pin) {
-					if ((AUDIOHD_SADA_OUTPUT |
-					    g_outport[i]) ==
-					    pin->sada_porttype)
-						AUDIOHD_ENABLE_PIN_OUT(statep,
-						    caddr,
-						    pin->wid);
-					pin = pin->next;
-				}
-			} else if (statep->hda_out_ports & g_outport[i]) {
-				tmp_port &= ~(g_outport[i]);
-				pin = codec->first_pin;
-				while (pin) {
-					if ((AUDIOHD_SADA_OUTPUT |
-					    g_outport[i]) ==
-					    pin->sada_porttype)
-						AUDIOHD_DISABLE_PIN_OUT(statep,
-						    caddr,
-						    pin->wid);
-					pin = pin->next;
+		return;
+	for (i = 0; i < statep->pathnum; i++) {
+		path = statep->path[i];
+		if (path->codec != codec || path->path_type != RECORD)
+			continue;
+		sumwgt = codec->widget[path->sum_wid];
+		if (path && sumwgt &&
+		    (sumwgt->type == WTYPE_AUDIO_SEL)) {
+			for (j = 0; j < path->pin_nums; j++) {
+				wid = path->pin_wid[j];
+				widget = codec->widget[wid];
+				if (widget == NULL)
+					return;
+				pin = (audiohd_pin_t *)widget->priv;
+				if (select &&
+				    pin->device == DTYPE_MIC_IN &&
+				    pin->wid == id &&
+				    (((pin->config >>
+				    AUDIOHD_PIN_CONTP_OFF) &
+				    AUDIOHD_PIN_CONTP_MASK) ==
+				    AUDIOHD_PIN_CON_JACK)) {
+					(void) audioha_codec_verb_get(
+					    statep,
+					    index,
+					    path->sum_wid,
+					    AUDIOHDC_VERB_SET_CONN_SEL,
+					    path->sum_selconn[j]);
+					statep->port[PORT_ADC]->index =
+					    path->tag;
+					return;
+				} else if (!select &&
+				    pin->device == DTYPE_MIC_IN &&
+				    pin->wid == id &&
+				    (((pin->config >>
+				    AUDIOHD_PIN_CONTP_OFF) &
+				    AUDIOHD_PIN_CONTP_MASK) ==
+				    AUDIOHD_PIN_CON_JACK)) {
+					(void) audioha_codec_verb_get(
+					    statep,
+					    index,
+					    path->sum_wid,
+					    AUDIOHDC_VERB_SET_CONN_SEL,
+					    path->sum_selconn[j]);
+					statep->port[PORT_ADC]->index =
+					    path->tag;
+					return;
 				}
 			}
+			if (path == NULL)
+				break;
+			sumwgt = codec->widget[path->sum_wid];
 		}
-		if (port & AUDIO_SPDIF_OUT) {
-			tmp_port |= AUDIO_SPDIF_OUT;
-			pin = codec->first_pin;
-			while (pin) {
-				if ((AUDIOHD_SADA_OUTPUT | AUDIO_SPDIF_OUT) ==
-				    pin->sada_porttype)
-					AUDIOHD_ENABLE_PIN_OUT(statep, caddr,
-					    pin->wid);
-				pin = pin->next;
-			}
-
-		} else if (statep->hda_out_ports & AUDIO_SPDIF_OUT) {
-			tmp_port &= ~(AUDIO_SPDIF_OUT);
-			pin = codec->first_pin;
-			while (pin) {
-				if ((AUDIOHD_SADA_OUTPUT | AUDIO_SPDIF_OUT) ==
-				    pin->sada_porttype)
-					AUDIOHD_DISABLE_PIN_OUT(statep, caddr,
-					    pin->wid);
-				pin = pin->next;
-			}
-		}
-		statep->hda_out_ports = tmp_port;
-		return (DDI_SUCCESS);
-	}
-
-	if (dir != AUDIO_RECORD) {
-		return (DDI_FAILURE);
-	}
-
-	/* now, we handle recording */
-	switch (port) {
-	case AUDIO_NONE:
-		for (pin = codec->first_pin; pin; pin = pin->next) {
-			if ((statep->hda_in_port | AUDIOHD_SADA_INPUT) ==
-			    pin->sada_porttype)
-				AUDIOHD_DISABLE_PIN_IN(statep, caddr, pin->wid);
-		}
-		tmp_port = 0;
-		pin_wid = 0;
-		break;
-	case AUDIO_MICROPHONE:
-	case AUDIO_LINE_IN:
-	case AUDIO_CD:
-	case AUDIO_AUX1_IN:
-		for (pin = codec->first_pin; pin; pin = pin->next) {
-			if ((port | AUDIOHD_SADA_INPUT) == pin->sada_porttype) {
-				AUDIOHD_ENABLE_PIN_IN(statep, caddr, pin->wid);
-				pin_wid = pin->wid;
-			} else if ((statep->hda_in_port | AUDIOHD_SADA_INPUT) ==
-			    pin->sada_porttype) {
-				AUDIOHD_DISABLE_PIN_IN(statep, caddr, pin->wid);
-			}
-		}
-		tmp_port = port;
-		break;
-	default:
-		break;
 	}
-	statep->hda_in_port = tmp_port;
-
-	/*
-	 * During path searching, we allow one real selector/mixer
-	 * in the input path. So, if there is a multi-input selector
-	 * we must set the right input connection for the selector.
-	 */
-	if (pin_wid == 0)
-		return (DDI_SUCCESS);
-	w = codec->widget[pin_wid];
-	pin = (audiohd_pin_t *)w->priv;
-
-	/* find ADC widget */
-	w = codec->widget[pin->adc_dac_wid];
-
-	istream = (audiohd_istream_t *)w->priv;
-
 	/*
-	 * If there is a real selector in this input stream,
-	 * we select the right one input for the selector.
+	 * If the input istream > 1, we should set the record stream tag
+	 * respectively. All the input streams sharing one tag may make the
+	 * record sound distorted.
 	 */
-	if (istream->sum_wid) {
-		w = codec->widget[istream->sum_wid];
-		if (w->type == WTYPE_AUDIO_SEL) {
-			for (i = 0; i < istream->pin_nums; i++)
-				if (istream->pin_wid[i] == pin_wid)
-					break;
-			if (i > istream->pin_nums) {
-				cmn_err(CE_WARN, "bug in istream-pin");
-			} else {
-				(void) audioha_codec_verb_get(
-				    statep, codec->index,
-				    istream->sum_wid,
-				    AUDIOHDC_VERB_SET_CONN_SEL,
-				    istream->sum_selconn[i]);
+	if (codec->nistream > 1) {
+		for (i = 0; i < statep->pathnum; i++) {
+			path = statep->path[i];
+			if (!path || path->path_type != RECORD)
+				continue;
+			for (j = 0; j < path->pin_nums; j++) {
+				wid = path->pin_wid[j];
+				widget = codec->widget[wid];
+				if (widget == NULL)
+					return;
+				pin = (audiohd_pin_t *)widget->priv;
+				if (select &&
+				    pin->device == DTYPE_MIC_IN &&
+				    pin->wid == id &&
+				    (((pin->config >>
+				    AUDIOHD_PIN_CONTP_OFF) &
+				    AUDIOHD_PIN_CONTP_MASK) ==
+				    AUDIOHD_PIN_CON_JACK)) {
+					statep->port[PORT_ADC]->index =
+					    path->tag;
+					return;
+				} else if (!select &&
+				    pin->device == DTYPE_MIC_IN &&
+				    (((pin->config >>
+				    AUDIOHD_PIN_CONTP_OFF) &
+				    AUDIOHD_PIN_CONTP_MASK) ==
+				    AUDIOHD_PIN_CON_FIXED)) {
+					statep->port[PORT_ADC]->index =
+					    path->tag;
+					return;
+				}
 			}
 		}
 	}
-
-	return (DDI_SUCCESS);
-
-}	/* audiohd_set_port() */
-
+}
 /*
- * audiohd_do_mute_outputs()
+ * audiohd_pin_sense()
  *
- * Description:
- *	Mute the output path.
+ * Description
+ *
+ * 	When the earphone is plugged into the jack associtated with the pin
+ * 	complex, we disable the built in speaker. When the earphone is plugged
+ * 	out of the jack, we enable the built in speaker.
  */
 static void
-audiohd_do_mute_outputs(audiohd_state_t *statep,
-    audiohd_ostream_t *ostream, hda_codec_t *codec,
-    boolean_t mute)
+audiohd_pin_sense(audiohd_state_t *statep, uint32_t resp, uint32_t respex)
 {
-	audiohd_widget_t	*w;
+	uint8_t			index;
+	uint8_t			id;
+	uint32_t		rs;
+	audiohd_widget_t	*widget;
 	audiohd_pin_t		*pin;
-	wid_t			wid;
-	int			lgain, rgain;
-	int			j;
-	uint_t			lg, rg;
-	uint16_t		caddr = codec->index;
-
-	if (mute == 0) {
-		if (ostream->mute_wid) {
-			if (ostream->mute_wid ==
-			    ostream->gain_wid) {
-				lgain = statep->hda_play_lgain *
-				    ostream->gain_bits /
-				    AUDIO_MAX_GAIN;
-				rgain = statep->hda_play_rgain *
-				    ostream->gain_bits /
-				    AUDIO_MAX_GAIN;
-			} else {
-				lgain = AUDIOHDC_GAIN_MAX;
-				rgain = AUDIOHDC_GAIN_MAX;
-			}
-			(void) audioha_codec_4bit_verb_get(statep,
-			    caddr,
-			    ostream->mute_wid,
-			    AUDIOHDC_VERB_SET_AMP_MUTE,
-			    ostream->mute_dir |
-			    AUDIOHDC_AMP_SET_LEFT |
-			    lgain);
-			(void) audioha_codec_4bit_verb_get(statep,
-			    caddr,
-			    ostream->mute_wid,
-			    AUDIOHDC_VERB_SET_AMP_MUTE,
-			    ostream->mute_dir |
-			    AUDIOHDC_AMP_SET_RIGHT |
-			    rgain);
-		} else {
-			for (j = 0; j < ostream->pin_nums;
-			    j++) {
-				wid = ostream->pin_wid[j];
-				w = codec->widget[wid];
-				lg = statep->hda_play_lgain;
-				rg = statep->hda_play_rgain;
-				pin = (audiohd_pin_t *)w->priv;
-				if (pin->mute_wid ==
-				    pin->gain_wid) {
-					lgain =
-					    lg *
-					    pin->gain_bits /
-					    AUDIO_MAX_GAIN;
-					rgain =
-					    rg *
-					    pin->gain_bits /
-					    AUDIO_MAX_GAIN;
-				} else {
-					lgain =
-					    AUDIOHDC_GAIN_MAX;
-					rgain =
-					    AUDIOHDC_GAIN_MAX;
-				}
-				(void) audioha_codec_4bit_verb_get(
-				    statep,
-				    caddr,
-				    pin->mute_wid,
-				    AUDIOHDC_VERB_SET_AMP_MUTE,
-				    pin->mute_dir |
-				    AUDIOHDC_AMP_SET_LEFT |
-				    lgain);
-				(void) audioha_codec_4bit_verb_get(
-				    statep,
-				    caddr,
-				    pin->mute_wid,
-				    AUDIOHDC_VERB_SET_AMP_MUTE,
-				    pin->mute_dir |
-				    AUDIOHDC_AMP_SET_RIGHT |
-				    rgain);
-			}
+	hda_codec_t		*codec;
+
+	index = respex & AUDIOHD_RIRB_CODEC_MASK;
+	id = resp >> (AUDIOHD_RIRB_WID_OFF - 1);
+
+	codec = statep->codec[index];
+	if (codec == NULL)
+		return;
+	widget = codec->widget[id];
+	if (widget == NULL)
+		return;
+
+	rs = audioha_codec_verb_get(statep, index, id,
+	    AUDIOHDC_VERB_GET_PIN_SENSE, 0);
+	if (rs >> (AUDIOHD_PIN_PRES_OFF - 1) & 1) {
+		/* A MIC is plugged in, we select the MIC as input */
+		if ((widget->type == WTYPE_PIN) &&
+		    (pin = (audiohd_pin_t *)widget->priv) &&
+		    (pin->device == DTYPE_MIC_IN)) {
+			audiohd_select_mic(statep, index, id, 1);
+			return;
 		}
+		/* output pin is plugged */
+		audiohd_change_speaker_state(statep, AUDIOHD_SP_OFF);
 	} else {
-		if (ostream->mute_wid) {
-			(void) audioha_codec_4bit_verb_get(statep,
-			    caddr, ostream->mute_wid,
-			    AUDIOHDC_VERB_SET_AMP_MUTE,
-			    ostream->mute_dir |
-			    AUDIOHDC_AMP_SET_LNR |
-			    AUDIOHDC_AMP_SET_MUTE);
-		} else {
-			for (j = 0; j < ostream->pin_nums;
-			    j++) {
-			wid = ostream->pin_wid[j];
-			w = codec->widget[wid];
-			pin = (audiohd_pin_t *)w->priv;
-			(void) audioha_codec_4bit_verb_get(
-			    statep,
-			    caddr,
-			    pin->wid,
-			    AUDIOHDC_VERB_SET_AMP_MUTE,
-			    pin->mute_dir |
-			    AUDIOHDC_AMP_SET_LNR |
-			    AUDIOHDC_AMP_SET_MUTE);
-			}
+		/*
+		 * A MIC is unplugged, we select the built in MIC
+		 * as input.
+		 */
+		if ((widget->type == WTYPE_PIN) &&
+		    (pin = (audiohd_pin_t *)widget->priv) &&
+		    (pin->device == DTYPE_MIC_IN)) {
+			audiohd_select_mic(statep, index, id, 0);
+			return;
 		}
+		/* output pin is unplugged */
+		audiohd_change_speaker_state(statep, AUDIOHD_SP_ON);
 	}
-}	/* audiohd_do_mute_outputs() */
 
+}
 /*
- * audiohd_mute_outputs()
+ * audiohd_intr()
  *
- * Description:
- *	Mute all the output streams
+ * Description
+ *
+ *
+ * Arguments:
+ *	caddr_t     arg Pointer to the interrupting device's state
+ *	            structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED    Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED  Interrupt not claimed, and thus ignored
  */
-static void
-audiohd_mute_outputs(audiohd_state_t *statep, boolean_t mute)
+static uint_t
+audiohd_intr(caddr_t arg)
 {
-	audiohd_ostream_t	*ostream;
-	hda_codec_t		*codec;
-	int			i;
+	audiohd_state_t	*statep = (audiohd_state_t *)arg;
+	uint32_t	status;
+	uint32_t	regbase;
+	uint32_t	resp, respex;
+	uint8_t		sdstatus, rirbsts;
+	int		i, ret;
+	audio_engine_t	*do_adc = NULL;
+	audio_engine_t	*do_dac = NULL;
 
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		codec = statep->codec[i];
-		if (codec == NULL)
-			continue;
-		for (ostream = codec->ostream; ostream;
-		    ostream = ostream->next_stream) {
-			audiohd_do_mute_outputs(statep, ostream, codec, mute);
-		}
-	}
 
-}	/* audiohd_mute_outputs() */
+	mutex_enter(&statep->hda_mutex);
+	if (statep->suspended) {
+		mutex_exit(&statep->hda_mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
 
-/*
- * audiohd_set_pin_monitor_gain()
- *
- * Description:
- *	Set the gain value for the monitor path.
- */
-static void
-audiohd_set_pin_monitor_gain(hda_codec_t *codec, audiohd_state_t *statep,
-    uint_t caddr, audiohd_pin_t *pin, int gain)
-{
-	int 			i, k;
-	uint_t			tmp;
-	audiohd_widget_t	*widget;
+	status = AUDIOHD_REG_GET32(AUDIOHD_REG_INTSTS);
+	if (status == 0) {
+		mutex_exit(&statep->hda_mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+	AUDIOHD_REG_SET32(AUDIOHD_REG_INTSTS, status);
 
-	for (k = 0; k < pin->num; k++) {
-		tmp = gain * pin->mg_gain[k]/
-		    AUDIO_MAX_GAIN;
-		widget = codec->widget[pin->mg_wid[k]];
-		if (pin->mg_dir[k] == AUDIOHDC_AMP_SET_OUTPUT) {
-			(void) audioha_codec_4bit_verb_get(
-			    statep,
-			    caddr,
-			    pin->mg_wid[k],
-			    AUDIOHDC_VERB_SET_AMP_MUTE,
-			    AUDIOHDC_AMP_SET_RIGHT|
-			    AUDIOHDC_AMP_SET_LEFT|
-			    pin->mg_dir[k] | tmp);
-		} else if (pin->mg_dir[k] == AUDIOHDC_AMP_SET_INPUT) {
-			for (i = 0; i < widget->used; i++) {
-				(void) audioha_codec_4bit_verb_get(
-				    statep,
-				    caddr,
-				    pin->mg_wid[k],
-				    AUDIOHDC_VERB_SET_AMP_MUTE,
-				    AUDIOHDC_AMP_SET_RIGHT|
-				    AUDIOHDC_AMP_SET_LEFT|
-				    widget->selmon[i]<<
-				    AUDIOHDC_AMP_SET_INDEX_OFFSET |
-				    pin->mg_dir[k] | tmp);
+	/*
+	 * unsolicited response from pins, maybe something plugged in or out
+	 * of the jack.
+	 */
+	if (status & AUDIOHD_CIS_MASK) {
+		/* clear the unsolicited response interrupt */
+		rirbsts = AUDIOHD_REG_GET8(AUDIOHD_REG_RIRBSTS);
+		AUDIOHD_REG_SET8(AUDIOHD_REG_RIRBSTS, rirbsts);
+		/*
+		 * We have to wait and try several times to make sure the
+		 * unsolicited response is generated by our pins.
+		 * we need to make it work for audiohd spec 0.9, which is
+		 * just a draft version and requires more time to wait.
+		 */
+		for (i = 0; i < AUDIOHD_TEST_TIMES; i++) {
+			ret = audiohd_response_from_codec(statep, &resp,
+			    &respex);
+			if ((ret == AUDIO_SUCCESS) &&
+			    (respex & AUDIOHD_RIRB_UR_MASK)) {
+				/*
+				 * A pin may generate more than one ur rirb,
+				 * we only need handle one of them, and clear
+				 * the other ones
+				 */
+				statep->hda_rirb_rp =
+				    AUDIOHD_REG_GET16(AUDIOHD_REG_RIRBWP) &
+				    AUDIOHD_RIRB_WPMASK;
+				break;
 			}
 		}
+		if ((ret == AUDIO_SUCCESS) &&
+		    (respex & AUDIOHD_RIRB_UR_MASK)) {
+			audiohd_pin_sense(statep, resp, respex);
+		}
+	}
+
+	/* stream intr */
+	for (i = 0; i < statep->hda_streams_nums; i++) {
+		if ((status & (1<<i)) == 0)
+			continue;
+
+		regbase = AUDIOHD_REG_SD_BASE + AUDIOHD_REG_SD_LEN * i;
+		sdstatus = AUDIOHD_REG_GET8(regbase + AUDIOHD_SDREG_OFFSET_STS);
+
+		/* clear intrs */
+		AUDIOHD_REG_SET8(regbase + AUDIOHD_SDREG_OFFSET_STS, sdstatus);
+		if (i < statep->hda_input_streams)
+			do_adc = statep->port[PORT_ADC]->engine;
+		else
+			do_dac = statep->port[PORT_DAC]->engine;
+	}
+
+	/* update the kernel interrupt statistics */
+	if (statep->hda_ksp) {
+		((kstat_intr_t *)
+		    (statep->hda_ksp->ks_data))->intrs[KSTAT_INTR_HARD]++;
 	}
-}	/* audiohd_set_pin_monitor_gain() */
+
+	mutex_exit(&statep->hda_mutex);
+
+	if (do_adc)
+		audio_engine_produce(do_adc);
+	if (do_dac)
+		audio_engine_consume(do_dac);
+	return (DDI_INTR_CLAIMED);
+}	/* audiohd_intr() */
+
 /*
- * audiohd_set_monitor_gain()
+ * audiohd_disable_intr()
  *
  * Description:
- *	Set the gain value for the monitor widget control
+ *	Disable all possible interrupts.
  */
-static int
-audiohd_set_monitor_gain(audiohd_state_t *statep, int gain)
+static void
+audiohd_disable_intr(audiohd_state_t *statep)
 {
-	int			ret;
-	int 			i, j;
-	audiohd_ostream_t	*ostream;
-	audiohd_pin_t		*pin;
-	hda_codec_t		*codec;
-	audiohd_widget_t	*w;
-	uint_t			caddr;
-	wid_t			wid;
-
-	ASSERT(statep);
-
-	if (gain > AUDIO_MAX_GAIN)
-		gain = AUDIO_MAX_GAIN;
-	else if (gain < AUDIO_MIN_GAIN)
-		gain = AUDIO_MIN_GAIN;
+	int		i;
+	uint32_t	base;
 
-	/*
-	 * Allow to set monitor gain even if no input is selected
-	 */
-	statep->hda_monitor_gain = gain;
-	for (i = 0; i < AUDIOHD_CODEC_MAX; i++) {
-		codec = statep->codec[i];
-		if (codec == NULL)
-			continue;
-		caddr = codec->index;
-		for (ostream = codec->ostream; ostream;
-		    ostream = ostream->next_stream) {
-			for (j = 0; j < ostream->pin_nums; j++) {
-				wid = ostream->pin_wid[j];
-				w = codec->widget[wid];
-				pin = (audiohd_pin_t *)w->priv;
-				audiohd_set_pin_monitor_gain(codec, statep,
-				    caddr, pin, gain);
-			}
-		}
+	AUDIOHD_REG_SET32(AUDIOHD_REG_INTCTL, 0);
+	base = AUDIOHD_REG_SD_BASE;
+	for (i = 0; i < statep->hda_streams_nums; i++) {
+		AUDIOHD_REG_SET8(base + AUDIOHD_SDREG_OFFSET_STS,
+		    AUDIOHDR_SD_STS_INTRS);
+		base += AUDIOHD_REG_SD_LEN;
 	}
+	AUDIOHD_REG_SET32(AUDIOHD_REG_INTSTS, (uint32_t)(-1));
 
+}	/* audiohd_disable_intr() */
 
-	ret = AUDIO_SUCCESS;
-	return (ret);
-
-}	/* audiohd_set_monitor_gain() */
 
 /*
  * audiohd_12bit_verb_to_codec()
@@ -5484,9 +5468,9 @@ audioha_codec_verb_get(void *arg, uint8_t caddr, uint8_t wid,
 		return (resp);
 	}
 
-	audio_sup_log(NULL, CE_WARN, "!%s: verb_get() timeout when get "
+	audio_dev_warn(statep->adev, "timeout when get "
 	    " response from codec: wid=%d, verb=0x%04x, param=0x%04x",
-	    audiohd_name, wid, verb, param);
+	    wid, verb, param);
 
 	return ((uint32_t)(-1));
 
@@ -5524,76 +5508,10 @@ audioha_codec_4bit_verb_get(void *arg, uint8_t caddr, uint8_t wid,
 		return (resp);
 	}
 
-	audio_sup_log(NULL, CE_WARN, "!%s: 4bit_verb_get() timeout when get "
+	audio_dev_warn(statep->adev,  "timeout when get "
 	    " response from codec: wid=%d, verb=0x%04x, param=0x%04x",
-	    audiohd_name, wid, verb, param);
+	    wid, verb, param);
 
 	return ((uint32_t)(-1));
 
 }	/* audioha_codec_4bit_verb_get() */
-
-/*
- * audiohd_set_busy()
- *
- * Description:
- *	This routine is called whenever a routine needs to guarantee
- *	that it will not be suspended. It will also block any routine
- *	while a suspend is going on.
- *	CAUTION: This routine cannot be called by routines that will
- *		block. Otherwise DDI_SUSPEND will be blocked for a
- *		long time. And that is the wrong thing to do.
- *
- * Arguments:
- *	audiohd_state_t *statep		The device's state structure
- * Returns:
- *	void
- */
-static void
-audiohd_set_busy(audiohd_state_t *statep)
-{
-	ASSERT(!mutex_owned(&statep->hda_mutex));
-
-	mutex_enter(&statep->hda_mutex);
-	/* block if we are suspended */
-	while (statep->suspended)
-		cv_wait(&statep->hda_cv, &statep->hda_mutex);
-	/*
-	 * OK, we aren't suspended, so mark as busy.
-	 * This will keep us from being suspended when we release the lock.
-	 */
-	ASSERT(statep->hda_busy_cnt >= 0);
-	statep->hda_busy_cnt++;
-
-	mutex_exit(&statep->hda_mutex);
-}	/* audiohd_set_busy() */
-
-/*
- * audiohd_set_idle()
- * Description:
- *	This routine reduces the busy count. It then does a cv_broadcast()
- *	if the count is 0 so a waiting DDI_SUSPEND will continue forward.
- * Arguments:
- *	audiohd_state_t *state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiohd_set_idle(audiohd_state_t *statep)
-{
-	ASSERT(!mutex_owned(&statep->hda_mutex));
-
-	mutex_enter(&statep->hda_mutex);
-
-	ASSERT(!statep->suspended);
-
-	/* decrement the busy count */
-	ASSERT(statep->hda_busy_cnt > 0);
-	statep->hda_busy_cnt--;
-
-	/* if no longer busy, then we wake up a waiting SUSPEND */
-	if (statep->hda_busy_cnt == 0)
-		cv_broadcast(&statep->hda_cv);
-
-	mutex_exit(&statep->hda_mutex);
-}	/* audiohd_set_idle() */
diff --git a/usr/src/uts/common/io/audio/sada/drv/audiohd/audiohd.conf b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.conf
index 105a8bb513..3b66afc3af 100644
--- a/usr/src/uts/common/io/audio/sada/drv/audiohd/audiohd.conf
+++ b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.conf
@@ -19,10 +19,9 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
 #
 # Configuration file for the audiohd audio driver.
 #
@@ -30,12 +29,6 @@
 #		may change at any time.
 
 #
-# Uncomment cdrom to enable the audiohd driver to use the CDROM analog
-# input. 
-#
-# cdrom=1;
-
-#
 # play-interrupts sets the number of interrupts per second when playing.
 # This affects the resolution of various things, such as sample counts.
 # record-interrupts does the same for record interrupts.
@@ -52,10 +45,3 @@
 #
 play-interrupts=175;
 record-interrupts=175;
-
-#
-# Uncomment reset-configuration to cause the audiohd driver's state to
-# be reset to the default when the driver is loaded. Otherwise this state
-# is retained across driver unload/reload cycles, but not across reboots.
-#
-#reset-configuration=1;
diff --git a/usr/src/uts/common/sys/audio/impl/audiohd_impl.h b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.h
index d05663c0a4..3e12767149 100644
--- a/usr/src/uts/common/sys/audio/impl/audiohd_impl.h
+++ b/usr/src/uts/common/io/audio/drv/audiohd/audiohd.h
@@ -19,25 +19,12 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 #ifndef _SYS_AUDIOHD_IMPL_H_
 #define	_SYS_AUDIOHD_IMPL_H_
 
-
-#include <sys/types.h>
-#include <sys/thread.h>
-#include <sys/synch.h>
-#include <sys/kstat.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/audio.h>
-#include <sys/audio/audio_support.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/audiohd.h>
-
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -49,7 +36,24 @@ extern "C" {
 #define	AUDIOHD_VID_ATI		0x1002
 #define	AUDIOHD_VID_NVIDIA	0x10de
 
+/*
+ * specific codec id used by specific vendors
+ */
+#define	AUDIOHD_CODECID_HP	0x111d7608
+#define	AUDIOHD_CODECID_SONY1	0x10ec0260
+#define	AUDIOHD_CODECID_SONY2	0x10ec0262
+
+#define	AUDIO_SUCCESS		(0)
+#define	AUDIO_FAILURE		(-1)
+
+#define	AUDIOHD_INTS		50
+#define	AUDIOHD_MAX_INTS	1500
+#define	AUDIOHD_MIN_INTS	32
 
+#define	AUDIOHD_DEV_CONFIG	"onboard1"
+#define	AUDIOHD_DEV_VERSION	"a"
+
+#define	AUDIOHD_FMT_PCM		0x001
 /*
  * Only for Intel hardware:
  * PCI Express traffic class select register in PCI configure space
@@ -90,6 +94,7 @@ extern "C" {
 #define	AUDIOHD_SP_ON		1
 #define	AUDIOHD_SP_OFF		0
 
+#define	AUDIOHD_PORT_MAX		15
 #define	AUDIOHD_CODEC_MAX	16
 #define	AUDIOHD_MEMIO_LEN	0x4000
 
@@ -140,6 +145,9 @@ extern "C" {
 #define	AUDIOHD_PIN_VREF_L4	0x02
 #define	AUDIOHD_PIN_VREF_OFF	8
 #define	AUDIOHD_PIN_VREF_MASK	0xff
+#define	AUDIOHD_PIN_CLR_MASK		0xf
+#define	AUDIOHD_PIN_CLR_OFF		12
+
 
 #define	AUDIOHD_VERB_ADDR_OFF	28
 #define	AUDIOHD_VERB_NID_OFF	20
@@ -163,20 +171,13 @@ extern "C" {
 #define	AUDIOHD_CODEC_NUM_MASK	0x000000ff
 #define	AUDIOHD_CODEC_TYPE_MASK	0x000000ff
 
+#define	AUDIOHD_FRAGFR_ALIGN	16
 #define	AUDIOHD_BDLE_BUF_ALIGN	128
 #define	AUDIOHD_CMDIO_ENT_MASK	0x00ff	/* 256 entries for CORB/RIRB */
 #define	AUDIOHD_CDBIO_CORB_LEN	1024	/* 256 entries for CORB, 1024B */
 #define	AUDIOHD_CDBIO_RIRB_LEN	2048	/* 256 entries for RIRB, 2048B */
 #define	AUDIOHD_BDLE_NUMS	4	/* 4 entires for record/play BD list */
 
-#define	AUDIOHD_MOD_NAME	"HD Audio Driver"
-#define	AUDIOHD_IDNUM		0x6176
-#define	AUDIOHD_NAME		"SUNW,audiohd"
-#define	AUDIOHD_MINPACKET	(0)
-#define	AUDIOHD_MAXPACKET	(1024)
-#define	AUDIOHD_HIWATER		(64*1024)
-#define	AUDIOHD_LOWATER		(32*1024)
-
 #define	AUDIOHD_PORT_UNMUTE	(0xffffffff)
 
 /*
@@ -354,8 +355,8 @@ extern "C" {
 #define	AUDIOHDC_VERB_SET_EAPD			0x70c
 
 #define	AUDIOHDC_VERB_GET_DEFAULT_CONF		0xf1c
-#define	AUDIOHDC_VERB_GET_SPDIF_CONTROL		0xf0d
-#define	AUDIOHDC_VERB_SET_SPDIF_LCONTROL	0x70d
+#define	AUDIOHDC_VERB_GET_SPDIF_CTL		0xf0d
+#define	AUDIOHDC_VERB_SET_SPDIF_LCL		0x70d
 
 #define	AUDIOHDC_VERB_SET_URCTRL		0x708
 #define	AUDIOHDC_VERB_GET_PIN_SENSE		0xf09
@@ -524,22 +525,38 @@ enum audiohda_device_type {
 	DTYPE_OTHER = 0x0f,
 };
 
+enum audiohd_pin_color {
+	AUDIOHD_PIN_UNKNOWN = 0,
+	AUDIOHD_PIN_BLACK,
+	AUDIOHD_PIN_GREY,
+	AUDIOHD_PIN_BLUE,
+	AUDIOHD_PIN_GREEN,
+	AUDIOHD_PIN_RED,
+	AUDIOHD_PIN_ORANGE,
+	AUDIOHD_PIN_YELLOW,
+	AUDIOHD_PIN_PURPLE,
+	AUDIOHD_PIN_PINK,
+	AUDIOHD_PIN_WHITE = 0xe,
+	AUDIOHD_PIN_OTHER = 0xf,
+};
+
+#define	CTRL_NUM	15
+
 /* values for audiohd_widget.path_flags */
 #define	AUDIOHD_PATH_DAC	(1 << 0)
 #define	AUDIOHD_PATH_ADC	(1 << 1)
 #define	AUDIOHD_PATH_MON	(1 << 2)
 #define	AUDIOHD_PATH_NOMON	(1 << 3)
 
-
-typedef struct audiohd_ostream	audiohd_ostream_t;
-typedef struct audiohd_istream	audiohd_istream_t;
+typedef struct audiohd_path		audiohd_path_t;
 typedef struct audiohd_widget	audiohd_widget_t;
-typedef struct audiohd_mixer	audiohd_mixer_t;
 typedef struct audiohd_state	audiohd_state_t;
 typedef struct audiohd_pin	audiohd_pin_t;
 typedef struct hda_codec	hda_codec_t;
 typedef uint32_t	wid_t;		/* id of widget */
 typedef	struct audiohd_entry_prop	audiohd_entry_prop_t;
+typedef	enum audiohda_device_type	audiohda_device_type_t;
+typedef	enum audiohd_pin_color		audiohd_pin_color_t;
 
 #define	AUDIOHD_MAX_WIDGET		128
 #define	AUDIOHD_MAX_CONN		16
@@ -603,74 +620,84 @@ struct audiohd_widget {
 
 #define	AUDIOHD_MAX_MIXER		5
 #define	AUDIOHD_MAX_PIN			4
-struct audiohd_ostream {
-	wid_t	dac_wid;	/* wid of DAC widget */
-	wid_t	pin_wid[AUDIOHD_MAX_PINS];
-	int	pin_nums;
-	/*
-	 * the mixer wid in each ostream path.
-	 * this is used to build the monitor path
-	 */
-	int	mon_wid[AUDIOHD_MAX_PIN][AUDIOHD_MAX_MIXER];
-	/*
-	 * The max num of mixer widget on the path of the pin
-	 */
-	int		maxmixer[AUDIOHD_MAX_PINS];
-	int		inuse;
-	wid_t		mute_wid;	/* node used to mute this pin */
-	int		mute_dir;	/* 1: input, 2: output */
-	wid_t		gain_wid;	/* node for gain control */
-	int		gain_dir;	/* _OUTPUT/_INPUT */
-	uint32_t	gain_bits;
-	uint32_t	pin_outputs;
-	audiohd_ostream_t	*next_stream;
 
-	/*
-	 * this is used to mark the ostream which is now
-	 * used by sada.
-	 */
-	boolean_t	in_use;
-};
+#define	PORT_DAC		0
+#define	PORT_ADC		1
+#define	PORT_MAX		2
+typedef enum {
+	PLAY = 0,
+	RECORD = 1,
+} path_type_t;
 
-struct audiohd_istream {
-	wid_t	adc_wid;
+struct audiohd_path {
+	wid_t			adda_wid;
 
-	/*
-	 * wid of sum, which can be a selector or mixer. And all pin
-	 * widgets whose wid is in pin_wid[] array can route to this
-	 * sum widget. If there is no such sum, sum_wid is 0, in this
-	 * case, pin_nums must be 1, that means, this stream has only
-	 * one pin widget input. But we should remember that sum_wid
-	 * can be non-zero when pin_nums is 1.
-	 */
-	wid_t	sum_wid;
+	wid_t			pin_wid[AUDIOHD_MAX_PINS];
+	int			sum_selconn[AUDIOHD_MAX_PINS];
+	int			mon_wid[AUDIOHD_MAX_PIN][AUDIOHD_MAX_MIXER];
+	int			pin_nums;
+	int			maxmixer[AUDIOHD_MAX_PINS];
 
-	/*
-	 * the index of sum widget inputs which can go through the
-	 * path and arrive input pin widget.
-	 */
-	int		sum_selconn[AUDIOHD_MAX_PINS];
+	path_type_t		path_type;
 
-	wid_t		pin_wid[AUDIOHD_MAX_PINS];
-	int		pin_nums;
+	wid_t			mute_wid;
+	int			mute_dir;
+	wid_t			gain_wid;
+	int			gain_dir;
+	uint32_t		gain_bits;
 
-	wid_t		mute_wid;	/* node used to mute this pin */
-	int		mute_dir;	/* 1: input, 2: output */
-	wid_t		gain_wid;	/* node for gain control */
-	int		gain_dir;	/* _OUTPUT/_INPUT */
-	uint32_t	gain_bits;
+	uint32_t		pin_outputs;
+	uint8_t			tag;
 
-	audiohd_istream_t	*next_stream;
+	hda_codec_t		*codec;
 
-	/*
-	 * this is used to mark the istream which is now
-	 * used by sada.
-	 */
+	wid_t			sum_wid;
 
-	boolean_t	in_use;
-	uint8_t		rtag;
+	audiohd_state_t		*statep;
 };
 
+typedef struct audiohd_port
+{
+	uint8_t			nchan;
+	int			index;
+	uint16_t		regoff;
+	boolean_t		started;
+	boolean_t		triggered;
+
+	unsigned		fragfr;
+	unsigned		nframes;
+	uint64_t		count;
+	int			curpos;
+	int			len;
+	int			intrs;
+
+	uint_t			format;
+	unsigned		sync_dir;
+
+	ddi_dma_handle_t	samp_dmah;
+	ddi_acc_handle_t	samp_acch;
+	size_t			samp_size;
+	caddr_t			samp_kaddr;
+	uint64_t		samp_paddr;
+
+	ddi_dma_handle_t	bdl_dmah;
+	ddi_acc_handle_t	bdl_acch;
+	size_t			bdl_size;
+	caddr_t			bdl_kaddr;
+	uint64_t		bdl_paddr;
+
+	audio_engine_t		*engine;
+	audiohd_state_t		*statep;
+}audiohd_port_t;
+
+typedef struct audiohd_ctrl
+{
+	audiohd_state_t		*statep;
+	audio_ctrl_t		*ctrl;
+	uint32_t		num;
+	uint64_t		val;
+} audiohd_ctrl_t;
+
 struct audiohd_pin {
 	audiohd_pin_t	*next;
 	wid_t		wid;
@@ -682,7 +709,6 @@ struct audiohd_pin {
 
 	uint8_t		vrefvalue;	/* value of VRef */
 
-	enum audiohda_device_type	device;
 	uint32_t	cap;
 	uint32_t	config;
 	uint32_t	ctrl;
@@ -690,6 +716,8 @@ struct audiohd_pin {
 	uint32_t	seq;
 	wid_t		adc_dac_wid; /* AD/DA wid which can route to this pin */
 	int		no_phys_conn;
+	enum audiohda_device_type	device;
+
 	/*
 	 * mg_dir, mg_gain, mg_wid are used to store the monitor gain control
 	 * widget wid.
@@ -700,20 +728,15 @@ struct audiohd_pin {
 	int		num;
 	int		finish;
 
-	/* this is used for SADA driver only. we should remove it later */
-#define	AUDIOHD_SADA_OUTPUT		0x8000
-#define	AUDIOHD_SADA_INPUT		0x4000
-#define	AUDIOHD_SADA_PORTMASK		0x00ff
-	uint_t	sada_porttype;
 };
 
 typedef struct {
 	ddi_dma_handle_t	ad_dmahdl;
 	ddi_acc_handle_t	ad_acchdl;
-	caddr_t		ad_vaddr;	/* virtual addr */
-	uint64_t	ad_paddr;	/* physical addr */
-	size_t		ad_req_sz;	/* required size of memory */
-	size_t		ad_real_sz;	/* real size of memory */
+	caddr_t			ad_vaddr;	/* virtual addr */
+	uint64_t		ad_paddr;	/* physical addr */
+	size_t			ad_req_sz;	/* required size of memory */
+	size_t			ad_real_sz;	/* real size of memory */
 } audiohd_dma_t;
 
 struct hda_codec {
@@ -735,70 +758,52 @@ struct hda_codec {
 
 	/* use wid as index to the array of widget pointers */
 	audiohd_widget_t	*widget[AUDIOHD_MAX_WIDGET];
-	audiohd_ostream_t	*ostream;
-	audiohd_istream_t	*istream;
+
+
+	audiohd_port_t		*port[AUDIOHD_PORT_MAX];
+	uint8_t			portnum;
 	audiohd_pin_t		*first_pin;
 };
 
+#define	AUDIOHD_MAX_ASSOC	15
 struct audiohd_state {
-	audiohdl_t	hda_ahandle;
 	dev_info_t	*hda_dip;
 	kstat_t		*hda_ksp;
 	kmutex_t	hda_mutex;
 	uint32_t	hda_flags;
 
-	caddr_t		hda_reg_base;
-	ddi_acc_handle_t	hda_pci_handle;
-	ddi_acc_handle_t	hda_reg_handle;
-	ddi_iblock_cookie_t	hda_intr_cookie;
+	caddr_t				hda_reg_base;
+	ddi_acc_handle_t		hda_pci_handle;
+	ddi_acc_handle_t		hda_reg_handle;
+	ddi_iblock_cookie_t		hda_intr_cookie;
 
 	audiohd_dma_t	hda_dma_corb;
 	audiohd_dma_t	hda_dma_rirb;
-	audiohd_dma_t	hda_dma_play_bd;
-	audiohd_dma_t	hda_dma_play_buf;
-	audiohd_dma_t	hda_dma_record_bd;
-	audiohd_dma_t	hda_dma_record_buf;
 
-	int		hda_input_streams;	/* # of input stream */
-	int		hda_output_streams;	/* # of output stream */
-	int		hda_streams_nums;	/* # of stream */
-	int		hda_pbuf_pos;		/* play buffer position */
-	int		hda_rbuf_pos;		/* record buffer position */
+
 	uint8_t		hda_rirb_rp;		/* read pointer for rirb */
 	uint16_t	hda_codec_mask;
 
-	am_ad_info_t	hda_ad_info;
-	audio_info_t	hda_info_defaults;
-	audio_device_t	hda_dev_info;
-
-	int		hda_csamples;
-	int		hda_psamples;
-	int		hda_psample_rate;
-	int		hda_pchannels;
-	int		hda_pprecision;
-	int		hda_csample_rate;
-	int		hda_cchannels;
-	int		hda_cprecision;
+
+	audio_dev_t	*adev;
+
+
 	int		hda_pint_freq;	/* play intr frequence */
 	int		hda_rint_freq;	/* record intr frequence */
-	int		hda_pbuf_size;	/* play buffer size */
-	int		hda_rbuf_size;	/* record buffer size */
 
-	boolean_t	hda_outputs_muted;
+	int		hda_input_streams;	/* # of input stream */
+	int		hda_output_streams;	/* # of output stream */
+	int		hda_streams_nums;	/* # of stream */
+
+	uint_t		hda_play_regbase;
+	uint_t		hda_record_regbase;
+
+	uint8_t		gpio_direct;
 
-	uint_t		hda_monitor_gain;
 	uint_t		hda_play_stag;		/* tag of playback stream */
 	uint_t		hda_record_stag;	/* tag of record stream */
-	uint_t		hda_play_regbase;	/* regbase for play stream */
-	uint_t		hda_record_regbase;	/* regbase for record stream */
 	uint_t		hda_play_lgain;		/* left gain for playback */
 	uint_t		hda_play_rgain;		/* right gain for playback */
-	uint_t		hda_record_lgain;	/* left gain for recording */
-	uint_t		hda_record_rgain;	/* right gain for recording */
-	uint_t		hda_play_format;
-	uint_t		hda_record_format;
-	uint_t		hda_out_ports;		/* active outputs */
-	uint_t		hda_in_port;		/* active inputs */
 
 	/*
 	 * Now, for the time being, we add some fields
@@ -809,8 +814,27 @@ struct audiohd_state {
 	 * Suspend/Resume used fields
 	 */
 	boolean_t	suspended;
-	int		hda_busy_cnt;
-	kcondvar_t	hda_cv;
+	boolean_t	monitor_unsupported;
+
+	audiohd_path_t	*path[AUDIOHD_PORT_MAX];
+	uint8_t		pathnum;
+	audiohd_port_t	*port[PORT_MAX];
+	uint8_t		pchan;
+	uint8_t		rchan;
+
+	uint64_t	inmask;
+
+	uint_t		hda_out_ports;
+	uint_t		in_port;
+
+	/*
+	 * Controls
+	 */
+	audiohd_ctrl_t		*controls[CTRL_NUM];
+
+	/* for multichannel */
+	uint8_t			chann[AUDIOHD_MAX_ASSOC];
+	uint8_t			assoc;
 
 };
 
diff --git a/usr/src/uts/common/io/audio/drv/audioixp/audioixp.c b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.c
new file mode 100644
index 0000000000..9c8b5e8e05
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.c
@@ -0,0 +1,1675 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * audioixp Audio Driver
+ *
+ * This driver supports audio hardware integrated in ATI IXP400 chipset.
+ *
+ * The IXP400 audio core is an AC'97 controller, which has independent
+ * channels for PCM in, PCM out. The AC'97 controller is a PCI bus master
+ * with scatter/gather support. Each channel has a DMA engine. Currently,
+ * we use only the PCM in and PCM out channels. Each DMA engine uses one
+ * buffer descriptor list.  Each entry contains a pointer to a data buffer,
+ * status, length of the buffer being pointed to and the pointer to the next
+ * entry. Length of the buffer is in number of bytes. Interrupt will be
+ * triggered each time a entry is processed by hardware.
+ *
+ * System power management is not yet supported by the driver.
+ *
+ * 	NOTE:
+ * 	This driver depends on the misc/ac97 and drv/audio modules being
+ *	loaded first.
+ */
+#include <sys/types.h>
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/conf.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/pci.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include "audioixp.h"
+
+/*
+ * Module linkage routines for the kernel
+ */
+static int audioixp_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audioixp_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+static int audioixp_quiesce(dev_info_t *);
+static int audioixp_resume(dev_info_t *);
+static int audioixp_suspend(dev_info_t *);
+
+/*
+ * Entry point routine prototypes
+ */
+static int audioixp_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void audioixp_close(void *);
+static int audioixp_start(void *);
+static void audioixp_stop(void *);
+static int audioixp_format(void *);
+static int audioixp_channels(void *);
+static int audioixp_rate(void *);
+static uint64_t audioixp_count(void *);
+static void audioixp_sync(void *, unsigned);
+static size_t audioixp_qlen(void *);
+
+static audio_engine_ops_t audioixp_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	audioixp_open,
+	audioixp_close,
+	audioixp_start,
+	audioixp_stop,
+	audioixp_count,
+	audioixp_format,
+	audioixp_channels,
+	audioixp_rate,
+	audioixp_sync,
+	audioixp_qlen
+};
+
+
+/*
+ * interrupt handler
+ */
+static uint_t	audioixp_intr(caddr_t);
+
+/*
+ * Local Routine Prototypes
+ */
+static int audioixp_attach(dev_info_t *);
+static int audioixp_detach(dev_info_t *);
+static int audioixp_alloc_port(audioixp_state_t *, int);
+static void audioixp_start_port(audioixp_port_t *);
+static void audioixp_stop_port(audioixp_port_t *);
+static void audioixp_reset_port(audioixp_port_t *);
+static void audioixp_update_port(audioixp_port_t *);
+
+static int audioixp_codec_sync(audioixp_state_t *);
+static void audioixp_wr97(void *, uint8_t, uint16_t);
+static uint16_t audioixp_rd97(void *, uint8_t);
+static int audioixp_reset_ac97(audioixp_state_t *);
+static int audioixp_map_regs(audioixp_state_t *);
+static void audioixp_unmap_regs(audioixp_state_t *);
+static int audioixp_chip_init(audioixp_state_t *);
+static void audioixp_destroy(audioixp_state_t *);
+
+/*
+ * Global variables, but used only by this file.
+ */
+
+/*
+ * DDI Structures
+ */
+
+/* Device operations structure */
+static struct dev_ops audioixp_dev_ops = {
+	DEVO_REV,		/* devo_rev */
+	0,			/* devo_refcnt */
+	NULL,			/* devo_getinfo */
+	nulldev,		/* devo_identify - obsolete */
+	nulldev,		/* devo_probe */
+	audioixp_ddi_attach,	/* devo_attach */
+	audioixp_ddi_detach,	/* devo_detach */
+	nodev,			/* devo_reset */
+	NULL,			/* devi_cb_ops */
+	NULL,			/* devo_bus_ops */
+	NULL,			/* devo_power */
+	audioixp_quiesce,	/* devo_quiesce */
+};
+
+/* Linkage structure for loadable drivers */
+static struct modldrv audioixp_modldrv = {
+	&mod_driverops,		/* drv_modops */
+	IXP_MOD_NAME,		/* drv_linkinfo */
+	&audioixp_dev_ops,	/* drv_dev_ops */
+};
+
+/* Module linkage structure */
+static struct modlinkage audioixp_modlinkage = {
+	MODREV_1,			/* ml_rev */
+	(void *)&audioixp_modldrv,	/* ml_linkage */
+	NULL				/* NULL terminates the list */
+};
+
+/*
+ * device access attributes for register mapping
+ */
+static struct ddi_device_acc_attr dev_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+static struct ddi_device_acc_attr buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * DMA attributes of buffer descriptor list
+ */
+static ddi_dma_attr_t	bdlist_dma_attr = {
+	DMA_ATTR_V0,	/* version */
+	0,		/* addr_lo */
+	0xffffffff,	/* addr_hi */
+	0x0000ffff,	/* count_max */
+	8,		/* align, BDL must be aligned on a 8-byte boundary */
+	0x3c,		/* burstsize */
+	8,		/* minxfer, set to the size of a BDlist entry */
+	0x0000ffff,	/* maxxfer */
+	0x00000fff,	/* seg, set to the RAM pagesize of intel platform */
+	1,		/* sgllen, there's no scatter-gather list */
+	8,		/* granular, set to the value of minxfer */
+	0		/* flags, use virtual address */
+};
+
+/*
+ * DMA attributes of buffers to be used to receive/send audio data
+ */
+static ddi_dma_attr_t	sample_buf_dma_attr = {
+	DMA_ATTR_V0,
+	0,		/* addr_lo */
+	0xffffffff,	/* addr_hi */
+	0x0001fffe,	/* count_max */
+	4,		/* align, data buffer is aligned on a 2-byte boundary */
+	0x3c,		/* burstsize */
+	4,		/* minxfer, set to the size of a sample data */
+	0x0001ffff,	/* maxxfer */
+	0x0001ffff,	/* seg */
+	1,		/* sgllen, no scatter-gather */
+	4,		/* granular, set to the value of minxfer */
+	0,		/* flags, use virtual address */
+};
+
+/*
+ * _init()
+ *
+ * Description:
+ *	Driver initialization, called when driver is first loaded.
+ *	This is how access is initially given to all the static structures.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
+ *	mod_install() status, see mod_install(9f)
+ */
+int
+_init(void)
+{
+	int	error;
+
+	audio_init_ops(&audioixp_dev_ops, IXP_NAME);
+
+	if ((error = mod_install(&audioixp_modlinkage)) != 0) {
+		audio_fini_ops(&audioixp_dev_ops);
+	}
+
+	return (error);
+}
+
+/*
+ * _fini()
+ *
+ * Description:
+ *	Module de-initialization, called when the driver is to be unloaded.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_remove() status, see mod_remove(9f)
+ */
+int
+_fini(void)
+{
+	int		error;
+
+	if ((error = mod_remove(&audioixp_modlinkage)) != 0) {
+		return (error);
+	}
+
+	audio_fini_ops(&audioixp_dev_ops);
+
+	return (0);
+}
+
+/*
+ * _info()
+ *
+ * Description:
+ *	Module information, returns information about the driver.
+ *
+ * Arguments:
+ *	modinfo		*modinfop	Pointer to the opaque modinfo structure
+ *
+ * Returns:
+ *	mod_info() status, see mod_info(9f)
+ */
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&audioixp_modlinkage, modinfop));
+}
+
+
+/* ******************* Driver Entry Points ********************************* */
+
+/*
+ * audioixp_ddi_attach()
+ *
+ * Description:
+ *	Attach an instance of the audioixp driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audioixp_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audioixp_attach(dip));
+
+	/*
+	 * now, no suspend/resume supported. we'll do it in the future.
+	 */
+	case DDI_RESUME:
+		return (audioixp_resume(dip));
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+/*
+ * audioixp_ddi_detach()
+ *
+ * Description:
+ *	Detach an instance of the audioixp driver.
+ *
+ * Arguments:
+ *	dev_info_t		*dip	Pointer to the device's dev_info struct
+ *	ddi_detach_cmd_t	cmd	Detach command
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audioixp_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audioixp_detach(dip));
+
+	/*
+	 * now, no suspend/resume supported. we'll do it in the future.
+	 */
+	case DDI_SUSPEND:
+		return (audioixp_suspend(dip));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+static void
+audioixp_stop_dma(audioixp_state_t *statep)
+{
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT_DMA);
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN_DMA);
+}
+
+static void
+audioixp_disable_intr(audioixp_state_t *statep)
+{
+	PUT32(IXP_AUDIO_INT, GET32(IXP_AUDIO_INT));
+	PUT32(IXP_AUDIO_INT_EN, 0);
+}
+
+/*
+ * quiesce(9E) entry point.
+ *
+ * This function is called when the system is single-threaded at high
+ * PIL with preemption disabled. Therefore, this function must not be blocked.
+ *
+ * This function returns DDI_SUCCESS on success, or DDI_FAILURE on failure.
+ * DDI_FAILURE indicates an error condition and should almost never happen.
+ */
+static int
+audioixp_quiesce(dev_info_t *dip)
+{
+	audioixp_state_t		*statep;
+
+	statep = ddi_get_driver_private(dip);
+	ASSERT(statep != NULL);
+
+	/* disable HW interrupt */
+	audioixp_disable_intr(statep);
+
+	/* stop DMA engines */
+	audioixp_stop_dma(statep);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audioixp_suspend(dev_info_t *dip)
+{
+	audioixp_state_t		*statep;
+
+	statep = ddi_get_driver_private(dip);
+	ASSERT(statep != NULL);
+
+	ac97_suspend(statep->ac97);
+	mutex_enter(&statep->inst_lock);
+
+	statep->suspended = B_TRUE;
+
+	audioixp_disable_intr(statep);
+	audioixp_stop_dma(statep);
+
+	mutex_exit(&statep->inst_lock);
+	return (DDI_SUCCESS);
+}
+
+static void
+audioixp_resume_port(audioixp_port_t *port)
+{
+	if (port != NULL) {
+		if (port->engine != NULL) {
+			audio_engine_reset(port->engine);
+		}
+	}
+	audioixp_reset_port(port);
+	if (port->started) {
+		audioixp_start_port(port);
+	} else {
+		audioixp_stop_port(port);
+	}
+}
+
+static int
+audioixp_resume(dev_info_t *dip)
+{
+	audioixp_state_t		*statep;
+	audio_dev_t			*adev;
+	audioixp_port_t			*rec_port, *play_port;
+
+	statep = ddi_get_driver_private(dip);
+	adev = statep->adev;
+	ASSERT(statep != NULL);
+
+	if (audioixp_chip_init(statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "DDI_RESUME failed to init chip");
+		return (DDI_SUCCESS);
+	}
+
+	ac97_resume(statep->ac97);
+	mutex_enter(&statep->inst_lock);
+	statep->suspended = B_FALSE;
+
+	rec_port = statep->rec_port;
+	play_port = statep->play_port;
+
+	audioixp_resume_port(rec_port);
+	audioixp_resume_port(play_port);
+
+	mutex_exit(&statep->inst_lock);
+	return (DDI_SUCCESS);
+}
+/*
+ * audioixp_intr()
+ *
+ * Description:
+ *	Interrupt service routine for both play and record. For play we
+ *	get the next buffers worth of audio. For record we send it on to
+ *	the mixer.
+ *
+ *	There's a hardware pointer which indicate memory location where
+ *	the hardware is processing. We check this pointer to decide whether
+ *	to handle the buffer and how many buffers should be handled.
+ *	Refer to ATI IXP400/450 Register Reference Manual, page 193,194.
+ *
+ * Arguments:
+ *	caddr_t		arg	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+audioixp_intr(caddr_t arg)
+{
+	audioixp_state_t	*statep;
+	uint32_t		sr;
+	int			claimed = DDI_INTR_UNCLAIMED;
+
+	statep = (void *)arg;
+	mutex_enter(&statep->inst_lock);
+
+	sr = GET32(IXP_AUDIO_INT);
+
+	/* PCM in interrupt */
+	if (sr & IXP_AUDIO_INT_IN_DMA) {
+		claimed = DDI_INTR_CLAIMED;
+		PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_IN_DMA);
+	}
+
+	/* PCM out interrupt */
+	if (sr & IXP_AUDIO_INT_OUT_DMA) {
+		claimed = DDI_INTR_CLAIMED;
+		PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_OUT_DMA);
+	}
+
+	/* system is too busy to process the input stream, ignore it */
+	if (sr & IXP_AUDIO_INT_IN_DMA_OVERFLOW) {
+		claimed = DDI_INTR_CLAIMED;
+		PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_IN_DMA_OVERFLOW);
+	}
+
+	/* System is too busy, ignore it */
+	if (sr & IXP_AUDIO_INT_OUT_DMA_UNDERFLOW) {
+		claimed = DDI_INTR_CLAIMED;
+		PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_OUT_DMA_UNDERFLOW);
+	}
+
+	/* update the kernel interrupt statistics */
+	if ((claimed == DDI_INTR_CLAIMED) && statep->ksp) {
+		IXP_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	mutex_exit(&statep->inst_lock);
+
+	if (sr & IXP_AUDIO_INT_IN_DMA) {
+		audio_engine_produce(statep->rec_port->engine);
+	}
+	if (sr & IXP_AUDIO_INT_OUT_DMA) {
+		audio_engine_consume(statep->play_port->engine);
+	}
+
+	return (claimed);
+}
+
+/*
+ * audioixp_open()
+ *
+ * Description:
+ *	Opens a DMA engine for use.
+ *
+ * Arguments:
+ *	void		*arg		The DMA engine to set up
+ *	int		flag		Open flags
+ *	unsigned	*fragfrp	Receives number of frames per fragment
+ *	unsigned	*nfragsp	Receives number of fragments
+ *	caddr_t		*bufp		Receives kernel data buffer
+ *
+ * Returns:
+ *	0	on success
+ *	errno	on failure
+ */
+static int
+audioixp_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	audioixp_port_t	*port = arg;
+
+	_NOTE(ARGUNUSED(flag));
+
+	port->started = B_FALSE;
+	port->count = 0;
+	port->offset = 0;
+	*fragfrp = port->fragfr;
+	*nfragsp = IXP_BD_NUMS;
+	*bufp = port->samp_kaddr;
+
+	mutex_enter(&port->statep->inst_lock);
+	audioixp_reset_port(port);
+	mutex_exit(&port->statep->inst_lock);
+
+	return (0);
+}
+
+/*
+ * audioixp_close()
+ *
+ * Description:
+ *	Closes an audio DMA engine that was previously opened.  Since
+ *	nobody is using it, we take this opportunity to possibly power
+ *	down the entire device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to shut down
+ */
+static void
+audioixp_close(void *arg)
+{
+	audioixp_port_t		*port = arg;
+	audioixp_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	audioixp_stop_port(port);
+	port->started = B_FALSE;
+	mutex_exit(&statep->inst_lock);
+}
+
+/*
+ * audioixp_stop()
+ *
+ * Description:
+ *	This is called by the framework to stop a port that is
+ *	transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to stop
+ */
+static void
+audioixp_stop(void *arg)
+{
+	audioixp_port_t		*port = arg;
+	audioixp_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	if (port->started) {
+		audioixp_stop_port(port);
+	}
+	port->started = B_FALSE;
+	mutex_exit(&statep->inst_lock);
+}
+
+/*
+ * audioixp_start()
+ *
+ * Description:
+ *	This is called by the framework to start a port transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to start
+ *
+ * Returns:
+ *	0 	on success (never fails, errno if it did)
+ */
+static int
+audioixp_start(void *arg)
+{
+	audioixp_port_t		*port = arg;
+	audioixp_state_t	*statep = port->statep;
+
+	mutex_enter(&statep->inst_lock);
+	if (!port->started) {
+		audioixp_start_port(port);
+		port->started = B_TRUE;
+	}
+	mutex_exit(&statep->inst_lock);
+	return (0);
+}
+
+/*
+ * audioixp_format()
+ *
+ * Description:
+ *	This is called by the framework to query the format for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	AUDIO_FORMAT_S16_LE
+ */
+static int
+audioixp_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+/*
+ * audioixp_channels()
+ *
+ * Description:
+ *	This is called by the framework to query the channels for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	Number of channels for the device.
+ */
+static int
+audioixp_channels(void *arg)
+{
+	audioixp_port_t *port = arg;
+
+	return (port->nchan);
+}
+
+/*
+ * audioixp_rate()
+ *
+ * Description:
+ *	This is called by the framework to query the rate of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	48000
+ */
+static int
+audioixp_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
+}
+
+/*
+ * audioixp_count()
+ *
+ * Description:
+ *	This is called by the framework to get the engine's frame counter
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	frame count for current engine
+ */
+static uint64_t
+audioixp_count(void *arg)
+{
+	audioixp_port_t		*port = arg;
+	audioixp_state_t	*statep = port->statep;
+	uint64_t		val;
+
+	mutex_enter(&statep->inst_lock);
+	audioixp_update_port(port);
+	val = port->count;
+	mutex_exit(&statep->inst_lock);
+
+	return (val);
+}
+
+/*
+ * audioixp_sync()
+ *
+ * Description:
+ *	This is called by the framework to synchronize DMA caches.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to sync
+ */
+static void
+audioixp_sync(void *arg, unsigned nframes)
+{
+	audioixp_port_t *port = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(port->samp_dmah, 0, 0, port->sync_dir);
+}
+
+/*
+ * audioixp_qlen()
+ *
+ * Description:
+ *	This is called by the framework to determine on-device queue length.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	hardware queue length not reported by count (0 for this device)
+ */
+static size_t
+audioixp_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (0);
+}
+
+
+/* *********************** Local Routines *************************** */
+
+/*
+ * audioixp_alloc_port()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use.  It also configures the BDL lists properly
+ *	for use.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audioixp_alloc_port(audioixp_state_t *statep, int num)
+{
+	ddi_dma_cookie_t	cookie;
+	uint_t			count;
+	int			dir;
+	unsigned		caps;
+	char			*prop;
+	audio_dev_t		*adev;
+	audioixp_port_t		*port;
+	uint32_t		paddr;
+	int			rc;
+	dev_info_t		*dip;
+	audioixp_bd_entry_t	*bdentry;
+
+	adev = statep->adev;
+	dip = statep->dip;
+
+	port = kmem_zalloc(sizeof (*port), KM_SLEEP);
+	port->statep = statep;
+	port->started = B_FALSE;
+	port->num = num;
+
+	switch (num) {
+	case IXP_REC:
+		statep->rec_port = port;
+		prop = "record-interrupts";
+		dir = DDI_DMA_READ;
+		caps = ENGINE_INPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORKERNEL;
+		port->nchan = 2;
+		break;
+	case IXP_PLAY:
+		statep->play_port = port;
+		prop = "play-interrupts";
+		dir = DDI_DMA_WRITE;
+		caps = ENGINE_OUTPUT_CAP;
+		port->sync_dir = DDI_DMA_SYNC_FORDEV;
+		/* This could possibly be conditionalized */
+		port->nchan = 6;
+		break;
+	default:
+		audio_dev_warn(adev, "bad port number (%d)!", num);
+		return (DDI_FAILURE);
+	}
+
+	port->intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, prop, IXP_INTS);
+
+	/* make sure the values are good */
+	if (port->intrs < IXP_MIN_INTS) {
+		audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+		    prop, port->intrs, IXP_INTS);
+		port->intrs = IXP_INTS;
+	} else if (port->intrs > IXP_MAX_INTS) {
+		audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+		    prop, port->intrs, IXP_INTS);
+		port->intrs = IXP_INTS;
+	}
+
+	/*
+	 * Figure out how much space we need.  Sample rate is 48kHz, and
+	 * we need to store 8 chunks.  (Note that this means that low
+	 * interrupt frequencies will require more RAM.)
+	 */
+	port->fragfr = 48000 / port->intrs;
+	port->fragfr = IXP_ROUNDUP(port->fragfr, IXP_MOD_SIZE);
+	port->fragsz = port->fragfr * port->nchan * 2;
+	port->samp_size = port->fragsz * IXP_BD_NUMS;
+
+	/* allocate dma handle */
+	rc = ddi_dma_alloc_handle(dip, &sample_buf_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->samp_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	/* allocate DMA buffer */
+	rc = ddi_dma_mem_alloc(port->samp_dmah, port->samp_size, &buf_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->samp_kaddr,
+	    &port->samp_size, &port->samp_acch);
+	if (rc == DDI_FAILURE) {
+		audio_dev_warn(adev, "dma_mem_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	/* bind DMA buffer */
+	rc = ddi_dma_addr_bind_handle(port->samp_dmah, NULL,
+	    port->samp_kaddr, port->samp_size, dir|DDI_DMA_CONSISTENT,
+	    DDI_DMA_SLEEP, NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev,
+		    "ddi_dma_addr_bind_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	port->samp_paddr = cookie.dmac_address;
+
+	/*
+	 * now, from here we allocate DMA memory for buffer descriptor list.
+	 * we allocate adjacent DMA memory for all DMA engines.
+	 */
+	rc = ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
+	    NULL, &port->bdl_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * we allocate all buffer descriptors lists in continuous dma memory.
+	 */
+	port->bdl_size = sizeof (audioixp_bd_entry_t) * IXP_BD_NUMS;
+	rc = ddi_dma_mem_alloc(port->bdl_dmah, port->bdl_size,
+	    &dev_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
+	    &port->bdl_kaddr, &port->bdl_size, &port->bdl_acch);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_mem_alloc(bdlist) failed");
+		return (DDI_FAILURE);
+	}
+
+	rc = ddi_dma_addr_bind_handle(port->bdl_dmah, NULL, port->bdl_kaddr,
+	    port->bdl_size, DDI_DMA_WRITE|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP,
+	    NULL, &cookie, &count);
+	if ((rc != DDI_DMA_MAPPED) || (count != 1)) {
+		audio_dev_warn(adev, "addr_bind_handle failed");
+		return (DDI_FAILURE);
+	}
+	port->bdl_paddr = cookie.dmac_address;
+
+	/*
+	 * Wire up the BD list.
+	 */
+	paddr = port->samp_paddr;
+	bdentry = (void *)port->bdl_kaddr;
+
+	for (int i = 0; i < IXP_BD_NUMS; i++) {
+
+		/* set base address of buffer */
+		ddi_put32(port->bdl_acch, &bdentry->buf_base, paddr);
+		ddi_put16(port->bdl_acch, &bdentry->status, 0);
+		ddi_put16(port->bdl_acch, &bdentry->buf_len, port->fragsz / 4);
+		ddi_put32(port->bdl_acch, &bdentry->next, port->bdl_paddr +
+		    (((i + 1) % IXP_BD_NUMS) * sizeof (audioixp_bd_entry_t)));
+		paddr += port->fragsz;
+		bdentry++;
+	}
+	ddi_dma_sync(port->bdl_dmah, 0, 0, DDI_DMA_SYNC_FORDEV);
+
+	port->engine = audio_engine_alloc(&audioixp_engine_ops, caps);
+	if (port->engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(port->engine, port);
+	audio_dev_add_engine(adev, port->engine);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audioixp_free_port()
+ *
+ * Description:
+ *	This routine unbinds the DMA cookies, frees the DMA buffers,
+ *	deallocates the DMA handles.
+ *
+ * Arguments:
+ *	audioixp_port_t	*port	The port structure for a DMA engine.
+ */
+static void
+audioixp_free_port(audioixp_port_t *port)
+{
+	if (port == NULL)
+		return;
+
+	if (port->engine) {
+		audio_dev_remove_engine(port->statep->adev, port->engine);
+		audio_engine_free(port->engine);
+	}
+	if (port->bdl_paddr) {
+		(void) ddi_dma_unbind_handle(port->bdl_dmah);
+	}
+	if (port->bdl_acch) {
+		ddi_dma_mem_free(&port->bdl_acch);
+	}
+	if (port->bdl_dmah) {
+		ddi_dma_free_handle(&port->bdl_dmah);
+	}
+	if (port->samp_paddr) {
+		(void) ddi_dma_unbind_handle(port->samp_dmah);
+	}
+	if (port->samp_acch) {
+		ddi_dma_mem_free(&port->samp_acch);
+	}
+	if (port->samp_dmah) {
+		ddi_dma_free_handle(&port->samp_dmah);
+	}
+	kmem_free(port, sizeof (*port));
+}
+
+/*
+ * audioixp_start_port()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ * Arguments:
+ *	audioixp_port_t	*port		Port of DMA engine to start.
+ */
+static void
+audioixp_start_port(audioixp_port_t *port)
+{
+	audioixp_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+
+	if (port->num == IXP_REC) {
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN);
+	} else {
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT);
+	}
+}
+
+/*
+ * audioixp_stop_port()
+ *
+ * Description:
+ *	This routine stops the DMA engine.
+ *
+ * Arguments:
+ *	audioixp_port_t	*port		Port of DMA engine to stop.
+ */
+static void
+audioixp_stop_port(audioixp_port_t *port)
+{
+	audioixp_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	/* if suspended, then do nothing else */
+	if (statep->suspended) {
+		return;
+	}
+	if (port->num == IXP_REC) {
+		CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN);
+	} else {
+		CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT);
+	}
+}
+
+/*
+ * audioixp_reset_port()
+ *
+ * Description:
+ *	This routine resets the DMA engine pareparing it for work.
+ *
+ * Arguments:
+ *	audioixp_port_t	*port		Port of DMA engine to reset.
+ */
+static void
+audioixp_reset_port(audioixp_port_t *port)
+{
+	audioixp_state_t	*statep = port->statep;
+
+	ASSERT(mutex_owned(&statep->inst_lock));
+
+	/*
+	 * XXX: reset counters.
+	 */
+	port->count = 0;
+
+	if (statep->suspended)
+		return;
+
+	/*
+	 * Perform full reset of the engine, and enable its interrupts
+	 * but leave it turned off.
+	 */
+	if (port->num == IXP_REC) {
+		PUT32(IXP_AUDIO_FIFO_FLUSH, IXP_AUDIO_FIFO_FLUSH_IN);
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_INTER_IN);
+
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN_DMA);
+		PUT32(IXP_AUDIO_IN_DMA_LINK_P,
+		    port->bdl_paddr | IXP_AUDIO_IN_DMA_LINK_P_EN);
+
+	} else {
+		uint32_t slot = GET32(IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD);
+		PUT32(IXP_AUDIO_FIFO_FLUSH, IXP_AUDIO_FIFO_FLUSH_OUT);
+		/* clear all slots */
+		slot &= ~ (IXP_AUDIO_OUT_DMA_SLOT_3 |
+		    IXP_AUDIO_OUT_DMA_SLOT_4 |
+		    IXP_AUDIO_OUT_DMA_SLOT_5 |
+		    IXP_AUDIO_OUT_DMA_SLOT_6 |
+		    IXP_AUDIO_OUT_DMA_SLOT_7 |
+		    IXP_AUDIO_OUT_DMA_SLOT_8 |
+		    IXP_AUDIO_OUT_DMA_SLOT_9 |
+		    IXP_AUDIO_OUT_DMA_SLOT_10 |
+		    IXP_AUDIO_OUT_DMA_SLOT_11 |
+		    IXP_AUDIO_OUT_DMA_SLOT_12);
+		/* enable 6 channel out, unconditional for now */
+		slot |= IXP_AUDIO_OUT_DMA_SLOT_3 |
+		    IXP_AUDIO_OUT_DMA_SLOT_4 |
+		    IXP_AUDIO_OUT_DMA_SLOT_6 |
+		    IXP_AUDIO_OUT_DMA_SLOT_9 |
+		    IXP_AUDIO_OUT_DMA_SLOT_7 |
+		    IXP_AUDIO_OUT_DMA_SLOT_8;
+
+		PUT32(IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD, slot);
+
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_INTER_OUT);
+
+		SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT_DMA);
+		PUT32(IXP_AUDIO_OUT_DMA_LINK_P,
+		    port->bdl_paddr | IXP_AUDIO_OUT_DMA_LINK_P_EN);
+	}
+}
+
+/*
+ * audioixp_update_port()
+ *
+ * Description:
+ *	This routine updates the ports frame counter from hardware, and
+ *	gracefully handles wraps.
+ *
+ * Arguments:
+ *	audioixp_port_t	*port		The port to update.
+ */
+static void
+audioixp_update_port(audioixp_port_t *port)
+{
+	audioixp_state_t	*statep = port->statep;
+	unsigned		regoff;
+	unsigned		n;
+	int			loop;
+	uint32_t		offset;
+	uint32_t		paddr;
+
+	if (statep->suspended) {
+		return;
+	}
+	if (port->num == IXP_REC) {
+		regoff = IXP_AUDIO_IN_DMA_DT_CUR;
+	} else {
+		regoff = IXP_AUDIO_OUT_DMA_DT_CUR;
+	}
+
+	/*
+	 * Apparently it may take several tries to get an update on the
+	 * position.  Is this a hardware bug?
+	 */
+	for (loop = 100; loop; loop--) {
+		paddr = GET32(regoff);
+
+		/* make sure address is reasonable */
+		if ((paddr < port->samp_paddr) ||
+		    (paddr >= (port->samp_paddr + port->samp_size))) {
+			continue;
+		}
+
+		offset = paddr - port->samp_paddr;
+
+		if (offset >= port->offset) {
+			n = offset - port->offset;
+		} else {
+			n = offset + (port->samp_size - port->offset);
+		}
+		port->offset = offset;
+		port->count += (n / (port->nchan * sizeof (uint16_t)));
+		return;
+	}
+
+	audio_dev_warn(statep->adev, "Unable to update count (h/w bug?)");
+}
+
+
+/*
+ * audioixp_map_regs()
+ *
+ * Description:
+ *	The registers are mapped in.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		  The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audioixp_map_regs(audioixp_state_t *statep)
+{
+	dev_info_t		*dip = statep->dip;
+
+	/* map PCI config space */
+	if (pci_config_setup(statep->dip, &statep->pcih) == DDI_FAILURE) {
+		audio_dev_warn(statep->adev, "unable to map PCI config space");
+		return (DDI_FAILURE);
+	}
+
+	/* map audio mixer register */
+	if ((ddi_regs_map_setup(dip, IXP_IO_AM_REGS, &statep->regsp, 0, 0,
+	    &dev_attr, &statep->regsh)) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "unable to map audio registers");
+		return (DDI_FAILURE);
+	}
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audioixp_unmap_regs()
+ *
+ * Description:
+ *	This routine unmaps control registers.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		The device's state structure
+ */
+static void
+audioixp_unmap_regs(audioixp_state_t *statep)
+{
+	if (statep->regsh) {
+		ddi_regs_map_free(&statep->regsh);
+	}
+
+	if (statep->pcih) {
+		pci_config_teardown(&statep->pcih);
+	}
+}
+
+/*
+ * audioixp_codec_ready()
+ *
+ * Description:
+ *	This routine checks the state of codecs.  It checks the flag to confirm
+ *	that primary codec is ready.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS	 codec is ready
+ *	DDI_FAILURE	 codec is not ready
+ */
+static int
+audioixp_codec_ready(audioixp_state_t *statep)
+{
+	uint32_t	sr;
+
+	PUT32(IXP_AUDIO_INT, 0xffffffff);
+	drv_usecwait(1000);
+
+	sr = GET32(IXP_AUDIO_INT);
+	if (sr & IXP_AUDIO_INT_CODEC0_NOT_READY) {
+		PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_CODEC0_NOT_READY);
+		audio_dev_warn(statep->adev, "primary codec not ready");
+
+		return (DDI_FAILURE);
+	}
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audioixp_codec_sync()
+ *
+ * Description:
+ *	Serialize access to the AC97 audio mixer registers.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Ready for an I/O access to the codec
+ *	DDI_FAILURE		An I/O access is currently in progress, can't
+ *				perform another I/O access.
+ */
+static int
+audioixp_codec_sync(audioixp_state_t *statep)
+{
+	int 		i;
+	uint32_t	cmd;
+
+	for (i = 0; i < 300; i++) {
+		cmd = GET32(IXP_AUDIO_OUT_PHY_ADDR_DATA);
+		if (!(cmd & IXP_AUDIO_OUT_PHY_EN)) {
+			return (DDI_SUCCESS);
+		}
+		drv_usecwait(10);
+	}
+
+	audio_dev_warn(statep->adev, "unable to synchronize codec");
+	return (DDI_FAILURE);
+}
+
+/*
+ * audioixp_rd97()
+ *
+ * Description:
+ *	Get the specific AC97 Codec register.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *
+ * Returns:
+ *	Register value.
+ */
+static uint16_t
+audioixp_rd97(void *arg, uint8_t reg)
+{
+	audioixp_state_t	*statep = arg;
+	uint32_t		value;
+	uint32_t		result;
+
+	if (audioixp_codec_sync(statep) != DDI_SUCCESS)
+		return (0xffff);
+
+	value = IXP_AUDIO_OUT_PHY_PRIMARY_CODEC |
+	    IXP_AUDIO_OUT_PHY_READ |
+	    IXP_AUDIO_OUT_PHY_EN |
+	    ((unsigned)reg << IXP_AUDIO_OUT_PHY_ADDR_SHIFT);
+	PUT32(IXP_AUDIO_OUT_PHY_ADDR_DATA, value);
+
+	if (audioixp_codec_sync(statep) != DDI_SUCCESS)
+		return (0xffff);
+
+	for (int i = 0; i < 300; i++) {
+		result = GET32(IXP_AUDIO_IN_PHY_ADDR_DATA);
+		if (result & IXP_AUDIO_IN_PHY_READY)	{
+			return (result >> IXP_AUDIO_IN_PHY_DATA_SHIFT);
+		}
+		drv_usecwait(10);
+	}
+
+done:
+	audio_dev_warn(statep->adev, "time out reading codec reg %d", reg);
+	return (0xffff);
+}
+
+/*
+ * audioixp_wr97()
+ *
+ * Description:
+ *	Set the specific AC97 Codec register.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint8_t		reg		AC97 register number
+ *	uint16_t	data		The data want to be set
+ */
+static void
+audioixp_wr97(void *arg, uint8_t reg, uint16_t data)
+{
+	audioixp_state_t	*statep = arg;
+	uint32_t		value;
+
+	if (audioixp_codec_sync(statep) != DDI_SUCCESS) {
+		return;
+	}
+
+	value = IXP_AUDIO_OUT_PHY_PRIMARY_CODEC |
+	    IXP_AUDIO_OUT_PHY_WRITE |
+	    IXP_AUDIO_OUT_PHY_EN |
+	    ((unsigned)reg << IXP_AUDIO_OUT_PHY_ADDR_SHIFT) |
+	    ((unsigned)data << IXP_AUDIO_OUT_PHY_DATA_SHIFT);
+	PUT32(IXP_AUDIO_OUT_PHY_ADDR_DATA, value);
+
+	(void) audioixp_rd97(statep, reg);
+}
+
+/*
+ * audioixp_reset_ac97()
+ *
+ * Description:
+ *	Reset AC97 Codec register.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Reset the codec successfully
+ *	DDI_FAILURE		Failed to reset the codec
+ */
+static int
+audioixp_reset_ac97(audioixp_state_t *statep)
+{
+	uint32_t	cmd;
+	int i;
+
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_POWER_DOWN);
+	drv_usecwait(10);
+
+	/* register reset */
+	SET32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_AC_SOFT_RESET);
+	/* force a read to flush caches */
+	(void) GET32(IXP_AUDIO_CMD);
+
+	drv_usecwait(10);
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_AC_SOFT_RESET);
+
+	/* cold reset */
+	for (i = 0; i < 300; i++) {
+		cmd = GET32(IXP_AUDIO_CMD);
+		if (cmd & IXP_AUDIO_CMD_AC_ACTIVE) {
+			cmd |= IXP_AUDIO_CMD_AC_RESET | IXP_AUDIO_CMD_AC_SYNC;
+			PUT32(IXP_AUDIO_CMD, cmd);
+			return (DDI_SUCCESS);
+		}
+		cmd &= ~IXP_AUDIO_CMD_AC_RESET;
+		cmd |= IXP_AUDIO_CMD_AC_SYNC;
+		PUT32(IXP_AUDIO_CMD, cmd);
+		(void) GET32(IXP_AUDIO_CMD);
+		drv_usecwait(10);
+		cmd |= IXP_AUDIO_CMD_AC_RESET;
+		PUT32(IXP_AUDIO_CMD, cmd);
+		drv_usecwait(10);
+	}
+
+	audio_dev_warn(statep->adev, "AC'97 reset timed out");
+	return (DDI_FAILURE);
+}
+
+/*
+ * audioixp_chip_init()
+ *
+ * Description:
+ *	This routine initializes ATI IXP audio controller and the AC97
+ *	codec.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS	The hardware was initialized properly
+ *	DDI_FAILURE	The hardware couldn't be initialized properly
+ */
+static int
+audioixp_chip_init(audioixp_state_t *statep)
+{
+	/*
+	 * put the audio controller into quiet state, everything off
+	 */
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT_DMA);
+	CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN_DMA);
+
+	/* AC97 reset */
+	if (audioixp_reset_ac97(statep) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "AC97 codec reset failed");
+		return (DDI_FAILURE);
+	}
+
+	if (audioixp_codec_ready(statep) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "AC97 codec not ready");
+		return (DDI_FAILURE);
+	}
+
+	/* enable interrupts */
+	PUT32(IXP_AUDIO_INT, 0xffffffff);
+	PUT32(
+	    IXP_AUDIO_INT_EN,
+	    IXP_AUDIO_INT_EN_IN_DMA_OVERFLOW |
+	    IXP_AUDIO_INT_EN_STATUS |
+	    IXP_AUDIO_INT_EN_OUT_DMA_UNDERFLOW);
+	return (DDI_SUCCESS);
+
+}	/* audioixp_chip_init() */
+
+/*
+ * audioixp_attach()
+ *
+ * Description:
+ *	Attach an instance of the audioixp driver. This routine does
+ * 	the device dependent attach tasks.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audioixp_attach(dev_info_t *dip)
+{
+	uint16_t		cmdeg;
+	audioixp_state_t	*statep;
+	audio_dev_t		*adev;
+	uint32_t		devid;
+	const char		*name;
+	const char		*rev;
+
+	/* we don't support high level interrupts in the driver */
+	if (ddi_intr_hilevel(dip, 0) != 0) {
+		cmn_err(CE_WARN,
+		    "!%s%d: unsupported high level interrupt",
+		    ddi_driver_name(dip), ddi_get_instance(dip));
+		return (DDI_FAILURE);
+	}
+
+	/* allocate the soft state structure */
+	statep = kmem_zalloc(sizeof (*statep), KM_SLEEP);
+	statep->dip = dip;
+	ddi_set_driver_private(dip, statep);
+
+	if (ddi_get_iblock_cookie(dip, 0, &statep->iblock) != DDI_SUCCESS) {
+		cmn_err(CE_WARN,
+		    "!%s%d: cannot get iblock cookie",
+		    ddi_driver_name(dip), ddi_get_instance(dip));
+		kmem_free(statep, sizeof (*statep));
+		return (DDI_FAILURE);
+	}
+	mutex_init(&statep->inst_lock, NULL, MUTEX_DRIVER, statep->iblock);
+
+	/* allocate framework audio device */
+	if ((adev = audio_dev_alloc(dip, 0)) == NULL) {
+		cmn_err(CE_WARN, "!%s%d: unable to allocate audio dev",
+		    ddi_driver_name(dip), ddi_get_instance(dip));
+		goto error;
+	}
+	statep->adev = adev;
+
+	/* map in the registers */
+	if (audioixp_map_regs(statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "couldn't map registers");
+		goto error;
+	}
+
+	/* set device information -- this could be smarter */
+	devid = ((pci_config_get16(statep->pcih, PCI_CONF_VENID)) << 16) |
+	    pci_config_get16(statep->pcih, PCI_CONF_DEVID);
+
+	name = "ATI AC'97";
+	switch (devid) {
+	case IXP_PCI_ID_200:
+		rev = "IXP150";
+		break;
+	case IXP_PCI_ID_300:
+		rev = "SB300";
+		break;
+	case IXP_PCI_ID_400:
+		if (pci_config_get8(statep->pcih, PCI_CONF_REVID) & 0x80) {
+			rev = "SB450";
+		} else {
+			rev = "SB400";
+		}
+		break;
+	case IXP_PCI_ID_SB600:
+		rev = "SB600";
+		break;
+	default:
+		rev = "Unknown";
+		break;
+	}
+	audio_dev_set_description(adev, name);
+	audio_dev_set_version(adev, rev);
+
+	/* allocate port structures */
+	if ((audioixp_alloc_port(statep, IXP_PLAY) != DDI_SUCCESS) ||
+	    (audioixp_alloc_port(statep, IXP_REC) != DDI_SUCCESS)) {
+		goto error;
+	}
+
+	statep->ac97 = ac97_alloc(dip, audioixp_rd97, audioixp_wr97, statep);
+	if (statep->ac97 == NULL) {
+		audio_dev_warn(adev, "failed to allocate ac97 handle");
+		goto error;
+	}
+
+	/* set PCI command register */
+	cmdeg = pci_config_get16(statep->pcih, PCI_CONF_COMM);
+	pci_config_put16(statep->pcih, PCI_CONF_COMM,
+	    cmdeg | PCI_COMM_IO | PCI_COMM_MAE);
+
+	/* set up kernel statistics */
+	if ((statep->ksp = kstat_create(IXP_NAME, ddi_get_instance(dip),
+	    IXP_NAME, "controller", KSTAT_TYPE_INTR, 1,
+	    KSTAT_FLAG_PERSISTENT)) != NULL) {
+		kstat_install(statep->ksp);
+	}
+
+
+	if (audioixp_chip_init(statep) != DDI_SUCCESS) {
+		audio_dev_warn(statep->adev, "failed to init chip");
+		goto error;
+	}
+
+	/* initialize the AC'97 part */
+	if (ac97_init(statep->ac97, adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ac'97 initialization failed");
+		goto error;
+	}
+
+	/* set up the interrupt handler */
+	if (ddi_add_intr(dip, 0, &statep->iblock, NULL, audioixp_intr,
+	    (caddr_t)statep) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "bad interrupt specification");
+	}
+	statep->intr_added = B_TRUE;
+
+	if (audio_dev_register(adev) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "unable to register with framework");
+		goto error;
+	}
+
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	audioixp_destroy(statep);
+	return (DDI_FAILURE);
+}
+
+/*
+ * audioixp_detach()
+ *
+ * Description:
+ *	Detach an instance of the audioixp driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS	The driver was detached
+ *	DDI_FAILURE	The driver couldn't be detached
+ */
+static int
+audioixp_detach(dev_info_t *dip)
+{
+	audioixp_state_t	*statep;
+
+	statep = ddi_get_driver_private(dip);
+
+	if (audio_dev_unregister(statep->adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	audioixp_destroy(statep);
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audioixp_destroy()
+ *
+ * Description:
+ *	This routine releases all resources held by the device instance,
+ *	as part of either detach or a failure in attach.
+ *
+ * Arguments:
+ *	audioixp_state_t	*state	The device soft state.
+ */
+void
+audioixp_destroy(audioixp_state_t *statep)
+{
+	if (!statep->suspended) {
+		PUT32(IXP_AUDIO_INT, GET32(IXP_AUDIO_INT));
+		PUT32(IXP_AUDIO_INT_EN, 0);
+
+		/*
+		 * put the audio controller into quiet state, everything off
+		 */
+		CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT_DMA);
+		CLR32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN_DMA);
+	}
+
+	if (statep->intr_added) {
+		ddi_remove_intr(statep->dip, 0, statep->iblock);
+	}
+	if (statep->ksp) {
+		kstat_delete(statep->ksp);
+	}
+
+	audioixp_free_port(statep->play_port);
+	audioixp_free_port(statep->rec_port);
+
+	audioixp_unmap_regs(statep);
+
+	if (statep->ac97) {
+		ac97_free(statep->ac97);
+	}
+
+	if (statep->adev) {
+		audio_dev_free(statep->adev);
+	}
+
+	mutex_destroy(&statep->inst_lock);
+	kmem_free(statep, sizeof (*statep));
+}
diff --git a/usr/src/uts/common/io/audio/sada/drv/audioixp/audioixp.conf b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.conf
index 01efaa0243..c9390e7375 100644
--- a/usr/src/uts/common/io/audio/sada/drv/audioixp/audioixp.conf
+++ b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.conf
@@ -19,34 +19,14 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Configuration file for the audioixp audio driver.
 #
 #	WARNING: This is an UNSTABLE configuration file. Its contents
 #		may change at any time.
 
-# mixer-mode determines the mode the mixer audio personality module
-# sets for this driver when it is loaded. The mixerctl(1) command may
-# be used to change modes on the fly.
-#
-# The compat mode (mixer disabled) may not work as expected on all systems,
-# please refer to the audioixp(7D) manpage. 
-#
-#       mixer-mode=1;   <-- mixer enabled
-#       mixer-mode=0;   <-- mixer disabled, old audio(7I) behavior
-#
-
-mixer-mode=1;
-
-#
-# Uncomment cdrom to enable the audioixp driver to use the CDROM analog
-# input. 
-#
-# cdrom=1;
 
 #
 # play-interrupts sets the number of interrupts per second when playing.
@@ -65,10 +45,3 @@ mixer-mode=1;
 #
 play-interrupts=175;
 record-interrupts=175;
-
-#
-# Uncomment reset-configuration to cause the audioixp driver's state to
-# be reset to the default when the driver is loaded. Otherwise this state
-# is retained across driver unload/reload cycles, but not across reboots.
-#
-#reset-configuration=1;
diff --git a/usr/src/uts/common/io/audio/drv/audioixp/audioixp.h b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.h
new file mode 100644
index 0000000000..b71f4a866f
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audioixp/audioixp.h
@@ -0,0 +1,286 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_AUDIOIXP_H_
+#define	_AUDIOIXP_H_
+
+/*
+ * Header file for the audioixp device driver
+ */
+
+#define	IXP_DEV_CONFIG		"onboard1"
+#define	IXP_DEV_VERSION		"a"
+
+/*
+ * Driver supported configuration information
+ */
+#define	IXP_NAME		"audioixp"
+#define	IXP_MOD_NAME		"ATI IXP audio driver"
+#define	IXP_CONFIG_REGS		(0)	/* PCI configure register */
+#define	IXP_IO_AM_REGS		(1)	/* PCI base register 0x10 */
+
+#define	IXP_PLAY		0
+#define	IXP_REC			1
+
+#define	IXP_ROUNDUP(x, algn)		(((x) + ((algn) - 1)) & ~((algn) - 1))
+#define	IXP_KIOP(X)			((kstat_intr_t *)(X->ksp->ks_data))
+
+#define	IXP_BD_NUMS			(8)
+#define	IXP_MOD_SIZE			(32)
+
+
+/*
+ * PCI configuration registers and bits
+ */
+#define	IXP_PCI_ID_200		(0x10024341U)
+#define	IXP_PCI_ID_300		(0x10024361U)
+#define	IXP_PCI_ID_400		(0x10024370U)
+#define	IXP_PCI_ID_SB600	(0x10024382U)
+
+/*
+ * Audio controller registers and bits
+ */
+#define	IXP_AUDIO_INT				(0x00)
+#define	IXP_AUDIO_INT_IN_DMA_OVERFLOW		(1U<<0)
+#define	IXP_AUDIO_INT_IN_DMA			(1U<<1)
+#define	IXP_AUDIO_INT_OUT_DMA_UNDERFLOW		(1U<<2)
+#define	IXP_AUDIO_INT_OUT_DMA			(1U<<3)
+#define	IXP_AUDIO_INT_CODEC0_NOT_READY		(1U<<10)
+#define	IXP_AUDIO_INT_CODEC1_NOT_READY		(1U<<11)
+#define	IXP_AUDIO_INT_CODEC2_NOT_READY		(1U<<12)
+#define	IXP_AUDIO_INT_NEW_FRAME			(1U<<13)
+
+#define	IXP_AUDIO_INT_EN			(0x04)
+#define	IXP_AUDIO_INT_EN_IN_DMA_OVERFLOW	(1U<<0)
+#define	IXP_AUDIO_INT_EN_STATUS			(1U<<1)
+#define	IXP_AUDIO_INT_EN_OUT_DMA_UNDERFLOW	(1U<<2)
+#define	IXP_AUDIO_INT_EN_CODEC0_NOT_READY	(1U<<10)
+#define	IXP_AUDIO_INT_EN_CODEC1_NOT_READY	(1U<<11)
+#define	IXP_AUDIO_INT_EN_CODEC2_NOT_READY	(1U<<12)
+#define	IXP_AUDIO_INT_EN_NEW_FRAME		(1U<<13)
+
+#define	IXP_AUDIO_CMD				(0x08)
+#define	IXP_AUDIO_CMD_POWER_DOWN		(1U<<0)
+#define	IXP_AUDIO_CMD_EN_IN			(1U<<1)
+#define	IXP_AUDIO_CMD_EN_OUT			(1U<<2)
+#define	IXP_AUDIO_CMD_EN_IN_DMA			(1U<<8)
+#define	IXP_AUDIO_CMD_EN_OUT_DMA		(1U<<9)
+#define	IXP_AUDIO_CMD_INTER_IN			(1U<<21)
+#define	IXP_AUDIO_CMD_INTER_OUT			(1U<<22)
+#define	IXP_AUDIO_CMD_BURST_EN			(1U<<25)
+#define	IXP_AUDIO_CMD_AC_ACTIVE			(1U<<28)
+#define	IXP_AUDIO_CMD_AC_SOFT_RESET		(1U<<29)
+#define	IXP_AUDIO_CMD_AC_SYNC			(1U<<30)
+#define	IXP_AUDIO_CMD_AC_RESET			(1U<<31)
+
+#define	IXP_AUDIO_OUT_PHY_ADDR_DATA		(0x0c)
+#define	IXP_AUDIO_OUT_PHY_PRIMARY_CODEC	(0u)
+#define	IXP_AUDIO_OUT_PHY_SECOND_CODEC	(1u)
+#define	IXP_AUDIO_OUT_PHY_THIRD_CODEC	(2u)
+#define	IXP_AUDIO_OUT_PHY_READ		(1u<<2)
+#define	IXP_AUDIO_OUT_PHY_WRITE		(0u)
+#define	IXP_AUDIO_OUT_PHY_EN			(1u<<8)
+#define	IXP_AUDIO_OUT_PHY_ADDR_SHIFT		(9)
+#define	IXP_AUDIO_OUT_PHY_ADDR_MASK		(0x7fu<<9)
+#define	IXP_AUDIO_OUT_PHY_DATA_SHIFT		(16)
+#define	IXP_AUDIO_OUT_PHY_DATA_MASK		(0xffffu<<16)
+
+#define	IXP_AUDIO_IN_PHY_ADDR_DATA		(0x10)
+#define	IXP_AUDIO_IN_PHY_READY			(1u<<8)
+#define	IXP_AUDIO_IN_PHY_ADDR_SHIFT		(9)
+#define	IXP_AUDIO_IN_PHY_ADDR_MASK		(0x7fu<<9)
+#define	IXP_AUDIO_IN_PHY_DATA_SHIFT		(16)
+#define	IXP_AUDIO_IN_PHY_DATA_MASK		(0xffffu<<16)
+
+#define	IXP_AUDIO_SLOTREQ			(0x14)
+#define	IXP_AUDIO_COUNTER			(0x18)
+#define	IXP_AUDIO_IN_FIFO_THRESHOLD		(0x1c)
+#define	IXP_AUDIO_IN_DMA_LINK_P			(0x20)
+#define	IXP_AUDIO_IN_DMA_LINK_P_EN		(1u<<0)
+
+#define	IXP_AUDIO_IN_DMA_DT_START		(0x24)
+#define	IXP_AUDIO_IN_DMA_DT_NEXT		(0x28)
+#define	IXP_AUDIO_IN_DMA_DT_CUR			(0x2c)
+#define	IXP_AUDIO_IN_DT_SIZE_FIFO_INFO		(0x30)
+
+#define	IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD	(0x34)
+#define	IXP_AUDIO_OUT_DMA_SLOT_3		(1U<<0)
+#define	IXP_AUDIO_OUT_DMA_SLOT_4		(1U<<1)
+#define	IXP_AUDIO_OUT_DMA_SLOT_5		(1U<<2)
+#define	IXP_AUDIO_OUT_DMA_SLOT_6		(1U<<3)
+#define	IXP_AUDIO_OUT_DMA_SLOT_7		(1U<<4)
+#define	IXP_AUDIO_OUT_DMA_SLOT_8		(1U<<5)
+#define	IXP_AUDIO_OUT_DMA_SLOT_9		(1U<<6)
+#define	IXP_AUDIO_OUT_DMA_SLOT_10		(1U<<7)
+#define	IXP_AUDIO_OUT_DMA_SLOT_11		(1U<<8)
+#define	IXP_AUDIO_OUT_DMA_SLOT_12		(1U<<9)
+#define	IXP_AUDIO_OUT_DMA_THRESHOLD_MASK	(0x7fU<<11)
+#define	IXP_AUDIO_OUT_DMA_THRESHOLD_SHIFT	(11)
+
+#define	IXP_AUDIO_OUT_DMA_LINK_P		(0x38)
+#define	IXP_AUDIO_OUT_DMA_LINK_P_EN		(1U<<0)
+
+#define	IXP_AUDIO_OUT_DMA_DT_START		(0x3c)
+#define	IXP_AUDIO_OUT_DMA_DT_NEXT		(0x40)
+#define	IXP_AUDIO_OUT_DMA_DT_CUR		(0x44)
+#define	IXP_AUDIO_OUT_DT_SIZE_USED_FREE		(0x48)
+#define	IXP_AUDIO_SPDIF_CMD			(0x4c)
+#define	IXP_AUDIO_SPDIF_LINK_P			(0x50)
+#define	IXP_AUDIO_SPDIF_DT_START		(0x54)
+#define	IXP_AUDIO_SPDIF_DT_NEXT			(0x58)
+#define	IXP_AUDIO_SPDIF_DT_CUR			(0x5c)
+#define	IXP_AUDIO_SPDIF_DT_SIZE_FIFO_INFO	(0x60)
+#define	IXP_AUDIO_MODEM_MIRROR			(0x7c)
+#define	IXP_AUDIO_AUDIO_MIRROR			(0x80)
+#define	IXP_AUDIO_6CH_RECORDER_EN		(0x84)
+#define	IXP_AUDIO_FIFO_FLUSH		(0x88)
+#define	IXP_AUDIO_FIFO_FLUSH_OUT		(1u<<0)
+#define	IXP_AUDIO_FIFO_FLUSH_IN			(1u<<1)
+
+#define	IXP_AUDIO_OUT_FIFO_INFO		(0x8c)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG1	(0x90)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG2	(0x94)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG3	(0x98)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG4	(0x9c)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG5	(0xa0)
+#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG6	(0xa4)
+#define	IXP_AUDIO_PHY_SEMA			(0xa8)
+
+/*
+ * AC97 status and link control registers are located
+ * in PCI configuration space.
+ */
+#define	IXP_REG_GSR				0x40
+#define	IXP_REG_GCR				0x41
+
+/* AC link interface status register */
+#define	IXP_GSR_PRI_READY			0x01
+#define	IXP_GSR_SEC_READY			0x04
+#define	IXP_GSR_TRI_READY			0x10
+#define	IXP_GSR_FOUR_READY			0x20
+
+/* AC link interface control register */
+#define	IXP_GCR_ENAC97				0x80
+#define	IXP_GCR_RST				0x40
+#define	IXP_GCR_RSYNCHI				0x20
+#define	IXP_GCR_SDO				0x10
+#define	IXP_GCR_VSR				0x08
+#define	IXP_GCR_3D_AUDIO_CHANNEL		0x04
+
+/*
+ * Macro for AD1980 codec
+ */
+#define	AD1980_VID1		0x4144
+#define	AD1980_VID2		0x5370
+#define	AD1985_VID2		0x5375
+#define	CODEC_AD_REG_MISC	0x76	/* offset of ad1980 misc control reg */
+#define	AD1980_MISC_LOSEL	0x0020	/* Line-out amplifier output selector */
+#define	AD1980_MISC_HPSEL	0x0400	/* HP-out amplifier output selector */
+
+struct audioixp_port {
+	int			num;
+	struct audioixp_state	*statep;
+	ddi_dma_handle_t	samp_dmah;
+	ddi_acc_handle_t	samp_acch;
+	size_t			samp_size;
+	caddr_t			samp_kaddr;
+	uint32_t		samp_paddr;
+
+	ddi_dma_handle_t	bdl_dmah;
+	ddi_acc_handle_t	bdl_acch;
+	size_t			bdl_size;
+	caddr_t			bdl_kaddr;
+	uint32_t		bdl_paddr;
+
+	unsigned		intrs;
+	unsigned		fragfr;
+	unsigned		fragsz;
+	uint64_t		count;
+	uint32_t		offset;
+	uint8_t			nchan;
+
+	unsigned		sync_dir;
+
+	boolean_t		started;
+
+	audio_engine_t		*engine;
+};
+typedef struct audioixp_port audioixp_port_t;
+
+/*
+ * buffer descriptor list entry, see datasheet
+ */
+struct audioixp_bd_entry {
+	uint32_t	buf_base;	/* the address of the buffer */
+	uint16_t	status;		/* status of the buffer */
+	uint16_t	buf_len;	/* size of the buffer in DWORD */
+	uint32_t	next;		/* physical addr of next bd_entry */
+};
+typedef struct	audioixp_bd_entry	audioixp_bd_entry_t;
+
+/*
+ * audioixp_state_t	-per instance state and operation data
+ */
+struct audioixp_state {
+	kmutex_t		inst_lock;	/* state protection lock */
+	ddi_iblock_cookie_t	iblock;
+	dev_info_t		*dip;
+	audio_dev_t		*adev;		/* audio handle */
+	ac97_t			*ac97;
+	audioixp_port_t		*play_port;
+	audioixp_port_t		*rec_port;
+
+	ddi_acc_handle_t	pcih;		/* pci configuration space */
+	ddi_acc_handle_t	regsh;		/* for audio mixer register */
+	caddr_t			regsp;		/* base of audio mixer regs */
+
+	boolean_t		suspended;
+	boolean_t		intr_added;
+	boolean_t		swap_out;	/* swap line-out and sur-out */
+
+	kstat_t			*ksp;		/* kernel statistics */
+
+	uint32_t		ixp_codec_not_ready_bits; /* for codec detect */
+};
+typedef struct audioixp_state	 audioixp_state_t;
+
+/*
+ * Useful bit twiddlers
+ */
+#define	GET32(reg)	\
+	ddi_get32(statep->regsh, (void *)(statep->regsp + (reg)))
+
+#define	PUT32(reg, val)	\
+	ddi_put32(statep->regsh, (void *)(statep->regsp + (reg)), (val))
+
+#define	SET32(reg, val)	PUT32(reg, GET32(reg) | ((uint32_t)(val)))
+
+#define	CLR32(reg, val)	PUT32(reg, GET32(reg) & ~((uint32_t)(val)))
+
+#define	IXP_INTS		(175)	/* default interrupt rate */
+#define	IXP_MIN_INTS		(24)	/* minimum interrupt rate */
+#define	IXP_MAX_INTS		(500)	/* maximum interrupt rate */
+
+#endif /* _AUDIOIXP_H_ */
diff --git a/usr/src/uts/common/io/audio/drv/audiopci/audiopci.c b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.c
new file mode 100644
index 0000000000..22b7975535
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.c
@@ -0,0 +1,1736 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * Purpose: Creative/Ensoniq AudioPCI  driver (ES1370)
+ *
+ * This driver is used with the original Ensoniq AudioPCI.
+ */
+
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#include <sys/audio/audio_driver.h>
+#include <sys/note.h>
+#include <sys/pci.h>
+#include "audiopci.h"
+
+/*
+ * The original OSS driver used a single duplex engine and a separate
+ * playback only engine.  Instead, we expose three engines, one for input
+ * and two for output.
+ */
+
+#define	ENSONIQ_VENDOR_ID	0x1274
+#define	CREATIVE_VENDOR_ID	0x1102
+#define	ENSONIQ_ES1370		0x5000
+
+#define	DEFINTS			75
+#define	DRVNAME			"audiopci"
+
+#define	INPUT_MIC	0
+#define	INPUT_LINEIN	1
+#define	INPUT_CD	2
+#define	INPUT_VIDEO	3
+#define	INPUT_PHONE	4
+#define	INSRCS		0x1f		/* bits 0-4 */
+
+static const char *audiopci_insrcs[] = {
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_CD,
+	AUDIO_PORT_VIDEO,
+	AUDIO_PORT_PHONE,
+	NULL
+};
+
+typedef struct audiopci_port
+{
+	/* Audio parameters */
+	boolean_t		trigger;
+
+	int			speed;
+	int			fmt;
+
+	int			num;
+#define	PORT_DAC		0
+#define	PORT_SYN		1
+#define	PORT_ADC		2
+#define	PORT_MAX		PORT_ADC
+
+	caddr_t			kaddr;
+	uint32_t		paddr;
+	ddi_acc_handle_t	acch;
+	ddi_dma_handle_t	dmah;
+	unsigned		fragfr;
+	unsigned		nfrags;
+	unsigned		nframes;
+	unsigned		frameno;
+	uint64_t		count;
+
+	struct audiopci_dev	*dev;
+	audio_engine_t	*engine;
+} audiopci_port_t;
+
+typedef enum {
+	CTL_VOLUME = 0,
+	CTL_FRONT,
+	CTL_MONO,
+	CTL_MIC,
+	CTL_LINE,
+	CTL_CD,
+	CTL_VID,
+	CTL_PHONE,
+	CTL_MICBOOST,
+	CTL_RECSRC,
+	CTL_MONSRC,
+	CTL_NUM		/* must be last */
+} audiopci_ctrl_num_t;
+
+typedef struct audiopci_ctrl
+{
+	struct audiopci_dev	*dev;
+	audio_ctrl_t		*ctrl;
+	audiopci_ctrl_num_t	num;
+	uint64_t		val;
+} audiopci_ctrl_t;
+
+
+typedef struct audiopci_dev
+{
+	audio_dev_t		*adev;
+	kmutex_t		mutex;
+	uint16_t		devid;
+	dev_info_t		*dip;
+	boolean_t		enabled;
+	boolean_t		suspended;
+
+	int			pintrs;
+	int			rintrs;
+
+	uint8_t			ak_regs[0x20];
+	int			micbias;
+
+	/*
+	 * Controls
+	 */
+	audiopci_ctrl_t		controls[CTL_NUM];
+#if 0
+	audiopci_ctrl_t		*micbias;
+#endif
+
+	kstat_t			*ksp;
+
+	audiopci_port_t		port[PORT_MAX + 1];
+
+
+	caddr_t			regs;
+	ddi_acc_handle_t	acch;
+	ddi_intr_handle_t	ihandle[1];
+} audiopci_dev_t;
+
+static ddi_device_acc_attr_t acc_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static ddi_device_acc_attr_t buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * The hardware appears to be able to address up to 16-bits worth of longwords,
+ * giving a total address space of 256K.  Note, however, that we will restrict
+ * this further when we do fragment and memory allocation.  At its very highest
+ * clock rate (48 kHz) and sample size (16-bit stereo), and lowest interrupt
+ * rate (32 Hz), we only need 6000 bytes per fragment.
+ *
+ * So with an allocated buffer size of 64K, we can support at least 10 frags,
+ * which is more than enough.  (The legacy Sun driver used only 2 fragments.)
+ */
+#define	AUDIOPCI_BUF_LEN	(65536)
+
+static ddi_dma_attr_t dma_attr = {
+	DMA_ATTR_VERSION,	/* dma_attr_version */
+	0x0,			/* dma_attr_addr_lo */
+	0xffffffffU,		/* dma_attr_addr_hi */
+	0x3ffff,		/* dma_attr_count_max */
+	0x8,			/* dma_attr_align */
+	0x7f,			/* dma_attr_burstsizes */
+	0x1,			/* dma_attr_minxfer */
+	0x3ffff,		/* dma_attr_maxxfer */
+	0x3ffff,		/* dma_attr_seg */
+	0x1,			/* dma_attr_sgllen */
+	0x1,			/* dma_attr_granular */
+	0			/* dma_attr_flags */
+};
+
+#define	GET8(dev, offset)	\
+	ddi_get8(dev->acch, (uint8_t *)(dev->regs + (offset)))
+#define	GET16(dev, offset)	\
+	ddi_get16(dev->acch, (uint16_t *)(void *)(dev->regs + (offset)))
+#define	GET32(dev, offset)	\
+	ddi_get32(dev->acch, (uint32_t *)(void *)(dev->regs + (offset)))
+#define	PUT8(dev, offset, v)	\
+	ddi_put8(dev->acch, (uint8_t *)(dev->regs + (offset)), v)
+#define	PUT16(dev, offset, v)	\
+	ddi_put16(dev->acch, (uint16_t *)(void *)(dev->regs + (offset)), v)
+#define	PUT32(dev, offset, v)	\
+	ddi_put32(dev->acch, (uint32_t *)(void *)(dev->regs + (offset)), v)
+
+#define	CLR8(dev, offset, v)	PUT8(dev, offset, GET8(dev, offset) & ~(v))
+#define	SET8(dev, offset, v)	PUT8(dev, offset, GET8(dev, offset) | (v))
+#define	CLR16(dev, offset, v)	PUT16(dev, offset, GET16(dev, offset) & ~(v))
+#define	SET16(dev, offset, v)	PUT16(dev, offset, GET16(dev, offset) | (v))
+
+#define	KSINTR(dev)	((kstat_intr_t *)((dev)->ksp->ks_data))
+
+static void audiopci_init_hw(audiopci_dev_t *);
+static void audiopci_init_port(audiopci_port_t *);
+static void audiopci_start_port(audiopci_port_t *);
+static void audiopci_stop_port(audiopci_port_t *);
+static void audiopci_update_port(audiopci_port_t *);
+static uint16_t audiopci_dac_rate(int);
+static int audiopci_add_controls(audiopci_dev_t *);
+static void audiopci_del_controls(audiopci_dev_t *);
+static void audiopci_ak_write(audiopci_dev_t *, uint16_t, uint8_t);
+
+static int
+audiopci_ak_wait(audiopci_dev_t *dev, uint8_t wstat)
+{
+	for (int i = 4000; i; i--) {
+		if (!(GET8(dev, CONC_bCODECSTAT_OFF) & wstat))
+			return (DDI_SUCCESS);
+		drv_usecwait(10);
+	}
+	return (DDI_FAILURE);
+}
+
+static void
+audiopci_ak_idle(audiopci_dev_t *dev)
+{
+	for (int i = 0; i < 5; i++) {
+		if (audiopci_ak_wait(dev, CONC_CSTAT_CSTAT) == DDI_SUCCESS)
+			return;
+	}
+	audio_dev_warn(dev->adev, "timed out waiting for codec to idle");
+}
+
+static void
+audiopci_ak_write(audiopci_dev_t *dev, uint16_t addr, uint8_t data)
+{
+	uint8_t	wstat;
+
+	if (dev->suspended)
+		return;
+
+	/* shadow the value */
+	dev->ak_regs[addr] = data;
+
+	wstat = addr == CODEC_RESET_PWRD ? CONC_CSTAT_CWRIP : CONC_CSTAT_CSTAT;
+
+	/* wait for codec to be available */
+	if (audiopci_ak_wait(dev, wstat) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "timeout waiting for codec");
+	}
+
+	PUT16(dev, CONC_wCODECCTL_OFF, (addr << 8) | data);
+}
+
+static void
+audiopci_writemem(audiopci_dev_t *dev, uint32_t page, uint32_t offs,
+    uint32_t data)
+{
+	/* Select memory page */
+	PUT32(dev, CONC_bMEMPAGE_OFF, page);
+	PUT32(dev, offs, data);
+}
+
+static uint32_t
+audiopci_readmem(audiopci_dev_t *dev, uint32_t page, uint32_t offs)
+{
+	PUT32(dev, CONC_bMEMPAGE_OFF, page);	/* Select memory page */
+	return (GET32(dev, offs));
+}
+
+static uint_t
+audiopci_intr(caddr_t arg1, caddr_t arg2)
+{
+	audiopci_dev_t *dev = (void *)arg1;
+	int stats;
+	int tmp;
+	unsigned char ackbits = 0;
+	audiopci_port_t *port;
+	audio_engine_t *do_syn, *do_dac, *do_adc;
+
+	_NOTE(ARGUNUSED(arg2));
+
+	/*
+	 * NB: The old driver didn't report spurious interrupts.  On
+	 * a system with shared interrupts (typical!) there will
+	 * normally be lots of these (each time the "other" device
+	 * interrupts).
+	 *
+	 * Also, because of the way the interrupt chain handling
+	 * works, reporting of spurious interrupts is probably not
+	 * terribly useful.
+	 *
+	 * However, we can count interrupts where the master interrupt
+	 * bit is set but none of the ackbits that we are prepared to
+	 * process is set.  That is probably useful.
+	 */
+	mutex_enter(&dev->mutex);
+	if (!dev->enabled) {
+
+		mutex_exit(&dev->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	stats = GET32(dev, CONC_dSTATUS_OFF);
+
+	if (!(stats & CONC_INTSTAT_PENDING)) {	/* No interrupt pending */
+		mutex_exit(&dev->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	do_syn = do_dac = do_adc = NULL;
+
+	/* synth interrupt */
+	if (stats & CONC_INTSTAT_SYNINT) {
+
+		ackbits |= CONC_SERCTL_SYNIE;
+		port = &dev->port[PORT_SYN];
+		if (port->trigger) {
+			do_syn = port->engine;
+		}
+	}
+
+	/* DAC interrupt */
+	if (stats & CONC_INTSTAT_DACINT) {
+
+		ackbits |= CONC_SERCTL_DACIE;
+		port = &dev->port[PORT_DAC];
+		if (port->trigger) {
+			do_dac = port->engine;
+		}
+	}
+
+	/* ADC interrupt */
+	if (stats & CONC_INTSTAT_ADCINT) {
+
+		ackbits |= CONC_SERCTL_ADCIE;
+		port = &dev->port[PORT_ADC];
+
+		if (port->trigger) {
+			do_adc = port->engine;
+		}
+	}
+
+	/* UART interrupt - we shouldn't get this! */
+	if (stats & CONC_INTSTAT_UARTINT) {
+		uint8_t uart_stat = GET8(dev, CONC_bUARTCSTAT_OFF);
+		while (uart_stat & CONC_UART_RXRDY)
+			uart_stat = GET8(dev, CONC_bUARTCSTAT_OFF);
+	}
+
+	/* Ack the interrupt */
+	tmp = GET8(dev, CONC_bSERCTL_OFF);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp & (~ackbits));	/* Clear bits */
+	PUT8(dev, CONC_bSERCTL_OFF, tmp | ackbits);	/* Turn them back on */
+
+	if (dev->ksp) {
+		if (ackbits == 0) {
+			KSINTR(dev)->intrs[KSTAT_INTR_SPURIOUS]++;
+		} else {
+			KSINTR(dev)->intrs[KSTAT_INTR_HARD]++;
+		}
+	}
+
+	mutex_exit(&dev->mutex);
+
+	if (do_syn)
+		audio_engine_consume(do_syn);
+	if (do_dac)
+		audio_engine_consume(do_dac);
+	if (do_adc)
+		audio_engine_produce(do_adc);
+
+	return (DDI_INTR_CLAIMED);
+}
+
+/*
+ * Audio routines
+ */
+
+static int
+audiopci_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+static int
+audiopci_channels(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (2);
+}
+
+static int
+audiopci_rate(void *arg)
+{
+	audiopci_port_t	*port = arg;
+
+	return (port->speed);
+}
+
+static void
+audiopci_init_port(audiopci_port_t *port)
+{
+	audiopci_dev_t	*dev = port->dev;
+	unsigned tmp;
+
+	if (dev->suspended)
+		return;
+
+	switch (port->num) {
+	case PORT_DAC:
+
+		/* Set physical address of the DMA buffer */
+		audiopci_writemem(dev, CONC_DACCTL_PAGE, CONC_dDACPADDR_OFF,
+		    port->paddr);
+
+		/* Set DAC rate */
+		PUT16(dev, CONC_wDACRATE_OFF, audiopci_dac_rate(48000));
+
+		/* Set format */
+		tmp = GET8(dev, CONC_bSERFMT_OFF);
+		tmp |= CONC_PCM_DAC_16BIT;
+		tmp |= CONC_PCM_DAC_STEREO;
+
+		PUT8(dev, CONC_bSKIPC_OFF, 0x10);
+		PUT8(dev, CONC_bSERFMT_OFF, tmp);
+
+		/* Set the frame count */
+		audiopci_writemem(dev, CONC_DACCTL_PAGE, CONC_wDACFC_OFF,
+		    port->nframes - 1);
+
+		/* Set # of frames between interrupts */
+		PUT16(dev, CONC_wDACIC_OFF, port->fragfr - 1);
+
+		break;
+
+	case PORT_SYN:
+
+		/* Set physical address of the DMA buffer */
+		audiopci_writemem(dev, CONC_SYNCTL_PAGE, CONC_dSYNPADDR_OFF,
+		    port->paddr);
+
+		/* Set rate - we force to 44.1 kHz */
+		SET8(dev, CONC_bMISCCTL_OFF, CONC_MISCCTL_SYN_44KHZ);
+
+		/* Set format */
+		tmp = GET8(dev, CONC_bSERFMT_OFF);
+		tmp |= CONC_PCM_SYN_16BIT;
+		tmp |= CONC_PCM_SYN_STEREO;
+
+		PUT8(dev, CONC_bSERFMT_OFF, tmp);
+
+		/* Set the frame count */
+		audiopci_writemem(dev, CONC_SYNCTL_PAGE, CONC_wSYNFC_OFF,
+		    port->nframes - 1);
+
+		/* Set # of frames between interrupts */
+		PUT16(dev, CONC_wSYNIC_OFF, port->fragfr - 1);
+
+		break;
+
+	case PORT_ADC:
+		/* Set physical address of the DMA buffer */
+		audiopci_writemem(dev, CONC_ADCCTL_PAGE, CONC_dADCPADDR_OFF,
+		    port->paddr);
+
+		/* Set ADC rate */
+		PUT16(dev, CONC_wDACRATE_OFF, audiopci_dac_rate(48000));
+
+		/* Set format - for input we only support 16 bit input */
+		tmp = GET8(dev, CONC_bSERFMT_OFF);
+		tmp |= CONC_PCM_ADC_16BIT;
+		tmp |= CONC_PCM_ADC_STEREO;
+
+		PUT8(dev, CONC_bSKIPC_OFF, 0x10);
+
+		PUT8(dev, CONC_bSERFMT_OFF, tmp);
+
+		/* Set the frame count */
+		audiopci_writemem(dev, CONC_ADCCTL_PAGE, CONC_wADCFC_OFF,
+		    port->nframes - 1);
+
+		/* Set # of frames between interrupts */
+		PUT16(dev, CONC_wADCIC_OFF, port->fragfr - 1);
+
+		break;
+	}
+
+	port->frameno = 0;
+}
+
+static int
+audiopci_open(void *arg, int flag, unsigned *fragfrp, unsigned *nfragsp,
+    caddr_t *bufp)
+{
+	audiopci_port_t	*port = arg;
+	audiopci_dev_t	*dev = port->dev;
+	int intrs;
+
+	_NOTE(ARGUNUSED(flag));
+
+
+	mutex_enter(&dev->mutex);
+	switch (port->num) {
+	case PORT_ADC:
+		intrs = dev->rintrs;
+		break;
+	case PORT_DAC:
+		intrs = dev->pintrs;
+		break;
+	case PORT_SYN:
+		intrs = dev->pintrs;
+		break;
+	}
+
+	/* interrupt at least at 25 Hz, and not more than 250 Hz */
+	intrs = min(250, max(25, intrs));
+
+	/* NB: frame size = 4 (16-bit stereo) */
+	port->fragfr = (port->speed / intrs);
+	port->nfrags = AUDIOPCI_BUF_LEN / (port->fragfr * 4);
+	port->nfrags = max(4, min(port->nfrags, 1024));
+	port->nframes = port->nfrags * port->fragfr;
+	port->trigger = B_FALSE;
+	port->count = 0;
+
+	audiopci_init_port(port);
+
+	*fragfrp = port->fragfr;
+	*nfragsp = port->nfrags;
+	*bufp = port->kaddr;
+	mutex_exit(&dev->mutex);
+
+	return (0);
+}
+
+static void
+audiopci_start_port(audiopci_port_t *port)
+{
+	audiopci_dev_t *dev = port->dev;
+
+	if (!dev->suspended) {
+		switch (port->num) {
+		case PORT_DAC:
+			SET8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_DAC_EN);
+			SET8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_DACIE);
+			break;
+		case PORT_SYN:
+			SET8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_SYN_EN);
+			SET8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_SYNIE);
+			break;
+		case PORT_ADC:
+			SET8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_ADC_EN);
+			SET8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_ADCIE);
+			break;
+		}
+	}
+}
+
+static void
+audiopci_stop_port(audiopci_port_t *port)
+{
+	audiopci_dev_t *dev = port->dev;
+
+	if (!dev->suspended) {
+		switch (port->num) {
+		case PORT_DAC:
+			CLR8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_DAC_EN);
+			CLR8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_DACIE);
+			break;
+		case PORT_SYN:
+			CLR8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_SYN_EN);
+			CLR8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_SYNIE);
+			break;
+		case PORT_ADC:
+			CLR8(dev, CONC_bDEVCTL_OFF, CONC_DEVCTL_ADC_EN);
+			CLR8(dev, CONC_bSERCTL_OFF, CONC_SERCTL_ADCIE);
+			break;
+		}
+	}
+}
+
+static int
+audiopci_start(void *arg)
+{
+	audiopci_port_t *port = arg;
+	audiopci_dev_t *dev = port->dev;
+
+	mutex_enter(&dev->mutex);
+	if (!port->trigger) {
+		port->trigger = B_TRUE;
+		audiopci_start_port(port);
+	}
+	mutex_exit(&dev->mutex);
+
+	return (0);
+}
+
+static void
+audiopci_stop(void *arg)
+{
+	audiopci_port_t *port = arg;
+	audiopci_dev_t *dev = port->dev;
+
+	mutex_enter(&dev->mutex);
+	if (port->trigger) {
+		port->trigger = B_FALSE;
+		audiopci_stop_port(port);
+	}
+	mutex_exit(&dev->mutex);
+}
+
+static void
+audiopci_update_port(audiopci_port_t *port)
+{
+	uint32_t page, offs;
+	int frameno, n;
+
+	switch (port->num) {
+	case PORT_DAC:
+		page = CONC_DACCTL_PAGE;
+		offs = CONC_wDACFC_OFF;
+		break;
+
+	case PORT_SYN:
+		page = CONC_SYNCTL_PAGE;
+		offs = CONC_wSYNFC_OFF;
+		break;
+
+	case PORT_ADC:
+		page = CONC_ADCCTL_PAGE;
+		offs = CONC_wADCFC_OFF;
+		break;
+	}
+
+	/*
+	 * Note that the current frame counter is in the high nybble.
+	 */
+	frameno = audiopci_readmem(port->dev, page, offs) >> 16;
+	n = frameno >= port->frameno ?
+	    frameno - port->frameno :
+	    frameno + port->nframes - port->frameno;
+	port->frameno = frameno;
+	port->count += n;
+}
+
+static uint64_t
+audiopci_count(void *arg)
+{
+	audiopci_port_t *port = arg;
+	audiopci_dev_t *dev = port->dev;
+	uint64_t val;
+
+	mutex_enter(&dev->mutex);
+	if (!dev->suspended) {
+		audiopci_update_port(port);
+	}
+	val = port->count;
+	mutex_exit(&dev->mutex);
+	return (val);
+}
+
+static void
+audiopci_close(void *arg)
+{
+	audiopci_port_t *port = arg;
+
+	audiopci_stop(port);
+}
+
+static void
+audiopci_sync(void *arg, unsigned nframes)
+{
+	audiopci_port_t *port = arg;
+
+	_NOTE(ARGUNUSED(nframes));
+
+	if (port->num == PORT_ADC) {
+		(void) ddi_dma_sync(port->dmah, 0, 0, DDI_DMA_SYNC_FORCPU);
+	} else {
+		(void) ddi_dma_sync(port->dmah, 0, 0, DDI_DMA_SYNC_FORDEV);
+	}
+}
+
+static size_t
+audiopci_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	/*
+	 * There is actually a small FIFO with 64 bytes -- 16 samples --
+	 * onboard, but it is probably not worth the effort to report it,
+	 * and the FIFO will not always be full.
+	 */
+	return (0);
+}
+
+audio_engine_ops_t audiopci_engine_ops = {
+	AUDIO_ENGINE_VERSION,		/* version number */
+	audiopci_open,
+	audiopci_close,
+	audiopci_start,
+	audiopci_stop,
+	audiopci_count,
+	audiopci_format,
+	audiopci_channels,
+	audiopci_rate,
+	audiopci_sync,
+	audiopci_qlen,
+	NULL,
+};
+
+static uint16_t
+audiopci_dac_rate(int samPerSec)
+{
+	unsigned short usTemp;
+
+	/* samPerSec /= 2; */
+
+	usTemp = (unsigned short) ((DAC_CLOCK_DIVIDE / 8) / samPerSec);
+
+	if (usTemp & 0x00000001) {
+		usTemp >>= 1;
+		usTemp -= 1;
+	} else {
+		usTemp >>= 1;
+		usTemp -= 2;
+	}
+	return (usTemp);
+}
+
+void
+audiopci_init_hw(audiopci_dev_t *dev)
+{
+	int tmp;
+
+	/* setup DAC frequency */
+	PUT16(dev, CONC_wDACRATE_OFF, audiopci_dac_rate(48000));
+
+	CLR8(dev, CONC_bMISCCTL_OFF, CONC_MISCCTL_CCB_INTRM);
+	SET8(dev, CONC_bMISCCTL_OFF, CONC_MISCCTL_SYN_44KHZ);
+
+	/* Turn on CODEC (UART and joystick left disabled) */
+	tmp = GET8(dev, CONC_bDEVCTL_OFF);
+	tmp |= CONC_DEVCTL_SERR_DIS;
+	tmp |= CONC_DEVCTL_CODEC_EN;
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+
+	/* Reset the UART */
+	PUT8(dev, CONC_bUARTCSTAT_OFF, 0x00);
+
+	/* Disable NMI */
+	PUT8(dev, CONC_bNMIENA_OFF, 0);
+	PUT16(dev, CONC_wNMISTAT_OFF, 0);
+
+	/* Initialize serial interface */
+	PUT8(dev, CONC_bSERCTL_OFF, 0);
+	PUT8(dev, CONC_bSERFMT_OFF,
+	    CONC_PCM_SYN_STEREO | CONC_PCM_SYN_16BIT);
+
+	/* Unmute codec */
+	CLR8(dev, CONC_bMISCCTL_OFF, CONC_MISCCTL_MUTE);
+
+	/* mixer initialization */
+	audiopci_ak_idle(dev);
+
+	/* power/reset down the codec */
+	audiopci_ak_write(dev, CODEC_RESET_PWRD, 0);
+	drv_usecwait(10);
+
+	/* now powerup and bring out of reset */
+	audiopci_ak_write(dev, CODEC_RESET_PWRD, 0x3);
+	audiopci_ak_idle(dev);
+
+	/* enable PLL for DAC2 */
+	audiopci_ak_write(dev, CODEC_CLKSELECT, 0);
+
+	/* select input mixer */
+	audiopci_ak_write(dev, CODEC_ADSELECT, 0);
+
+	/* mark FM for output mixer */
+	audiopci_ak_write(dev, CODEC_OUT_SW1, CODEC_OUT_ENABLE_SYNTH);
+	audiopci_ak_write(dev, CODEC_OUT_SW2, CODEC_OUT_ENABLE_WAVE);
+
+	/* initialize some reasonable values for the WAVE and SYNTH inputs */
+	audiopci_ak_write(dev, CODEC_VOL_WAVE_L, 6);
+	audiopci_ak_write(dev, CODEC_VOL_WAVE_R, 6);
+	audiopci_ak_write(dev, CODEC_VOL_SYNTH_L, 6);
+	audiopci_ak_write(dev, CODEC_VOL_SYNTH_R, 6);
+
+	/* enable microphone phantom power */
+	if (dev->micbias) {
+		SET16(dev, 2, CONC_DEVCTL_MICBIAS);
+	}
+
+	dev->enabled = B_TRUE;
+}
+
+static int
+audiopci_init(audiopci_dev_t *dev)
+{
+	dev->micbias = 1;
+
+	audiopci_init_hw(dev);
+
+	dev->pintrs = ddi_prop_get_int(DDI_DEV_T_ANY, dev->dip,
+	    DDI_PROP_DONTPASS, "play-interrupts", DEFINTS);
+
+	dev->rintrs = ddi_prop_get_int(DDI_DEV_T_ANY, dev->dip,
+	    DDI_PROP_DONTPASS, "record-interrupts", DEFINTS);
+
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audiopci_port_t *port;
+		unsigned caps;
+		unsigned dmaflags;
+		size_t rlen;
+		ddi_dma_cookie_t c;
+		unsigned ccnt;
+
+		port = &dev->port[i];
+		port->dev = dev;
+
+		switch (i) {
+		case PORT_SYN:
+			caps = ENGINE_OUTPUT_CAP;
+			dmaflags = DDI_DMA_WRITE | DDI_DMA_CONSISTENT;
+			port->speed = 44100;
+			break;
+
+		case PORT_DAC:
+			caps = ENGINE_OUTPUT_CAP;
+			dmaflags = DDI_DMA_WRITE | DDI_DMA_CONSISTENT;
+			port->speed = 48000;
+			break;
+
+		case PORT_ADC:
+			caps = ENGINE_INPUT_CAP;
+			dmaflags = DDI_DMA_READ | DDI_DMA_CONSISTENT;
+			port->speed = 48000;
+			break;
+		}
+
+		port->num = i;
+
+		/*
+		 * Allocate DMA resources.
+		 */
+
+		if (ddi_dma_alloc_handle(dev->dip, &dma_attr, DDI_DMA_SLEEP,
+		    NULL, &port->dmah) != DDI_SUCCESS) {
+			audio_dev_warn(dev->adev,
+			    "port %d: dma handle allocation failed", i);
+			return (DDI_FAILURE);
+		}
+		if (ddi_dma_mem_alloc(port->dmah, AUDIOPCI_BUF_LEN, &buf_attr,
+		    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->kaddr,
+		    &rlen, &port->acch) != DDI_SUCCESS) {
+			audio_dev_warn(dev->adev,
+			    "port %d: dma memory allocation failed", i);
+			return (DDI_FAILURE);
+		}
+		/* ensure that the buffer is zeroed out properly */
+		bzero(port->kaddr, rlen);
+		if (ddi_dma_addr_bind_handle(port->dmah, NULL, port->kaddr,
+		    AUDIOPCI_BUF_LEN, dmaflags, DDI_DMA_SLEEP, NULL,
+		    &c, &ccnt) != DDI_DMA_MAPPED) {
+			audio_dev_warn(dev->adev,
+			    "port %d: dma binding failed", i);
+			return (DDI_FAILURE);
+		}
+		port->paddr = c.dmac_address;
+
+		/*
+		 * Allocate and configure audio engine.
+		 */
+		port->engine = audio_engine_alloc(&audiopci_engine_ops, caps);
+		if (port->engine == NULL) {
+			audio_dev_warn(dev->adev,
+			    "port %d: audio_engine_alloc failed", i);
+			return (DDI_FAILURE);
+		}
+
+		audio_engine_set_private(port->engine, port);
+		audio_dev_add_engine(dev->adev, port->engine);
+	}
+
+	/*
+	 * Register audio controls.
+	 */
+	if (audiopci_add_controls(dev) == DDI_FAILURE) {
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * Set up kstats for interrupt reporting.
+	 */
+	dev->ksp = kstat_create(ddi_driver_name(dev->dip),
+	    ddi_get_instance(dev->dip), ddi_driver_name(dev->dip),
+	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT);
+	if (dev->ksp != NULL) {
+		kstat_install(dev->ksp);
+	}
+
+	if (audio_dev_register(dev->adev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev,
+		    "unable to register with audio framework");
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+}
+
+int
+audiopci_setup_interrupts(audiopci_dev_t *dev)
+{
+	int actual;
+	uint_t ipri;
+
+	if ((ddi_intr_alloc(dev->dip, dev->ihandle, DDI_INTR_TYPE_FIXED,
+	    0, 1, &actual, DDI_INTR_ALLOC_NORMAL) != DDI_SUCCESS) ||
+	    (actual != 1)) {
+		audio_dev_warn(dev->adev, "can't alloc intr handle");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_get_pri(dev->ihandle[0], &ipri) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev,  "can't determine intr priority");
+		(void) ddi_intr_free(dev->ihandle[0]);
+		dev->ihandle[0] = NULL;
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_add_handler(dev->ihandle[0], audiopci_intr, dev,
+	    NULL) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "can't add intr handler");
+		(void) ddi_intr_free(dev->ihandle[0]);
+		dev->ihandle[0] = NULL;
+		return (DDI_FAILURE);
+	}
+
+	mutex_init(&dev->mutex, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+
+	return (DDI_SUCCESS);
+}
+
+void
+audiopci_destroy(audiopci_dev_t *dev)
+{
+	int	i;
+
+	if (dev->ihandle[0] != NULL) {
+		(void) ddi_intr_disable(dev->ihandle[0]);
+		(void) ddi_intr_remove_handler(dev->ihandle[0]);
+		(void) ddi_intr_free(dev->ihandle[0]);
+		mutex_destroy(&dev->mutex);
+	}
+
+	if (dev->ksp != NULL) {
+		kstat_delete(dev->ksp);
+	}
+
+	/* free up ports, including DMA resources for ports */
+	for (i = 0; i <= PORT_MAX; i++) {
+		audiopci_port_t	*port = &dev->port[i];
+
+		if (port->paddr != 0)
+			(void) ddi_dma_unbind_handle(port->dmah);
+		if (port->acch != NULL)
+			ddi_dma_mem_free(&port->acch);
+		if (port->dmah != NULL)
+			ddi_dma_free_handle(&port->dmah);
+
+		if (port->engine != NULL) {
+			audio_dev_remove_engine(dev->adev, port->engine);
+			audio_engine_free(port->engine);
+		}
+	}
+
+	if (dev->acch != NULL) {
+		ddi_regs_map_free(&dev->acch);
+	}
+
+	audiopci_del_controls(dev);
+
+	if (dev->adev != NULL) {
+		audio_dev_free(dev->adev);
+	}
+
+	kmem_free(dev, sizeof (*dev));
+}
+
+static void
+audiopci_stereo(audiopci_dev_t *dev, audiopci_ctrl_num_t num, uint8_t lreg)
+{
+	uint8_t		lval, rval;
+	uint8_t		lmute, rmute;
+	uint64_t	val;
+	uint8_t		rreg;
+
+	rreg = lreg + 1;
+	val = dev->controls[num].val;
+	lval = (val & 0xff00) >> 8;
+	rval = val & 0xff;
+
+	lmute = lval ? 0 : CODEC_ATT_MUTE;
+	rmute = rval ? 0 : CODEC_ATT_MUTE;
+
+	/* convert to attenuation & apply mute if appropriate */
+	lval = ((((100U - lval) * CODEC_ATT_MAX) / 100) & 0xff) | lmute;
+	rval = ((((100U - rval) * CODEC_ATT_MAX) / 100) & 0xff) | rmute;
+
+	audiopci_ak_write(dev, lreg, lval);
+	audiopci_ak_write(dev, rreg, rval);
+}
+
+static void
+audiopci_mono(audiopci_dev_t *dev, audiopci_ctrl_num_t num, uint8_t reg)
+{
+	uint64_t val = (dev->controls[num].val & 0xff);
+	uint8_t mute;
+
+	mute = val ? 0 : CODEC_ATT_MUTE;
+	val = ((((100U - val) * CODEC_ATT_MAX) / 100) & 0xff) | mute;
+
+	audiopci_ak_write(dev, reg, val);
+}
+
+static void
+audiopci_mono8(audiopci_dev_t *dev, audiopci_ctrl_num_t num, uint8_t reg)
+{
+	uint64_t val = (dev->controls[num].val & 0xff);
+	uint8_t mute;
+
+	mute = val ? 0 : CODEC_ATT_MUTE;
+	val = ((((100U - val) * CODEC_ATT_MONO) / 100) & 0xff) | mute;
+
+	audiopci_ak_write(dev, reg, val);
+}
+
+static int
+audiopci_get_value(void *arg, uint64_t *val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	mutex_enter(&dev->mutex);
+	*val = pc->val;
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static void
+audiopci_configure_output(audiopci_dev_t *dev)
+{
+	uint64_t val;
+	uint8_t	tmp;
+
+	/* PCM/Wave level */
+	audiopci_mono(dev, CTL_VOLUME, CODEC_VOL_WAVE_L);
+	audiopci_mono(dev, CTL_VOLUME, CODEC_VOL_WAVE_R);
+	audiopci_mono(dev, CTL_VOLUME, CODEC_VOL_SYNTH_L);
+	audiopci_mono(dev, CTL_VOLUME, CODEC_VOL_SYNTH_R);
+
+	/* front & mono outputs */
+	audiopci_stereo(dev, CTL_FRONT, CODEC_VOL_MASTER_L);
+	audiopci_mono8(dev, CTL_MONO, CODEC_VOL_MONO);
+
+	val = dev->controls[CTL_MONSRC].val;
+
+	/* setup output monitoring as well */
+	tmp = CODEC_OUT_ENABLE_SYNTH;
+	if (val & (1U << INPUT_MIC))
+		tmp |= CODEC_OUT_ENABLE_MIC;
+	if (val & (1U << INPUT_CD))
+		tmp |= CODEC_OUT_ENABLE_CD;
+	if (val & (1U << INPUT_LINEIN))
+		tmp |= CODEC_OUT_ENABLE_AUX;
+	audiopci_ak_write(dev, CODEC_OUT_SW1, tmp);
+
+	tmp = CODEC_OUT_ENABLE_WAVE;
+	if (val & (1U << INPUT_VIDEO))
+		tmp |= CODEC_OUT_ENABLE_TV;
+	if (val & (1U << INPUT_PHONE))
+		tmp |= CODEC_OUT_ENABLE_TAD;
+	audiopci_ak_write(dev, CODEC_OUT_SW2, tmp);
+}
+
+static void
+audiopci_configure_input(audiopci_dev_t *dev)
+{
+	uint64_t	val = dev->controls[CTL_RECSRC].val;
+	uint8_t		tmp;
+
+	tmp = 0;
+	if (val & (1U << INPUT_LINEIN))
+		tmp |= CODEC_IN_ENABLE_AUX_L;
+	if (val & (1U << INPUT_CD))
+		tmp |= CODEC_IN_ENABLE_CD_L;
+	if (val & (1U << INPUT_MIC))
+		tmp |= CODEC_IN_ENABLE_MIC;
+	if (val & (1U << INPUT_PHONE))
+		tmp |= CODEC_IN_ENABLE_TAD;
+	audiopci_ak_write(dev, CODEC_LIN_SW1, tmp);
+
+	tmp = 0;
+	if (val & (1U << INPUT_LINEIN))
+		tmp |= CODEC_IN_ENABLE_AUX_R;
+	if (val & (1U << INPUT_CD))
+		tmp |= CODEC_IN_ENABLE_CD_R;
+	if (val & (1U << INPUT_PHONE))
+		tmp |= CODEC_IN_ENABLE_TAD;
+	if (val & (1U << INPUT_MIC))
+		tmp |= CODEC_IN_ENABLE_MIC;
+	audiopci_ak_write(dev, CODEC_RIN_SW1, tmp);
+
+	tmp = 0;
+	if (val & (1U << INPUT_VIDEO))
+		tmp |= CODEC_IN_ENABLE_TV_L;
+	if (val & (1U << INPUT_MIC))
+		tmp |= CODEC_IN_ENABLE_TMIC;
+	audiopci_ak_write(dev, CODEC_LIN_SW2, tmp);
+
+	tmp = 0;
+	if (val & (1U << INPUT_VIDEO))
+		tmp |= CODEC_IN_ENABLE_TV_R;
+	if (val & (1U << INPUT_MIC))
+		tmp |= CODEC_IN_ENABLE_TMIC;
+	audiopci_ak_write(dev, CODEC_RIN_SW2, tmp);
+
+	/* configure volumes */
+	audiopci_mono(dev, CTL_MIC, CODEC_VOL_MIC);
+	audiopci_mono(dev, CTL_PHONE, CODEC_VOL_TAD);
+	audiopci_stereo(dev, CTL_LINE, CODEC_VOL_AUX_L);
+	audiopci_stereo(dev, CTL_CD, CODEC_VOL_CD_L);
+	audiopci_stereo(dev, CTL_VID, CODEC_VOL_TV_L);
+
+	/* activate 30dB mic boost */
+	audiopci_ak_write(dev, CODEC_MICBOOST,
+	    dev->controls[CTL_MICBOOST].val ? 1 : 0);
+}
+
+static int
+audiopci_set_reclevel(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+	uint8_t		l;
+	uint8_t		r;
+
+	l = (val & 0xff00) >> 8;
+	r = val & 0xff;
+
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_input(dev);
+
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static int
+audiopci_set_micboost(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_input(dev);
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static int
+audiopci_set_monsrc(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	if ((val & ~INSRCS) != 0)
+		return (EINVAL);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_output(dev);
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static int
+audiopci_set_recsrc(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	if ((val & ~INSRCS) != 0)
+		return (EINVAL);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_input(dev);
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static int
+audiopci_set_volume(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	val &= 0xff;
+	if (val > 100)
+		return (EINVAL);
+
+	val = (val & 0xff) | ((val & 0xff) << 8);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_output(dev);
+	mutex_exit(&dev->mutex);
+
+	return (0);
+}
+
+static int
+audiopci_set_front(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+	uint8_t		l;
+	uint8_t		r;
+
+	l = (val & 0xff00) >> 8;
+	r = val & 0xff;
+
+	if ((l > 100) || (r > 100))
+		return (EINVAL);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_output(dev);
+
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+static int
+audiopci_set_speaker(void *arg, uint64_t val)
+{
+	audiopci_ctrl_t	*pc = arg;
+	audiopci_dev_t	*dev = pc->dev;
+
+	val &= 0xff;
+
+	if (val > 100)
+		return (EINVAL);
+
+	mutex_enter(&dev->mutex);
+	pc->val = val;
+	audiopci_configure_output(dev);
+
+	mutex_exit(&dev->mutex);
+	return (0);
+}
+
+#define	PLAYCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_PLAY)
+#define	RECCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_REC)
+#define	MONCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_MONITOR)
+#define	PCMVOL	(PLAYCTL | AUDIO_CTRL_FLAG_PCMVOL)
+#define	MAINVOL	(PLAYCTL | AUDIO_CTRL_FLAG_MAINVOL)
+#define	RECVOL	(RECCTL | AUDIO_CTRL_FLAG_RECVOL)
+
+static void
+audiopci_alloc_ctrl(audiopci_dev_t *dev, uint32_t num, uint64_t val)
+{
+	audio_ctrl_desc_t	desc;
+	audio_ctrl_wr_t		fn;
+	audiopci_ctrl_t		*pc;
+
+	bzero(&desc, sizeof (desc));
+
+	pc = &dev->controls[num];
+	pc->num = num;
+	pc->dev = dev;
+
+	switch (num) {
+	case CTL_VOLUME:
+		desc.acd_name = AUDIO_CTRL_ID_VOLUME;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = PCMVOL;
+		fn = audiopci_set_volume;
+		break;
+
+	case CTL_FRONT:
+		desc.acd_name = AUDIO_CTRL_ID_LINEOUT;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiopci_set_front;
+		break;
+
+	case CTL_MONO:
+		desc.acd_name = AUDIO_CTRL_ID_SPEAKER;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MAINVOL;
+		fn = audiopci_set_speaker;
+		break;
+
+	case CTL_MIC:
+		desc.acd_name = AUDIO_CTRL_ID_MIC;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiopci_set_reclevel;
+		break;
+
+	case CTL_LINE:
+		desc.acd_name = AUDIO_CTRL_ID_LINEIN;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiopci_set_reclevel;
+		break;
+
+	case CTL_CD:
+		desc.acd_name = AUDIO_CTRL_ID_CD;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiopci_set_reclevel;
+		break;
+
+	case CTL_VID:
+		desc.acd_name = AUDIO_CTRL_ID_VIDEO;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiopci_set_reclevel;
+		break;
+
+	case CTL_PHONE:
+		desc.acd_name = AUDIO_CTRL_ID_PHONE;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiopci_set_reclevel;
+		break;
+
+	case CTL_RECSRC:
+		desc.acd_name = AUDIO_CTRL_ID_RECSRC;
+		desc.acd_type = AUDIO_CTRL_TYPE_ENUM;
+		desc.acd_minvalue = INSRCS;
+		desc.acd_maxvalue = INSRCS;
+		desc.acd_flags = RECCTL | AUDIO_CTRL_FLAG_MULTI;
+		for (int i = 0; audiopci_insrcs[i]; i++) {
+			desc.acd_enum[i] = audiopci_insrcs[i];
+		}
+		fn = audiopci_set_recsrc;
+		break;
+
+	case CTL_MONSRC:
+		desc.acd_name = AUDIO_CTRL_ID_MONSRC;
+		desc.acd_type = AUDIO_CTRL_TYPE_ENUM;
+		desc.acd_minvalue = INSRCS;
+		desc.acd_maxvalue = INSRCS;
+		desc.acd_flags = MONCTL | AUDIO_CTRL_FLAG_MULTI;
+		for (int i = 0; audiopci_insrcs[i]; i++) {
+			desc.acd_enum[i] = audiopci_insrcs[i];
+		}
+		fn = audiopci_set_monsrc;
+		break;
+
+	case CTL_MICBOOST:
+		desc.acd_name = AUDIO_CTRL_ID_MICBOOST;
+		desc.acd_type = AUDIO_CTRL_TYPE_BOOLEAN;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECCTL;
+		fn = audiopci_set_micboost;
+		break;
+	}
+
+	pc->val = val;
+	pc->ctrl = audio_dev_add_control(dev->adev, &desc,
+	    audiopci_get_value, fn, pc);
+}
+
+static int
+audiopci_add_controls(audiopci_dev_t *dev)
+{
+	audiopci_alloc_ctrl(dev, CTL_VOLUME, 75);
+	audiopci_alloc_ctrl(dev, CTL_FRONT, ((75) | (75 << 8)));
+	audiopci_alloc_ctrl(dev, CTL_MONO, 75);
+	audiopci_alloc_ctrl(dev, CTL_MIC, 50);
+	audiopci_alloc_ctrl(dev, CTL_LINE, 0);
+	audiopci_alloc_ctrl(dev, CTL_CD, 0);
+	audiopci_alloc_ctrl(dev, CTL_VID, 0);
+	audiopci_alloc_ctrl(dev, CTL_PHONE, 0);
+	audiopci_alloc_ctrl(dev, CTL_RECSRC, (1U << INPUT_MIC));
+	audiopci_alloc_ctrl(dev, CTL_MONSRC, 0);
+	audiopci_alloc_ctrl(dev, CTL_MICBOOST, 1);
+
+	audiopci_configure_output(dev);
+	audiopci_configure_input(dev);
+
+	return (DDI_SUCCESS);
+}
+
+void
+audiopci_del_controls(audiopci_dev_t *dev)
+{
+	for (int i = 0; i < CTL_NUM; i++) {
+		if (dev->controls[i].ctrl) {
+			audio_dev_del_control(dev->controls[i].ctrl);
+		}
+	}
+}
+
+int
+audiopci_attach(dev_info_t *dip)
+{
+	uint16_t pci_command, vendor, device;
+	audiopci_dev_t *dev;
+	ddi_acc_handle_t pcih;
+
+	dev = kmem_zalloc(sizeof (*dev), KM_SLEEP);
+	dev->dip = dip;
+	ddi_set_driver_private(dip, dev);
+
+	if (pci_config_setup(dip, &pcih) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "pci_config_setup failed");
+		kmem_free(dev, sizeof (*dev));
+		return (DDI_FAILURE);
+	}
+
+	vendor = pci_config_get16(pcih, PCI_CONF_VENID);
+	device = pci_config_get16(pcih, PCI_CONF_DEVID);
+
+	if ((vendor != ENSONIQ_VENDOR_ID && vendor != CREATIVE_VENDOR_ID) ||
+	    (device != ENSONIQ_ES1370))
+		goto err_exit;
+
+	dev->devid = device;
+
+	dev->adev = audio_dev_alloc(dip, 0);
+	if (dev->adev == NULL) {
+		goto err_exit;
+	}
+
+	audio_dev_set_description(dev->adev, "AudioPCI");
+	audio_dev_set_version(dev->adev, "ES1370");
+	audio_dev_add_info(dev->adev, "Legacy codec: Asahi Kasei AK4531");
+
+	/* activate the device */
+	pci_command = pci_config_get16(pcih, PCI_CONF_COMM);
+	pci_command |= PCI_COMM_ME | PCI_COMM_IO;
+	pci_config_put16(pcih, PCI_CONF_COMM, pci_command);
+
+	/* map registers */
+	if (ddi_regs_map_setup(dip, 1, &dev->regs, 0, 0, &acc_attr,
+	    &dev->acch) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "can't map registers");
+		goto err_exit;
+	}
+
+	if (audiopci_setup_interrupts(dev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "can't register interrupts");
+		goto err_exit;
+	}
+
+	/* This allocates and configures the engines */
+	if (audiopci_init(dev) != DDI_SUCCESS) {
+		audio_dev_warn(dev->adev, "can't init device");
+		goto err_exit;
+	}
+
+	(void) ddi_intr_enable(dev->ihandle[0]);
+
+	pci_config_teardown(&pcih);
+
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+err_exit:
+	pci_config_teardown(&pcih);
+
+	audiopci_destroy(dev);
+
+	return (DDI_FAILURE);
+}
+
+int
+audiopci_detach(audiopci_dev_t *dev)
+{
+	int tmp;
+
+	/* first unregister us from the DDI framework, might be busy */
+	if (audio_dev_unregister(dev->adev) != DDI_SUCCESS)
+		return (DDI_FAILURE);
+
+	mutex_enter(&dev->mutex);
+
+	tmp = GET8(dev, CONC_bSERCTL_OFF) &
+	    ~(CONC_SERCTL_DACIE | CONC_SERCTL_SYNIE | CONC_SERCTL_ADCIE);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+
+	tmp = GET8(dev, CONC_bDEVCTL_OFF) &
+	    ~(CONC_DEVCTL_DAC_EN | CONC_DEVCTL_ADC_EN | CONC_DEVCTL_SYN_EN);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+
+	dev->enabled = B_FALSE;
+
+	mutex_exit(&dev->mutex);
+
+	audiopci_destroy(dev);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audiopci_resume(audiopci_dev_t *dev)
+{
+	/* ask framework to reset/relocate engine data */
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audio_engine_reset(dev->port[i].engine);
+	}
+
+	mutex_enter(&dev->mutex);
+	dev->suspended = B_FALSE;
+
+	/* reinitialize hardware */
+	audiopci_init_hw(dev);
+
+	/* restore mixer settings */
+	audiopci_configure_output(dev);
+	audiopci_configure_input(dev);
+
+	/* restart ports */
+	for (int i = 0; i < PORT_MAX; i++) {
+		audiopci_port_t	*port = &dev->port[i];
+		audiopci_init_port(port);
+		/* possibly start it up if was going when we suspended */
+		if (port->trigger) {
+			audiopci_start_port(port);
+
+			/* signal callbacks on resume */
+			if (port->num == PORT_ADC) {
+				audio_engine_produce(port->engine);
+			} else {
+				audio_engine_consume(port->engine);
+			}
+		}
+	}
+	mutex_exit(&dev->mutex);
+	return (DDI_SUCCESS);
+}
+
+static int
+audiopci_suspend(audiopci_dev_t *dev)
+{
+	/*
+	 * Stop all engines/DMA data.
+	 */
+	mutex_enter(&dev->mutex);
+	for (int i = 0; i <= PORT_MAX; i++) {
+		audiopci_stop_port(&dev->port[i]);
+		audiopci_update_port(&dev->port[i]);
+	}
+	dev->suspended = B_TRUE;
+	dev->enabled = B_FALSE;
+	mutex_exit(&dev->mutex);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audiopci_quiesce(dev_info_t *dip)
+{
+	audiopci_dev_t	*dev;
+	uint8_t		tmp;
+
+	if ((dev = ddi_get_driver_private(dip)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	/* This disables all DMA engines and interrupts */
+	tmp = GET8(dev, CONC_bSERCTL_OFF) &
+	    ~(CONC_SERCTL_DACIE | CONC_SERCTL_SYNIE | CONC_SERCTL_ADCIE);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+	PUT8(dev, CONC_bSERCTL_OFF, tmp);
+
+	tmp = GET8(dev, CONC_bDEVCTL_OFF) &
+	    ~(CONC_DEVCTL_DAC_EN | CONC_DEVCTL_ADC_EN | CONC_DEVCTL_SYN_EN);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+	PUT8(dev, CONC_bDEVCTL_OFF, tmp);
+
+	return (DDI_SUCCESS);
+}
+
+
+static int
+audiopci_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	audiopci_dev_t *dev;
+
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audiopci_attach(dip));
+
+	case DDI_RESUME:
+		if ((dev = ddi_get_driver_private(dip)) == NULL) {
+			return (DDI_FAILURE);
+		}
+		return (audiopci_resume(dev));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+static int
+audiopci_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	audiopci_dev_t *dev;
+
+	if ((dev = ddi_get_driver_private(dip)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audiopci_detach(dev));
+
+	case DDI_SUSPEND:
+		return (audiopci_suspend(dev));
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+static struct dev_ops audiopci_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	NULL,			/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	audiopci_ddi_attach,	/* attach */
+	audiopci_ddi_detach,	/* detach */
+	nodev,			/* reset */
+	NULL,			/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+	audiopci_quiesce,	/* quiesce */
+};
+
+static struct modldrv audiopci_modldrv = {
+	&mod_driverops,			/* drv_modops */
+	"Ensoniq 1370 Audio",		/* linkinfo */
+	&audiopci_dev_ops,		/* dev_ops */
+};
+
+static struct modlinkage modlinkage = {
+	MODREV_1,
+	{ &audiopci_modldrv, NULL }
+};
+
+int
+_init(void)
+{
+	int	rv;
+
+	audio_init_ops(&audiopci_dev_ops, DRVNAME);
+	if ((rv = mod_install(&modlinkage)) != 0) {
+		audio_fini_ops(&audiopci_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_fini(void)
+{
+	int	rv;
+
+	if ((rv = mod_remove(&modlinkage)) == 0) {
+		audio_fini_ops(&audiopci_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&modlinkage, modinfop));
+}
diff --git a/usr/src/uts/common/io/audio/drv/audiopci/audiopci.conf b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.conf
new file mode 100644
index 0000000000..ca6a4c8726
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.conf
@@ -0,0 +1,46 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Configuration file for the audiopci audio driver.
+#
+#	WARNING: This is an UNSTABLE configuration file. Its contents
+#		may change at any time.
+
+#
+# play-interrupts sets the number of interrupts per second when playing.
+# This affects the resolution of various things, such as sample counts.
+# record-interrupts does the same for record interrupts.
+#
+# These may be tuned to get more accurate information by increasing the
+# count. However, the larger the interrupts per second the larger the
+# load on the system. So use this capability cautiously. The audiopci
+# driver enforces a maximum and minimum count.
+#
+# It should also be understood that not all interrupt rates are legal.
+# The hardware is restricted to DMA buffers being allocated on certain
+# boundaries. If those boundaries are violated then the value specified
+# will be ignored.
+#
+play-interrupts=75;
+record-interrupts=75;
diff --git a/usr/src/uts/common/io/audio/drv/audiopci/audiopci.h b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.h
new file mode 100644
index 0000000000..c922dc2cc4
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiopci/audiopci.h
@@ -0,0 +1,233 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+/*
+ * Purpose: Definitions for the Creative/Ensoniq AudioPCI97 driver.
+ */
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#ifndef	_AUDIOPCI_H
+#define	_AUDIOPCI_H
+
+/* CONCERT PCI-SIG defines */
+#define	CONC_PCI_VENDID		0x1274U
+#define	CONC_PCI_DEVID		0x5000U
+
+/* used for the development board only!! */
+#define	CONC_DEV_PCI_VENDID	0x1274U
+#define	CONC_DEV_PCI_DEVID	0x5000U
+
+
+/*
+ * CONCERT Registers
+ */
+
+#define	DAC_CLOCK_DIVIDE	22579200UL	/* DAC2 (CODEC) clock divide */
+
+/* Concert direct register offset defines */
+
+#define	CONC_bDEVCTL_OFF	0x00	/* Device control/enable */
+#define	CONC_bMISCCTL_OFF	0x01	/* Miscellaneous control */
+#define	CONC_wDACRATE_OFF	0x02	/* CODEC clock divider for PLL */
+#define	CONC_dSTATUS_OFF	0x04	/* long status register */
+#define	CONC_bCODECSTAT_OFF	0x05	/* CODEC interface status */
+#define	CONC_bUARTDATA_OFF	0x08	/* UART data R/W - read clears RX int */
+#define	CONC_bUARTCSTAT_OFF	0x09	/* UART control and status */
+#define	CONC_bMEMPAGE_OFF	0x0c	/* Memory page select */
+#define	CONC_wCODECCTL_OFF	0x10	/* CODEC control - word-write-only */
+#define	CONC_wNMISTAT_OFF	0x18	/* Legacy NMI status */
+#define	CONC_bNMIENA_OFF	0x1a	/* Legacy NMI enable */
+#define	CONC_bNMICTL_OFF	0x1b	/* Legacy control */
+#define	CONC_bSERFMT_OFF	0x20	/* Serial device control */
+#define	CONC_bSERCTL_OFF	0x21	/* Serial device format */
+#define	CONC_bSKIPC_OFF		0x22	/* Skip counts for DAC (wave) */
+#define	CONC_wSYNIC_OFF		0x24	/* Synth int count in sample frames */
+#define	CONC_wSYNCIC_OFF	0x26	/* Synth current int count */
+#define	CONC_wDACIC_OFF		0x28	/* DAC int count in sample frames */
+#define	CONC_wDACCIC_OFF	0x2a	/* DAC current int count */
+#define	CONC_wADCIC_OFF		0x2c	/* ADC int count in sample frames */
+#define	CONC_wADCCIC_OFF	0x2e	/* ADC current int count */
+#define	CONC_MEMBASE_OFF	0x30	/* Memory window base - 16 bytes */
+
+/* Device Control defines */
+#define	CONC_DEVCTL_SERR_DIS	0x01	/* internal PCI serr bus enable */
+#define	CONC_DEVCTL_CODEC_EN	0x02	/* CoDec Enable */
+#define	CONC_DEVCTL_JSTICK_EN	0x04	/* Joystick Enable */
+#define	CONC_DEVCTL_UART_EN	0x08	/* UART Enable */
+#define	CONC_DEVCTL_ADC_EN	0x10	/* ADC Enable (record) */
+#define	CONC_DEVCTL_DAC_EN	0x20	/* DAC Enable (playback) */
+#define	CONC_DEVCTL_SYN_EN	0x40	/* Synth Enable */
+#define	CONC_DEVCTL_MICBIAS	0x4000L	/* mic bias switch */
+
+/* Misc Control defines */
+#define	CONC_MISCCTL_MUTE	0x01	/* XTL0 wired to mute */
+#define	CONC_MISCCTL_CCB_INTRM	0x04	/* CCB interrupt mask */
+#define	CONC_MISCCTL_SYN_5KHZ	0x00	/* synth: 5512 Hz */
+#define	CONC_MISCCTL_SYN_11KHZ	0x10	/* synth: 11025 Hz */
+#define	CONC_MISCCTL_SYN_22KHZ	0x20	/* synth: 22050 Hz */
+#define	CONC_MISCCTL_SYN_44KHZ	0x30	/* synth: 44100 Hz */
+
+/* Interrupt Status defines */
+#define	CONC_INTSTAT_ADCINT	0x01	/* A/D interrupt pending bit */
+#define	CONC_INTSTAT_DACINT	0x02	/* DAC interrupt pending bit */
+#define	CONC_INTSTAT_SYNINT	0x04	/* synth interrupt pending bit */
+#define	CONC_INTSTAT_UARTINT	0x08	/* UART interrupt pending bit */
+#define	CONC_INTSTAT_PENDING	0x80000000
+			/* this bit set high while'st we have an interrupt */
+
+/* Codec Status defines */
+#define	CONC_CSTAT_CSTAT	0x4
+#define	CONC_CSTAT_CBUSY	0x2
+#define	CONC_CSTAT_CWRIP	0x1
+
+/* SERFMT PCM format defines */
+#define	CONC_PCM_SYN_STEREO	0x01
+#define	CONC_PCM_SYN_16BIT	0x02
+#define	CONC_PCM_DAC_STEREO	0x04
+#define	CONC_PCM_DAC_16BIT	0x08
+#define	CONC_PCM_ADC_STEREO	0x10
+#define	CONC_PCM_ADC_16BIT	0x20
+
+/* Serial Control defines */
+#define	CONC_SERCTL_SYNIE	0x01	/* synth int enable */
+#define	CONC_SERCTL_DACIE	0x02	/* playback interrupt enable */
+#define	CONC_SERCTL_ADCIE	0x04	/* record interrupt enable */
+#define	CONC_SERCTL_SYNPAUSE	0x10	/* playback pause */
+
+/* Concert memory page-banked register offset defines */
+#define	CONC_dSYNPADDR_OFF	0x30	/* Synth host frame PCI phys addr */
+#define	CONC_wSYNFC_OFF		0x34	/* Synth host frame count in DWORDS */
+#define	CONC_wSYNCFC_OFF	0x36	/* Synth host current frame count */
+#define	CONC_dDACPADDR_OFF	0x38	/* DAC host frame PCI phys addr */
+#define	CONC_wDACFC_OFF		0x3c	/* DAC host frame count in DWORDS */
+#define	CONC_wDACCFC_OFF	0x3e	/* DAC host current frame count */
+#define	CONC_dADCPADDR_OFF	0x30	/* ADC host frame PCI phys addr */
+#define	CONC_wADCFC_OFF		0x34	/* ADC host frame count in DWORDS */
+#define	CONC_wADCCFC_OFF	0x36	/* ADC host current frame count */
+
+/* Concert memory page number defines */
+#define	CONC_SYNRAM_PAGE	0x00	/* Synth host/serial I/F RAM */
+#define	CONC_DACRAM_PAGE	0x04	/* DAC host/serial I/F RAM */
+#define	CONC_ADCRAM_PAGE	0x08	/* ADC host/serial I/F RAM */
+#define	CONC_SYNCTL_PAGE	0x0c	/* Page bank for synth host control */
+#define	CONC_DACCTL_PAGE	0x0c	/* Page bank for DAC host control */
+#define	CONC_ADCCTL_PAGE	0x0d	/* Page bank for ADC host control */
+#define	CONC_FIFO0_PAGE		0x0e	/* page 0 of UART "FIFO" (rx stash) */
+#define	CONC_FIFO1_PAGE		0x0f	/* page 1 of UART "FIFO" (rx stash) */
+
+/* UARTCSTAT register masks  */
+#define	CONC_UART_RXRDY		0x01
+#define	CONC_UART_TXRDY		0x02
+#define	CONC_UART_TXINT		0x04
+#define	CONC_UART_RXINT		0x80
+
+#define	CONC_UART_CTL		0x03
+#define	CONC_UART_TXINTEN	0x20
+#define	CONC_UART_RXINTEN	0x80
+
+/*
+ * CODEC register map
+ */
+#define	NUMREGS			32	/* total number of registers */
+#define	NUMVOLS			16	/* number of vol regs */
+
+/* Source and output volume control defines */
+#define	CODEC_VOL_MASTER_L	0x00U	/* Master out, left */
+#define	CODEC_VOL_MASTER_R	0x01U	/* Master out, right */
+#define	CODEC_VOL_WAVE_L	0x02U	/* Wave DAC, left */
+#define	CODEC_VOL_WAVE_R	0x03U	/* Wave DAC, right */
+#define	CODEC_VOL_SYNTH_L	0x04U	/* Synth DAC, left */
+#define	CODEC_VOL_SYNTH_R	0x05U	/* Synth DAC, right */
+#define	CODEC_VOL_CD_L		0x06U	/* CD audio, left */
+#define	CODEC_VOL_CD_R		0x07U	/* CD audio, right */
+#define	CODEC_VOL_AUX_L		0x08U	/* Aux line source, left */
+#define	CODEC_VOL_AUX_R		0x09U	/* Aux line source, right */
+#define	CODEC_VOL_TV_L		0x0aU	/* TV Tuner, left */
+#define	CODEC_VOL_TV_R		0x0bU	/* TV Tuner, right */
+#define	CODEC_VOL_TAD		0x0cU	/* TAD monitor, mono */
+#define	CODEC_VOL_MONO2		0x0dU	/* Unused MONO2 */
+#define	CODEC_VOL_MIC		0x0eU	/* Mic, mono */
+#define	CODEC_VOL_MONO		0x0fU	/* Mono out volume */
+
+/* Input bus enable defines -SW1 */
+#define	CODEC_IN_ENABLE_MIC	0x01U	/* Mic enable, mono */
+#define	CODEC_IN_ENABLE_CD_R	0x02U	/* CD audio enable, right */
+#define	CODEC_IN_ENABLE_CD_L	0x04U	/* CD audio enable, left */
+#define	CODEC_IN_ENABLE_AUX_R	0x08U	/* Aux line source enable, right */
+#define	CODEC_IN_ENABLE_AUX_L	0x10U	/* Aux line source enable, left */
+#define	CODEC_IN_ENABLE_SYNTH_R 0x20U	/* Synth DAC enable, right */
+#define	CODEC_IN_ENABLE_SYNTH_L 0x40U	/* Synth DAC enable, left */
+
+/* Input bus enable defines - SW2 */
+#define	CODEC_IN_ENABLE_TAD	0x01U	/* TAD monitor enable, mono */
+#define	CODEC_IN_ENABLE_MONO2	0x02U	/* Unused MONO2 enable, mono */
+#define	CODEC_IN_ENABLE_WAVE	0x04U	/* Wave DAC enable */
+#define	CODEC_IN_ENABLE_TV_R	0x08U	/* TV Tuner enable, right */
+#define	CODEC_IN_ENABLE_TV_L	0x10U	/* TV Tuner enable, left */
+#define	CODEC_IN_ENABLE_TMONO2	0x20U	/* unboosted MONO2 */
+#define	CODEC_IN_ENABLE_TMONO1	0x40U	/* unboosted MONO1 */
+#define	CODEC_IN_ENABLE_TMIC	0x80U	/* unboosted MONO3 (mic) */
+
+/* Output bus enable defines - SW1 */
+#define	CODEC_OUT_ENABLE_MIC	0x01U	/* Mic enable, mono */
+#define	CODEC_OUT_ENABLE_CD	0x06U	/* CD audio enable, stereo */
+#define	CODEC_OUT_ENABLE_AUX	0x18U	/* Aux line source enable, stereo */
+#define	CODEC_OUT_ENABLE_SYNTH	0x60U	/* Synth DAC enable, stereo */
+
+/* Output bus enable defines - SW2 */
+#define	CODEC_OUT_ENABLE_TAD	0x01U	/* TAD monitor enable, mono */
+#define	CODEC_OUT_ENABLE_MONO2	0x02U	/* Unused MONO2 enable, mono */
+#define	CODEC_OUT_ENABLE_WAVE	0x0cU	/* Wave DAC enable, stereo */
+#define	CODEC_OUT_ENABLE_TV	0x30U	/* TV Tuner enable, stereo */
+
+/* Volume setting constants */
+#define	CODEC_ATT_MUTE		0x80U
+#define	CODEC_ATT_MAX		0x1fU
+#define	CODEC_ATT_MONO		0x07U
+
+/* Control function defines */
+#define	CODEC_CTL_4SPKR		0x00U	/* 4-spkr output mode enable */
+#define	CODEC_CTL_MICBOOST	0x01U	/* Mic boost (+30 dB) enable */
+
+/* Miscellaneous CODEC defines for internal use */
+#define	CODEC_OUT_SW1		0x10U
+#define	CODEC_OUT_SW2		0x11U
+#define	CODEC_LIN_SW1		0x12U
+#define	CODEC_RIN_SW1		0x13U
+#define	CODEC_LIN_SW2		0x14U
+#define	CODEC_RIN_SW2		0x15U
+#define	CODEC_RESET_PWRD	0x16U
+#define	CODEC_CLKSELECT		0x17U
+#define	CODEC_ADSELECT		0x18U
+#define	CODEC_MICBOOST		0x19U
+
+#endif	/* _AUDIOPCI_H */
diff --git a/usr/src/uts/common/io/audio/drv/audiots/audiots.c b/usr/src/uts/common/io/audio/drv/audiots/audiots.c
new file mode 100644
index 0000000000..4c5a19f87b
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiots/audiots.c
@@ -0,0 +1,2015 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+/*
+ * audiots Audio Driver
+ *
+ * This Audio Driver controls the T2 audio core in the ALI M1553
+ * southbridge chip. This chip supports multiple play streams, but just
+ * a single record stream. It also supports wave table synthesis and
+ * hardware MIDI and joystick ports. Unfortunately the MIDI ports are
+ * not available because their pins have been re-assigned to expose
+ * interrupts. We also aren't going to do anything with the joystick
+ * ports. The audio core controls an AC-97 V2.1 Codec.
+ *
+ * The DMA engine uses a single buffer which is large enough to hold
+ * two interrupts worth of data. When it gets to the mid point an
+ * interrupt is generated and data is either sent (for record) or
+ * requested and put in that half of the buffer (for play). When the
+ * second half is played we do the same, but the audio core loops the
+ * pointer back to the beginning.
+ *
+ * The audio core has a bug in silicon that doesn't let it read the AC-97
+ * Codec's register. T2 has provided an algorithm that attempts to read the
+ * the Codec several times. This is probably heuristic and thus isn't
+ * absolutely guaranteed to work. However we do have to place a limit on
+ * the looping, otherwise when we read a valid 0x00 we would never exit
+ * the loop. Unfortunately there is also a problem with writing the AC-97
+ * Codec's registers as well. Thus we read it back to verify the write.
+ *
+ * The AC'97 common code provides shadow state for AC'97 registers for us,
+ * so we only need to read those registers during early startup (primarily
+ * to determine codec id and capabilities.)
+ *
+ * We don't save any of the audio controller registers during normal
+ * operation. When we need to save register state we only have to save
+ * the aram and eram. The rest of the controller state is never modified
+ * from the initial programming. Thus restoring the controller state
+ * can be done from audiots_chip_init() as well.
+ *
+ *
+ * WARNING: The SME birdsnest platform uses a PCI bridge chip between the
+ *	CPU and the southbridge containing the audio core. There is
+ *	a bug in silicon that causes a bogus parity error. With the mixer
+ *	reimplementation project, Bug 4374774, the audio driver is always
+ *	set to the best precision and number of channels. Thus when turning
+ *	the mixer on and off the only thing that changes is the sample rate.
+ *	This change in programming doesn't trigger the silicon error.
+ *	Thus the supported channels must always be 2 and the precision
+ *	must always be 16-bits. This will keep any future change in the
+ *	mixer from exposing this bug.
+ *
+ * Due to a hardware bug, system power management is not supported by this
+ * driver.
+ *
+ *	CAUTION: If audio controller state is changed outside of aram
+ *		and eram then that information must be saved and restored
+ *		during power management shutdown and bringup.
+ *
+ *	NOTE: The AC-97 Codec's reset pin is set to PCI reset, so we
+ *		can't power down the Codec all the way.
+ *
+ *	NOTE: This driver depends on the drv/audio and misc/ac97
+ *		modules being loaded first.
+ *
+ *	NOTE: Don't OR the ap_stop register to stop a play or record. This
+ *		will just stop all active channels because a read of ap_stop
+ *		returns ap_start. Just set the ap_stop register with the
+ *		channels you want to stop. The same goes for ap_start.
+ *
+ *	NOTE: There is a hardware problem with P2 rev motherboards. After
+ *		prolonged use, reading the AC97 register will always return
+ *		busy. The AC97 register is now useless. Consequently, we are no
+ *		longer able to program the Codec. This work around disables
+ *		audio when this state is detected. It's not great, but its
+ *		better than having audio blasting out at 100% all the time.
+ *
+ *	NOTE: Power Management testing has also exposed this AC97 timeout
+ *		problem. Management has decided this is too risky for customers
+ *		and hence they want power management support removed from the
+ *		audio subsystem. All PM support is now removed.
+ */
+
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/pci.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/debug.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+#include "audiots.h"
+
+/*
+ * Module linkage routines for the kernel
+ */
+static int audiots_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audiots_detach(dev_info_t *, ddi_detach_cmd_t);
+static int audiots_quiesce(dev_info_t *);
+
+/*
+ * Entry point routine prototypes
+ */
+static int audiots_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void audiots_close(void *);
+static int audiots_start(void *);
+static void audiots_stop(void *);
+static int audiots_format(void *);
+static int audiots_channels(void *);
+static int audiots_rate(void *);
+static void audiots_chinfo(void *, int, unsigned *, unsigned *);
+static uint64_t audiots_count(void *);
+static void audiots_sync(void *, unsigned);
+static size_t audiots_qlen(void *);
+
+static audio_engine_ops_t	audiots_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	audiots_open,
+	audiots_close,
+	audiots_start,
+	audiots_stop,
+	audiots_count,
+	audiots_format,
+	audiots_channels,
+	audiots_rate,
+	audiots_sync,
+	audiots_qlen,
+	audiots_chinfo
+};
+
+/*
+ * Local Routine Prototypes
+ */
+static void audiots_power_up(audiots_state_t *);
+static void audiots_chip_init(audiots_state_t *);
+static uint16_t audiots_get_ac97(void *, uint8_t);
+static void audiots_set_ac97(void *, uint8_t, uint16_t);
+static int audiots_init_state(audiots_state_t *, dev_info_t *);
+static uint_t audiots_intr(caddr_t);
+static int audiots_map_regs(dev_info_t *, audiots_state_t *);
+static void audiots_update_port(audiots_port_t *);
+static void audiots_start_port(audiots_port_t *);
+static void audiots_stop_port(audiots_port_t *);
+static uint16_t audiots_read_ac97(audiots_state_t *, int);
+static void audiots_stop_everything(audiots_state_t *);
+static void audiots_destroy(audiots_state_t *);
+static int audiots_alloc_port(audiots_state_t *, int);
+static void audiots_reset_port(audiots_port_t *);
+
+/*
+ * Global variables, but viewable only by this file.
+ */
+
+/* anchor for soft state structures */
+static void *audiots_statep;
+
+/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
+static char *audiots_name = TS_NAME;
+
+/*
+ * DDI Structures
+ */
+
+/* Device operations structure */
+static struct dev_ops audiots_dev_ops = {
+	DEVO_REV,		/* devo_rev */
+	0,			/* devo_refcnt */
+	NULL,			/* devo_getinfo */
+	nulldev,		/* devo_identify - obsolete */
+	nulldev,		/* devo_probe */
+	audiots_attach,		/* devo_attach */
+	audiots_detach,		/* devo_detach */
+	nodev,			/* devo_reset */
+	NULL,			/* devo_cb_ops */
+	NULL,			/* devo_bus_ops */
+	NULL,			/* devo_power */
+	audiots_quiesce,	/* devo_quiesce */
+};
+
+/* Linkage structure for loadable drivers */
+static struct modldrv audiots_modldrv = {
+	&mod_driverops,		/* drv_modops */
+	TS_MOD_NAME,		/* drv_linkinfo */
+	&audiots_dev_ops	/* drv_dev_ops */
+};
+
+/* Module linkage structure */
+static struct modlinkage audiots_modlinkage = {
+	MODREV_1,			/* ml_rev */
+	(void *)&audiots_modldrv,	/* ml_linkage */
+	NULL				/* NULL terminates the list */
+};
+
+
+/*
+ * NOTE: Grover OBP v4.0.166 and rev G of the ALI Southbridge chip force the
+ * audiots driver to use the upper 2 GB DMA address range. However to maintain
+ * backwards compatibility with older systems/OBP, we're going to try the full
+ * 4 GB DMA range.
+ *
+ * Eventually, this will be set back to using the proper high 2 GB DMA range.
+ */
+
+/* Device attribute structure - full 4 gig address range */
+static ddi_dma_attr_t audiots_attr = {
+	DMA_ATTR_VERSION,		/* version */
+	0x0000000000000000LL,		/* dlim_addr_lo */
+	0x00000000ffffffffLL,		/* dlim_addr_hi */
+	0x0000000000003fffLL,		/* DMA counter register - 16 bits */
+	0x0000000000000008LL,		/* DMA address alignment, 64-bit */
+	0x0000007f,			/* 1 through 64 byte burst sizes */
+	0x00000001,			/* min effective DMA size */
+	0x0000000000003fffLL,		/* maximum transfer size, 16k */
+	0x000000000000ffffLL,		/* segment boundary, 64k */
+	0x00000001,			/* s/g list length, no s/g */
+	0x00000001,			/* granularity of device, don't care */
+	0				/* DMA flags */
+};
+
+static ddi_device_acc_attr_t ts_acc_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static ddi_device_acc_attr_t ts_regs_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * _init()
+ *
+ * Description:
+ *	Driver initialization, called when driver is first loaded.
+ *	This is how access is initially given to all the static structures.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
+ *	mod_install() status, see mod_install(9f)
+ */
+int
+_init(void)
+{
+	int		error;
+
+	audio_init_ops(&audiots_dev_ops, TS_NAME);
+
+	/* initialize the soft state */
+	if ((error = ddi_soft_state_init(&audiots_statep,
+	    sizeof (audiots_state_t), 1)) != 0) {
+		audio_fini_ops(&audiots_dev_ops);
+		return (error);
+	}
+
+	if ((error = mod_install(&audiots_modlinkage)) != 0) {
+		audio_fini_ops(&audiots_dev_ops);
+		ddi_soft_state_fini(&audiots_statep);
+	}
+
+	return (error);
+}
+
+/*
+ * _fini()
+ *
+ * Description:
+ *	Module de-initialization, called when the driver is to be unloaded.
+ *
+ * Arguments:
+ *	None
+ *
+ * Returns:
+ *	mod_remove() status, see mod_remove(9f)
+ */
+int
+_fini(void)
+{
+	int		error;
+
+	if ((error = mod_remove(&audiots_modlinkage)) != 0) {
+		return (error);
+	}
+
+	/* free the soft state internal structures */
+	ddi_soft_state_fini(&audiots_statep);
+
+	/* clean up ops */
+	audio_fini_ops(&audiots_dev_ops);
+
+	return (0);
+}
+
+/*
+ * _info()
+ *
+ * Description:
+ *	Module information, returns infomation about the driver.
+ *
+ * Arguments:
+ *	modinfo *modinfop	Pointer to the opaque modinfo structure
+ *
+ * Returns:
+ *	mod_info() status, see mod_info(9f)
+ */
+int
+_info(struct modinfo *modinfop)
+{
+	int		error;
+
+	error = mod_info(&audiots_modlinkage, modinfop);
+
+	return (error);
+}
+
+
+/*
+ * audiots_attach()
+ *
+ * Description:
+ *	Attach an instance of the audiots driver. This routine does the
+ *	device dependent attach tasks.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audiots_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	audiots_state_t		*state;
+	int			instance;
+
+	instance = ddi_get_instance(dip);
+
+	switch (cmd) {
+	case DDI_ATTACH:
+		break;
+	case DDI_RESUME:
+
+		/* we've already allocated the state structure so get ptr */
+		if ((state = ddi_get_soft_state(audiots_statep, instance)) ==
+		    NULL) {
+			/* this will probably panic */
+			cmn_err(CE_WARN,
+			    "!%s%d: RESUME get soft state failed",
+			    audiots_name, instance);
+			return (DDI_FAILURE);
+		}
+
+		ASSERT(dip == state->ts_dip);
+
+		/* suspend/resume resets the chip, so we have no more faults */
+		if (state->ts_flags & TS_AUDIO_READ_FAILED) {
+			ddi_dev_report_fault(state->ts_dip,
+			    DDI_SERVICE_RESTORED,
+			    DDI_DEVICE_FAULT,
+			    "check port, gain, balance, and mute settings");
+			/* and clear the fault state flags */
+			state->ts_flags &=
+			    ~(TS_AUDIO_READ_FAILED|TS_READ_FAILURE_PRINTED);
+		}
+
+		audiots_power_up(state);
+		audiots_chip_init(state);
+		ac97_resume(state->ts_ac97);
+
+		mutex_enter(&state->ts_lock);
+		/*
+		 * Initialize/reset ports.  Done under the lock, to
+		 * avoid race with interrupt service routine.
+		 */
+		state->ts_suspended = B_FALSE;
+		for (int i = 0; i < TS_NUM_PORTS; i++) {
+			audiots_port_t	*port = state->ts_ports[i];
+			if (port != NULL) {
+				/* relocate any streams properly */
+				if (port->tp_engine)
+					audio_engine_reset(port->tp_engine);
+
+				/* do a hardware reset on the port */
+				audiots_reset_port(port);
+				if (port->tp_started) {
+					audiots_start_port(port);
+				} else {
+					audiots_stop_port(port);
+				}
+			}
+		}
+		mutex_exit(&state->ts_lock);
+
+		return (DDI_SUCCESS);
+
+	default:
+		cmn_err(CE_WARN, "!%s%d: attach() unknown command: 0x%x",
+		    audiots_name, instance, cmd);
+		return (DDI_FAILURE);
+	}
+
+	/* before we do anything make sure that we haven't had a h/w failure */
+	if (ddi_get_devstate(dip) == DDI_DEVSTATE_DOWN) {
+		cmn_err(CE_WARN, "%s%d: The audio hardware has "
+		    "been disabled.", audiots_name, instance);
+		cmn_err(CE_CONT, "Please reboot to restore audio.");
+		return (DDI_FAILURE);
+	}
+
+	/* we don't support high level interrupts in this driver */
+	if (ddi_intr_hilevel(dip, 0) != 0) {
+		cmn_err(CE_WARN, "!%s%d: unsupported high level interrupt",
+		    audiots_name, instance);
+		return (DDI_FAILURE);
+	}
+
+	/* allocate the state structure */
+	if (ddi_soft_state_zalloc(audiots_statep, instance) == DDI_FAILURE) {
+		cmn_err(CE_WARN, "!%s%d: soft state allocate failed",
+		    audiots_name, instance);
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * WARNING: From here on all errors require that we free memory,
+	 *	including the state structure.
+	 */
+
+	/* get the state structure - cannot fail */
+	state = ddi_get_soft_state(audiots_statep, instance);
+	ASSERT(state != NULL);
+
+	if ((state->ts_adev = audio_dev_alloc(dip, 0)) == NULL) {
+		cmn_err(CE_WARN, "unable to allocate audio dev");
+		goto error;
+	}
+
+	/* map in the registers, allocate DMA buffers, etc. */
+	if (audiots_map_regs(dip, state) == DDI_FAILURE) {
+		audio_dev_warn(state->ts_adev, "unable to map registers");
+		goto error;
+	}
+
+	/* initialize the audio state structures */
+	if (audiots_init_state(state, dip) == DDI_FAILURE) {
+		audio_dev_warn(state->ts_adev, "init state structure failed");
+		goto error;
+	}
+
+	/* power up */
+	audiots_power_up(state);
+
+	/* initialize the audio controller */
+	audiots_chip_init(state);
+
+	/* initialize the AC-97 Codec */
+	ac97_init(state->ts_ac97, state->ts_adev);
+
+	/* put the engine interrupts into a known state -- all off */
+	ddi_put32(state->ts_acch, &state->ts_regs->aud_regs.ap_ainten,
+	    TS_ALL_DMA_OFF);
+
+	/* call the framework attach routine */
+	if (audio_dev_register(state->ts_adev) != DDI_SUCCESS) {
+		audio_dev_warn(state->ts_adev, "unable to register audio");
+		goto error;
+	}
+
+	/* set up kernel statistics */
+	state->ts_ksp = kstat_create(TS_NAME, instance, TS_NAME,
+	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT);
+	if (state->ts_ksp != NULL) {
+		kstat_install(state->ts_ksp);
+	}
+
+	/* set up the interrupt handler */
+	if (ddi_add_intr(dip, 0, NULL, NULL, audiots_intr,
+	    (caddr_t)state) != DDI_SUCCESS) {
+		audio_dev_warn(state->ts_adev,
+		    "failed to register interrupt handler");
+		goto error;
+	}
+	state->ts_flags |= TS_INTR_INSTALLED;
+
+	/* everything worked out, so report the device */
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	audiots_destroy(state);
+	return (DDI_FAILURE);
+}
+
+/*
+ * audiots_detach()
+ *
+ * Description:
+ *	Detach an instance of the audiots driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_detach_cmd_t cmd	Detach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was detached
+ *	DDI_FAILURE		The driver couldn't be detached
+ */
+static int
+audiots_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	audiots_state_t		*state;
+	int			instance;
+
+	instance = ddi_get_instance(dip);
+
+	/* get the state structure */
+	if ((state = ddi_get_soft_state(audiots_statep, instance)) == NULL) {
+		cmn_err(CE_WARN, "!%s%d: detach get soft state failed",
+		    audiots_name, instance);
+		return (DDI_FAILURE);
+	}
+
+	switch (cmd) {
+	case DDI_DETACH:
+		break;
+	case DDI_SUSPEND:
+
+		ac97_suspend(state->ts_ac97);
+
+		mutex_enter(&state->ts_lock);
+
+		state->ts_suspended = B_TRUE;	/* stop new ops */
+
+		/* we may already be powered down, so only save state if up */
+
+		/* stop playing and recording */
+		(void) audiots_stop_everything(state);
+
+		mutex_exit(&state->ts_lock);
+
+		return (DDI_SUCCESS);
+
+	default:
+		return (DDI_FAILURE);
+	}
+
+	/* attempt to unregister from the framework first */
+	if (audio_dev_unregister(state->ts_adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	audiots_destroy(state);
+
+	return (DDI_SUCCESS);
+
+}
+
+/*
+ * audiots_quiesce()
+ *
+ * Description:
+ *	Quiesce an instance of the audiots driver. Stops all DMA and
+ *	interrupts.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was quiesced
+ *	DDI_SUCCESS		The driver was NOT quiesced
+ */
+static int
+audiots_quiesce(dev_info_t *dip)
+{
+	audiots_state_t		*state;
+	int			instance;
+
+	instance = ddi_get_instance(dip);
+
+	/* get the state structure */
+	if ((state = ddi_get_soft_state(audiots_statep, instance)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	audiots_stop_everything(state);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiots_power_up()
+ *
+ * Description
+ *	Ensure that the device is running in PCI power state D0.
+ */
+static void
+audiots_power_up(audiots_state_t *state)
+{
+	ddi_acc_handle_t	pcih = state->ts_pcih;
+	uint8_t			ptr;
+	uint16_t		pmcsr;
+
+	if ((pci_config_get16(pcih, PCI_CONF_STAT) & PCI_STAT_CAP) == 0) {
+		/* does not implement PCI capabilities -- no PM */
+		return;
+	}
+
+	ptr = pci_config_get8(pcih, PCI_CONF_CAP_PTR);
+	for (;;) {
+		if (ptr == PCI_CAP_NEXT_PTR_NULL) {
+			/* PM capability not found */
+			return;
+		}
+		if (pci_config_get8(pcih, ptr + PCI_CAP_ID) == PCI_CAP_ID_PM) {
+			/* found it */
+			break;
+		}
+		ptr = pci_config_get8(pcih, ptr + PCI_CAP_NEXT_PTR);
+	}
+
+	/* if we got here, then got valid PMCSR pointer */
+	ptr += PCI_PMCSR;
+
+	/* check to see if we are already in state D0 */
+	pmcsr = pci_config_get16(pcih, ptr);
+	if ((pmcsr & PCI_PMCSR_STATE_MASK) != PCI_PMCSR_D0) {
+
+		/* D3hot (or any other state) -> D0 */
+		pmcsr &= ~PCI_PMCSR_STATE_MASK;
+		pmcsr |= PCI_PMCSR_D0;
+		pci_config_put16(pcih, ptr, pmcsr);
+	}
+
+	/*
+	 * Wait for it to power up - PCI spec says 10 ms is enough.
+	 * We double it.  Note that no locks are held when this routine
+	 * is called, so we can sleep (we are in attach context only).
+	 *
+	 * We do this delay even if already powerd up, just to make
+	 * sure we aren't seeing something that *just* transitioned
+	 * into D0 state.
+	 */
+	delay(drv_usectohz(TS_20MS));
+
+	/* clear PME# flag */
+	pmcsr = pci_config_get16(pcih, ptr);
+	pci_config_put16(pcih, ptr, pmcsr | PCI_PMCSR_PME_STAT);
+}
+
+/*
+ * audiots_chip_init()
+ *
+ * Description:
+ *	Initialize the audio core.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_chip_init(audiots_state_t *state)
+{
+	ddi_acc_handle_t	handle = state->ts_acch;
+	audiots_regs_t		*regs = state->ts_regs;
+	int			str;
+
+	/* start with all interrupts & dma channels disabled */
+	ddi_put32(handle, &regs->aud_regs.ap_stop, TS_ALL_DMA_ENGINES);
+	ddi_put32(handle, &regs->aud_regs.ap_ainten, TS_ALL_DMA_OFF);
+
+	/* set global music and wave volume to 0dB */
+	ddi_put32(handle, &regs->aud_regs.ap_volume, 0x0);
+
+	/* enable end interrupts for all channels. */
+	ddi_put32(handle, &regs->aud_regs.ap_cir_gc, AP_CIR_GC_ENDLP_IE);
+
+	/* for each stream, set gain and vol settings */
+	for (str = 0; str < TS_MAX_HW_CHANNELS; str++) {
+		/*
+		 * Set volume to all off, 1st left and then right.
+		 * These are never changed, so we don't have to save them.
+		 */
+		ddi_put16(handle,
+		    &regs->aud_ram[str].eram.eram_gvsel_pan_vol,
+		    (ERAM_WAVE_VOL|ERAM_PAN_LEFT|ERAM_PAN_0dB|
+		    ERAM_VOL_MAX_ATTEN));
+		ddi_put16(handle,
+		    &regs->aud_ram[str].eram.eram_gvsel_pan_vol,
+		    (ERAM_WAVE_VOL|ERAM_PAN_RIGHT|ERAM_PAN_0dB|
+		    ERAM_VOL_MAX_ATTEN));
+
+		/*
+		 * The envelope engine *MUST* remain in still mode (off).
+		 * Otherwise bad things like gain randomly disappearing might
+		 * happen. See bug #4332773.
+		 */
+
+		ddi_put32(handle, &regs->aud_ram[str].eram.eram_ebuf1,
+		    ERAM_EBUF_STILL);
+		ddi_put32(handle, &regs->aud_ram[str].eram.eram_ebuf2,
+		    ERAM_EBUF_STILL);
+
+		/* program the initial eram and aram rate */
+		ddi_put16(handle, &regs->aud_ram[str].aram.aram_delta,
+		    1 << TS_SRC_SHIFT);
+		ddi_put16(handle, &regs->aud_ram[str].eram.eram_ctrl_ec,
+		    ERAM_16_BITS | ERAM_STEREO | ERAM_LOOP_MODE |
+		    ERAM_SIGNED_PCM);
+	}
+
+	/* program channel 31 for record */
+	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_global_control,
+	    (AP_CLOGAL_CTRL_E_PCMIN_CH31|AP_CLOGAL_CTRL_PCM_OUT_AC97|
+	    AP_CLOGAL_CTRL_MMC_FROM_MIXER|AP_CLOGAL_CTRL_PCM_OUT_TO_AC97));
+
+	/* do a warm reset, which powers up the Codec */
+	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_sctrl,
+	    AP_SCTRL_WRST_CODEC);
+	drv_usecwait(2);
+	AND_SET_WORD(handle, &state->ts_regs->aud_regs.ap_sctrl,
+	    ~AP_SCTRL_WRST_CODEC);
+
+	/* do a warm reset via the Codec, yes, I'm being paranoid! */
+	audiots_set_ac97(state, AC97_RESET_REGISTER, 0);
+
+	/* Make sure the Codec is powered up. */
+	int i = TS_WAIT_CNT;
+	while ((audiots_get_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER) &
+	    PCSR_POWERD_UP) != PCSR_POWERD_UP && i--) {
+		drv_usecwait(1);
+	}
+
+}
+
+/*
+ * audiots_get_ac97()
+ *
+ * Description:
+ *	Get the value in the specified AC-97 Codec register. There is a
+ *	bug in silicon which forces us to do multiple reads of the Codec's
+ *	register. This algorithm was provided by T2 and is heuristic in
+ *	nature. Unfortunately we have no guarantees that the real answer
+ *	isn't 0x0000, which is what we get when a read fails. So we loop
+ *	TS_LOOP_CNT times before we give up. We just have to hope this is
+ *	sufficient to give us the correct value.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *	int		reg		AC-97 register number
+ *
+ * Returns:
+ *	unsigned short		The value in the specified register
+ */
+static uint16_t
+audiots_get_ac97(void *arg, uint8_t reg)
+{
+	audiots_state_t		*state = arg;
+	ddi_acc_handle_t	handle = state->ts_acch;
+	uint16_t		*data;
+	int			count;
+	int			delay;
+	uint16_t		first;
+	uint16_t		next;
+
+	if (state->ts_revid == AC_REV_ID1) {
+		data = &state->ts_regs->aud_regs.ap_acrd_35D_data;
+	} else {
+		data = &state->ts_regs->aud_regs.ap_acrdwr_data;
+	}
+
+	/* make sure the register is good */
+	reg &= AP_ACRD_INDEX_MASK;
+	for (count = TS_LOOP_CNT; count--; ) {
+		if ((first = audiots_read_ac97(state, reg)) != 0) {
+			next = first;
+			break;
+		}
+
+		delay = TS_DELAY_CNT;
+		while (delay--) {
+			(void) ddi_get16(handle, data);
+		}
+
+		if ((next = audiots_read_ac97(state, reg)) != 0) {
+			break;
+		}
+	}
+
+	/*
+	 * Arggg, if you let the next read happen too soon then it fails.
+	 * 12 usec fails, 13 usec succeeds. So set it to 20 for safety.
+	 */
+	drv_usecwait(TS_20US);
+
+	return (next);
+
+}
+
+/*
+ * audiots_init_state()
+ *
+ * Description:
+ *	This routine initializes the audio driver's state structure.
+ *	This includes reading the properties.
+ *
+ *	CAUTION: This routine cannot allocate resources, unless it frees
+ *		them before returning for an error. Also, error_destroy:
+ *		in audiots_attach() would need to be fixed as well.
+ *
+ *	NOTE: birdsnest supports CD ROM input. We check for the cdrom
+ *		property. If there we turn it on.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *	dev_info_t	*dip		Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS			State structure initialized
+ *	DDI_FAILURE			State structure not initialized
+ */
+static int
+audiots_init_state(audiots_state_t *state, dev_info_t *dip)
+{
+	state->ts_ac97 = ac97_alloc(dip, audiots_get_ac97,
+	    audiots_set_ac97, state);
+
+	if (state->ts_ac97 == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	/* save the device info pointer */
+	state->ts_dip = dip;
+
+	/* get the iblock cookie needed for interrupt context */
+	if (ddi_get_iblock_cookie(dip, 0, &state->ts_iblock) != DDI_SUCCESS) {
+		audio_dev_warn(state->ts_adev,
+		    "cannot get iblock cookie");
+		return (DDI_FAILURE);
+	}
+
+	/* initialize the state mutexes and condition variables */
+	mutex_init(&state->ts_lock, NULL, MUTEX_DRIVER, state->ts_iblock);
+	state->ts_flags |= TS_MUTEX_INIT;
+
+	for (int i = 0; i < TS_NUM_PORTS; i++) {
+		if (audiots_alloc_port(state, i) != DDI_SUCCESS) {
+			return (DDI_FAILURE);
+		}
+	}
+	/* init power management state */
+	state->ts_suspended = B_FALSE;
+
+	return (DDI_SUCCESS);
+
+}
+
+/*
+ * audiots_intr()
+ *
+ * Description:
+ *	Interrupt service routine for both play and record. For play we
+ *	get the next buffers worth of audio. For record we send it on to
+ *	the mixer.
+ *
+ *	NOTE: This device needs to make sure any PIO access required to clear
+ *	its interrupt has made it out on the PCI bus before returning from its
+ *	interrupt handler so that the interrupt has been deasserted. This is
+ *	done by rereading the address engine interrupt register.
+ *
+ * Arguments:
+ *	caddr_t		T	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+audiots_intr(caddr_t T)
+{
+	audiots_state_t		*state = (void *)T;
+	audiots_regs_t		*regs = state->ts_regs;
+	ddi_acc_handle_t	handle = state->ts_acch;
+	uint32_t		interrupts;
+
+	mutex_enter(&state->ts_lock);
+
+	if (state->ts_suspended) {
+		mutex_exit(&state->ts_lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	interrupts = ddi_get32(handle, &regs->aud_regs.ap_aint);
+	if (interrupts == 0) {
+		mutex_exit(&state->ts_lock);
+		/* no interrupts to process, so it's not us */
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	/*
+	 * Clear the interrupts to acknowledge.  Also, reread the
+	 * interrupt reg to ensure that PIO write has completed.
+	 */
+	ddi_put32(handle, &regs->aud_regs.ap_aint, interrupts);
+	(void) ddi_get32(handle, &regs->aud_regs.ap_aint);
+
+	/* update the kernel interrupt statistics */
+	if (state->ts_ksp) {
+		TS_KIOP(state)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	mutex_exit(&state->ts_lock);
+
+	for (int i = 0; i < TS_NUM_PORTS; i++) {
+		audiots_port_t	*port = state->ts_ports[i];
+
+		if (((interrupts & port->tp_int_mask) == 0) ||
+		    (!port->tp_started))
+			continue;
+
+		if (i == TS_INPUT_PORT) {
+			audio_engine_produce(port->tp_engine);
+		} else {
+			audio_engine_consume(port->tp_engine);
+		}
+	}
+
+	return (DDI_INTR_CLAIMED);
+
+}
+
+/*
+ * audiots_map_regs()
+ *
+ * Description:
+ *	This routine maps the registers in.
+ *
+ *	Once the config space registers are mapped in we determine if the
+ *	audio core may be power managed. It should, but if it doesn't,
+ *	then trying to may cause the core to hang.
+ *
+ *	CAUTION: Make sure all errors call audio_dev_warn().
+ *
+ * Arguments:
+ *	dev_info_t	*dip            Pointer to the device's devinfo
+ *	audiots_state_t	*state          The device's state structure
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+audiots_map_regs(dev_info_t *dip, audiots_state_t *state)
+{
+	char	rev[16];
+	char	*name;
+
+	/* map in the registers, the config and memory mapped registers */
+	if (pci_config_setup(dip, &state->ts_pcih) != DDI_SUCCESS) {
+		audio_dev_warn(state->ts_adev,
+		    "unable to map PCI configuration space");
+		return (DDI_FAILURE);
+	}
+
+	/* Read the Audio Controller's vendor, device, and revision IDs */
+	state->ts_devid =
+	    (pci_config_get16(state->ts_pcih, PCI_CONF_VENID) << 16) |
+	    pci_config_get16(state->ts_pcih, PCI_CONF_DEVID);
+	state->ts_revid = pci_config_get8(state->ts_pcih, PCI_CONF_REVID);
+
+	if (ddi_regs_map_setup(dip, TS_MEM_MAPPED_REGS,
+	    (caddr_t *)&state->ts_regs, 0, 0, &ts_regs_attr, &state->ts_acch) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(state->ts_adev,
+		    "unable to map PCI device registers");
+		return (DDI_FAILURE);
+	}
+
+	switch (state->ts_devid) {
+	case 0x10b95451:
+		name = "ALI M5451";
+		break;
+	default:
+		name = "audiots";
+		break;
+	}
+	(void) snprintf(rev, sizeof (rev), "Rev %x", state->ts_revid);
+	audio_dev_set_description(state->ts_adev, name);
+	audio_dev_set_version(state->ts_adev, rev);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiots_alloc_port()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use. It then binds each of the buffers to its
+ *	respective handle, getting a DMA cookie.
+ *
+ *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
+ *		memory. This means these calls should always succeed.
+ *
+ *	NOTE: ddi_dma_alloc_handle() attempts to use the full 4 GB DMA address
+ *		range. This is to work around Southbridge rev E/G OBP issues.
+ *		(See Grover OBP note above)
+ *
+ *	CAUTION: Make sure all errors call audio_dev_warn().
+ *
+ * Arguments:
+ *	audiots_port_t	*state          The port structure for a device stream
+ *	int		num		The port number
+ *
+ * Returns:
+ *	DDI_SUCCESS		DMA resources mapped
+ *	DDI_FAILURE		DMA resources not successfully mapped
+ */
+int
+audiots_alloc_port(audiots_state_t *state, int num)
+{
+	audiots_port_t		*port;
+	dev_info_t		*dip = state->ts_dip;
+	audio_dev_t		*adev = state->ts_adev;
+	char			*prop;
+	int			dir;
+	unsigned		caps;
+	ddi_dma_cookie_t	cookie;
+	unsigned		count;
+	int			rc;
+	ddi_acc_handle_t	regsh = state->ts_acch;
+	uint32_t		*gcptr = &state->ts_regs->aud_regs.ap_cir_gc;
+	char			*namestr;
+
+	port = kmem_zalloc(sizeof (*port), KM_SLEEP);
+	state->ts_ports[num] = port;
+	port->tp_num = num;
+	port->tp_state = state;
+	port->tp_started = B_FALSE;
+	port->tp_rate = 48000;
+
+	if (num == TS_INPUT_PORT) {
+		prop = "record-interrupts";
+		dir = DDI_DMA_READ;
+		caps = ENGINE_INPUT_CAP;
+		port->tp_dma_stream = 31;
+		port->tp_int_stream = 2;
+		port->tp_sync_dir = DDI_DMA_SYNC_FORKERNEL;
+	} else {
+		prop = "play-interrupts";
+		dir = DDI_DMA_WRITE;
+		caps = ENGINE_OUTPUT_CAP;
+		port->tp_dma_stream = 0;
+		port->tp_int_stream = 1;
+		port->tp_sync_dir = DDI_DMA_SYNC_FORDEV;
+	}
+	port->tp_int_mask = (1U << port->tp_int_stream);
+	port->tp_dma_mask = (1U << port->tp_dma_stream);
+
+	/* get the number of interrupts per second */
+	port->tp_intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, prop, TS_INTS);
+
+	/* make sure the values are good */
+	if (port->tp_intrs < TS_MIN_INTS) {
+		audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+		    prop, port->tp_intrs, TS_INTS);
+		port->tp_intrs = TS_INTS;
+	} else if (port->tp_intrs > TS_MAX_INTS) {
+		audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+		    prop, port->tp_intrs, TS_INTS);
+		port->tp_intrs = TS_INTS;
+	}
+
+	/*
+	 * Now allocate space.  We configure for the worst case.  The
+	 * worst (biggest) case is 48000 kHz, at 4 bytes per frame
+	 * (16-bit stereo), with the lowest interrupt frequency.  We
+	 * need two fragments though, and each half has to be rounded
+	 * up to allow for alignment considerations.
+	 */
+
+	/* allocate dma handle */
+	rc = ddi_dma_alloc_handle(dip, &audiots_attr, DDI_DMA_SLEEP,
+	    NULL, &port->tp_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	/* allocate DMA buffer */
+	rc = ddi_dma_mem_alloc(port->tp_dmah, TS_BUFSZ, &ts_acc_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &port->tp_kaddr,
+	    &port->tp_size, &port->tp_acch);
+	if (rc == DDI_FAILURE) {
+		audio_dev_warn(adev, "dma_mem_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	/* bind DMA buffer */
+	rc = ddi_dma_addr_bind_handle(port->tp_dmah, NULL,
+	    port->tp_kaddr, port->tp_size, dir|DDI_DMA_CONSISTENT,
+	    DDI_DMA_SLEEP, NULL, &cookie, &count);
+	if (rc != DDI_DMA_MAPPED) {
+		audio_dev_warn(adev,
+		    "ddi_dma_addr_bind_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	ASSERT(count == 1);
+
+	port->tp_paddr = cookie.dmac_address;
+	if ((unsigned)port->tp_paddr & 0x80000000U) {
+		ddi_put32(regsh, gcptr,
+		    ddi_get32(regsh, gcptr) | AP_CIR_GC_SYS_MEM_4G_ENABLE);
+	} else {
+		ddi_put32(regsh, gcptr,
+		    ddi_get32(regsh, gcptr) & ~(AP_CIR_GC_SYS_MEM_4G_ENABLE));
+	}
+	port->tp_engine = audio_engine_alloc(&audiots_engine_ops, caps);
+	if (port->tp_engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(port->tp_engine, port);
+	audio_dev_add_engine(adev, port->tp_engine);
+
+	state->ts_swapped = B_FALSE;
+
+	/*
+	 * SPARCLE platform specific hack.  For reasons that I can't
+	 * seem to fathom, the SPARCLE platform only gets the
+	 * endianness correct when transferring whole 32-bit words and
+	 * using little endian mapping.  That isn't compatible with
+	 * the audio framework's access mode, so we have to set up
+	 * explicit swapping of the left and right channels.
+	 *
+	 * The real mystery here is why this is required for Tadpole
+	 * SPARCLE, but not for any other standard Sun platforms that
+	 * I've tested.
+	 *
+	 * "swap_channels" property can be used in driver.conf to
+	 * force the left/right as well.
+	 */
+	rc = ddi_prop_lookup_string(DDI_DEV_T_ANY, ddi_root_node(), 0,
+	    "name", &namestr);
+	if (rc == DDI_PROP_SUCCESS) {
+		if (strcmp(namestr, "TAD,SPARCLE") == 0) {
+			state->ts_swapped = B_TRUE;
+		}
+		ddi_prop_free(namestr);
+	}
+
+	state->ts_swapped = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_NOTPROM | DDI_PROP_DONTPASS, "swap_channels",
+	    state->ts_swapped);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiots_read_ac97()
+ *
+ * Description:
+ *	This routine actually reads the AC-97 Codec's register. It may
+ *	be called several times to succeed.
+ *
+ * NOTE:
+ * 	Revision M1535D B1-C of the ALI SouthBridge includes a workaround for
+ *	the broken busy flag. Resetting the busy flag requires a software tweak
+ *	to go with the worked around hardware. When we detect failure, we make
+ *	10 attempts to reset the chip before we fail. This should reset the new
+ *	SB systems. On all SB systems, this will increse the read delay
+ *	slightly, but shouldn't bother it otherwise.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *	int		reg		AC-97 register number
+ *
+ * Returns:
+ *	unsigned short		The value in the specified register
+ */
+static uint16_t
+audiots_read_ac97(audiots_state_t *state, int reg)
+{
+	ddi_acc_handle_t	acch = state->ts_acch;
+	uint16_t		*addr;
+	uint16_t		*data;
+	uint32_t		*stimer = &state->ts_regs->aud_regs.ap_stimer;
+	uint32_t		chk1;
+	uint32_t		chk2;
+	int			resets = 0;
+	int			i;
+
+	if (state->ts_revid == AC_REV_ID1) {
+		addr = &state->ts_regs->aud_regs.ap_acrd_35D_reg;
+		data = &state->ts_regs->aud_regs.ap_acrd_35D_data;
+	} else {
+		addr = &state->ts_regs->aud_regs.ap_acrdwr_reg;
+		data = &state->ts_regs->aud_regs.ap_acrdwr_data;
+	}
+
+first_read:
+	/* wait for ready to send read request */
+	for (i = 0; i < TS_READ_TRIES; i++) {
+		if (!(ddi_get16(acch, addr) & AP_ACRD_R_READ_BUSY)) {
+			break;
+		}
+		/* don't beat on the bus */
+		drv_usecwait(1);
+	}
+	if (i >= TS_READ_TRIES) {
+		if (resets < TS_RESET_TRIES) {
+			/* Attempt to reset */
+			drv_usecwait(TS_20US);
+			ddi_put16(acch, addr, TS_SB_RESET);
+			resets++;
+			goto first_read;
+		} else {
+			state->ts_flags |= TS_AUDIO_READ_FAILED;
+			if (!(state->ts_flags & TS_READ_FAILURE_PRINTED)) {
+				ddi_dev_report_fault(state->ts_dip,
+				    DDI_SERVICE_LOST, DDI_DEVICE_FAULT,
+				    "Unable to communicate with AC97 CODEC");
+				audio_dev_warn(state->ts_adev,
+				    "The audio AC97 register has timed out.");
+				audio_dev_warn(state->ts_adev,
+				    "Audio is now disabled.");
+				audio_dev_warn(state->ts_adev,
+				    "Please reboot to restore audio.");
+
+				/* Don't flood the console */
+				state->ts_flags |= TS_READ_FAILURE_PRINTED;
+			}
+		}
+		return (0);
+	}
+
+	/* program the register to read */
+	ddi_put16(acch, addr, (reg|AP_ACRD_W_PRIMARY_CODEC|
+	    AP_ACRD_W_READ_MIXER_REG|AP_ACRD_W_AUDIO_READ_REQ&
+	    (~AP_ACWR_W_SELECT_WRITE)));
+
+	/* hardware bug work around */
+	chk1 = ddi_get32(acch, stimer);
+	chk2 = ddi_get32(acch, stimer);
+	i = TS_WAIT_CNT;
+	while (chk1 == chk2 && i) {
+		chk2 = ddi_get32(acch, stimer);
+		i--;
+	}
+	OR_SET_SHORT(acch, addr, AP_ACRD_W_READ_MIXER_REG);
+	resets = 0;
+
+second_read:
+	/* wait again for read to send read request */
+	for (i = 0; i < TS_READ_TRIES; i++) {
+		if (!(ddi_get16(acch, addr) & AP_ACRD_R_READ_BUSY)) {
+			break;
+		}
+		/* don't beat on the bus */
+		drv_usecwait(1);
+	}
+	if (i >= TS_READ_TRIES) {
+		if (resets < TS_RESET_TRIES) {
+			/* Attempt to reset */
+			drv_usecwait(TS_20US);
+			ddi_put16(acch, addr, TS_SB_RESET);
+			resets++;
+			goto second_read;
+		} else {
+			state->ts_flags |= TS_AUDIO_READ_FAILED;
+			if (!(state->ts_flags & TS_READ_FAILURE_PRINTED)) {
+				ddi_dev_report_fault(state->ts_dip,
+				    DDI_SERVICE_LOST, DDI_DEVICE_FAULT,
+				    "Unable to communicate with AC97 CODEC");
+				audio_dev_warn(state->ts_adev,
+				    "The audio AC97 register has timed out.");
+				audio_dev_warn(state->ts_adev,
+				    "Audio is now disabled.");
+				audio_dev_warn(state->ts_adev,
+				    "Please reboot to restore audio.");
+
+				/* Don't flood the console */
+				state->ts_flags |= TS_READ_FAILURE_PRINTED;
+			}
+		}
+		return (0);
+	}
+
+	return (ddi_get16(acch, data));
+
+}	/* audiots_read_ac97() */
+
+/*
+ * audiots_set_ac97()
+ *
+ * Description:
+ *	Set the value in the specified AC-97 Codec register. Just like
+ *	reading the AC-97 Codec, it is possible there is a problem writing
+ *	it as well. So we loop.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *	int		reg		AC-97 register number
+ *	uint16_t	value		The value to write
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_set_ac97(void *arg, uint8_t reg8, uint16_t data)
+{
+	audiots_state_t	*state = arg;
+	ddi_acc_handle_t handle = state->ts_acch;
+	uint16_t	*data_addr = &state->ts_regs->aud_regs.ap_acrdwr_data;
+	uint16_t	*reg_addr = &state->ts_regs->aud_regs.ap_acrdwr_reg;
+	int		count;
+	int		i;
+	uint16_t	tmp_short;
+	uint16_t	reg = reg8;
+
+	reg &= AP_ACWR_INDEX_MASK;
+
+	/* Don't touch the reserved bits on the pre 35D+ SouthBridge */
+	if (state->ts_revid == AC_REV_ID1) {
+		reg |= AP_ACWR_W_PRIMARY_CODEC|AP_ACWR_W_WRITE_MIXER_REG;
+	} else {
+		reg |= AP_ACWR_W_PRIMARY_CODEC|AP_ACWR_W_WRITE_MIXER_REG|
+		    AP_ACWR_W_SELECT_WRITE;
+	}
+
+	for (count = TS_LOOP_CNT; count--; ) {
+		/* wait for ready to write */
+		for (i = 0; i < TS_WAIT_CNT; i++) {
+			if (!(ddi_get16(handle, reg_addr) &
+			    AP_ACWR_R_WRITE_BUSY)) {
+				/* ready to write */
+				ddi_put16(handle, reg_addr, reg);
+
+				/* Write the data */
+				ddi_put16(handle, data_addr, data);
+				break;
+			}
+		}
+		if (i >= TS_WAIT_CNT) {
+			/* try again */
+			continue;
+		}
+
+		/* wait for write to complete */
+		for (i = 0; i < TS_WAIT_CNT; i++) {
+			if (!(ddi_get16(handle, reg_addr) &
+			    AP_ACWR_R_WRITE_BUSY)) {
+				/* done writing */
+				break;
+			}
+		}
+
+		/* verify the value written */
+		tmp_short = audiots_get_ac97(state, reg8);
+		if (data == tmp_short) {
+			/* successfully loaded, so we can return */
+			return;
+		}
+	}
+
+}	/* audiots_set_ac97() */
+
+/*
+ * audiots_reset_port()
+ *
+ * Description:
+ *	Initializes the hardware for a DMA engine.
+ *	We only support stereo 16-bit linear PCM (signed native endian).
+ *
+ *	The audio core uses a single DMA buffer which is divided into two
+ *	halves. An interrupt is generated when the middle of the buffer has
+ *	been reached and at the end. The audio core resets the pointer back
+ *	to the beginning automatically. After the interrupt the driver clears
+ *	the buffer and asks the mixer for more audio samples. If there aren't
+ *	enough then silence is played out.
+ *
+ * Arguments:
+ *	audiots_port_t	*port		The DMA engine to reset
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_reset_port(audiots_port_t *port)
+{
+	audiots_state_t		*state = port->tp_state;
+	ddi_acc_handle_t	handle = state->ts_acch;
+	audiots_regs_t		*regs = state->ts_regs;
+	audiots_aram_t		*aram;
+	audiots_eram_t		*eram;
+	unsigned		delta;
+	uint16_t		ctrl;
+	uint16_t		gvsel;
+	uint16_t		eso;
+
+	if (state->ts_suspended)
+		return;
+
+	port->tp_cso = 0;
+
+	gvsel = ERAM_WAVE_VOL | ERAM_PAN_0dB | ERAM_VOL_DEFAULT;
+	ctrl = ERAM_16_BITS | ERAM_STEREO | ERAM_LOOP_MODE | ERAM_SIGNED_PCM;
+	for (int i = 0; i < 2; i++) {
+
+		delta = (port->tp_rate << TS_SRC_SHIFT) / TS_RATE;
+
+		if (i == 0) {
+			/* first do the DMA stream */
+			aram = &regs->aud_ram[port->tp_dma_stream].aram;
+			eram = &regs->aud_ram[port->tp_dma_stream].eram;
+			if (port->tp_num == TS_INPUT_PORT) {
+				delta = (TS_RATE << TS_SRC_SHIFT) /
+				    port->tp_rate;
+			}
+			eso = port->tp_nframes - 1;
+		} else {
+			/* else do the interrupt stream */
+			aram = &regs->aud_ram[port->tp_int_stream].aram;
+			eram = &regs->aud_ram[port->tp_int_stream].eram;
+			/* interrupt stream is silent */
+			gvsel |= ERAM_VOL_MAX_ATTEN;
+			eso = port->tp_fragfr - 1;
+		}
+
+		/* program the sample rate */
+		ddi_put16(handle, &aram->aram_delta, (uint16_t)delta);
+
+		/* program the precision, number of channels and loop mode */
+		ddi_put16(handle, &eram->eram_ctrl_ec, ctrl);
+
+		/* program the volume settings */
+		ddi_put16(handle, &eram->eram_gvsel_pan_vol, gvsel);
+
+		/* set ALPHA and FMS to 0 */
+		ddi_put16(handle, &aram->aram_alpha_fms, 0x0);
+
+		/* set CSO to 0 */
+		ddi_put16(handle, &aram->aram_cso, 0x0);
+
+		/* set LBA */
+		ddi_put32(handle, &aram->aram_cptr_lba,
+		    port->tp_paddr & ARAM_LBA_MASK);
+
+		/* set ESO */
+		ddi_put16(handle, &aram->aram_eso, eso);
+	}
+
+	/* stop the DMA & interrupt engines */
+	ddi_put32(handle, &regs->aud_regs.ap_stop,
+	    port->tp_int_mask | port->tp_dma_mask);
+
+	/* enable interrupts */
+	OR_SET_WORD(handle, &regs->aud_regs.ap_ainten, port->tp_int_mask);
+}
+
+/*
+ * audiots_open()
+ *
+ * Description:
+ *	Opens a DMA engine for use.  Will also ensure the device is powered
+ *	up if not already done so.
+ *
+ * Arguments:
+ *	void		*arg		The DMA engine to set up
+ *	int		flag		Open flags
+ *	unsigned	*fragfrp	Receives number of frames per fragment
+ *	unsigned	*nfragsp	Receives number of fragments
+ *	caddr_t		*bufp		Receives kernel data buffer
+ *
+ * Returns:
+ *	0	on success
+ *	errno	on failure
+ */
+static int
+audiots_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	audiots_port_t	*port = arg;
+	unsigned	nfrag;
+
+	_NOTE(ARGUNUSED(flag));
+
+	/*
+	 * Round up - we have to have a sample that is a whole number
+	 * of 64-bit words.  Since our frames are 4 bytes wide, we
+	 * just need an even number of frames.
+	 */
+	port->tp_fragfr = port->tp_rate / port->tp_intrs;
+	port->tp_fragfr = (port->tp_fragfr + 1) & ~1;
+	nfrag = port->tp_size / (port->tp_fragfr * TS_FRAMESZ);
+	port->tp_nframes = nfrag * port->tp_fragfr;
+	port->tp_started = B_FALSE;
+	port->tp_count = 0;
+	port->tp_cso = 0;
+	*fragfrp = port->tp_fragfr;
+	*nfragsp = nfrag;
+	*bufp = port->tp_kaddr;
+
+	/*
+	 * This should always be true because we used a worst case
+	 * assumption when calculating the port->tp_size.
+	 */
+	ASSERT((port->tp_fragfr * nfrag) <= port->tp_size);
+
+	mutex_enter(&port->tp_state->ts_lock);
+	audiots_reset_port(port);
+	mutex_exit(&port->tp_state->ts_lock);
+
+	return (0);
+}
+
+/*
+ * audiots_close()
+ *
+ * Description:
+ *	Closes an audio DMA engine that was previously opened.  Since
+ *	nobody is using it, we take this opportunity to possibly power
+ *	down the entire device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to shut down
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_close(void *arg)
+{
+	audiots_port_t	*port = arg;
+	audiots_state_t	*state = port->tp_state;
+
+	mutex_enter(&state->ts_lock);
+	audiots_stop_port(port);
+	port->tp_started = B_FALSE;
+	mutex_exit(&state->ts_lock);
+}
+
+/*
+ * audiots_stop()
+ *
+ * Description:
+ *	This is called by the framework to stop a port that is
+ *	transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to stop
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_stop(void *arg)
+{
+	audiots_port_t	*port = arg;
+	audiots_state_t	*state = port->tp_state;
+
+	mutex_enter(&state->ts_lock);
+	if (port->tp_started) {
+		audiots_stop_port(port);
+	}
+	port->tp_started = B_FALSE;
+	mutex_exit(&state->ts_lock);
+}
+
+/*
+ * audiots_start()
+ *
+ * Description:
+ *	This is called by the framework to start a port transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to start
+ *
+ * Returns:
+ *	0 	on success (never fails, errno if it did)
+ */
+static int
+audiots_start(void *arg)
+{
+	audiots_port_t	*port = arg;
+	audiots_state_t	*state = port->tp_state;
+
+	mutex_enter(&state->ts_lock);
+	if (!port->tp_started) {
+		audiots_start_port(port);
+		port->tp_started = B_TRUE;
+	}
+	mutex_exit(&state->ts_lock);
+	return (0);
+}
+
+/*
+ * audiots_chinfo()
+ *
+ * Description:
+ *	This is called by the framework to query the channel offsets
+ *	and ordering.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *	int	chan		Channel number.
+ *	unsigned *offset	Starting offset of channel.
+ *	unsigned *incr		Increment (in samples) between frames.
+ *
+ * Returns:
+ *	0 indicating rate array is range instead of enumeration
+ */
+
+static void
+audiots_chinfo(void *arg, int chan, unsigned *offset, unsigned *incr)
+{
+	audiots_port_t *port = arg;
+
+	/* if channels are swapped (SPARCLE), then deal with it */
+	if (port->tp_state->ts_swapped) {
+		*offset = (chan ? 0 : 1);
+	} else {
+		*offset = chan;
+	}
+	*incr = 2;
+}
+
+/*
+ * audiots_format()
+ *
+ * Description:
+ *	Called by the framework to query the format for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	AUDIO_FORMAT_S16_LE.
+ */
+static int
+audiots_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+
+/*
+ * audiots_channels()
+ *
+ * Description:
+ *	Called by the framework to query the channnels for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	2 (Stereo).
+ */
+static int
+audiots_channels(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (2);
+}
+
+/*
+ * audiots_rate()
+ *
+ * Description:
+ *	Called by the framework to query the sample rates for the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	Sample rate in HZ (always 48000).
+ */
+static int
+audiots_rate(void *arg)
+{
+	audiots_port_t *port = arg;
+
+	return (port->tp_rate);
+}
+
+/*
+ * audiots_count()
+ *
+ * Description:
+ *	This is called by the framework to get the engine's frame counter
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	frame count for current engine
+ */
+static uint64_t
+audiots_count(void *arg)
+{
+	audiots_port_t	*port = arg;
+	audiots_state_t	*state = port->tp_state;
+	uint64_t	val;
+
+	mutex_enter(&state->ts_lock);
+	audiots_update_port(port);
+
+	val = port->tp_count;
+	mutex_exit(&state->ts_lock);
+	return (val);
+}
+
+/*
+ * audiots_sync()
+ *
+ * Description:
+ *	This is called by the framework to synchronize DMA caches.
+ *	We also leverage this do some endian swapping, because on SPARC
+ *	the chip accesses the DMA region using 32-bit little-endian
+ *	accesses.  Its not enough to just use the framework's sample
+ *	conversion logic, because the channels will also be backwards.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to sync
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_sync(void *arg, unsigned nframes)
+{
+	audiots_port_t *port = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(port->tp_dmah, 0, 0, port->tp_sync_dir);
+}
+
+/*
+ * audiots_qlen()
+ *
+ * Description:
+ *	This is called by the framework to determine on-device queue length.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	hardware queue length not reported by count (0 for this device)
+ */
+static size_t
+audiots_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (0);
+}
+
+/*
+ * audiots_start_port()
+ *
+ * Description:
+ *	The audio core uses a single DMA buffer which is divided into two
+ *	halves. An interrupt is generated when the middle of the buffer has
+ *	been reached and at the end. The audio core resets the pointer back
+ *	to the beginning automatically. After the interrupt the driver clears
+ *	the buffer and asks the mixer for more audio samples. If there aren't
+ *	enough then silence is played out.
+ *
+ * Arguments:
+ *	audiots_port_t	*port		The DMA engine to start up
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_start_port(audiots_port_t *port)
+{
+	audiots_state_t		*state = port->tp_state;
+	audiots_regs_t		*regs = state->ts_regs;
+	ddi_acc_handle_t	handle = state->ts_acch;
+
+	ASSERT(mutex_owned(&state->ts_lock));
+
+	/* if suspended then do nothing else */
+	if (state->ts_suspended)  {
+		return;
+	}
+
+	/* make sure it starts playing */
+	ddi_put32(handle, &regs->aud_regs.ap_start,
+	    port->tp_dma_mask | port->tp_int_mask);
+
+	ASSERT(mutex_owned(&state->ts_lock));
+}
+
+/*
+ * audiots_stop_port()
+ *
+ * Description:
+ *	This routine stops a DMA engine.
+ *
+ * Arguments:
+ *	audiots_port_t	*port		The port to stop
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_stop_port(audiots_port_t *port)
+{
+	audiots_state_t *state = port->tp_state;
+
+	ASSERT(mutex_owned(&state->ts_lock));
+
+	if (state->ts_suspended)
+		return;
+
+	ddi_put32(state->ts_acch, &state->ts_regs->aud_regs.ap_stop,
+	    port->tp_int_mask | port->tp_dma_mask);
+
+	ASSERT(mutex_owned(&state->ts_lock));
+}
+
+/*
+ * audiots_update_port()
+ *
+ * Description:
+ *	This routine updates the ports frame counter from hardware, and
+ *	gracefully handles wraps.
+ *
+ * Arguments:
+ *	audiots_port_t	*port		The port to stop
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_update_port(audiots_port_t *port)
+{
+	audiots_state_t		*state = port->tp_state;
+
+	uint16_t		cso;
+	unsigned		n;
+
+	ASSERT(mutex_owned(&state->ts_lock));
+
+	if (state->ts_suspended)
+		return;
+
+	cso = ddi_get16(state->ts_acch,
+	    &state->ts_regs->aud_ram[port->tp_dma_stream].aram.aram_cso);
+
+	n = (cso >= port->tp_cso) ?
+	    cso - port->tp_cso :
+	    cso + port->tp_nframes - port->tp_cso;
+
+	port->tp_cso = cso;
+	port->tp_count += n;
+}
+
+/*
+ * audiots_stop_everything()
+ *
+ * Description:
+ *	This routine disables the address engine interrupt for all 32 DMA
+ *	engines. Just to be sure, it then explicitly issues a stop command to
+ *	the address engine and envelope engines for all 32 channels.
+ *
+ * NOTE:
+ *
+ * 	There is a hardware bug that generates a spurious interrupt
+ *	when the DMA engines are stopped. It's not consistent - it
+ *	happens every 1 out of 6 stops or so. It will show up as a
+ *	record interrupt. The problem is that once the driver is
+ *	detached or if the system goes into low power mode, nobody
+ *	will service that interrupt. The system will eventually become
+ *	unusable.
+ *
+ * Arguments:
+ *	audiots_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+audiots_stop_everything(audiots_state_t *state)
+{
+	if (state->ts_acch == NULL)
+		return;
+
+	ddi_put32(state->ts_acch, &state->ts_regs->aud_regs.ap_ainten,
+	    TS_ALL_DMA_OFF);
+
+	ddi_put32(state->ts_acch, &state->ts_regs->aud_regs.ap_stop,
+	    TS_ALL_DMA_ENGINES);
+
+	ddi_put32(state->ts_acch, &state->ts_regs->aud_regs.ap_aint,
+	    TS_ALL_DMA_ENGINES);
+}
+
+/*
+ * audiots_free_port()
+ *
+ * Description:
+ *	This routine unbinds the DMA cookies, frees the DMA buffers,
+ *	deallocates the DMA handles.
+ *
+ * Arguments:
+ *	audiots_port_t	*port	The port structure for a device stream.
+ *
+ * Returns:
+ *	None
+ */
+void
+audiots_free_port(audiots_port_t *port)
+{
+	if (port == NULL)
+		return;
+
+	if (port->tp_engine) {
+		audio_dev_remove_engine(port->tp_state->ts_adev,
+		    port->tp_engine);
+		audio_engine_free(port->tp_engine);
+	}
+	if (port->tp_paddr) {
+		(void) ddi_dma_unbind_handle(port->tp_dmah);
+	}
+	if (port->tp_acch) {
+		ddi_dma_mem_free(&port->tp_acch);
+	}
+	if (port->tp_dmah) {
+		ddi_dma_free_handle(&port->tp_dmah);
+	}
+	kmem_free(port, sizeof (*port));
+}
+
+/*
+ * audiots_destroy()
+ *
+ * Description:
+ *	This routine releases all resources held by the device instance,
+ *	as part of either detach or a failure in attach.
+ *
+ * Arguments:
+ *	audiots_state_t	*state	The device soft state.
+ *
+ * Returns:
+ *	None
+ */
+void
+audiots_destroy(audiots_state_t *state)
+{
+	audiots_stop_everything(state);
+
+	if (state->ts_flags & TS_INTR_INSTALLED)
+		ddi_remove_intr(state->ts_dip, 0, NULL);
+
+	if (state->ts_ksp)
+		kstat_delete(state->ts_ksp);
+
+	for (int i = 0; i < TS_NUM_PORTS; i++)
+		audiots_free_port(state->ts_ports[i]);
+
+	if (state->ts_acch)
+		ddi_regs_map_free(&state->ts_acch);
+
+	if (state->ts_pcih)
+		pci_config_teardown(&state->ts_pcih);
+
+	if (state->ts_ac97)
+		ac97_free(state->ts_ac97);
+
+	if (state->ts_adev)
+		audio_dev_free(state->ts_adev);
+
+	if (state->ts_flags & TS_MUTEX_INIT) {
+		mutex_destroy(&state->ts_lock);
+	}
+
+	ddi_soft_state_free(audiots_statep, ddi_get_instance(state->ts_dip));
+}
diff --git a/usr/src/uts/common/io/audio/sada/drv/audiots/audiots.conf b/usr/src/uts/common/io/audio/drv/audiots/audiots.conf
index 09bb31c4f3..598c38c531 100644
--- a/usr/src/uts/common/io/audio/sada/drv/audiots/audiots.conf
+++ b/usr/src/uts/common/io/audio/drv/audiots/audiots.conf
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,25 +19,14 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2000-2002 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Configuration file for the audiots audio driver.
 #
 #	WARNING: This is an UNSTABLE configuration file. Its contents
 #		may change at any time.
 #
-# mixer-mode determines the mode the mixer audio personality module
-# sets for this driver when it is loaded. The mixerctl(1) command may
-# be used to change modes on the fly.
-#
-#	mixer-mode=1;	<-- mixer enabled
-#	mixer-mode=0;	<-- mixer disabled, old audio(7I) behavior
-#
-
-mixer-mode=1;
 
 #
 # play-interrupts sets the number of interrupts per second when playing.
@@ -55,12 +43,10 @@ mixer-mode=1;
 # boundaries. If those boundaries are violated the driver will not be
 # loaded and an error message is entered into the messages log
 #
-play-interrupts=175;
-record-interrupts=175;
+#play-interrupts=175;
+#record-interrupts=175;
 
 #
-# Uncomment reset-configuration to cause the audiots driver's state to
-# be reset to the default when the driver is loaded. Otherwise this state
-# is retained across driver unload/reload cycles, but not across reboots.
+# ac97-amplifier enables the use of any onboard external amplifier.
 #
-#reset-configuration=1;
+#ac97-amplifier=1;
diff --git a/usr/src/uts/common/sys/audio/impl/audiots_impl.h b/usr/src/uts/common/io/audio/drv/audiots/audiots.h
index 1723ad89b7..f239e8f608 100644
--- a/usr/src/uts/common/sys/audio/impl/audiots_impl.h
+++ b/usr/src/uts/common/io/audio/drv/audiots/audiots.h
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,134 +19,83 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2000-2003 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#ifndef	_SYS_AUDIOTS_IMPL_H
-#define	_SYS_AUDIOTS_IMPL_H
+#ifndef	_AUDIOTS_H
+#define	_AUDIOTS_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
+/*
+ * Header file for the audiots device driver.
+ */
 
-#ifdef	__cplusplus
-extern "C" {
-#endif
+/*
+ * Values returned by the AUDIO_GETDEV ioctl()
+ */
+#define	TS_DEV_NAME		"SUNW,audiots"
+#define	TS_DEV_CONFIG		"onboard1"
+#define	TS_DEV_VERSION_A	"a"		/* grover, sngl strming play */
+#define	TS_DEV_VERSION_B	"b"		/* birdsnest, sngl strm play */
 
 /*
- * Implementation specific header file for the audiots device driver.
+ * Driver supported configuration information
  */
+#define	TS_NAME			"audiots"
+#define	TS_MOD_NAME		"audiots audio driver"
 
-#ifdef _KERNEL
+#define	TS_RATE			(48000)
+#define	TS_STEREO		(2)		/* stereo */
+#define	TS_FRAMESZ		(4)		/* 16-bit stereo */
+/*
+ * This is how much buffer space we at minimum need.  The worst
+ * (biggest) case is 48000 kHz, at 4 bytes per frame (16-bit stereo),
+ * with the lowest interrupt frequency.  48000 x 4 bytes per sample /
+ * 24 is 8000 bytes per fragment.  Two such fragments would be 16000.
+ * (At higher data rates, we get better buffering, with more frags.)
+ */
+#define	TS_BUFSZ		(16000)		/* maximum buffer size */
+#define	TS_INTS			(175)		/* default interrupt rate */
+#define	TS_MIN_INTS		(24)		/* minimum interrupt rate */
+#define	TS_MAX_INTS		(2000)		/* maximum interrupt rate */
 
 /*
  * Misc. defines
  */
-#define	TS_CONFIG_REGS		(0)
 #define	TS_IO_MAPPED_REGS	(1)
 #define	TS_MEM_MAPPED_REGS	(2)
 #define	TS_ALL_DMA_ENGINES	(0xffffffff)
 #define	TS_ALL_DMA_OFF		(0x00000000)
 
-#define	TS_IDNUM		(0x6175)
-#define	TS_MINPACKET		(0)
-#define	TS_MAXPACKET		(1*1024)
-#define	TS_HIWATER		(64*1024)
-#define	TS_LOWATER		(32*1024)
-
-#define	TS_BSIZE		(8*1024)
 #define	TS_20MS			(20000)		/* 20,000 microseconds */
 #define	TS_20US			(20)		/* 20 microseconds */
-#define	TS_GAIN_SHIFT3		(3)		/* 5 significan bits */
-#define	TS_GAIN_SHIFT4		(4)		/* 4 significan bits */
-#define	TS_BYTE_SHIFT		(8)
-#define	TS_LAST_AC_REG		(0x23)
-#define	TS_REG_SIZE		(2)
-#define	TS_MOD_SIZE		(16)
-#define	TS_BUF_HALVES		(2)
-#define	TS_1ST_HALF		(0)
-#define	TS_2ND_HALF		(1)
 #define	TS_SRC_SHIFT		(12)
-#define	TS_NOT_SUSPENDED	(0)
-#define	TS_SUSPENDED		(~TS_NOT_SUSPENDED)
-#define	TS_VENDOR_ID1_MASK	(0x0000ffff);
-#define	TS_PORT_MASK		(0x0ff)
 
-#define	TS_MAX_CHANNELS		(200)		/* force max # chs */
+#define	TS_NUM_PORTS		(2)
+#define	TS_OUTPUT_PORT		(0)
+#define	TS_INPUT_PORT		(1)
+
 #define	TS_MAX_HW_CHANNELS	(32)
-#define	TS_MAX_IN_CHANNELS	1
-#define	TS_MAX_OUT_CHANNELS	(TS_MAX_HW_CHANNELS - TS_MAX_IN_CHANNELS)
-#define	TS_INPUT_STREAM		(31)
-#define	TS_INPUT_CHANNEL_MASK	(0x80000000u)
-#define	TS_INPUT_CHANNEL	(TS_INPUT_CHANNEL_MASK)
-#define	TS_OUTPUT_STREAM	(0)
-#define	TS_OUTPUT_CHANNEL	(0x00000001u)
-#define	TS_CH_ARRAY_SZIE	(TS_MAX_HW_CHANNELS * sizeof (int))
-#define	TS_COMPONENT		(0)
-#define	TS_PWR_OFF		(0)
-#define	TS_PWR_ON		(1)
-#define	TS_NO_PWR_MANAGE	(0)
-#define	TS_PWR_MANAGE		(1)
-#define	TS_NORMAL_POWER		(1)
+
 #define	TS_KIOP(X)		((kstat_intr_t *)(X->ts_ksp->ks_data))
 #define	TS_WAIT_CNT		(512)
 #define	TS_LOOP_CNT		(10)
 #define	TS_DELAY_CNT		(25)
-#define	TS_INIT_RESTORE		(0)
-#define	TS_INIT_NO_RESTORE	~TS_INIT_RESTORE
 #define	TS_CODEC_REG(r)		((r) >> 1)
-#define	TS_PORT_UNMUTE		(0xffffffff)
-#define	TS_AC97_ATTEN_HP	(0x1818)
-#define	TS_AC97_ATTEN_LINE	(0x0404)
-#define	TS_AC97_ATTEN_SPKR	(0x0001)
-#define	TS_FIFO_SIZE		(8)
 
 /* ALI Hacks */
 #define	TS_READ_TRIES		(TS_WAIT_CNT/10)
 #define	TS_RESET_TRIES		(16)
 #define	TS_SB_RESET		(0x7fff)
 
-
-/*
- * audiots_config_t	- Config space registers
- */
-struct audiots_config {
-	uint16_t	tsc_vendor_id;			/* 00h - 01h */
-	uint16_t	tsc_device_id;			/* 02h - 03h */
-	uint16_t	tsc_command;			/* 04h - 05h */
-	uint16_t	tsc_status;			/* 06h - 07h */
-	uint32_t	tsc_class_code_rev_id;		/* 08h - 0bh */
-	uint32_t	tsc_bist_cache;			/* 0ch - 0fh */
-	uint32_t	tsc_audio_io_base;		/* 10h - 13h */
-	uint32_t	tsc_audio_mem_base;		/* 14h - 17h */
-	uint32_t	tsc_rsvd1[5];			/* 18h - 2bh */
-	uint32_t	tsc_sub_ids;			/* 2ch - 2fh */
-	uint32_t	tsc_rsvd2;			/* 30h - 33h */
-	uint16_t	tsc_cap_ptr;			/* 34h - 35h */
-	uint16_t	tsc_rsvd3;			/* 36h - 37h */
-	uint32_t	tsc_rsvd4;			/* 38h - 3bh */
-	uint32_t	tsc_compat1;			/* 3ch - 3fh */
-	uint32_t	tsc_ddma_slave;			/* 40h - 43h */
-	uint32_t	tsc_compat2;			/* 44h - 47h */
-	uint32_t	tsc_compat3;			/* 48h - 4bh */
-	uint32_t	tsc_rsvd5[36];			/* 4ch - dbh */
-	uint16_t	tsc_pm_id_next_ptr;		/* dch - ddh */
-	uint16_t	tsc_pmc;			/* deh - dfh */
-	uint16_t	tsc_pmcsr;			/* e0h - e1h */
-	uint16_t	tsc_pmcsr_bse_base;		/* e2h - e3h */
-	uint32_t	tsc_rsvd6[7];			/* e4h - ffh */
-};
-typedef struct audiots_config audiots_config_t;
+/* These are the devices we know about */
+#define	TS_DEVICE_ALI_M5451			((0x10b9 << 16) | (0x5451))
+#define	TS_DEVICE_TRIDENT_4DWAVEDX		((0x1023 << 16) | (0x2000))
+#define	TS_DEVICE_TRIDENT_4DWAVENX		((0x1023 << 16) | (0x2001))
+#define	TS_DEVICE_SIS_7018			((0x1039 << 16) | (0x7018))
 
 #define	TS_AD_CODEC_ID				0x4144
 #define	TS_AD_REV_ID				0x5340
-#define	TS_CONFIG_VENDOR_ID			0x10b9
-#define	TS_CONFIG_DEVICE_ID			0x5451
-#define	TS_CONFIG_SUBSYSTEM_VENDOR_ID		0x10b9
-#define	TS_CONFIG_SUBSYSTEM_ID			0x5451
-#define	TRI_CONFIG_VENDOR_ID			0x1023
-#define	TRI_CONFIG_DEVICE_ID			0x2000
-#define	TRI_CONFIG_SUBSYSTEM_VENDOR_ID		0x1023
-#define	TRI_CONFIG_SUBSYSTEM_ID			0x2000
 #define	AC_REV_ID1				0x0001
 #define	AC_REV_ID2				0x0002
 
@@ -377,6 +325,7 @@ typedef struct audiots_processor audiots_processor_t;
 #define	AP_CIR_GC_MIDLP_IE			0x00002000
 #define	AP_CIR_GC_ETOG_IE			0x00004000
 #define	AP_CIR_GC_EDROP_IE			0x00008000
+#define	AP_CIR_GC_SYS_MEM_4G_ENABLE		0x00010000
 #define	AP_CIR_GC_PCM_FIFO			0x00000000
 #define	AP_CIR_GC_MMC_BUFFER			0x00100000
 #define	AP_CIR_GC_NORMAL_MODE			0x00000000
@@ -485,90 +434,74 @@ typedef struct audiots_regs audiots_regs_t;
 /*
  * audiots_save_regs_t	- Saved audio controller registers.
  */
-struct audiots_save_regs {
-	uint16_t	aram_delta;
-	uint16_t	eram_ctrl_ec;
+
+struct audiots_port {
+	struct audiots_state	*tp_state;
+	int			tp_num;
+	int			tp_dma_stream;
+	int			tp_int_stream;
+
+	uint32_t		tp_dma_mask;
+	uint32_t		tp_int_mask;
+
+	boolean_t		tp_started;
+	unsigned		tp_intrs;
+	unsigned		tp_fragfr;
+	unsigned		tp_nframes;
+	unsigned		tp_rate;
+	uint64_t		tp_count;
+	size_t			tp_size;
+	caddr_t			tp_kaddr;
+	uint32_t		tp_paddr;
+	ddi_dma_handle_t	tp_dmah;
+	ddi_acc_handle_t	tp_acch;
+	unsigned		tp_sync_dir;
+
+	audio_engine_t		*tp_engine;
+	/* counting cso */
+	uint16_t		tp_cso;
 };
-typedef struct audiots_save_regs audiots_save_regs_t;
+typedef struct audiots_port audiots_port_t;
 
 /*
  * audiots_state_t	- per instance state and operation data
  */
 struct audiots_state {
 	kmutex_t		ts_lock;	/* state protection lock */
-	kcondvar_t		ts_cv;		/* suspend/resume cond. var. */
 	ddi_iblock_cookie_t	ts_iblock;	/* iblock cookie */
 	uint_t			ts_flags;	/* flags */
 	kstat_t			*ts_ksp;	/* kernel statistics */
 	dev_info_t		*ts_dip;	/* used by ts_getinfo() */
-	audiohdl_t		ts_ahandle;	/* audio handle */
-	audio_info_t		ts_defaults;	/* default state for the dev */
-	am_ad_info_t		ts_ad_info;	/* audio device info state */
-	audiots_config_t	*ts_config;	/* configuration registers */
+	audio_dev_t		*ts_adev;	/* audio device handle */
+	ac97_t			*ts_ac97;	/* ac97 common handle */
 	audiots_regs_t		*ts_regs;	/* memory mapped registers */
-	audiots_save_regs_t	ts_save_regs[TS_MAX_HW_CHANNELS];
-						/* saved controller regs */
-	ddi_acc_handle_t	ts_chandle;	/* handle to config regs */
-	ddi_acc_handle_t	ts_handle;	/* handle to mapped regs */
-	audio_device_t		ts_dev_info;	/* device info strings */
-	int			ts_instance;	/* device instance */
-	uint16_t		ts_shadow[64];	/* ac97 shadow registers */
-	boolean_t		ts_pm_core;	/* power manage audio core? */
-	int			ts_suspended;	/* power management state */
-	int			ts_powered;	/* device powered up? */
-	int			ts_busy_cnt;	/* device busy count */
-	ddi_dma_attr_t		*ts_dma_attr;	/* DMA attributes */
-	ddi_dma_handle_t	ts_ph;		/* play DMA handles */
-	ddi_dma_handle_t	ts_ch;		/* capture DMA handles */
-	ddi_dma_cookie_t	ts_pc;		/* play DMA cookies */
-	ddi_dma_cookie_t	ts_cc;		/* capture DMA cookies */
-	ddi_acc_handle_t	ts_pmh;		/* play DMA memory handles */
-	ddi_acc_handle_t	ts_cmh;		/* capture DMA memory handles */
-	size_t			ts_pbuf_size;	/* size of play buffer */
-	size_t			ts_pml;		/* play DMA memory length */
-	size_t			ts_cbuf_size;	/* size of capture buffer */
-	size_t			ts_cml;		/* capture DMA memory length */
-	caddr_t			ts_pb;		/* play DMA buffers */
-	caddr_t			ts_cb;		/* capture DMA buffers */
-	uint16_t		*ts_tcbuf;	/* temp capture buffer */
-	int			ts_pcnt[2];	/* play count, in bytes */
-	int			ts_ccnt;	/* capture count, in bytes */
-	int			ts_psamples[2];	/* play samples */
-	uint_t			ts_output_muted; /* output muted */
-	uint_t			ts_psample_rate; /* play sample rate */
-	uint_t			ts_pchannels;	/* play channels */
-	uint_t			ts_pprecision;	/* play precision */
-	uint_t			ts_csample_rate; /* capture sample rate */
-	uint_t			ts_cchannels;	/* capture channels */
-	uint_t			ts_cprecision;	/* capture precision */
-	uint_t			ts_output_port;	/* current output port */
-	uint_t			ts_input_port;	/* current input port */
-	uint16_t		ts_monitor_gain; /* monitor gain */
-	int			ts_rev_id;	/* SB Chip Revision ID */
+	audiots_port_t		*ts_ports[TS_NUM_PORTS];
+						/* hardware engines */
+
+	ddi_acc_handle_t	ts_pcih;	/* handle to config regs */
+	ddi_acc_handle_t	ts_acch;	/* handle to mapped regs */
+	boolean_t		ts_suspended;	/* power management state */
+	boolean_t		ts_swapped;	/* swapped left/right? */
+
+	uint32_t		ts_devid;
+	uint8_t			ts_revid;	/* SB Chip Revision ID */
+
 };
 typedef struct audiots_state audiots_state_t;
 
 _NOTE(MUTEX_PROTECTS_DATA(audiots_state::ts_lock, audiots_state))
 _NOTE(READ_ONLY_DATA(audiots_state::ts_instance))
 _NOTE(READ_ONLY_DATA(audiots_state::ts_dip))
-_NOTE(READ_ONLY_DATA(audiots_state::ts_ahandle))
+_NOTE(READ_ONLY_DATA(audiots_state::ts_adev))
 
 /* audiots_state.ts_flags defines */
-#define	TS_DMA_ENGINE_INITIALIZED	0x0001u /* play DMA eng. initialized */
-#define	TS_DMA_ENGINE_PAUSED		0x0002u /* play DMA engine paused */
-#define	TS_DMA_ENGINE_EMPTY		0x0004u /* play DMA engine empty */
-#define	TS_DMA_RECORD_START		0x0008u /* record DMA engine started */
-#define	TS_INTR_PENDING			0x0010u /* interrupt pending at DMA */
-						/* stop */
+#define	TS_MUTEX_INIT			0x0001u	/* mutex initialized */
+#define	TS_INTR_INSTALLED		0x0002u	/* intr handler installeld */
 #define	TS_AUDIO_READ_FAILED		0x0020u /* reading the AC97 register */
 						/* has stopped working */
 #define	TS_READ_FAILURE_PRINTED		0x0040u /* Flag to avoid flooding the */
 						/* console with AC97 failure */
 						/* messages */
-#define	TS_ATTACH_PWR			0x0080u	/* Power raised at */
-						/* attach time */
-#define	TS_PM_SUPPORTED			0x0100u	/* PM supported? */
-#define	TS_PLAY_ACTIVE			0x0200u	/* play active, for PM */
 
 /*
  * Read and write the AC-97 Codec's registers
@@ -603,10 +536,4 @@ _NOTE(READ_ONLY_DATA(audiots_state::ts_ahandle))
 	ddi_put32((handle), (uint32_t *)(addr),				\
 		(ddi_get32((handle), (uint32_t *)(addr)) & (uint32_t)(val)));
 
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIOTS_IMPL_H */
+#endif	/* _AUDIOTS_H */
diff --git a/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.c b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.c
new file mode 100644
index 0000000000..f1b05a5a25
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.c
@@ -0,0 +1,1070 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: Driver for the VIA8233/8235 AC97 audio controller
+ */
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#include <sys/types.h>
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/conf.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/pci.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include <sys/audio/ac97.h>
+
+#include "audiovia823x.h"
+
+static struct ddi_device_acc_attr dev_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static struct ddi_device_acc_attr buf_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_NEVERSWAP_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static ddi_dma_attr_t dma_attr_sgd = {
+	DMA_ATTR_V0,		/* version number */
+	0x00000000,		/* low DMA address range */
+	0xffffffff,		/* high DMA address range */
+	0x0000ffff,		/* DMA counter register */
+	8,			/* DMA address alignment */
+	0x3c,			/* DMA burstsizes */
+	8,			/* min effective DMA size */
+	0xffffffff,		/* max DMA xfer size */
+	0x00000fff,		/* segment boundary */
+	1,			/* s/g length */
+	8,			/* granularity of device */
+	0			/* Bus specific DMA flags */
+};
+
+static ddi_dma_attr_t dma_attr_buf = {
+	DMA_ATTR_V0,		/* version number */
+	0x00000000,		/* low DMA address range */
+	0xffffffff,		/* high DMA address range */
+	0x0001fffe,		/* DMA counter register */
+	4,			/* DMA address alignment */
+	0x3c,			/* DMA burstsizes */
+	4,			/* min effective DMA size */
+	0x0001ffff,		/* max DMA xfer size */
+	0x0001ffff,		/* segment boundary */
+	1,			/* s/g length */
+	4,			/* granularity of device */
+	0			/* Bus specific DMA flags */
+};
+
+static int auvia_attach(dev_info_t *);
+static int auvia_resume(dev_info_t *);
+static int auvia_detach(auvia_devc_t *);
+static int auvia_suspend(auvia_devc_t *);
+
+static int auvia_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void auvia_close(void *);
+static int auvia_start(void *);
+static void auvia_stop(void *);
+static int auvia_format(void *);
+static int auvia_channels(void *);
+static int auvia_rate(void *);
+static uint64_t auvia_count(void *);
+static void auvia_sync(void *, unsigned);
+static size_t auvia_qlen(void *);
+
+static uint16_t auvia_read_ac97(void *, uint8_t);
+static void auvia_write_ac97(void *, uint8_t, uint16_t);
+static int auvia_alloc_port(auvia_devc_t *, int);
+static void auvia_start_port(auvia_portc_t *);
+static void auvia_stop_port(auvia_portc_t *);
+static void auvia_update_port(auvia_portc_t *);
+static void auvia_reset_input(auvia_portc_t *);
+static void auvia_reset_output(auvia_portc_t *);
+static void auvia_destroy(auvia_devc_t *);
+static int auvia_setup_intrs(auvia_devc_t *);
+static void auvia_hwinit(auvia_devc_t *);
+static uint_t auvia_intr(caddr_t, caddr_t);
+
+static audio_engine_ops_t auvia_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	auvia_open,
+	auvia_close,
+	auvia_start,
+	auvia_stop,
+	auvia_count,
+	auvia_format,
+	auvia_channels,
+	auvia_rate,
+	auvia_sync,
+	auvia_qlen
+};
+
+static uint16_t
+auvia_read_ac97(void *arg, uint8_t index)
+{
+	auvia_devc_t *devc = arg;
+	uint32_t val = 0;
+	int i;
+
+	mutex_enter(&devc->low_mutex);
+
+	val = ((uint32_t)index << 16) | CODEC_RD;
+	OUTL(devc, devc->base + REG_CODEC, val);
+	drv_usecwait(100);
+
+	/* Check AC CODEC access time out */
+	for (i = 0; i < CODEC_TIMEOUT_COUNT; i++) {
+
+		/* if send command over, break */
+		if (INL(devc, devc->base + REG_CODEC) & CODEC_STA_VALID)
+			break;
+		drv_usecwait(50);
+	}
+
+	if (i == CODEC_TIMEOUT_COUNT) {
+		goto failed;
+	}
+
+	/* Check if Index still ours? If yes, return data, else return FAIL */
+	val = INL(devc, devc->base + REG_CODEC);
+	OUTB(devc, devc->base + REG_CODEC + 3, 0x02);
+	if (((val & CODEC_INDEX) >> 16) == index) {
+		mutex_exit(&devc->low_mutex);
+		return (val & CODEC_DATA);
+	}
+
+failed:
+	mutex_exit(&devc->low_mutex);
+	return (0xffff);
+}
+
+static void
+auvia_write_ac97(void *arg, uint8_t index, uint16_t data)
+{
+	auvia_devc_t *devc = arg;
+	uint32_t val = 0;
+	int i = 0;
+
+	mutex_enter(&devc->low_mutex);
+
+	val = ((uint32_t)index << 16) | data | CODEC_WR;
+	OUTL(devc, devc->base + REG_CODEC, val);
+	drv_usecwait(100);
+
+	/* Check AC CODEC access time out */
+	for (i = 0; i < CODEC_TIMEOUT_COUNT; i++) {
+		/* if send command over, break */
+		if (!(INL(devc, devc->base + REG_CODEC) & CODEC_IN_CMD))
+			break;
+		drv_usecwait(50);
+	}
+
+	mutex_exit(&devc->low_mutex);
+}
+
+static uint_t
+auvia_intr(caddr_t argp, caddr_t nocare)
+{
+	auvia_devc_t	*devc = (void *)argp;
+	auvia_portc_t	*portc;
+	uint32_t	gstat;
+	uint8_t		status;
+	unsigned	intrs = 0;
+
+	_NOTE(ARGUNUSED(nocare));
+
+	mutex_enter(&devc->mutex);
+	if (devc->suspended) {
+		mutex_exit(&devc->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	gstat = INL(devc, devc->base + REG_GSTAT);
+	if (gstat == 0) {
+		mutex_exit(&devc->mutex);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	for (int i = 0; i < AUVIA_NUM_PORTC; i++) {
+
+		portc = devc->portc[i];
+
+		status = INB(devc, portc->base + OFF_STATUS);
+		if ((status & STATUS_INTR) == 0) {
+			/* clear any other interrupts */
+			continue;
+		}
+
+		/*
+		 * NB: The old code did some goofy things to update
+		 * the last valid SGD.  However, since we don't ever
+		 * reach the last valid SGD (because we loop first), I
+		 * don't believe we need to do that.  It would appear
+		 * that NetBSD does the same.
+		 */
+		/* port interrupt */
+		if (portc->started) {
+			intrs |= (1U << i);
+		}
+		/* XXX: do we really need to do this? */
+		OUTB(devc, portc->base + OFF_STATUS, status);
+	}
+
+	OUTL(devc, devc->base + REG_GSTAT, gstat);
+
+	mutex_exit(&devc->mutex);
+	if (intrs & (1U << AUVIA_PLAY_SGD_NUM)) {
+		audio_engine_consume(devc->portc[AUVIA_PLAY_SGD_NUM]->engine);
+	}
+	if (intrs & (1U << AUVIA_REC_SGD_NUM)) {
+		audio_engine_produce(devc->portc[AUVIA_REC_SGD_NUM]->engine);
+	}
+	if (devc->ksp) {
+		AUVIA_KIOP(devc)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	return (DDI_INTR_CLAIMED);
+}
+
+/*
+ * Audio routines
+ */
+
+int
+auvia_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	auvia_portc_t	 *portc = arg;
+	auvia_devc_t	 *devc = portc->devc;
+
+	_NOTE(ARGUNUSED(flag));
+
+	portc->started = B_FALSE;
+	portc->count = 0;
+	*fragfrp = portc->fragfr;
+	*nfragsp = AUVIA_NUM_SGD;
+	*bufp = portc->buf_kaddr;
+
+	mutex_enter(&devc->mutex);
+	portc->reset(portc);
+	mutex_exit(&devc->mutex);
+
+	return (0);
+}
+
+void
+auvia_close(void *arg)
+{
+	auvia_portc_t	 *portc = arg;
+	auvia_devc_t	 *devc = portc->devc;
+
+	mutex_enter(&devc->mutex);
+	auvia_stop_port(portc);
+	portc->started = B_FALSE;
+	mutex_exit(&devc->mutex);
+}
+
+int
+auvia_start(void *arg)
+{
+	auvia_portc_t	*portc = arg;
+	auvia_devc_t	*devc = portc->devc;
+
+	mutex_enter(&devc->mutex);
+	if (!portc->started) {
+		auvia_start_port(portc);
+		portc->started = B_TRUE;
+	}
+	mutex_exit(&devc->mutex);
+	return (0);
+}
+
+void
+auvia_stop(void *arg)
+{
+	auvia_portc_t	*portc = arg;
+	auvia_devc_t	*devc = portc->devc;
+
+	mutex_enter(&devc->mutex);
+	if (portc->started) {
+		auvia_stop_port(portc);
+		portc->started = B_FALSE;
+	}
+	mutex_exit(&devc->mutex);
+}
+
+int
+auvia_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_LE);
+}
+
+int
+auvia_channels(void *arg)
+{
+	auvia_portc_t	*portc = arg;
+
+	return (portc->nchan);
+}
+
+int
+auvia_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
+}
+
+void
+auvia_sync(void *arg, unsigned nframes)
+{
+	auvia_portc_t *portc = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(portc->buf_dmah, 0, 0, portc->syncdir);
+}
+
+size_t
+auvia_qlen(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+	return (0);
+}
+
+uint64_t
+auvia_count(void *arg)
+{
+	auvia_portc_t	*portc = arg;
+	auvia_devc_t	*devc = portc->devc;
+	uint64_t	val;
+
+	mutex_enter(&devc->mutex);
+	auvia_update_port(portc);
+	/*
+	 * The residual is in bytes.  We have to convert to frames,
+	 * and then subtract it from the fragment size to get the
+	 * number of frames processed.  It is somewhat unfortunate thta
+	 * this (the division) has to happen under the lock.  If we
+	 * restricted ourself to stereo out, this would be a simple
+	 * shift.
+	 */
+	val = portc->count +
+	    (portc->fragfr - (portc->resid / (portc->nchan * 2)));
+	mutex_exit(&devc->mutex);
+
+	return (val);
+}
+
+
+/* private implementation bits */
+
+void
+auvia_start_port(auvia_portc_t *portc)
+{
+	auvia_devc_t	*devc = portc->devc;
+
+	ASSERT(mutex_owned(&devc->mutex));
+
+	if (devc->suspended)
+		return;
+
+	/*
+	 * Start with autoinit and SGD flag
+	 * interrupts enabled.
+	 */
+	OUTB(devc, portc->base + OFF_CTRL,
+	    CTRL_START | CTRL_AUTOSTART | CTRL_FLAG);
+}
+
+void
+auvia_stop_port(auvia_portc_t *portc)
+{
+	auvia_devc_t	*devc = portc->devc;
+
+	if (devc->suspended)
+		return;
+
+	OUTB(devc, portc->base + OFF_CTRL, CTRL_TERMINATE);
+}
+
+void
+auvia_update_port(auvia_portc_t *portc)
+{
+	auvia_devc_t	*devc = portc->devc;
+	uint32_t	frag;
+	uint32_t	n;
+
+	ASSERT(mutex_owned(&devc->mutex));
+	if (devc->suspended) {
+		portc->cur_frag = 0;
+		portc->resid = portc->fragsz;
+		n = 0;
+	} else {
+		frag = INL(devc, portc->base + OFF_COUNT);
+		portc->resid = (frag & 0xffffff);
+		frag >>= 24;
+		frag &= 0xff;
+
+		if (frag >= portc->cur_frag) {
+			n = frag - portc->cur_frag;
+		} else {
+			n = frag + AUVIA_NUM_SGD - portc->cur_frag;
+		}
+		portc->count += (n * portc->fragfr);
+		portc->cur_frag = frag;
+	}
+}
+
+void
+auvia_reset_output(auvia_portc_t *portc)
+{
+	auvia_devc_t	*devc = portc->devc;
+	uint32_t	cmap;
+
+	portc->cur_frag = 0;
+	portc->resid = portc->fragsz;
+
+	if (devc->suspended)
+		return;
+
+	OUTB(devc, portc->base + OFF_CTRL, CTRL_TERMINATE);	/* Stop */
+	OUTL(devc, portc->base + OFF_DMA, portc->sgd_paddr);
+
+	OUTB(devc, portc->base + OFF_PLAYFMT,
+	    PLAYFMT_16BIT | (portc->nchan << 4));
+
+	/* Select channel assignment - not valid for 8233A */
+	if (devc->chip_type != CHIP_8233A) {
+		/*
+		 * Undocumented slot mapping table:
+		 *
+		 * slot 3 = 1 (left)
+		 * slot 4 = 2 (right)
+		 * slot 6 = 5 (center)
+		 * slot 9 = 6 (lfe)
+		 * slot 7 = 3 (left rear)
+		 * slot 8 = 4 (right rear)
+		 */
+		switch (portc->nchan) {
+		case 1:
+			cmap = (1 << 0) | (1 << 4);
+			break;
+		case 2:
+			cmap = (1 << 0) | (2 << 4);
+			break;
+		case 4:
+			cmap = (1 << 0) | (2 << 4) | (3 << 8) | (4 << 12);
+			break;
+		case 6:
+			cmap = (1 << 0) | (2 << 4) |
+			    (5 << 8) | (6 << 12) | (3 << 16) | (4 << 20);
+			break;
+		default:
+			cmap = 0;
+			break;
+		}
+		OUTL(devc, portc->base + OFF_CHANNELS, cmap | 0xFF000000U);
+	}
+}
+
+static void
+auvia_reset_input(auvia_portc_t *portc)
+{
+	auvia_devc_t	*devc = portc->devc;
+	uint32_t	fmt;
+
+	portc->cur_frag = 0;
+	portc->resid = portc->fragsz;
+
+	if (devc->suspended)
+		return;
+
+	OUTB(devc, portc->base + OFF_CTRL, CTRL_TERMINATE);	/* Stop */
+	OUTL(devc, portc->base + OFF_DMA, portc->sgd_paddr);
+
+	fmt = RECFMT_STEREO | RECFMT_16BIT;
+
+	if (devc->chip_type != CHIP_8233A) {
+		fmt |= RECFMT_48K;
+	}
+	fmt |= (0xffU << 24);
+	OUTB(devc, portc->base + OFF_RECFIFO, RECFIFO_ENABLE);
+	OUTL(devc, portc->base + OFF_RECFMT, fmt);
+}
+
+int
+auvia_alloc_port(auvia_devc_t *devc, int num)
+{
+	auvia_portc_t		*portc;
+	size_t			len;
+	ddi_dma_cookie_t	cookie;
+	uint_t			count;
+	int			dir;
+	char			*prop;
+	unsigned		caps;
+	audio_dev_t		*adev;
+	uint32_t		*desc;
+	uint32_t		paddr;
+
+	adev = devc->adev;
+	portc = kmem_zalloc(sizeof (*portc), KM_SLEEP);
+	devc->portc[num] = portc;
+	portc->devc = devc;
+	portc->started = B_FALSE;
+
+	switch (num) {
+	case AUVIA_REC_SGD_NUM:
+		prop = "record-interrupts";
+		portc->base = devc->base + REG_RECBASE;
+		portc->syncdir = DDI_DMA_SYNC_FORKERNEL;
+		portc->nchan = 2;
+		portc->reset = auvia_reset_input;
+		caps = ENGINE_INPUT_CAP;
+		dir = DDI_DMA_READ;
+		break;
+	case AUVIA_PLAY_SGD_NUM:
+		prop = "play-interrupts";
+		portc->base = devc->base + REG_PLAYBASE;
+		portc->syncdir = DDI_DMA_SYNC_FORDEV;
+		portc->nchan = 6;
+		portc->reset = auvia_reset_output;
+		caps = ENGINE_OUTPUT_CAP;
+		dir = DDI_DMA_WRITE;
+		break;
+	default:
+		return (DDI_FAILURE);
+	}
+
+	/* make sure port is shut down */
+	OUTB(portc->devc, portc->base + OFF_CTRL, CTRL_TERMINATE);
+
+	/* figure out fragment configuration */
+	portc->intrs = ddi_prop_get_int(DDI_DEV_T_ANY, devc->dip,
+	    DDI_PROP_DONTPASS, prop, AUVIA_INTRS);
+
+	/* make sure the values are good */
+	if (portc->intrs < AUVIA_MIN_INTRS) {
+		audio_dev_warn(adev, "%s too low, %d, reset to %d",
+		    prop, portc->intrs, AUVIA_INTRS);
+		portc->intrs = AUVIA_INTRS;
+	} else if (portc->intrs > AUVIA_MAX_INTRS) {
+		audio_dev_warn(adev, "%s too high, %d, reset to %d",
+		    prop, portc->intrs, AUVIA_INTRS);
+		portc->intrs = AUVIA_INTRS;
+	}
+
+	portc->fragfr = 48000 / portc->intrs;
+	portc->fragsz = portc->fragfr * portc->nchan * 2;
+	portc->buf_size = portc->fragsz * AUVIA_NUM_SGD;
+
+	/* first allocate up space for SGD list */
+	if (ddi_dma_alloc_handle(devc->dip, &dma_attr_sgd,
+	    DDI_DMA_SLEEP, NULL, &portc->sgd_dmah) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to allocate SGD handle");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_dma_mem_alloc(portc->sgd_dmah,
+	    AUVIA_NUM_SGD * 2 * sizeof (uint32_t), &dev_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &portc->sgd_kaddr,
+	    &len, &portc->sgd_acch) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to allocate SGD memory");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_dma_addr_bind_handle(portc->sgd_dmah, NULL,
+	    portc->sgd_kaddr, len, DDI_DMA_CONSISTENT | DDI_DMA_WRITE,
+	    DDI_DMA_SLEEP, NULL, &cookie, &count) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed binding SGD DMA handle");
+		return (DDI_FAILURE);
+	}
+	portc->sgd_paddr = cookie.dmac_address;
+
+	/* now buffers */
+	if (ddi_dma_alloc_handle(devc->dip, &dma_attr_buf, DDI_DMA_SLEEP, NULL,
+	    &portc->buf_dmah) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to allocate BUF handle");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_dma_mem_alloc(portc->buf_dmah, portc->buf_size,
+	    &buf_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
+	    &portc->buf_kaddr, &len, &portc->buf_acch) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed to allocate BUF memory");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_dma_addr_bind_handle(portc->buf_dmah, NULL, portc->buf_kaddr,
+	    len, DDI_DMA_CONSISTENT | dir, DDI_DMA_SLEEP, NULL, &cookie,
+	    &count) != DDI_SUCCESS) {
+		audio_dev_warn(adev, "failed binding BUF DMA handle");
+		return (DDI_FAILURE);
+	}
+	portc->buf_paddr = cookie.dmac_address;
+
+	/* now wire descriptors up */
+	desc = (void *)portc->sgd_kaddr;
+	paddr = portc->buf_paddr;
+	for (int i = 0; i < AUVIA_NUM_SGD; i++) {
+		uint32_t	flags;
+
+		flags = AUVIA_SGD_FLAG | portc->fragsz;
+
+		if (i == (AUVIA_NUM_SGD - 1)) {
+			flags |= AUVIA_SGD_EOL;
+		}
+		ddi_put32(portc->sgd_acch, desc++, paddr);
+		ddi_put32(portc->sgd_acch, desc++, flags);
+		paddr += portc->fragsz;
+	}
+
+	ddi_dma_sync(portc->sgd_dmah, 0, 0, DDI_DMA_SYNC_FORDEV);
+
+	portc->engine = audio_engine_alloc(&auvia_engine_ops, caps);
+	if (portc->engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(portc->engine, portc);
+	audio_dev_add_engine(adev, portc->engine);
+
+	return (DDI_SUCCESS);
+}
+
+int
+auvia_setup_intrs(auvia_devc_t *devc)
+{
+	uint_t			ipri;
+	int			actual;
+	int			rv;
+	ddi_intr_handle_t	ih[1];
+
+	rv = ddi_intr_alloc(devc->dip, ih, DDI_INTR_TYPE_FIXED,
+	    0, 1, &actual, DDI_INTR_ALLOC_STRICT);
+	if ((rv != DDI_SUCCESS) || (actual != 1)) {
+		audio_dev_warn(devc->adev,
+		    "Can't alloc interrupt handle (rv %d actual %d)",
+		    rv, actual);
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_get_pri(ih[0], &ipri) != DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "Can't get interrupt priority");
+		(void) ddi_intr_free(ih[0]);
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_intr_add_handler(ih[0], auvia_intr, devc, NULL) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "Can't add interrupt handler");
+		(void) ddi_intr_free(ih[0]);
+		return (DDI_FAILURE);
+	}
+
+	devc->ih = ih[0];
+	mutex_init(&devc->mutex, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+	mutex_init(&devc->low_mutex, NULL, MUTEX_DRIVER, DDI_INTR_PRI(ipri));
+	return (DDI_SUCCESS);
+}
+
+void
+auvia_destroy(auvia_devc_t *devc)
+{
+	if (devc->ih != NULL) {
+		(void) ddi_intr_disable(devc->ih);
+		(void) ddi_intr_remove_handler(devc->ih);
+		(void) ddi_intr_free(devc->ih);
+		mutex_destroy(&devc->mutex);
+		mutex_destroy(&devc->low_mutex);
+	}
+
+	if (devc->ksp) {
+		kstat_delete(devc->ksp);
+	}
+
+	for (int i = 0; i < AUVIA_NUM_PORTC; i++) {
+		auvia_portc_t *portc = devc->portc[i];
+		if (!portc)
+			continue;
+		if (portc->engine) {
+			audio_dev_remove_engine(devc->adev, portc->engine);
+			audio_engine_free(portc->engine);
+		}
+		if (portc->sgd_paddr) {
+			(void) ddi_dma_unbind_handle(portc->sgd_dmah);
+		}
+		if (portc->sgd_acch) {
+			ddi_dma_mem_free(&portc->sgd_acch);
+		}
+		if (portc->sgd_dmah) {
+			ddi_dma_free_handle(&portc->sgd_dmah);
+		}
+		if (portc->buf_paddr) {
+			(void) ddi_dma_unbind_handle(portc->buf_dmah);
+		}
+		if (portc->buf_acch) {
+			ddi_dma_mem_free(&portc->buf_acch);
+		}
+		if (portc->buf_dmah) {
+			ddi_dma_free_handle(&portc->buf_dmah);
+		}
+		kmem_free(portc, sizeof (*portc));
+	}
+
+	if (devc->ac97 != NULL) {
+		ac97_free(devc->ac97);
+	}
+	if (devc->adev != NULL) {
+		audio_dev_free(devc->adev);
+	}
+	if (devc->regsh != NULL) {
+		ddi_regs_map_free(&devc->regsh);
+	}
+	if (devc->pcih != NULL) {
+		pci_config_teardown(&devc->pcih);
+	}
+	kmem_free(devc, sizeof (*devc));
+}
+
+void
+auvia_hwinit(auvia_devc_t *devc)
+{
+	ddi_acc_handle_t	pcih = devc->pcih;
+	uint32_t		val;
+
+	val = pci_config_get32(pcih, AUVIA_PCICFG);
+	/* we want to disable all legacy */
+	val &= ~AUVIA_PCICFG_LEGACY;
+	val &= ~(AUVIA_PCICFG_FMEN | AUVIA_PCICFG_SBEN);
+
+	/* enable AC'97 link and clear the reset bit */
+	val |= (AUVIA_PCICFG_ACLINKEN | AUVIA_PCICFG_NRST);
+	/* disable SRC (we won't use it) */
+	val &= ~AUVIA_PCICFG_SRCEN;
+	/* enable the SGD engines */
+	val |= AUVIA_PCICFG_SGDEN;
+
+	pci_config_put32(pcih, AUVIA_PCICFG, val);
+
+	drv_usecwait(10);
+}
+
+int
+auvia_attach(dev_info_t *dip)
+{
+	uint8_t 	pci_revision;
+	uint16_t	pci_command, vendor, device;
+	auvia_devc_t	*devc;
+	ddi_acc_handle_t pcih;
+	const char	*version;
+
+	devc = kmem_zalloc(sizeof (*devc), KM_SLEEP);
+	devc->dip = dip;
+	ddi_set_driver_private(dip, devc);
+
+	if ((devc->adev = audio_dev_alloc(dip, 0)) == NULL) {
+		cmn_err(CE_WARN, "audio_dev_alloc failed");
+		goto error;
+	}
+
+	if (pci_config_setup(dip, &pcih) != DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "pci_config_setup failed");
+		goto error;
+	}
+	devc->pcih = pcih;
+
+	vendor = pci_config_get16(pcih, PCI_CONF_VENID);
+	device = pci_config_get16(pcih, PCI_CONF_DEVID);
+	if ((vendor != VIA_VENDOR_ID) || (device != VIA_8233_ID &&
+	    device != VIA_8233A_ID)) {
+		audio_dev_warn(devc->adev, "Hardware not recognized "
+		    "(vendor=%x, dev=%x)", vendor, device);
+		goto error;
+	}
+
+	devc->chip_type = CHIP_8233;
+	devc->chip_name = "VIA VT8233";
+	version = "8233";
+
+	pci_revision = pci_config_get8(pcih, PCI_CONF_REVID);
+
+	if (pci_revision == 0x50) {
+		devc->chip_name = "VIA VT8235";
+		version = "8235";
+	}
+
+	if (pci_revision == 0x60) {
+		devc->chip_name = "VIA VT8237";
+		version = "8237";
+	}
+
+	if ((device == VIA_8233A_ID) ||
+	    (device == VIA_8233_ID && pci_revision == 0x40)) {
+		devc->chip_type = CHIP_8233A;
+		devc->chip_name = "VIA VT8233A";
+		version = "8233A";
+	}
+	audio_dev_set_description(devc->adev, devc->chip_name);
+	audio_dev_set_version(devc->adev, version);
+
+	pci_command = pci_config_get16(pcih, PCI_CONF_COMM);
+	pci_command |= PCI_COMM_ME | PCI_COMM_IO | PCI_COMM_MAE;
+	pci_config_put16(pcih, PCI_CONF_COMM, pci_command);
+
+	if ((ddi_regs_map_setup(dip, 1, &devc->base, 0, 0, &dev_attr,
+	    &devc->regsh)) != DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "failed to map registers");
+		goto error;
+	}
+
+	auvia_hwinit(devc);
+
+	if ((auvia_alloc_port(devc, AUVIA_PLAY_SGD_NUM) != DDI_SUCCESS) ||
+	    (auvia_alloc_port(devc, AUVIA_REC_SGD_NUM) != DDI_SUCCESS)) {
+		goto error;
+	}
+
+	if (auvia_setup_intrs(devc) != DDI_SUCCESS) {
+		goto error;
+	}
+
+	devc->ac97 = ac97_alloc(dip, auvia_read_ac97, auvia_write_ac97, devc);
+	if (devc->ac97 == NULL) {
+		audio_dev_warn(devc->adev, "failed to allocate ac97 handle");
+		goto error;
+	}
+
+	if (ac97_init(devc->ac97, devc->adev) != DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "failed to init ac97");
+		goto error;
+	}
+
+	/* set up kernel statistics */
+	if ((devc->ksp = kstat_create(AUVIA_NAME, ddi_get_instance(dip),
+	    AUVIA_NAME, "controller", KSTAT_TYPE_INTR, 1,
+	    KSTAT_FLAG_PERSISTENT)) != NULL) {
+		kstat_install(devc->ksp);
+	}
+
+	if (audio_dev_register(devc->adev) != DDI_SUCCESS) {
+		audio_dev_warn(devc->adev, "unable to register with framework");
+		goto error;
+	}
+
+	(void) ddi_intr_enable(devc->ih);
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	auvia_destroy(devc);
+	return (DDI_FAILURE);
+}
+
+int
+auvia_resume(dev_info_t *dip)
+{
+	auvia_devc_t *devc;
+
+	devc = ddi_get_driver_private(dip);
+
+	auvia_hwinit(devc);
+
+	/* allow ac97 operations again */
+	ac97_resume(devc->ac97);
+
+	mutex_enter(&devc->mutex);
+	devc->suspended = B_TRUE;
+	for (int i = 0; i < AUVIA_NUM_PORTC; i++) {
+
+		auvia_portc_t *portc = devc->portc[i];
+
+		if (portc->engine != NULL)
+			audio_engine_reset(portc->engine);
+
+		/* reset the port */
+		portc->reset(portc);
+
+		if (portc->started) {
+			auvia_start_port(portc);
+		} else {
+			auvia_stop_port(portc);
+		}
+	}
+	mutex_exit(&devc->mutex);
+	return (DDI_SUCCESS);
+}
+
+
+int
+auvia_detach(auvia_devc_t *devc)
+{
+	if (audio_dev_unregister(devc->adev) != DDI_SUCCESS)
+		return (DDI_FAILURE);
+
+	auvia_destroy(devc);
+	return (DDI_SUCCESS);
+}
+
+int
+auvia_suspend(auvia_devc_t *devc)
+{
+	ac97_suspend(devc->ac97);
+
+	mutex_enter(&devc->mutex);
+	for (int i = 0; i < AUVIA_NUM_PORTC; i++) {
+
+		auvia_portc_t *portc = devc->portc[i];
+		auvia_stop_port(portc);
+	}
+	devc->suspended = B_TRUE;
+	mutex_exit(&devc->mutex);
+	return (DDI_SUCCESS);
+}
+
+static int auvia_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int auvia_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+static int auvia_ddi_quiesce(dev_info_t *);
+
+static struct dev_ops auvia_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	NULL,			/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	auvia_ddi_attach,	/* attach */
+	auvia_ddi_detach,	/* detach */
+	nodev,			/* reset */
+	NULL,			/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+	auvia_ddi_quiesce,	/* quiesce */
+};
+
+static struct modldrv auvia_modldrv = {
+	&mod_driverops,			/* drv_modops */
+	"Via 823x Audio",		/* linkinfo */
+	&auvia_dev_ops,			/* dev_ops */
+};
+
+static struct modlinkage modlinkage = {
+	MODREV_1,
+	{ &auvia_modldrv, NULL }
+};
+
+int
+_init(void)
+{
+	int	rv;
+
+	audio_init_ops(&auvia_dev_ops, AUVIA_NAME);
+	if ((rv = mod_install(&modlinkage)) != 0) {
+		audio_fini_ops(&auvia_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_fini(void)
+{
+	int	rv;
+
+	if ((rv = mod_remove(&modlinkage)) == 0) {
+		audio_fini_ops(&auvia_dev_ops);
+	}
+	return (rv);
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&modlinkage, modinfop));
+}
+
+int
+auvia_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (auvia_attach(dip));
+
+	case DDI_RESUME:
+		return (auvia_resume(dip));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+int
+auvia_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	auvia_devc_t *devc;
+
+	devc = ddi_get_driver_private(dip);
+
+	switch (cmd) {
+	case DDI_DETACH:
+		return (auvia_detach(devc));
+
+	case DDI_SUSPEND:
+		return (auvia_suspend(devc));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+int
+auvia_ddi_quiesce(dev_info_t *dip)
+{
+	auvia_devc_t	*devc;
+
+	devc = ddi_get_driver_private(dip);
+
+	for (int i = 0; i < AUVIA_NUM_PORTC; i++) {
+
+		auvia_portc_t *portc = devc->portc[i];
+		auvia_stop_port(portc);
+	}
+	return (DDI_SUCCESS);
+}
diff --git a/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.conf b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.conf
new file mode 100644
index 0000000000..4b98468d88
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.conf
@@ -0,0 +1,41 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Configuration file for the audiovia823x audio driver.
+#
+#	WARNING: This is an UNSTABLE configuration file. Its contents
+#		may change at any time.
+
+#
+# play-interrupts sets the number of interrupts per second when playing.
+# This affects the resolution of various things, such as sample counts.
+# record-interrupts does the same for record interrupts.
+#
+# These may be tuned to get more accurate information by increasing the
+# count. However, the larger the interrupts per second the larger the
+# load on the system. So use this capability cautiously. The audiovia823x
+# driver enforces a maximum and minimum count.
+#
+play-interrupts=175;
+record-interrupts=175;
diff --git a/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.h b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.h
new file mode 100644
index 0000000000..f2eb52ce50
--- /dev/null
+++ b/usr/src/uts/common/io/audio/drv/audiovia823x/audiovia823x.h
@@ -0,0 +1,191 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: Definitions for the via8233 driver
+ */
+/*
+ * This file is part of Open Sound System
+ *
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * This software is released under CDDL 1.0 source license.
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+#ifndef	AUDIOVIA823X_H
+#define	AUDIOVIA823X_H
+
+#define	AUVIA_NAME		"audiovia823x"
+
+#define	VIA_VENDOR_ID		0x1106
+#define	VIA_8233_ID		0x3059
+#define	VIA_8233A_ID		0x7059
+
+/* pci configuration registers */
+#define	AUVIA_PCICFG		0x40		/* Via chip specific cfg reg */
+#define	AUVIA_PCICFG_LEGACY	0x00ff0000	/* legacy enables mask */
+#define	AUVIA_PCICFG_ACLINKEN	0x00008000	/* AC'97 link enable */
+#define	AUVIA_PCICFG_NRST	0x00004000	/* inverse of AC'97 reset */
+#define	AUVIA_PCICFG_ACSYNC	0x00002000	/* AC'97 sync */
+#define	AUVIA_PCICFG_SRCEN	0x00000800	/* sample rate converter en */
+#define	AUVIA_PCICFG_SGDEN	0x00000400	/* SGD enable */
+#define	AUVIA_PCICFG_FMEN	0x00000200 	/* FM synth enable (legacy) */
+#define	AUVIA_PCICFG_SBEN	0x00000100	/* SB compat enable (legacy) */
+#define	AUVIA_PCICFG_PRIVALID	0x00000001	/* primary codec ready */
+
+#define	AUVIA_PLAY_SGD_NUM	1
+#define	AUVIA_REC_SGD_NUM	0
+#define	AUVIA_NUM_PORTC		2
+#define	AUVIA_NUM_SGD		16	/* number of fragments */
+
+#define	AUVIA_MAX_INTRS		256
+#define	AUVIA_MIN_INTRS		24
+#define	AUVIA_INTRS		175
+
+#define	AUVIA_SGD_EOL		0x80000000
+#define	AUVIA_SGD_FLAG		0x40000000
+
+#define	CODEC_TIMEOUT_COUNT		500
+
+#define	REG_PLAYBASE		0x40	/* Multichannel SGD */
+#define	REG_RECBASE		0x60
+#define	REG_CODEC		0x80	/* Access AC97 Codec */
+#define	REG_GSTAT		0x84	/* Global status */
+
+/* REG_CODEC */
+#define	CODEC_IN_CMD		0x01000000	/* busy in sending */
+#define	CODEC_STA_VALID		0x02000000	/* 1:status data is valid */
+#define	CODEC_RD		0x00800000	/* Read CODEC status */
+#define	CODEC_WR		0x00000000	/* Write CODEC status */
+#define	CODEC_INDEX		0x007F0000	/* Index of command register */
+#define	CODEC_DATA		0x0000FFFF	/* AC97 status register data */
+
+/* registers that are offsets relative to a port */
+#define	OFF_STATUS		0x00
+#define	OFF_CTRL		0x01
+#define	OFF_PLAYFMT		0x02
+#define	OFF_RECFIFO		0x02
+#define	OFF_DMA			0x04
+#define	OFF_CHANNELS		0x08
+#define	OFF_RECFMT		0x08
+#define	OFF_COUNT		0x0C
+
+/* bits for above offsets */
+#define	STATUS_INTR		0x3
+
+#define	CTRL_START		0x80
+#define	CTRL_TERMINATE		0x40
+#define	CTRL_AUTOSTART		0x20
+#define	CTRL_MULTICHORDER	0x10	/* SGD 0x40 only, Center/LFE order */
+#define	CTRL_FLAG		0x01
+
+#define	PLAYFMT_16BIT		0x80
+#define	PLAYFMT_STEREO		0x20	/* Num channels (1-6), upper nybble */
+#define	PLAYFMT_6CH		0x60
+#define	PLAYFMT_4CH		0x40
+
+#define	RECFIFO_ENABLE		0x40
+
+#define	RECFMT_48K		0x00ffffff
+#define	RECFMT_STEREO		0x00100000
+#define	RECFMT_16BIT		0x00200000
+
+
+typedef struct {
+	unsigned int phaddr;
+	unsigned int flags;
+} SGD_entry;
+
+typedef struct auvia_portc auvia_portc_t;
+typedef struct auvia_devc auvia_devc_t;
+
+struct auvia_portc {
+	auvia_devc_t		*devc;
+	audio_engine_t		*engine;
+	caddr_t			base;		/* base for registers */
+	boolean_t		started;
+	int			nchan;
+
+	ddi_dma_handle_t	sgd_dmah;	/* dma for descriptors */
+	ddi_acc_handle_t	sgd_acch;
+	uint32_t		sgd_paddr;
+	caddr_t			sgd_kaddr;
+
+	ddi_dma_handle_t	buf_dmah;	/* dma for buffers */
+	ddi_acc_handle_t	buf_acch;
+	uint32_t		buf_paddr;
+	caddr_t			buf_kaddr;
+	size_t			buf_size;
+	int			syncdir;
+
+	unsigned		intrs;
+	unsigned		fragfr;
+	unsigned		fragsz;
+	unsigned		cur_frag;
+	unsigned		resid;
+
+	uint64_t		count;
+
+	/* helper functions */
+	void			(*reset)(auvia_portc_t *);
+};
+
+
+struct auvia_devc {
+	dev_info_t		*dip;
+	audio_dev_t		*adev;
+	ac97_t			*ac97;
+	kstat_t			*ksp;
+	boolean_t		suspended;
+
+	char			*chip_name;
+	int			chip_type;
+#define	CHIP_8233		0
+#define	CHIP_8233A		1
+
+	/* registers */
+	ddi_acc_handle_t	pcih;
+	ddi_acc_handle_t	regsh;
+	caddr_t			base;
+
+	kmutex_t		mutex;		/* For normal locking */
+	kmutex_t		low_mutex;	/* For low level routines */
+	ddi_intr_handle_t	ih;
+	auvia_portc_t		*portc[AUVIA_NUM_PORTC];
+};
+
+#define	AUVIA_KIOP(X)	((kstat_intr_t *)(X->ksp->ks_data))
+
+#define	INL(devc, reg)		ddi_get32(devc->regsh, (void *)(reg))
+
+#define	INB(devc, reg)		ddi_get8(devc->regsh, (void *)(reg))
+
+#define	OUTL(devc, reg, val)	ddi_put32(devc->regsh, (void *)(reg), (val))
+
+#define	OUTB(devc, reg, val)	ddi_put8(devc->regsh, (void *)(reg), (val))
+
+#endif /* AUDIOVIA823X_H */
diff --git a/usr/src/uts/common/io/audio/impl/audio.conf b/usr/src/uts/common/io/audio/impl/audio.conf
new file mode 100644
index 0000000000..ca5a83e886
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio.conf
@@ -0,0 +1,38 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Configuration file for the common audio driver.
+#
+
+#
+# We are a pseudo driver.
+#
+name="audio" parent="pseudo" instance=0;
+
+#
+# We want to always be present and never removed.  This is important
+# to ensure that device numbering is not lost due to the driver being
+# forced out of memory automatically.
+#
+ddi-forceattach=1;
+ddi-no-autodetach=1;
diff --git a/usr/src/uts/common/io/audio/impl/audio_client.c b/usr/src/uts/common/io/audio/impl/audio_client.c
new file mode 100644
index 0000000000..f3514df49b
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_client.c
@@ -0,0 +1,1675 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/sysmacros.h>
+#include <sys/list.h>
+#include <sys/file.h>
+#include <sys/open.h>
+#include <sys/stat.h>
+#include <sys/errno.h>
+#include <sys/atomic.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+
+#include "audio_impl.h"
+
+/*
+ * Audio Client implementation.
+ */
+
+/*
+ * Attenuation table for dB->linear conversion. Indexed in steps of
+ * 0.5 dB.  Table size is 25 dB (first entry is handled as mute).
+ *
+ * Notably, the last item in table is taken as 0 dB (i.e. maximum volume).
+ *
+ * Table contents can be calculated as follows (requires sunmath library):
+ *
+ * scale = AUDIO_VOL_SCALE;
+ * for (i = -50; i <= 0; i++) {
+ *     x = exp10(0.05 * i);
+ *     printf("%d: %f %.0f\n", i,  x, trunc(x * scale));
+ * }
+ *
+ */
+
+const uint16_t auimpl_db_table[AUDIO_DB_SIZE + 1] = {
+	0,   0,   1,   1,   1,   1,   1,   1,   2,   2,
+	2,   2,   3,   3,   4,   4,   5,   5,   6,   7,
+	8,   9,   10,  11,  12,  14,  16,  18,  20,  22,
+	25,  28,  32,  36,  40,  45,  51,  57,  64,  72,
+	80,  90,  101, 114, 128, 143, 161, 181, 203, 228,
+	256
+};
+
+void *
+auclnt_get_private(audio_client_t *c)
+{
+	return (c->c_private);
+}
+
+void
+auclnt_set_private(audio_client_t *c, void *private)
+{
+	c->c_private = private;
+}
+
+int
+auclnt_set_rate(audio_stream_t *sp, int rate)
+{
+	audio_parms_t	parms;
+	int		rv = 0;
+
+	/* basic sanity checks! */
+	if ((rate < 5000) || (rate > 192000)) {
+		return (EINVAL);
+	}
+	mutex_enter(&sp->s_lock);
+	parms = *sp->s_user_parms;
+	if (rate != parms.p_rate) {
+		parms.p_rate = rate;
+		rv = auimpl_format_setup(sp, &parms);
+	}
+	mutex_exit(&sp->s_lock);
+	return (rv);
+}
+
+int
+auclnt_get_rate(audio_stream_t *sp)
+{
+	return (sp->s_user_parms->p_rate);
+}
+
+unsigned
+auclnt_get_fragsz(audio_stream_t *sp)
+{
+	return (sp->s_fragbytes);
+}
+
+unsigned
+auclnt_get_framesz(audio_stream_t *sp)
+{
+	return (sp->s_framesz);
+}
+
+unsigned
+auclnt_get_nfrags(audio_stream_t *sp)
+{
+	return (sp->s_nfrags);
+}
+
+unsigned
+auclnt_get_nframes(audio_stream_t *sp)
+{
+	return (sp->s_nframes);
+}
+
+uint64_t
+auclnt_get_head(audio_stream_t *sp)
+{
+	return (sp->s_head);
+}
+
+uint64_t
+auclnt_get_tail(audio_stream_t *sp)
+{
+	return (sp->s_tail);
+}
+
+unsigned
+auclnt_get_hidx(audio_stream_t *sp)
+{
+	return (sp->s_hidx);
+}
+
+unsigned
+auclnt_get_tidx(audio_stream_t *sp)
+{
+	return (sp->s_tidx);
+}
+
+audio_stream_t *
+auclnt_input_stream(audio_client_t *c)
+{
+	return (&c->c_istream);
+}
+
+audio_stream_t *
+auclnt_output_stream(audio_client_t *c)
+{
+	return (&c->c_ostream);
+}
+
+unsigned
+auclnt_get_count(audio_stream_t *sp)
+{
+	unsigned	count;
+
+	mutex_enter(&sp->s_lock);
+	ASSERT((sp->s_head - sp->s_tail) <= sp->s_nframes);
+	count = (unsigned)(sp->s_head - sp->s_tail);
+	mutex_exit(&sp->s_lock);
+
+	return (count);
+}
+
+unsigned
+auclnt_consume(audio_stream_t *sp, unsigned n)
+{
+	mutex_enter(&sp->s_lock);
+
+	ASSERT(sp == &sp->s_client->c_istream);
+	n = max(n, sp->s_head - sp->s_tail);
+	sp->s_tail += n;
+	sp->s_tidx += n;
+	if (sp->s_tidx >= sp->s_nframes) {
+		sp->s_tidx -= sp->s_nframes;
+	}
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < sp->s_nframes);
+
+	mutex_exit(&sp->s_lock);
+
+	return (n);
+}
+
+unsigned
+auclnt_consume_data(audio_stream_t *sp, caddr_t dst, unsigned n)
+{
+	unsigned nframes;
+	unsigned framesz;
+	unsigned cnt;
+	caddr_t	data;
+
+	mutex_enter(&sp->s_lock);
+
+	nframes = sp->s_nframes;
+	framesz = sp->s_framesz;
+
+	ASSERT(sp == &sp->s_client->c_istream);
+	ASSERT(sp->s_head >= sp->s_tail);
+	ASSERT(sp->s_tidx < nframes);
+	ASSERT(sp->s_hidx < nframes);
+
+	cnt = n = min(n, sp->s_head - sp->s_tail);
+	data = sp->s_data + (sp->s_tidx * framesz);
+	do {
+		unsigned nf, nb;
+
+		nf = min(nframes - sp->s_tidx, n);
+		nb = nf * framesz;
+
+		bcopy(data, dst, nb);
+		dst += nb;
+		data += nb;
+
+		n -= nf;
+		sp->s_tail += nf;
+		sp->s_tidx += nf;
+		if (sp->s_tidx == nframes) {
+			sp->s_tidx = 0;
+			data = sp->s_data;
+		}
+	} while (n);
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_tidx < nframes);
+
+	mutex_exit(&sp->s_lock);
+
+	return (cnt);
+}
+
+unsigned
+auclnt_produce(audio_stream_t *sp, unsigned n)
+{
+	mutex_enter(&sp->s_lock);
+
+	ASSERT(sp == &sp->s_client->c_ostream);
+	n = max(n, sp->s_nframes - (sp->s_head - sp->s_tail));
+	sp->s_head += n;
+	sp->s_hidx += n;
+	if (sp->s_hidx >= sp->s_nframes) {
+		sp->s_hidx -= sp->s_nframes;
+	}
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < sp->s_nframes);
+
+	mutex_exit(&sp->s_lock);
+
+	return (n);
+}
+
+unsigned
+auclnt_produce_data(audio_stream_t *sp, caddr_t src, unsigned n)
+{
+	unsigned nframes;
+	unsigned framesz;
+	unsigned cnt;
+	caddr_t data;
+
+	mutex_enter(&sp->s_lock);
+
+	nframes = sp->s_nframes;
+	framesz = sp->s_framesz;
+
+	ASSERT(sp == &sp->s_client->c_ostream);
+	ASSERT(sp->s_head >= sp->s_tail);
+	ASSERT(sp->s_tidx < nframes);
+	ASSERT(sp->s_hidx < nframes);
+
+	cnt = n = min(n, nframes - (sp->s_head - sp->s_tail));
+	data = sp->s_data + (sp->s_hidx * framesz);
+	do {
+		unsigned nf, nb;
+
+		nf = min(nframes - sp->s_hidx, n);
+		nb = nf * framesz;
+
+		bcopy(src, data, nb);
+
+		src += nb;
+		data += nb;
+
+		n -= nf;
+		sp->s_head += nf;
+		sp->s_hidx += nf;
+		if (sp->s_hidx == nframes) {
+			sp->s_hidx = 0;
+			data = sp->s_data;
+		}
+	} while (n);
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < nframes);
+
+	mutex_exit(&sp->s_lock);
+
+	return (cnt);
+}
+
+int
+auclnt_read(audio_client_t *c, struct uio *uio)
+{
+	audio_stream_t	*sp = &c->c_istream;
+	unsigned	cnt;
+	int		rv = 0;
+	offset_t	loff;
+	int		eagain;
+
+	loff = uio->uio_loffset;
+	eagain = EAGAIN;
+
+	mutex_enter(&sp->s_lock);
+
+	if ((!sp->s_paused) && (!sp->s_running)) {
+		mutex_exit(&sp->s_lock);
+		auclnt_start(sp);
+		mutex_enter(&sp->s_lock);
+	}
+
+	ASSERT(sp->s_head >= sp->s_tail);
+	ASSERT(sp->s_tidx < sp->s_nframes);
+	ASSERT(sp->s_hidx < sp->s_nframes);
+
+	while (uio->uio_resid >= sp->s_framesz) {
+
+		while ((cnt = (sp->s_head - sp->s_tail)) == 0) {
+			if (uio->uio_fmode & (FNONBLOCK|FNDELAY)) {
+				mutex_exit(&sp->s_lock);
+				return (eagain);
+			}
+			if (cv_wait_sig(&sp->s_cv, &sp->s_lock) == 0) {
+				mutex_exit(&sp->s_lock);
+				return (EINTR);
+			}
+		}
+
+		cnt = min(cnt, sp->s_nframes - sp->s_tidx);
+		cnt = min(cnt, (uio->uio_resid / sp->s_framesz));
+
+		rv = uiomove(sp->s_data + (sp->s_tidx * sp->s_framesz),
+		    cnt * sp->s_framesz, UIO_READ, uio);
+		uio->uio_loffset = loff;
+		eagain = 0;
+
+		if (rv != 0) {
+			mutex_exit(&sp->s_lock);
+			return (rv);
+		}
+
+		sp->s_tail += cnt;
+		sp->s_tidx += cnt;
+		if (sp->s_tidx == sp->s_nframes) {
+			sp->s_tidx = 0;
+		}
+	}
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_tidx < sp->s_nframes);
+
+	/* round off any remaining partial bits */
+	uio->uio_resid = 0;
+
+	mutex_exit(&sp->s_lock);
+
+	return (rv);
+}
+
+int
+auclnt_write(audio_client_t *c, struct uio *uio)
+{
+	audio_stream_t *sp = &c->c_ostream;
+	unsigned	cnt;
+	int		rv = 0;
+	offset_t	loff;
+	int		eagain;
+
+	loff = uio->uio_loffset;
+	eagain = EAGAIN;
+
+	mutex_enter(&sp->s_lock);
+
+	ASSERT(sp->s_head >= sp->s_tail);
+	ASSERT(sp->s_tidx < sp->s_nframes);
+	ASSERT(sp->s_hidx < sp->s_nframes);
+
+	while (uio->uio_resid >= sp->s_framesz) {
+
+		while ((cnt = sp->s_nframes - (sp->s_head - sp->s_tail)) == 0) {
+			if (uio->uio_fmode & (FNONBLOCK|FNDELAY)) {
+				mutex_exit(&sp->s_lock);
+				return (eagain);
+			}
+			if (cv_wait_sig(&sp->s_cv, &sp->s_lock) == 0) {
+				mutex_exit(&sp->s_lock);
+				return (EINTR);
+			}
+		}
+
+		cnt = min(cnt, sp->s_nframes - sp->s_hidx);
+		cnt = min(cnt, (uio->uio_resid / sp->s_framesz));
+
+		rv = uiomove(sp->s_data + (sp->s_hidx * sp->s_framesz),
+		    cnt * sp->s_framesz, UIO_WRITE, uio);
+		uio->uio_loffset = loff;
+		eagain = 0;
+
+		if (rv != 0) {
+			mutex_exit(&sp->s_lock);
+			return (rv);
+		}
+
+		sp->s_head += cnt;
+		sp->s_hidx += cnt;
+		if (sp->s_hidx == sp->s_nframes) {
+			sp->s_hidx = 0;
+		}
+
+		if ((!sp->s_paused) && (!sp->s_running) &&
+		    ((sp->s_head - sp->s_tail) > sp->s_fragfr)) {
+			mutex_exit(&sp->s_lock);
+			auclnt_start(sp);
+			mutex_enter(&sp->s_lock);
+		}
+	}
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < sp->s_nframes);
+
+	/* round off any remaining partial bits */
+	uio->uio_resid = 0;
+
+	mutex_exit(&sp->s_lock);
+
+	return (rv);
+}
+
+int
+auclnt_chpoll(audio_client_t *c, short events, int anyyet, short *reventsp,
+    struct pollhead **phpp)
+{
+	audio_stream_t	*sp;
+	short nev = 0;
+
+	if (events & (POLLIN | POLLRDNORM)) {
+		sp = &c->c_istream;
+		mutex_enter(&sp->s_lock);
+		if ((sp->s_head - sp->s_tail) > sp->s_fragfr) {
+			nev = POLLIN | POLLRDNORM;
+		}
+		mutex_exit(&sp->s_lock);
+	}
+
+	if (events & POLLOUT) {
+		sp = &c->c_ostream;
+		mutex_enter(&sp->s_lock);
+		if ((sp->s_nframes - (sp->s_head - sp->s_tail)) >
+		    sp->s_fragfr) {
+			nev = POLLOUT;
+		}
+		mutex_exit(&sp->s_lock);
+	}
+
+	if (nev) {
+		*reventsp = nev & events;
+	} else {
+		*reventsp = 0;
+		if (!anyyet) {
+			*phpp = &c->c_pollhead;
+		}
+	}
+	return (0);
+}
+
+void
+auclnt_pollwakeup(audio_client_t *c, short events)
+{
+	pollwakeup(&c->c_pollhead, events);
+}
+
+void
+auclnt_get_output_qlen(audio_client_t *c, unsigned *slen, unsigned *flen)
+{
+	audio_stream_t	*sp = &c->c_ostream;
+	audio_engine_t	*e = sp->s_engine;
+	uint64_t	el, sl;
+	unsigned	cnt, er, sr;
+
+	if (e == NULL) {
+		/* if no output engine, can't do it! */
+		*slen = 0;
+		*flen = 0;
+		return;
+	}
+
+	mutex_enter(&e->e_lock);
+	mutex_enter(&sp->s_lock);
+	el = ENG_QLEN(e) + (e->e_head - e->e_tail);
+	er = e->e_rate;
+	sl = sp->s_cnv_cnt;
+	sr = sp->s_user_parms->p_rate;
+	cnt = (unsigned)(sp->s_head - sp->s_tail);
+	mutex_exit(&sp->s_lock);
+	mutex_exit(&e->e_lock);
+
+	/* engine frames converted to stream rate, plus stream frames */
+	*slen = cnt;
+	*flen = ((unsigned)(((el * sr) / er) + sl));
+}
+
+int
+auclnt_set_format(audio_stream_t *sp, int fmt)
+{
+	audio_parms_t	parms;
+	int		rv = 0;
+
+	/*
+	 * AC3: If we select an AC3 format, then we have to allocate
+	 * another engine.  Normally this will be an output only
+	 * engine.  However, for now we aren't supporting AC3
+	 * passthru.
+	 */
+
+	switch (fmt) {
+	case AUDIO_FORMAT_U8:
+	case AUDIO_FORMAT_ULAW:
+	case AUDIO_FORMAT_ALAW:
+	case AUDIO_FORMAT_S8:
+	case AUDIO_FORMAT_S16_LE:
+	case AUDIO_FORMAT_S16_BE:
+	case AUDIO_FORMAT_U16_LE:
+	case AUDIO_FORMAT_U16_BE:
+	case AUDIO_FORMAT_S24_LE:
+	case AUDIO_FORMAT_S24_BE:
+	case AUDIO_FORMAT_S32_LE:
+	case AUDIO_FORMAT_S32_BE:
+	case AUDIO_FORMAT_S24_PACKED:
+		break;
+
+	case AUDIO_FORMAT_AC3:		/* AC3: PASSTHRU */
+	default:
+		return (ENOTSUP);
+	}
+
+
+	mutex_enter(&sp->s_lock);
+	parms = *sp->s_user_parms;
+
+	/*
+	 * Optimization.  Some personalities send us the same format
+	 * over and over again.  (Sun personality does this
+	 * repeatedly.)  setup_src is potentially expensive, so we
+	 * avoid doing it unless we really need to.
+	 */
+	if (fmt != parms.p_format) {
+		/*
+		 * Note that setting the format doesn't check that the
+		 * audio streams have been paused.  As a result, any
+		 * data still playing or recording will probably get
+		 * misinterpreted.  It would be smart if the client
+		 * application paused/stopped playback before changing
+		 * formats.
+		 */
+		parms.p_format = fmt;
+		rv = auimpl_format_setup(sp, &parms);
+	}
+	mutex_exit(&sp->s_lock);
+
+	return (rv);
+}
+
+int
+auclnt_get_format(audio_stream_t *sp)
+{
+	return (sp->s_user_parms->p_format);
+}
+
+int
+auclnt_get_output_format(audio_client_t *c)
+{
+	return (c->c_ostream.s_user_parms->p_format);
+}
+
+int
+auclnt_get_input_format(audio_client_t *c)
+{
+	return (c->c_istream.s_user_parms->p_format);
+}
+
+int
+auclnt_set_channels(audio_stream_t *sp, int nchan)
+{
+	audio_parms_t	parms;
+	int		rv = 0;
+
+	/* Validate setting */
+	if ((nchan > AUDIO_MAX_CHANNELS) || (nchan < 1)) {
+		return (EINVAL);
+	}
+
+	mutex_enter(&sp->s_lock);
+	parms = *sp->s_user_parms;
+	if (nchan != parms.p_nchan) {
+		parms.p_nchan = nchan;
+		rv = auimpl_format_setup(sp, &parms);
+	}
+	mutex_exit(&sp->s_lock);
+
+	return (rv);
+}
+
+int
+auclnt_get_channels(audio_stream_t *sp)
+{
+	return (sp->s_user_parms->p_nchan);
+}
+
+void
+auimpl_set_gain_master(audio_stream_t *sp, uint8_t gain)
+{
+	uint32_t	scaled;
+
+	if (gain > 100) {
+		gain = 0;
+	}
+
+	mutex_enter(&sp->s_lock);
+	if (sp->s_gain_master == gain) {
+		mutex_exit(&sp->s_lock);
+		return;
+	}
+
+	/*
+	 * calculate the scaled values.  Done now to avoid calculations
+	 * later.
+	 */
+	scaled = (gain * sp->s_gain_pct * AUDIO_DB_SIZE) / (100 * 100);
+
+	sp->s_gain_master = gain;
+	sp->s_gain_scaled = auimpl_db_table[scaled];
+
+	if (!sp->s_muted) {
+		sp->s_gain_eff = sp->s_gain_scaled;
+	}
+	mutex_exit(&sp->s_lock);
+
+	/*
+	 * No need to notify clients, since they can't see this update.
+	 */
+}
+
+void
+auclnt_set_gain(audio_stream_t *sp, uint8_t gain)
+{
+	uint32_t	scaled;
+
+	if (gain > 100) {
+		gain = 0;
+	}
+
+	mutex_enter(&sp->s_lock);
+
+	/* if no change, don't bother doing updates */
+	if (sp->s_gain_pct == gain) {
+		mutex_exit(&sp->s_lock);
+		return;
+	}
+
+	/*
+	 * calculate the scaled values.  Done now to avoid calculations
+	 * later.
+	 */
+	scaled = (gain * sp->s_gain_master * AUDIO_DB_SIZE) / (100 * 100);
+
+	sp->s_gain_pct = gain;
+	sp->s_gain_scaled = auimpl_db_table[scaled];
+
+	if (!sp->s_muted) {
+		sp->s_gain_eff = sp->s_gain_scaled;
+	}
+	mutex_exit(&sp->s_lock);
+
+	auclnt_notify_dev(sp->s_client->c_dev);
+}
+
+uint8_t
+auclnt_get_gain(audio_stream_t *sp)
+{
+	return (sp->s_gain_pct);
+}
+
+void
+auclnt_set_muted(audio_stream_t *sp, boolean_t muted)
+{
+	mutex_enter(&sp->s_lock);
+
+	/* if no work change, don't bother doing updates */
+	if (sp->s_muted == muted) {
+		mutex_exit(&sp->s_lock);
+		return;
+	}
+
+	sp->s_muted = muted;
+	if (muted) {
+		sp->s_gain_eff = 0;
+	} else {
+		sp->s_gain_eff = sp->s_gain_scaled;
+	}
+	mutex_exit(&sp->s_lock);
+
+	auclnt_notify_dev(sp->s_client->c_dev);
+}
+
+boolean_t
+auclnt_get_muted(audio_stream_t *sp)
+{
+	return (sp->s_muted);
+}
+
+void
+auclnt_start(audio_stream_t *sp)
+{
+	mutex_enter(&sp->s_lock);
+	sp->s_running = B_TRUE;
+	mutex_exit(&sp->s_lock);
+}
+
+void
+auclnt_stop(audio_stream_t *sp)
+{
+	mutex_enter(&sp->s_lock);
+	/* if running, then stop it */
+	if (sp->s_running) {
+		sp->s_running = B_FALSE;
+		/*
+		 * if we stopped the engine, we might need to wake up
+		 * a thread that is waiting for drain to complete.
+		 */
+		cv_broadcast(&sp->s_cv);
+	}
+	mutex_exit(&sp->s_lock);
+}
+
+/*
+ * When pausing, no new data will be played after the most recently
+ * mixed samples have played.  However, the audio engine will continue
+ * to play (possibly just silence).
+ *
+ * Note that we don't reference count the device, or release/close the
+ * engine here.  Once fired up, the engine continues running unil it
+ * is closed.
+ */
+void
+auclnt_set_paused(audio_stream_t *sp)
+{
+	mutex_enter(&sp->s_lock);
+	if (sp->s_paused) {
+		mutex_exit(&sp->s_lock);
+		return;
+	}
+	sp->s_paused = B_TRUE;
+	mutex_exit(&sp->s_lock);
+
+	auclnt_stop(sp);
+
+	auclnt_notify_dev(sp->s_client->c_dev);
+}
+
+void
+auclnt_clear_paused(audio_stream_t *sp)
+{
+	mutex_enter(&sp->s_lock);
+	if (!sp->s_paused) {
+		mutex_exit(&sp->s_lock);
+		return;
+	}
+	sp->s_paused = B_FALSE;
+	mutex_exit(&sp->s_lock);
+}
+
+boolean_t
+auclnt_is_paused(audio_stream_t *sp)
+{
+	return (sp->s_paused);
+}
+
+void
+auclnt_flush(audio_stream_t *sp)
+{
+	mutex_enter(&sp->s_lock);
+	if (sp == &sp->s_client->c_ostream) {
+		sp->s_tail = sp->s_head;
+		sp->s_tidx = sp->s_hidx;
+	} else {
+		sp->s_head = sp->s_tail;
+		sp->s_hidx = sp->s_tidx;
+	}
+	sp->s_cnv_cnt = 0;
+	mutex_exit(&sp->s_lock);
+}
+
+int
+auclnt_get_oflag(audio_client_t *c)
+{
+	return (c->c_omode);
+}
+
+/*
+ * These routines should not be accessed by client "personality"
+ * implementations, but are for private framework use only.
+ */
+
+static list_t			auimpl_clients;
+static krwlock_t		auimpl_client_lock;
+static audio_client_ops_t	*audio_client_ops[AUDIO_MN_TYPE_MASK + 1];
+
+void
+auimpl_client_init(void)
+{
+	rw_init(&auimpl_client_lock, NULL, RW_DRIVER, NULL);
+	list_create(&auimpl_clients, sizeof (struct audio_client),
+	    offsetof(struct audio_client, c_global_linkage));
+}
+
+void
+auimpl_client_fini(void)
+{
+	rw_destroy(&auimpl_client_lock);
+	list_destroy(&auimpl_clients);
+}
+
+static int
+auimpl_stream_init(audio_stream_t *sp, audio_client_t *c)
+{
+	mutex_init(&sp->s_lock, NULL, MUTEX_DRIVER, NULL);
+	cv_init(&sp->s_cv, NULL, CV_DRIVER, NULL);
+	sp->s_client = c;
+
+	if (sp == &c->c_ostream) {
+		sp->s_user_parms = &sp->s_cnv_src_parms;
+		sp->s_phys_parms = &sp->s_cnv_dst_parms;
+		sp->s_engcap = ENGINE_OUTPUT_CAP;
+	} else {
+		ASSERT(sp == &c->c_istream);
+		sp->s_user_parms = &sp->s_cnv_dst_parms;
+		sp->s_phys_parms = &sp->s_cnv_src_parms;
+		sp->s_engcap = ENGINE_INPUT_CAP;
+	}
+
+	/* for now initialize conversion parameters */
+	sp->s_src_quality = 3;	/* reasonable compromise for now */
+	sp->s_cnv_dst_nchan = 2;
+	sp->s_cnv_dst_format = AUDIO_FORMAT_S24_NE;
+	sp->s_cnv_dst_rate = 48000;
+	sp->s_cnv_src_nchan = 2;
+	sp->s_cnv_src_format = AUDIO_FORMAT_S24_NE;
+	sp->s_cnv_src_rate = 48000;
+
+	/* set volume/gain all the way up */
+	sp->s_muted = B_FALSE;
+	sp->s_gain_pct = 0;
+	sp->s_gain_scaled = AUDIO_VOL_SCALE;
+	sp->s_gain_eff = AUDIO_VOL_SCALE;
+
+	/*
+	 * We have to start off with a reasonable buffer and
+	 * interrupt configuration.
+	 */
+	sp->s_allocsz = 65536;
+	sp->s_data = ddi_umem_alloc(sp->s_allocsz, DDI_UMEM_NOSLEEP,
+	    &sp->s_cookie);
+	if (sp->s_data == NULL) {
+		sp->s_allocsz = 0;
+		audio_dev_warn(c->c_dev, "ddi_umem_alloc failed");
+		return (ENOMEM);
+	}
+	/* make sure no stale data left in stream */
+	bzero(sp->s_data, sp->s_allocsz);
+
+	/*
+	 * Allocate SRC and data conversion state.
+	 */
+	mutex_enter(&sp->s_lock);
+	if (auimpl_format_alloc(sp) != 0) {
+		mutex_exit(&sp->s_lock);
+		return (ENOMEM);
+	}
+
+	mutex_exit(&sp->s_lock);
+
+	return (0);
+}
+
+
+static void
+audio_stream_fini(audio_stream_t *sp)
+{
+	auimpl_format_free(sp);
+	if (sp->s_cnv_buf0)
+		kmem_free(sp->s_cnv_buf0, sp->s_cnv_max);
+	if (sp->s_cnv_buf1)
+		kmem_free(sp->s_cnv_buf1, sp->s_cnv_max);
+	mutex_destroy(&sp->s_lock);
+	cv_destroy(&sp->s_cv);
+	if (sp->s_data != NULL) {
+		ddi_umem_free(sp->s_cookie);
+		sp->s_data = NULL;
+	}
+}
+
+void
+auimpl_client_task(void *arg)
+{
+	audio_client_t		*c = arg;
+
+	mutex_enter(&c->c_lock);
+
+	for (;;) {
+		if (c->c_closing) {
+			break;
+		}
+
+		if (c->c_do_output) {
+			c->c_do_output = B_FALSE;
+
+			mutex_exit(&c->c_lock);
+			if (c->c_output != NULL)
+				c->c_output(c);
+			mutex_enter(&c->c_lock);
+			continue;
+		}
+
+		if (c->c_do_input) {
+			c->c_do_input = B_FALSE;
+
+			mutex_exit(&c->c_lock);
+			if (c->c_input != NULL)
+				c->c_input(c);
+			mutex_enter(&c->c_lock);
+			continue;
+		}
+
+		if (c->c_do_notify) {
+			c->c_do_notify = B_FALSE;
+
+			mutex_exit(&c->c_lock);
+			if (c->c_notify != NULL)
+				c->c_notify(c);
+			mutex_enter(&c->c_lock);
+			continue;
+		}
+
+		if (c->c_do_drain) {
+			c->c_do_drain = B_FALSE;
+
+			mutex_exit(&c->c_lock);
+			if (c->c_drain != NULL)
+				c->c_drain(c);
+			mutex_enter(&c->c_lock);
+			continue;
+		}
+
+		/* if we got here, we had no work to do */
+		cv_wait(&c->c_cv, &c->c_lock);
+	}
+	mutex_exit(&c->c_lock);
+}
+
+int
+auclnt_start_drain(audio_client_t *c)
+{
+	audio_stream_t	*sp;
+	int		rv;
+
+	sp = &c->c_ostream;
+
+	/* start an asynchronous drain operation. */
+	mutex_enter(&sp->s_lock);
+	if (sp->s_paused || !sp->s_running) {
+		rv = EALREADY;
+	} else {
+		sp->s_draining = B_TRUE;
+		rv = 0;
+	}
+	mutex_exit(&sp->s_lock);
+	return (rv);
+}
+
+int
+auclnt_drain(audio_client_t *c)
+{
+	audio_stream_t	*sp;
+
+	sp = &c->c_ostream;
+
+	/*
+	 * Note: Drain logic will automatically "stop" the stream when
+	 * the drain threshold has been reached.  So all we have to do
+	 * is wait for the stream to stop.
+	 */
+	mutex_enter(&sp->s_lock);
+	sp->s_draining = B_TRUE;
+	while (sp->s_draining && sp->s_running && !sp->s_paused) {
+		if (cv_wait_sig(&sp->s_cv, &sp->s_lock) == 0) {
+			mutex_exit(&sp->s_lock);
+			return (EINTR);
+		}
+	}
+	mutex_exit(&sp->s_lock);
+	return (0);
+}
+
+audio_client_t *
+auimpl_client_create(dev_t dev)
+{
+	audio_client_ops_t	*ops;
+	audio_client_t		*c;
+	audio_client_t		*next;
+	list_t			*list = &auimpl_clients;
+	minor_t			minor;
+	audio_dev_t		*d;
+	char			scratch[80];
+	static uint64_t		unique = 0;
+
+	/* validate minor number */
+	minor = getminor(dev) & AUDIO_MN_TYPE_MASK;
+	if ((ops = audio_client_ops[minor]) == NULL) {
+		return (NULL);
+	}
+
+	/* lookup device instance */
+	if ((d = auimpl_dev_hold_by_devt(dev)) == NULL) {
+		audio_dev_warn(NULL, "no audio_dev for dev_t %d,%d",
+		    getmajor(dev), getminor(dev));
+		return (NULL);
+	}
+
+	if ((c = kmem_zalloc(sizeof (*c), KM_NOSLEEP)) == NULL) {
+		audio_dev_warn(d, "unable to allocate client structure");
+		auimpl_dev_release(d);
+		return (NULL);
+	}
+	c->c_dev = d;
+
+	mutex_init(&c->c_lock, NULL, MUTEX_DRIVER, NULL);
+	cv_init(&c->c_cv, NULL, CV_DRIVER, NULL);
+
+	if ((auimpl_stream_init(&c->c_ostream, c) != 0) ||
+	    (auimpl_stream_init(&c->c_istream, c) != 0)) {
+		goto failed;
+	}
+
+	c->c_major =		getmajor(dev);
+	c->c_origminor =	getminor(dev);
+	c->c_ops =		*ops;
+
+	(void) snprintf(scratch, sizeof (scratch), "auclnt%" PRIx64,
+	    atomic_inc_64_nv(&unique));
+	c->c_tq = ddi_taskq_create(NULL, scratch, 1, TASKQ_DEFAULTPRI, 0);
+	if (c->c_tq == NULL) {
+		audio_dev_warn(d, "client taskq_create failed");
+		goto failed;
+	}
+
+	/*
+	 * We hold the client lock here.
+	 */
+	rw_enter(&auimpl_client_lock, RW_WRITER);
+
+	minor = AUDIO_MN_CLONE_MASK;
+	for (next = list_head(list); next; next = list_next(list, next)) {
+		if (next->c_minor > minor) {
+			break;
+		}
+		minor++;
+	}
+	if (minor >= MAXMIN32) {
+		rw_exit(&auimpl_client_lock);
+		goto failed;
+	}
+	c->c_minor = minor;
+	list_insert_before(list, next, c);
+
+	rw_exit(&auimpl_client_lock);
+
+
+	return (c);
+
+failed:
+	auimpl_dev_release(d);
+	if (c->c_tq != NULL) {
+		ddi_taskq_destroy(c->c_tq);
+	}
+	audio_stream_fini(&c->c_ostream);
+	audio_stream_fini(&c->c_istream);
+	mutex_destroy(&c->c_lock);
+	cv_destroy(&c->c_cv);
+	kmem_free(c, sizeof (*c));
+	return (NULL);
+}
+
+void
+auimpl_client_destroy(audio_client_t *c)
+{
+	/* remove us from the global list */
+	rw_enter(&auimpl_client_lock, RW_WRITER);
+	list_remove(&auimpl_clients, c);
+	rw_exit(&auimpl_client_lock);
+
+	ASSERT(!c->c_istream.s_running);
+	ASSERT(!c->c_istream.s_running);
+
+	/* release the device reference count */
+	auimpl_dev_release(c->c_dev);
+	c->c_dev = NULL;
+
+	ddi_taskq_destroy(c->c_tq);
+
+	mutex_destroy(&c->c_lock);
+	cv_destroy(&c->c_cv);
+
+	audio_stream_fini(&c->c_istream);
+	audio_stream_fini(&c->c_ostream);
+	kmem_free(c, sizeof (*c));
+}
+
+void
+auclnt_close(audio_client_t *c)
+{
+	audio_dev_t	*d = c->c_dev;
+
+	/* stop the engines if they are running */
+	auclnt_stop(&c->c_istream);
+	auclnt_stop(&c->c_ostream);
+
+	rw_enter(&d->d_clnt_lock, RW_WRITER);
+	list_remove(&d->d_clients, c);
+	rw_exit(&d->d_clnt_lock);
+
+	mutex_enter(&c->c_lock);
+	/* if in transition need to wait for other thread to release */
+	while (c->c_refcnt) {
+		cv_wait(&c->c_cv, &c->c_lock);
+	}
+	c->c_closing = B_TRUE;
+	cv_broadcast(&c->c_cv);
+	mutex_exit(&c->c_lock);
+
+	/* make sure taskq has drained */
+	ddi_taskq_wait(c->c_tq);
+
+	/* release any engines that we were holding */
+	auimpl_engine_close(&c->c_ostream);
+	auimpl_engine_close(&c->c_istream);
+}
+
+audio_dev_t *
+auclnt_hold_dev_by_index(int index)
+{
+	return (auimpl_dev_hold_by_index(index));
+}
+
+void
+auclnt_release_dev(audio_dev_t *dev)
+{
+	auimpl_dev_release(dev);
+}
+
+audio_client_t *
+auclnt_hold_by_devt(dev_t dev)
+{
+	minor_t	mn = getminor(dev);
+	major_t mj = getmajor(dev);
+	list_t *list;
+	audio_client_t *c;
+
+	list = &auimpl_clients;
+	/* linked list search is kind of inefficient, but it works */
+	rw_enter(&auimpl_client_lock, RW_READER);
+	for (c = list_head(list); c != NULL; c = list_next(list, c)) {
+		if ((c->c_major == mj) && (c->c_minor == mn)) {
+			mutex_enter(&c->c_lock);
+			if (c->c_is_open) {
+				c->c_refcnt++;
+				mutex_exit(&c->c_lock);
+			} else {
+				mutex_exit(&c->c_lock);
+				c = NULL;
+			}
+			break;
+		}
+	}
+	rw_exit(&auimpl_client_lock);
+	return (c);
+}
+
+void
+auclnt_release(audio_client_t *c)
+{
+	mutex_enter(&c->c_lock);
+	c->c_refcnt--;
+	if (c->c_refcnt == 0)
+		cv_broadcast(&c->c_cv);
+	mutex_exit(&c->c_lock);
+}
+
+void
+auclnt_dev_walk_clients(audio_dev_t *d,
+    int (*walker)(audio_client_t *, void *),
+    void *arg)
+{
+	list_t		*l = &d->d_clients;
+	audio_client_t	*c;
+	int		rv;
+
+	rw_enter(&d->d_clnt_lock, RW_READER);
+restart:
+	for (c = list_head(l); c != NULL; c = list_next(l, c)) {
+		rv = (walker(c, arg));
+		if (rv == AUDIO_WALK_STOP) {
+			break;
+		} else if (rv == AUDIO_WALK_RESTART) {
+			goto restart;
+		}
+	}
+	rw_exit(&d->d_clnt_lock);
+}
+
+
+int
+auclnt_open(audio_client_t *c, unsigned fmts, int oflag)
+{
+	audio_stream_t	*sp;
+	audio_dev_t	*d = c->c_dev;
+	int		rv = 0;
+	int		flags;
+	audio_parms_t	parms;
+
+	flags = 0;
+	if (oflag & FNDELAY)
+		flags |= ENGINE_NDELAY;
+
+	if (oflag & FWRITE) {
+		sp = &c->c_ostream;
+		rv = auimpl_engine_open(d, fmts, flags | ENGINE_OUTPUT, sp);
+
+		if (rv != 0) {
+			goto done;
+		}
+		mutex_enter(&sp->s_lock);
+		parms = *sp->s_user_parms;
+		rv = auimpl_format_setup(sp, &parms);
+		mutex_exit(&sp->s_lock);
+		if (rv != 0) {
+			goto done;
+		}
+	}
+
+	if (oflag & FREAD) {
+		sp = &c->c_istream;
+		rv = auimpl_engine_open(d, fmts, flags | ENGINE_INPUT, sp);
+
+		if (rv != 0) {
+			goto done;
+		}
+		mutex_enter(&sp->s_lock);
+		parms = *sp->s_user_parms;
+		rv = auimpl_format_setup(sp, &parms);
+		mutex_exit(&sp->s_lock);
+		if (rv != 0) {
+			goto done;
+		}
+	}
+
+	if (ddi_taskq_dispatch(c->c_tq, auimpl_client_task, c, DDI_NOSLEEP) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(d, "unable to start client taskq");
+		rv = ENOMEM;
+	}
+
+done:
+	if (rv != 0) {
+		/* close any engines that we opened */
+		auimpl_engine_close(&c->c_ostream);
+		auimpl_engine_close(&c->c_istream);
+	} else {
+		rw_enter(&d->d_clnt_lock, RW_WRITER);
+		list_insert_tail(&d->d_clients, c);
+		c->c_ostream.s_gain_master = d->d_pcmvol;
+		c->c_istream.s_gain_master = 100;
+		rw_exit(&d->d_clnt_lock);
+		auclnt_set_gain(&c->c_ostream, 100);
+		auclnt_set_gain(&c->c_istream, 100);
+	}
+
+	return (rv);
+}
+
+minor_t
+auclnt_get_minor(audio_client_t *c)
+{
+	return (c->c_minor);
+}
+
+minor_t
+auclnt_get_original_minor(audio_client_t *c)
+{
+	return (c->c_origminor);
+}
+
+minor_t
+auclnt_get_minor_type(audio_client_t *c)
+{
+	return (c->c_origminor & AUDIO_MN_TYPE_MASK);
+}
+
+pid_t
+auclnt_get_pid(audio_client_t *c)
+{
+	return (c->c_pid);
+}
+
+cred_t *
+auclnt_get_cred(audio_client_t *c)
+{
+	return (c->c_cred);
+}
+
+audio_dev_t *
+auclnt_get_dev(audio_client_t *c)
+{
+	return (c->c_dev);
+}
+
+int
+auclnt_get_dev_number(audio_dev_t *dev)
+{
+	return (dev->d_number);
+}
+
+int
+auclnt_get_dev_index(audio_dev_t *dev)
+{
+	return (dev->d_index);
+}
+
+const char *
+auclnt_get_dev_name(audio_dev_t *dev)
+{
+	return (dev->d_name);
+}
+
+const char *
+auclnt_get_dev_driver(audio_dev_t *dev)
+{
+	return (ddi_driver_name(dev->d_dip));
+}
+
+dev_info_t *
+auclnt_get_dev_devinfo(audio_dev_t *dev)
+{
+	return (dev->d_dip);
+}
+
+const char *
+auclnt_get_dev_hw_info(audio_dev_t *dev, void **iter)
+{
+	struct audio_infostr *isp = *iter;
+	if (isp == NULL) {
+		isp = list_head(&dev->d_hwinfo);
+	} else {
+		isp = list_next(&dev->d_hwinfo, isp);
+	}
+
+	*iter = isp;
+	return (isp ? isp->i_line : NULL);
+}
+
+int
+auclnt_get_dev_instance(audio_dev_t *dev)
+{
+	return (dev->d_instance);
+}
+
+const char *
+auclnt_get_dev_description(audio_dev_t *dev)
+{
+	return (dev->d_desc);
+}
+
+const char *
+auclnt_get_dev_version(audio_dev_t *dev)
+{
+	return (dev->d_vers);
+}
+
+unsigned
+auclnt_get_dev_capab(audio_dev_t *dev)
+{
+	uint32_t	flags;
+	unsigned	caps = 0;
+
+	flags = dev->d_flags;
+
+	if (flags & DEV_OUTPUT_CAP)
+		caps |= AUDIO_CLIENT_CAP_PLAY;
+	if (flags & DEV_INPUT_CAP)
+		caps |= AUDIO_CLIENT_CAP_RECORD;
+	if (flags & DEV_DUPLEX_CAP)
+		caps |= AUDIO_CLIENT_CAP_DUPLEX;
+
+	/* AC3: deal with formats that don't support mixing */
+
+	return (caps);
+}
+
+void
+auclnt_notify_dev(audio_dev_t *dev)
+{
+	list_t *l = &dev->d_clients;
+	audio_client_t *c;
+
+	rw_enter(&dev->d_clnt_lock, RW_READER);
+	for (c = list_head(l); c != NULL; c = list_next(l, c)) {
+		mutex_enter(&c->c_lock);
+		c->c_do_notify = B_TRUE;
+		cv_broadcast(&c->c_cv);
+		mutex_exit(&c->c_lock);
+	}
+	rw_exit(&dev->d_clnt_lock);
+}
+
+uint64_t
+auclnt_get_samples(audio_stream_t *sp)
+{
+	uint64_t	n;
+
+	mutex_enter(&sp->s_lock);
+	n = sp->s_samples;
+	mutex_exit(&sp->s_lock);
+	return (n);
+}
+
+void
+auclnt_set_samples(audio_stream_t *sp, uint64_t n)
+{
+	mutex_enter(&sp->s_lock);
+	sp->s_samples = n;
+	mutex_exit(&sp->s_lock);
+}
+
+uint64_t
+auclnt_get_errors(audio_stream_t *sp)
+{
+	uint64_t	n;
+	mutex_enter(&sp->s_lock);
+	n = sp->s_errors;
+	mutex_exit(&sp->s_lock);
+	return (n);
+}
+
+void
+auclnt_set_errors(audio_stream_t *sp, uint64_t n)
+{
+	mutex_enter(&sp->s_lock);
+	sp->s_errors = n;
+	mutex_exit(&sp->s_lock);
+}
+
+void
+auclnt_register_ops(minor_t minor, audio_client_ops_t *ops)
+{
+	/* we control minor number allocations, no need for runtime checks */
+	ASSERT(minor <= AUDIO_MN_TYPE_MASK);
+
+	audio_client_ops[minor] = ops;
+}
+
+int
+auimpl_create_minors(audio_dev_t *d)
+{
+	char			path[MAXPATHLEN];
+	int			rv = 0;
+	minor_t			minor;
+	audio_client_ops_t	*ops;
+	char			*nt;
+
+	for (int i = 0; i <= AUDIO_MN_TYPE_MASK; i++) {
+
+		if ((ops = audio_client_ops[i]) == NULL)
+			continue;
+
+		if (ops->aco_dev_init != NULL)
+			d->d_minor_data[i] = ops->aco_dev_init(d);
+
+		switch (i) {
+		case AUDIO_MINOR_SNDSTAT:
+			if (!(d->d_flags & DEV_SNDSTAT_CAP)) {
+				continue;
+			}
+			nt = DDI_PSEUDO;
+			break;
+
+		default:
+			if (!(d->d_flags & (DEV_INPUT_CAP| DEV_OUTPUT_CAP))) {
+				continue;
+			}
+			nt = DDI_NT_AUDIO;
+			break;
+		}
+
+		if (ops->aco_minor_prefix != NULL) {
+
+			minor = AUDIO_MKMN(d->d_instance, i);
+			(void) snprintf(path, sizeof (path),
+			    "%s%d", ops->aco_minor_prefix, d->d_instance);
+
+			rv = ddi_create_minor_node(d->d_dip, path, S_IFCHR,
+			    minor, nt, 0);
+
+			if (rv != 0)
+				break;
+		}
+	}
+	return (rv);
+}
+
+void
+auimpl_remove_minors(audio_dev_t *d)
+{
+	char			path[MAXPATHLEN];
+	audio_client_ops_t	*ops;
+
+	for (int i = 0; i <= AUDIO_MN_TYPE_MASK; i++) {
+		if ((ops = audio_client_ops[i]) == NULL)
+			continue;
+		if (ops->aco_minor_prefix != NULL) {
+			(void) snprintf(path, sizeof (path), "%s%d",
+			    ops->aco_minor_prefix, d->d_instance);
+			(void) ddi_remove_minor_node(d->d_dip, path);
+		}
+
+		if (ops->aco_dev_fini != NULL)
+			ops->aco_dev_fini(d->d_minor_data[i]);
+	}
+}
+
+void *
+auclnt_get_dev_minor_data(audio_dev_t *d, minor_t mn)
+{
+	ASSERT(mn < (1U << AUDIO_MN_TYPE_NBITS));
+	return (d->d_minor_data[mn]);
+}
+
+void *
+auclnt_get_minor_data(audio_client_t *c, minor_t mn)
+{
+	ASSERT(mn < (1U << AUDIO_MN_TYPE_NBITS));
+	return (c->c_dev->d_minor_data[mn]);
+}
+
+/*
+ * This will walk all controls registered to a clients device and callback
+ * to walker for each one with its audio_ctrl. Note this data
+ * must be considered read only by walker.
+ *
+ * Note that walk_func may return values to continue (AUDIO_WALK_CONTINUE)
+ * or stop walk (AUDIO_WALK_STOP).
+ *
+ */
+void
+auclnt_walk_controls(audio_dev_t *d,
+    int (*walker)(audio_ctrl_t *, void *),
+    void *arg)
+{
+	audio_ctrl_t *ctrl;
+
+	rw_enter(&d->d_ctrl_lock, RW_READER);
+	for (ctrl = list_head(&d->d_controls); ctrl;
+	    ctrl = list_next(&d->d_controls, ctrl)) {
+		if (walker(ctrl, arg) == AUDIO_WALK_STOP)
+			break;
+	}
+	rw_exit(&d->d_ctrl_lock);
+}
+
+/*
+ * This will search all controls attached to an
+ * audio device for a control with the desired name.
+ *
+ * d    - the audio device to look on
+ * name - name of the control being looked for.
+ *
+ * On successful return a ctrl handle will be returned. On
+ * failure NULL is returned.
+ */
+audio_ctrl_t *
+auclnt_find_control(audio_dev_t *d, const char *name)
+{
+	audio_ctrl_t *ctrl;
+
+	/* Verify argument */
+	ASSERT(d);
+
+	rw_enter(&d->d_ctrl_lock, RW_READER);
+	for (ctrl = list_head(&d->d_controls); ctrl;
+	    ctrl = list_next(&d->d_controls, ctrl)) {
+		if (strcmp(ctrl->ctrl_name, name) == 0) {
+			rw_exit(&d->d_ctrl_lock);
+			return (ctrl);
+		}
+	}
+	rw_exit(&d->d_ctrl_lock);
+	return (NULL);
+}
+
+/*
+ * Given a known control, get its attributes.
+ *
+ * The caller must supply a audio_ctrl_desc_t structure.  Also the
+ * values in the structure are ignored when making the call and filled
+ * in by this function. All data pointed to by elements of desc should
+ * be assumed read only.
+ *
+ * If an error occurs then a non-zero is returned.
+ *
+ */
+int
+auclnt_control_describe(audio_ctrl_t *ctrl, audio_ctrl_desc_t *desc)
+{
+	ASSERT(ctrl);
+	ASSERT(desc);
+
+	bcopy(&ctrl->ctrl_des, desc, sizeof (*desc));
+	return (0);
+}
+
+int
+auclnt_control_read(audio_ctrl_t *ctrl, uint64_t *value)
+{
+	return (audio_control_read(ctrl, value));
+}
+
+int
+auclnt_control_write(audio_ctrl_t *ctrl, uint64_t value)
+{
+	return (audio_control_write(ctrl, value));
+}
+
+void
+auclnt_warn(audio_client_t *c, const char *fmt, ...)
+{
+	va_list va;
+
+	va_start(va, fmt);
+	auimpl_dev_vwarn(c ? c->c_dev : NULL, fmt, va);
+	va_end(va);
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_client.h b/usr/src/uts/common/io/audio/impl/audio_client.h
new file mode 100644
index 0000000000..d38b1485a8
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_client.h
@@ -0,0 +1,260 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_AUDIO_CLIENT_H
+#define	_AUDIO_CLIENT_H
+
+/*
+ * Structure implementation in audio_impl.h
+ */
+#include <sys/audio/audio_common.h>
+
+typedef struct audio_client_ops {
+	const char	*aco_minor_prefix;
+	void		*(*aco_dev_init)(audio_dev_t *);
+	void		(*aco_dev_fini)(void *);
+	int		(*aco_open)(audio_client_t *, int);
+	void		(*aco_close)(audio_client_t *);
+	int		(*aco_read)(audio_client_t *, struct uio *, cred_t *);
+	int		(*aco_write)(audio_client_t *, struct uio *, cred_t *);
+	int		(*aco_ioctl)(audio_client_t *, int, intptr_t, int,
+			    cred_t *, int *);
+	int		(*aco_chpoll)(audio_client_t *, short, int, short *,
+			    struct pollhead **);
+	int		(*aco_mmap)(audio_client_t *, ...);
+	void		(*aco_input)(audio_client_t *);
+	void		(*aco_output)(audio_client_t *);
+	void		(*aco_notify)(audio_client_t *);
+	void		(*aco_drain)(audio_client_t *);
+} audio_client_ops_t;
+
+void *auclnt_get_private(audio_client_t *);
+void auclnt_set_private(audio_client_t *, void *);
+
+int auclnt_drain(audio_client_t *);
+int auclnt_start_drain(audio_client_t *);
+
+int auclnt_set_rate(audio_stream_t *, int);
+int auclnt_get_rate(audio_stream_t *);
+
+int auclnt_set_format(audio_stream_t *, int);
+int auclnt_get_format(audio_stream_t *);
+
+int auclnt_set_channels(audio_stream_t *, int);
+int auclnt_get_channels(audio_stream_t *);
+
+void auclnt_set_gain(audio_stream_t *, uint8_t);
+uint8_t auclnt_get_gain(audio_stream_t *);
+
+void auclnt_set_muted(audio_stream_t *, boolean_t);
+boolean_t auclnt_get_muted(audio_stream_t *);
+
+uint64_t auclnt_get_samples(audio_stream_t *);
+void auclnt_set_samples(audio_stream_t *, uint64_t);
+
+uint64_t auclnt_get_errors(audio_stream_t *);
+void auclnt_set_errors(audio_stream_t *, uint64_t);
+
+uint64_t auclnt_get_eof(audio_stream_t *);
+void auclnt_set_eof(audio_stream_t *, uint64_t);
+
+void auclnt_start(audio_stream_t *);
+void auclnt_stop(audio_stream_t *);
+
+void auclnt_set_paused(audio_stream_t *);
+void auclnt_clear_paused(audio_stream_t *);
+boolean_t auclnt_is_paused(audio_stream_t *);
+
+void auclnt_flush(audio_stream_t *);
+
+void auclnt_get_output_qlen(audio_client_t *, unsigned *, unsigned *);
+
+unsigned auclnt_get_fragsz(audio_stream_t *);
+unsigned auclnt_get_framesz(audio_stream_t *);
+unsigned auclnt_get_nfrags(audio_stream_t *);
+unsigned auclnt_get_nframes(audio_stream_t *);
+unsigned auclnt_get_count(audio_stream_t *);
+uint64_t auclnt_get_head(audio_stream_t *);
+uint64_t auclnt_get_tail(audio_stream_t *);
+unsigned auclnt_get_hidx(audio_stream_t *);
+unsigned auclnt_get_tidx(audio_stream_t *);
+
+audio_stream_t *auclnt_input_stream(audio_client_t *);
+audio_stream_t *auclnt_output_stream(audio_client_t *);
+
+int auclnt_get_oflag(audio_client_t *);
+
+int auclnt_open(audio_client_t *, unsigned, int);
+void auclnt_close(audio_client_t *);
+
+void auclnt_register_ops(minor_t, audio_client_ops_t *);
+
+minor_t	auclnt_get_minor(audio_client_t *);
+minor_t auclnt_get_original_minor(audio_client_t *);
+minor_t auclnt_get_minor_type(audio_client_t *);
+
+unsigned auclnt_produce(audio_stream_t *, unsigned);
+unsigned auclnt_produce_data(audio_stream_t *, caddr_t, unsigned);
+unsigned auclnt_consume(audio_stream_t *, unsigned);
+unsigned auclnt_consume_data(audio_stream_t *, caddr_t, unsigned);
+int auclnt_read(audio_client_t *, struct uio *);
+int auclnt_write(audio_client_t *, struct uio *);
+int auclnt_chpoll(audio_client_t *, short, int, short *, struct pollhead **);
+void auclnt_pollwakeup(audio_client_t *, short);
+
+/*
+ * Return the process id that performed the original open() of the client.
+ */
+pid_t auclnt_get_pid(audio_client_t *);
+
+/*
+ * Return the credentials of the process that opened the client.
+ */
+cred_t *auclnt_get_cred(audio_client_t *);
+
+/*
+ * Get an opaque handle the underlying device for an audio client.
+ */
+audio_dev_t *auclnt_get_dev(audio_client_t *);
+audio_dev_t *auclnt_hold_dev_by_index(int);
+void auclnt_release_dev(audio_dev_t *);
+void auclnt_notify_dev(audio_dev_t *);
+int auclnt_get_dev_index(audio_dev_t *);
+int auclnt_get_dev_number(audio_dev_t *);
+void auclnt_set_dev_number(audio_dev_t *, int);
+const char *auclnt_get_dev_name(audio_dev_t *);
+const char *auclnt_get_dev_driver(audio_dev_t *);
+dev_info_t *auclnt_get_dev_devinfo(audio_dev_t *);
+int auclnt_get_dev_instance(audio_dev_t *);
+const char *auclnt_get_dev_description(audio_dev_t *);
+const char *auclnt_get_dev_version(audio_dev_t *);
+const char *auclnt_get_dev_hw_info(audio_dev_t *, void **);
+unsigned auclnt_get_dev_capab(audio_dev_t *);
+#define	AUDIO_CLIENT_CAP_PLAY		(1U << 0)
+#define	AUDIO_CLIENT_CAP_RECORD		(1U << 1)
+#define	AUDIO_CLIENT_CAP_DUPLEX		(1U << 2)
+#define	AUDIO_CLIENT_CAP_SNDSTAT	(1U << 3)
+
+/*
+ * Walk all the open client structures for a named audio device.
+ * Clients can use this to find "peer" clients accessing the same
+ * audio device.  (This is useful for implementing special linkages,
+ * e.g. between /dev/audio and /dev/audioctl.)
+ */
+void auclnt_dev_walk_clients(audio_dev_t *,
+    int (*)(audio_client_t *, void *), void *);
+
+/*
+ * Audio control functions for use by clients.
+ */
+
+/*
+ * This will walk all controls registered to my device and callback
+ * to walker for each one with its audio_ctrl_desc_t..
+ *
+ * Note that walk_func may return values to continue (AUDIO_WALK_CONTINUE)
+ * or stop walk (AUDIO_WALK_STOP).
+ *
+ */
+void auclnt_walk_controls(audio_dev_t *,
+    int (*)(audio_ctrl_t *, void *), void *);
+
+/*
+ * This will search all controls attached to a clients
+ * audio device for a control with the desired name.
+ *
+ * On successful return a ctrl handle will be returned. On
+ * failure NULL is returned.
+ */
+audio_ctrl_t *auclnt_find_control(audio_dev_t *, const char *);
+
+/*
+ * Given a known control, get its attributes.
+ *
+ * The caller must supply a audio_ctrl_desc_t structure.  Also the
+ * values in the structure are ignored when making the call and filled
+ * in by this function.
+ *
+ * If an error occurs then a non-zero is returned.
+ */
+int auclnt_control_describe(audio_ctrl_t *, audio_ctrl_desc_t *);
+
+
+/*
+ * This is used to read the current value of a control.
+ * Note, this will cause a callback into the driver to get the value.
+ *
+ * On return zero is returned on success else errno is returned.
+ */
+int auclnt_control_read(audio_ctrl_t *, uint64_t *);
+
+/*
+ * This is used to write a value to a control.
+ * Note, this will cause a callback into the driver to write the value.
+ *
+ * On return zero is returned on success else errno is returned.
+ *
+ */
+int auclnt_control_write(audio_ctrl_t *, uint64_t);
+
+/*
+ * Walk all the audio devices on the system.  Useful for clients
+ * like sndstat, which may need to inquire about every audio device
+ * on the system.
+ */
+void auclnt_walk_devs(int (*walker)(audio_dev_t *, void *), void *);
+void auclnt_walk_devs_by_number(int (*walker)(audio_dev_t *, void *), void *);
+
+audio_client_t *auclnt_hold_by_devt(dev_t);
+void auclnt_release(audio_client_t *);
+
+/*
+ * Engine rlated accesses.  Note that normally clients don't need this level
+ * of information.
+ */
+void auclnt_dev_walk_engines(audio_dev_t *,
+    int (*)(audio_engine_t *, void *), void *);
+int auclnt_engine_get_format(audio_engine_t *);
+int auclnt_engine_get_rate(audio_engine_t *);
+int auclnt_engine_get_channels(audio_engine_t *);
+unsigned auclnt_engine_get_capab(audio_engine_t *);
+
+/*
+ * Retrieve minor-specific data for the instance.  This allows for
+ * personality modules to store persistent state data on a physical
+ * device (e.g. to store persistent settings.)  Synchronization of
+ * stored settings between personality modules is up to the
+ * personality modules themselves.
+ */
+void *auclnt_get_minor_data(audio_client_t *, minor_t);
+void *auclnt_get_dev_minor_data(audio_dev_t *, minor_t);
+
+/*
+ * Simpler warning message, alternative to cmn_err.
+ */
+void auclnt_warn(audio_client_t *, const char *fmt, ...);
+
+#endif	/* _AUDIO_CLIENT_H */
diff --git a/usr/src/uts/common/io/audio/impl/audio_ctrl.c b/usr/src/uts/common/io/audio/impl/audio_ctrl.c
new file mode 100644
index 0000000000..64e7ca5549
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_ctrl.c
@@ -0,0 +1,375 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/list.h>
+#include <sys/sysmacros.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+
+#include "audio_impl.h"
+
+/*
+ * Audio Control functions.
+ */
+
+/*
+ * Given a control structure - free all names
+ * strings allocated to it.
+ *
+ * ctrl             - The control who's names that will be free'd.
+ */
+static void
+audio_control_freenames(audio_ctrl_t *ctrl)
+{
+	int	indx;
+
+	if (ctrl->ctrl_name != NULL)
+		strfree((char *)ctrl->ctrl_name);
+	ctrl->ctrl_name = NULL;
+
+	for (indx = 0; indx < 64; indx++) {
+		if (ctrl->ctrl_enum[indx] != NULL) {
+			strfree((char *)ctrl->ctrl_enum[indx]);
+			ctrl->ctrl_enum[indx] = NULL;
+		}
+	}
+}
+
+/*
+ * This will allocate and register a control for my audio device.
+ *
+ * d                - The audio device the control will be attached to.
+ * desc             - Attributes about this new control
+ * read_fn          - Callback function in driver to read control
+ * write_fn         - Callback function in driver to write control.
+ * arg              - driver private context passed to read_fn/write_fn
+ *
+ * On success this will return a control structure else NULL.
+ *
+ * The value passed in for a control number in the audio_ctrl_desc_t
+ * has some special meaning. If it is less then AUDIO_CONTROL_EXTBASE
+ * then the control is assumed to be a known control. If it is
+ * AUDIO_CONTROL_EXTBASE then the framework will allocate a unique
+ * control number and replace it in the audio_ctrl_desc_t structure
+ * and this control is considered an extended driver private control.
+ * The number that is replaced in audio_ctrl_desc_t will be greater
+ * then AUDIO_CONTROL_EXTBASE.
+ *
+ */
+audio_ctrl_t *
+audio_dev_add_control(audio_dev_t *d, audio_ctrl_desc_t *desc,
+    audio_ctrl_rd_t read_fn, audio_ctrl_wr_t write_fn, void *arg)
+{
+	audio_ctrl_t *ctrl;
+	audio_ctrl_desc_t *new_desc;
+	char	scratch[16];
+	const char	*name;
+
+	/* Verify arguments */
+	ASSERT(d);
+	ASSERT(desc);
+
+	/* We cannot deal with unnamed controls */
+	if ((name = desc->acd_name) == NULL) {
+		return (NULL);
+	}
+
+	/*
+	 * If this was called with a control name that was already
+	 * added, then we do some special things. First we reuse the
+	 * control audio_ctrl_t and as far as outside users are
+	 * concerned the handle is reused. To users this looks like we
+	 * are changing the controls attributes. But what we really do
+	 * is free every thing allocated to the control and then
+	 * reinit everything.  That way the same code can get used for
+	 * both.
+	 *
+	 * We verify anything that could fail before we change the
+	 * control or commit to any changes. If there is something bad
+	 * return null to indicate an error but the original control
+	 * is still usable and untouched.
+	 */
+	ctrl = auclnt_find_control(d, name);
+
+	if (ctrl == NULL) {
+		/* Allocate a new control */
+		ctrl = kmem_zalloc(sizeof (*ctrl), KM_SLEEP);
+	} else {
+		/* Re-configure an existing control */
+		switch (desc->acd_type) {
+		case AUDIO_CTRL_TYPE_BOOLEAN:
+		case AUDIO_CTRL_TYPE_STEREO:
+		case AUDIO_CTRL_TYPE_MONO:
+		case AUDIO_CTRL_TYPE_METER:
+		case AUDIO_CTRL_TYPE_ENUM:
+			break;
+		default:
+			audio_dev_warn(d, "bad control type %d for %s "
+			    "not replaced", desc->acd_type, desc->acd_name);
+			return (NULL);
+		}
+
+		/*
+		 * By removing it from the list we prevent the need to lock
+		 * and check for locks on the control itself.
+		 * Also by doing this we can use the normal add code to do
+		 * what it normally does below.
+		 */
+		rw_enter(&d->d_ctrl_lock, RW_WRITER);
+		list_remove(&d->d_controls, ctrl);
+		rw_exit(&d->d_ctrl_lock);
+
+		audio_control_freenames(ctrl);
+		ctrl->ctrl_read_fn = NULL;
+		ctrl->ctrl_write_fn = NULL;
+		ctrl->ctrl_arg = NULL;
+		ctrl->ctrl_dev = NULL;
+		mutex_destroy(&ctrl->ctrl_lock);
+	}
+	new_desc = &ctrl->ctrl_des;
+
+	/* Fill in new control description */
+	new_desc->acd_type = desc->acd_type;
+	new_desc->acd_flags = desc->acd_flags;
+	new_desc->acd_maxvalue = desc->acd_maxvalue;
+	new_desc->acd_minvalue = desc->acd_minvalue;
+	new_desc->acd_name = strdup(name);
+
+	/* Process type of control special actions, if any */
+	switch (desc->acd_type) {
+	case AUDIO_CTRL_TYPE_BOOLEAN:
+	case AUDIO_CTRL_TYPE_STEREO:
+	case AUDIO_CTRL_TYPE_MONO:
+	case AUDIO_CTRL_TYPE_METER:
+		break;
+
+	case AUDIO_CTRL_TYPE_ENUM:
+		for (int bit = 0; bit < 64; bit++) {
+			if (((1U << bit) & desc->acd_maxvalue) == 0)
+				continue;
+			name = desc->acd_enum[bit];
+			if (name == NULL) {
+				(void) snprintf(scratch, sizeof (scratch),
+				    "bit%d", bit);
+				name = scratch;
+			}
+			new_desc->acd_enum[bit] = strdup(name);
+		}
+		break;
+	default:
+		audio_dev_warn(d, "bad control type %d for %s",
+		    desc->acd_type, desc->acd_name);
+		goto ctrl_fail;
+	}
+
+	ctrl->ctrl_dev = d;
+	if (new_desc->acd_flags & AUDIO_CTRL_FLAG_READABLE) {
+		ASSERT(read_fn);
+		ctrl->ctrl_read_fn = read_fn;
+		ctrl->ctrl_arg = arg;
+	}
+	if (new_desc->acd_flags & AUDIO_CTRL_FLAG_WRITEABLE) {
+		ASSERT(write_fn);
+		ctrl->ctrl_write_fn = write_fn;
+		ctrl->ctrl_arg = arg;
+	}
+
+	mutex_init(&ctrl->ctrl_lock, NULL, MUTEX_DRIVER, NULL);
+
+	rw_enter(&d->d_ctrl_lock, RW_WRITER);
+	list_insert_tail(&d->d_controls, ctrl);
+	rw_exit(&d->d_ctrl_lock);
+
+	return (ctrl);
+
+
+ctrl_fail:
+	if (ctrl) {
+		audio_control_freenames(ctrl);
+		kmem_free(ctrl, sizeof (*ctrl));
+	}
+	return (NULL);
+}
+
+/*
+ * This will remove a control from my audio device.
+ *
+ * ctrl             - The control will be removed.
+ */
+void
+audio_dev_del_control(audio_ctrl_t *ctrl)
+{
+	audio_dev_t *d;
+
+	/* Verify argument */
+	ASSERT(ctrl);
+	d = ctrl->ctrl_dev;
+	ASSERT(d);
+
+	rw_enter(&d->d_ctrl_lock, RW_WRITER);
+	list_remove(&d->d_controls, ctrl);
+	rw_exit(&d->d_ctrl_lock);
+
+	mutex_destroy(&ctrl->ctrl_lock);
+
+	audio_control_freenames(ctrl);
+	kmem_free(ctrl, sizeof (*ctrl));
+}
+
+static int
+auimpl_set_pcmvol(void *arg, uint64_t val)
+{
+	audio_dev_t	*d = arg;
+	list_t		*l = &d->d_clients;
+	audio_client_t	*c;
+
+	if (val > 100) {
+		return (EINVAL);
+	}
+	rw_enter(&d->d_clnt_lock, RW_WRITER);
+	d->d_pcmvol = val & 0xff;
+	for (c = list_head(l); c; c = list_next(l, c)) {
+		auimpl_set_gain_master(&c->c_ostream, (uint8_t)val);
+	}
+	rw_exit(&d->d_clnt_lock);
+	return (0);
+}
+
+static int
+auimpl_get_pcmvol(void *arg, uint64_t *val)
+{
+	audio_dev_t	*d = arg;
+
+	*val = d->d_pcmvol;
+	return (0);
+}
+
+int
+audio_dev_add_soft_volume(audio_dev_t *d)
+{
+	audio_ctrl_desc_t	desc;
+
+	bzero(&desc, sizeof (desc));
+	if (d->d_pcmvol_ctrl == NULL) {
+		desc.acd_name = AUDIO_CTRL_ID_VOLUME;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_PLAY |
+		    AUDIO_CTRL_FLAG_PCMVOL;
+		d->d_pcmvol_ctrl = audio_dev_add_control(d, &desc,
+		    auimpl_get_pcmvol, auimpl_set_pcmvol, d);
+		d->d_pcmvol = 75;
+	}
+	return (0);
+}
+
+/*
+ * This will notify clients of need to reread control
+ * values since they have changed.
+ *
+ * There will be a routine that allows a client to register
+ * a callback. All callbacks registered to this device should
+ * get called from this routine.
+ *
+ * d                - The device that needs updates.
+ */
+void
+audio_dev_update_controls(audio_dev_t *d)
+{
+	auclnt_notify_dev(d);
+}
+
+
+/*
+ * This is used to read the current value of a control.
+ * Note, this will cause a callback into the driver to get the value.
+ *
+ * ctrl        - should be the valid control being read.
+ * value       - is a pointer to the place that will contain the value read.
+ *
+ * On return zero is returned on success else errno is returned.
+ *
+ */
+int
+audio_control_read(audio_ctrl_t *ctrl, uint64_t *value)
+{
+	uint64_t my_value;
+	int ret;
+
+	/* Verify arguments */
+	ASSERT(ctrl);
+	ASSERT(value);
+	ASSERT(ctrl->ctrl_dev);
+
+	if (!(ctrl->ctrl_flags & AUDIO_CTRL_FLAG_READABLE)) {
+		return (ENXIO);
+	}
+
+	ASSERT(ctrl->ctrl_read_fn);
+
+	if ((ret = ctrl->ctrl_read_fn(ctrl->ctrl_arg, &my_value)) != 0) {
+		return (ret);
+	}
+
+	*value = my_value;
+
+	return (ret);
+}
+
+/*
+ * This is used to write a value to a control.
+ * Note, this will cause a callback into the driver to write the value.
+ *
+ * ctrl        - should be the valid control being written.
+ * value       - is value to set the control to.
+ *
+ * On return zero is returned on success else errno is returned.
+ *
+ */
+int
+audio_control_write(audio_ctrl_t *ctrl, uint64_t value)
+{
+	int		ret;
+	audio_dev_t	*d = ctrl->ctrl_dev;
+
+	/* Verify arguments */
+	ASSERT(ctrl);
+	ASSERT(d);
+
+	if (!(ctrl->ctrl_flags & AUDIO_CTRL_FLAG_WRITEABLE)) {
+		return (ENXIO);
+	}
+
+	ASSERT(ctrl->ctrl_write_fn);
+
+	ret = ctrl->ctrl_write_fn(ctrl->ctrl_arg, value);
+
+	if (ret == 0)
+		audio_dev_update_controls(d);
+
+	return (ret);
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_ddi.c b/usr/src/uts/common/io/audio/impl/audio_ddi.c
new file mode 100644
index 0000000000..56138bc28c
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_ddi.c
@@ -0,0 +1,465 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/sysmacros.h>
+#include <sys/stropts.h>
+#include <sys/strsun.h>
+#include <sys/list.h>
+#include <sys/mkdev.h>
+#include <sys/conf.h>
+#include <sys/note.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+
+#include "audio_impl.h"
+
+/*
+ * Audio DDI glue implementation.
+ */
+
+/*
+ * The audio module is itself a pseudo driver, as it contains the
+ * logic to support un-associated nodes.  (Think generic /dev/mixer
+ * and /dev/sndstat used by OSS.)
+ */
+static int
+audio_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	audio_dev_t	*adev;
+
+	/* pseudo devices don't need S/R support */
+	if ((cmd != DDI_ATTACH) || (dip == NULL)) {
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_get_instance(dip) != 0) {
+		return (DDI_FAILURE);
+	}
+
+	/* this can't fail */
+	adev = audio_dev_alloc(dip, 0);
+	adev->d_flags = DEV_SNDSTAT_CAP;
+	audio_dev_set_description(adev, "Audio Common Code");
+	audio_dev_set_version(adev, "pseudo");
+	ddi_set_driver_private(dip, adev);
+
+	/* look up our properties! */
+
+	if (audio_dev_register(adev) != NULL) {
+		audio_dev_free(adev);
+		return (DDI_FAILURE);
+	}
+
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audio_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	audio_dev_t	*adev;
+
+	/* pseudo devices don't need S/R support */
+	if (cmd != DDI_DETACH) {
+		return (DDI_FAILURE);
+	}
+
+	if (dip == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	if ((adev = ddi_get_driver_private(dip)) == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	if (audio_dev_unregister(adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	audio_dev_free(adev);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+audio_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd, void *arg, void **resp)
+{
+	dip = NULL;
+
+	if (getminor((dev_t)arg) & AUDIO_MN_CLONE_MASK) {
+		audio_client_t *c;
+		c = auclnt_hold_by_devt((dev_t)arg);
+		if (c != NULL) {
+			dip = c->c_dev->d_dip;
+			auclnt_release(c);
+		}
+	} else {
+		audio_dev_t	*adev;
+		if ((adev = auimpl_dev_hold_by_devt((dev_t)arg)) != NULL) {
+			dip = adev->d_dip;
+			auimpl_dev_release(adev);
+		}
+	}
+
+	if (dip == NULL) {
+		return (DDI_FAILURE);
+	}
+
+	switch (cmd) {
+	case DDI_INFO_DEVT2DEVINFO:
+		*resp = dip;
+		break;
+	case DDI_INFO_DEVT2INSTANCE:
+		*resp = (void *)(uintptr_t)ddi_get_instance(dip);
+		break;
+	default:
+		*resp = NULL;
+		return (DDI_FAILURE);
+	}
+	return (DDI_SUCCESS);
+}
+
+static int
+audio_open(dev_t *devp, int oflag, int otyp, cred_t *credp)
+{
+	int			rv;
+	audio_client_t		*c;
+
+	if (otyp == OTYP_BLK) {
+		return (ENXIO);
+	}
+
+	if ((c = auimpl_client_create(*devp)) == NULL) {
+		audio_dev_warn(NULL, "client create failed");
+		return (ENXIO);
+	}
+
+	c->c_omode = oflag;
+	c->c_pid = ddi_get_pid();
+	c->c_cred = credp;
+
+	/*
+	 * Call client/personality specific open handler.  Note that
+	 * we "insist" that there is an open.  The personality layer
+	 * will initialize/allocate any engines required.
+	 *
+	 * Hmm... do we need to pass in the cred?
+	 */
+	if ((rv = c->c_open(c, oflag)) != 0) {
+		audio_dev_warn(c->c_dev, "open failed (rv %d)", rv);
+		auimpl_client_destroy(c);
+		return (rv);
+	}
+
+	/* we do device cloning! */
+	*devp = makedevice(c->c_major, c->c_minor);
+
+	mutex_enter(&c->c_lock);
+	c->c_is_open = B_TRUE;
+	mutex_exit(&c->c_lock);
+
+	auclnt_notify_dev(c->c_dev);
+
+	return (0);
+}
+
+static int
+audio_close(dev_t dev, int flag, int otyp, cred_t *credp)
+{
+	audio_client_t	*c;
+	audio_dev_t	*d;
+
+	_NOTE(ARGUNUSED(flag));
+	_NOTE(ARGUNUSED(credp));
+	_NOTE(ARGUNUSED(otyp));
+
+	if ((c = auclnt_hold_by_devt(dev)) == NULL) {
+		audio_dev_warn(NULL, "close on bugs devt %x,%x",
+		    getmajor(dev), getminor(dev));
+		return (ENXIO);
+	}
+
+	mutex_enter(&c->c_lock);
+	c->c_is_open = B_FALSE;
+	mutex_exit(&c->c_lock);
+
+	/*
+	 * Pick up any data sitting around in input buffers.  This
+	 * avoids leaving record data stuck in queues.
+	 */
+	if (c->c_istream.s_engine != NULL)
+		audio_engine_produce(c->c_istream.s_engine);
+
+	/* get a local hold on the device */
+	d = c->c_dev;
+	auimpl_dev_hold(c->c_dev);
+
+	/*
+	 * NB: This must be done before c->c_close, since it calls
+	 * auclnt_close which will block waiting for the refence count
+	 * to drop to zero.
+	 */
+	auclnt_release(c);
+
+	/* Call personality specific close handler */
+	c->c_close(c);
+
+	auimpl_client_destroy(c);
+
+	/* notify peers that a change has occurred */
+	auclnt_notify_dev(d);
+
+	/* now we can drop the release we had on the device */
+	auimpl_dev_release(d);
+
+	return (0);
+}
+
+static int
+audio_write(dev_t dev, struct uio *uio, cred_t *credp)
+{
+	audio_client_t *c;
+	int rv;
+
+	if ((c = auclnt_hold_by_devt(dev)) == NULL) {
+		return (ENXIO);
+	}
+	rv = (c->c_write == NULL) ? ENXIO : c->c_write(c, uio, credp);
+	auclnt_release(c);
+
+	return (rv);
+}
+
+static int
+audio_read(dev_t dev, struct uio *uio, cred_t *credp)
+{
+	audio_client_t *c;
+	int rv;
+
+	if ((c = auclnt_hold_by_devt(dev)) == NULL) {
+		return (ENXIO);
+	}
+	rv = (c->c_read == NULL) ? ENXIO : c->c_read(c, uio, credp);
+	auclnt_release(c);
+
+	return (rv);
+}
+
+static int
+audio_ioctl(dev_t dev, int cmd, intptr_t arg, int mode, cred_t *credp,
+    int *rvalp)
+{
+	audio_client_t *c;
+	int rv;
+
+	if ((c = auclnt_hold_by_devt(dev)) == NULL) {
+		return (ENXIO);
+	}
+	rv = (c->c_ioctl == NULL) ? ENXIO : c->c_ioctl(c, cmd, arg, mode,
+	    credp, rvalp);
+	auclnt_release(c);
+
+	return (rv);
+}
+
+static int
+audio_chpoll(dev_t dev, short events, int anyyet, short *reventsp,
+    struct pollhead **phpp)
+{
+	audio_client_t *c;
+	int rv;
+
+	if ((c = auclnt_hold_by_devt(dev)) == NULL) {
+		return (ENXIO);
+	}
+	rv = (c->c_chpoll == NULL) ?
+	    ENXIO :
+	    c->c_chpoll(c, events, anyyet, reventsp, phpp);
+	auclnt_release(c);
+
+	return (rv);
+}
+
+struct cb_ops audio_cb_ops = {
+	audio_open,		/* open */
+	audio_close,		/* close */
+	nodev,		/* strategy */
+	nodev,		/* print */
+	nodev,		/* dump */
+	audio_read,		/* read */
+	audio_write,		/* write */
+	audio_ioctl,		/* ioctl */
+	nodev,		/* devmap */
+	nodev,		/* mmap */
+	nodev,		/* segmap */
+	audio_chpoll,	/* chpoll */
+	ddi_prop_op,	/* prop_op */
+	NULL,		/* str */
+	D_MP | D_64BIT,		/* flag */
+	CB_REV, 	/* rev */
+	nodev,		/* aread */
+	nodev,		/* awrite */
+};
+
+static struct dev_ops audio_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	audio_getinfo,		/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	audio_attach,		/* attach */
+	audio_detach,		/* detach */
+	nodev,			/* reset */
+	&audio_cb_ops,		/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+};
+
+static struct modldrv modldrv = {
+	&mod_driverops,
+	"Audio Framework",
+	&audio_dev_ops,
+};
+
+static struct modlinkage modlinkage = {
+	MODREV_1,			/* MODREV_1 indicated by manual */
+	&modldrv,
+	NULL
+};
+
+struct audio_ops_helper {
+	struct cb_ops		cbops;
+};
+
+void
+audio_init_ops(struct dev_ops *devops, const char *name)
+{
+	_NOTE(ARGUNUSED(name));
+
+	struct audio_ops_helper	*helper;
+
+	helper = kmem_zalloc(sizeof (*helper), KM_SLEEP);
+
+	helper->cbops.cb_open = audio_open;
+	helper->cbops.cb_close = audio_close;
+	helper->cbops.cb_strategy = nodev;
+	helper->cbops.cb_print = nodev;
+	helper->cbops.cb_dump = nodev;
+	helper->cbops.cb_read = audio_read;
+	helper->cbops.cb_write = audio_write;
+	helper->cbops.cb_ioctl = audio_ioctl;
+	helper->cbops.cb_devmap = nodev;
+	helper->cbops.cb_mmap = nodev;
+	helper->cbops.cb_segmap = nodev;
+	helper->cbops.cb_chpoll = audio_chpoll;
+	helper->cbops.cb_prop_op = ddi_prop_op;
+	helper->cbops.cb_str = NULL;
+	helper->cbops.cb_flag = D_MP | D_64BIT;
+	helper->cbops.cb_rev = CB_REV;
+	helper->cbops.cb_aread = nodev;
+	helper->cbops.cb_awrite = nodev;
+
+	devops->devo_cb_ops = &helper->cbops;
+	devops->devo_getinfo = audio_getinfo;
+}
+
+void
+audio_fini_ops(struct dev_ops *devops)
+{
+	kmem_free(devops->devo_cb_ops, sizeof (struct audio_ops_helper));
+	devops->devo_cb_ops = NULL;
+	devops->devo_getinfo = NULL;
+}
+
+void
+auimpl_dev_vwarn(audio_dev_t *dev, const char *fmt, va_list va)
+{
+	char	buf[256];
+
+	if (dev != NULL) {
+		(void) snprintf(buf, sizeof (buf), "%s#%d: %s",
+		    ddi_driver_name(dev->d_dip), ddi_get_instance(dev->d_dip),
+		    fmt);
+	} else {
+		(void) snprintf(buf, sizeof (buf), "audio: %s", fmt);
+	}
+
+	vcmn_err(CE_WARN, buf, va);
+}
+
+
+void
+audio_dev_warn(audio_dev_t *dev, const char *fmt, ...)
+{
+	va_list	va;
+
+	va_start(va, fmt);
+	auimpl_dev_vwarn(dev, fmt, va);
+	va_end(va);
+}
+
+/*
+ * _init, _info, and _fini DDI glue.
+ */
+int
+_init(void)
+{
+	int	rv;
+
+	auimpl_client_init();
+	auimpl_dev_init();
+	auimpl_sun_init();
+	auimpl_oss_init();
+
+	if ((rv = mod_install(&modlinkage)) != 0) {
+		auimpl_dev_fini();
+		auimpl_client_fini();
+	}
+	return (rv);
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&modlinkage, modinfop));
+}
+
+int
+_fini(void)
+{
+	int rv;
+
+	if ((rv = mod_remove(&modlinkage)) != 0)
+		return (rv);
+
+	auimpl_dev_fini();
+	auimpl_client_fini();
+
+	return (rv);
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_engine.c b/usr/src/uts/common/io/audio/impl/audio_engine.c
new file mode 100644
index 0000000000..0d48587991
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_engine.c
@@ -0,0 +1,1197 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/list.h>
+#include <sys/sysmacros.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/callb.h>
+#include <sys/kstat.h>
+#include <sys/note.h>
+
+#include "audio_impl.h"
+
+/*
+ * Audio Engine functions.
+ */
+
+audio_dev_t *
+audio_dev_alloc(dev_info_t *dip, int instance)
+{
+	audio_dev_t *d;
+
+	/*
+	 * For a card with multiple independent audio ports on it, we
+	 * allow the driver to provide a different instance numbering
+	 * scheme than the standard DDI instance number.  (This is
+	 * sort of like the PPA numbering scheme used by NIC drivers
+	 * -- by default PPA == instance, but sometimes we need more
+	 * flexibility.)
+	 */
+	if (instance == 0) {
+		instance = ddi_get_instance(dip);
+	}
+	/* generally this shouldn't occur */
+	if (instance > AUDIO_MN_INST_MASK) {
+		audio_dev_warn(NULL, "bad instance number for %s (%d)",
+		    ddi_driver_name(dip), instance);
+		return (NULL);
+	}
+
+	if ((d = kmem_zalloc(sizeof (*d), KM_NOSLEEP)) == NULL) {
+		audio_dev_warn(NULL, "unable to allocate audio device struct");
+		return (NULL);
+	}
+	d->d_dip = dip;
+	d->d_number = -1;
+	d->d_major = ddi_driver_major(dip);
+	d->d_instance = instance;
+	d->d_pcmvol = 100;
+	mutex_init(&d->d_lock, NULL, MUTEX_DRIVER, NULL);
+	cv_init(&d->d_cv, NULL, CV_DRIVER, NULL);
+	rw_init(&d->d_ctrl_lock, NULL, RW_DRIVER, NULL);
+	rw_init(&d->d_clnt_lock, NULL, RW_DRIVER, NULL);
+	list_create(&d->d_clients, sizeof (struct audio_client),
+	    offsetof(struct audio_client, c_dev_linkage));
+	list_create(&d->d_engines, sizeof (struct audio_engine),
+	    offsetof(struct audio_engine, e_dev_linkage));
+	list_create(&d->d_controls, sizeof (struct audio_ctrl),
+	    offsetof(struct audio_ctrl, ctrl_linkage));
+	list_create(&d->d_hwinfo, sizeof (struct audio_infostr),
+	    offsetof(struct audio_infostr, i_linkage));
+	(void) snprintf(d->d_name, sizeof (d->d_name), "%s#%d",
+	    ddi_driver_name(dip), instance);
+
+	return (d);
+}
+
+void
+audio_dev_free(audio_dev_t *d)
+{
+	struct audio_infostr *isp;
+	while ((isp = list_remove_head(&d->d_hwinfo)) != NULL) {
+		kmem_free(isp, sizeof (*isp));
+	}
+	if (d->d_pcmvol_ctrl != NULL) {
+		audio_dev_del_control(d->d_pcmvol_ctrl);
+	}
+	list_destroy(&d->d_hwinfo);
+	list_destroy(&d->d_engines);
+	list_destroy(&d->d_controls);
+	list_destroy(&d->d_clients);
+	rw_destroy(&d->d_ctrl_lock);
+	mutex_destroy(&d->d_lock);
+	cv_destroy(&d->d_cv);
+	rw_destroy(&d->d_clnt_lock);
+	kmem_free(d, sizeof (*d));
+}
+
+void
+audio_dev_set_description(audio_dev_t *d, const char *desc)
+{
+	(void) strlcpy(d->d_desc, desc, sizeof (d->d_desc));
+}
+
+void
+audio_dev_set_version(audio_dev_t *d, const char *vers)
+{
+	(void) strlcpy(d->d_vers, vers, sizeof (d->d_vers));
+}
+
+void
+audio_dev_add_info(audio_dev_t *d, const char *info)
+{
+	struct audio_infostr *isp;
+
+	/* failure to add information structure is not critical */
+	isp = kmem_zalloc(sizeof (*isp), KM_NOSLEEP);
+	if (isp == NULL) {
+		audio_dev_warn(d, "unable to allocate information structure");
+	} else {
+		(void) snprintf(isp->i_line, sizeof (isp->i_line), info);
+		list_insert_tail(&d->d_hwinfo, isp);
+	}
+}
+
+void
+audio_engine_consume(audio_engine_t *e)
+{
+	mutex_enter(&e->e_lock);
+	e->e_tail = ENG_COUNT(e);
+	if (e->e_tail > e->e_head) {
+		/* want more data than we have, not much we can do */
+		e->e_errors++;
+	}
+	auimpl_output_callback(e);
+	mutex_exit(&e->e_lock);
+}
+
+void
+audio_engine_produce(audio_engine_t *e)
+{
+	mutex_enter(&e->e_lock);
+	e->e_head = ENG_COUNT(e);
+	if ((e->e_head - e->e_tail) > e->e_nframes) {
+		/* no room for engine data, not much we can do */
+		e->e_errors++;
+	}
+	auimpl_input_callback(e);
+	mutex_exit(&e->e_lock);
+}
+
+void
+audio_engine_reset(audio_engine_t *e)
+{
+	char	*buf;
+	char	*ptr;
+	int	nfr;
+	int	tail;
+
+
+	if ((e->e_flags & (ENGINE_INPUT | ENGINE_OUTPUT)) == 0) {
+		/* engine not open, nothing to do */
+		return;
+	}
+
+	buf = kmem_alloc(e->e_nbytes, KM_SLEEP);
+	ptr = buf;
+
+	mutex_enter(&e->e_lock);
+
+	tail = e->e_tidx;
+	nfr = min(e->e_head - e->e_tail, e->e_nframes);
+	while (nfr) {
+		int	cnt;
+		int	nbytes;
+
+		cnt = min((e->e_nframes - tail), nfr);
+		nbytes = cnt * e->e_framesz;
+
+		bcopy(e->e_data + (tail * e->e_framesz), ptr, nbytes);
+		ptr += nbytes;
+		tail += cnt;
+		if (tail >= e->e_framesz) {
+			tail -= e->e_framesz;
+		}
+		nfr -= cnt;
+	}
+
+	nfr = min(e->e_head - e->e_tail, e->e_nframes);
+	if (e->e_flags & ENGINE_INPUT) {
+		/* record */
+		e->e_hidx = 0;
+		e->e_tidx = (e->e_nframes - nfr) % e->e_nframes;
+	} else {
+		/* play */
+		e->e_hidx = nfr % e->e_nframes;
+		e->e_tidx = 0;
+	}
+
+	/* relocate from scratch area to destination */
+	bcopy(buf, e->e_data + (e->e_tidx * e->e_framesz), nfr * e->e_framesz);
+	mutex_exit(&e->e_lock);
+
+	kmem_free(buf, e->e_nbytes);
+}
+
+audio_engine_t *
+audio_engine_alloc(audio_engine_ops_t *ops, unsigned flags)
+{
+	int i;
+	audio_engine_t *e;
+
+	if (ops->audio_engine_version != AUDIO_ENGINE_VERSION) {
+		audio_dev_warn(NULL, "audio engine version mismatch: %d != %d",
+		    ops->audio_engine_version, AUDIO_ENGINE_VERSION);
+		return (NULL);
+	}
+
+	/* NB: The ops vector must be held in persistent storage! */
+	e = kmem_zalloc(sizeof (audio_engine_t), KM_NOSLEEP);
+	if (e == NULL) {
+		audio_dev_warn(NULL, "unable to allocate engine struct");
+		return (NULL);
+	}
+	e->e_ops = *ops;
+	mutex_init(&e->e_lock, NULL, MUTEX_DRIVER, NULL);
+	list_create(&e->e_streams, sizeof (struct audio_stream),
+	    offsetof(struct audio_stream, s_eng_linkage));
+
+	for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
+		e->e_chbufs[i] = kmem_zalloc(sizeof (int32_t) * AUDIO_CHBUFS,
+		    KM_NOSLEEP);
+		if (e->e_chbufs[i] == NULL) {
+			audio_dev_warn(NULL, "unable to allocate channel buf");
+			audio_engine_free(e);
+			return (NULL);
+		}
+	}
+
+	e->e_flags = flags & ENGINE_DRIVER_FLAGS;
+	return (e);
+}
+
+void
+audio_engine_free(audio_engine_t *e)
+{
+	int i;
+
+	for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
+		if (e->e_chbufs[i] != NULL) {
+			kmem_free(e->e_chbufs[i],
+			    sizeof (int32_t) * AUDIO_CHBUFS);
+		}
+	}
+	list_destroy(&e->e_streams);
+	mutex_destroy(&e->e_lock);
+	kmem_free(e, sizeof (*e));
+}
+
+static list_t auimpl_devs_by_index;
+static list_t auimpl_devs_by_number;
+static krwlock_t auimpl_dev_lock;
+
+/*
+ * Not for public consumption: Private interfaces.
+ */
+void
+auimpl_dev_hold(audio_dev_t *d)
+{
+	/* bump the reference count */
+	mutex_enter(&d->d_lock);
+	d->d_refcnt++;
+	mutex_exit(&d->d_lock);
+}
+
+audio_dev_t *
+auimpl_dev_hold_by_devt(dev_t dev)
+{
+	audio_dev_t *d;
+	major_t major;
+	int instance;
+	list_t *l = &auimpl_devs_by_index;
+
+	major = getmajor(dev);
+	instance = (getminor(dev) >> AUDIO_MN_INST_SHIFT) & AUDIO_MN_INST_MASK;
+
+	rw_enter(&auimpl_dev_lock, RW_READER);
+
+	for (d = list_head(l); d; d = list_next(l, d)) {
+		if ((d->d_major == major) && (d->d_instance == instance)) {
+			auimpl_dev_hold(d);
+			break;
+		}
+	}
+
+	rw_exit(&auimpl_dev_lock);
+	return (d);
+}
+
+audio_dev_t *
+auimpl_dev_hold_by_index(int index)
+{
+	audio_dev_t *d;
+	list_t *l = &auimpl_devs_by_index;
+
+	rw_enter(&auimpl_dev_lock, RW_READER);
+
+	for (d = list_head(l); d; d = list_next(l, d)) {
+		if (d->d_index == index) {
+			auimpl_dev_hold(d);
+			break;
+		}
+	}
+
+	rw_exit(&auimpl_dev_lock);
+	return (d);
+}
+
+void
+auimpl_dev_release(audio_dev_t *d)
+{
+	mutex_enter(&d->d_lock);
+	d->d_refcnt--;
+	mutex_exit(&d->d_lock);
+}
+
+int
+auimpl_choose_format(int fmts)
+{
+	/*
+	 * Choose the very best format we can.  We choose 24 bit in
+	 * preference to 32 bit because we mix in 24 bit.  We do that
+	 * to allow overflows to fit within 32-bits.  (Very few humans
+	 * can tell a difference between 24 and 32 bit audio anyway.)
+	 */
+	if (fmts & AUDIO_FORMAT_S24_NE)
+		return (AUDIO_FORMAT_S24_NE);
+
+	if (fmts & AUDIO_FORMAT_S32_NE)
+		return (AUDIO_FORMAT_S32_NE);
+
+	if (fmts & AUDIO_FORMAT_S24_OE)
+		return (AUDIO_FORMAT_S24_OE);
+
+	if (fmts & AUDIO_FORMAT_S32_OE)
+		return (AUDIO_FORMAT_S32_OE);
+
+	if (fmts & AUDIO_FORMAT_S16_NE)
+		return (AUDIO_FORMAT_S16_NE);
+
+	if (fmts & AUDIO_FORMAT_S16_OE)
+		return (AUDIO_FORMAT_S16_OE);
+
+	if (fmts & AUDIO_FORMAT_AC3)
+		return (AUDIO_FORMAT_AC3);
+
+	return (AUDIO_FORMAT_NONE);
+}
+
+int
+auimpl_engine_open(audio_dev_t *d, int fmts, int flags, audio_stream_t *sp)
+{
+	audio_engine_t	*e = NULL;
+	list_t		*list;
+	unsigned	caps;
+	int		priority = 0;
+	int		rv = ENODEV;
+	int		sampsz;
+	int		i;
+
+	/*
+	 * Engine selection:
+	 *
+	 * We try hard to avoid consuming an engine that can be used
+	 * for another purpose.
+	 *
+	 */
+
+	/*
+	 * Which direction are we opening.  (We must open exactly
+	 * one direction, otherwise the open is meaningless.)
+	 */
+	if (flags & ENGINE_OUTPUT)
+		caps = ENGINE_OUTPUT_CAP;
+	else if (flags & ENGINE_INPUT)
+		caps = ENGINE_INPUT_CAP;
+	else
+		return (EINVAL);
+
+	list = &d->d_engines;
+
+	mutex_enter(&d->d_lock);
+
+	/*
+	 * First we want to know if we already have "default" input
+	 * and output engines.
+	 */
+
+again:
+
+	for (audio_engine_t *t = list_head(list); t; t = list_next(list, t)) {
+		int	mypri;
+
+		/* make sure the engine can do what we want it to */
+		mutex_enter(&t->e_lock);
+		if ((((t->e_flags & caps) & caps) == 0) ||
+		    ((ENG_FORMAT(t) & fmts) == 0)) {
+			mutex_exit(&t->e_lock);
+			continue;
+		}
+
+		/* if engine is in exclusive use, can't do it */
+		if (t->e_flags & ENGINE_EXCLUSIVE) {
+			mutex_exit(&t->e_lock);
+			rv = EBUSY;
+			continue;
+		}
+
+		/* if engine is in incompatible use, can't do it */
+		if (((flags & ENGINE_INPUT) && (t->e_flags & ENGINE_OUTPUT)) ||
+		    ((flags & ENGINE_OUTPUT) && (t->e_flags & ENGINE_INPUT))) {
+			mutex_exit(&t->e_lock);
+			rv = EBUSY;
+			continue;
+		}
+
+		/*
+		 * In order to support as many different possible
+		 * output streams (e.g. AC3 passthru or AC3 decode),
+		 * or multiple exclusive outputs, we treat audio
+		 * engines as *precious*.
+		 *
+		 * This means that we will try hard to reuse an
+		 * existing allocated engine.  This may not be the
+		 * optimal performance configuration (especially if we
+		 * wanted to avoid rate conversion, for example), but
+		 * it should have fewer cases where the configuration
+		 * results in denying service to any client.
+		 */
+
+		rv = 0;
+		mypri = 2000;
+
+		/* try not to pick on idle engines */
+		if (list_is_empty(&t->e_streams)) {
+			mypri -= 1000;
+		}
+
+		/* try not to pick on duplex engines first */
+		if ((t->e_flags & ENGINE_CAPS) != caps) {
+			mypri -= 100;
+		}
+
+		/* try not to pick on engines that can do other formats */
+		if (t->e_format & ~fmts) {
+			mypri -= 10;
+		}
+
+		if (mypri > priority) {
+			if (e != NULL) {
+				mutex_exit(&e->e_lock);
+			}
+			e = t;
+			priority = mypri;
+		} else {
+			mutex_exit(&t->e_lock);
+		}
+	}
+
+	if ((rv == EBUSY) && ((flags & ENGINE_NDELAY) == 0)) {
+		ASSERT(e == NULL);
+		if (cv_wait_sig(&d->d_cv, &d->d_lock) == 0) {
+			mutex_exit(&d->d_lock);
+			return (EINTR);
+		}
+		goto again;
+	}
+
+	if (rv != 0) {
+		ASSERT(e == NULL);
+		mutex_exit(&d->d_lock);
+		return (rv);
+	}
+
+	ASSERT(e != NULL);
+	ASSERT(mutex_owned(&e->e_lock));
+
+	/*
+	 * If the engine is already open, there is no need for further
+	 * work.  The first open will be relatively expensive, but
+	 * subsequent opens should be as cheap as possible.
+	 */
+	if (!list_is_empty(&e->e_streams)) {
+		rv = 0;
+		goto ok;
+	}
+
+	e->e_format = ENG_FORMAT(e);
+	e->e_nchan = ENG_CHANNELS(e);
+	e->e_rate = ENG_RATE(e);
+
+	/* Find out the "best" sample format supported by the device */
+	switch (e->e_format) {
+	case AUDIO_FORMAT_S24_NE:
+		e->e_export = auimpl_export_24ne;
+		e->e_import = auimpl_import_24ne;
+		sampsz = 4;
+		break;
+	case AUDIO_FORMAT_S32_NE:
+		e->e_export = auimpl_export_32ne;
+		e->e_import = auimpl_import_32ne;
+		sampsz = 4;
+		break;
+	case AUDIO_FORMAT_S24_OE:
+		e->e_export = auimpl_export_24oe;
+		e->e_import = auimpl_import_24oe;
+		sampsz = 4;
+		break;
+	case AUDIO_FORMAT_S32_OE:
+		e->e_export = auimpl_export_32oe;
+		e->e_import = auimpl_import_32oe;
+		sampsz = 4;
+		break;
+	case AUDIO_FORMAT_S16_NE:
+		e->e_export = auimpl_export_16ne;
+		e->e_import = auimpl_import_16ne;
+		sampsz = 2;
+		break;
+	case AUDIO_FORMAT_S16_OE:
+		e->e_export = auimpl_export_16oe;
+		e->e_import = auimpl_import_16oe;
+		sampsz = 2;
+		break;
+	case AUDIO_FORMAT_AC3:
+		e->e_export = auimpl_export_24ne;
+		e->e_import = auimpl_import_24ne;
+		flags |= ENGINE_EXCLUSIVE;
+		sampsz = 2;
+		break;
+	default:
+		audio_dev_warn(d, "bad format");
+		rv = ENOTSUP;
+		goto done;
+	}
+
+	/* sanity test a few values */
+	if ((e->e_nchan < 0) || (e->e_nchan > AUDIO_MAX_CHANNELS) ||
+	    (e->e_rate < 5000) || (e->e_rate > 192000)) {
+		audio_dev_warn(d, "bad engine channels or rate");
+		rv = EINVAL;
+		goto done;
+	}
+
+	rv = ENG_OPEN(e, &e->e_fragfr, &e->e_nfrags, &e->e_data);
+	if (rv != 0) {
+		audio_dev_warn(d, "unable to open engine");
+		goto done;
+	}
+	if ((e->e_fragfr < 1) || (e->e_data == NULL)) {
+		audio_dev_warn(d, "improper engine configuration");
+		rv = EINVAL;
+		goto done;
+	}
+
+	if ((e->e_fragfr > AUDIO_CHBUFS) || (e->e_nfrags < 2)) {
+		rv = EINVAL;
+		audio_dev_warn(d, "invalid fragment configuration");
+		goto done;
+	}
+
+	e->e_framesz = e->e_nchan * sampsz;
+	e->e_fragbytes = e->e_fragfr * e->e_framesz;
+	e->e_nframes = e->e_nfrags * e->e_fragfr;
+	e->e_intrs = e->e_rate / e->e_fragfr;
+	e->e_nbytes = e->e_nframes * e->e_framesz;
+	e->e_head = 0;
+	e->e_tail = 0;
+	e->e_hidx = 0;
+	e->e_tidx = 0;
+	e->e_limiter_state = 0x10000;
+	bzero(e->e_data, e->e_nbytes);
+
+	for (i = 0; i < e->e_nchan; i++) {
+		if (e->e_ops.audio_engine_chinfo == NULL) {
+			e->e_choffs[i] = i;
+			e->e_chincr[i] = e->e_nchan;
+		} else {
+			ENG_CHINFO(e, i, &e->e_choffs[i], &e->e_chincr[i]);
+		}
+	}
+
+	e->e_flags |= (ENGINE_OPEN | (flags & (ENGINE_OUTPUT | ENGINE_INPUT)));
+
+	/*
+	 * Start the engine up now.
+	 *
+	 * AC3: Note that this will need to be modified for AC3, since
+	 * for AC3 we can't start the device until we actually have
+	 * some data for it from the application.  Probably the best
+	 * way to do this would be to add a flag, ENGINE_DEFERRED or
+	 * somesuch.
+	 */
+	if (e->e_ops.audio_engine_start != NULL) {
+		rv = ENG_START(e);
+		if (rv != 0) {
+			ENG_CLOSE(e);
+			goto done;
+		}
+	}
+
+ok:
+	sp->s_phys_parms->p_rate = e->e_rate;
+	sp->s_phys_parms->p_nchan = e->e_nchan;
+
+	list_insert_tail(&e->e_streams, sp);
+	sp->s_engine = e;
+
+done:
+	mutex_exit(&e->e_lock);
+	mutex_exit(&d->d_lock);
+	return (rv);
+}
+
+void
+auimpl_engine_close(audio_stream_t *sp)
+{
+	audio_engine_t	*e = sp->s_engine;
+	audio_dev_t	*d;
+
+	if (e == NULL)
+		return;
+
+	d = e->e_dev;
+
+	mutex_enter(&d->d_lock);
+	mutex_enter(&e->e_lock);
+	sp->s_engine = NULL;
+	list_remove(&e->e_streams, sp);
+	if (list_is_empty(&e->e_streams)) {
+		/* if last client holding engine open, close it all down */
+		if (e->e_ops.audio_engine_stop != NULL)
+			ENG_STOP(e);
+		e->e_flags &= ENGINE_DRIVER_FLAGS;
+		ENG_CLOSE(e);
+	}
+	mutex_exit(&e->e_lock);
+
+	cv_broadcast(&d->d_cv);
+	mutex_exit(&d->d_lock);
+}
+
+int
+audio_dev_register(audio_dev_t *d)
+{
+	list_t *l;
+	audio_dev_t *srch;
+	int start;
+
+	/*
+	 * Make sure we don't automatically unload.  This prevents
+	 * loss of hardware settings when no audio clients are
+	 * running.
+	 */
+	(void) ddi_prop_update_int(DDI_DEV_T_NONE, d->d_dip,
+	    DDI_NO_AUTODETACH, 1);
+
+	/*
+	 * This does an in-order insertion, finding the first available
+	 * free index.  "Special" devices (ones without any actual engines)
+	 * are all numbered 0.  There should only be one of them anyway.
+	 * All others start at one.
+	 */
+	if (d->d_flags & DEV_SNDSTAT_CAP) {
+		start = 0;
+	} else {
+		start = 1;
+	}
+	d->d_index = start;
+	rw_enter(&auimpl_dev_lock, RW_WRITER);
+	l = &auimpl_devs_by_index;
+	for (srch = list_head(l); srch; srch = list_next(l, srch)) {
+		/* skip over special nodes */
+		if (srch->d_index < start)
+			continue;
+		if (srch->d_index > d->d_index) {
+			/* found a free spot! */
+			break;
+		}
+		d->d_index++;
+	}
+	/*
+	 * NB: If srch is NULL, then list_insert_before puts
+	 * it on the tail of the list.  So if we didn't find a
+	 * hole, then that's where we want it.
+	 */
+	list_insert_before(l, srch, d);
+
+	/* insert in order by number */
+	l = &auimpl_devs_by_number;
+	for (srch = list_head(l); srch; srch = list_next(l, srch)) {
+		if (srch->d_number >= d->d_number) {
+			break;
+		}
+	}
+	list_insert_before(l, srch, d);
+
+	rw_exit(&auimpl_dev_lock);
+
+	if (auimpl_create_minors(d) != 0) {
+		rw_enter(&auimpl_dev_lock, RW_WRITER);
+		auimpl_remove_minors(d);
+		list_remove(&auimpl_devs_by_index, d);
+		list_remove(&auimpl_devs_by_number, d);
+		rw_exit(&auimpl_dev_lock);
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+}
+
+int
+audio_dev_unregister(audio_dev_t *d)
+{
+	rw_enter(&auimpl_dev_lock, RW_WRITER);
+
+	mutex_enter(&d->d_lock);
+	/* if we are still in use, we can't unregister */
+	if (d->d_refcnt) {
+		mutex_exit(&d->d_lock);
+		rw_exit(&auimpl_dev_lock);
+		return (DDI_FAILURE);
+	}
+	auimpl_remove_minors(d);
+	list_remove(&auimpl_devs_by_index, d);
+	list_remove(&auimpl_devs_by_number, d);
+	mutex_exit(&d->d_lock);
+
+	rw_exit(&auimpl_dev_lock);
+
+	return (DDI_SUCCESS);
+}
+
+static int
+auimpl_engine_ksupdate(kstat_t *ksp, int rw)
+{
+	audio_engine_t *e = ksp->ks_private;
+	struct audio_stats *st = &e->e_stats;
+
+	if (rw == KSTAT_WRITE) {
+		return (EACCES);
+	}
+
+	mutex_enter(&e->e_lock);
+	st->st_head.value.ui64 = e->e_head;
+	st->st_tail.value.ui64 = e->e_tail;
+	st->st_flags.value.ui32 = e->e_flags;
+	st->st_fragfr.value.ui32 = e->e_fragfr;
+	st->st_nfrags.value.ui32 = e->e_nfrags;
+	st->st_framesz.value.ui32 = e->e_framesz;
+	st->st_nbytes.value.ui32 = e->e_nbytes;
+	st->st_hidx.value.ui32 = e->e_hidx;
+	st->st_tidx.value.ui32 = e->e_tidx;
+	st->st_format.value.ui32 = e->e_format;
+	st->st_nchan.value.ui32 = e->e_nchan;
+	st->st_rate.value.ui32 = e->e_rate;
+	st->st_intrs.value.ui32 = e->e_intrs;
+	st->st_errors.value.ui32 = e->e_errors;
+	st->st_suspended.value.ui32 = e->e_suspended;
+	mutex_exit(&e->e_lock);
+
+	return (0);
+}
+
+static void
+auimpl_engine_ksinit(audio_dev_t *d, audio_engine_t *e)
+{
+	char			name[32];
+	struct audio_stats	*st;
+
+	(void) snprintf(name, sizeof (name), "engine_%d", e->e_num);
+
+	e->e_ksp = kstat_create(ddi_driver_name(d->d_dip), d->d_instance,
+	    name, "misc", KSTAT_TYPE_NAMED,
+	    sizeof (struct audio_stats) / sizeof (kstat_named_t), 0);
+
+	if (e->e_ksp == NULL) {
+		audio_dev_warn(d, "unable to initialize kstats");
+		return;
+	}
+
+	st = &e->e_stats;
+	e->e_ksp->ks_data = st;
+	e->e_ksp->ks_private = e;
+	e->e_ksp->ks_lock = NULL;
+	e->e_ksp->ks_update = auimpl_engine_ksupdate;
+	kstat_named_init(&st->st_head, "head", KSTAT_DATA_UINT64);
+	kstat_named_init(&st->st_tail, "tail", KSTAT_DATA_UINT64);
+	kstat_named_init(&st->st_flags, "flags", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_fragfr, "fragfr", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_nfrags, "nfrags", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_framesz, "framesz", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_nbytes, "nbytes", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_hidx, "hidx", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_tidx, "tidx", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_format, "format", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_nchan, "channels", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_rate, "rate", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_intrs, "intrhz", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_errors, "errors", KSTAT_DATA_UINT32);
+	kstat_named_init(&st->st_suspended, "suspended", KSTAT_DATA_UINT32);
+	kstat_install(e->e_ksp);
+}
+
+void
+audio_dev_add_engine(audio_dev_t *d, audio_engine_t *e)
+{
+	e->e_num = d->d_engno++;
+
+	mutex_enter(&d->d_lock);
+
+	auimpl_engine_ksinit(d, e);
+
+	/* check for duplex */
+	if ((e->e_flags & ENGINE_OUTPUT_CAP) && (d->d_flags & DEV_INPUT_CAP)) {
+		d->d_flags |= DEV_DUPLEX_CAP;
+	}
+	if ((e->e_flags & ENGINE_INPUT_CAP) && (d->d_flags & DEV_OUTPUT_CAP)) {
+		d->d_flags |= DEV_DUPLEX_CAP;
+	}
+	/* add in the direction caps -- must be done after duplex above */
+	if (e->e_flags & ENGINE_OUTPUT_CAP) {
+		d->d_flags |= DEV_OUTPUT_CAP;
+	}
+	if (e->e_flags & ENGINE_INPUT_CAP) {
+		d->d_flags |= DEV_INPUT_CAP;
+	}
+
+	list_insert_tail(&d->d_engines, e);
+	e->e_dev = d;
+	mutex_exit(&d->d_lock);
+}
+
+void
+audio_dev_remove_engine(audio_dev_t *d, audio_engine_t *e)
+{
+	mutex_enter(&d->d_lock);
+	list_remove(&d->d_engines, e);
+	e->e_dev = NULL;
+	if (e->e_ksp)
+		kstat_delete(e->e_ksp);
+	e->e_ksp = NULL;
+	mutex_exit(&d->d_lock);
+}
+
+/*
+ * Change the number.
+ */
+void
+auclnt_set_dev_number(audio_dev_t *d, int num)
+{
+	list_t		*l = &auimpl_devs_by_number;
+	audio_dev_t	*srch;
+
+	/* reorder our list */
+	rw_enter(&auimpl_dev_lock, RW_WRITER);
+	d->d_number = num;
+	list_remove(l, d);
+	for (srch = list_head(l); srch; srch = list_next(l, srch)) {
+		if (srch->d_number >= d->d_number) {
+			break;
+		}
+	}
+	list_insert_before(l, srch, d);
+
+	rw_exit(&auimpl_dev_lock);
+}
+
+void
+auclnt_walk_devs(int (*walker)(audio_dev_t *, void *), void *arg)
+{
+	audio_dev_t	*d;
+	boolean_t	cont;
+	list_t		*l;
+
+	l = &auimpl_devs_by_index;
+	rw_enter(&auimpl_dev_lock, RW_READER);
+	for (d = list_head(l); d; d = list_next(l, d)) {
+		mutex_enter(&d->d_lock);
+		cont = walker(d, arg);
+		mutex_exit(&d->d_lock);
+		if (cont == AUDIO_WALK_STOP)
+			break;
+	}
+	rw_exit(&auimpl_dev_lock);
+}
+
+void
+auclnt_walk_devs_by_number(int (*walker)(audio_dev_t *, void *), void *arg)
+{
+	audio_dev_t	*d;
+	boolean_t	cont;
+	list_t		*l;
+
+	l = &auimpl_devs_by_number;
+	rw_enter(&auimpl_dev_lock, RW_READER);
+	for (d = list_head(l); d; d = list_next(l, d)) {
+		mutex_enter(&d->d_lock);
+		cont = walker(d, arg);
+		mutex_exit(&d->d_lock);
+		if (cont == AUDIO_WALK_STOP)
+			break;
+	}
+	rw_exit(&auimpl_dev_lock);
+}
+
+void
+auclnt_dev_walk_engines(audio_dev_t *d,
+    int (*walker)(audio_engine_t *, void *),
+    void *arg)
+{
+	audio_engine_t *e;
+	list_t *l = &d->d_engines;
+
+	mutex_enter(&d->d_lock);
+	for (e = list_head(l); e != NULL; e = list_next(l, e)) {
+		if (walker(e, arg) == AUDIO_WALK_STOP) {
+			break;
+		}
+	}
+	mutex_exit(&d->d_lock);
+}
+
+int
+auclnt_engine_get_format(audio_engine_t *e)
+{
+	return (ENG_FORMAT(e));
+}
+
+int
+auclnt_engine_get_channels(audio_engine_t *e)
+{
+	return (ENG_CHANNELS(e));
+}
+
+int
+auclnt_engine_get_rate(audio_engine_t *e)
+{
+	return (ENG_RATE(e));
+}
+
+unsigned
+auclnt_engine_get_capab(audio_engine_t *e)
+{
+	unsigned capab = 0;
+
+	if (e->e_flags & ENGINE_INPUT_CAP) {
+		capab |= AUDIO_CLIENT_CAP_RECORD;
+	}
+	if (e->e_flags & ENGINE_OUTPUT_CAP) {
+		capab |= AUDIO_CLIENT_CAP_PLAY;
+	}
+	return (capab);
+}
+
+static void
+auimpl_walk_engines(int (*walker)(audio_engine_t *, void *), void *arg)
+{
+	audio_dev_t	*d;
+	audio_engine_t	*e;
+	list_t		*l1;
+	list_t		*l2;
+	boolean_t	done = B_FALSE;
+
+	rw_enter(&auimpl_dev_lock, RW_READER);
+	l1 = &auimpl_devs_by_index;
+	for (d = list_head(l1); d; d = list_next(l1, d)) {
+		mutex_enter(&d->d_lock);
+		l2 = &d->d_engines;
+		for (e = list_head(l2); e; e = list_next(l2, e)) {
+			if (walker(e, arg) == AUDIO_WALK_STOP) {
+				done = B_TRUE;
+				break;
+			}
+		}
+		mutex_exit(&d->d_lock);
+		if (done)
+			break;
+	}
+	rw_exit(&auimpl_dev_lock);
+}
+
+/*
+ * This function suspends an engine.  The intent is to pause the
+ * engine temporarily so that it does not underrun while user threads
+ * are suspended.  The driver is still responsible for actually doing
+ * the driver suspend work -- all this does is put the engine in a
+ * paused state.  It does not prevent, for example, threads from
+ * accessing the hardware.
+ *
+ * A properly implemented driver won't even be aware of the existence
+ * of this routine -- the driver will just handle the suspend &
+ * resume.  At the point of suspend & resume, the driver will see that
+ * the engines are not running (as if all threads had "paused" it).
+ *
+ * Failure to execute either of the routines below is not critical,
+ * but will probably lead to underruns and overflows as the kernel
+ * driver gets resumed well in advance of the time when user threads
+ * are ready to start operation.
+ */
+static int
+auimpl_engine_suspend(audio_engine_t *e, void *dontcare)
+{
+	_NOTE(ARGUNUSED(dontcare));
+
+	mutex_enter(&e->e_lock);
+	e->e_suspended = B_TRUE;
+	mutex_exit(&e->e_lock);
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+auimpl_engine_resume(audio_engine_t *e, void *dontcare)
+{
+	_NOTE(ARGUNUSED(dontcare));
+	mutex_enter(&e->e_lock);
+	e->e_suspended = B_FALSE;
+	mutex_exit(&e->e_lock);
+	return (AUDIO_WALK_CONTINUE);
+}
+
+boolean_t
+auimpl_cpr(void *arg, int code)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	switch (code) {
+	case CB_CODE_CPR_CHKPT:
+		auimpl_walk_engines(auimpl_engine_suspend, NULL);
+		return (B_TRUE);
+
+	case CB_CODE_CPR_RESUME:
+		auimpl_walk_engines(auimpl_engine_resume, NULL);
+		return (B_TRUE);
+
+	default:
+		return (B_FALSE);
+	}
+}
+
+static callb_id_t	auimpl_cpr_id = 0;
+
+void
+auimpl_dev_init(void)
+{
+	rw_init(&auimpl_dev_lock, NULL, RW_DRIVER, NULL);
+	list_create(&auimpl_devs_by_index, sizeof (struct audio_dev),
+	    offsetof(struct audio_dev, d_by_index));
+	list_create(&auimpl_devs_by_number, sizeof (struct audio_dev),
+	    offsetof(struct audio_dev, d_by_number));
+
+	/*
+	 * We "borrow" the CB_CL_CPR_PM class, which gets executed at
+	 * about the right time for us.  It would be nice to have a
+	 * new CB_CL_CPR_AUDIO class, but it isn't critical at this
+	 * point.
+	 *
+	 * Note that we don't care about our thread id.
+	 */
+	auimpl_cpr_id = callb_add(auimpl_cpr, NULL, CB_CL_CPR_PM, "audio_cpr");
+}
+
+void
+auimpl_dev_fini(void)
+{
+	(void) callb_delete(auimpl_cpr_id);
+	list_destroy(&auimpl_devs_by_index);
+	list_destroy(&auimpl_devs_by_number);
+	rw_destroy(&auimpl_dev_lock);
+}
+
+void
+audio_engine_set_private(audio_engine_t *eng, void *prv)
+{
+	eng->e_private = prv;
+}
+
+void *
+audio_engine_get_private(audio_engine_t *eng)
+{
+	return (eng->e_private);
+}
+
+void
+audio_dump_bytes(const uint8_t *w, int dcount)
+{
+	char		line[64];
+	char		*s;
+	int		i;
+	const int	wrap = 16;
+
+	s = line;
+	line[0] = 0;
+
+	cmn_err(CE_NOTE, "starting @ %p", (void *)w);
+	for (i = 0; i < dcount; i++) {
+
+		(void) sprintf(s, " %02x", *w);
+		s += strlen(s);
+		w++;
+
+		if ((i % wrap) == (wrap - 1)) {
+			cmn_err(CE_NOTE, "%08x:%s", i - (wrap - 1), line);
+			line[0] = 0;
+			s = line;
+		}
+	}
+
+	if ((i % wrap) != 0) {
+		cmn_err(CE_NOTE, "%08x:%s", i - (i % wrap), line);
+	}
+}
+
+void
+audio_dump_words(const uint16_t *w, int dcount)
+{
+	char		line[64];
+	char		*s;
+	int		i;
+	const int	wrap = 8;
+
+	s = line;
+	line[0] = 0;
+
+	cmn_err(CE_NOTE, "starting @ %p", (void *)w);
+	for (i = 0; i < dcount; i++) {
+
+		(void) sprintf(s, " %04x", *w);
+		s += strlen(s);
+		w++;
+
+		if ((i % wrap) == (wrap - 1)) {
+			cmn_err(CE_NOTE, "%08x:%s", i - (wrap - 1), line);
+			line[0] = 0;
+			s = line;
+		}
+	}
+
+	if ((i % wrap) != 0) {
+		cmn_err(CE_NOTE, "%08x:%s", i - (i % wrap), line);
+	}
+}
+
+void
+audio_dump_dwords(const uint32_t *w, int dcount)
+{
+	char		line[128];
+	char		*s;
+	int		i;
+	const int	wrap = 4;
+
+	s = line;
+	line[0] = 0;
+
+	cmn_err(CE_NOTE, "starting @ %p", (void *)w);
+	for (i = 0; i < dcount; i++) {
+
+		(void) sprintf(s, " %08x", *w);
+		s += strlen(s);
+		w++;
+
+		if ((i % wrap) == (wrap - 1)) {
+			cmn_err(CE_NOTE, "%08x:%s", i - (wrap - 1), line);
+			line[0] = 0;
+			s = line;
+		}
+	}
+
+	if ((i % wrap) != 0) {
+		cmn_err(CE_NOTE, "%08x:%s", i - (i % wrap), line);
+	}
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_fltdata.c b/usr/src/uts/common/io/audio/impl/audio_fltdata.c
new file mode 100644
index 0000000000..8c5887085e
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_fltdata.c
@@ -0,0 +1,9283 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+
+/*
+ * Purpose: Filter coefficient tables for GRC3
+ */
+
+const int32_t filter_data_L[4097] = {
+	0, -43570, -87593, -132073, -177009, -222405, -268261, -314581,
+	-361366, -408617, -456336, -504526, -553189, -602325, -651938, -702028,
+	-752598, -803650, -855185, -907205, -959713, -1012709, -1066196,
+	-1120176, -1174650, -1229621, -1285089, -1341058, -1397528, -1454502,
+	-1511982, -1569968, -1628464, -1687470, -1746989, -1807023, -1867572,
+	-1928640, -1990227, -2052336, -2114969, -2178126, -2241810, -2306023,
+	-2370767, -2436042, -2501851, -2568196, -2635079, -2702500, -2770463,
+	-2838968, -2908017, -2977612, -3047755, -3118448, -3189691, -3261488,
+	-3333839, -3406746, -3480211, -3554236, -3628822, -3703971, -3779685,
+	-3855965, -3932813, -4010230, -4088218, -4166780, -4245915, -4325627,
+	-4405917, -4486785, -4568235, -4650267, -4732883, -4816085, -4899874,
+	-4984252, -5069220, -5154780, -5240933, -5327682, -5415027, -5502970,
+	-5591512, -5680656, -5770402, -5860752, -5951708, -6043271, -6135443,
+	-6228224, -6321617, -6415623, -6510244, -6605480, -6701333, -6797805,
+	-6894897, -6992611, -7090947, -7189908, -7289494, -7389707, -7490549,
+	-7592020, -7694123, -7796857, -7900226, -8004230, -8108870, -8214147,
+	-8320064, -8426621, -8533819, -8641661, -8750146, -8859276, -8969053,
+	-9079478, -9190552, -9302276, -9414651, -9527679, -9641360, -9755697,
+	-9870689, -9986339, -10102647, -10219615, -10337243, -10455532,
+	-10574485, -10694101, -10814383, -10935330, -11056944, -11179227,
+	-11302178, -11425800, -11550093, -11675058, -11800696, -11927008,
+	-12053996, -12181659, -12309999, -12439017, -12568714, -12699091,
+	-12830148, -12961887, -13094307, -13227412, -13361200, -13495673,
+	-13630832, -13766677, -13903209, -14040430, -14178339, -14316938,
+	-14456228, -14596208, -14736880, -14878245, -15020302, -15163054,
+	-15306500, -15450641, -15595478, -15741012, -15887242, -16034170,
+	-16181797, -16330122, -16479146, -16628871, -16779295, -16930421,
+	-17082248, -17234777, -17388008, -17541942, -17696580, -17851921,
+	-18007966, -18164716, -18322170, -18480330, -18639196, -18798767,
+	-18959045, -19120029, -19281720, -19444119, -19607224, -19771038,
+	-19935559, -20100788, -20266726, -20433372, -20600727, -20768791,
+	-20937563, -21107045, -21277236, -21448136, -21619746, -21792065,
+	-21965093, -22138831, -22313278, -22488435, -22664301, -22840876,
+	-23018161, -23196155, -23374859, -23554271, -23734392, -23915223,
+	-24096761, -24279008, -24461964, -24645627, -24829998, -25015077,
+	-25200863, -25387356, -25574556, -25762462, -25951074, -26140392,
+	-26330415, -26521143, -26712576, -26904712, -27097553, -27291096,
+	-27485342, -27680290, -27875940, -28072291, -28269343, -28467095,
+	-28665546, -28864695, -29064543, -29265089, -29466331, -29668269,
+	-29870903, -30074231, -30278253, -30482969, -30688377, -30894476,
+	-31101266, -31308746, -31516915, -31725772, -31935317, -32145547,
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+	39686069, 38400797, 37120729, 35845860, 34576189, 33311711, 32052423,
+	30798323, 29549406, 28305668, 27067108, 25833720, 24605501, 23382448,
+	22164556, 20951823, 19744244, 18541815, 17344532, 16152391, 14965389,
+	13783521, 12606782, 11435170, 10268679, 9107305, 7951044, 6799891,
+	5653843, 4512893, 3377039, 2246275, 1120597, 0, -1115521, -2225970,
+	-3331353, -4431675, -5526940, -6617153, -7702319, -8782443, -9857532,
+	-10927588, -11992619, -13052629, -14107623, -15157608, -16202587,
+	-17242567, -18277554, -19307552, -20332568, -21352606, -22367673,
+	-23377775, -24382917, -25383105, -26378345, -27368642, -28354004,
+	-29334435, -30309941, -31280530, -32246207, -33206977, -34162849,
+	-35113826, -36059917, -37001127, -37937462, -38868930, -39795536,
+	-40717287, -41634190, -42546251, -43453477, -44355874, -45253449,
+	-46146210, -47034162, -47917312, -48795669, -49669237, -50538025,
+	-51402039, -52261287, -53115775, -53965511, -54810501, -55650754,
+	-56486275, -57317073, -58143155, -58964527, -59781199, -60593176,
+	-61400466, -62203077, -63001017, -63794292, -64582911, -65366882,
+	-66146211, -66920907, -67690977, -68456429, -69217271, -69973511,
+	-70725156, -71472216, -72214696, -72952607, -73685955, -74414749,
+	-75138996, -75858706, -76573885, -77284543, -77990687, -78692326,
+	-79389469, -80082122, -80770296, -81453997, -82133236, -82808019,
+	-83478356, -84144255, -84805725, -85462774, -86115411, -86763645,
+	-87407483, -88046936, -88682011, -89312718, -89939065, -90561061,
+	-91178715, -91792036, -92401033, -93005714, -93606089, -94202167,
+	-94793956, -95381466, -95964706, -96543685, -97118412, -97688897,
+	-98255148, -98817175, -99374986, -99928593, -100478002, -101023225,
+	-101564269, -102101146, -102633863, -103162432, -103686860, -104207157,
+	-104723333, -105235398, -105743361, -106247232, -106747020, -107242735,
+	-107734386, -108221984, -108705537, -109185057, -109660551, -110132031,
+	-110599506, -111062985, -111522479, -111977998, -112429551, -112877148,
+	-113320800, -113760516, -114196305, -114628179, -115056147, -115480219,
+	-115900406, -116316716, -116729161, -117137751, -117542495, -117943403,
+	-118340487, -118733756, -119123220, -119508889, -119890775, -120268886,
+	-120643234, -121013828, -121380679, -121743798, -122103193, -122458877,
+	-122810860, -123159150, -123503761, -123844700, -124181980, -124515610,
+	-124845601, -125171963, -125494708, -125813845, -126129385, -126441338,
+	-126749716, -127054528, -127355785, -127653499, -127947679, -128238336,
+	-128525481, -128809125, -129089277, -129365950, -129639153, -129908897,
+	-130175193, -130438052, -130697485, -130953501, -131206113, -131455330,
+	-131701164, -131943626, -132182725, -132418474, -132650882, -132879961,
+	-133105722, -133328175, -133547331, -133763202, -133975798, -134185129,
+	-134391208, -134594044, -134793650, -134990034, -135183210, -135373187,
+	-135559977, -135743590, -135924037, -136101331, -136275480, -136446497,
+	-136614393, -136779178, -136940863, -137099461, -137254980, -137407433,
+	-137556831, -137703185, -137846505, -137986803, -138124090, -138258377,
+	-138389675, -138517995, -138643348, -138765745, -138885198, -139001717,
+	-139115313, -139225998, -139333783, -139438679, -139540696, -139639847,
+	-139736141, -139829591, -139920208, -140008001, -140092984, -140175166,
+	-140254559, -140331174, -140405022, -140476114, -140544461, -140610075,
+	-140672967, -140733147, -140790628, -140845419, -140897532, -140946979,
+	-140993770, -141037916, -141079429, -141118320, -141154599, -141188279,
+	-141219370, -141247883, -141273829, -141297220, -141318066, -141336379,
+	-141352170, -141365450, -141376229, -141384520, -141390333, -141393679,
+	-141394570, -141393015, -141389028, -141382618, -141373797, -141362575,
+	-141348965, -141332976, -141314620, -141293908, -141270851, -141245460,
+	-141217746, -141187721, -141155394, -141120778, -141083882, -141044719,
+	-141003299, -140959634, -140913733, -140865608, -140815271, -140762732,
+	-140708001, -140651091, -140592011, -140530774, -140467389, -140401868,
+	-140334221, -140264460, -140192595, -140118638, -140042598, -139964488,
+	-139884317, -139802098, -139717840, -139631554, -139543251, -139452943,
+	-139360639, -139266351, -139170090, -139071865, -138971689, -138869571,
+	-138765523, -138659555, -138551678, -138441902, -138330239, -138216698,
+	-138101292, -137984030, -137864922, -137743980, -137621215, -137496636,
+	-137370255, -137242082, -137112128, -136980402, -136846917, -136711682,
+	-136574707, -136436004, -136295583, -136153454, -136009628, -135864115,
+	-135716926, -135568071, -135417561, -135265405, -135111615, -134956201,
+	-134799172, -134640541, -134480316, -134318508, -134155127, -133990184,
+	-133823690, -133655653, -133486085, -133314995, -133142395, -132968293,
+	-132792701, -132615628, -132437085, -132257082, -132075628, -131892735,
+	-131708411, -131522667, -131335514, -131146961, -130957018, -130765695,
+	-130573002, -130378950, -130183547, -129986805, -129788732, -129589339,
+	-129388636, -129186633, -128983338, -128778764, -128572918, -128365810,
+	-128157452, -127947852, -127737019, -127524965, -127311698, -127097228,
+	-126881565, -126664718, -126446697, -126227512, -126007172, -125785687,
+	-125563066, -125339319, -125114455, -124888484, -124661415, -124433258,
+	-124204022, -123973717, -123742351, -123509935, -123276477, -123041988,
+	-122806475, -122569949, -122332418, -122093893, -121854382, -121613894,
+	-121372438, -121130025, -120886662, -120642359, -120397125, -120150969,
+	-119903900, -119655927, -119407060, -119157307, -118906676, -118655178,
+	-118402820, -118149613, -117895564, -117640682, -117384977, -117128457,
+	-116871131, -116613008, -116354096, -116094404, -115833940, -115572715,
+	-115310735, -115048010, -114784548, -114520359, -114255449, -113989829,
+	-113723506, -113456489, -113188786, -112920406, -112651358, -112381649,
+	-112111288, -111840283, -111568643, -111296375, -111023489, -110749992,
+	-110475893, -110201200, -109925920, -109650063, -109373636, -109096647,
+	-108819105, -108541017, -108262391, -107983236, -107703560, -107423370,
+	-107142674, -106861480, -106579796, -106297631, -106014991, -105731884,
+	-105448319, -105164303, -104879843, -104594949, -104309626, -104023883,
+	-103737727, -103451167, -103164209, -102876861, -102589131, -102301026,
+	-102012553, -101723721, -101434536, -101145006, -100855138, -100564940,
+	-100274419, -99983581, -99692436, -99400988, -99109247, -98817219,
+	-98524911, -98232330, -97939484, -97646380, -97353024, -97059423,
+	-96765585, -96471517, -96177226, -95882717, -95588000, -95293079,
+	-94997963, -94702657, -94407169, -94111506, -93815673, -93519679,
+	-93223529, -92927231, -92630790, -92334214, -92037509, -91740681,
+	-91443738, -91146685, -90849530, -90552278, -90254936, -89957510,
+	-89660007, -89362433, -89064795, -88767098, -88469349, -88171554,
+	-87873720, -87575852, -87277957, -86980041, -86682109, -86384169,
+	-86086225, -85788285, -85490354, -85192437, -84894542, -84596673,
+	-84298837, -84001040, -83703287, -83405584, -83107937, -82810352,
+	-82512834, -82215389, -81918024, -81620742, -81323551, -81026455,
+	-80729460, -80432572, -80135797, -79839139, -79542604, -79246197,
+	-78949925, -78653792, -78357803, -78061964, -77766281, -77470758,
+	-77175401, -76880215, -76585205, -76290376, -75995733, -75701282,
+	-75407028, -75112975, -74819129, -74525494, -74232076, -73938880,
+	-73645909, -73353171, -73060668, -72768406, -72476390, -72184624,
+	-71893114, -71601864, -71310878, -71020162, -70729720, -70439556,
+	-70149675, -69860083, -69570782, -69281779, -68993076, -68704680,
+	-68416593, -68128821, -67841368, -67554238, -67267436, -66980965,
+	-66694831, -66409037, -66123588, -65838487, -65553739, -65269349,
+	-64985319, -64701655, -64418360, -64135438, -63852894, -63570731,
+	-63288953, -63007564, -62726569, -62445970, -62165772, -61885979,
+	-61606594, -61327621, -61049064, -60770926, -60493212, -60215925,
+	-59939068, -59662646, -59386661, -59111118, -58836019, -58561369,
+	-58287171, -58013428, -57740143, -57467321, -57194964, -56923077,
+	-56651661, -56380720, -56110259, -55840279, -55570785, -55301779,
+	-55033265, -54765245, -54497724, -54230703, -53964186, -53698176,
+	-53432677, -53167690, -52903220, -52639268, -52375839, -52112934,
+	-51850557, -51588711, -51327398, -51066621, -50806383, -50546687,
+	-50287535, -50028931, -49770876, -49513373, -49256426, -49000036,
+	-48744207, -48488940, -48234239, -47980105, -47726542, -47473552,
+	-47221137, -46969299, -46718042, -46467367, -46217276, -45967773,
+	-45718859, -45470536, -45222807, -44975675, -44729140, -44483206,
+	-44237875, -43993148, -43749028, -43505516, -43262616, -43020329,
+	-42778656, -42537601, -42297164, -42057349, -41818156, -41579587,
+	-41341645, -41104332, -40867649, -40631598, -40396180, -40161398,
+	-39927254, -39693748, -39460883, -39228661, -38997082, -38766149,
+	-38535863, -38306227, -38077240, -37848905, -37621224, -37394198,
+	-37167828, -36942116, -36717063, -36492671, -36268941, -36045874,
+	-35823472, -35601737, -35380668, -35160268, -34940539, -34721480,
+	-34503094, -34285381, -34068343, -33851981, -33636296, -33421289,
+	-33206961, -32993313, -32780347, -32568064, -32356463, -32145547,
+	-31935317, -31725772, -31516915, -31308746, -31101266, -30894476,
+	-30688377, -30482969, -30278253, -30074231, -29870903, -29668269,
+	-29466331, -29265089, -29064543, -28864695, -28665546, -28467095,
+	-28269343, -28072291, -27875940, -27680290, -27485342, -27291096,
+	-27097553, -26904712, -26712576, -26521143, -26330415, -26140392,
+	-25951074, -25762462, -25574556, -25387356, -25200863, -25015077,
+	-24829998, -24645627, -24461964, -24279008, -24096761, -23915223,
+	-23734392, -23554271, -23374859, -23196155, -23018161, -22840876,
+	-22664301, -22488435, -22313278, -22138831, -21965093, -21792065,
+	-21619746, -21448136, -21277236, -21107045, -20937563, -20768791,
+	-20600727, -20433372, -20266726, -20100788, -19935559, -19771038,
+	-19607224, -19444119, -19281720, -19120029, -18959045, -18798767,
+	-18639196, -18480330, -18322170, -18164716, -18007966, -17851921,
+	-17696580, -17541942, -17388008, -17234777, -17082248, -16930421,
+	-16779295, -16628871, -16479146, -16330122, -16181797, -16034170,
+	-15887242, -15741012, -15595478, -15450641, -15306500, -15163054,
+	-15020302, -14878245, -14736880, -14596208, -14456228, -14316938,
+	-14178339, -14040430, -13903209, -13766677, -13630832, -13495673,
+	-13361200, -13227412, -13094307, -12961887, -12830148, -12699091,
+	-12568714, -12439017, -12309999, -12181659, -12053996, -11927008,
+	-11800696, -11675058, -11550093, -11425800, -11302178, -11179227,
+	-11056944, -10935330, -10814383, -10694101, -10574485, -10455532,
+	-10337243, -10219615, -10102647, -9986339, -9870689, -9755697,
+	-9641360, -9527679, -9414651, -9302276, -9190552, -9079478, -8969053,
+	-8859276, -8750146, -8641661, -8533819, -8426621, -8320064, -8214147,
+	-8108870, -8004230, -7900226, -7796857, -7694123, -7592020, -7490549,
+	-7389707, -7289494, -7189908, -7090947, -6992611, -6894897, -6797805,
+	-6701333, -6605480, -6510244, -6415623, -6321617, -6228224, -6135443,
+	-6043271, -5951708, -5860752, -5770402, -5680656, -5591512, -5502970,
+	-5415027, -5327682, -5240933, -5154780, -5069220, -4984252, -4899874,
+	-4816085, -4732883, -4650267, -4568235, -4486785, -4405917, -4325627,
+	-4245915, -4166780, -4088218, -4010230, -3932813, -3855965, -3779685,
+	-3703971, -3628822, -3554236, -3480211, -3406746, -3333839, -3261488,
+	-3189691, -3118448, -3047755, -2977612, -2908017, -2838968, -2770463,
+	-2702500, -2635079, -2568196, -2501851, -2436042, -2370767, -2306023,
+	-2241810, -2178126, -2114969, -2052336, -1990227, -1928640, -1867572,
+	-1807023, -1746989, -1687470, -1628464, -1569968, -1511982, -1454502,
+	-1397528, -1341058, -1285089, -1229621, -1174650, -1120176, -1066196,
+	-1012709, -959713, -907205, -855185, -803650, -752598, -702028,
+	-651938, -602325, -553189, -504526, -456336, -408617, -361366, -314581,
+	-268261, -222405, -177009, -132073, -87593, -43570, 0
+};
+
+const int32_t filter_data_M[8193] = {
+	0, -5008, -10058, -15150, -20285, -25462, -30682, -35945, -41250,
+	-46598, -51990, -57425, -62903, -68425, -73990, -79599, -85252, -90949,
+	-96690, -102475, -108304, -114178, -120097, -126060, -132068, -138121,
+	-144219, -150362, -156550, -162784, -169063, -175387, -181758, -188174,
+	-194636, -201145, -207699, -214300, -220947, -227641, -234381, -241168,
+	-248001, -254882, -261810, -268785, -275807, -282876, -289993, -297157,
+	-304369, -311629, -318937, -326293, -333697, -341149, -348649, -356197,
+	-363794, -371440, -379134, -386877, -394669, -402510, -410400, -418339,
+	-426328, -434365, -442452, -450589, -458775, -467011, -475297, -483632,
+	-492018, -500453, -508939, -517475, -526061, -534698, -543385, -552122,
+	-560911, -569750, -578639, -587580, -596572, -605615, -614708, -623853,
+	-633050, -642297, -651597, -660947, -670349, -679803, -689309, -698866,
+	-708475, -718136, -727849, -737615, -747432, -757301, -767223, -777197,
+	-787224, -797303, -807434, -817618, -827855, -838144, -848486, -858881,
+	-869329, -879830, -890384, -900990, -911650, -922363, -933129, -943949,
+	-954822, -965748, -976727, -987760, -998846, -1009986, -1021179,
+	-1032426, -1043727, -1055082, -1066490, -1077952, -1089467, -1101037,
+	-1112660, -1124337, -1136069, -1147854, -1159693, -1171587, -1183534,
+	-1195535, -1207591, -1219701, -1231865, -1244083, -1256355, -1268682,
+	-1281063, -1293498, -1305988, -1318532, -1331130, -1343783, -1356490,
+	-1369251, -1382067, -1394937, -1407862, -1420841, -1433874, -1446962,
+	-1460105, -1473302, -1486553, -1499859, -1513219, -1526634, -1540103,
+	-1553627, -1567205, -1580838, -1594525, -1608267, -1622063, -1635913,
+	-1649818, -1663778, -1677791, -1691859, -1705982, -1720159, -1734390,
+	-1748676, -1763015, -1777410, -1791858, -1806361, -1820917, -1835528,
+	-1850194, -1864913, -1879686, -1894514, -1909395, -1924331, -1939320,
+	-1954364, -1969461, -1984612, -1999817, -2015076, -2030388, -2045755,
+	-2061174, -2076648, -2092175, -2107755, -2123389, -2139076, -2154817,
+	-2170611, -2186458, -2202358, -2218312, -2234318, -2250378, -2266490,
+	-2282656, -2298874, -2315144, -2331468, -2347844, -2364273, -2380754,
+	-2397287, -2413873, -2430511, -2447201, -2463943, -2480737, -2497583,
+	-2514481, -2531431, -2548432, -2565485, -2582589, -2599745, -2616951,
+	-2634210, -2651519, -2668879, -2686290, -2703752, -2721265, -2738828,
+	-2756441, -2774105, -2791820, -2809584, -2827399, -2845263, -2863178,
+	-2881142, -2899155, -2917218, -2935331, -2953492, -2971703, -2989963,
+	-3008272, -3026629, -3045035, -3063490, -3081993, -3100544, -3119143,
+	-3137790, -3156485, -3175227, -3194017, -3212855, -3231739, -3250671,
+	-3269650, -3288675, -3307747, -3326866, -3346031, -3365242, -3384499,
+	-3403802, -3423151, -3442545, -3461985, -3481469, -3500999, -3520574,
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+	-10412153, -10432465, -10452530, -10472349, -10491921, -10511249,
+	-10530332, -10549172, -10567768, -10586122, -10604234, -10622105,
+	-10639735, -10657126, -10674278, -10691191, -10707866, -10724305,
+	-10740507, -10756473, -10772204, -10787701, -10802964, -10817995,
+	-10832793, -10847359, -10861695, -10875800, -10889676, -10903323,
+	-10916742, -10929933, -10942897, -10955636, -10968149, -10980438,
+	-10992502, -11004344, -11015963, -11027360, -11038536, -11049491,
+	-11060227, -11070743, -11081042, -11091123, -11100987, -11110634,
+	-11120067, -11129284, -11138288, -11147078, -11155656, -11164022,
+	-11172176, -11180121, -11187855, -11195381, -11202698, -11209808,
+	-11216710, -11223407, -11229898, -11236185, -11242268, -11248148,
+	-11253825, -11259300, -11264574, -11269648, -11274523, -11279199,
+	-11283676, -11287956, -11292040, -11295928, -11299620, -11303118,
+	-11306422, -11309534, -11312453, -11315181, -11317717, -11320064,
+	-11322222, -11324191, -11325972, -11327566, -11328974, -11330196,
+	-11331233, -11332087, -11332756, -11333243, -11333549, -11333673,
+	-11333616, -11333380, -11332965, -11332372, -11331601, -11330654,
+	-11329531, -11328232, -11326759, -11325112, -11323292, -11321300,
+	-11319136, -11316802, -11314298, -11311624, -11308781, -11305771,
+	-11302594, -11299250, -11295741, -11292067, -11288229, -11284228,
+	-11280064, -11275738, -11271250, -11266603, -11261796, -11256830,
+	-11251705, -11246424, -11240985, -11235391, -11229642, -11223738,
+	-11217680, -11211469, -11205107, -11198592, -11191927, -11185112,
+	-11178147, -11171034, -11163774, -11156366, -11148811, -11141112,
+	-11133267, -11125278, -11117146, -11108871, -11100455, -11091897,
+	-11083198, -11074360, -11065383, -11056268, -11047015, -11037626,
+	-11028100, -11018440, -11008644, -10998715, -10988653, -10978458,
+	-10968131, -10957674, -10947087, -10936370, -10925524, -10914550,
+	-10903449, -10892221, -10880868, -10869389, -10857786, -10846060,
+	-10834210, -10822238, -10810145, -10797931, -10785597, -10773143,
+	-10760571, -10747881, -10735074, -10722150, -10709110, -10695956,
+	-10682687, -10669304, -10655809, -10642201, -10628482, -10614652,
+	-10600712, -10586662, -10572504, -10558238, -10543865, -10529385,
+	-10514799, -10500109, -10485313, -10470414, -10455412, -10440308,
+	-10425102, -10409794, -10394387, -10378880, -10363275, -10347571,
+	-10331769, -10315871, -10299877, -10283787, -10267603, -10251324,
+	-10234952, -10218488, -10201931, -10185284, -10168545, -10151717,
+	-10134799, -10117793, -10100699, -10083518, -10066250, -10048896,
+	-10031457, -10013933, -9996326, -9978635, -9960862, -9943007, -9925071,
+	-9907054, -9888957, -9870782, -9852527, -9834195, -9815785, -9797299,
+	-9778737, -9760100, -9741388, -9722602, -9703743, -9684811, -9665807,
+	-9646732, -9627586, -9608370, -9589084, -9569730, -9550308, -9530818,
+	-9511261, -9491638, -9471949, -9452196, -9432378, -9412496, -9392551,
+	-9372544, -9352475, -9332344, -9312154, -9291903, -9271593, -9251224,
+	-9230797, -9210312, -9189771, -9169174, -9148520, -9127812, -9107050,
+	-9086233, -9065363, -9044441, -9023467, -9002441, -8981365, -8960238,
+	-8939062, -8917837, -8896563, -8875242, -8853873, -8832458, -8810996,
+	-8789490, -8767938, -8746342, -8724702, -8703020, -8681294, -8659527,
+	-8637718, -8615869, -8593979, -8572050, -8550081, -8528074, -8506029,
+	-8483947, -8461827, -8439672, -8417481, -8395254, -8372993, -8350698,
+	-8328370, -8306008, -8283614, -8261189, -8238732, -8216244, -8193726,
+	-8171178, -8148602, -8125996, -8103363, -8080702, -8058014, -8035300,
+	-8012560, -7989794, -7967004, -7944189, -7921350, -7898488, -7875604,
+	-7852697, -7829768, -7806818, -7783848, -7760857, -7737847, -7714817,
+	-7691769, -7668702, -7645618, -7622517, -7599399, -7576265, -7553115,
+	-7529951, -7506771, -7483577, -7460370, -7437149, -7413916, -7390670,
+	-7367413, -7344144, -7320865, -7297575, -7274275, -7250966, -7227648,
+	-7204322, -7180988, -7157646, -7134297, -7110942, -7087580, -7064213,
+	-7040841, -7017464, -6994083, -6970697, -6947309, -6923917, -6900523,
+	-6877127, -6853730, -6830331, -6806931, -6783532, -6760132, -6736733,
+	-6713335, -6689938, -6666544, -6643151, -6619762, -6596375, -6572992,
+	-6549613, -6526239, -6502869, -6479504, -6456146, -6432793, -6409447,
+	-6386107, -6362775, -6339451, -6316134, -6292827, -6269528, -6246238,
+	-6222958, -6199688, -6176428, -6153180, -6129942, -6106717, -6083503,
+	-6060301, -6037113, -6013937, -5990775, -5967627, -5944493, -5921374,
+	-5898270, -5875181, -5852108, -5829050, -5806010, -5782986, -5759979,
+	-5736990, -5714019, -5691066, -5668131, -5645216, -5622320, -5599443,
+	-5576586, -5553750, -5530934, -5508139, -5485366, -5462614, -5439884,
+	-5417177, -5394492, -5371830, -5349191, -5326576, -5303985, -5281418,
+	-5258876, -5236358, -5213866, -5191399, -5168958, -5146544, -5124155,
+	-5101793, -5079459, -5057152, -5034872, -5012620, -4990397, -4968202,
+	-4946036, -4923898, -4901791, -4879713, -4857665, -4835647, -4813660,
+	-4791703, -4769778, -4747884, -4726021, -4704191, -4682393, -4660627,
+	-4638894, -4617193, -4595526, -4573893, -4552293, -4530728, -4509196,
+	-4487699, -4466237, -4444810, -4423418, -4402062, -4380741, -4359457,
+	-4338209, -4316997, -4295822, -4274684, -4253583, -4232520, -4211494,
+	-4190506, -4169557, -4148645, -4127772, -4106939, -4086144, -4065388,
+	-4044672, -4023995, -4003359, -3982762, -3962206, -3941690, -3921215,
+	-3900781, -3880388, -3860037, -3839727, -3819458, -3799232, -3779048,
+	-3758906, -3738807, -3718750, -3698736, -3678766, -3658838, -3638955,
+	-3619114, -3599318, -3579565, -3559857, -3540193, -3520574, -3500999,
+	-3481469, -3461985, -3442545, -3423151, -3403802, -3384499, -3365242,
+	-3346031, -3326866, -3307747, -3288675, -3269650, -3250671, -3231739,
+	-3212855, -3194017, -3175227, -3156485, -3137790, -3119143, -3100544,
+	-3081993, -3063490, -3045035, -3026629, -3008272, -2989963, -2971703,
+	-2953492, -2935331, -2917218, -2899155, -2881142, -2863178, -2845263,
+	-2827399, -2809584, -2791820, -2774105, -2756441, -2738828, -2721265,
+	-2703752, -2686290, -2668879, -2651519, -2634210, -2616951, -2599745,
+	-2582589, -2565485, -2548432, -2531431, -2514481, -2497583, -2480737,
+	-2463943, -2447201, -2430511, -2413873, -2397287, -2380754, -2364273,
+	-2347844, -2331468, -2315144, -2298874, -2282656, -2266490, -2250378,
+	-2234318, -2218312, -2202358, -2186458, -2170611, -2154817, -2139076,
+	-2123389, -2107755, -2092175, -2076648, -2061174, -2045755, -2030388,
+	-2015076, -1999817, -1984612, -1969461, -1954364, -1939320, -1924331,
+	-1909395, -1894514, -1879686, -1864913, -1850194, -1835528, -1820917,
+	-1806361, -1791858, -1777410, -1763015, -1748676, -1734390, -1720159,
+	-1705982, -1691859, -1677791, -1663778, -1649818, -1635913, -1622063,
+	-1608267, -1594525, -1580838, -1567205, -1553627, -1540103, -1526634,
+	-1513219, -1499859, -1486553, -1473302, -1460105, -1446962, -1433874,
+	-1420841, -1407862, -1394937, -1382067, -1369251, -1356490, -1343783,
+	-1331130, -1318532, -1305988, -1293498, -1281063, -1268682, -1256355,
+	-1244083, -1231865, -1219701, -1207591, -1195535, -1183534, -1171587,
+	-1159693, -1147854, -1136069, -1124337, -1112660, -1101037, -1089467,
+	-1077952, -1066490, -1055082, -1043727, -1032426, -1021179, -1009986,
+	-998846, -987760, -976727, -965748, -954822, -943949, -933129, -922363,
+	-911650, -900990, -890384, -879830, -869329, -858881, -848486, -838144,
+	-827855, -817618, -807434, -797303, -787224, -777197, -767223, -757301,
+	-747432, -737615, -727849, -718136, -708475, -698866, -689309, -679803,
+	-670349, -660947, -651597, -642297, -633050, -623853, -614708, -605615,
+	-596572, -587580, -578639, -569750, -560911, -552122, -543385, -534698,
+	-526061, -517475, -508939, -500453, -492018, -483632, -475297, -467011,
+	-458775, -450589, -442452, -434365, -426328, -418339, -410400, -402510,
+	-394669, -386877, -379134, -371440, -363794, -356197, -348649, -341149,
+	-333697, -326293, -318937, -311629, -304369, -297157, -289993, -282876,
+	-275807, -268785, -261810, -254882, -248001, -241168, -234381, -227641,
+	-220947, -214300, -207699, -201145, -194636, -188174, -181758, -175387,
+	-169063, -162784, -156550, -150362, -144219, -138121, -132068, -126060,
+	-120097, -114178, -108304, -102475, -96690, -90949, -85252, -79599,
+	-73990, -68425, -62903, -57425, -51990, -46598, -41250, -35945, -30682,
+	-25462, -20285, -15150, -10058, -5008, 0
+};
+
+const int32_t filter_data_H[16385] = {
+	0, -2498, -5007, -7527, -10057, -12598, -15149, -17711, -20283, -22866,
+	-25459, -28063, -30677, -33301, -35936, -38582, -41238, -43904, -46581,
+	-49268, -51966, -54674, -57392, -60121, -62860, -65610, -68370, -71141,
+	-73922, -76713, -79515, -82327, -85149, -87982, -90825, -93678, -96542,
+	-99416, -102301, -105195, -108100, -111016, -113941, -116877, -119823,
+	-122779, -125746, -128723, -131710, -134707, -137714, -140732, -143760,
+	-146798, -149846, -152904, -155972, -159051, -162139, -165238, -168347,
+	-171466, -174594, -177733, -180882, -184041, -187210, -190389, -193578,
+	-196777, -199986, -203205, -206433, -209672, -212920, -216179, -219447,
+	-222725, -226013, -229310, -232618, -235935, -239262, -242599, -245945,
+	-249301, -252667, -256042, -259427, -262822, -266227, -269641, -273064,
+	-276497, -279940, -283392, -286854, -290325, -293805, -297295, -300795,
+	-304304, -307822, -311350, -314886, -318433, -321988, -325553, -329127,
+	-332710, -336303, -339904, -343515, -347135, -350764, -354402, -358050,
+	-361706, -365371, -369045, -372728, -376421, -380122, -383832, -387550,
+	-391278, -395014, -398760, -402514, -406276, -410048, -413828, -417617,
+	-421414, -425220, -429035, -432858, -436690, -440530, -444378, -448235,
+	-452101, -455974, -459857, -463747, -467646, -471553, -475468, -479391,
+	-483323, -487262, -491210, -495166, -499130, -503102, -507082, -511069,
+	-515065, -519069, -523080, -527099, -531126, -535161, -539203, -543253,
+	-547311, -551376, -555449, -559530, -563618, -567713, -571816, -575926,
+	-580044, -584169, -588301, -592440, -596587, -600741, -604902, -609070,
+	-613245, -617427, -621616, -625812, -630015, -634225, -638442, -642665,
+	-646895, -651132, -655376, -659626, -663883, -668146, -672416, -676693,
+	-680975, -685265, -689560, -693862, -698170, -702484, -706805, -711132,
+	-715464, -719803, -724148, -728499, -732855, -737218, -741586, -745961,
+	-750341, -754726, -759117, -763514, -767917, -772325, -776738, -781157,
+	-785581, -790011, -794446, -798886, -803331, -807782, -812237, -816698,
+	-821164, -825634, -830110, -834590, -839075, -843565, -848060, -852559,
+	-857063, -861571, -866084, -870602, -875123, -879650, -884180, -888715,
+	-893254, -897797, -902344, -906895, -911451, -916010, -920573, -925140,
+	-929711, -934285, -938863, -943445, -948031, -952620, -957212, -961808,
+	-966407, -971009, -975615, -980224, -984836, -989451, -994069, -998690,
+	-1003314, -1007941, -1012571, -1017203, -1021838, -1026476, -1031116,
+	-1035759, -1040404, -1045052, -1049702, -1054354, -1059009, -1063665,
+	-1068324, -1072985, -1077648, -1082313, -1086979, -1091648, -1096318,
+	-1100990, -1105663, -1110338, -1115015, -1119693, -1124372, -1129053,
+	-1133735, -1138418, -1143102, -1147787, -1152474, -1157161, -1161849,
+	-1166538, -1171228, -1175918, -1180609, -1185301, -1189993, -1194686,
+	-1199379, -1204072, -1208765, -1213459, -1218153, -1222847, -1227540,
+	-1232234, -1236928, -1241621, -1246314, -1251007, -1255699, -1260391,
+	-1265082, -1269773, -1274463, -1279152, -1283840, -1288528, -1293214,
+	-1297900, -1302584, -1307267, -1311950, -1316630, -1321310, -1325987,
+	-1330664, -1335339, -1340012, -1344683, -1349353, -1354021, -1358687,
+	-1363350, -1368012, -1372672, -1377329, -1381984, -1386637, -1391288,
+	-1395935, -1400581, -1405223, -1409863, -1414500, -1419135, -1423766,
+	-1428394, -1433019, -1437641, -1442260, -1446876, -1451488, -1456097,
+	-1460702, -1465303, -1469901, -1474495, -1479086, -1483672, -1488255,
+	-1492833, -1497407, -1501977, -1506543, -1511105, -1515662, -1520214,
+	-1524762, -1529305, -1533844, -1538377, -1542906, -1547430, -1551949,
+	-1556463, -1560971, -1565474, -1569972, -1574465, -1578952, -1583433,
+	-1587909, -1592378, -1596843, -1601301, -1605753, -1610199, -1614639,
+	-1619073, -1623500, -1627921, -1632336, -1636744, -1641146, -1645540,
+	-1649929, -1654310, -1658684, -1663051, -1667411, -1671764, -1676110,
+	-1680448, -1684779, -1689102, -1693418, -1697726, -1702027, -1706319,
+	-1710604, -1714880, -1719149, -1723409, -1727662, -1731905, -1736141,
+	-1740368, -1744586, -1748796, -1752997, -1757189, -1761372, -1765546,
+	-1769711, -1773867, -1778014, -1782152, -1786280, -1790398, -1794507,
+	-1798606, -1802696, -1806776, -1810845, -1814905, -1818955, -1822995,
+	-1827024, -1831043, -1835051, -1839050, -1843037, -1847014, -1850980,
+	-1854935, -1858879, -1862812, -1866734, -1870645, -1874545, -1878433,
+	-1882310, -1886175, -1890029, -1893871, -1897701, -1901520, -1905326,
+	-1909120, -1912902, -1916672, -1920430, -1924175, -1927908, -1931628,
+	-1935336, -1939030, -1942712, -1946381, -1950037, -1953680, -1957310,
+	-1960926, -1964529, -1968119, -1971695, -1975258, -1978806, -1982341,
+	-1985863, -1989370, -1992863, -1996342, -1999806, -2003257, -2006693,
+	-2010114, -2013521, -2016913, -2020291, -2023654, -2027001, -2030334,
+	-2033651, -2036954, -2040241, -2043513, -2046769, -2050009, -2053234,
+	-2056444, -2059637, -2062815, -2065976, -2069122, -2072251, -2075364,
+	-2078461, -2081541, -2084605, -2087652, -2090682, -2093696, -2096693,
+	-2099672, -2102635, -2105581, -2108509, -2111420, -2114313, -2117189,
+	-2120048, -2122889, -2125712, -2128517, -2131304, -2134073, -2136824,
+	-2139557, -2142272, -2144968, -2147646, -2150305, -2152945, -2155567,
+	-2158170, -2160754, -2163319, -2165865, -2168391, -2170899, -2173387,
+	-2175855, -2178304, -2180734, -2183144, -2185533, -2187904, -2190254,
+	-2192584, -2194893, -2197183, -2199452, -2201701, -2203929, -2206137,
+	-2208324, -2210490, -2212636, -2214760, -2216864, -2218946, -2221007,
+	-2223047, -2225066, -2227063, -2229038, -2230992, -2232924, -2234834,
+	-2236722, -2238589, -2240433, -2242255, -2244055, -2245832, -2247587,
+	-2249320, -2251030, -2252717, -2254382, -2256023, -2257642, -2259238,
+	-2260810, -2262360, -2263886, -2265388, -2266868, -2268323, -2269755,
+	-2271164, -2272548, -2273909, -2275246, -2276558, -2277847, -2279111,
+	-2280351, -2281567, -2282758, -2283924, -2285066, -2286183, -2287276,
+	-2288343, -2289386, -2290403, -2291395, -2292362, -2293304, -2294220,
+	-2295111, -2295976, -2296816, -2297630, -2298418, -2299180, -2299916,
+	-2300626, -2301310, -2301968, -2302600, -2303205, -2303783, -2304335,
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+	2608839, 2591610, 2574395, 2557193, 2540004, 2522829, 2505667, 2488520,
+	2471386, 2454267, 2437162, 2420072, 2402997, 2385936, 2368891, 2351861,
+	2334846, 2317847, 2300864, 2283896, 2266945, 2250010, 2233091, 2216189,
+	2199303, 2182435, 2165583, 2148749, 2131932, 2115133, 2098351, 2081587,
+	2064841, 2048113, 2031404, 2014713, 1998040, 1981387, 1964752, 1948136,
+	1931540, 1914963, 1898405, 1881867, 1865349, 1848851, 1832373, 1815915,
+	1799477, 1783060, 1766664, 1750289, 1733935, 1717601, 1701289, 1684999,
+	1668730, 1652483, 1636257, 1620054, 1603872, 1587713, 1571577, 1555462,
+	1539371, 1523302, 1507257, 1491234, 1475234, 1459258, 1443306, 1427377,
+	1411471, 1395590, 1379733, 1363899, 1348091, 1332306, 1316546, 1300811,
+	1285100, 1269414, 1253754, 1238118, 1222508, 1206923, 1191364, 1175831,
+	1160323, 1144841, 1129385, 1113956, 1098552, 1083175, 1067825, 1052501,
+	1037204, 1021934, 1006691, 991475, 976286, 961124, 945990, 930884,
+	915805, 900754, 885731, 870735, 855768, 840830, 825919, 811037, 796184,
+	781359, 766563, 751796, 737058, 722349, 707669, 693019, 678398, 663806,
+	649244, 634712, 620210, 605737, 591295, 576883, 562501, 548149, 533828,
+	519537, 505277, 491047, 476848, 462681, 448544, 434438, 420364, 406320,
+	392308, 378328, 364379, 350462, 336576, 322723, 308901, 295111, 281353,
+	267628, 253934, 240274, 226645, 213049, 199485, 185955, 172457, 158991,
+	145559, 132160, 118794, 105461, 92161, 78894, 65661, 52462, 39296,
+	26163, 13064, 0, -13031, -26028, -38991, -51920, -64815, -77676,
+	-90503, -103295, -116053, -128776, -141465, -154119, -166738, -179323,
+	-191873, -204388, -216868, -229313, -241723, -254097, -266437, -278741,
+	-291010, -303244, -315442, -327605, -339732, -351823, -363879, -375899,
+	-387884, -399832, -411745, -423621, -435462, -447266, -459035, -470767,
+	-482463, -494122, -505746, -517333, -528883, -540397, -551875, -563316,
+	-574720, -586088, -597419, -608713, -619971, -631191, -642375, -653522,
+	-664632, -675704, -686740, -697739, -708700, -719625, -730512, -741362,
+	-752174, -762949, -773687, -784387, -795050, -805676, -816264, -826814,
+	-837327, -847802, -858240, -868640, -879002, -889326, -899613, -909862,
+	-920073, -930246, -940381, -950479, -960538, -970560, -980543, -990488,
+	-1000396, -1010265, -1020096, -1029889, -1039644, -1049361, -1059039,
+	-1068680, -1078282, -1087846, -1097371, -1106859, -1116308, -1125718,
+	-1135091, -1144425, -1153720, -1162977, -1172196, -1181376, -1190518,
+	-1199622, -1208687, -1217713, -1226701, -1235651, -1244562, -1253434,
+	-1262268, -1271063, -1279820, -1288538, -1297218, -1305859, -1314462,
+	-1323026, -1331551, -1340038, -1348486, -1356895, -1365266, -1373598,
+	-1381892, -1390147, -1398363, -1406540, -1414680, -1422780, -1430842,
+	-1438865, -1446849, -1454795, -1462702, -1470571, -1478401, -1486192,
+	-1493945, -1501659, -1509334, -1516971, -1524569, -1532129, -1539650,
+	-1547132, -1554576, -1561981, -1569348, -1576676, -1583966, -1591217,
+	-1598429, -1605603, -1612739, -1619836, -1626894, -1633914, -1640896,
+	-1647839, -1654744, -1661610, -1668438, -1675228, -1681979, -1688692,
+	-1695366, -1702002, -1708600, -1715160, -1721681, -1728164, -1734609,
+	-1741016, -1747384, -1753714, -1760007, -1766261, -1772476, -1778654,
+	-1784794, -1790896, -1796959, -1802985, -1808973, -1814923, -1820835,
+	-1826709, -1832545, -1838343, -1844104, -1849826, -1855511, -1861159,
+	-1866768, -1872340, -1877874, -1883371, -1888830, -1894252, -1899636,
+	-1904982, -1910291, -1915563, -1920797, -1925994, -1931154, -1936276,
+	-1941361, -1946409, -1951420, -1956393, -1961329, -1966229, -1971091,
+	-1975916, -1980705, -1985456, -1990171, -1994848, -1999489, -2004093,
+	-2008660, -2013191, -2017685, -2022142, -2026563, -2030947, -2035295,
+	-2039607, -2043881, -2048120, -2052322, -2056488, -2060618, -2064712,
+	-2068769, -2072790, -2076776, -2080725, -2084638, -2088516, -2092357,
+	-2096163, -2099933, -2103667, -2107366, -2111029, -2114656, -2118248,
+	-2121804, -2125325, -2128811, -2132261, -2135676, -2139056, -2142400,
+	-2145710, -2148984, -2152224, -2155428, -2158598, -2161732, -2164832,
+	-2167897, -2170928, -2173923, -2176885, -2179811, -2182703, -2185561,
+	-2188385, -2191174, -2193929, -2196649, -2199336, -2201988, -2204607,
+	-2207191, -2209742, -2212258, -2214741, -2217191, -2219606, -2221988,
+	-2224337, -2226652, -2228933, -2231181, -2233396, -2235578, -2237727,
+	-2239842, -2241924, -2243974, -2245990, -2247974, -2249924, -2251842,
+	-2253728, -2255581, -2257401, -2259189, -2260944, -2262667, -2264358,
+	-2266016, -2267643, -2269237, -2270800, -2272330, -2273828, -2275295,
+	-2276730, -2278133, -2279505, -2280845, -2282154, -2283431, -2284677,
+	-2285892, -2287076, -2288228, -2289350, -2290440, -2291499, -2292528,
+	-2293526, -2294493, -2295430, -2296336, -2297211, -2298057, -2298871,
+	-2299656, -2300410, -2301135, -2301829, -2302493, -2303128, -2303732,
+	-2304307, -2304852, -2305368, -2305854, -2306310, -2306738, -2307135,
+	-2307504, -2307844, -2308154, -2308436, -2308688, -2308912, -2309107,
+	-2309273, -2309411, -2309520, -2309601, -2309653, -2309677, -2309673,
+	-2309640, -2309580, -2309492, -2309375, -2309231, -2309059, -2308860,
+	-2308633, -2308378, -2308096, -2307786, -2307449, -2307086, -2306694,
+	-2306276, -2305831, -2305359, -2304861, -2304335, -2303783, -2303205,
+	-2302600, -2301968, -2301310, -2300626, -2299916, -2299180, -2298418,
+	-2297630, -2296816, -2295976, -2295111, -2294220, -2293304, -2292362,
+	-2291395, -2290403, -2289386, -2288343, -2287276, -2286183, -2285066,
+	-2283924, -2282758, -2281567, -2280351, -2279111, -2277847, -2276558,
+	-2275246, -2273909, -2272548, -2271164, -2269755, -2268323, -2266868,
+	-2265388, -2263886, -2262360, -2260810, -2259238, -2257642, -2256023,
+	-2254382, -2252717, -2251030, -2249320, -2247587, -2245832, -2244055,
+	-2242255, -2240433, -2238589, -2236722, -2234834, -2232924, -2230992,
+	-2229038, -2227063, -2225066, -2223047, -2221007, -2218946, -2216864,
+	-2214760, -2212636, -2210490, -2208324, -2206137, -2203929, -2201701,
+	-2199452, -2197183, -2194893, -2192584, -2190254, -2187904, -2185533,
+	-2183144, -2180734, -2178304, -2175855, -2173387, -2170899, -2168391,
+	-2165865, -2163319, -2160754, -2158170, -2155567, -2152945, -2150305,
+	-2147646, -2144968, -2142272, -2139557, -2136824, -2134073, -2131304,
+	-2128517, -2125712, -2122889, -2120048, -2117189, -2114313, -2111420,
+	-2108509, -2105581, -2102635, -2099672, -2096693, -2093696, -2090682,
+	-2087652, -2084605, -2081541, -2078461, -2075364, -2072251, -2069122,
+	-2065976, -2062815, -2059637, -2056444, -2053234, -2050009, -2046769,
+	-2043513, -2040241, -2036954, -2033651, -2030334, -2027001, -2023654,
+	-2020291, -2016913, -2013521, -2010114, -2006693, -2003257, -1999806,
+	-1996342, -1992863, -1989370, -1985863, -1982341, -1978806, -1975258,
+	-1971695, -1968119, -1964529, -1960926, -1957310, -1953680, -1950037,
+	-1946381, -1942712, -1939030, -1935336, -1931628, -1927908, -1924175,
+	-1920430, -1916672, -1912902, -1909120, -1905326, -1901520, -1897701,
+	-1893871, -1890029, -1886175, -1882310, -1878433, -1874545, -1870645,
+	-1866734, -1862812, -1858879, -1854935, -1850980, -1847014, -1843037,
+	-1839050, -1835051, -1831043, -1827024, -1822995, -1818955, -1814905,
+	-1810845, -1806776, -1802696, -1798606, -1794507, -1790398, -1786280,
+	-1782152, -1778014, -1773867, -1769711, -1765546, -1761372, -1757189,
+	-1752997, -1748796, -1744586, -1740368, -1736141, -1731905, -1727662,
+	-1723409, -1719149, -1714880, -1710604, -1706319, -1702027, -1697726,
+	-1693418, -1689102, -1684779, -1680448, -1676110, -1671764, -1667411,
+	-1663051, -1658684, -1654310, -1649929, -1645540, -1641146, -1636744,
+	-1632336, -1627921, -1623500, -1619073, -1614639, -1610199, -1605753,
+	-1601301, -1596843, -1592378, -1587909, -1583433, -1578952, -1574465,
+	-1569972, -1565474, -1560971, -1556463, -1551949, -1547430, -1542906,
+	-1538377, -1533844, -1529305, -1524762, -1520214, -1515662, -1511105,
+	-1506543, -1501977, -1497407, -1492833, -1488255, -1483672, -1479086,
+	-1474495, -1469901, -1465303, -1460702, -1456097, -1451488, -1446876,
+	-1442260, -1437641, -1433019, -1428394, -1423766, -1419135, -1414500,
+	-1409863, -1405223, -1400581, -1395935, -1391288, -1386637, -1381984,
+	-1377329, -1372672, -1368012, -1363350, -1358687, -1354021, -1349353,
+	-1344683, -1340012, -1335339, -1330664, -1325987, -1321310, -1316630,
+	-1311950, -1307267, -1302584, -1297900, -1293214, -1288528, -1283840,
+	-1279152, -1274463, -1269773, -1265082, -1260391, -1255699, -1251007,
+	-1246314, -1241621, -1236928, -1232234, -1227540, -1222847, -1218153,
+	-1213459, -1208765, -1204072, -1199379, -1194686, -1189993, -1185301,
+	-1180609, -1175918, -1171228, -1166538, -1161849, -1157161, -1152474,
+	-1147787, -1143102, -1138418, -1133735, -1129053, -1124372, -1119693,
+	-1115015, -1110338, -1105663, -1100990, -1096318, -1091648, -1086979,
+	-1082313, -1077648, -1072985, -1068324, -1063665, -1059009, -1054354,
+	-1049702, -1045052, -1040404, -1035759, -1031116, -1026476, -1021838,
+	-1017203, -1012571, -1007941, -1003314, -998690, -994069, -989451,
+	-984836, -980224, -975615, -971009, -966407, -961808, -957212, -952620,
+	-948031, -943445, -938863, -934285, -929711, -925140, -920573, -916010,
+	-911451, -906895, -902344, -897797, -893254, -888715, -884180, -879650,
+	-875123, -870602, -866084, -861571, -857063, -852559, -848060, -843565,
+	-839075, -834590, -830110, -825634, -821164, -816698, -812237, -807782,
+	-803331, -798886, -794446, -790011, -785581, -781157, -776738, -772325,
+	-767917, -763514, -759117, -754726, -750341, -745961, -741586, -737218,
+	-732855, -728499, -724148, -719803, -715464, -711132, -706805, -702484,
+	-698170, -693862, -689560, -685265, -680975, -676693, -672416, -668146,
+	-663883, -659626, -655376, -651132, -646895, -642665, -638442, -634225,
+	-630015, -625812, -621616, -617427, -613245, -609070, -604902, -600741,
+	-596587, -592440, -588301, -584169, -580044, -575926, -571816, -567713,
+	-563618, -559530, -555449, -551376, -547311, -543253, -539203, -535161,
+	-531126, -527099, -523080, -519069, -515065, -511069, -507082, -503102,
+	-499130, -495166, -491210, -487262, -483323, -479391, -475468, -471553,
+	-467646, -463747, -459857, -455974, -452101, -448235, -444378, -440530,
+	-436690, -432858, -429035, -425220, -421414, -417617, -413828, -410048,
+	-406276, -402514, -398760, -395014, -391278, -387550, -383832, -380122,
+	-376421, -372728, -369045, -365371, -361706, -358050, -354402, -350764,
+	-347135, -343515, -339904, -336303, -332710, -329127, -325553, -321988,
+	-318433, -314886, -311350, -307822, -304304, -300795, -297295, -293805,
+	-290325, -286854, -283392, -279940, -276497, -273064, -269641, -266227,
+	-262822, -259427, -256042, -252667, -249301, -245945, -242599, -239262,
+	-235935, -232618, -229310, -226013, -222725, -219447, -216179, -212920,
+	-209672, -206433, -203205, -199986, -196777, -193578, -190389, -187210,
+	-184041, -180882, -177733, -174594, -171466, -168347, -165238, -162139,
+	-159051, -155972, -152904, -149846, -146798, -143760, -140732, -137714,
+	-134707, -131710, -128723, -125746, -122779, -119823, -116877, -113941,
+	-111016, -108100, -105195, -102301, -99416, -96542, -93678, -90825,
+	-87982, -85149, -82327, -79515, -76713, -73922, -71141, -68370, -65610,
+	-62860, -60121, -57392, -54674, -51966, -49268, -46581, -43904, -41238,
+	-38582, -35936, -33301, -30677, -28063, -25459, -22866, -20283, -17711,
+	-15149, -12598, -10057, -7527, -5007, -2498, 0
+};
+
+const int32_t filter_data_P[32769] = {
+	0, -1248, -2498, -3752, -5007, -6266, -7527, -8790, -10057, -11325,
+	-12597, -13870, -15147, -16426, -17707, -18991, -20277, -21566, -22857,
+	-24151, -25447, -26746, -28047, -29350, -30656, -31964, -33275, -34588,
+	-35903, -37221, -38541, -39863, -41188, -42515, -43844, -45176, -46510,
+	-47846, -49184, -50525, -51868, -53213, -54560, -55910, -57261, -58615,
+	-59971, -61330, -62690, -64053, -65417, -66784, -68153, -69524, -70897,
+	-72272, -73649, -75028, -76410, -77793, -79178, -80565, -81955, -83346,
+	-84739, -86134, -87531, -88930, -90331, -91734, -93139, -94545, -95954,
+	-97364, -98776, -100190, -101606, -103024, -104443, -105864, -107287,
+	-108712, -110138, -111567, -112997, -114428, -115861, -117296, -118733,
+	-120171, -121611, -123053, -124496, -125940, -127387, -128835, -130284,
+	-131735, -133188, -134642, -136097, -137554, -139013, -140473, -141934,
+	-143397, -144861, -146327, -147794, -149262, -150732, -152203, -153676,
+	-155150, -156625, -158101, -159579, -161058, -162538, -164020, -165503,
+	-166986, -168472, -169958, -171445, -172934, -174424, -175915, -177407,
+	-178900, -180394, -181889, -183385, -184883, -186381, -187880, -189381,
+	-190882, -192384, -193888, -195392, -196897, -198403, -199910, -201417,
+	-202926, -204435, -205945, -207456, -208968, -210481, -211994, -213508,
+	-215023, -216539, -218055, -219572, -221090, -222608, -224127, -225646,
+	-227166, -228687, -230208, -231730, -233253, -234776, -236299, -237823,
+	-239347, -240872, -242397, -243923, -245449, -246976, -248503, -250030,
+	-251557, -253085, -254614, -256142, -257671, -259200, -260729, -262259,
+	-263789, -265319, -266849, -268379, -269910, -271440, -272971, -274502,
+	-276033, -277564, -279094, -280625, -282157, -283688, -285218, -286749,
+	-288280, -289811, -291342, -292872, -294403, -295933, -297463, -298993,
+	-300523, -302053, -303582, -305111, -306640, -308169, -309697, -311225,
+	-312752, -314280, -315806, -317333, -318859, -320385, -321910, -323435,
+	-324959, -326483, -328006, -329528, -331051, -332572, -334093, -335614,
+	-337133, -338652, -340171, -341689, -343206, -344722, -346238, -347753,
+	-349267, -350780, -352293, -353804, -355315, -356825, -358334, -359843,
+	-361350, -362856, -364362, -365866, -367370, -368872, -370374, -371874,
+	-373373, -374872, -376369, -377865, -379360, -380853, -382346, -383837,
+	-385327, -386816, -388304, -389790, -391275, -392759, -394242, -395723,
+	-397203, -398681, -400158, -401633, -403107, -404580, -406051, -407521,
+	-408989, -410456, -411921, -413384, -414846, -416306, -417765, -419222,
+	-420677, -422131, -423583, -425033, -426482, -427928, -429373, -430816,
+	-432258, -433697, -435135, -436570, -438004, -439436, -440866, -442294,
+	-443720, -445144, -446566, -447986, -449404, -450820, -452234, -453645,
+	-455055, -456462, -457867, -459270, -460671, -462069, -463466, -464860,
+	-466251, -467641, -469028, -470413, -471795, -473175, -474553, -475928,
+	-477300, -478671, -480038, -481404, -482766, -484127, -485484, -486839,
+	-488192, -489542, -490889, -492233, -493575, -494914, -496251, -497584,
+	-498915, -500243, -501569, -502891, -504211, -505528, -506842, -508153,
+	-509461, -510766, -512068, -513367, -514664, -515957, -517247, -518534,
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+	-618955, -620057, -621151, -622237, -623314, -624384, -625445, -626497,
+	-627542, -628578, -629606, -630626, -631637, -632641, -633636, -634623,
+	-635601, -636572, -637534, -638488, -639434, -640371, -641301, -642222,
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+	-650144, -650983, -651814, -652637, -653452, -654259, -655057, -655848,
+	-656630, -657404, -658171, -658929, -659678, -660420, -661154, -661880,
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+	-682708, -682299, -681883, -681461, -681033, -680598, -680157, -679709,
+	-679255, -678795, -678329, -677856, -677377, -676892, -676401, -675904,
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+	-656124, -655426, -654722, -654013, -653298, -652578, -651852, -651120,
+	-650383, -649641, -648893, -648140, -647381, -646617, -645848, -645073,
+	-644293, -643507, -642716, -641920, -641118, -640312, -639500, -638682,
+	-637860, -637032, -636200, -635362, -634519, -633670, -632817, -631959,
+	-631095, -630226, -629353, -628474, -627591, -626702, -625809, -624910,
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+	-600898, -599877, -598852, -597823, -596790, -595752, -594709, -593663,
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+	-584050, -582961, -581868, -580771, -579670, -578564, -577455, -576341,
+	-575224, -574102, -572977, -571848, -570714, -569577, -568436, -567291,
+	-566142, -564989, -563833, -562672, -561508, -560340, -559169, -557993,
+	-556814, -555631, -554445, -553255, -552061, -550864, -549663, -548459,
+	-547251, -546039, -544824, -543606, -542384, -541158, -539930, -538697,
+	-537462, -536223, -534980, -533735, -532486, -531234, -529978, -528719,
+	-527457, -526192, -524924, -523652, -522377, -521099, -519818, -518534,
+	-517247, -515957, -514664, -513367, -512068, -510766, -509461, -508153,
+	-506842, -505528, -504211, -502891, -501569, -500243, -498915, -497584,
+	-496251, -494914, -493575, -492233, -490889, -489542, -488192, -486839,
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+	-474553, -473175, -471795, -470413, -469028, -467641, -466251, -464860,
+	-463466, -462069, -460671, -459270, -457867, -456462, -455055, -453645,
+	-452234, -450820, -449404, -447986, -446566, -445144, -443720, -442294,
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+	-429373, -427928, -426482, -425033, -423583, -422131, -420677, -419222,
+	-417765, -416306, -414846, -413384, -411921, -410456, -408989, -407521,
+	-406051, -404580, -403107, -401633, -400158, -398681, -397203, -395723,
+	-394242, -392759, -391275, -389790, -388304, -386816, -385327, -383837,
+	-382346, -380853, -379360, -377865, -376369, -374872, -373373, -371874,
+	-370374, -368872, -367370, -365866, -364362, -362856, -361350, -359843,
+	-358334, -356825, -355315, -353804, -352293, -350780, -349267, -347753,
+	-346238, -344722, -343206, -341689, -340171, -338652, -337133, -335614,
+	-334093, -332572, -331051, -329528, -328006, -326483, -324959, -323435,
+	-321910, -320385, -318859, -317333, -315806, -314280, -312752, -311225,
+	-309697, -308169, -306640, -305111, -303582, -302053, -300523, -298993,
+	-297463, -295933, -294403, -292872, -291342, -289811, -288280, -286749,
+	-285218, -283688, -282157, -280625, -279094, -277564, -276033, -274502,
+	-272971, -271440, -269910, -268379, -266849, -265319, -263789, -262259,
+	-260729, -259200, -257671, -256142, -254614, -253085, -251557, -250030,
+	-248503, -246976, -245449, -243923, -242397, -240872, -239347, -237823,
+	-236299, -234776, -233253, -231730, -230208, -228687, -227166, -225646,
+	-224127, -222608, -221090, -219572, -218055, -216539, -215023, -213508,
+	-211994, -210481, -208968, -207456, -205945, -204435, -202926, -201417,
+	-199910, -198403, -196897, -195392, -193888, -192384, -190882, -189381,
+	-187880, -186381, -184883, -183385, -181889, -180394, -178900, -177407,
+	-175915, -174424, -172934, -171445, -169958, -168472, -166986, -165503,
+	-164020, -162538, -161058, -159579, -158101, -156625, -155150, -153676,
+	-152203, -150732, -149262, -147794, -146327, -144861, -143397, -141934,
+	-140473, -139013, -137554, -136097, -134642, -133188, -131735, -130284,
+	-128835, -127387, -125940, -124496, -123053, -121611, -120171, -118733,
+	-117296, -115861, -114428, -112997, -111567, -110138, -108712, -107287,
+	-105864, -104443, -103024, -101606, -100190, -98776, -97364, -95954,
+	-94545, -93139, -91734, -90331, -88930, -87531, -86134, -84739, -83346,
+	-81955, -80565, -79178, -77793, -76410, -75028, -73649, -72272, -70897,
+	-69524, -68153, -66784, -65417, -64053, -62690, -61330, -59971, -58615,
+	-57261, -55910, -54560, -53213, -51868, -50525, -49184, -47846, -46510,
+	-45176, -43844, -42515, -41188, -39863, -38541, -37221, -35903, -34588,
+	-33275, -31964, -30656, -29350, -28047, -26746, -25447, -24151, -22857,
+	-21566, -20277, -18991, -17707, -16426, -15147, -13870, -12597, -11325,
+	-10057, -8790, -7527, -6266, -5007, -3752, -2498, -1248, 0
+};
diff --git a/usr/src/uts/common/io/audio/impl/audio_format.c b/usr/src/uts/common/io/audio/impl/audio_format.c
new file mode 100644
index 0000000000..b7385e336c
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_format.c
@@ -0,0 +1,894 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: Audio format conversion routines used by audio.c
+ */
+
+#include <sys/types.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/g711.h>
+
+#include "audio_impl.h"
+#include "audio_grc3.h"
+
+
+/*
+ * Note: In the function below, the division by the number of channels is
+ * probably fairly expensive.  It turns out that we usually deal with stereo
+ * or mono data, so perhaps it would be useful to build custom versions of
+ * this function that only dealt with stereo or mono.
+ */
+static int
+do_src(audio_stream_t *sp, void *p1, void *p2, int len, int nchan)
+{
+	int ch, size;
+
+	/*
+	 * Note that we presume that we are doing sample rate
+	 * conversions on AUDIO_FORMAT_S24_NE, which means that have 4
+	 * byte and 32-bit samples.
+	 */
+	size = sp->s_cnv_max / 4;		/* sample size is 4 */
+	size /= nchan;
+
+	for (ch = 0; ch < nchan; ch++) {
+		grc3_convert(sp->s_src_state[ch], sp->s_src_quality,
+		    p1, p2, len, size, nchan, ch);
+	}
+	return (((grc3state_t *)sp->s_src_state[0])->outsz);
+}
+
+static void
+setup_src(audio_stream_t *sp, int srate, int trate, int sch, int tch)
+{
+	int ch, nch;
+
+	nch = min(sch, tch);
+
+	ASSERT(nch <= AUDIO_MAX_CHANNELS);
+
+	if (sp->s_src_quality < 1)
+		sp->s_src_quality = 1;
+	if (sp->s_src_quality > 5)
+		sp->s_src_quality = 5;
+
+	for (ch = 0; ch < nch; ch++) {
+		grc3_reset(sp->s_src_state[ch]);
+		grc3_setup(sp->s_src_state[ch], srate, trate);
+	}
+}
+
+static int
+cnv_srconly(audio_stream_t *sp, int len)
+{
+	void *src = sp->s_cnv_src;
+	void *dst = sp->s_cnv_dst;
+
+	/*
+	 * We must be using 24-bit native signed.
+	 */
+	len = do_src(sp, src, dst, len, sp->s_cnv_src_nchan);
+
+	sp->s_cnv_src = dst;
+	sp->s_cnv_dst = src;
+
+	return (len);
+}
+
+static int
+cnv_s24oe(audio_stream_t *sp, int len)
+{
+	/*
+	 * Endian switch works in both directions.  We do it in place.
+	 */
+	int32_t *src = sp->s_cnv_src;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--) {
+		*src = ddi_swap32(*src);
+		src++;
+	}
+
+	return (len);
+}
+
+
+static int
+cnv_from_s8(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	int8_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (*src++) << 16;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_u8(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint8_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (int8_t)((*src++) ^ 0x80) << 16;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_ulaw(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint8_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--) {
+		*dst++ = _8ulaw2linear16[(*src++)] << 8;
+	}
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_alaw(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint8_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--) {
+		*dst++ = _8alaw2linear16[(*src++)] << 8;
+	}
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_s16ne(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	int16_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (*src++) << 8;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_s16oe(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	int16_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (int16_t)(ddi_swap16(*src++)) << 8;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_u16ne(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint16_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (int16_t)((*src++) ^ 0x8000) << 8;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_u16oe(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint16_t *src = s;
+	int32_t *dst = d;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--)
+		*dst++ = (int16_t)(ddi_swap16((*src++) ^ 0x8000)) << 8;
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_s24p(audio_stream_t *sp, int len)
+{
+	void *s = sp->s_cnv_src;
+	void *d = sp->s_cnv_dst;
+	uint8_t *src = s;
+	int32_t *dst = d;
+	int32_t tmp;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--) {
+		/* NB: this is a little endian format */
+		tmp = (*src++);
+		tmp |= (*src++) << 8;
+		tmp |= (*src++) << 16;
+		*dst++ = tmp;
+	}
+
+	sp->s_cnv_src = d;
+	sp->s_cnv_dst = s;
+	return (len);
+}
+
+static int
+cnv_from_s32ne(audio_stream_t *sp, int len)
+{
+	/* 32-bit conversions can be done in place */
+	int32_t *src = sp->s_cnv_src;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--, src++)
+		*src = *src >> 8;
+
+	return (len);
+}
+
+static int
+cnv_from_s32oe(audio_stream_t *sp, int len)
+{
+	/* 32-bit conversions can be done in place */
+	int32_t *src = sp->s_cnv_src;
+
+	for (int i = len * sp->s_cnv_src_nchan; i; i--, src++)
+		*src = (int32_t)(ddi_swap32(*src)) >> 8;
+
+	return (len);
+}
+
+/*
+ * NB: All the destination format conversions use the same or fewer
+ * bytes as the 24-bit unpacked (32-bits used per sample), so we can
+ * convert all of them in place.
+ */
+
+static int
+cnv_to_u8(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint8_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = (*src++ >> 16) ^ 0x80;
+
+	return (len);
+}
+
+static int
+cnv_to_s8(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	int8_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = *src++ >> 16;
+
+	return (len);
+}
+
+static int
+cnv_to_ulaw(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint8_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
+		int idx = *src++;
+		idx >>= 10;
+		idx += G711_ULAW_MIDPOINT;
+		idx &= 0x3fff;	/* safety precaution */
+		*dst++ = _14linear2ulaw8[idx];
+	}
+
+	return (len);
+}
+
+static int
+cnv_to_alaw(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint8_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
+		int idx = *src++;
+		idx >>= 11;
+		idx += G711_ALAW_MIDPOINT;
+		idx &= 0x1fff;	/* safety precaution */
+		*dst++ = _13linear2alaw8[idx];
+	}
+
+	return (len);
+}
+
+static int
+cnv_to_s16ne(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	int16_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = *src++ >> 8;
+
+	return (len);
+}
+
+static int
+cnv_to_s16oe(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	int16_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = ddi_swap16(*src++ >> 8);
+
+	return (len);
+}
+
+static int
+cnv_to_u16ne(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint16_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = (*src++ >> 8) ^ 0x8000;
+
+	return (len);
+}
+
+static int
+cnv_to_u16oe(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint16_t *dst = (void *)src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--)
+		*dst++ = ddi_swap16(*src++ >> 8) ^ 0x8000;
+
+	return (len);
+}
+
+static int
+cnv_to_s24p(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+	uint8_t *dst = (void *)src;
+	int32_t d;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
+		/* NB: this is a little endian format */
+		d = *src++;
+		*dst++ = d & 0xff;
+		*dst++ = (d >> 8) & 0xff;
+		*dst++ = (d >> 16) & 0xff;
+	}
+
+	return (len);
+}
+
+static int
+cnv_to_s32ne(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--, src++)
+		*src = *src << 8;
+
+	return (len);
+}
+
+static int
+cnv_to_s32oe(audio_stream_t *sp, int len)
+{
+	int32_t *src = sp->s_cnv_src;
+
+	for (int i = len * sp->s_cnv_dst_nchan; i; i--, src++)
+		*src = ddi_swap32(*src << 8);
+
+	return (len);
+}
+
+static int
+cnv_default(audio_stream_t *sp, int len)
+{
+	/*
+	 * Note that the formats were already preverified during
+	 * select_converter, to ensure that only supported formats are
+	 * used.
+	 */
+
+	/*
+	 * Convert samples to 24 bit (32 bit lsb aligned) if
+	 * necessary.
+	 */
+
+	switch (sp->s_cnv_src_format) {
+
+	case AUDIO_FORMAT_U8:
+		len = cnv_from_u8(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S8:
+		len = cnv_from_s8(sp, len);
+		break;
+
+	case AUDIO_FORMAT_ULAW:
+		len = cnv_from_ulaw(sp, len);
+		break;
+
+	case AUDIO_FORMAT_ALAW:
+		len = cnv_from_alaw(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S16_NE:
+		len = cnv_from_s16ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S16_OE:
+		len = cnv_from_s16oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_U16_NE:
+		len = cnv_from_u16ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_U16_OE:
+		len = cnv_from_u16oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S32_NE:
+		len = cnv_from_s32ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S32_OE:
+		len = cnv_from_s32oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S24_OE:
+		len = cnv_s24oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S24_PACKED:
+		len = cnv_from_s24p(sp, len);
+		break;
+	}
+
+	/*
+	 * If we aren't decreasing the number of channels, then do the
+	 * SRC now.  (We prefer to do SRC on the smaller number of channels.)
+	 */
+	if (sp->s_cnv_src_rate != sp->s_cnv_dst_rate &&
+	    sp->s_cnv_src_nchan <= sp->s_cnv_dst_nchan) {
+		int32_t *src = sp->s_cnv_src;
+		int32_t *dst = sp->s_cnv_dst;
+
+		len = do_src(sp, src, dst, len, sp->s_cnv_src_nchan);
+
+		sp->s_cnv_src = dst;
+		sp->s_cnv_dst = src;
+	}
+
+	/*
+	 * Convert between mono and stereo
+	 */
+
+	if (sp->s_cnv_src_nchan != sp->s_cnv_dst_nchan) {
+		int32_t *src = sp->s_cnv_src;
+		int32_t *dst = sp->s_cnv_dst;
+		int tc = sp->s_cnv_dst_nchan;
+		int sc = sp->s_cnv_src_nchan;
+		int nc;
+		int i;
+
+		sp->s_cnv_src = dst;
+		sp->s_cnv_dst = src;
+
+		if (sc == 1) {
+			/*
+			 * Mono expansion.  We expand into the stereo
+			 * channel, and leave other channels silent.
+			 */
+			for (i = len; i; i--) {
+				*dst++ = *src;
+				*dst++ = *src++;
+				for (int j = tc - 2; j > 0; j--) {
+					*dst++ = 0;
+				}
+
+			}
+
+		} else if (sc == 2 && tc == 1) {
+			/*
+			 * Stereo -> mono.  We do stereo separately to make
+			 * the division fast (div by const 2 is just shift).
+			 */
+			for (i = len; i; i--) {
+				/*
+				 * Take just the left channel sample,
+				 * discard the right channel.
+				 */
+				*dst++ = *src++;	/* left */
+				src++;			/* right */
+			}
+		} else {
+			/*
+			 * Multi channel conversions.  We just copy the
+			 * minimum number of channels.
+			 */
+
+			/* Calculate number of frames */
+
+			nc = min(sc, tc);
+
+			/* Clear destination */
+			bzero(dst, (len * tc * sizeof (int32_t)));
+
+			for (i = len; i; i--) {
+				int c;
+
+				for (c = 0; c < nc; c++)
+					dst[c] = src[c];
+
+				src += sc;
+				dst += tc;
+			}
+		}
+	}
+
+	/*
+	 * If we didn't do SRC pre-conversion, then do it now.
+	 */
+	if (sp->s_cnv_src_rate != sp->s_cnv_dst_rate &&
+	    sp->s_cnv_src_nchan > sp->s_cnv_dst_nchan) {
+
+		int32_t *src = sp->s_cnv_src;
+		int32_t *dst = sp->s_cnv_dst;
+
+		len = do_src(sp, src, dst, len, sp->s_cnv_dst_nchan);
+
+		sp->s_cnv_src = dst;
+		sp->s_cnv_dst = src;
+	}
+
+	/*
+	 * Finally convert samples from internal 24 bit format to target format
+	 */
+
+	switch (sp->s_cnv_dst_format) {
+	case AUDIO_FORMAT_U8:
+		len = cnv_to_u8(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S8:
+		len = cnv_to_s8(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S16_NE:
+		len = cnv_to_s16ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S16_OE:
+		len = cnv_to_s16oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_U16_NE:
+		len = cnv_to_u16ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_U16_OE:
+		len = cnv_to_u16oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S24_OE:
+		len = cnv_s24oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S24_PACKED:
+		len = cnv_to_s24p(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S32_NE:
+		len = cnv_to_s32ne(sp, len);
+		break;
+
+	case AUDIO_FORMAT_S32_OE:
+		len = cnv_to_s32oe(sp, len);
+		break;
+
+	case AUDIO_FORMAT_ULAW:
+		len = cnv_to_ulaw(sp, len);
+		break;
+
+	case AUDIO_FORMAT_ALAW:
+		len = cnv_to_alaw(sp, len);
+		break;
+	}
+
+	return (len);
+}
+
+static const struct audio_format_info {
+	unsigned		format;
+	int			sampsize;
+	audio_cnv_func_t	from;
+	audio_cnv_func_t	to;
+} audio_format_info[] = {
+	{ AUDIO_FORMAT_S8,	1,	cnv_from_s8,		cnv_to_s8 },
+	{ AUDIO_FORMAT_U8,	1,	cnv_from_u8,		cnv_to_u8 },
+	{ AUDIO_FORMAT_ULAW,	1,	cnv_from_ulaw,		cnv_to_ulaw },
+	{ AUDIO_FORMAT_ALAW,	1,	cnv_from_alaw,		cnv_to_alaw },
+	{ AUDIO_FORMAT_S16_NE,	2,	cnv_from_s16ne,		cnv_to_s16ne },
+	{ AUDIO_FORMAT_S16_OE,	2,	cnv_from_s16oe,		cnv_to_s16oe },
+	{ AUDIO_FORMAT_U16_NE,	2,	cnv_from_u16ne,		cnv_to_u16ne },
+	{ AUDIO_FORMAT_U16_OE,	2,	cnv_from_u16oe,		cnv_to_u16oe },
+	{ AUDIO_FORMAT_S32_NE,	4,	cnv_from_s32ne,		cnv_to_s32ne },
+	{ AUDIO_FORMAT_S32_OE,	4,	cnv_from_s32oe,		cnv_to_s32oe },
+
+	/* 24-bit formats are "special" */
+	{ AUDIO_FORMAT_S24_NE,	4,	NULL,			NULL },
+	{ AUDIO_FORMAT_S24_OE,	4,	cnv_s24oe,		cnv_s24oe },
+	{ AUDIO_FORMAT_S24_PACKED, 3,	cnv_from_s24p,		cnv_to_s24p },
+
+	/* sentinel */
+	{ AUDIO_FORMAT_NONE,	0,	NULL,			NULL }
+};
+
+int
+auimpl_format_setup(audio_stream_t *sp, audio_parms_t *uparms)
+{
+	audio_parms_t			*source = &sp->s_cnv_src_parms;
+	audio_parms_t			*target = &sp->s_cnv_dst_parms;
+	audio_cnv_func_t		converter = NULL;
+	const struct audio_format_info	*info;
+	int				expand = AUDIO_UNIT_EXPAND;
+	unsigned			cnv_sampsz = sizeof (uint32_t);
+	unsigned			cnv_max;
+
+	if (sp == &sp->s_client->c_ostream) {
+		source = uparms;
+	} else {
+		target = uparms;
+	}
+
+	ASSERT(mutex_owned(&sp->s_lock));
+
+	/*
+	 * At least one of the source or target are S24_NE.
+	 *
+	 * If we have a signed/native endian format, then pick an
+	 * optimized converter.  While at it, ensure that a valid
+	 * format is selected.
+	 */
+	if (source->p_format != AUDIO_FORMAT_S24_NE) {
+		for (info = &audio_format_info[0]; info->sampsize; info++) {
+			if (source->p_format == info->format) {
+				converter = info->from;
+				expand *= sizeof (int32_t);
+				expand /= info->sampsize;
+				/* save source frame size */
+				cnv_sampsz = info->sampsize;
+				break;
+			}
+		}
+		if (info->format == AUDIO_FORMAT_NONE) {
+			audio_dev_warn(sp->s_client->c_dev,
+			    "invalid source format selected");
+			return (EINVAL);
+		}
+
+	} else if (target->p_format != AUDIO_FORMAT_S24_NE) {
+		for (info = &audio_format_info[0]; info->sampsize; info++) {
+			if (target->p_format == info->format) {
+				converter = info->to;
+				expand *= info->sampsize;
+				expand /= sizeof (int32_t);
+				break;
+			}
+		}
+		if (info->format == AUDIO_FORMAT_NONE) {
+			audio_dev_warn(sp->s_client->c_dev,
+			    "invalid target format selected");
+			return (EINVAL);
+		}
+	}
+
+	if (source->p_nchan != target->p_nchan) {
+		/*
+		 * if channels need conversion, then we must use the
+		 * default.
+		 */
+		converter = cnv_default;
+		expand *= target->p_nchan;
+		expand /= source->p_nchan;
+	}
+
+	if (source->p_rate != target->p_rate) {
+		/*
+		 * We need SRC; if we can avoid data conversion, do so.
+		 */
+		setup_src(sp, source->p_rate, target->p_rate,
+		    source->p_nchan, target->p_nchan);
+
+		converter = (converter == NULL) ? cnv_srconly : cnv_default;
+
+		expand *= target->p_rate;
+		expand /= source->p_rate;
+	}
+
+	ASSERT(sp->s_engine);
+	ASSERT(sp->s_engine->e_intrs);
+
+	/*
+	 * Figure out the size of the conversion buffer we need.  We
+	 * assume room for two full source fragments, which ought to
+	 * be enough, even with rounding errors.
+	 */
+	cnv_max = 2 * (source->p_rate / sp->s_engine->e_intrs) *
+	    cnv_sampsz * source->p_nchan;
+
+	/*
+	 * If the conversion will cause us to expand fragments, then
+	 * we need to increase cnv_max.  Scale by AUDIO_UNIT_EXPAND to
+	 * avoid rouding errors or losing bits when doing reducing
+	 * conversions.
+	 */
+	if (expand > AUDIO_UNIT_EXPAND) {
+		cnv_max *= expand;
+		cnv_max /= AUDIO_UNIT_EXPAND;
+	}
+
+	/*
+	 * Now we need to allocate space.
+	 */
+	if (sp->s_cnv_max < cnv_max) {
+		uint32_t *buf0, *buf1;
+
+		buf0 = kmem_alloc(cnv_max, KM_NOSLEEP);
+		buf1 = kmem_alloc(cnv_max, KM_NOSLEEP);
+		if ((buf0 == NULL) || (buf1 == NULL)) {
+			audio_dev_warn(sp->s_client->c_dev,
+			    "failed to allocate audio conversion buffer "
+			    "(%u bytes)", cnv_max);
+			if (buf0)
+				kmem_free(buf0, cnv_max);
+			if (buf1)
+				kmem_free(buf1, cnv_max);
+			return (ENOMEM);
+		}
+
+		if (sp->s_cnv_buf0)
+			kmem_free(sp->s_cnv_buf0, sp->s_cnv_max);
+		if (sp->s_cnv_buf1)
+			kmem_free(sp->s_cnv_buf1, sp->s_cnv_max);
+
+		sp->s_cnv_buf0 = buf0;
+		sp->s_cnv_buf1 = buf1;
+		sp->s_cnv_max = cnv_max;
+	}
+
+	/*
+	 * NB: From here on, we must not fail.
+	 */
+
+	/*
+	 * Configure default fragment setup.
+	 */
+	for (info = &audio_format_info[0]; info->sampsize; info++) {
+		if (uparms->p_format == info->format) {
+			break;
+		}
+	}
+	ASSERT(info->sampsize);
+
+	sp->s_framesz = info->sampsize * uparms->p_nchan;
+	sp->s_fragfr = (uparms->p_rate / sp->s_engine->e_intrs);
+	sp->s_fragbytes = sp->s_fragfr * sp->s_framesz;
+	sp->s_nfrags = sp->s_allocsz / sp->s_fragbytes;
+	sp->s_nframes = sp->s_nfrags * sp->s_fragfr;
+	sp->s_nbytes = sp->s_nframes * sp->s_framesz;
+	*sp->s_user_parms = *uparms;
+	sp->s_converter = converter;
+
+	/*
+	 * Ensure that we toss any stale data -- probably wrong format.
+	 * Note that as a consequence of this, all of the offsets and
+	 * counters get reset.  Clients should not rely on these values
+	 * being preserved when changing formats.
+	 *
+	 * Its critical that we reset the indices, in particular,
+	 * because not only will the data be the wrong format, but the
+	 * indices themselves are quite possibly going to be invalid.
+	 */
+	sp->s_cnv_cnt = 0;
+	sp->s_tail = sp->s_head = 0;
+	sp->s_tidx = sp->s_hidx = 0;
+
+	return (0);
+}
+
+int
+auimpl_format_alloc(audio_stream_t *sp)
+{
+	int	i;
+
+	ASSERT(mutex_owned(&sp->s_lock));
+	for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
+		sp->s_src_state[i] =
+		    kmem_zalloc(sizeof (grc3state_t), KM_NOSLEEP);
+		if (sp->s_src_state[i] == NULL) {
+			audio_dev_warn(sp->s_client->c_dev,
+			    "unable to allocate SRC state structures");
+			return (ENOMEM);
+		}
+	}
+	return (0);
+}
+
+void
+auimpl_format_free(audio_stream_t *sp)
+{
+	int	i;
+
+	for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
+		if (sp->s_src_state[i] != NULL) {
+			kmem_free(sp->s_src_state[i], sizeof (grc3state_t));
+			sp->s_src_state[i] = NULL;
+		}
+	}
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_grc3.c b/usr/src/uts/common/io/audio/impl/audio_grc3.c
new file mode 100644
index 0000000000..3d42ab8425
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_grc3.c
@@ -0,0 +1,388 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: GRC3 Sample Rate Converter
+ *
+ * GRC library version 3.1
+ */
+
+#include <sys/types.h>
+#include "audio_grc3.h"
+
+extern const int32_t filter_data_L[];
+extern const int32_t filter_data_M[];
+extern const int32_t filter_data_H[];
+extern const int32_t filter_data_P[];
+
+#define	filter_data_HX  filter_data_H
+#define	filter_data_PX  filter_data_P
+
+static int32_t
+_muldivu64(uint32_t a, uint32_t val1, uint32_t val2)
+{
+	uint64_t v = ((uint64_t)a) * val1 / val2;
+	return ((uint32_t)(v));
+}
+
+
+static int32_t
+_grc_sat6(int32_t a, int32_t b)
+{
+	int64_t v = ((int64_t)a) * b + (1 << 5);
+	return ((int32_t)(v >> 6));
+}
+
+static int32_t
+_grc_sat31(int32_t a, int32_t b)
+{
+	int64_t v = ((int64_t)a) * b + (1 << 30);
+	return ((int32_t)(v >> 31));
+}
+
+
+#define	DEFINE_FILTER(T)						\
+static int32_t								\
+_filt31_##T(int32_t a, int32_t idx)					\
+{									\
+	int64_t v = ((int64_t)a) * filter_data_##T[idx >> 15];		\
+	return ((int32_t)(v >> 31));					\
+}
+
+#define	DEFINE_FILTER_HQ(T)						\
+static int32_t								\
+_filt31_##T(int32_t a, int32_t idx)					\
+{									\
+	int32_t idx2 = idx>>15;						\
+	int64_t v = ((int64_t)a) *					\
+									\
+	    (filter_data_##T[idx2] +					\
+	    (((int64_t)(idx & 32767)) * (filter_data_##T[idx2 + 1] -	\
+	    filter_data_##T[idx2]) >> 15));				\
+	return ((int32_t)(v>>31));					\
+}
+
+
+DEFINE_FILTER(L)
+DEFINE_FILTER(M)
+DEFINE_FILTER(H)
+DEFINE_FILTER_HQ(HX)
+DEFINE_FILTER(P)
+DEFINE_FILTER_HQ(PX)
+
+#define	DEFINE_CONVD(T, SZ)						\
+static int32_t								\
+_conv31d_##T(int32_t *history,  uint32_t filter, uint32_t incv)		\
+{									\
+	int32_t accum = 0;						\
+									\
+	filter = (1024 << 15) - filter;					\
+									\
+	while (filter < ((uint32_t)(SZ << 15))) {			\
+		accum += _filt31_##T(*history, filter);			\
+		filter += incv;						\
+		history--;						\
+	}								\
+									\
+	return (accum);							\
+}
+
+DEFINE_CONVD(L, 4096)
+DEFINE_CONVD(M, 8192)
+DEFINE_CONVD(H, 16384)
+DEFINE_CONVD(HX, 16384)
+DEFINE_CONVD(P, 32768)
+DEFINE_CONVD(PX, 32768)
+
+static int32_t
+_conv31_L(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+#define	ITERATION(p)				\
+	accum += _filt31_##p(*history, filter);	\
+	filter += (1024 << 15);			\
+	history--
+
+	ITERATION(L); ITERATION(L); ITERATION(L); ITERATION(L);
+	return (accum);
+}
+
+
+static int32_t
+_conv31_M(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+	ITERATION(M); ITERATION(M); ITERATION(M); ITERATION(M);
+	ITERATION(M); ITERATION(M); ITERATION(M); ITERATION(M);
+	return (accum);
+}
+
+static int32_t
+_conv31_H(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+	ITERATION(H); ITERATION(H); ITERATION(H); ITERATION(H);
+	ITERATION(H); ITERATION(H); ITERATION(H); ITERATION(H);
+	ITERATION(H); ITERATION(H); ITERATION(H); ITERATION(H);
+	ITERATION(H); ITERATION(H); ITERATION(H); ITERATION(H);
+	return (accum);
+}
+
+static int32_t
+_conv31_HX(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+	ITERATION(HX); ITERATION(HX); ITERATION(HX); ITERATION(HX);
+	ITERATION(HX); ITERATION(HX); ITERATION(HX); ITERATION(HX);
+	ITERATION(HX); ITERATION(HX); ITERATION(HX); ITERATION(HX);
+	ITERATION(HX); ITERATION(HX); ITERATION(HX); ITERATION(HX);
+	return (accum);
+}
+
+static int32_t
+_conv31_P(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	ITERATION(P); ITERATION(P); ITERATION(P); ITERATION(P);
+	return (accum);
+}
+
+static int32_t
+_conv31_PX(int32_t *history, uint32_t filter)
+{
+	int32_t accum = 0;
+
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	ITERATION(PX); ITERATION(PX); ITERATION(PX); ITERATION(PX);
+	return (accum);
+}
+
+#define	GRC3_RESAMPLE(QUAL)						\
+static void								\
+grc3_upsample_##QUAL(grc3state_t *grc, const int32_t *src,		\
+    int32_t *dst, uint32_t sz, uint32_t bufsz, int inc, int offset)	\
+{									\
+	int32_t ptr = grc->ptr;						\
+	int32_t srcrate = grc->srcrate;					\
+	int32_t dstrate = grc->dstrate;					\
+	int32_t *history = grc->historyptr;				\
+	int32_t filtfactor = grc->filtfactor;				\
+	uint32_t dstsz = 0;						\
+									\
+	src += offset;							\
+	dst += offset;							\
+									\
+	while (sz > 0) {						\
+		while (ptr < dstrate) {					\
+			if (dstsz >= bufsz)				\
+				goto endloop;				\
+			dst[0] = (_conv31_##QUAL(history,		\
+				_grc_sat6(ptr, filtfactor)));		\
+			ptr += srcrate;					\
+			dst += inc;					\
+			dstsz++;					\
+		}							\
+									\
+		history++;						\
+		if (history >= (grc->history + GRC3_MAXHISTORY * 2))	\
+			history -= GRC3_MAXHISTORY;			\
+									\
+		history[0] = history[-GRC3_MAXHISTORY] = (*src);	\
+									\
+		ptr -= dstrate;						\
+									\
+		sz--;							\
+		src += inc;						\
+	}								\
+endloop:								\
+									\
+	grc->ptr = ptr;							\
+	grc->historyptr = history;					\
+	grc->outsz = dstsz;						\
+}									\
+									\
+static void								\
+grc3_dnsample_##QUAL(grc3state_t *grc, const int32_t *src,		\
+    int32_t *dst, uint32_t sz, uint32_t bufsz, int inc, int offset)	\
+{									\
+	int32_t ptr = grc->ptr;						\
+	int32_t srcrate = grc->srcrate;					\
+	int32_t dstrate = grc->dstrate;					\
+	int32_t sat = grc->sat;						\
+	int32_t *history = grc->historyptr;				\
+	int32_t filtfactor = grc->filtfactor;				\
+	uint32_t dstsz = 0;						\
+									\
+	src += offset;							\
+	dst += offset;							\
+									\
+	while (sz > 0) {						\
+		while (ptr >= srcrate) {				\
+			if (dstsz >= bufsz)				\
+				goto endloop;				\
+			ptr -= srcrate;					\
+			dst[0] = (_conv31d_##QUAL(history,		\
+			    _grc_sat6(ptr, filtfactor),			\
+				grc->ptr_incv));			\
+			dst += inc;					\
+			dstsz++;					\
+		}							\
+									\
+		history++;						\
+		if (history >= (grc->history + GRC3_MAXHISTORY * 2))	\
+			history -= GRC3_MAXHISTORY;			\
+									\
+		/*							\
+		 * TODO: for better quality multiplier is worth moving	\
+		 * to output cascade					\
+		 */							\
+		history[0] = history[-GRC3_MAXHISTORY] =		\
+		    _grc_sat31((*src), sat);				\
+									\
+		ptr += dstrate;						\
+									\
+		sz--;							\
+		src += inc;						\
+	}								\
+endloop:								\
+									\
+	grc->ptr = ptr;							\
+	grc->historyptr = history;					\
+	grc->outsz = dstsz;						\
+}									\
+									\
+static void								\
+grc3_resample_##QUAL(grc3state_t *grc, const void *src, void *dst,	\
+    uint32_t sz, uint32_t bufsz, int inc, int  offset)			\
+{									\
+	if (grc->srcrate <= grc->dstrate)				\
+		grc3_upsample_##QUAL(grc, src, dst, sz,			\
+		    bufsz, inc, offset);				\
+	else								\
+		grc3_dnsample_##QUAL(grc, src, dst, sz,			\
+		    bufsz, inc, offset);				\
+}
+
+GRC3_RESAMPLE(L)
+GRC3_RESAMPLE(M)
+GRC3_RESAMPLE(H)
+GRC3_RESAMPLE(HX)
+GRC3_RESAMPLE(P)
+GRC3_RESAMPLE(PX)
+
+/*
+ * For performance reasons, we only support 24-bit SRC.
+ */
+void
+grc3_convert(grc3state_t *grc, int quality, const void *src,
+    void *dst, int sz, int bufsz, int inc, int offset)
+{
+
+	switch (quality) {
+	default:
+	case 0:
+	case 1:
+		grc3_resample_L(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	case 2:
+		grc3_resample_M(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	case 3:
+		grc3_resample_H(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	case 4:
+		grc3_resample_HX(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	case 5:
+		grc3_resample_P(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	case 6:
+		grc3_resample_PX(grc, src, dst, sz, bufsz, inc, offset);
+		break;
+	}
+}
+
+void
+grc3_reset(grc3state_t *grc)
+{
+	int32_t t;
+	grc->ptr = 0;
+	grc->historyptr = grc->history + GRC3_MAXHISTORY;
+
+	for (t = 0; t < GRC3_MAXHISTORY * 2; t++)
+		grc->history[t] = 0;
+}
+
+static void
+grc3_setup_up(grc3state_t *grc, uint32_t fromRate, uint32_t toRate)
+{
+	grc->srcrate = fromRate;
+	grc->dstrate = toRate;
+	grc->filtfactor = 0x80000000U / toRate;
+}
+
+static void
+grc3_setup_dn(grc3state_t *grc, uint32_t fromRate, uint32_t toRate)
+{
+	grc->srcrate = fromRate;
+	grc->dstrate = toRate;
+	grc->filtfactor = 0x80000000U / fromRate;
+	grc->ptr_incv = _muldivu64(1024 << 15, toRate, fromRate);
+	grc->sat = _muldivu64(0x80000000U, toRate, fromRate);
+}
+
+void
+grc3_setup(grc3state_t *grc, uint32_t fromRate, uint32_t toRate)
+{
+	while ((!(fromRate & 1)) && (!(toRate & 1)) && (fromRate > 0)) {
+		fromRate >>= 1;
+		toRate >>= 1;
+	}
+
+	if (fromRate <= toRate)
+		grc3_setup_up(grc, fromRate, toRate);
+	else
+		grc3_setup_dn(grc, fromRate, toRate);
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_grc3.h b/usr/src/uts/common/io/audio/impl/audio_grc3.h
new file mode 100644
index 0000000000..0003dc1574
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_grc3.h
@@ -0,0 +1,275 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: GRC library version 3.1 internal definitions
+ *
+ * GRC3 is a high quality sample rate conversion module that uses fixed point
+ * arithmetic.
+ */
+
+#ifndef AUDIO_GRC3_H
+#define	AUDIO_GRC3_H
+
+#define	GRC3_MAXHISTORY 4096
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+typedef struct grc3state {
+	uint32_t srcrate;
+	uint32_t dstrate;
+	uint32_t ptr;
+	uint32_t ptr_incv;
+
+	uint32_t sat;
+	uint32_t filtfactor;
+	int32_t *historyptr;
+	int32_t dummy_pad1;
+
+	int32_t history[GRC3_MAXHISTORY * 2];
+
+	uint32_t outsz;
+} grc3state_t;
+
+
+/* BEGIN CSTYLED */
+/*****************************************************************************
+
+    Tutorial on how to use GRC3 rate conversion
+    
+1.  First, you create an instance of grc3state_t for each channel. If you 
+    are working with stereo files - you will need 2 of such instances,
+    for quadro - 4.
+    
+    The instances may be allocated in either static or dynamic memory - that
+    makes no difference to the convertor. So, if your program has to process
+    one stereo stream, there's no reason why should you use malloc/free to 
+    allocate/deallocate structures. Also, in device drivers, you can 
+    use static variables as well:
+    
+	static grc3state_t grc[2]; // for two channels
+
+    
+2.  Before starting any conversion, grc3state_t instances should be initialized
+    properly, and you do this with grc3_setup function. Function itself does
+    not allocate additional memory or change anything except grc3state_t
+    structure, so this is thread safe, and you don't have to do additional
+    "deinitialization".
+    
+    If you are doing interleaved audio (stereo/quadro/whatever) conversion, 
+    you should do setup on each of the channels, and should have separate
+    instance of grc3state_t for each channel. As you will understand further,
+    such conversion is done separately. And now, the setup function:
+    
+	int grc3_setup( grc3state_t *grc, 
+		    uint32_t	fromRate,
+		uint32_t    toRate );
+	       
+	grc	  - pointer to grc3state_t instance
+	fromRate  - source sample rate
+	toRate	  - destination sample rate
+	
+    Note, that sample rates itself are not important - the important thing 
+    is ratio between those sample rates. So, for example, if you have to
+    convert from 24000Hz to 48000Hz, it's ok to write:
+    
+	grc3_setup( &grc[0], 240, 480 );
+    
+    Sometimes (in MIDI synths) it would be desired to use fractional sample
+    rates. For example, setup for conversion from 33100.78 to 48000 may look 
+    like this:
+    
+	grc3_setup( &grc[0], 3310078, 4800000);
+    
+    Note, that on stereo, GRC3 setup will look like this:
+    
+	static grc3state_t grc[2];
+    
+    // ...
+    
+	grc3_setup( &grc[0], 3310078, 4800000)
+        grc3_setup( &grc[1], 3310078, 4800000);	   
+    
+
+    Note, that you should not rely on grc3_setup's fast execution or any
+    execution timing. It may contain some massive arithmetic and even huge 
+    loops, so avoid putting grc3_setup to inner loops and calling in 
+    latency-dependent code.
+
+	
+3.  Next, before running a stream through grc3_convert function, you should
+    reset each of grc3state_t instance used:
+    
+	void grc3_reset(grc3state_t *grc);
+    
+    
+	grc	- pointer to GRC3 instance variable
+ 
+    So, for stereo, this appears to be:
+    
+	static grc3state_t grc[2]; 
+    
+    // ...
+    
+	grc3_reset( &grc[0] );	
+    grc3_reset( &grc[1] );
+	
+
+4.  Finally, doing conversion is easy:
+
+	void grc3_convert( grc3state_t *grc,
+			  int	       domain, 
+	      int	   quality, 
+			  const void  *src, 
+	      void	  *dst, 
+		  int	       maxInSize, 
+	      int	   maxOutSize, 
+	      int	   interleave,
+	      int	   offset );
+	      
+	      
+	grc	   - pointer to initialized grc3state_t instance; you
+		     can specify NULL to check whether a particular
+		 domain/quality pair is supported, check return value
+	
+	quality	   - quality to use for conversion, supported values are:
+	
+		     0 - D lowest quality (normally equals to low quality)
+		 1 - L	low quality    (spline interpolation)
+		 2 - M	medium quality (lagrange interpolation)
+		 3 - H	high quality   
+		 4 - HX high quality   (high quality with extra precision)
+		 5 - P	production quality
+
+		 6 - PX production quality (prod quality with extra precision)
+		     (PX is currently disabled because it causes a crash)
+	       
+	src	   - source audio buffer	  
+	
+	dst	   - destination audio buffer; 
+	
+	maxInSize  - size of input buffer (in samples per channel!)
+	
+	maxOutSize - size of output buffer (in samples per channel!)
+		     (will never overrun this size)
+	
+	interleave - interleave factor; for MONO or non-interleaved data
+		     it should be equal to 1;
+	     
+	     2 - STEREO interleaved audio
+	     4 - QUADRO interleaved audio
+	     
+	     So, basically, this parameter should be equal to number
+	     of interleaved channels
+	     
+	offset	   - number of interleaved channel currently processing, 
+		     starting from 0; for MONO or non-interleaved data
+	     it should be equal to 0
+	     
+	     
+	     For unsupported quality values, it will fall back to
+	     "D" quality (the lowest one)
+		     
+		     also on return it sets:
+	     
+	     grc->outsz	 == number of output samples
+	     
+	     Note, that if quality is not supported,
+	     calling the function with real data will fall back
+	     to the worst quality available.		 
+
+	     Note that this version of GRC3 only supports 24-bit
+	     native endian.  (Modified by Sun for performance.)
+
+	     
+	     
+5.  Interleaved processing of N channels is done like this:
+
+    
+	static grc3state_t grc[N]; 
+    int t;  
+
+    //...
+    
+
+    for(t=0; t<N; t++)
+    {
+	grc3_setup( &grc[t], 22050, 48000 );
+	
+	grc3_reset( &grc[t] );
+    }
+
+    
+    //...
+    
+	while (...) {
+
+	for(t = 0; t < N; t++) {
+	    grc3_convert(&grc[t],   // instance pointer
+		      4,            // quality
+		      in_buffer,    // input buffer
+		  out_buffer,       // input buffer
+		  in_samples_count, // number of samples
+				    // in in_buffer
+		  2048,             // size of out_buffer
+		  N, t              // num of channels, channel#
+		);
+	}
+	
+	
+	// Normally, for interleaved data, ->outsz of all instances will
+	// be the same for the same stream
+	
+	put_sound_somewhere(out_buffer, 
+		     grc[0]->outsz * N * sizeof(out_buffer[0]) ); 
+    }
+    
+    
+6.  If you use the same storage and the same setup for processing few separate
+    non-related sounds, to prevent the feedback of sound1's tail to sound2's 
+    beginning - do grc3_reset on the state instances before calling 
+    grc_convert.
+    
+*****************************************************************************
+*/
+/* END CSTYLED */
+
+void grc3_setup(grc3state_t *, uint32_t fromRate, uint32_t toRate);
+
+void grc3_reset(grc3state_t *);
+
+void grc3_convert(grc3state_t *, int quality,
+    const void *src, void *dst, int sz, int bufsz, int inc, int offset);
+
+#ifdef __cplusplus
+};
+#endif
+
+#endif	/* AUDIO_GRC3_H */
diff --git a/usr/src/uts/common/io/audio/impl/audio_impl.h b/usr/src/uts/common/io/audio/impl/audio_impl.h
new file mode 100644
index 0000000000..7c8099ea88
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_impl.h
@@ -0,0 +1,472 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_AUDIO_IMPL_H
+#define	_AUDIO_IMPL_H
+
+#include <sys/types.h>
+#include <sys/list.h>
+#include <sys/poll.h>
+
+#include <sys/audio/audio_driver.h>
+#include "audio_client.h"
+
+#define	AUDIO_MAX_OPENS		256
+#define	AUDIO_MAX_CHANNELS	16
+#define	AUDIO_UNIT_EXPAND	1024
+#define	AUDIO_CHBUFS		2048	/* samples for mixing */
+#define	AUDIO_VOL_SCALE		256
+#define	AUDIO_DB_SIZE		50
+
+struct audio_parms {
+	int		p_format;
+	int		p_rate;
+	int		p_nchan;
+};
+
+typedef int (*audio_cnv_func_t)(audio_stream_t *, int);
+
+struct audio_buffer {
+	caddr_t			b_data;
+	uint64_t		b_head;
+	uint64_t		b_tail;
+	unsigned		b_hidx;		/* head % nframes */
+	unsigned		b_tidx;		/* tail % nframes */
+	unsigned		b_fragfr;	/* frames per frag */
+	unsigned		b_fragbytes;	/* bytes per frag */
+	unsigned		b_nframes;	/* total frames */
+	unsigned		b_nbytes;	/* total bytes */
+	unsigned		b_nfrags;	/* total frags */
+	unsigned		b_framesz;	/* bytes per frame  */
+};
+
+/*
+ * struct audio_stream: This structure represents a virtual stream exposed
+ * to a single client.  Each client will have at most two of these (one for
+ * record, one for playback.)
+ */
+struct audio_stream {
+	audio_buffer_t		s_buf;
+#define	s_data			s_buf.b_data
+#define	s_bufsz			s_buf.b_size
+#define	s_head			s_buf.b_head
+#define	s_tail			s_buf.b_tail
+#define	s_nfrags		s_buf.b_nfrags
+#define	s_framesz		s_buf.b_framesz
+#define	s_fragfr		s_buf.b_fragfr
+#define	s_fragbytes		s_buf.b_fragbytes
+#define	s_nframes		s_buf.b_nframes
+#define	s_nbytes		s_buf.b_nbytes
+#define	s_tidx			s_buf.b_tidx
+#define	s_hidx			s_buf.b_hidx
+	ddi_umem_cookie_t	s_cookie;
+	size_t			s_allocsz;
+
+	/*
+	 * Various counters.
+	 */
+	uint64_t		s_samples;
+	uint64_t		s_errors;	/* underrun or overrun count */
+
+	boolean_t		s_running;
+	boolean_t		s_paused;	/* stream paused */
+	boolean_t		s_draining;	/* stream draining */
+
+	/*
+	 * Sample rate conversion (SRC) and format conversion details.
+	 */
+	struct grc3state	*s_src_state[AUDIO_MAX_CHANNELS];
+	unsigned		s_src_quality;
+	int			s_cnv_max;
+	audio_cnv_func_t	s_converter;
+	uint32_t		*s_cnv_buf0;
+	uint32_t		*s_cnv_buf1;
+	void			*s_cnv_src;
+	void			*s_cnv_dst;
+	audio_parms_t		s_cnv_src_parms;
+#define	s_cnv_src_nchan		s_cnv_src_parms.p_nchan
+#define	s_cnv_src_rate		s_cnv_src_parms.p_rate
+#define	s_cnv_src_format	s_cnv_src_parms.p_format
+
+	audio_parms_t		s_cnv_dst_parms;
+#define	s_cnv_dst_nchan		s_cnv_dst_parms.p_nchan
+#define	s_cnv_dst_rate		s_cnv_dst_parms.p_rate
+#define	s_cnv_dst_format	s_cnv_dst_parms.p_format
+
+	size_t			s_cnv_cnt;
+	int32_t			*s_cnv_ptr;
+
+	audio_parms_t		*s_user_parms;
+	audio_parms_t		*s_phys_parms;
+
+	/*
+	 * Volume.
+	 */
+	uint8_t			s_gain_master;
+	uint8_t			s_gain_pct;
+	uint16_t		s_gain_scaled;
+	uint16_t		s_gain_eff;
+	boolean_t		s_muted;
+
+	/*
+	 * Callbacks.
+	 */
+	uint64_t		s_drain_idx;	/* engine index */
+
+	/*
+	 * Other per stream details, e.g. channel offset, etc.
+	 */
+	kmutex_t		s_lock;
+	kcondvar_t		s_cv;
+	list_node_t		s_eng_linkage;	/*  place on engine list */
+	audio_client_t		*s_client;
+	audio_engine_t		*s_engine;
+	int			s_choffs;
+
+	/*
+	 * Other bits.
+	 */
+	unsigned		s_engcap;	/* ENGINE_xxx_CAP */
+};
+
+/*
+ * struct audio_client: This structure represents a logical port,
+ * associated with an open file, etc.  These are the entities that are
+ * mixed.
+ */
+struct audio_client {
+	audio_stream_t		c_istream;
+	audio_stream_t		c_ostream;
+	void			*c_private;
+
+	/*
+	 * DDI support.
+	 */
+	major_t			c_major;
+	minor_t			c_minor;
+	minor_t			c_origminor;
+
+	/*
+	 * Linkage for per-device list of clients.
+	 */
+	list_node_t		c_global_linkage;
+	list_node_t		c_dev_linkage;
+	int			c_refcnt;
+
+	kmutex_t		c_lock;
+	kcondvar_t		c_cv;
+	ddi_taskq_t		*c_tq;
+	boolean_t		c_do_output;
+	boolean_t		c_do_input;
+	boolean_t		c_do_notify;
+	boolean_t		c_do_drain;
+	boolean_t		c_closing;
+	boolean_t		c_is_open;
+
+	/*
+	 * Client wide settings... e.g. ops vector, etc.
+	 */
+	unsigned		c_omode;	/* open mode */
+	pid_t			c_pid;		/* opening process id */
+	audio_dev_t		*c_dev;
+	cred_t			*c_cred;
+	audio_client_ops_t	c_ops;
+#define	c_open			c_ops.aco_open
+#define	c_close			c_ops.aco_close
+#define	c_read			c_ops.aco_read
+#define	c_write			c_ops.aco_write
+#define	c_ioctl			c_ops.aco_ioctl
+#define	c_chpoll		c_ops.aco_chpoll
+#define	c_output		c_ops.aco_output
+#define	c_input			c_ops.aco_input
+#define	c_notify		c_ops.aco_notify
+#define	c_drain			c_ops.aco_drain
+
+	struct pollhead		c_pollhead;
+
+};
+
+struct audio_infostr {
+	char			i_line[100];
+	list_node_t		i_linkage;
+};
+
+struct audio_stats {
+	kstat_named_t		st_head;
+	kstat_named_t		st_tail;
+	kstat_named_t		st_flags;
+	kstat_named_t		st_fragfr;
+	kstat_named_t		st_nfrags;
+	kstat_named_t		st_framesz;
+	kstat_named_t		st_nbytes;
+	kstat_named_t		st_hidx;
+	kstat_named_t		st_tidx;
+	kstat_named_t		st_format;
+	kstat_named_t		st_nchan;
+	kstat_named_t		st_rate;
+	kstat_named_t		st_intrs;
+	kstat_named_t		st_errors;
+	kstat_named_t		st_suspended;
+};
+
+/*
+ * An audio engine corresponds to a single DMA transfer channel.  It can
+ * represent either record or playback, but not both at the same time.
+ * A device that supports simultaneous record and playback will register
+ * separate channels.
+ */
+struct audio_engine {
+	audio_engine_ops_t	e_ops;
+	void			*e_private;
+	unsigned		e_flags;
+
+	/*
+	 * Mixing related fields.
+	 */
+	unsigned		e_limiter_state;
+	int32_t			*e_chbufs[AUDIO_MAX_CHANNELS];
+	unsigned		e_choffs[AUDIO_MAX_CHANNELS];
+	unsigned		e_chincr[AUDIO_MAX_CHANNELS];
+	void			(*e_export)(audio_engine_t *);
+	void			(*e_import)(audio_engine_t *, audio_stream_t *);
+
+	/*
+	 * Underlying physical buffer shared with device driver.
+	 */
+	audio_buffer_t		e_buf;
+#define	e_head			e_buf.b_head
+#define	e_tail			e_buf.b_tail
+#define	e_data			e_buf.b_data
+#define	e_fragfr		e_buf.b_fragfr
+#define	e_fragbytes		e_buf.b_fragbytes
+#define	e_framesz		e_buf.b_framesz
+#define	e_nbytes		e_buf.b_nbytes
+#define	e_nframes		e_buf.b_nframes
+#define	e_nfrags		e_buf.b_nfrags
+#define	e_hidx			e_buf.b_hidx
+#define	e_tidx			e_buf.b_tidx
+
+	int			e_intrs;
+	int			e_errors;
+
+	audio_parms_t		e_parms;
+#define	e_format		e_parms.p_format
+#define	e_nchan			e_parms.p_nchan
+#define	e_rate			e_parms.p_rate
+
+	/*
+	 * Statistics.
+	 */
+	kstat_t			*e_ksp;
+	struct audio_stats	e_stats;
+
+
+	/*
+	 * Synchronization.
+	 */
+	kmutex_t		e_lock;
+
+	/*
+	 * Linkage for per-device list.
+	 */
+	list_node_t		e_dev_linkage;
+	audio_dev_t		*e_dev;
+	int			e_num;	/* arbitrary engine number */
+
+	/*
+	 * List of of streams attached to this engine.
+	 */
+	list_t			e_streams;
+	int			e_nrunning;
+	boolean_t		e_suspended;
+};
+
+struct audio_dev {
+	dev_info_t		*d_dip;
+	major_t			d_major;
+	int			d_instance;
+
+	uint32_t		d_flags;
+#define	DEV_OUTPUT_CAP		(1U << 0)
+#define	DEV_INPUT_CAP		(1U << 1)
+#define	DEV_DUPLEX_CAP		(1U << 2)
+#define	DEV_SNDSTAT_CAP		(1U << 3)
+
+	char			d_name[128];	/* generic description */
+	char			d_desc[128];	/* detailed config descr */
+	char			d_vers[128];	/* detailed version descr */
+	int			d_number;	/* global /dev/audioXX # */
+	int			d_index;	/* master device index */
+	int			d_engno;	/* engine counter */
+
+	list_t			d_hwinfo;	/* strings of hw info */
+
+	/*
+	 * Synchronization.
+	 */
+	kmutex_t		d_lock;
+	kcondvar_t		d_cv;
+	krwlock_t		d_ctrl_lock;	/* leaf lock */
+	krwlock_t		d_clnt_lock;
+	unsigned		d_refcnt;
+
+	/*
+	 * Lists of virtual clients, controls and engines.  Protected by
+	 * the d_lock field above.
+	 */
+	list_t			d_clients;
+	list_t			d_engines;
+	list_t			d_controls;
+	audio_ctrl_t		*d_pcmvol_ctrl;
+	uint64_t		d_pcmvol;
+
+	/*
+	 * Linkage onto global list of devices.
+	 */
+	list_node_t		d_by_index;
+	list_node_t		d_by_number;
+
+	/*
+	 * Personality specific data.
+	 */
+	void			*d_minor_data[1 << AUDIO_MN_TYPE_NBITS];
+};
+
+/*
+ * Each audio_dev optionally can have controls attached to it.
+ * Controls are separate from audio engines. They are methods of
+ * adjusting pharameters or reading metrics that usually relate to
+ * hardware on devices engine by the driver. They can be things like
+ * master volume for example.
+ *
+ * If the driver does not support controls then it must insure
+ * that any hardware controls are initialized to a usable state.
+ *
+ * For the framework/user-apps to be able to change controls
+ * the driver must create, enable and configure controls with
+ * control API's.
+ *
+ * There are a number of common controls (well-known) that most
+ * hardware supports. These have known names and known ctrl numbers.
+ * In addition a driver can have any number of extention
+ * controls (device-private). These can have any name and any ctrl
+ * number other then the ones, defined as well-knonw ones.
+ *
+ * Only controls created through control API's will be available,
+ * well-known or device-private.
+ */
+struct	audio_ctrl {
+	audio_ctrl_desc_t	ctrl_des;
+#define	ctrl_name		ctrl_des.acd_name
+#define	ctrl_type		ctrl_des.acd_type
+#define	ctrl_enum		ctrl_des.acd_enum
+#define	ctrl_flags		ctrl_des.acd_flags
+	audio_dev_t		*ctrl_dev;
+	audio_ctrl_rd_t		ctrl_read_fn;
+	audio_ctrl_wr_t		ctrl_write_fn;
+	list_node_t		ctrl_linkage;
+	kmutex_t		ctrl_lock;
+	void			*ctrl_arg;
+};
+
+
+/*
+ * Prototypes.
+ */
+
+/* audio_format.c */
+int auimpl_format_alloc(audio_stream_t *);
+void auimpl_format_free(audio_stream_t *);
+int auimpl_format_setup(audio_stream_t *, audio_parms_t *);
+
+/* audio_output.c */
+void auimpl_export_16ne(audio_engine_t *);
+void auimpl_export_16oe(audio_engine_t *);
+void auimpl_export_24ne(audio_engine_t *);
+void auimpl_export_24oe(audio_engine_t *);
+void auimpl_export_32ne(audio_engine_t *);
+void auimpl_export_32oe(audio_engine_t *);
+void auimpl_output_callback(audio_engine_t *);
+
+/* audio_input.c */
+void auimpl_import_16ne(audio_engine_t *, audio_stream_t *);
+void auimpl_import_16oe(audio_engine_t *, audio_stream_t *);
+void auimpl_import_24ne(audio_engine_t *, audio_stream_t *);
+void auimpl_import_24oe(audio_engine_t *, audio_stream_t *);
+void auimpl_import_32ne(audio_engine_t *, audio_stream_t *);
+void auimpl_import_32oe(audio_engine_t *, audio_stream_t *);
+void auimpl_input_callback(audio_engine_t *);
+int auimpl_input_drain(audio_stream_t *);
+
+/* audio_client.c */
+void auimpl_client_init(void);
+void auimpl_client_fini(void);
+audio_client_t *auimpl_client_create(dev_t);
+void auimpl_client_destroy(audio_client_t *);
+int auimpl_create_minors(audio_dev_t *);
+void auimpl_remove_minors(audio_dev_t *);
+void auimpl_set_gain_master(audio_stream_t *, uint8_t);
+
+/* audio_engine.c */
+void auimpl_dev_init(void);
+void auimpl_dev_fini(void);
+void auimpl_dev_hold(audio_dev_t *);
+audio_dev_t *auimpl_dev_hold_by_devt(dev_t);
+audio_dev_t *auimpl_dev_hold_by_index(int);
+void auimpl_dev_release(audio_dev_t *);
+int auimpl_choose_format(int);
+
+int auimpl_engine_open(audio_dev_t *, int, int, audio_stream_t *);
+void auimpl_engine_close(audio_stream_t *);
+
+void auimpl_dev_walk_engines(audio_dev_t *,
+    int (*)(audio_engine_t *, void *), void *);
+
+void auimpl_dev_vwarn(audio_dev_t *, const char *, va_list);
+
+/* engine operations */
+#define	E_OP(e, entry)		((e)->e_ops.audio_engine_##entry)
+#define	E_PRV(e)		((e)->e_private)
+#define	ENG_FORMAT(e)		E_OP(e, format)(E_PRV(e))
+#define	ENG_RATE(e)		E_OP(e, rate)(E_PRV(e))
+#define	ENG_CHANNELS(e)		E_OP(e, channels)(E_PRV(e))
+#define	ENG_SYNC(e, num)	E_OP(e, sync)(E_PRV(e), num)
+#define	ENG_START(e)		E_OP(e, start)(E_PRV(e))
+#define	ENG_STOP(e)		E_OP(e, stop)(E_PRV(e))
+#define	ENG_COUNT(e)		E_OP(e, count)(E_PRV(e))
+#define	ENG_QLEN(e)		E_OP(e, qlen)(E_PRV(e))
+#define	ENG_CLOSE(e)		E_OP(e, close)(E_PRV(e))
+#define	ENG_OPEN(e, s, nf, d) 	E_OP(e, open)(E_PRV(e), e->e_flags, s, nf, d)
+#define	ENG_CHINFO(e, c, o, i)	E_OP(e, chinfo(E_PRV(e), c, o, i))
+
+/* audio_sun.c */
+void auimpl_sun_init(void);
+
+/* audio_oss.c */
+void auimpl_oss_init(void);
+
+#endif	/* _AUDIO_IMPL_H */
diff --git a/usr/src/uts/common/io/audio/impl/audio_input.c b/usr/src/uts/common/io/audio/impl/audio_input.c
new file mode 100644
index 0000000000..8646707880
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_input.c
@@ -0,0 +1,229 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: Virtual mixing audio input routines
+ *
+ * This file contains the actual mixing and resampling engine for input.
+ */
+
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/sysmacros.h>
+#include "audio_impl.h"
+
+#ifdef	DEBUG
+int	audio_overruns = 0;
+#endif
+
+#define	DECL_AUDIO_IMPORT(NAME, TYPE, SWAP, SHIFT)			\
+void									\
+auimpl_import_##NAME(audio_engine_t *eng, audio_stream_t *sp)		\
+{									\
+	int	nch = eng->e_nchan;					\
+	int32_t *out;							\
+	TYPE	*in;							\
+	int	ch;							\
+	void	*data;							\
+	int	vol;							\
+									\
+	data = sp->s_cnv_src;						\
+	ch = 0;								\
+	in = (void *)(eng->e_data + (eng->e_tidx * eng->e_framesz));	\
+	out = data;							\
+	vol = sp->s_gain_eff;						\
+									\
+	do {	/* for each channel */					\
+		TYPE *ip;						\
+		int32_t *op;						\
+		int i;							\
+									\
+		/* get value and adjust next channel offset */		\
+		op = out++;						\
+		ip = in++;						\
+									\
+		i = eng->e_fragfr;					\
+									\
+		do {	/* for each frame */				\
+			int32_t	sample = (TYPE)SWAP(*ip);		\
+			int32_t	scaled = sample SHIFT;			\
+									\
+			scaled *= vol;					\
+			scaled /= AUDIO_VOL_SCALE;			\
+									\
+			*op = scaled;					\
+			ip += nch;					\
+			op += nch;					\
+									\
+		} while (--i);						\
+		ch++;							\
+	} while (ch < nch);						\
+}
+
+DECL_AUDIO_IMPORT(16ne, int16_t, /* nop */, << 8)
+DECL_AUDIO_IMPORT(16oe, int16_t, ddi_swap16, << 8)
+DECL_AUDIO_IMPORT(32ne, int32_t, /* nop */, >> 8)
+DECL_AUDIO_IMPORT(32oe, int32_t, ddi_swap32, >> 8)
+DECL_AUDIO_IMPORT(24ne, int32_t, /* nop */, /* nop */)
+DECL_AUDIO_IMPORT(24oe, int32_t, ddi_swap32, /* nop */)
+
+/*
+ * Produce a fragment's worth of data.  This is called when the data in
+ * the conversion buffer is exhausted, and we need to refill it from the
+ * source buffer.  We always consume data from the client in quantities of
+ * a fragment at a time (assuming that a fragment is available.)
+ */
+static void
+auimpl_produce_fragment(audio_stream_t *sp, unsigned count)
+{
+	unsigned	nframes;
+	unsigned	framesz;
+	caddr_t		cnvsrc;
+	caddr_t		data;
+
+	nframes = sp->s_nframes;
+	framesz = sp->s_framesz;
+
+	ASSERT(sp->s_head >= sp->s_tail);
+	ASSERT(sp->s_hidx < nframes);
+	ASSERT(sp->s_tidx < nframes);
+
+	/*
+	 * Copy data.  We deal properly with wraps.  Done as a
+	 * do...while to minimize the number of tests.
+	 */
+	cnvsrc = sp->s_cnv_src;
+	data = sp->s_data + (sp->s_hidx * framesz);
+	do {
+		unsigned nf;
+		unsigned nb;
+
+		ASSERT(sp->s_hidx < nframes);
+		nf = min(nframes - sp->s_hidx, count);
+		nb = nf * framesz;
+
+		bcopy(cnvsrc, data, nb);
+		data += nb;
+		cnvsrc += nb;
+		sp->s_hidx += nf;
+		sp->s_head += nf;
+		count -= nf;
+		sp->s_samples += nf;
+		if (sp->s_hidx >= nframes) {
+			sp->s_hidx -= nframes;
+			data -= sp->s_nbytes;
+		}
+	} while (count);
+
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < nframes);
+	ASSERT(sp->s_tail <= sp->s_head);
+	ASSERT(sp->s_hidx < nframes);
+}
+
+void
+auimpl_input_callback(audio_engine_t *eng)
+{
+	int		fragfr = eng->e_fragfr;
+	boolean_t	overrun;
+	audio_client_t	*c;
+
+	/* consume all fragments in the buffer */
+	while ((eng->e_head - eng->e_tail) > fragfr) {
+
+		/*
+		 * Consider doing the SYNC outside of the lock.
+		 */
+		ENG_SYNC(eng, fragfr);
+
+		for (audio_stream_t *sp = list_head(&eng->e_streams);
+		    sp != NULL;
+		    sp = list_next(&eng->e_streams, sp)) {
+			int space;
+			int count;
+
+			c = sp->s_client;
+
+			mutex_enter(&sp->s_lock);
+			/* skip over streams paused or not running */
+			if (sp->s_paused || (!sp->s_running) ||
+			    eng->e_suspended) {
+				mutex_exit(&sp->s_lock);
+				continue;
+			}
+			sp->s_cnv_src = sp->s_cnv_buf0;
+			sp->s_cnv_dst = sp->s_cnv_buf1;
+			eng->e_import(eng, sp);
+
+			/*
+			 * Optionally convert fragment to requested sample
+			 * format and rate.
+			 */
+			if (sp->s_converter != NULL) {
+				count = sp->s_converter(sp, fragfr);
+			} else {
+				count = fragfr;
+			}
+
+			space = sp->s_nframes - (sp->s_head - sp->s_tail);
+			if (count > space) {
+#ifdef	DEBUG
+				audio_overruns++;
+#endif
+				sp->s_errors += count - space;
+				count = space;
+				overrun = B_TRUE;
+			} else {
+				overrun = B_FALSE;
+			}
+
+			auimpl_produce_fragment(sp, count);
+
+			/* wake blocked threads (blocking reads, etc.) */
+			cv_broadcast(&sp->s_cv);
+
+			mutex_exit(&sp->s_lock);
+
+			mutex_enter(&c->c_lock);
+			if (overrun) {
+				c->c_do_notify = B_TRUE;
+			}
+			c->c_do_input = B_TRUE;
+			cv_broadcast(&c->c_cv);
+			mutex_exit(&c->c_lock);
+		}
+
+		/*
+		 * Update the tail pointer, and the data pointer.
+		 */
+		eng->e_tail += fragfr;
+		eng->e_tidx += fragfr;
+		if (eng->e_tidx >= eng->e_nframes) {
+			eng->e_tidx -= eng->e_nframes;
+		}
+	}
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_oss.c b/usr/src/uts/common/io/audio/impl/audio_oss.c
new file mode 100644
index 0000000000..a77fcd1196
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_oss.c
@@ -0,0 +1,2577 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/types.h>
+#include <sys/open.h>
+#include <sys/errno.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_oss.h>
+#include <sys/file.h>
+#include <sys/note.h>
+#include <sys/sysmacros.h>
+#include <sys/list.h>
+#include "audio_client.h"
+
+#define	OSS_FMT		AFMT_S16_LE
+#define	OSS_RATE	48000
+#define	OSS_CHANNELS	2
+
+typedef struct ossclient ossclient_t;
+typedef struct ossdev ossdev_t;
+
+static const struct {
+	int	oss;
+	int	fmt;
+} oss_formats[] = {
+	{ AFMT_MU_LAW,		AUDIO_FORMAT_ULAW },
+	{ AFMT_A_LAW,		AUDIO_FORMAT_ALAW },
+	{ AFMT_U8,		AUDIO_FORMAT_U8 },
+	{ AFMT_S8,		AUDIO_FORMAT_S8 },
+	{ AFMT_S16_BE,		AUDIO_FORMAT_S16_BE },
+	{ AFMT_S16_LE,		AUDIO_FORMAT_S16_LE },
+	{ AFMT_U16_BE,		AUDIO_FORMAT_U16_BE },
+	{ AFMT_U16_LE,		AUDIO_FORMAT_U16_LE },
+	{ AFMT_S24_BE,		AUDIO_FORMAT_S24_BE },
+	{ AFMT_S24_LE,		AUDIO_FORMAT_S24_LE },
+	{ AFMT_S32_BE,		AUDIO_FORMAT_S32_BE },
+	{ AFMT_S32_LE,		AUDIO_FORMAT_S32_LE },
+	{ AFMT_S24_PACKED,	AUDIO_FORMAT_S24_PACKED },
+	{ AFMT_AC3,		AUDIO_FORMAT_AC3 },
+	{ AFMT_QUERY,		AUDIO_FORMAT_NONE }
+};
+
+/* common structure shared between both mixer and dsp nodes */
+struct ossclient {
+	ossdev_t		*o_ossdev;
+	audio_client_t		*o_client;
+	/* sndstat */
+	kmutex_t		o_ss_lock;
+	char			*o_ss_buf;
+	size_t			o_ss_len;
+	size_t			o_ss_sz;
+	size_t			o_ss_off;
+};
+
+struct ossdev {
+	audio_dev_t		*d_dev;
+
+	uint_t			d_modify_cnt;	/* flag apps of ctrl changes */
+	uint_t			d_nctrl;	/* num actual controls */
+	uint_t			d_nalloc;	/* num allocated controls */
+	audio_ctrl_t		**d_ctrls;	/* array of control handles */
+	oss_mixext		*d_exts;	/* array of mixer descs */
+
+	int			d_play_grp;
+	int			d_rec_grp;
+	int			d_mon_grp;
+	int			d_misc_grp;
+
+	kmutex_t		d_mx;
+	kcondvar_t		d_cv;
+};
+
+static int
+oss_cnt_controls(audio_ctrl_t *ctrl, void *arg)
+{
+	int			*pint = (int *)arg;
+	int			cnt;
+	audio_ctrl_desc_t	desc;
+
+	cnt = *pint;
+	cnt++;
+	*pint = cnt;
+
+	if (auclnt_control_describe(ctrl, &desc) != 0)
+		return (AUDIO_WALK_CONTINUE);
+
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_MULTI) {
+		for (uint64_t mask = desc.acd_maxvalue; mask; mask >>= 1) {
+			if (mask & 1) {
+				cnt++;
+			}
+		}
+		*pint = cnt;
+	}
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+/*
+ * Add one entry to the OSS user control table to internal control
+ * helper table.
+ *
+ * This is used with auimpl_walk_controls. The table must be pre-
+ * allocated before it is walk'd. This includes the root and
+ * extended control markers!
+ */
+static int
+oss_add_control(audio_ctrl_t *ctrl, void *arg)
+{
+	ossdev_t		*odev = arg;
+	audio_ctrl_desc_t	desc;
+	oss_mixext		*ext;
+	int			bit;
+	uint64_t		mask;
+	const char		*name;
+	int			parent;
+	int			flags;
+	unsigned		scope;
+
+	if (auclnt_control_describe(ctrl, &desc))
+		return (AUDIO_WALK_CONTINUE);
+
+	parent = 0;
+
+	/*
+	 * Add appropriate group if not already done so.
+	 */
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_PLAY) {
+		if (!odev->d_play_grp) {
+			ext = &odev->d_exts[odev->d_nctrl];
+			ext->ctrl = odev->d_nctrl;
+			ext->control_no = -1;
+			ext->type = MIXT_GROUP;
+			ext->desc = MIXEXT_SCOPE_OUTPUT;
+			ext->timestamp = gethrtime();
+			(void) snprintf(ext->id, sizeof (ext->id), "PLAYBACK");
+			odev->d_play_grp = odev->d_nctrl;
+			odev->d_nctrl++;
+		}
+		scope = MIXEXT_SCOPE_OUTPUT;
+		parent = odev->d_play_grp;
+	} else if (desc.acd_flags & AUDIO_CTRL_FLAG_REC) {
+		if (!odev->d_rec_grp) {
+			ext = &odev->d_exts[odev->d_nctrl];
+			ext->ctrl = odev->d_nctrl;
+			ext->control_no = -1;
+			ext->type = MIXT_GROUP;
+			ext->desc = MIXEXT_SCOPE_INPUT;
+			ext->timestamp = gethrtime();
+			(void) snprintf(ext->id, sizeof (ext->id), "RECORD");
+			odev->d_rec_grp = odev->d_nctrl;
+			odev->d_nctrl++;
+		}
+		scope = MIXEXT_SCOPE_INPUT;
+		parent = odev->d_rec_grp;
+	} else if (desc.acd_flags & AUDIO_CTRL_FLAG_MONITOR) {
+		if (!odev->d_mon_grp) {
+			ext = &odev->d_exts[odev->d_nctrl];
+			ext->ctrl = odev->d_nctrl;
+			ext->control_no = -1;
+			ext->type = MIXT_GROUP;
+			ext->desc = MIXEXT_SCOPE_MONITOR;
+			ext->timestamp = gethrtime();
+			(void) snprintf(ext->id, sizeof (ext->id), "MONITOR");
+			odev->d_mon_grp = odev->d_nctrl;
+			odev->d_nctrl++;
+		}
+		scope = MIXEXT_SCOPE_MONITOR;
+		parent = odev->d_mon_grp;
+	} else {
+		if (!odev->d_misc_grp) {
+			ext = &odev->d_exts[odev->d_nctrl];
+			ext->ctrl = odev->d_nctrl;
+			ext->control_no = -1;
+			ext->type = MIXT_GROUP;
+			ext->desc = MIXEXT_SCOPE_OTHER;
+			ext->timestamp = gethrtime();
+			(void) snprintf(ext->id, sizeof (ext->id), "MISC");
+			odev->d_misc_grp = odev->d_nctrl;
+			odev->d_nctrl++;
+		}
+		scope = MIXEXT_SCOPE_OTHER;
+		parent = odev->d_misc_grp;
+	}
+
+	name = desc.acd_name ? desc.acd_name : "";
+
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_MULTI) {
+		ext = &odev->d_exts[odev->d_nctrl];
+		ext->ctrl = odev->d_nctrl;
+		ext->control_no = -1;
+		ext->type = MIXT_GROUP;
+		ext->timestamp = gethrtime();
+		ext->parent = parent;
+		ext->desc = scope;
+		(void) snprintf(ext->id, sizeof (ext->id), "%s", name);
+		(void) snprintf(ext->extname, sizeof (ext->extname),
+		    "%s", name);
+		parent = odev->d_nctrl++;
+	}
+
+	/* Next available open entry */
+	ext = &odev->d_exts[odev->d_nctrl];
+
+	/* Record the underlying control handle */
+	odev->d_ctrls[odev->d_nctrl] = ctrl;
+
+	/*
+	 * Now setup the oss entry
+	 */
+
+	ext->ctrl = odev->d_nctrl;
+	ext->control_no = -1;
+	ext->maxvalue = (int)desc.acd_maxvalue;
+	ext->minvalue = (int)desc.acd_minvalue;
+	ext->timestamp = gethrtime();
+	ext->parent = parent;
+	ext->desc = scope;
+	/* all controls should be pollable for now */
+	flags = MIXF_POLL;
+
+	/*
+	 * The following flags are intended to help out applications
+	 * which need to figure out where to place certain controls.
+	 * A few further words of guidance:
+	 *
+	 * Apps that just want a single master volume control should
+	 * adjust the control(s) that are labelled with MIXF_PCMVOL if
+	 * present.  They can fall back to adjusting all MAINVOL
+	 * levels instead, if no PCMVOL is present.
+	 *
+	 * Controls that are one type on a certain device might be a
+	 * different type on another device.  For example,
+	 * audiopci/ak4531 can adjust input gains for individual
+	 * levels, but lacks a master record gain.  AC'97, on the
+	 * other hand, has individual monitor gains for inputs, but
+	 * only a single master recording gain.
+	 */
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_READABLE)
+		flags |= MIXF_READABLE;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_WRITEABLE)
+		flags |= MIXF_WRITEABLE;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_CENTIBEL)
+		flags |= MIXF_CENTIBEL;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_DECIBEL)
+		flags |= MIXF_DECIBEL;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_MAINVOL)
+		flags |= MIXF_MAINVOL;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_PCMVOL)
+		flags |= MIXF_PCMVOL;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_RECVOL)
+		flags |= MIXF_RECVOL;
+	if (desc.acd_flags & AUDIO_CTRL_FLAG_MONVOL)
+		flags |= MIXF_MONVOL;
+	ext->flags = flags;
+
+	(void) snprintf(ext->id, sizeof (ext->id), "%s", name);
+
+	/*
+	 * For now just use the same extname as the real name.
+	 */
+	(void) snprintf(ext->extname, sizeof (ext->extname), name);
+
+	/*
+	 * Now we deal with various control types.
+	 */
+	switch (desc.acd_type) {
+	case AUDIO_CTRL_TYPE_BOOLEAN:
+		ext->type = MIXT_ONOFF;
+		ext->enumbit = -1;
+		break;
+	case AUDIO_CTRL_TYPE_STEREO:
+		ext->type = MIXT_STEREOSLIDER;
+		break;
+	case AUDIO_CTRL_TYPE_MONO:
+		ext->type = MIXT_MONOSLIDER;
+		break;
+	case AUDIO_CTRL_TYPE_ENUM:
+
+		if (desc.acd_flags & AUDIO_CTRL_FLAG_MULTI) {
+			/*
+			 * We turn AUDIO_CTRL_FLAG_MULTI into a group
+			 * of checkboxes, since OSS can't represent it
+			 * natively.
+			 */
+			mask = desc.acd_maxvalue;
+			bit = 0;
+			while (mask) {
+				if (mask & 1) {
+					ext = &odev->d_exts[odev->d_nctrl];
+					(void) snprintf(ext->extname,
+					    sizeof (ext->extname), "%s.%s",
+					    name, desc.acd_enum[bit]);
+					(void) snprintf(ext->id,
+					    sizeof (ext->id), "%s",
+					    desc.acd_enum[bit]);
+					ext->ctrl = odev->d_nctrl;
+					ext->control_no = -1;
+					ext->parent = parent;
+					ext->timestamp = gethrtime();
+					ext->type = MIXT_ONOFF;
+					ext->minvalue = 0;
+					ext->maxvalue = 1;
+					ext->enumbit = bit;
+					ext->flags = flags;
+					odev->d_ctrls[odev->d_nctrl] = ctrl;
+					odev->d_nctrl++;
+				}
+				bit++;
+				mask >>= 1;
+			}
+			return (AUDIO_WALK_CONTINUE);
+		} else {
+			/*
+			 * NB: This is sufficient only for controls
+			 * with a single value.  It cannot express the
+			 * richer bitmask capabilities.
+			 */
+			ext->type = MIXT_ENUM;
+			ext->minvalue = 0;
+
+			/*
+			 * For an enumaration, we need to figure out
+			 * which values are present, and set the
+			 * appropriate mask and max value.
+			 */
+			bzero(ext->enum_present, sizeof (ext->enum_present));
+			mask = desc.acd_maxvalue;
+			bit = 0;
+			while (mask) {
+				if (mask & 1) {
+					ext->enum_present[bit / 8] |=
+					    (1 << (bit % 8));
+				}
+				mask >>= 1;
+				bit++;
+			}
+			ext->maxvalue = bit;
+		}
+		break;
+
+	case AUDIO_CTRL_TYPE_METER:
+	default:
+		/* Its an unknown or unsupported (for now) control, skip */
+		return (AUDIO_WALK_CONTINUE);
+	}
+
+	odev->d_nctrl++;
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+/*
+ * Free up an OSS user land control to internal control,
+ * helper table.
+ */
+static void
+oss_free_controls(ossdev_t *odev)
+{
+	kmem_free(odev->d_ctrls, sizeof (audio_ctrl_t *) * odev->d_nalloc);
+	kmem_free(odev->d_exts, sizeof (oss_mixext) * odev->d_nalloc);
+	odev->d_nctrl = 0;
+	odev->d_nalloc = 0;
+}
+
+/*
+ * Allocate and fill in an OSS user land controls to internal controls
+ * helper table. This is done on one audio_dev device.
+ */
+static void
+oss_alloc_controls(ossdev_t *odev)
+{
+	audio_dev_t		*d = odev->d_dev;
+	int			nctrl = 0;
+	oss_mixext		*ext;
+	oss_mixext_root		*root_data;
+
+	/* Find out who many entries we need */
+	auclnt_walk_controls(d, oss_cnt_controls, &nctrl);
+	nctrl++;		/* Needs space for the device root node */
+	nctrl++;		/* Needs space for the device ext marker */
+	nctrl++;		/* Needs space for the play group */
+	nctrl++;		/* Needs space for the record group */
+	nctrl++;		/* Needs space for the monitor group */
+	nctrl++;		/* Needs space for the tone group */
+	nctrl++;		/* Needs space for the 3D group */
+	nctrl++;		/* Needs space for the misc group */
+
+	/* Allocate the OSS to boomer helper table */
+	odev->d_nalloc = nctrl;
+	odev->d_ctrls = kmem_zalloc(sizeof (audio_ctrl_t *) * nctrl, KM_SLEEP);
+	odev->d_exts = kmem_zalloc(sizeof (oss_mixext) * nctrl, KM_SLEEP);
+
+	/*
+	 * Setup special case outputs to output OSS routes helper tables
+	 */
+
+	/*
+	 * Root node is first, that way all others parent is this one
+	 */
+	ext = &odev->d_exts[odev->d_nctrl];
+	ext->ctrl = 0;
+	ext->parent = -1;
+	ext->type = MIXT_DEVROOT;
+	ext->timestamp = gethrtime();
+	(void) snprintf(ext->id, sizeof (ext->id), "DEVROOT");
+	/*
+	 * Root data... nobody should be using this though.
+	 */
+	root_data = (oss_mixext_root *)&ext->data;
+	(void) snprintf(root_data->name, sizeof (root_data->name), "%s",
+	    auclnt_get_dev_name(d));
+	(void) snprintf(root_data->id, sizeof (root_data->id), "%s",
+	    auclnt_get_dev_name(d));
+
+	odev->d_nctrl++;
+
+	/*
+	 * Insert an extra marker -- needed to keep layout apps hapy.
+	 * This prevents some apps from assuming we are in "LEGACY" mode.
+	 */
+	ext = &odev->d_exts[odev->d_nctrl];
+	ext->ctrl = odev->d_nctrl;
+	ext->control_no = -1;
+	ext->type = MIXT_MARKER;
+	ext->timestamp = gethrtime();
+	ext->parent = 0;
+	odev->d_nctrl++;
+
+	/* Fill in the complete table now */
+	auclnt_walk_controls(d, oss_add_control, odev);
+
+	/* Update the update_counter reference counter for groups */
+	for (nctrl = 0; nctrl < odev->d_nctrl; nctrl++) {
+		int i;
+
+		ext = &odev->d_exts[nctrl];
+		i = ext->parent;
+		while ((i >= 0) && (i < odev->d_nctrl)) {
+
+			ext = &odev->d_exts[i];
+			ASSERT(ext->parent < i);
+			ASSERT((ext->type == MIXT_GROUP) ||
+			    (ext->type == MIXT_DEVROOT));
+			ext->update_counter++;
+			i = ext->parent;
+		}
+	}
+
+	ASSERT(odev->d_nctrl <= odev->d_nalloc);
+}
+
+static int
+oss_open(audio_client_t *c, int oflag)
+{
+	int		rv;
+	ossdev_t	*odev;
+	ossclient_t	*sc;
+	audio_stream_t	*isp, *osp;
+
+	isp = auclnt_input_stream(c);
+	osp = auclnt_output_stream(c);
+
+	/* note that OSS always uses nonblocking open() semantics */
+	if ((rv = auclnt_open(c, AUDIO_FORMAT_PCM, oflag | FNDELAY)) != 0) {
+		return (rv);
+	}
+
+	if ((sc = kmem_zalloc(sizeof (*sc), KM_NOSLEEP)) == NULL) {
+		auclnt_close(c);
+		return (ENOMEM);
+	}
+	auclnt_set_private(c, sc);
+
+	odev = auclnt_get_minor_data(c, AUDIO_MINOR_DSP);
+
+	/* set a couple of common fields */
+	sc->o_client = c;
+	sc->o_ossdev = odev;
+
+	/* set all default parameters */
+	if (oflag & FWRITE) {
+		if (((rv = auclnt_set_format(osp, OSS_FMT)) != 0) ||
+		    ((rv = auclnt_set_rate(osp, OSS_RATE)) != 0) ||
+		    ((rv = auclnt_set_channels(osp, OSS_CHANNELS)) != 0)) {
+			goto failed;
+		}
+	}
+
+	if (oflag & FREAD) {
+		if (((rv = auclnt_set_format(isp, OSS_FMT)) != 0) ||
+		    ((rv = auclnt_set_rate(isp, OSS_RATE)) != 0) ||
+		    ((rv = auclnt_set_channels(isp, OSS_CHANNELS)) != 0)) {
+			goto failed;
+		}
+	}
+
+	return (0);
+
+failed:
+	auclnt_close(c);
+	return (rv);
+}
+
+static void
+oss_close(audio_client_t *c)
+{
+	ossclient_t	*sc;
+
+	sc = auclnt_get_private(c);
+
+	if (ddi_can_receive_sig() || (ddi_get_pid() == 0)) {
+		(void) auclnt_drain(c);
+	}
+
+	kmem_free(sc, sizeof (*sc));
+
+	auclnt_close(c);
+}
+
+/*
+ * This is used to generate an array of names for an enumeration
+ */
+static ushort_t
+oss_set_enum(oss_mixer_enuminfo *ei, ushort_t nxt, const char *name)
+{
+	uint32_t	n;
+
+	/* Get current entry to fill in */
+	n = ei->nvalues;
+	(void) snprintf(&ei->strings[nxt], ((sizeof (ei->strings) - nxt) - 1),
+	    "%s", name);
+	ei->strindex[n] = nxt;
+
+	/* Adjust everything for next entry */
+	nxt += strnlen(name, ((sizeof (ei->strings) - nxt) - 1));
+	ei->strings[nxt++] = '\0';
+
+	ei->nvalues++;
+	return (nxt);
+}
+
+/*
+ * The following two functions are used to count the number of devices
+ * in under the boomer framework.
+ *
+ * We actually report the highest "index", and then if an audio device
+ * is not found, we report a bogus removed device for it in the actual
+ * ioctls.  This goofiness is required to make the OSS API happy.
+ */
+int
+oss_dev_walker(audio_dev_t *d, void *arg)
+{
+	int		*pcnt = arg;
+	int		cnt;
+	int		index;
+
+	cnt = *pcnt;
+	index = auclnt_get_dev_index(d);
+	if ((index + 1) > cnt) {
+		cnt = index + 1;
+		*pcnt = cnt;
+	}
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+oss_cnt_devs(void)
+{
+	int cnt = 0;
+
+	auclnt_walk_devs(oss_dev_walker, &cnt);
+	return (cnt);
+}
+
+static int
+sndctl_dsp_speed(audio_client_t *c, int *ratep)
+{
+	int		rv;
+	int		rate;
+	int		oflag;
+
+	rate = *ratep;
+
+	oflag = auclnt_get_oflag(c);
+	if (oflag & FREAD) {
+		if ((rv = auclnt_set_rate(auclnt_input_stream(c), rate)) != 0)
+			return (rv);
+	}
+
+	if (oflag & FWRITE) {
+		if ((rv = auclnt_set_rate(auclnt_output_stream(c), rate)) != 0)
+			return (rv);
+	}
+
+	return (0);
+}
+
+static int
+sndctl_dsp_setfmt(audio_client_t *c, int *fmtp)
+{
+	int		rv;
+	int		fmt;
+	int		i;
+	int		oflag;
+
+	oflag = auclnt_get_oflag(c);
+
+	if (*fmtp != AFMT_QUERY) {
+		/* convert from OSS */
+		for (i = 0; oss_formats[i].fmt != AUDIO_FORMAT_NONE; i++) {
+			if (oss_formats[i].oss == *fmtp) {
+				fmt = oss_formats[i].fmt;
+				break;
+			}
+		}
+		if (fmt == AUDIO_FORMAT_NONE) {
+			/* if format not known, return ENOTSUP */
+			return (ENOTSUP);
+		}
+
+		if (oflag & FWRITE) {
+			rv = auclnt_set_format(auclnt_output_stream(c), fmt);
+			if (rv != 0)
+				return (rv);
+		}
+
+		if (oflag & FREAD) {
+			rv = auclnt_set_format(auclnt_input_stream(c), fmt);
+			if (rv != 0)
+				return (rv);
+		}
+	}
+
+	if (oflag & FWRITE) {
+		fmt = auclnt_get_format(auclnt_output_stream(c));
+	} else if (oflag & FREAD) {
+		fmt = auclnt_get_format(auclnt_input_stream(c));
+	}
+
+	/* convert back to OSS */
+	*(int *)fmtp = AFMT_QUERY;
+	for (i = 0; oss_formats[i].fmt != AUDIO_FORMAT_NONE; i++) {
+		if (oss_formats[i].fmt == fmt) {
+			*(int *)fmtp = oss_formats[i].oss;
+		}
+	}
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getfmts(audio_client_t *c, int *fmtsp)
+{
+	_NOTE(ARGUNUSED(c));
+
+	/*
+	 * For now, we support all the standard ones.  Later we might
+	 * add in conditional support for AC3.
+	 */
+	*fmtsp = (AFMT_MU_LAW | AFMT_A_LAW |
+	    AFMT_U8 | AFMT_S8 |
+	    AFMT_S16_LE |AFMT_S16_BE |
+	    AFMT_S24_LE | AFMT_S24_BE |
+	    AFMT_S32_LE | AFMT_S32_BE |
+	    AFMT_S24_PACKED);
+
+	return (0);
+}
+
+static int
+sndctl_dsp_channels(audio_client_t *c, int *chanp)
+{
+	int		rv;
+	int		nchan;
+	int		oflag;
+
+	oflag = auclnt_get_oflag(c);
+
+	nchan = *chanp;
+	if (nchan != 0) {
+		if (oflag & FWRITE) {
+			rv = auclnt_set_channels(auclnt_output_stream(c),
+			    nchan);
+			if (rv != 0)
+				return (rv);
+		}
+
+		if (oflag & FREAD) {
+			rv = auclnt_set_channels(auclnt_input_stream(c), nchan);
+			if (rv != 0)
+				return (rv);
+		}
+	}
+
+	if (oflag & FWRITE) {
+		nchan = auclnt_get_channels(auclnt_output_stream(c));
+	} else if (oflag & FREAD) {
+		nchan = auclnt_get_channels(auclnt_input_stream(c));
+	}
+	*chanp = nchan;
+	return (0);
+}
+
+static int
+sndctl_dsp_stereo(audio_client_t *c, int *onoff)
+{
+	int	nchan;
+
+	switch (*onoff) {
+	case 0:
+		nchan = 1;
+		break;
+	case 1:
+		nchan = 2;
+		break;
+	default:
+		return (EINVAL);
+	}
+
+	return (sndctl_dsp_channels(c, &nchan));
+}
+
+static int
+sndctl_dsp_post(audio_client_t *c)
+{
+	if (auclnt_get_oflag(c) & FWRITE) {
+		audio_stream_t	*sp = auclnt_output_stream(c);
+		auclnt_flush(sp);
+		auclnt_clear_paused(sp);
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_getcaps(audio_client_t *c, int *capsp)
+{
+	int		ncaps;
+	int		osscaps = 0;
+
+	ncaps = auclnt_get_dev_capab(auclnt_get_dev(c));
+
+	if (ncaps & AUDIO_CLIENT_CAP_PLAY)
+		osscaps |= PCM_CAP_OUTPUT;
+	if (ncaps & AUDIO_CLIENT_CAP_RECORD)
+		osscaps |= PCM_CAP_INPUT;
+	if (ncaps & AUDIO_CLIENT_CAP_DUPLEX)
+		osscaps |= PCM_CAP_DUPLEX;
+
+	*capsp = osscaps;
+	return (0);
+}
+
+static int
+sndctl_dsp_gettrigger(audio_client_t *c, int *trigp)
+{
+	int		triggers = 0;
+	int		oflag;
+
+	oflag = auclnt_get_oflag(c);
+
+	if (oflag & FWRITE) {
+		if (!auclnt_is_paused(auclnt_output_stream(c))) {
+			triggers |= PCM_ENABLE_OUTPUT;
+		}
+	}
+
+	if (oflag & FREAD) {
+		if (!auclnt_is_paused(auclnt_input_stream(c))) {
+			triggers |= PCM_ENABLE_INPUT;
+		}
+	}
+	*trigp = triggers;
+
+	return (0);
+}
+
+static int
+sndctl_dsp_settrigger(audio_client_t *c, int *trigp)
+{
+	int		triggers;
+	int		oflag;
+
+	oflag = auclnt_get_oflag(c);
+	triggers = *trigp;
+
+	if ((oflag & FWRITE) && (triggers & PCM_ENABLE_OUTPUT)) {
+		auclnt_clear_paused(auclnt_output_stream(c));
+	}
+
+	if ((oflag & FREAD) && (triggers & PCM_ENABLE_INPUT)) {
+		auclnt_clear_paused(auclnt_input_stream(c));
+	}
+
+	return (0);
+}
+
+struct oss_legacy_volume {
+	pid_t		pid;
+	uint8_t		ogain;
+	uint8_t		igain;
+};
+
+static int
+oss_legacy_volume_walker(audio_client_t *c, void *arg)
+{
+	struct oss_legacy_volume	 *olv = arg;
+
+	if (auclnt_get_pid(c) == olv->pid) {
+		if (olv->ogain <= 100) {
+			auclnt_set_gain(auclnt_output_stream(c), olv->ogain);
+		}
+		if (olv->igain <= 100) {
+			auclnt_set_gain(auclnt_input_stream(c), olv->igain);
+		}
+	}
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static void
+oss_set_legacy_volume(audio_client_t *c, uint8_t ogain, uint8_t igain)
+{
+	struct oss_legacy_volume olv;
+
+	olv.pid = auclnt_get_pid(c);
+	olv.ogain = ogain;
+	olv.igain = igain;
+	auclnt_dev_walk_clients(auclnt_get_dev(c),
+	    oss_legacy_volume_walker, &olv);
+}
+
+static int
+sndctl_dsp_getplayvol(audio_client_t *c, int *volp)
+{
+	int	vol;
+
+	/* convert monophonic soft value to OSS stereo value */
+	vol = auclnt_get_gain(auclnt_output_stream(c));
+	*volp = vol | (vol << 8);
+	return (0);
+}
+
+static int
+sndctl_dsp_setplayvol(audio_client_t *c, int *volp)
+{
+	uint8_t		vol;
+
+	vol = *volp & 0xff;
+	if (vol > 100) {
+		return (EINVAL);
+	}
+
+	auclnt_set_gain(auclnt_output_stream(c), vol);
+	*volp = (vol | (vol << 8));
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getrecvol(audio_client_t *c, int *volp)
+{
+	int	vol;
+
+	vol = auclnt_get_gain(auclnt_input_stream(c));
+	*volp = (vol | (vol << 8));
+	return (0);
+}
+
+static int
+sndctl_dsp_setrecvol(audio_client_t *c, int *volp)
+{
+	uint8_t		vol;
+
+	vol = *volp & 0xff;
+	if (vol > 100) {
+		return (EINVAL);
+	}
+
+	auclnt_set_gain(auclnt_input_stream(c), vol);
+	*volp = (vol | (vol << 8));
+
+	return (0);
+}
+
+static int
+sound_mixer_write_ogain(audio_client_t *c, int *volp)
+{
+	uint8_t		vol;
+
+	vol = *volp & 0xff;
+	if (vol > 100) {
+		return (EINVAL);
+	}
+	oss_set_legacy_volume(c, vol, 255);
+	*volp = (vol | (vol << 8));
+	return (0);
+}
+
+static int
+sound_mixer_write_igain(audio_client_t *c, int *volp)
+{
+	uint8_t		vol;
+
+	vol = *volp & 0xff;
+	if (vol > 100) {
+		return (EINVAL);
+	}
+	oss_set_legacy_volume(c, 255, vol);
+	*volp = (vol | (vol << 8));
+	return (0);
+}
+
+static int
+sndctl_dsp_readctl(audio_client_t *c, oss_digital_control *ctl)
+{
+	/* SPDIF: need to add support with spdif */
+	_NOTE(ARGUNUSED(c));
+	_NOTE(ARGUNUSED(ctl));
+	return (ENOTSUP);
+}
+
+static int
+sndctl_dsp_writectl(audio_client_t *c, oss_digital_control *ctl)
+{
+	/* SPDIF: need to add support with spdif */
+	_NOTE(ARGUNUSED(c));
+	_NOTE(ARGUNUSED(ctl));
+	return (ENOTSUP);
+}
+
+static int
+sndctl_dsp_cookedmode(audio_client_t *c, int *rvp)
+{
+	_NOTE(ARGUNUSED(c));
+
+	/* We are *always* in cooked mode -- at least until we have AC3. */
+	if (*rvp == 0) {
+		return (ENOTSUP);
+	} else {
+		return (0);
+	}
+}
+
+static int
+sndctl_dsp_silence(audio_client_t *c)
+{
+	if (auclnt_get_oflag(c) & FWRITE) {
+		audio_stream_t	*sp = auclnt_output_stream(c);
+		auclnt_set_paused(sp);
+		auclnt_flush(sp);
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_skip(audio_client_t *c)
+{
+	if (auclnt_get_oflag(c) & FWRITE) {
+		audio_stream_t	*sp = auclnt_output_stream(c);
+		auclnt_set_paused(sp);
+		auclnt_flush(sp);
+		auclnt_clear_paused(sp);
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_halt_input(audio_client_t *c)
+{
+	if (auclnt_get_oflag(c) & FREAD) {
+		audio_stream_t	*sp = auclnt_input_stream(c);
+		auclnt_set_paused(sp);
+		auclnt_flush(sp);
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_halt_output(audio_client_t *c)
+{
+	if (auclnt_get_oflag(c) & FWRITE) {
+		audio_stream_t	*sp = auclnt_output_stream(c);
+		auclnt_set_paused(sp);
+		auclnt_flush(sp);
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_halt(audio_client_t *c)
+{
+	(void) sndctl_dsp_halt_input(c);
+	(void) sndctl_dsp_halt_output(c);
+	return (0);
+}
+
+static int
+sndctl_dsp_sync(audio_client_t *c)
+{
+	return (auclnt_drain(c));
+}
+
+static int
+sndctl_dsp_setfragment(audio_client_t *c, int *fragp)
+{
+	_NOTE(ARGUNUSED(c));
+	_NOTE(ARGUNUSED(fragp));
+	/*
+	 * We don't really implement this "properly" at this time.
+	 * The problems with this ioctl are various: the API insists
+	 * that fragment sizes be a power of two -- and we can't cope
+	 * with accurately reporting fragment sizes in the face of
+	 * mixing and format conversion.
+	 *
+	 * Well behaved applications should really not use this API.
+	 *
+	 * According to the OSS API documentation, the values provided
+	 * are nothing more than a "hint" and not to be relied upon
+	 * anyway.  And we aren't obligated to report the actual
+	 * values back!
+	 */
+	return (0);
+}
+
+/*
+ * A word about recsrc, and playtgt ioctls: We don't allow ordinary DSP
+ * applications to change port configurations, because these could have a
+ * bad effect for other applications.  Instead, these settings have to
+ * be changed using the master mixer panel.  In order to make applications
+ * happy, we just present a single "default" source/target.
+ */
+static int
+sndctl_dsp_get_recsrc_names(audio_client_t *c, oss_mixer_enuminfo *ei)
+{
+	_NOTE(ARGUNUSED(c));
+
+	ei->nvalues = 1;
+	(void) snprintf(ei->strings, sizeof (ei->strings), "default");
+	ei->strindex[0] = 0;
+
+	return (0);
+}
+
+static int
+sndctl_dsp_get_recsrc(audio_client_t *c, int *srcp)
+{
+	_NOTE(ARGUNUSED(c));
+	*srcp = 0;
+	return (0);
+}
+
+static int
+sndctl_dsp_set_recsrc(audio_client_t *c, int *srcp)
+{
+	_NOTE(ARGUNUSED(c));
+	*srcp = 0;
+	return (0);
+}
+
+static int
+sndctl_dsp_get_playtgt_names(audio_client_t *c, oss_mixer_enuminfo *ei)
+{
+	_NOTE(ARGUNUSED(c));
+
+	ei->nvalues = 1;
+	(void) snprintf(ei->strings, sizeof (ei->strings), "default");
+	ei->strindex[0] = 0;
+
+	return (0);
+}
+
+static int
+sndctl_dsp_get_playtgt(audio_client_t *c, int *tgtp)
+{
+	_NOTE(ARGUNUSED(c));
+	*tgtp = 0;
+	return (0);
+}
+
+static int
+sndctl_dsp_set_playtgt(audio_client_t *c, int *tgtp)
+{
+	_NOTE(ARGUNUSED(c));
+	*tgtp = 0;
+	return (0);
+}
+
+static int
+sndctl_sysinfo(oss_sysinfo *si)
+{
+	bzero(si, sizeof (*si));
+	(void) snprintf(si->product, sizeof (si->product), "SunOS Audio");
+	(void) snprintf(si->version, sizeof (si->version), "4.0");
+	si->versionnum = OSS_VERSION;
+	si->numcards = oss_cnt_devs();
+	si->nummixers = si->numcards - 1;
+	si->numaudios = si->numcards - 1;
+	si->numaudioengines = si->numaudios;
+	(void) snprintf(si->license, sizeof (si->license), "CDDL");
+	return (0);
+}
+
+static int
+sndctl_cardinfo(audio_client_t *c, oss_card_info *ci)
+{
+	audio_dev_t	*d;
+	void		*iter;
+	const char 	*info;
+	int		n;
+	boolean_t	release;
+
+	if ((n = ci->card) == -1) {
+		release = B_FALSE;
+		d = auclnt_get_dev(c);
+		n = auclnt_get_dev_index(d);
+	} else {
+		release = B_TRUE;
+		d = auclnt_hold_dev_by_index(n);
+	}
+
+	bzero(ci, sizeof (*ci));
+	ci->card = n;
+
+	if (d == NULL) {
+		/*
+		 * If device removed (e.g. for DR), then
+		 * report a bogus removed entry.
+		 */
+		(void) snprintf(ci->shortname, sizeof (ci->shortname),
+		    "<removed>");
+		(void) snprintf(ci->longname, sizeof (ci->longname),
+		    "<removed>");
+		return (0);
+	}
+
+	(void) snprintf(ci->shortname, sizeof (ci->shortname),
+	    "%s", auclnt_get_dev_name(d));
+	(void) snprintf(ci->longname, sizeof (ci->longname),
+	    "%s (%s)", auclnt_get_dev_description(d),
+	    auclnt_get_dev_version(d));
+
+	iter = NULL;
+	while ((info = auclnt_get_dev_hw_info(d, &iter)) != NULL) {
+		(void) strlcat(ci->hw_info, info, sizeof (ci->hw_info));
+		(void) strlcat(ci->hw_info, "\n", sizeof (ci->hw_info));
+	}
+
+	/*
+	 * We don't report interrupt counts, ack counts (which are
+	 * just "read" interrupts, not spurious), or any other flags.
+	 * Nothing should be using any of this data anyway ... these
+	 * values were intended for 4Front's debugging purposes.  In
+	 * Solaris, drivers should use interrupt kstats to report
+	 * interrupt related statistics.
+	 */
+	if (release)
+		auclnt_release_dev(d);
+	return (0);
+}
+
+static int
+audioinfo_walker(audio_engine_t *e, void *a)
+{
+	oss_audioinfo *si = a;
+	int fmt, nchan, rate, cap;
+
+	fmt = auclnt_engine_get_format(e);
+	nchan = auclnt_engine_get_channels(e);
+	rate = auclnt_engine_get_rate(e);
+	cap = auclnt_engine_get_capab(e);
+
+	for (int i = 0; oss_formats[i].fmt != AUDIO_FORMAT_NONE; i++) {
+		if (fmt == oss_formats[i].fmt) {
+			if (cap & AUDIO_CLIENT_CAP_PLAY) {
+				si->oformats |= oss_formats[i].oss;
+			}
+			if (cap & AUDIO_CLIENT_CAP_RECORD) {
+				si->iformats |= oss_formats[i].oss;
+			}
+			break;
+		}
+	}
+	si->max_channels = max(nchan, si->max_channels);
+	si->max_rate = max(rate, si->max_rate);
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+sndctl_audioinfo(audio_client_t *c, oss_audioinfo *si)
+{
+	audio_dev_t		*d;
+	const char		*name;
+	int			n;
+	boolean_t		release;
+	unsigned		cap;
+
+	if ((n = si->dev) == -1) {
+		release = B_FALSE;
+		d = auclnt_get_dev(c);
+		n = auclnt_get_dev_index(d);
+	} else {
+		release = B_TRUE;
+		n++;	/* skip pseudo device */
+		d = auclnt_hold_dev_by_index(n);
+	}
+
+	bzero(si, sizeof (*si));
+	si->dev = n - 1;
+
+	if (d == NULL) {
+		/* if device not present, forge a false entry */
+		si->card_number = n;
+		si->mixer_dev = n - 1;
+		si->legacy_device = -1;
+		si->enabled = 0;
+		(void) snprintf(si->name, sizeof (si->name), "<removed>");
+		return (0);
+	}
+
+	name = auclnt_get_dev_name(d);
+	(void) snprintf(si->name, sizeof (si->name), "%s", name);
+
+	si->legacy_device = auclnt_get_dev_number(d);
+	si->caps = 0;
+
+	auclnt_dev_walk_engines(d, audioinfo_walker, si);
+
+	cap = auclnt_get_dev_capab(d);
+
+	if (cap	& AUDIO_CLIENT_CAP_DUPLEX) {
+		si->caps |= PCM_CAP_DUPLEX;
+	}
+	if (cap & AUDIO_CLIENT_CAP_PLAY) {
+		si->caps |= PCM_CAP_OUTPUT;
+	}
+	if (cap & AUDIO_CLIENT_CAP_RECORD) {
+		si->caps |= PCM_CAP_INPUT;
+	}
+
+	if (si->caps != 0) {
+		/* AC3: PCM_CAP_MULTI would be wrong for an AC3 only device */
+		si->caps |= PCM_CAP_BATCH | PCM_CAP_TRIGGER | PCM_CAP_MULTI;
+		/* MMAP: we add PCM_CAP_MMAP when we we support it */
+		si->enabled = 1;
+		si->rate_source = si->dev;
+
+		/* we can convert PCM formats */
+		if ((si->iformats | si->oformats) &
+		    AUDIO_FORMAT_PCM) {
+			si->min_channels = min(2, si->max_channels);
+			si->min_rate = min(5000, si->max_rate);
+			si->caps |= PCM_CAP_FREERATE;
+		}
+		(void) snprintf(si->devnode, sizeof (si->devnode),
+		    "/dev/sound/%s:%ddsp",
+		    auclnt_get_dev_driver(d), auclnt_get_dev_instance(d));
+	} else {
+		si->enabled = 0;	/* stops apps from using us directly */
+		si->caps = PCM_CAP_VIRTUAL;
+		(void) snprintf(si->devnode, sizeof (si->devnode),
+		    "/dev/sndstat");
+	}
+
+	si->pid = -1;
+	(void) snprintf(si->handle, sizeof (si->handle), "%s", name);
+	(void) snprintf(si->label, sizeof (si->label), "%s", name);
+	si->latency = -1;
+	si->card_number = n;
+	si->mixer_dev = n - 1;
+
+	if (release)
+		auclnt_release_dev(d);
+
+	return (0);
+}
+
+static int
+sound_mixer_info(audio_client_t *c, mixer_info *mi)
+{
+	audio_dev_t	*d;
+	ossdev_t	*odev;
+	ossclient_t	*sc;
+	const char	*name;
+
+	sc = auclnt_get_private(c);
+	odev = sc->o_ossdev;
+
+	d = auclnt_get_dev(c);
+
+	name = auclnt_get_dev_name(d);
+	(void) snprintf(mi->id, sizeof (mi->id), "%s", name);
+	(void) snprintf(mi->name, sizeof (mi->name), "%s", name);
+	(void) snprintf(mi->handle, sizeof (mi->handle), "%s", name);
+	mi->modify_counter = odev->d_modify_cnt;
+	mi->card_number = auclnt_get_dev_index(d);
+	mi->port_number = 0;
+	return (0);
+}
+
+static int
+sound_mixer_read_devmask(audio_client_t *c, int *devmask)
+{
+	_NOTE(ARGUNUSED(c));
+	*devmask = SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_IGAIN;
+	return (0);
+}
+
+static int
+sound_mixer_read_recmask(audio_client_t *c, int *recmask)
+{
+	_NOTE(ARGUNUSED(c));
+	*recmask = 0;
+	return (0);
+}
+
+static int
+sound_mixer_read_recsrc(audio_client_t *c, int *recsrc)
+{
+	_NOTE(ARGUNUSED(c));
+	*recsrc = 0;
+	return (0);
+}
+
+static int
+sound_mixer_read_caps(audio_client_t *c, int *caps)
+{
+	_NOTE(ARGUNUSED(c));
+	/* single recording source... sort of */
+	*caps = SOUND_CAP_EXCL_INPUT;
+	return (0);
+}
+
+static int
+sndctl_mixerinfo(audio_client_t *c, oss_mixerinfo *mi)
+{
+	audio_dev_t		*d;
+	ossdev_t 		*odev;
+	const char		*name;
+	int			n;
+	boolean_t		release = B_FALSE;
+
+	if ((n = mi->dev) == -1) {
+		release = B_FALSE;
+		d = auclnt_get_dev(c);
+		n = auclnt_get_dev_index(d);
+	} else {
+		release = B_TRUE;
+		n++;
+		d = auclnt_hold_dev_by_index(n);
+	}
+
+	bzero(mi, sizeof (*mi));
+	mi->dev = n - 1;
+
+	if (d == NULL) {
+		mi->card_number = n;
+		mi->enabled = 0;
+		mi->legacy_device = -1;
+		(void) snprintf(mi->name, sizeof (mi->name), "<removed>");
+		(void) snprintf(mi->id, sizeof (mi->id), "<removed>");
+		return (0);
+	}
+
+	if ((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) {
+		if (release)
+			auclnt_release_dev(d);
+		return (EINVAL);
+	}
+
+	name = auclnt_get_dev_name(d);
+	(void) snprintf(mi->name, sizeof (mi->name), "%s", name);
+	(void) snprintf(mi->id, sizeof (mi->id), "%s", name);
+	(void) snprintf(mi->handle, sizeof (mi->handle), "%s", name);
+	mi->modify_counter = odev->d_modify_cnt;
+	mi->card_number = auclnt_get_dev_index(d);
+	mi->legacy_device = auclnt_get_dev_number(d);
+	if (mi->legacy_device >= 0) {
+		(void) snprintf(mi->devnode, sizeof (mi->devnode),
+		    "/dev/sound/%s:%dmixer",
+		    auclnt_get_dev_driver(d), auclnt_get_dev_instance(d));
+		mi->enabled = 1;
+	} else {
+		/* special nodes use generic sndstat node */
+		(void) snprintf(mi->devnode, sizeof (mi->devnode),
+		    "/dev/sndstat");
+		mi->enabled = 0;
+	}
+	mi->nrext = odev->d_nctrl;
+
+	if (release)
+		auclnt_release_dev(d);
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getblksize(audio_client_t *c, int *fragsz)
+{
+	int	oflag = auclnt_get_oflag(c);
+
+	if (oflag & FWRITE)
+		*fragsz  = auclnt_get_fragsz(auclnt_output_stream(c));
+	else if (oflag & FREAD)
+		*fragsz  = auclnt_get_fragsz(auclnt_input_stream(c));
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getospace(audio_client_t *c, audio_buf_info *bi)
+{
+	audio_stream_t	*sp;
+	unsigned	n;
+
+	if ((auclnt_get_oflag(c) & FWRITE) == 0) {
+		return (EACCES);
+	}
+
+	sp = auclnt_output_stream(c);
+	n = auclnt_get_nframes(sp) - auclnt_get_count(sp);
+
+	bi->fragsize  = auclnt_get_fragsz(sp);
+	bi->fragstotal = auclnt_get_nfrags(sp);
+	bi->bytes = (n * auclnt_get_framesz(sp));
+	bi->fragments = bi->bytes / bi->fragsize;
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getispace(audio_client_t *c, audio_buf_info *bi)
+{
+	audio_stream_t	*sp;
+	unsigned	n;
+
+	if ((auclnt_get_oflag(c) & FREAD) == 0) {
+		return (EACCES);
+	}
+
+	sp = auclnt_input_stream(c);
+	n = auclnt_get_count(sp);
+
+	bi->fragsize  = auclnt_get_fragsz(sp);
+	bi->fragstotal = auclnt_get_nfrags(sp);
+	bi->bytes = (n * auclnt_get_framesz(sp));
+	bi->fragments = bi->bytes / bi->fragsize;
+
+	return (0);
+}
+
+static int
+sndctl_dsp_getodelay(audio_client_t *c, int *bytes)
+{
+	unsigned	framesz;
+	unsigned	slen, flen;
+
+	if (auclnt_get_oflag(c) & FWRITE) {
+		audio_stream_t	*sp = auclnt_output_stream(c);
+		framesz = auclnt_get_framesz(sp);
+		auclnt_get_output_qlen(c, &slen, &flen);
+		*bytes = (slen + flen) * framesz;
+	} else {
+		*bytes = 0;
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_current_iptr(audio_client_t *c, oss_count_t *count)
+{
+	if (auclnt_get_oflag(c) & FREAD) {
+		count->samples = auclnt_get_samples(auclnt_input_stream(c));
+		count->fifo_samples = 0;	/* not quite accurate */
+	} else {
+		count->samples = 0;
+		count->fifo_samples = 0;
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_current_optr(audio_client_t *c, oss_count_t *count)
+{
+	unsigned samples, fifo;
+
+	if (auclnt_get_oflag(c) & FWRITE) {
+		auclnt_get_output_qlen(c, &samples, &fifo);
+		count->samples = samples;
+		count->fifo_samples = fifo;
+	} else {
+		count->samples = 0;
+		count->fifo_samples = 0;
+	}
+	return (0);
+}
+
+static int
+sndctl_dsp_getoptr(audio_client_t *c, count_info *ci)
+{
+	audio_stream_t	*sp;
+	unsigned	framesz;
+	unsigned	fragsz;
+
+	bzero(ci, sizeof (*ci));
+	if ((auclnt_get_oflag(c) & FWRITE) == 0) {
+		return (0);
+	}
+	sp = auclnt_output_stream(c);
+	framesz = auclnt_get_framesz(sp);
+	fragsz = auclnt_get_fragsz(sp);
+	ci->blocks = auclnt_get_samples(sp) * framesz / fragsz;
+	auclnt_set_samples(sp, 0);
+	ci->bytes = auclnt_get_tail(sp) * framesz;
+	ci->ptr = auclnt_get_tidx(sp) * framesz;
+	return (0);
+}
+
+static int
+sndctl_dsp_getiptr(audio_client_t *c, count_info *ci)
+{
+	audio_stream_t	*sp;
+	unsigned	framesz;
+	unsigned	fragsz;
+
+	bzero(ci, sizeof (*ci));
+	if ((auclnt_get_oflag(c) & FREAD) == 0) {
+		return (0);
+	}
+	sp = auclnt_input_stream(c);
+	framesz = auclnt_get_framesz(sp);
+	fragsz = auclnt_get_fragsz(sp);
+	ci->blocks = auclnt_get_samples(sp) * framesz / fragsz;
+	auclnt_set_samples(sp, 0);
+	ci->bytes = auclnt_get_head(sp) * framesz;
+	ci->ptr = auclnt_get_hidx(sp) * framesz;
+	return (0);
+}
+
+static int
+sndctl_dsp_geterror(audio_client_t *c, audio_errinfo *bi)
+{
+	audio_stream_t	*sp;
+	unsigned	fragsz;
+	/*
+	 * Note: The use of this structure is unsafe... different
+	 * meanings for error codes are used by different implementations,
+	 * according to the spec.  (Even different versions of the same
+	 * implementation could have different values.)
+	 *
+	 * Rather than try to come up with a reliable solution here, we
+	 * don't use it.  If you want to report errors, or see the result
+	 * of errors, use syslog.
+	 */
+	bzero(bi, sizeof (*bi));
+
+	sp = auclnt_output_stream(c);
+	fragsz = max(auclnt_get_fragsz(sp), 1);
+	bi->play_underruns = (int)((auclnt_get_errors(sp) + (fragsz - 1)) /
+	    fragsz);
+	auclnt_set_errors(sp, 0);
+
+	sp = auclnt_input_stream(c);
+	fragsz = max(auclnt_get_fragsz(sp), 1);
+	bi->rec_overruns = (int)((auclnt_get_errors(sp) + (fragsz - 1)) /
+	    fragsz);
+	auclnt_set_errors(sp, 0);
+
+	return (0);
+}
+
+static int
+sndctl_sun_send_number(audio_client_t *c, int *num, cred_t *cr)
+{
+	audio_dev_t	*dev;
+	int		rv;
+
+	if ((rv = drv_priv(cr)) != 0) {
+		return (rv);
+	}
+
+	dev = auclnt_get_dev(c);
+	auclnt_set_dev_number(dev, *num);
+	return (0);
+}
+
+static int
+oss_getversion(int *versp)
+{
+	*versp = OSS_VERSION;
+	return (0);
+}
+
+static int
+oss_ioctl(audio_client_t *c, int cmd, intptr_t arg, int mode, cred_t *credp,
+    int *rvalp)
+{
+	int	sz;
+	void	*data;
+	int	rv = 0;
+
+	_NOTE(ARGUNUSED(credp));
+
+	sz = OSSIOC_GETSZ(cmd);
+
+	if ((cmd & (OSSIOC_IN | OSSIOC_OUT)) && sz) {
+		if ((data = kmem_zalloc(sz, KM_NOSLEEP)) == NULL) {
+			return (ENOMEM);
+		}
+	} else {
+		sz = 0;
+	}
+
+	if (cmd & OSSIOC_IN) {
+		if ((rv = ddi_copyin((void *)arg, data, sz, mode)) != 0) {
+			goto done;
+		}
+	}
+
+	switch (cmd) {
+		/*
+		 * DSP specific ioctls
+		 */
+	case SNDCTL_DSP_HALT:
+		rv = sndctl_dsp_halt(c);
+		break;
+
+	case SNDCTL_DSP_SYNC:
+		rv = sndctl_dsp_sync(c);
+		break;
+
+	case SNDCTL_DSP_SPEED:
+		rv = sndctl_dsp_speed(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SETFMT:
+		rv = sndctl_dsp_setfmt(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETFMTS:
+		rv = sndctl_dsp_getfmts(c, (int *)data);
+		break;
+	case SNDCTL_DSP_STEREO:
+		rv = sndctl_dsp_stereo(c, (int *)data);
+		break;
+	case SNDCTL_DSP_CHANNELS:
+		rv = sndctl_dsp_channels(c, (int *)data);
+		break;
+	case SNDCTL_DSP_POST:
+		rv = sndctl_dsp_post(c);
+		break;
+	case SNDCTL_DSP_GETCAPS:
+		rv = sndctl_dsp_getcaps(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETTRIGGER:
+		rv = sndctl_dsp_gettrigger(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SETTRIGGER:
+		rv = sndctl_dsp_settrigger(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETPLAYVOL:
+	case SOUND_MIXER_READ_VOLUME:	/* legacy mixer on dsp */
+	case SOUND_MIXER_READ_PCM:	/* legacy mixer on dsp */
+	case SOUND_MIXER_READ_OGAIN:	/* legacy mixer on dsp */
+		rv = sndctl_dsp_getplayvol(c, (int *)data);
+		break;
+	case SOUND_MIXER_WRITE_VOLUME:	/* legacy mixer on dsp */
+	case SOUND_MIXER_WRITE_PCM:	/* legacy mixer on dsp */
+	case SOUND_MIXER_WRITE_OGAIN:	/* legacy mixer on dsp */
+		rv = sound_mixer_write_ogain(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SETPLAYVOL:
+		rv = sndctl_dsp_setplayvol(c, (int *)data);
+		break;
+	case SNDCTL_DSP_READCTL:
+		rv = sndctl_dsp_readctl(c, (oss_digital_control *)data);
+		break;
+	case SNDCTL_DSP_WRITECTL:
+		rv = sndctl_dsp_writectl(c, (oss_digital_control *)data);
+		break;
+	case SNDCTL_DSP_COOKEDMODE:
+		rv = sndctl_dsp_cookedmode(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SILENCE:
+		rv = sndctl_dsp_silence(c);
+		break;
+	case SNDCTL_DSP_SKIP:
+		rv = sndctl_dsp_skip(c);
+		break;
+	case SNDCTL_DSP_HALT_INPUT:
+		rv = sndctl_dsp_halt_input(c);
+		break;
+	case SNDCTL_DSP_HALT_OUTPUT:
+		rv = sndctl_dsp_halt_output(c);
+		break;
+	case SNDCTL_DSP_GET_RECSRC_NAMES:
+		rv = sndctl_dsp_get_recsrc_names(c, (oss_mixer_enuminfo *)data);
+		break;
+	case SNDCTL_DSP_SUBDIVIDE:
+	case SNDCTL_DSP_SETFRAGMENT:
+		rv = sndctl_dsp_setfragment(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GET_RECSRC:
+		rv = sndctl_dsp_get_recsrc(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SET_RECSRC:
+		rv = sndctl_dsp_set_recsrc(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GET_PLAYTGT_NAMES:
+		rv = sndctl_dsp_get_playtgt_names(c,
+		    (oss_mixer_enuminfo *)data);
+		break;
+	case SNDCTL_DSP_GET_PLAYTGT:
+		rv = sndctl_dsp_get_playtgt(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SET_PLAYTGT:
+		rv = sndctl_dsp_set_playtgt(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETRECVOL:
+	case SOUND_MIXER_READ_RECGAIN:	/* legacy mixer on dsp */
+	case SOUND_MIXER_READ_RECLEV:	/* legacy mixer on dsp */
+	case SOUND_MIXER_READ_IGAIN:	/* legacy mixer on dsp */
+		rv = sndctl_dsp_getrecvol(c, (int *)data);
+		break;
+	case SOUND_MIXER_WRITE_RECGAIN:	/* legacy mixer on dsp */
+	case SOUND_MIXER_WRITE_RECLEV:	/* legacy mixer on dsp */
+	case SOUND_MIXER_WRITE_IGAIN:	/* legacy mixer on dsp */
+		rv = sound_mixer_write_igain(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SETRECVOL:
+		rv = sndctl_dsp_setrecvol(c, (int *)data);
+		break;
+	case SNDCTL_DSP_SETDUPLEX:	/* Ignored */
+	case SNDCTL_DSP_LOW_WATER:	/* Ignored */
+	case SNDCTL_DSP_POLICY:		/* Ignored */
+	case SNDCTL_DSP_PROFILE:	/* Ignored */
+		rv = 0;
+		break;
+	case SNDCTL_DSP_GETBLKSIZE:
+		rv = sndctl_dsp_getblksize(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETOSPACE:
+		rv = sndctl_dsp_getospace(c, (audio_buf_info *)data);
+		break;
+	case SNDCTL_DSP_GETISPACE:
+		rv = sndctl_dsp_getispace(c, (audio_buf_info *)data);
+		break;
+	case SNDCTL_DSP_GETODELAY:
+		rv = sndctl_dsp_getodelay(c, (int *)data);
+		break;
+	case SNDCTL_DSP_GETOPTR:
+		rv = sndctl_dsp_getoptr(c, (count_info *)data);
+		break;
+	case SNDCTL_DSP_GETIPTR:
+		rv = sndctl_dsp_getiptr(c, (count_info *)data);
+		break;
+	case SNDCTL_DSP_GETERROR:
+		rv = sndctl_dsp_geterror(c, (audio_errinfo *)data);
+		break;
+	case SNDCTL_DSP_CURRENT_IPTR:
+		rv = sndctl_dsp_current_iptr(c, (oss_count_t *)data);
+		break;
+	case SNDCTL_DSP_CURRENT_OPTR:
+		rv = sndctl_dsp_current_optr(c, (oss_count_t *)data);
+		break;
+
+		/*
+		 * Shared ioctls with /dev/mixer.
+		 */
+	case OSS_GETVERSION:
+		rv = oss_getversion((int *)data);
+		break;
+	case SNDCTL_CARDINFO:
+		rv = sndctl_cardinfo(c, (oss_card_info *)data);
+		break;
+	case SNDCTL_ENGINEINFO:
+	case SNDCTL_AUDIOINFO:
+	case SNDCTL_AUDIOINFO_EX:
+		rv = sndctl_audioinfo(c, (oss_audioinfo *)data);
+		break;
+	case SNDCTL_SYSINFO:
+		rv = sndctl_sysinfo((oss_sysinfo *)data);
+		break;
+	case SNDCTL_MIXERINFO:
+		rv = sndctl_mixerinfo(c, (oss_mixerinfo *)data);
+		break;
+	case SOUND_MIXER_INFO:
+		rv = sound_mixer_info(c, (mixer_info *)data);
+		break;
+
+		/*
+		 * These are mixer ioctls that are virtualized for the DSP
+		 * device.  They are accessible via either /dev/mixer or
+		 * /dev/dsp.
+		 */
+	case SOUND_MIXER_READ_RECSRC:
+	case SOUND_MIXER_WRITE_RECSRC:
+		rv = sound_mixer_read_recsrc(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_DEVMASK:
+	case SOUND_MIXER_READ_STEREODEVS:
+		rv = sound_mixer_read_devmask(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_RECMASK:
+		rv = sound_mixer_read_recmask(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_CAPS:
+		rv = sound_mixer_read_caps(c, (int *)data);
+		break;
+
+		/*
+		 * Ioctls we have chosen not to support for now.  Some
+		 * of these are of legacy interest only.
+		 */
+	case SNDCTL_SETSONG:
+	case SNDCTL_GETSONG:
+	case SNDCTL_DSP_SYNCGROUP:
+	case SNDCTL_DSP_SYNCSTART:
+	case SNDCTL_DSP_GET_CHNORDER:
+	case SNDCTL_DSP_SET_CHNORDER:
+	case SNDCTL_DSP_GETIPEAKS:
+	case SNDCTL_DSP_GETOPEAKS:
+	case SNDCTL_DSP_GETCHANNELMASK:
+	case SNDCTL_DSP_BIND_CHANNEL:
+	case SNDCTL_DSP_SETSYNCRO:
+	default:
+		rv = EINVAL;
+		break;
+	}
+
+	if ((rv == 0) && (cmd & OSSIOC_OUT)) {
+		rv = ddi_copyout(data, (void *)arg, sz, mode);
+	}
+	if (rv == 0) {
+		*rvalp = 0;
+	}
+
+done:
+	if (sz) {
+		kmem_free(data, sz);
+	}
+	return (rv);
+}
+
+static void
+oss_output(audio_client_t *c)
+{
+	auclnt_pollwakeup(c, POLLOUT);
+}
+
+static void
+oss_input(audio_client_t *c)
+{
+	auclnt_pollwakeup(c, POLLIN | POLLRDNORM);
+}
+
+static void
+oss_notify(audio_client_t *c)
+{
+	audio_dev_t	*d;
+	ossdev_t	*odev;
+
+	d = auclnt_get_dev(c);
+	if ((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) {
+		return;
+	}
+	odev->d_modify_cnt++;
+}
+
+static int
+ossmix_open(audio_client_t *c, int oflag)
+{
+	int		rv;
+	ossclient_t	*sc;
+	ossdev_t	*odev;
+
+	_NOTE(ARGUNUSED(oflag));
+
+	if ((rv = auclnt_open(c, AUDIO_FORMAT_NONE, 0)) != 0) {
+		return (rv);
+	}
+
+	if ((sc = kmem_zalloc(sizeof (*sc), KM_NOSLEEP)) == NULL) {
+		return (ENOMEM);
+	}
+	sc->o_ss_sz = 8192;
+	if ((sc->o_ss_buf = kmem_zalloc(sc->o_ss_sz, KM_NOSLEEP)) == NULL) {
+		kmem_free(sc, sizeof (*sc));
+		return (ENOMEM);
+	}
+	auclnt_set_private(c, sc);
+
+	odev = auclnt_get_minor_data(c, AUDIO_MINOR_DSP);
+
+	/* set a couple of common fields */
+	sc->o_client = c;
+	sc->o_ossdev = odev;
+
+	return (rv);
+}
+
+static void
+ossmix_close(audio_client_t *c)
+{
+	ossclient_t	*sc;
+
+	sc = auclnt_get_private(c);
+
+	kmem_free(sc->o_ss_buf, sc->o_ss_sz);
+	kmem_free(sc, sizeof (*sc));
+
+	auclnt_close(c);
+}
+
+static int
+sndctl_mix_nrext(audio_client_t *c, int *ncp)
+{
+	audio_dev_t	*d;
+	ossdev_t	*odev;
+
+	d = auclnt_get_dev(c);
+
+	if ((*ncp != -1) && (*ncp != (auclnt_get_dev_index(d) - 1))) {
+		return (ENXIO);
+	}
+
+	if ((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) {
+		return (EINVAL);
+	}
+
+	*ncp = odev->d_nctrl;
+
+	return (0);
+}
+
+static int
+sndctl_mix_extinfo(audio_client_t *c, oss_mixext *pext)
+{
+	audio_dev_t		*d;
+	ossdev_t		*odev;
+	int			rv = 0;
+	int			dev;
+
+	d = auclnt_get_dev(c);
+
+	if (((dev = pext->dev) != -1) && (dev != (auclnt_get_dev_index(d) - 1)))
+		return (ENXIO);
+
+	if (((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) ||
+	    (pext->ctrl >= odev->d_nctrl)) {
+		return (EINVAL);
+	}
+
+	bcopy(&odev->d_exts[pext->ctrl], pext, sizeof (*pext));
+	pext->enumbit = 0;
+	pext->dev = dev;
+
+	return (rv);
+}
+
+static int
+sndctl_mix_enuminfo(audio_client_t *c, oss_mixer_enuminfo *ei)
+{
+	audio_dev_t		*d;
+	audio_ctrl_desc_t	desc;
+	audio_ctrl_t		*ctrl;
+	ossdev_t		*odev;
+	uint64_t		mask;
+	int			bit;
+	ushort_t		nxt;
+
+	d = auclnt_get_dev(c);
+
+	if ((ei->dev != -1) && (ei->dev != (auclnt_get_dev_index(d) - 1)))
+		return (ENXIO);
+
+	if (((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) ||
+	    (ei->ctrl >= odev->d_nctrl) ||
+	    (odev->d_exts[ei->ctrl].type != MIXT_ENUM) ||
+	    ((ctrl = odev->d_ctrls[ei->ctrl]) == NULL) ||
+	    (auclnt_control_describe(ctrl, &desc) != 0)) {
+		return (EINVAL);
+	}
+
+	mask = desc.acd_maxvalue;
+	bit = 0;
+	nxt = 0;
+	ei->nvalues = 0;
+	bzero(ei->strings, sizeof (ei->strings));
+	bzero(ei->strindex, sizeof (ei->strindex));
+
+	while (mask) {
+		const char *name = desc.acd_enum[bit];
+		nxt = oss_set_enum(ei, nxt, name ? name : "");
+		mask >>= 1;
+		bit++;
+	}
+
+	return (0);
+}
+
+static int
+sndctl_mix_read(audio_client_t *c, oss_mixer_value *vr)
+{
+	int			rv;
+	uint64_t		v;
+	audio_dev_t		*d;
+	audio_ctrl_t		*ctrl;
+	ossdev_t		*odev;
+
+	d = auclnt_get_dev(c);
+
+	if ((vr->dev != -1) && (vr->dev != (auclnt_get_dev_index(d) - 1)))
+		return (ENXIO);
+
+	if (((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) ||
+	    (vr->ctrl >= odev->d_nctrl) ||
+	    ((ctrl = odev->d_ctrls[vr->ctrl]) == NULL)) {
+		return (EINVAL);
+	}
+	if ((rv = auclnt_control_read(ctrl, &v)) == 0) {
+		switch (odev->d_exts[vr->ctrl].type) {
+		case MIXT_ENUM:
+			/* translate this from an enum style bit mask */
+			vr->value = ddi_ffs((unsigned long)v) - 1;
+			break;
+		case MIXT_STEREOSLIDER:
+			vr->value = (int)ddi_swap16(v & 0xffff);
+			break;
+		case MIXT_MONOSLIDER:
+			vr->value = (int)(v | (v << 8));
+			break;
+		case MIXT_ONOFF:
+			/* this could be simple, or could be part of a multi */
+			if (odev->d_exts[vr->ctrl].enumbit >= 0) {
+				uint64_t mask;
+				mask = 1;
+				mask <<= (odev->d_exts[vr->ctrl].enumbit);
+				vr->value = (v & mask) ? 1 : 0;
+			} else {
+				vr->value = v ? 1 : 0;
+			}
+			break;
+
+		default:
+			vr->value = (int)v;
+			break;
+		}
+	}
+
+	return (rv);
+}
+
+static int
+sndctl_mix_write(audio_client_t *c, oss_mixer_value *vr)
+{
+	int			rv;
+	uint64_t		v;
+	audio_dev_t		*d;
+	audio_ctrl_t		*ctrl;
+	ossdev_t		*odev;
+
+	d = auclnt_get_dev(c);
+
+	if ((vr->dev != -1) && (vr->dev != (auclnt_get_dev_index(d) - 1)))
+		return (ENXIO);
+
+	if (((odev = auclnt_get_dev_minor_data(d, AUDIO_MINOR_DSP)) == NULL) ||
+	    (vr->ctrl >= odev->d_nctrl) ||
+	    ((ctrl = odev->d_ctrls[vr->ctrl]) == NULL)) {
+		return (EINVAL);
+	}
+
+	switch (odev->d_exts[vr->ctrl].type) {
+	case MIXT_ONOFF:
+		/* this could be standalone, or it could be part of a multi */
+		if (odev->d_exts[vr->ctrl].enumbit >= 0) {
+			uint64_t mask;
+			if ((rv = auclnt_control_read(ctrl, &v)) != 0) {
+				return (EINVAL);
+			}
+			mask = 1;
+			mask <<= (odev->d_exts[vr->ctrl].enumbit);
+			if (vr->value) {
+				v |= mask;
+			} else {
+				v &= ~mask;
+			}
+		} else {
+			v = vr->value;
+		}
+		break;
+	case MIXT_ENUM:
+		/* translate this to an enum style bit mask */
+		v = 1U << vr->value;
+		break;
+	case MIXT_MONOSLIDER:
+		/* mask off high order bits */
+		v = vr->value & 0xff;
+		break;
+	case MIXT_STEREOSLIDER:
+		/* OSS uses reverse byte ordering */
+		v = vr->value;
+		v = ddi_swap16(vr->value & 0xffff);
+		break;
+	default:
+		v = vr->value;
+	}
+	rv = auclnt_control_write(ctrl, v);
+
+	return (rv);
+}
+
+static int
+sndctl_mix_nrmix(audio_client_t *c, int *nmixp)
+{
+	_NOTE(ARGUNUSED(c));
+	*nmixp = oss_cnt_devs() - 1;
+	return (0);
+}
+
+static int
+ossmix_ioctl(audio_client_t *c, int cmd, intptr_t arg, int mode, cred_t *credp,
+    int *rvalp)
+{
+	int	sz;
+	void	*data;
+	int	rv = 0;
+
+	sz = OSSIOC_GETSZ(cmd);
+
+	if ((cmd & (OSSIOC_IN | OSSIOC_OUT)) && sz) {
+		if ((data = kmem_zalloc(sz, KM_NOSLEEP)) == NULL) {
+			return (ENOMEM);
+		}
+	} else {
+		sz = 0;
+	}
+
+	if (cmd & OSSIOC_IN) {
+		if ((rv = ddi_copyin((void *)arg, data, sz, mode)) != 0) {
+			goto done;
+		}
+	}
+
+	switch (cmd) {
+		/*
+		 * Mixer specific ioctls
+		 */
+	case SNDCTL_MIX_NREXT:
+		rv = sndctl_mix_nrext(c, (int *)data);
+		break;
+	case SNDCTL_MIX_EXTINFO:
+		rv = sndctl_mix_extinfo(c, (oss_mixext *)data);
+		break;
+	case SNDCTL_MIX_ENUMINFO:
+		rv = sndctl_mix_enuminfo(c, (oss_mixer_enuminfo *)data);
+		break;
+	case SNDCTL_MIX_READ:
+		rv = sndctl_mix_read(c, (oss_mixer_value *)data);
+		break;
+	case SNDCTL_MIX_WRITE:
+		rv = sndctl_mix_write(c, (oss_mixer_value *)data);
+		break;
+	case SNDCTL_MIX_NRMIX:
+		rv = sndctl_mix_nrmix(c, (int *)data);
+		break;
+
+		/*
+		 * Legacy ioctls.  These are treated as soft values only,
+		 * and do not affect global hardware state.  For use by
+		 * legacy DSP applications.
+		 */
+	case SOUND_MIXER_READ_VOLUME:
+	case SOUND_MIXER_READ_PCM:
+	case SOUND_MIXER_READ_OGAIN:
+		rv = sndctl_dsp_getplayvol(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_WRITE_VOLUME:
+	case SOUND_MIXER_WRITE_PCM:
+	case SOUND_MIXER_WRITE_OGAIN:
+		rv = sound_mixer_write_ogain(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_RECGAIN:
+	case SOUND_MIXER_READ_RECLEV:
+	case SOUND_MIXER_READ_IGAIN:
+		rv = sndctl_dsp_getrecvol(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_WRITE_RECGAIN:
+	case SOUND_MIXER_WRITE_RECLEV:
+	case SOUND_MIXER_WRITE_IGAIN:
+		rv = sound_mixer_write_igain(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_RECSRC:
+	case SOUND_MIXER_WRITE_RECSRC:
+		rv = sound_mixer_read_recsrc(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_DEVMASK:
+	case SOUND_MIXER_READ_STEREODEVS:
+		rv = sound_mixer_read_devmask(c, (int *)data);
+		break;
+
+	case SOUND_MIXER_READ_RECMASK:
+		rv = sound_mixer_read_recmask(c, (int *)data);
+		break;
+
+		/*
+		 * Common ioctls shared with DSP
+		 */
+	case OSS_GETVERSION:
+		rv = oss_getversion((int *)data);
+		break;
+
+	case SNDCTL_CARDINFO:
+		rv = sndctl_cardinfo(c, (oss_card_info *)data);
+		break;
+
+	case SNDCTL_ENGINEINFO:
+	case SNDCTL_AUDIOINFO:
+	case SNDCTL_AUDIOINFO_EX:
+		rv = sndctl_audioinfo(c, (oss_audioinfo *)data);
+		break;
+
+	case SNDCTL_SYSINFO:
+		rv = sndctl_sysinfo((oss_sysinfo *)data);
+		break;
+
+	case SNDCTL_MIXERINFO:
+		rv = sndctl_mixerinfo(c, (oss_mixerinfo *)data);
+		break;
+
+	case SOUND_MIXER_INFO:
+		rv = sound_mixer_info(c, (mixer_info *)data);
+		break;
+
+	case SNDCTL_MIX_DESCRIPTION:	/* NOT SUPPORTED: tooltip */
+		rv = EIO;	/* OSS returns EIO for this one */
+		break;
+
+		/*
+		 * Special implementation-private ioctls.
+		 */
+	case SNDCTL_SUN_SEND_NUMBER:
+		rv = sndctl_sun_send_number(c, (int *)data, credp);
+		break;
+
+		/*
+		 * Legacy ioctls we don't support.
+		 */
+	case SOUND_MIXER_WRITE_MONGAIN:
+	case SOUND_MIXER_READ_MONGAIN:
+	case SOUND_MIXER_READ_BASS:
+	case SOUND_MIXER_READ_TREBLE:
+	case SOUND_MIXER_READ_SPEAKER:
+	case SOUND_MIXER_READ_LINE:
+	case SOUND_MIXER_READ_MIC:
+	case SOUND_MIXER_READ_CD:
+	case SOUND_MIXER_READ_IMIX:
+	case SOUND_MIXER_READ_ALTPCM:
+	case SOUND_MIXER_READ_SYNTH:
+	case SOUND_MIXER_READ_LINE1:
+	case SOUND_MIXER_READ_LINE2:
+	case SOUND_MIXER_READ_LINE3:
+	case SOUND_MIXER_WRITE_BASS:
+	case SOUND_MIXER_WRITE_TREBLE:
+	case SOUND_MIXER_WRITE_SPEAKER:
+	case SOUND_MIXER_WRITE_LINE:
+	case SOUND_MIXER_WRITE_MIC:
+	case SOUND_MIXER_WRITE_CD:
+	case SOUND_MIXER_WRITE_IMIX:
+	case SOUND_MIXER_WRITE_ALTPCM:
+	case SOUND_MIXER_WRITE_SYNTH:
+	case SOUND_MIXER_WRITE_LINE1:
+	case SOUND_MIXER_WRITE_LINE2:
+	case SOUND_MIXER_WRITE_LINE3:
+		/*
+		 * Additional ioctls we *could* support, but don't.
+		 */
+	case SNDCTL_SETSONG:
+	case SNDCTL_SETLABEL:
+	case SNDCTL_GETSONG:
+	case SNDCTL_GETLABEL:
+	case SNDCTL_MIDIINFO:
+	case SNDCTL_SETNAME:
+	default:
+		rv = EINVAL;
+		break;
+	}
+
+	if ((rv == 0) && (cmd & OSSIOC_OUT)) {
+		rv = ddi_copyout(data, (void *)arg, sz, mode);
+	}
+	if (rv == 0) {
+		*rvalp = 0;
+	}
+
+done:
+	if (sz) {
+		kmem_free(data, sz);
+	}
+	return (rv);
+}
+
+static void *
+oss_dev_init(audio_dev_t *dev)
+{
+	ossdev_t	*odev;
+
+	odev = kmem_zalloc(sizeof (*odev), KM_SLEEP);
+	odev->d_dev = dev;
+
+	mutex_init(&odev->d_mx, NULL, MUTEX_DRIVER, NULL);
+	cv_init(&odev->d_cv, NULL, CV_DRIVER, NULL);
+	oss_alloc_controls(odev);
+
+	return (odev);
+}
+
+static void
+oss_dev_fini(void *arg)
+{
+	ossdev_t	*odev = arg;
+
+	if (odev != NULL) {
+		oss_free_controls(odev);
+		mutex_destroy(&odev->d_mx);
+		cv_destroy(&odev->d_cv);
+		kmem_free(odev, sizeof (*odev));
+	}
+}
+
+static void
+sndstat_printf(ossclient_t *oc, const char *fmt, ...)
+{
+	va_list	va;
+
+	va_start(va, fmt);
+	(void) vsnprintf(oc->o_ss_buf + oc->o_ss_len,
+	    oc->o_ss_sz - oc->o_ss_len, fmt, va);
+	va_end(va);
+	oc->o_ss_len = strlen(oc->o_ss_buf);
+}
+
+static int
+sndstat_dev_walker(audio_dev_t *d, void *arg)
+{
+	ossclient_t	*oc = arg;
+	const char	*capstr;
+	unsigned	cap;
+
+	cap = auclnt_get_dev_capab(d);
+
+	if (cap	& AUDIO_CLIENT_CAP_DUPLEX) {
+		capstr = "DUPLEX";
+	} else if ((cap & AUDIO_CLIENT_CAP_PLAY) &&
+	    (cap & AUDIO_CLIENT_CAP_RECORD)) {
+		capstr = "INPUT,OUTPUT";
+	} else if (cap & AUDIO_CLIENT_CAP_PLAY) {
+		capstr = "OUTPUT";
+	} else if (cap & AUDIO_CLIENT_CAP_RECORD) {
+		capstr = "INPUT";
+	} else {
+		capstr = NULL;
+	}
+
+	if (capstr == NULL)
+		return (AUDIO_WALK_CONTINUE);
+
+	sndstat_printf(oc, "%d: %s %s, %s (%s)\n",
+	    auclnt_get_dev_number(d), auclnt_get_dev_name(d),
+	    auclnt_get_dev_description(d), auclnt_get_dev_version(d), capstr);
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+sndstat_mixer_walker(audio_dev_t *d, void *arg)
+{
+	ossclient_t	*oc = arg;
+	unsigned	cap;
+	void		*iter;
+	const char	*info;
+
+	cap = auclnt_get_dev_capab(d);
+
+	if ((cap & (AUDIO_CLIENT_CAP_PLAY|AUDIO_CLIENT_CAP_RECORD)) == 0)
+		return (AUDIO_WALK_CONTINUE);
+
+	sndstat_printf(oc, "%d: %s %s, %s\n",
+	    auclnt_get_dev_number(d), auclnt_get_dev_name(d),
+	    auclnt_get_dev_description(d), auclnt_get_dev_version(d));
+	iter = NULL;
+	while ((info = auclnt_get_dev_hw_info(d, &iter)) != NULL) {
+		sndstat_printf(oc, "\t%s\n", info);
+	}
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+ossmix_write(audio_client_t *c, struct uio *uio, cred_t *cr)
+{
+	/* write on sndstat is a no-op */
+	_NOTE(ARGUNUSED(c));
+	_NOTE(ARGUNUSED(uio));
+	_NOTE(ARGUNUSED(cr));
+
+	return (0);
+}
+
+static int
+ossmix_read(audio_client_t *c, struct uio *uio, cred_t *cr)
+{
+	ossclient_t	*oc;
+	unsigned	n;
+	int		rv;
+
+	_NOTE(ARGUNUSED(cr));
+
+	if (uio->uio_resid == 0) {
+		return (0);
+	}
+
+	oc = auclnt_get_private(c);
+
+	mutex_enter(&oc->o_ss_lock);
+
+	if (oc->o_ss_off == 0) {
+
+		sndstat_printf(oc, "SunOS Audio Framework\n");
+
+		sndstat_printf(oc, "\nAudio Devices:\n");
+		auclnt_walk_devs_by_number(sndstat_dev_walker, oc);
+
+		sndstat_printf(oc, "\nMixers:\n");
+		auclnt_walk_devs_by_number(sndstat_mixer_walker, oc);
+	}
+
+	/*
+	 * For simplicity's sake, we implement a non-seekable device.  We could
+	 * support seekability, but offsets would be rather meaningless between
+	 * changes.
+	 */
+	n = min(uio->uio_resid, (oc->o_ss_len - oc->o_ss_off));
+
+	rv = uiomove(oc->o_ss_buf + oc->o_ss_off, n, UIO_READ, uio);
+	if (rv != 0) {
+		n = 0;
+	}
+	oc->o_ss_off += n;
+
+	if (n == 0) {
+		/*
+		 * end-of-file reached... clear the sndstat buffer so that
+		 * subsequent reads will get the latest data.
+		 */
+		oc->o_ss_off = oc->o_ss_len = 0;
+	}
+	mutex_exit(&oc->o_ss_lock);
+	return (rv);
+}
+
+int
+oss_read(audio_client_t *c, struct uio *uio, cred_t *cr)
+{
+	_NOTE(ARGUNUSED(cr));
+
+	auclnt_clear_paused(auclnt_input_stream(c));
+
+	return (auclnt_read(c, uio));
+}
+
+int
+oss_write(audio_client_t *c, struct uio *uio, cred_t *cr)
+{
+	_NOTE(ARGUNUSED(cr));
+
+	auclnt_clear_paused(auclnt_output_stream(c));
+
+	return (auclnt_write(c, uio));
+}
+
+int
+oss_chpoll(audio_client_t *c, short events, int anyyet, short *reventsp,
+    struct pollhead **phpp)
+{
+	return (auclnt_chpoll(c, events, anyyet, reventsp, phpp));
+}
+
+static struct audio_client_ops oss_ops = {
+	"sound,dsp",
+	oss_dev_init,
+	oss_dev_fini,
+	oss_open,
+	oss_close,
+	oss_read,
+	oss_write,
+	oss_ioctl,
+	oss_chpoll,
+	NULL,		/* mmap */
+	oss_input,
+	oss_output,
+	NULL,		/* notify */
+	NULL,		/* drain */
+};
+
+static struct audio_client_ops ossmix_ops = {
+	"sound,mixer",
+	NULL,
+	NULL,
+	ossmix_open,
+	ossmix_close,
+	ossmix_read,
+	ossmix_write,
+	ossmix_ioctl,
+	NULL,	/* chpoll */
+	NULL,   /* mmap */
+	NULL,	/* input */
+	NULL,   /* output */
+	oss_notify,
+	NULL,	/* drain */
+};
+
+/* nearly the same as ossxmix; different minor name helps devfsadm */
+static struct audio_client_ops sndstat_ops = {
+	"sound,sndstat",
+	NULL,	/* dev_init */
+	NULL,	/* dev_fini */
+	ossmix_open,
+	ossmix_close,
+	ossmix_read,
+	ossmix_write,
+	ossmix_ioctl,
+	NULL,	/* chpoll */
+	NULL,	/* mmap */
+	NULL,	/* input */
+	NULL,	/* output */
+	NULL,	/* notify */
+	NULL,	/* drain */
+};
+
+void
+auimpl_oss_init(void)
+{
+	auclnt_register_ops(AUDIO_MINOR_DSP, &oss_ops);
+	auclnt_register_ops(AUDIO_MINOR_MIXER, &ossmix_ops);
+	auclnt_register_ops(AUDIO_MINOR_SNDSTAT, &sndstat_ops);
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_output.c b/usr/src/uts/common/io/audio/impl/audio_output.c
new file mode 100644
index 0000000000..be3c66412a
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_output.c
@@ -0,0 +1,463 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Purpose: Virtual mixing audio output routines
+ *
+ * This file contains the actual mixing and resampling engine for output.
+ */
+
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/sysmacros.h>
+#include "audio_impl.h"
+
+#ifdef	DEBUG
+int audio_underruns = 0;
+#endif
+
+#define	DECL_AUDIO_EXPORT(NAME, TYPE, SAMPLE)				\
+void									\
+auimpl_export_##NAME(audio_engine_t *eng)				\
+{									\
+	int		fragfr = eng->e_fragfr;				\
+	int		nch = eng->e_nchan;				\
+	unsigned	hidx = eng->e_hidx;				\
+	TYPE		*out = (void *)eng->e_data;			\
+	int		ch = 0;						\
+									\
+	do {	/* for each channel */					\
+		int32_t *ip;						\
+		TYPE	*op;						\
+		int	i;						\
+		int	incr = eng->e_chincr[ch];			\
+									\
+		/* get value and adjust next channel offset */		\
+		op = out + eng->e_choffs[ch] + (hidx * incr);		\
+		ip = eng->e_chbufs[ch];					\
+									\
+		i = fragfr;						\
+									\
+		do {	/* for each frame */				\
+			int32_t sample = *ip;				\
+									\
+			*op = SAMPLE;					\
+			op += incr;					\
+			ip++;						\
+									\
+		} while (--i);						\
+									\
+		ch++;							\
+	} while (ch < nch);						\
+}
+
+DECL_AUDIO_EXPORT(16ne, int16_t, sample >> 8)
+DECL_AUDIO_EXPORT(16oe, int16_t, ddi_swap16(sample >> 8))
+DECL_AUDIO_EXPORT(32ne, int32_t, sample << 8)
+DECL_AUDIO_EXPORT(32oe, int32_t, ddi_swap32(sample << 8))
+DECL_AUDIO_EXPORT(24ne, int32_t, sample)
+DECL_AUDIO_EXPORT(24oe, int32_t, ddi_swap32(sample))
+
+/*
+ * Simple limiter to prevent overflows when using fixed point computations
+ */
+static void
+auimpl_output_limiter(audio_engine_t *eng)
+{
+	int k, t;
+	unsigned int q, amp, amp2;
+	int nchan = eng->e_nchan;
+	int fragfr = eng->e_fragfr;
+	int32_t **chbufs = eng->e_chbufs;
+	unsigned int statevar = eng->e_limiter_state;
+
+	for (t = 0; t < fragfr; t++) {
+
+		amp = (unsigned)ABS(chbufs[0][t]);
+
+		for (k = 1; k < nchan; k++)	{
+			amp2 = (unsigned)ABS(chbufs[k][t]);
+			if (amp2 > amp)
+				amp = amp2;
+		}
+
+		amp >>= 8;
+		q = 0x10000;
+
+		if (amp > 0x7FFF)
+			q = 0x7FFF0000 / amp;
+
+		if (statevar > q) {
+			statevar = q;
+		} else {
+			q = statevar;
+
+			/*
+			 * Simplier (linear) tracking algo
+			 * (gives less distortion, but more pumping)
+			 */
+			statevar += 2;
+			if (statevar > 0x10000)
+				statevar = 0x10000;
+
+			/*
+			 * Classic tracking algo
+			 * gives more distortion with no-lookahead
+			 * statevar=0x10000-((0x10000-statevar)*0xFFF4>>16);
+			 */
+		}
+
+		for (k = 0; k < nchan; k++) {
+			int32_t in = chbufs[k][t];
+			int32_t out = 0;
+			unsigned int p;
+
+			if (in >= 0) {
+				p = in;
+				p = ((p & 0xFFFF) * (q >> 4) >> 12) +
+				    (p >> 16) * q;
+				out = p;
+			} else {
+				p = -in;
+				p = ((p & 0xFFFF) * (q >> 4) >> 12) +
+				    (p >> 16) * q;
+				out = -p;
+			}
+			/* safety code */
+			/*
+			 * if output after limiter is clamped, then it
+			 * can be dropped
+			 */
+			if (out > 0x7FFFFF)
+				out = 0x7FFFFF;
+			else if (out < -0x7FFFFF)
+				out = -0x7FFFFF;
+
+			chbufs[k][t] = out;
+		}
+	}
+
+	eng->e_limiter_state = statevar;
+}
+
+/*
+ * Output mixing function.  Assumption: all work is done in 24-bit native PCM.
+ */
+static void
+auimpl_output_mix(audio_stream_t *sp, int offset, int nfr)
+{
+	audio_engine_t *eng = sp->s_engine;
+	const int32_t *src;
+	int choffs;
+	int nch;
+	int vol;
+
+	/*
+	 * Initial setup.
+	 */
+
+	src = sp->s_cnv_ptr;
+	choffs = sp->s_choffs;
+	nch = sp->s_cnv_dst_nchan;
+	vol = sp->s_gain_eff;
+
+	/*
+	 * Do the mixing.  We de-interleave the source stream at the
+	 * same time.
+	 */
+	for (int ch = 0; ch < nch; ch++) {
+		int32_t *op;
+		const int32_t *ip;
+
+
+		ip = src + ch;
+		op = eng->e_chbufs[ch + choffs];
+		op += offset;
+
+		for (int i = nfr; i; i--) {
+
+			int64_t	samp;
+
+			samp = *ip;
+			samp *= vol;
+			samp /= AUDIO_VOL_SCALE;
+
+			ip += nch;
+			*op += (int32_t)samp;
+			op++;
+		}
+	}
+
+	sp->s_cnv_cnt -= nfr;
+	sp->s_cnv_ptr += (nch * nfr);
+}
+
+/*
+ * Consume a fragment's worth of data.  This is called when the data in
+ * the conversion buffer is exhausted, and we need to refill it from the
+ * source buffer.  We always consume data from the client in quantities of
+ * a fragment at a time (assuming that a fragment is available.)
+ */
+static void
+auimpl_consume_fragment(audio_stream_t *sp)
+{
+	unsigned	count;
+	unsigned	avail;
+	unsigned	nframes;
+	unsigned	fragfr;
+	unsigned	framesz;
+	caddr_t		cnvbuf;
+
+	sp->s_cnv_src = sp->s_cnv_buf0;
+	sp->s_cnv_dst = sp->s_cnv_buf1;
+
+	fragfr = sp->s_fragfr;
+	nframes = sp->s_nframes;
+	framesz = sp->s_framesz;
+
+	ASSERT(sp->s_head >= sp->s_tail);
+
+	avail = sp->s_head - sp->s_tail;
+	cnvbuf = sp->s_cnv_src;
+
+	count = min(avail, fragfr);
+
+	/*
+	 * Copy data.  We deal properly with wraps.  Done as a
+	 * do...while to minimize the number of tests.
+	 */
+	do {
+		unsigned n;
+		unsigned nbytes;
+
+		n = min(nframes - sp->s_tidx, count);
+		nbytes = framesz * n;
+		bcopy(sp->s_data + (sp->s_tidx * framesz), cnvbuf, nbytes);
+		cnvbuf += nbytes;
+		count -= n;
+		sp->s_samples += n;
+		sp->s_tail += n;
+		sp->s_tidx += n;
+		if (sp->s_tidx >= nframes)
+			sp->s_tidx -= nframes;
+	} while (count);
+
+	/* Note: data conversion is optional! */
+	count = min(avail, fragfr);
+	if (sp->s_converter != NULL) {
+		sp->s_cnv_cnt = sp->s_converter(sp, count);
+	} else {
+		sp->s_cnv_cnt = count;
+	}
+}
+
+static void
+auimpl_output_callback_impl(audio_engine_t *eng)
+{
+	int		fragfr = eng->e_fragfr;
+
+	/* clear any preexisting mix results */
+	for (int i = 0; i < eng->e_nchan; i++)
+		bzero(eng->e_chbufs[i], AUDIO_CHBUFS * sizeof (int32_t));
+
+	for (audio_stream_t *sp = list_head(&eng->e_streams);
+	    sp != NULL;
+	    sp = list_next(&eng->e_streams, sp)) {
+
+		int		need;
+		int		avail;
+		int		used;
+		int		offset;
+		boolean_t	underrun = B_FALSE;
+		boolean_t	drained = B_FALSE;
+		audio_client_t	*c = sp->s_client;
+
+		/*
+		 * We need/want a full fragment.  If the client has
+		 * less than that available, it will cause a client
+		 * underrun in auimpl_consume_fragment, but in such a
+		 * case we should get silence bytes.  Assignments done
+		 * ahead of the lock to minimize lock contention.
+		 */
+		need = fragfr;
+		offset = 0;
+
+		mutex_enter(&sp->s_lock);
+		/* skip over streams not running or paused */
+		if ((!sp->s_running) || (sp->s_paused) || eng->e_suspended) {
+			mutex_exit(&sp->s_lock);
+			continue;
+		}
+
+		do {
+			/* make sure we have data to chew on */
+			if ((avail = sp->s_cnv_cnt) == 0) {
+				auimpl_consume_fragment(sp);
+				sp->s_cnv_ptr = sp->s_cnv_src;
+				avail = sp->s_cnv_cnt;
+			}
+
+			/*
+			 * We might have got more data than we need
+			 * right now.  (E.g. 8kHz expanding to 48kHz.)
+			 * Take only what we need.
+			 */
+			used = min(avail, need);
+
+			/*
+			 * Mix the results, as much data as we can use
+			 * this round.
+			 */
+			auimpl_output_mix(sp, offset, used);
+
+			/*
+			 * Save the offset for the next round, so we don't
+			 * remix into the same location.
+			 */
+			offset += used;
+
+			/*
+			 * Okay, we mixed some data, but it might not
+			 * have been all we need.  This can happen
+			 * either because we just mixed up some
+			 * partial/residual data, or because the
+			 * client has a fragment size which expands to
+			 * less than a full fragment for us. (Such as
+			 * a client wanting to operate at a higher
+			 * data rate than the engine.)
+			 */
+			need -= used;
+
+		} while (need && avail);
+
+		if (avail == 0) {
+			/* underrun or end of data */
+			if (sp->s_draining) {
+				if (sp->s_drain_idx == 0) {
+					sp->s_drain_idx = eng->e_head;
+				}
+				if (eng->e_tail >= sp->s_drain_idx) {
+					sp->s_drain_idx = 0;
+					sp->s_draining = B_FALSE;
+					/*
+					 * After draining, stop the
+					 * stream cleanly.  This
+					 * prevents underrun errors.
+					 *
+					 * (Stream will auto-start if
+					 * client submits more data to
+					 * it.)
+					 *
+					 * AC3: When an AC3 stream
+					 * drains we should probably
+					 * stop the actual hardware
+					 * engine.
+					 */
+					ASSERT(mutex_owned(&eng->e_lock));
+					sp->s_running = B_FALSE;
+					drained = B_TRUE;
+				}
+			} else {
+#ifdef	DEBUG
+				audio_underruns++;
+#endif
+				underrun = B_TRUE;
+				sp->s_errors += need;
+			}
+		}
+
+		/* wake threads waiting for stream (blocking writes, etc.) */
+		cv_broadcast(&sp->s_cv);
+
+		mutex_exit(&sp->s_lock);
+
+
+		/*
+		 * Asynchronously notify clients.  We do as much as
+		 * possible of this outside of the lock, it avoids
+		 * s_lock and c_lock contention and eliminates any
+		 * chance of deadlock.
+		 */
+
+		mutex_enter(&c->c_lock);
+		c->c_do_output = B_TRUE;
+
+		if (underrun)
+			c->c_do_notify = B_TRUE;
+
+		if (drained)
+			c->c_do_drain = B_TRUE;
+
+		cv_broadcast(&c->c_cv);
+		mutex_exit(&c->c_lock);
+	}
+
+	/*
+	 * Deal with 24-bit overflows (from mixing) gracefully.
+	 */
+	auimpl_output_limiter(eng);
+
+	/*
+	 * Export the data (a whole fragment) to the device.
+	 */
+	eng->e_export(eng);
+
+	/*
+	 * Update the head and offset.  The head counts without
+	 * wrapping, whereas the offset wraps.  Note that the test +
+	 * subtraction is faster for dealing with wrap than modulo.
+	 */
+	eng->e_head += fragfr;
+	eng->e_hidx += fragfr;
+	if (eng->e_hidx >= eng->e_nframes)
+		eng->e_hidx -= eng->e_nframes;
+
+	/*
+	 * Consider doing the SYNC outside of the lock.
+	 */
+	ENG_SYNC(eng, fragfr);
+}
+
+/*
+ * Outer loop attempts to keep playing until we hit maximum playahead.
+ */
+
+void
+auimpl_output_callback(audio_engine_t *eng)
+{
+	unsigned fragfr;
+	int64_t cnt;
+
+	fragfr = eng->e_fragfr;
+	cnt = eng->e_head - eng->e_tail;
+
+	/* stay a bit ahead */
+	while (cnt < (fragfr * 4)) {
+		auimpl_output_callback_impl(eng);
+		cnt = eng->e_head - eng->e_tail;
+	}
+}
diff --git a/usr/src/uts/common/io/audio/impl/audio_sun.c b/usr/src/uts/common/io/audio/impl/audio_sun.c
new file mode 100644
index 0000000000..d75366f439
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/audio_sun.c
@@ -0,0 +1,1967 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Sun audio(7I) and mixer(7I) personality.
+ *
+ * There are some "undocumented" details of how legacy Sun audio
+ * interfaces work.  The following "rules" were derived from reading the
+ * legacy Sun mixer code, and to the best of our knowledge are not
+ * documented elsewhere.
+ *
+ * - We create a "fake" audio device, which behaves like a classic
+ *   exclusive audio device, for each PID, as determined during open(2).
+ *
+ * - Different processes don't interfere with each other.  Even though
+ *   they are running concurrently, they each think they have exclusive
+ *   control over the audio device.
+ *
+ * - Read and write directions operate independent of each other.  That
+ *   is, a device open for reading won't intefere with a future open for
+ *   writing, and vice versa.  This is true even within the same process.
+ *
+ * - Because the virtualization is by PID, strange behavior may occur
+ *   if a process tries to open an audio device at the same time it
+ *   has already received a file descriptor from another process (such
+ *   through inheritence via fork()).
+ *
+ * - The "fake" audio device has no control over physical settings.
+ *   It sees only the software attenuation-based volumes for play and
+ *   record, and has no support for alternate input or output ports or
+ *   access to the monitoring features of the hardware.
+ *
+ * - Explicit notificaton signals (SIGPOLL) are only ever sent up the
+ *   audioctl node -- never up a regular audio node.  (The stream head
+ *   may still issue SIGPOLL based on readability/writability of
+ *   course.)
+ *
+ * - Corollary: processes that want asynch. notifications will open
+ *   /dev/audioctl as well as /dev/audio.
+ *
+ * - We don't support the MIXER mode at all.
+ *
+ * - By corollary, a process is only allowed to open /dev/audio once
+ *   (in each direction.)
+ *
+ * - Attempts to open /dev/audio in duplex mode (O_RDWR) fail (EBUSY)
+ *   if the device cannot support duplex operation.
+ *
+ * - Attempts to open a device with FREAD set fail if the device is not
+ *   capable of recording.  (Likewise for FWRITE and playback.)
+ *
+ * - No data transfer is permitted for audioctl nodes.  (No actual
+ *   record or play.)
+ *
+ * - Sun audio does not support any formats other than linear and
+ *   ULAW/ALAW.  I.e. it will never support AC3 or other "opaque"
+ *   streams which require special handling.
+ *
+ * - Sun audio only supports stereo or monophonic data streams.
+ */
+
+#include <sys/types.h>
+#include <sys/open.h>
+#include <sys/errno.h>
+#include <sys/audio.h>
+#include <sys/mixer.h>
+#include <sys/file.h>
+#include <sys/stropts.h>
+#include <sys/strsun.h>
+#include <sys/sysmacros.h>
+#include <sys/list.h>
+#include <sys/note.h>
+#include <sys/stat.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include "audio_client.h"
+
+typedef struct sclient sclient_t;
+typedef struct sdev sdev_t;
+typedef struct sproc sproc_t;
+typedef struct sioc sioc_t;
+
+typedef enum {
+	COPYIN,
+	COPYOUT,
+	IOCTL,
+	ACK,
+	NAK,
+	FINI
+} sioc_state_t;
+
+struct sioc {
+	sclient_t		*i_sc;
+	int			i_cmd;
+	size_t			i_size;
+	void			*i_data;
+	mblk_t			*i_bcont;
+	int			i_step;
+	uint_t			i_model;
+	sioc_state_t		i_state;
+	mblk_t			*i_mp;
+	caddr_t			i_addr;
+	int			i_error;
+};
+
+/* common structure shared between both audioctl and audio nodes */
+struct sclient {
+	sproc_t			*s_proc;
+	sdev_t			*s_sdev;
+	audio_client_t		*s_client;
+	queue_t			*s_rq;
+	queue_t			*s_wq;
+	ldi_handle_t		s_lh;
+	unsigned		s_eof;
+	list_t			s_eofcnt;
+	kmutex_t		s_lock;
+	mblk_t			*s_draining;
+};
+
+struct eofcnt {
+	list_node_t		linkage;
+	uint64_t		tail;
+};
+
+struct sdev {
+	audio_dev_t		*d_dev;
+
+	list_t			d_procs;
+	kmutex_t		d_mx;
+	kcondvar_t		d_cv;
+};
+
+struct sproc {
+	pid_t			p_id;
+	struct audio_info	p_info;
+	int			p_refcnt;
+	int			p_oflag;
+	list_node_t		p_linkage;
+	sdev_t			*p_sdev;
+	sclient_t		*p_writer;
+	sclient_t		*p_reader;
+};
+
+int sproc_hold(audio_client_t *, int);
+void sproc_release(sclient_t *);
+static void sproc_update(sproc_t *);
+
+
+static kmutex_t	sdev_lock;
+static dev_info_t *sdev_dip;
+
+/*
+ * Alloc extra room for ioctl buffer, in case none was supplied or copyin was
+ * shorter than we need for the whole struct.  On failure, returns an
+ * appropriate errno, zero on success.  Any original data is preserved.
+ */
+static int
+sioc_alloc(sioc_t *ip, size_t size)
+{
+	mblk_t			*nmp;
+
+	/* if we already have enough, just use what we've got */
+	if (ip->i_size >= size)
+		return (0);
+
+	if ((nmp = allocb(size, BPRI_MED)) == NULL) {
+		ip->i_state = NAK;
+		ip->i_error = ENOMEM;
+		return (ENOMEM);
+	}
+	bzero(nmp->b_rptr, size);
+
+	/* if there was already some data present, preserve it */
+	if (ip->i_size != 0) {
+		bcopy(ip->i_data, nmp->b_rptr, ip->i_size);
+		freemsg(ip->i_bcont);
+	}
+	ip->i_bcont = nmp;
+	ip->i_data = nmp->b_rptr;
+	ip->i_size = size;
+
+	return (0);
+}
+
+static void
+sioc_copyin(sioc_t *ip, size_t size)
+{
+	ip->i_state = COPYIN;
+	ip->i_size = size;
+	if (ip->i_bcont != NULL) {
+		freemsg(ip->i_bcont);
+		ip->i_bcont = NULL;
+	}
+
+	mcopyin(ip->i_mp, ip, size, ip->i_addr);
+}
+
+static void
+sioc_copyout(sioc_t *ip, size_t size)
+{
+	mblk_t			*bcont;
+
+	ASSERT(ip->i_size >= size);
+
+	bcont = ip->i_bcont;
+
+	ip->i_state = COPYOUT;
+	ip->i_bcont = NULL;
+
+	mcopyout(ip->i_mp, ip, size, ip->i_addr, bcont);
+}
+
+static void
+sioc_error(sioc_t *ip, int error)
+{
+	ip->i_state = NAK;
+	ip->i_error = error;
+}
+
+static void
+sioc_success(sioc_t *ip)
+{
+	ip->i_state = ACK;
+}
+
+static void
+sioc_fini(sioc_t *ip)
+{
+	if (ip->i_bcont != NULL)
+		freemsg(ip->i_bcont);
+
+	kmem_free(ip, sizeof (*ip));
+}
+
+static void
+sioc_finish(sioc_t *ip)
+{
+	mblk_t		*mp;
+	sclient_t	*sc;
+
+	sc = ip->i_sc;
+	mp = ip->i_mp;
+	ip->i_mp = NULL;
+
+	switch (ip->i_state) {
+	case ACK:
+		miocack(sc->s_wq, mp, 0, 0);
+		break;
+
+	case IOCTL:	/* caller didn't use sioc_success */
+		ip->i_error = ECANCELED;
+		miocnak(sc->s_wq, mp, 0, ip->i_error);
+		break;
+
+	case NAK:
+		miocnak(sc->s_wq, mp, 0, ip->i_error);
+		break;
+
+	case COPYOUT:
+	case COPYIN:
+		/* data copy to be done */
+		qreply(sc->s_wq, mp);
+		return;
+
+	case FINI:
+		if (mp != NULL) {
+			freemsg(mp);
+		}
+		break;
+	}
+
+	sioc_fini(ip);
+}
+
+static int
+sun_compose_format(audio_prinfo_t *prinfo)
+{
+	switch (prinfo->precision) {
+	case 8:
+		switch (prinfo->encoding) {
+		case AUDIO_ENCODING_ULAW:
+			return (AUDIO_FORMAT_ULAW);
+		case AUDIO_ENCODING_ALAW:
+			return (AUDIO_FORMAT_ALAW);
+		case AUDIO_ENCODING_LINEAR8:
+			return (AUDIO_FORMAT_U8);
+		case AUDIO_ENCODING_LINEAR:
+			return (AUDIO_FORMAT_S8);
+		}
+		break;
+	case 16:
+		if (prinfo->encoding == AUDIO_ENCODING_LINEAR)
+			return (AUDIO_FORMAT_S16_NE);
+		break;
+	case 32:
+		if (prinfo->encoding == AUDIO_ENCODING_LINEAR)
+			return (AUDIO_FORMAT_S32_NE);
+		break;
+	}
+	return (AUDIO_FORMAT_NONE);
+
+}
+
+static void
+sun_decompose_format(audio_prinfo_t *prinfo, int afmt)
+{
+	int	e, p;
+
+	/*
+	 * N.B.: Even though some of the formats below can't be set by
+	 * this personality, reporting them (using the closest match)
+	 * allows this personality to roughly approximate settings for
+	 * other streams.  It would be incredibly poor form for any
+	 * personality to modify the format settings for a different
+	 * personality, so we don't worry about that case.
+	 */
+
+	switch (afmt) {
+	case AUDIO_FORMAT_ULAW:
+		e = AUDIO_ENCODING_ULAW;
+		p = 8;
+		break;
+
+	case AUDIO_FORMAT_ALAW:
+		e = AUDIO_ENCODING_ALAW;
+		p = 8;
+		break;
+
+	case AUDIO_FORMAT_U8:
+		e = AUDIO_ENCODING_LINEAR8;
+		p = 8;
+		break;
+
+	case AUDIO_FORMAT_S8:
+		e = AUDIO_ENCODING_LINEAR;
+		p = 8;
+		break;
+
+	case AUDIO_FORMAT_S16_NE:
+	case AUDIO_FORMAT_S16_OE:
+	case AUDIO_FORMAT_U16_NE:
+	case AUDIO_FORMAT_U16_OE:
+		e = AUDIO_ENCODING_LINEAR;
+		p = 16;
+		break;
+
+	case AUDIO_FORMAT_S24_NE:
+	case AUDIO_FORMAT_S24_OE:
+	case AUDIO_FORMAT_S24_PACKED:
+		e = AUDIO_ENCODING_LINEAR;
+		p = 24;
+		break;
+
+	case AUDIO_FORMAT_S32_NE:
+	case AUDIO_FORMAT_S32_OE:
+		e = AUDIO_ENCODING_LINEAR;
+		p = 32;
+		break;
+
+	default:
+		/* all other formats (e.g. AC3) are uninterpreted */
+		e = AUDIO_ENCODING_NONE;
+		p = 32;
+		break;
+	}
+
+	prinfo->encoding = e;
+	prinfo->precision = p;
+}
+
+static sproc_t *
+sproc_alloc(sclient_t *sc)
+{
+	audio_client_t	*c;
+	audio_info_t	*info;
+	audio_prinfo_t	*prinfo;
+	uint32_t	caps;
+	sproc_t		*proc;
+
+	c = sc->s_client;
+	if ((proc = kmem_zalloc(sizeof (*proc), KM_NOSLEEP)) == NULL) {
+		return (NULL);
+	}
+	info = &proc->p_info;
+
+	/*
+	 * audio(7I) says: Upon the initial open() of the audio
+	 * device, the driver resets the data format of the device to
+	 * the default state of 8-bit, 8Khz, mono u-Law data.
+	 */
+	prinfo = &info->play;
+	prinfo->channels =	1;
+	prinfo->sample_rate =	8000;
+	prinfo->encoding =	AUDIO_ENCODING_ULAW;
+	prinfo->precision =	8;
+	prinfo->gain =		AUDIO_MAX_GAIN;
+	prinfo->balance =	AUDIO_MID_BALANCE;
+	prinfo->buffer_size =	8192;
+	prinfo->pause =		B_FALSE;
+	prinfo->waiting =	B_FALSE;
+	prinfo->open =		B_FALSE;
+	prinfo->active =	B_FALSE;
+	prinfo->samples =	0;
+	prinfo->eof =		0;
+	prinfo->error =		0;
+	prinfo->minordev =	0;
+	prinfo->port =		AUDIO_SPEAKER;
+	prinfo->avail_ports =	AUDIO_SPEAKER;
+	prinfo->mod_ports =	AUDIO_NONE;
+	prinfo->_xxx =		0;
+
+	prinfo = &info->record;
+	prinfo->channels =	1;
+	prinfo->sample_rate =	8000;
+	prinfo->encoding =	AUDIO_ENCODING_ULAW;
+	prinfo->precision =	8;
+	prinfo->gain =		AUDIO_MAX_GAIN;
+	prinfo->balance =	AUDIO_MID_BALANCE;
+	prinfo->buffer_size =	8192;
+	prinfo->waiting =	B_FALSE;
+	prinfo->open =  	B_FALSE;
+	prinfo->active =	B_FALSE;
+	prinfo->samples =	0;
+	prinfo->eof =		0;
+	prinfo->error =		0;
+	prinfo->minordev =	0;
+	prinfo->port =		AUDIO_MICROPHONE;
+	prinfo->avail_ports =	AUDIO_MICROPHONE;
+	prinfo->mod_ports =	AUDIO_MICROPHONE;
+
+	info->output_muted =	B_FALSE;
+	/* pretend we don't have a software mixer - we don't support the API */
+	info->hw_features =	0;
+	info->sw_features =	0;
+	info->sw_features_enabled = 0;
+
+	caps = auclnt_get_dev_capab(auclnt_get_dev(c));
+	if (caps & AUDIO_CLIENT_CAP_PLAY)
+		info->hw_features |= AUDIO_HWFEATURE_PLAY;
+	if (caps & AUDIO_CLIENT_CAP_RECORD)
+		info->hw_features |= AUDIO_HWFEATURE_RECORD;
+	if (caps & AUDIO_CLIENT_CAP_DUPLEX)
+		info->hw_features |= AUDIO_HWFEATURE_DUPLEX;
+
+	return (proc);
+}
+
+static void
+sproc_free(sproc_t *proc)
+{
+	kmem_free(proc, sizeof (*proc));
+}
+
+int
+sproc_hold(audio_client_t *c, int oflag)
+{
+	pid_t		pid;
+	sproc_t		*proc;
+	sdev_t		*sdev;
+	sclient_t	*sc;
+	list_t		*l;
+	audio_dev_t	*adev;
+	int		rv;
+
+	adev = auclnt_get_dev(c);
+
+	/* first allocate and initialize the sclient private data */
+	if ((sc = kmem_zalloc(sizeof (*sc), KM_NOSLEEP)) == NULL) {
+		return (ENOMEM);
+	}
+
+	mutex_init(&sc->s_lock, NULL, MUTEX_DRIVER, NULL);
+	list_create(&sc->s_eofcnt, sizeof (struct eofcnt),
+	    offsetof(struct eofcnt, linkage));
+	auclnt_set_private(c, sc);
+
+	sdev = auclnt_get_dev_minor_data(adev, AUDIO_MINOR_DEVAUDIO);
+	l = &sdev->d_procs;
+	pid = auclnt_get_pid(c);
+
+	/* set a couple of common fields */
+	sc->s_client = c;
+	sc->s_sdev = sdev;
+
+	mutex_enter(&sdev->d_mx);
+	for (proc = list_head(l); proc != NULL; proc = list_next(l, proc)) {
+		if (proc->p_id == pid) {
+			proc->p_refcnt++;
+			break;
+		}
+	}
+	if (proc == NULL) {
+		if ((proc = sproc_alloc(sc)) == NULL) {
+			rv = ENOMEM;
+			goto failed;
+		}
+		proc->p_refcnt = 1;
+		proc->p_id = pid;
+		proc->p_sdev = sdev;
+		list_insert_tail(l, proc);
+	}
+
+	sc->s_proc = proc;
+
+	while (proc->p_oflag & oflag) {
+
+		if (oflag & (FNDELAY|FNONBLOCK)) {
+			rv = EBUSY;
+			goto failed;
+		}
+		if (oflag & FWRITE)
+			proc->p_info.play.waiting++;
+		if (oflag & FREAD)
+			proc->p_info.record.waiting++;
+		if (cv_wait_sig(&sdev->d_cv, &sdev->d_mx) == 0) {
+			/* interrupted! */
+			if (oflag & FWRITE)
+				proc->p_info.play.waiting--;
+			if (oflag & FREAD)
+				proc->p_info.record.waiting--;
+			rv = EINTR;
+			goto failed;
+		}
+		if (oflag & FWRITE)
+			proc->p_info.play.waiting--;
+		if (oflag & FREAD)
+			proc->p_info.record.waiting--;
+	}
+
+	if (oflag & FWRITE) {
+		audio_prinfo_t	*play = &proc->p_info.play;
+		audio_stream_t	*sp = auclnt_output_stream(c);
+
+		if (((rv = auclnt_set_rate(sp, 8000)) != 0) ||
+		    ((rv = auclnt_set_format(sp, AUDIO_FORMAT_ULAW)) != 0) ||
+		    ((rv = auclnt_set_channels(sp, 1)) != 0)) {
+			goto failed;
+		}
+
+		auclnt_set_samples(sp, 0);
+		auclnt_set_errors(sp, 0);
+		play->eof = 0;
+		play->buffer_size = 8192;
+
+		auclnt_set_gain(sp, ((play->gain * 100) / AUDIO_MAX_GAIN));
+		auclnt_set_muted(sp, proc->p_info.output_muted);
+		play->open = B_TRUE;
+		proc->p_writer = sc;
+		proc->p_oflag |= FWRITE;
+	}
+
+	if (oflag & FREAD) {
+		audio_prinfo_t	*rec = &proc->p_info.record;
+		audio_stream_t	*sp = auclnt_input_stream(c);
+
+		if (((rv = auclnt_set_rate(sp, 8000)) != 0) ||
+		    ((rv = auclnt_set_format(sp, AUDIO_FORMAT_ULAW)) != 0) ||
+		    ((rv = auclnt_set_channels(sp, 1)) != 0)) {
+			goto failed;
+		}
+
+		auclnt_set_samples(sp, 0);
+		auclnt_set_errors(sp, 0);
+		rec->eof = 0;
+		rec->buffer_size = 8192;
+
+		auclnt_set_gain(sp, ((rec->gain * 100) / AUDIO_MAX_GAIN));
+		rec->open = B_TRUE;
+		proc->p_reader = sc;
+		proc->p_oflag |= FREAD;
+	}
+
+	sproc_update(proc);
+
+	mutex_exit(&sdev->d_mx);
+
+	return (0);
+
+failed:
+	mutex_exit(&sdev->d_mx);
+	sproc_release(sc);
+	return (rv);
+
+}
+
+static void
+sun_clear_eof(sclient_t *sc)
+{
+	struct eofcnt *eof;
+	mutex_enter(&sc->s_lock);
+	while ((eof = list_remove_head(&sc->s_eofcnt)) != NULL) {
+		kmem_free(eof, sizeof (*eof));
+	}
+	mutex_exit(&sc->s_lock);
+}
+
+void
+sproc_release(sclient_t *sc)
+{
+	sproc_t		*proc;
+	sdev_t		*sdev;
+	mblk_t		*mp;
+
+	proc = sc->s_proc;
+	sdev = sc->s_sdev;
+	sc->s_proc = NULL;
+
+	mutex_enter(&sdev->d_mx);
+
+	if (proc != NULL) {
+		proc->p_refcnt--;
+		ASSERT(proc->p_refcnt >= 0);
+
+		if (sc == proc->p_writer) {
+			proc->p_oflag &= ~FWRITE;
+			proc->p_writer = NULL;
+		}
+		if (sc == proc->p_reader) {
+			proc->p_oflag &= ~FREAD;
+			proc->p_reader = NULL;
+		}
+		cv_broadcast(&sdev->d_cv);
+
+		if (proc->p_refcnt == 0) {
+			list_remove(&sdev->d_procs, proc);
+			sproc_free(proc);
+		}
+		sc->s_proc = NULL;
+	}
+
+	mutex_exit(&sdev->d_mx);
+
+	sun_clear_eof(sc);
+
+	while ((mp = sc->s_draining) != NULL) {
+		sc->s_draining = mp->b_next;
+		mp->b_next = NULL;
+		freemsg(mp);
+	}
+
+	mutex_destroy(&sc->s_lock);
+	list_destroy(&sc->s_eofcnt);
+	kmem_free(sc, sizeof (*sc));
+}
+
+static void
+sun_sendup(audio_client_t *c)
+{
+	audio_stream_t	*sp = auclnt_input_stream(c);
+	sclient_t	*sc = auclnt_get_private(c);
+	unsigned	framesz = auclnt_get_framesz(sp);
+	queue_t		*rq = sc->s_rq;
+	mblk_t		*mp;
+	unsigned	nbytes = sc->s_proc->p_info.record.buffer_size;
+	unsigned	count = nbytes / framesz;
+
+	/*
+	 * Potentially send a message upstream with the record data.
+	 * We collect this up in chunks of the buffer size requested
+	 * by the client.
+	 */
+
+	while (auclnt_get_count(sp) >= count) {
+
+		if ((!canputnext(rq)) ||
+		    ((mp = allocb(nbytes, BPRI_MED)) == NULL)) {
+			/*
+			 * This will apply back pressure to the
+			 * buffer.  We haven't yet lost any data, we
+			 * just can't send it up.  The point at which
+			 * we have an unrecoverable overrun is in the
+			 * buffer, not in the streams queue.  So, no
+			 * need to do anything right now.
+			 *
+			 * Note that since recording is enabled, we
+			 * expect that the callback routine will be
+			 * called repeatedly & regularly, so we don't
+			 * have to worry about leaving data orphaned
+			 * in the queue.
+			 */
+			break;
+		}
+
+		(void) auclnt_consume_data(sp, (caddr_t)mp->b_wptr, count);
+		mp->b_wptr += nbytes;
+		putnext(rq, mp);
+	}
+}
+
+static int
+sun_open(audio_client_t *c, int oflag)
+{
+	_NOTE(ARGUNUSED(c));
+	_NOTE(ARGUNUSED(oflag));
+	return (0);
+}
+
+static void
+sun_close(audio_client_t *c)
+{
+	_NOTE(ARGUNUSED(c));
+}
+
+static void
+sproc_update(sproc_t *proc)
+{
+	audio_info_t	*info;
+	audio_stream_t	*sp;
+	sclient_t	*sc;
+
+	info = &proc->p_info;
+
+	ASSERT(mutex_owned(&proc->p_sdev->d_mx));
+
+	if ((sc = proc->p_writer) != NULL) {
+		sp = auclnt_output_stream(sc->s_client);
+
+		info->play.sample_rate = auclnt_get_rate(sp);
+		info->play.channels = auclnt_get_channels(sp);
+		sun_decompose_format(&info->play, auclnt_get_format(sp));
+
+		info->play.gain =
+		    (auclnt_get_gain(sp) * AUDIO_MAX_GAIN) / 100;
+		info->play.pause = auclnt_is_paused(sp);
+		info->play.active = !info->play.pause;
+		info->play.samples = auclnt_get_samples(sp);
+		info->play.error = auclnt_get_errors(sp) ? B_TRUE : B_FALSE;
+		info->output_muted = auclnt_get_muted(sp);
+	}
+
+	if ((sc = proc->p_reader) != NULL) {
+		sp = auclnt_input_stream(sc->s_client);
+
+		info->record.sample_rate = auclnt_get_rate(sp);
+		info->record.channels = auclnt_get_channels(sp);
+		sun_decompose_format(&info->record, auclnt_get_format(sp));
+
+		info->record.gain =
+		    (auclnt_get_gain(sp) * AUDIO_MAX_GAIN) / 100;
+		info->record.pause = auclnt_is_paused(sp);
+		info->record.active = !info->record.pause;
+		info->record.samples = auclnt_get_samples(sp);
+		info->record.error = auclnt_get_errors(sp) ? B_TRUE : B_FALSE;
+	}
+}
+
+static void
+sioc_getinfo(sioc_t *ip)
+{
+	sclient_t	*sc = ip->i_sc;
+	sproc_t		*proc = sc->s_proc;
+	int		rv;
+
+	switch (ip->i_step) {
+	case 0:
+		if ((rv = sioc_alloc(ip, sizeof (audio_info_t))) != 0) {
+			sioc_error(ip, rv);
+			break;
+		}
+
+		mutex_enter(&sc->s_sdev->d_mx);
+		sproc_update(proc);
+		mutex_exit(&sc->s_sdev->d_mx);
+
+		bcopy(&proc->p_info, ip->i_data, sizeof (audio_info_t));
+		sioc_copyout(ip, sizeof (audio_info_t));
+		break;
+	case 1:
+		sioc_success(ip);
+		break;
+	}
+
+	ip->i_step++;
+	sioc_finish(ip);
+}
+
+#define	CHANGED(new, old, field)			\
+	((new->field != ((uint32_t)~0)) && (new->field != old->field))
+#define	CHANGED8(new, old, field)			\
+	((new->field != ((uint8_t)~0)) && (new->field != old->field))
+
+static int
+sun_setinfo(sclient_t *sc, audio_info_t *ninfo)
+{
+	sproc_t		*proc = sc->s_proc;
+	audio_info_t	*oinfo = &proc->p_info;
+	audio_prinfo_t	*npr;
+	audio_prinfo_t	*opr;
+
+	int		pfmt = AUDIO_FORMAT_NONE;
+	int		rfmt = AUDIO_FORMAT_NONE;
+
+	boolean_t	reader;
+	boolean_t	writer;
+	boolean_t	isctl;
+	audio_stream_t	*sp;
+	int		rv;
+
+	if (auclnt_get_minor_type(sc->s_client) == AUDIO_MINOR_DEVAUDIOCTL) {
+		/* control node can do both read and write fields */
+		isctl = B_TRUE;
+		reader = B_TRUE;
+		writer = B_TRUE;
+	} else {
+		isctl = B_FALSE;
+		writer = sc == proc->p_writer;
+		reader = sc == proc->p_reader;
+	}
+
+	/*
+	 * Start by validating settings.
+	 */
+	npr = &ninfo->play;
+	opr = &oinfo->play;
+
+	if (writer && CHANGED(npr, opr, sample_rate)) {
+		if ((isctl) ||
+		    (npr->sample_rate < 5500) || (npr->sample_rate > 48000)) {
+			return (EINVAL);
+		}
+	}
+	if (writer && CHANGED(npr, opr, channels)) {
+		if ((isctl) || (npr->channels < 1) || (npr->channels > 2)) {
+			return (EINVAL);
+		}
+	}
+	if (writer &&
+	    (CHANGED(npr, opr, encoding) || CHANGED(npr, opr, precision))) {
+		if (npr->encoding == (uint32_t)~0)
+			npr->encoding = opr->encoding;
+		if (npr->precision == (uint32_t)~0)
+			npr->precision = opr->precision;
+		pfmt = sun_compose_format(npr);
+		if ((isctl) || (pfmt == AUDIO_FORMAT_NONE)) {
+			return (EINVAL);
+		}
+	}
+
+	/* play fields that anyone can modify */
+	if (CHANGED(npr, opr, gain)) {
+		if (npr->gain > AUDIO_MAX_GAIN) {
+			return (EINVAL);
+		}
+	}
+
+
+	npr = &ninfo->record;
+	opr = &oinfo->record;
+
+	if (reader && CHANGED(npr, opr, sample_rate)) {
+		if ((isctl) ||
+		    (npr->sample_rate < 5500) ||
+		    (npr->sample_rate > 48000)) {
+			return (EINVAL);
+		}
+	}
+	if (reader && CHANGED(npr, opr, channels)) {
+		if ((isctl) || (npr->channels < 1) || (npr->channels > 2)) {
+			return (EINVAL);
+		}
+	}
+	if (reader &&
+	    (CHANGED(npr, opr, encoding) || CHANGED(npr, opr, precision))) {
+		if (npr->encoding == (uint32_t)~0)
+			npr->encoding = opr->encoding;
+		if (npr->precision == (uint32_t)~0)
+			npr->precision = opr->precision;
+		rfmt = sun_compose_format(npr);
+		if ((isctl) || (rfmt == AUDIO_FORMAT_NONE)) {
+			return (EINVAL);
+		}
+	}
+	if (reader && CHANGED(npr, opr, buffer_size)) {
+		if (isctl) {
+			return (EINVAL);
+		}
+		/* make sure we can support 16-bit stereo samples */
+		if ((npr->buffer_size % 4) != 0) {
+			npr->buffer_size = (npr->buffer_size + 3) & ~3;
+		}
+		/* limit the maximum buffer size somewhat */
+		if (npr->buffer_size > 16384) {
+			npr->buffer_size = 16384;
+		}
+	}
+
+	/* record fields that anyone can modify */
+	if (CHANGED(npr, opr, gain)) {
+		if (npr->gain > AUDIO_MAX_GAIN) {
+			return (EINVAL);
+		}
+	}
+
+	/*
+	 * Now apply the changes.
+	 */
+	if (proc->p_writer != NULL) {
+		sp = auclnt_output_stream(proc->p_writer->s_client);
+		npr = &ninfo->play;
+		opr = &oinfo->play;
+
+		if (CHANGED(npr, opr, sample_rate)) {
+			rv = auclnt_set_rate(sp, npr->sample_rate);
+			if (rv != 0)
+				return (rv);
+		}
+		if (CHANGED(npr, opr, channels)) {
+			rv = auclnt_set_channels(sp, npr->channels);
+			if (rv != 0)
+				return (rv);
+		}
+		if (pfmt != AUDIO_FORMAT_NONE) {
+			rv = auclnt_set_format(sp, pfmt);
+			if (rv != 0)
+				return (rv);
+		}
+		if (CHANGED(npr, opr, samples)) {
+			auclnt_set_samples(sp, npr->samples);
+		}
+		if (CHANGED(npr, opr, eof)) {
+			/*
+			 * This ugly special case code is required to
+			 * prevent problems with realaudio.
+			 */
+			if (npr->eof == 0) {
+				sun_clear_eof(proc->p_writer);
+			}
+			opr->eof = npr->eof;
+		}
+		if (CHANGED8(npr, opr, pause)) {
+			if (npr->pause) {
+				auclnt_set_paused(sp);
+			} else {
+				auclnt_clear_paused(sp);
+				/* qenable to start up the playback */
+				qenable(proc->p_writer->s_wq);
+			}
+		}
+		if (CHANGED8(npr, opr, waiting) && (npr->waiting)) {
+			opr->waiting = npr->waiting;
+		}
+		if (CHANGED8(npr, opr, error)) {
+			auclnt_set_errors(sp, npr->error);
+		}
+		if (CHANGED(npr, opr, gain)) {
+			auclnt_set_gain(sp, (npr->gain * 100) / AUDIO_MAX_GAIN);
+		}
+		if (CHANGED8(ninfo, oinfo, output_muted)) {
+			auclnt_set_muted(sp, ninfo->output_muted);
+		}
+		if (CHANGED(npr, opr, buffer_size)) {
+			/*
+			 * No checks on the buffer size are performed
+			 * for play side.  The value of the buffer size
+			 * is meaningless for play side anyway.
+			 */
+			opr->buffer_size = npr->buffer_size;
+		}
+	} else {
+		/* these vaalues are preserved even if /dev/audio not open */
+		if (CHANGED(npr, opr, gain)) {
+			opr->gain = npr->gain;
+		}
+		if (CHANGED8(ninfo, oinfo, output_muted)) {
+			oinfo->output_muted = ninfo->output_muted;
+		}
+	}
+
+	if (proc->p_reader != NULL) {
+		sp = auclnt_input_stream(proc->p_reader->s_client);
+		npr = &ninfo->record;
+		opr = &oinfo->record;
+
+		if (CHANGED(npr, opr, sample_rate)) {
+			rv = auclnt_set_rate(sp, npr->sample_rate);
+			if (rv != 0)
+				return (rv);
+		}
+		if (CHANGED(npr, opr, channels)) {
+			rv = auclnt_set_channels(sp, npr->channels);
+			if (rv != 0)
+				return (rv);
+		}
+		if (rfmt != AUDIO_FORMAT_NONE) {
+			rv = auclnt_set_format(sp, rfmt);
+			if (rv != 0)
+				return (rv);
+		}
+		if (CHANGED(npr, opr, samples)) {
+			auclnt_set_samples(sp, npr->samples);
+		}
+		if (CHANGED(npr, opr, eof)) {
+			opr->eof = npr->eof;
+		}
+		if (CHANGED8(npr, opr, pause)) {
+			if (npr->pause) {
+				auclnt_set_paused(sp);
+			} else {
+				auclnt_clear_paused(sp);
+				auclnt_start(sp);
+			}
+		}
+		if (CHANGED8(npr, opr, waiting) && (npr->waiting)) {
+			opr->waiting = npr->waiting;
+		}
+		if (CHANGED8(npr, opr, error)) {
+			auclnt_set_errors(sp, npr->error);
+		}
+		if (CHANGED(npr, opr, buffer_size)) {
+			opr->buffer_size = npr->buffer_size;
+		}
+		if (CHANGED(npr, opr, gain)) {
+			auclnt_set_gain(sp, (npr->gain * 100) / AUDIO_MAX_GAIN);
+		}
+	} else {
+		/* these values are preserved even if /dev/audio not open */
+		if (CHANGED(npr, opr, gain)) {
+			opr->gain = npr->gain;
+		}
+	}
+
+	return (0);
+}
+
+static void
+sioc_setinfo(sioc_t *ip)
+{
+	int		rv;
+	sclient_t	*sc = ip->i_sc;
+	audio_info_t	*ninfo;
+
+	switch (ip->i_step) {
+	case 0:
+		sioc_copyin(ip, sizeof (audio_info_t));
+		break;
+
+	case 1:
+		ninfo = (audio_info_t *)ip->i_data;
+
+		mutex_enter(&sc->s_sdev->d_mx);
+		rv = sun_setinfo(ip->i_sc, ninfo);
+
+		if (rv != 0) {
+			sioc_error(ip, rv);
+		} else {
+			sproc_update(sc->s_proc);
+
+			bcopy(&sc->s_proc->p_info, ninfo, sizeof (*ninfo));
+			sioc_copyout(ip, sizeof (audio_info_t));
+		}
+		mutex_exit(&sc->s_sdev->d_mx);
+		break;
+
+	case 2:
+		sioc_success(ip);
+		break;
+	}
+
+	ip->i_step++;
+	sioc_finish(ip);
+}
+
+static void
+sioc_getdev(sioc_t *ip)
+{
+	int		rv;
+	sclient_t	*sc = ip->i_sc;
+	audio_client_t	*c = sc->s_client;
+	audio_dev_t	*d = auclnt_get_dev(c);
+
+	switch (ip->i_step) {
+	case 0:
+		rv = sioc_alloc(ip, sizeof (audio_device_t));
+		if (rv == 0) {
+			audio_device_t *a = ip->i_data;
+
+			(void) snprintf(a->name, sizeof (a->name),
+			    "SUNW,%s", auclnt_get_dev_name(d));
+			(void) strlcpy(a->config,
+			    auclnt_get_dev_description(d), sizeof (a->config));
+			(void) strlcpy(a->version,
+			    auclnt_get_dev_version(d),  sizeof (a->version));
+			sioc_copyout(ip, sizeof (*a));
+		} else {
+			sioc_error(ip, rv);
+		}
+		break;
+
+	case 1:
+		sioc_success(ip);
+		break;
+	}
+
+	ip->i_step++;
+	sioc_finish(ip);
+}
+
+static void
+sunstr_ioctl(sioc_t *ip)
+{
+	switch (ip->i_cmd) {
+	case AUDIO_GETINFO:
+		sioc_getinfo(ip);
+		break;
+
+	case AUDIO_SETINFO:
+		sioc_setinfo(ip);
+		break;
+
+	case AUDIO_GETDEV:
+		sioc_getdev(ip);
+		break;
+
+	case AUDIO_DIAG_LOOPBACK:
+		/* we don't support this one */
+		sioc_error(ip, ENOTTY);
+		sioc_finish(ip);
+		break;
+
+	case AUDIO_MIXERCTL_GET_MODE:
+	case AUDIO_MIXERCTL_SET_MODE:
+	case AUDIO_MIXERCTL_GET_CHINFO:
+	case AUDIO_MIXERCTL_SET_CHINFO:
+	case AUDIO_MIXERCTL_GETINFO:
+	case AUDIO_MIXERCTL_SETINFO:
+	case AUDIO_GET_NUM_CHS:
+	case AUDIO_GET_CH_NUMBER:
+	case AUDIO_GET_CH_TYPE:
+	case AUDIO_MIXER_SINGLE_OPEN:
+	case AUDIO_MIXER_MULTIPLE_OPEN:
+	case AUDIO_MIXER_GET_SAMPLE_RATES:
+	default:
+		sioc_error(ip, EINVAL);
+		sioc_finish(ip);
+		break;
+	}
+}
+
+static int
+sun_sigpoll(audio_client_t *c, void *arg)
+{
+	sproc_t		*proc = arg;
+	sclient_t	*sc;
+
+	if (auclnt_get_minor_type(c) == AUDIO_MINOR_DEVAUDIOCTL) {
+		sc = auclnt_get_private(c);
+		/* we only need to notify peers in our own process */
+		if ((sc != NULL) && (sc->s_proc == proc)) {
+			(void) putnextctl1(sc->s_rq, M_PCSIG, SIGPOLL);
+		}
+	}
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static void
+sun_drain(audio_client_t *c)
+{
+	sclient_t	*sc = auclnt_get_private(c);
+	mblk_t		*mplist, *mp;
+
+	mutex_enter(&sc->s_lock);
+	mplist = sc->s_draining;
+	sc->s_draining = NULL;
+	mutex_exit(&sc->s_lock);
+
+	while ((mp = mplist) != NULL) {
+		mplist = mp->b_next;
+		mp->b_next = NULL;
+		miocack(sc->s_wq, mp, 0, 0);
+	}
+}
+
+static void
+sun_output(audio_client_t *c)
+{
+	sclient_t	*sc = auclnt_get_private(c);
+	sproc_t		*proc = sc->s_proc;
+	uint64_t	tail;
+	struct eofcnt	*eof;
+	int		eofs = 0;
+
+	tail = auclnt_get_tail(auclnt_output_stream(c));
+
+	/* get more data! (do this early) */
+	qenable(sc->s_wq);
+
+	mutex_enter(&sc->s_lock);
+	while (((eof = list_head(&sc->s_eofcnt)) != NULL) &&
+	    (eof->tail < tail)) {
+		list_remove(&sc->s_eofcnt, eof);
+		kmem_free(eof, sizeof (*eof));
+		eofs++;
+	}
+	proc->p_info.play.eof += eofs;
+	mutex_exit(&sc->s_lock);
+
+	if (eofs) {
+		auclnt_dev_walk_clients(auclnt_get_dev(c),
+		    sun_sigpoll, proc);
+	}
+}
+
+static void
+sun_input(audio_client_t *c)
+{
+	sun_sendup(c);
+}
+
+static int
+sun_create_minors(audio_dev_t *adev, void *notused)
+{
+	char		path[MAXPATHLEN];
+	minor_t		minor;
+	int		inst;
+	int		index;
+	const char	*driver;
+	unsigned	cap;
+
+	_NOTE(ARGUNUSED(notused));
+
+	ASSERT(mutex_owned(&sdev_lock));
+
+	/* don't create device nodes for sndstat device */
+	cap = auclnt_get_dev_capab(adev);
+	if ((cap & (AUDIO_CLIENT_CAP_PLAY | AUDIO_CLIENT_CAP_RECORD)) == 0) {
+		return (AUDIO_WALK_CONTINUE);
+	}
+
+	index = auclnt_get_dev_index(adev);
+	inst = auclnt_get_dev_instance(adev);
+	driver = auclnt_get_dev_driver(adev);
+
+	if (sdev_dip != NULL) {
+
+		minor = AUDIO_MKMN(index, AUDIO_MINOR_DEVAUDIO);
+		(void) snprintf(path, sizeof (path), "sound,%s,audio%d",
+		    driver, inst);
+		(void) ddi_create_minor_node(sdev_dip, path, S_IFCHR, minor,
+		    DDI_NT_AUDIO, 0);
+
+		minor = AUDIO_MKMN(index, AUDIO_MINOR_DEVAUDIOCTL);
+		(void) snprintf(path, sizeof (path), "sound,%s,audioctl%d",
+		    driver, inst);
+		(void) ddi_create_minor_node(sdev_dip, path, S_IFCHR, minor,
+		    DDI_NT_AUDIO, 0);
+	}
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static int
+sun_remove_minors(audio_dev_t *adev, void *notused)
+{
+	char		path[MAXPATHLEN];
+	int		inst;
+	const char	*driver;
+	unsigned	cap;
+
+	_NOTE(ARGUNUSED(notused));
+
+	ASSERT(mutex_owned(&sdev_lock));
+
+	cap = auclnt_get_dev_capab(adev);
+	/* if not a play or record device, don't bother creating minors */
+	if ((cap & (AUDIO_CLIENT_CAP_PLAY | AUDIO_CLIENT_CAP_RECORD)) == 0) {
+		return (AUDIO_WALK_CONTINUE);
+	}
+
+	inst = auclnt_get_dev_instance(adev);
+	driver = auclnt_get_dev_driver(adev);
+
+	if (sdev_dip != NULL) {
+
+		(void) snprintf(path, sizeof (path), "sound,%s,audio%d",
+		    driver, inst);
+		ddi_remove_minor_node(sdev_dip, path);
+
+		(void) snprintf(path, sizeof (path), "sound,%s,audioctl%d",
+		    driver, inst);
+		ddi_remove_minor_node(sdev_dip, path);
+	}
+
+	return (AUDIO_WALK_CONTINUE);
+}
+
+static void *
+sun_dev_init(audio_dev_t *adev)
+{
+	sdev_t		*sdev;
+	unsigned	cap;
+
+	cap = auclnt_get_dev_capab(adev);
+	/* if not a play or record device, don't bother initializing it */
+	if ((cap & (AUDIO_CLIENT_CAP_PLAY | AUDIO_CLIENT_CAP_RECORD)) == 0) {
+		return (NULL);
+	}
+
+	sdev = kmem_zalloc(sizeof (*sdev), KM_SLEEP);
+	sdev->d_dev = adev;
+	mutex_init(&sdev->d_mx, NULL, MUTEX_DRIVER, NULL);
+	cv_init(&sdev->d_cv, NULL, CV_DRIVER, NULL);
+	list_create(&sdev->d_procs, sizeof (struct sproc),
+	    offsetof(struct sproc, p_linkage));
+
+	mutex_enter(&sdev_lock);
+	(void) sun_create_minors(adev, NULL);
+	mutex_exit(&sdev_lock);
+
+	return (sdev);
+}
+
+static void
+sun_dev_fini(void *arg)
+{
+	sdev_t	*sdev = arg;
+
+	if (sdev != NULL) {
+
+		/* remove minor nodes */
+		mutex_enter(&sdev_lock);
+		(void) sun_remove_minors(sdev->d_dev, NULL);
+		mutex_exit(&sdev_lock);
+
+		mutex_destroy(&sdev->d_mx);
+		cv_destroy(&sdev->d_cv);
+		list_destroy(&sdev->d_procs);
+		kmem_free(sdev, sizeof (*sdev));
+	}
+}
+
+static struct audio_client_ops sun_ops = {
+	"internal,audio",
+	sun_dev_init,
+	sun_dev_fini,
+	sun_open,
+	sun_close,
+	NULL,	/* read */
+	NULL,	/* write */
+	NULL,	/* ioctl */
+	NULL,	/* chpoll */
+	NULL,	/* mmap */
+	sun_input,
+	sun_output,
+	NULL,	/* notify */
+	sun_drain,
+};
+
+static struct audio_client_ops sunctl_ops = {
+	"internal,audioctl",
+	NULL,	/* dev_init */
+	NULL,	/* dev_fini */
+	sun_open,
+	sun_close,
+	NULL,	/* read */
+	NULL,	/* write */
+	NULL,	/* ioctl */
+	NULL,	/* chpoll */
+	NULL,	/* mmap */
+	NULL,	/* output */
+	NULL,	/* input */
+	NULL,	/* notify */
+	NULL,	/* drain */
+};
+
+void
+auimpl_sun_init(void)
+{
+	mutex_init(&sdev_lock, NULL, MUTEX_DRIVER, NULL);
+	sdev_dip = NULL;
+	auclnt_register_ops(AUDIO_MINOR_DEVAUDIO, &sun_ops);
+	auclnt_register_ops(AUDIO_MINOR_DEVAUDIOCTL, &sunctl_ops);
+}
+
+/*
+ * This is the operations entry points that are streams specific...
+ * We map "instance" numbers.
+ */
+
+static int
+sunstr_open(queue_t *rq, dev_t *devp, int flag, int sflag, cred_t *cr)
+{
+	int			rv;
+	minor_t			minor;
+	minor_t			index;
+	minor_t			type;
+	dev_t			physdev;
+	ldi_ident_t		lid;
+	ldi_handle_t		lh = NULL;
+	audio_client_t		*c = NULL;
+	sclient_t		*sc = NULL;
+	audio_dev_t		*adev;
+	unsigned		fmt;
+	int			oflag;
+	boolean_t		isopen = B_FALSE;
+
+	if (sflag != 0) {
+		/* no direct clone or module opens */
+		return (EINVAL);
+	}
+
+	/*
+	 * NB: We reuse the partitioning that the core framework is
+	 * using for instance numbering.  This does mean that we are
+	 * limited to at most AUDIO_MN_INST_MASK devices, but this
+	 * number is sufficiently large (8192) that not to be a concern.
+	 */
+
+	minor = getminor(*devp);
+	index = (minor >> AUDIO_MN_INST_SHIFT) & AUDIO_MN_INST_MASK;
+	type = (minor >> AUDIO_MN_TYPE_SHIFT) & AUDIO_MN_TYPE_MASK;
+
+	/* can't directly open a cloned node! */
+	if (minor & AUDIO_MN_CLONE_MASK) {
+		return (ENXIO);
+	}
+
+	switch (type) {
+	case AUDIO_MINOR_DEVAUDIOCTL:
+		fmt = AUDIO_FORMAT_NONE;
+		oflag = flag & ~(FWRITE | FREAD);
+		break;
+	case AUDIO_MINOR_DEVAUDIO:
+		fmt = AUDIO_FORMAT_PCM;
+		oflag = flag;
+		break;
+	default:
+		/* these minor types are not legal */
+		return (ENXIO);
+	}
+
+	/* look up and hold the matching audio device */
+	adev = auclnt_hold_dev_by_index(index);
+	if (adev == NULL) {
+		return (ENXIO);
+	}
+	/* find the matching physical devt */
+	physdev = makedevice(ddi_driver_major(auclnt_get_dev_devinfo(adev)),
+	    AUDIO_MKMN(auclnt_get_dev_instance(adev), type));
+
+	if ((rv = ldi_ident_from_stream(rq, &lid)) == 0) {
+		rv = ldi_open_by_dev(&physdev, OTYP_CHR, flag, cr, &lh, lid);
+	}
+
+	/* ldi open is done, lh holds device, and we can release our hold */
+	auclnt_release_dev(adev);
+
+	if (rv != 0) {
+		goto fail;
+	}
+	/* phys layer clones a device for us */
+	ASSERT((getminor(physdev) & AUDIO_MN_CLONE_MASK) != 0);
+
+	c = auclnt_hold_by_devt(physdev);
+	ASSERT(c != NULL);
+	if ((rv = auclnt_open(c, fmt, oflag)) != 0) {
+		goto fail;
+	}
+	isopen = B_TRUE;
+
+	if ((rv = sproc_hold(c, oflag)) != 0) {
+		goto fail;
+	}
+
+	sc = auclnt_get_private(c);
+	WR(rq)->q_ptr = rq->q_ptr = sc;
+	sc->s_lh = lh;
+	sc->s_rq = rq;
+	sc->s_wq = WR(rq);
+
+	/* start up the input */
+	if (oflag & FREAD) {
+		auclnt_start(auclnt_input_stream(c));
+	}
+
+	auclnt_release(c);
+
+	/* we just reuse same minor number that phys layer used */
+	*devp = makedevice(getmajor(*devp), getminor(physdev));
+
+	qprocson(rq);
+
+	return (0);
+
+fail:
+	if (sc != NULL) {
+		sproc_release(sc);
+	}
+	if (isopen) {
+		auclnt_close(c);
+	}
+	if (lh != NULL) {
+		auclnt_release(c);
+		(void) ldi_close(lh, flag, cr);
+	}
+
+	return (rv);
+}
+
+static int
+sunstr_close(queue_t *rq, int flag, cred_t *cr)
+{
+	sclient_t	*sc;
+	audio_client_t	*c;
+	int		rv;
+
+	sc = rq->q_ptr;
+	c = sc->s_client;
+
+	if ((auclnt_get_minor_type(c) == AUDIO_MINOR_DEVAUDIO) &&
+	    (ddi_can_receive_sig() || (ddi_get_pid() == 0))) {
+		rv = auclnt_drain(c);
+	}
+
+	auclnt_stop(auclnt_output_stream(c));
+	auclnt_stop(auclnt_input_stream(c));
+
+	auclnt_close(c);
+
+	qprocsoff(rq);
+
+	(void) ldi_close(sc->s_lh, flag, cr);
+
+	sproc_release(sc);
+
+	return (rv);
+}
+
+static void
+sunstr_miocdata(sclient_t *sc, mblk_t *mp)
+{
+	struct copyresp		*csp;
+	sioc_t			*ip;
+	mblk_t			*bcont;
+
+	csp = (void *)mp->b_rptr;
+
+	/*
+	 * If no state, then something "bad" has happened.
+	 */
+	if (((ip = (void *)csp->cp_private) == NULL) || (ip->i_sc != sc)) {
+		miocnak(sc->s_wq, mp, 0, EFAULT);
+		return;
+	}
+
+	/*
+	 * If we failed to transfer data to/from userland, then we are
+	 * done.  (Stream head will have notified userland.)
+	 */
+	if (csp->cp_rval != 0) {
+		ip->i_state = FINI;
+		ip->i_mp = mp;
+		sioc_finish(ip);
+		return;
+	}
+
+	/*
+	 * Buffer area for ioctl is attached to chain.
+	 * For an ioctl that didn't have any data to copyin,
+	 * we might need to allocate a new buffer area.
+	 */
+	bcont = mp->b_cont;
+	ip->i_bcont = bcont;
+	mp->b_cont = NULL;
+
+	if (bcont != NULL) {
+		ip->i_data = bcont->b_rptr;
+	}
+
+	/*
+	 * Meaty part of data processing.
+	 */
+	ip->i_state = IOCTL;
+	ip->i_mp = mp;
+
+	/* now, call the handler ioctl */
+	sunstr_ioctl(ip);
+}
+
+static void
+sunstr_mioctl(sclient_t *sc, mblk_t *mp)
+{
+	struct iocblk	*iocp = (void *)mp->b_rptr;
+	sioc_t		*ip;
+
+	/* BSD legacy here: we only support transparent ioctls */
+	if (iocp->ioc_count != TRANSPARENT) {
+		miocnak(sc->s_wq, mp, 0, EINVAL);
+		return;
+	}
+
+	ip = kmem_zalloc(sizeof (*ip), KM_NOSLEEP);
+	if (ip == NULL) {
+		miocnak(sc->s_wq, mp, 0, ENOMEM);
+		return;
+	}
+
+	/* make sure everything is setup in case we need to do copyin/out */
+	ip->i_sc = sc;
+	ip->i_model = iocp->ioc_flag;
+	ip->i_cmd = iocp->ioc_cmd;
+	ip->i_addr = *(caddr_t *)(void *)mp->b_cont->b_rptr;
+	ip->i_state = IOCTL;
+	ip->i_mp = mp;
+	freemsg(mp->b_cont);
+	mp->b_cont = NULL;
+
+	/* now, call the handler ioctl */
+	sunstr_ioctl(ip);
+}
+
+static int
+sunstr_wput(queue_t *wq, mblk_t *mp)
+{
+	sclient_t	*sc = wq->q_ptr;
+	struct iocblk	*iocp;
+
+	switch (DB_TYPE(mp)) {
+	case M_IOCTL:
+		/* Drain ioctl needs to be handled on the service queue */
+		iocp = (void *)mp->b_rptr;
+		if (iocp->ioc_cmd == AUDIO_DRAIN) {
+			if (auclnt_get_minor_type(sc->s_client) ==
+			    AUDIO_MINOR_DEVAUDIO) {
+				(void) putq(wq, mp);
+			} else {
+				miocnak(wq, mp, 0, EINVAL);
+			}
+		} else {
+			sunstr_mioctl(sc, mp);
+		}
+		break;
+
+	case M_IOCDATA:
+		sunstr_miocdata(sc, mp);
+		break;
+
+	case M_FLUSH:
+		/*
+		 * We don't flush the engine.  The reason is that
+		 * other streams might be using the engine.  This is
+		 * fundamentally no different from the case where the
+		 * engine hardware has data buffered in an
+		 * inaccessible FIFO.
+		 *
+		 * Clients that want to ensure no more data is coming
+		 * should stop the stream before flushing.
+		 */
+		if (*mp->b_rptr & FLUSHW) {
+			flushq(wq, FLUSHALL);
+			auclnt_flush(auclnt_output_stream(sc->s_client));
+			*mp->b_rptr &= ~FLUSHW;
+		}
+		if (*mp->b_rptr & FLUSHR) {
+			flushq(RD(wq), FLUSHALL);
+			auclnt_flush(auclnt_input_stream(sc->s_client));
+			qreply(wq, mp);
+		} else {
+			freemsg(mp);
+		}
+		break;
+
+	case M_DATA:
+		/*
+		 * If we don't have an engine, then we can't accept
+		 * write() data.  audio(7i) says we just ignore it,
+		 * so we toss it.
+		 */
+		if (auclnt_get_minor_type(sc->s_client) !=
+		    AUDIO_MINOR_DEVAUDIO) {
+			freemsg(mp);
+		} else {
+			/*
+			 * Defer processing to the queue.  This keeps
+			 * the data ordered, and allows the wsrv
+			 * routine to gather multiple mblks at once.
+			 */
+			if (mp->b_cont != NULL) {
+
+				/*
+				 * If we need to pullup, do it here to
+				 * simplify the rest of the processing
+				 * later.  This should rarely (if
+				 * ever) be necessary.
+				 */
+				mblk_t	*nmp;
+
+				if ((nmp = msgpullup(mp, -1)) == NULL) {
+					freemsg(mp);
+				} else {
+					freemsg(mp);
+					(void) putq(wq, nmp);
+				}
+			} else {
+				(void) putq(wq, mp);
+			}
+		}
+		break;
+
+	default:
+		freemsg(mp);
+		break;
+	}
+	return (0);
+}
+
+static int
+sunstr_wsrv(queue_t *wq)
+{
+	sclient_t	*sc = wq->q_ptr;
+	audio_client_t	*c = sc->s_client;
+	audio_stream_t	*sp;
+	mblk_t		*mp;
+	unsigned	framesz;
+
+	sp = auclnt_output_stream(c);
+
+	framesz = auclnt_get_framesz(sp);
+
+	while ((mp = getq(wq)) != NULL) {
+
+		unsigned	count;
+
+		/* got a message */
+
+		/* if its a drain ioctl, we need to process it here */
+		if (DB_TYPE(mp) == M_IOCTL) {
+			ASSERT((*(int *)(void *)mp->b_rptr) == AUDIO_DRAIN);
+			mutex_enter(&sc->s_lock);
+			mp->b_next = sc->s_draining;
+			sc->s_draining = mp;
+			mutex_exit(&sc->s_lock);
+
+			if (auclnt_start_drain(c) != 0) {
+				sun_drain(c);
+			}
+			continue;
+		}
+
+		ASSERT(DB_TYPE(mp) == M_DATA);
+
+		/*
+		 * Empty mblk require special handling, since they
+		 * indicate EOF.  We treat them separate from the main
+		 * processing loop.
+		 */
+		if (MBLKL(mp) == 0) {
+			struct eofcnt	*eof;
+
+			eof = kmem_zalloc(sizeof (*eof), KM_NOSLEEP);
+			if (eof != NULL) {
+				eof->tail = auclnt_get_head(sp);
+				mutex_enter(&sc->s_lock);
+				list_insert_tail(&sc->s_eofcnt, eof);
+				mutex_exit(&sc->s_lock);
+			}
+			freemsg(mp);
+			continue;
+		}
+
+		count = auclnt_produce_data(sp, (caddr_t)mp->b_rptr,
+		    MBLKL(mp) / framesz);
+
+		mp->b_rptr += count * framesz;
+
+		if (MBLKL(mp) >= framesz) {
+			(void) putbq(wq, mp);
+			break;
+		} else {
+			freemsg(mp);
+		}
+	}
+
+	/* if the stream isn't running yet, start it up */
+	if (!auclnt_is_paused(sp))
+		auclnt_start(sp);
+
+	return (0);
+}
+
+static int
+sunstr_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	if ((cmd != DDI_ATTACH) || (dip == NULL)) {
+		return (DDI_FAILURE);
+	}
+	if (ddi_get_instance(dip) != 0) {
+		return (DDI_FAILURE);
+	}
+
+	mutex_enter(&sdev_lock);
+	sdev_dip = dip;
+	auclnt_walk_devs(sun_create_minors, NULL);
+	mutex_exit(&sdev_lock);
+	ddi_report_dev(dip);
+
+	return (0);
+}
+
+static int
+sunstr_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	if ((cmd != DDI_DETACH) || (dip == NULL)) {
+		return (DDI_FAILURE);
+	}
+	if (ddi_get_instance(dip) != 0) {
+		return (DDI_FAILURE);
+	}
+
+	mutex_enter(&sdev_lock);
+	/* remove all minors */
+	auclnt_walk_devs(sun_remove_minors, NULL);
+	sdev_dip = NULL;
+	mutex_exit(&sdev_lock);
+
+	return (0);
+}
+
+static int
+sunstr_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd, void *arg, void **result)
+{
+	int		error;
+
+	_NOTE(ARGUNUSED(dip));
+	_NOTE(ARGUNUSED(arg));
+
+	switch (cmd) {
+	case DDI_INFO_DEVT2DEVINFO:
+		*result = sdev_dip;
+		error = DDI_SUCCESS;
+		break;
+	case DDI_INFO_DEVT2INSTANCE:
+		*result = 0;
+		error = DDI_SUCCESS;
+		break;
+	default:
+		*result = NULL;
+		error = DDI_FAILURE;
+	}
+	return (error);
+}
+
+static struct module_info sunstr_minfo = {
+	0,		/* used for strlog(1M) only, which we don't use */
+	"austr",
+	0,		/* min pkt size */
+	2048,		/* max pkt size */
+	65536,		/* hi water */
+	32768,		/* lo water */
+};
+
+static struct qinit sunstr_rqinit = {
+	NULL,		/* qi_putp */
+	NULL,		/* qi_srvp */
+	sunstr_open,	/* qi_qopen */
+	sunstr_close,	/* qi_qclose */
+	NULL,		/* qi_qadmin */
+	&sunstr_minfo,	/* qi_minfo */
+	NULL,		/* qi_mstat */
+};
+
+static struct qinit sunstr_wqinit = {
+	sunstr_wput,	/* qi_putp */
+	sunstr_wsrv,	/* qi_srvp */
+	NULL,		/* qi_qopen */
+	NULL,		/* qi_qclose */
+	NULL,		/* qi_qadmin */
+	&sunstr_minfo,	/* qi_minfo */
+	NULL,		/* qi_mstat */
+};
+
+static struct streamtab sunstr_strtab = {
+	&sunstr_rqinit,
+	&sunstr_wqinit,
+	NULL,
+	NULL
+};
+
+struct cb_ops sunstr_cb_ops = {
+	nodev,		/* open */
+	nodev,		/* close */
+	nodev,		/* strategy */
+	nodev,		/* print */
+	nodev,		/* dump */
+	nodev,		/* read */
+	nodev,		/* write */
+	nodev,		/* ioctl */
+	nodev,		/* devmap */
+	nodev,		/* mmap */
+	nodev,		/* segmap */
+	nochpoll,	/* chpoll */
+	ddi_prop_op,	/* prop_op */
+	&sunstr_strtab,	/* str */
+	D_MP,		/* flag */
+	CB_REV, 	/* rev */
+	nodev,		/* aread */
+	nodev,		/* awrite */
+};
+
+static struct dev_ops sunstr_dev_ops = {
+	DEVO_REV,		/* rev */
+	0,			/* refcnt */
+	sunstr_getinfo,		/* getinfo */
+	nulldev,		/* identify */
+	nulldev,		/* probe */
+	sunstr_attach,		/* attach */
+	sunstr_detach,		/* detach */
+	nodev,			/* reset */
+	&sunstr_cb_ops,		/* cb_ops */
+	NULL,			/* bus_ops */
+	NULL,			/* power */
+};
+
+static struct modldrv sunstr_modldrv = {
+	&mod_driverops,
+	"Audio Streams Support",
+	&sunstr_dev_ops,
+};
+
+static struct modlinkage sunstr_modlinkage = {
+	MODREV_1,			/* MODREV_1 indicated by manual */
+	&sunstr_modldrv,
+	NULL
+};
+
+int
+sunstr_init(void)
+{
+	/*
+	 * NB: This *must* be called after the "audio" module's
+	 * _init routine has called auimpl_sun_init().
+	 */
+	return (mod_install(&sunstr_modlinkage));
+}
+
+int
+sunstr_fini(void)
+{
+	return (mod_remove(&sunstr_modlinkage));
+}
+
+int
+sunstr_info(struct modinfo *modinfop)
+{
+	return (mod_info(&sunstr_modlinkage, modinfop));
+}
diff --git a/usr/src/uts/common/io/audio/impl/austr.c b/usr/src/uts/common/io/audio/impl/austr.c
new file mode 100644
index 0000000000..5f85c483b7
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/austr.c
@@ -0,0 +1,62 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Audio Streams Support driver.  This relies on the common audio module.
+ *
+ * The entire driver is a tiny shim for the audio framework, and exists
+ * solely to provide a device node that we can hang STREAMs minor nodes from,
+ * since we can't mix and match STREAMs and character devices in the same
+ * driver, due to limitations in the DDI.
+ */
+
+#include <sys/types.h>
+#include <sys/modctl.h>
+#include <sys/conf.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+
+/* our entire API with "audio" is miniscule */
+extern int sunstr_init(void);
+extern int sunstr_fini(void);
+extern int sunstr_info(struct modinfo *);
+
+int
+_init(void)
+{
+	return (sunstr_init());
+}
+
+int
+_fini(void)
+{
+	return (sunstr_fini());
+}
+
+int
+_info(struct modinfo *modinfop)
+{
+	return (sunstr_info(modinfop));
+}
diff --git a/usr/src/uts/common/io/audio/impl/austr.conf b/usr/src/uts/common/io/audio/impl/austr.conf
new file mode 100644
index 0000000000..4796f135b1
--- /dev/null
+++ b/usr/src/uts/common/io/audio/impl/austr.conf
@@ -0,0 +1,38 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+# Configuration file for the common austr driver.
+#
+
+#
+# We are a pseudo driver.
+#
+name="austr" parent="pseudo" instance=0;
+
+#
+# We want to always be present and never removed.  This is important
+# to ensure that device numbering is not lost due to the driver being
+# forced out of memory automatically.
+#
+ddi-forceattach=1;
+ddi-no-autodetach=1;
diff --git a/usr/src/uts/common/io/audio/inc.flg b/usr/src/uts/common/io/audio/inc.flg
deleted file mode 100644
index 09f67a654c..0000000000
--- a/usr/src/uts/common/io/audio/inc.flg
+++ /dev/null
@@ -1,128 +0,0 @@
-#!/bin/sh
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
-
-# This file brings down all that is needed to build just 
-# audio drivers.
-#
-
-find_files "s.*"	\
-	usr/src/uts/sun/io/audio
-
-find_files "s.*.h"	\
-	usr/src/uts/common/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/sparc/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/sparc/v7/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/sparc/v9/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/intel/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/intel/ia32/sys
-
-find_files "s.*"	\
-	usr/src/uts/intel/ia32/ml
-
-find_files "s.*.h"	\
-	usr/src/uts/sun/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/i86pc/sys
-
-find_files "s.*.h"	\
-	usr/src/uts/common/vm
-
-find_files "s.*"	\
-	usr/src/uts/common/io/warlock
-
-find_files "s.*"	\
-	usr/src/uts/sparc/ml
-
-find_files "s.*"	\
-	usr/src/uts/sparc/warlock	\
-	usr/src/uts/sparc/audiocs	\
-	usr/src/uts/sparc/audiots	\
-	usr/src/uts/sparc/audio810	\
-	usr/src/uts/sparc/audio1575	\
-	usr/src/uts/sparc/audiohd	\
-	usr/src/uts/sparc/audiosup	\
-	usr/src/uts/sparc/mixer		\
-	usr/src/uts/sparc/amsrc2	\
-	usr/src/uts/intel/audio810	\
-	usr/src/uts/intel/audiohd	\
-	usr/src/uts/intel/audiosup	\
-	usr/src/uts/intel/mixer	
-
-echo_file usr/src/uts/sparc/os/driver_classes
-echo_file usr/src/uts/sparc/os/driver_aliases
-echo_file usr/src/uts/sparc/os/name_to_major
-echo_file usr/src/uts/intel/os/driver_classes
-echo_file usr/src/uts/intel/os/driver_aliases
-echo_file usr/src/uts/intel/os/name_to_major
-
-echo_file  usr/src/Makefile.master
-echo_file  usr/src/Makefile.master.64
-echo_file  usr/src/req.flg
-echo_file  usr/src/Makefile.psm
-echo_file  usr/src/Makefile.psm.targ
-echo_file  usr/src/uts/Makefile
-echo_file  usr/src/uts/Makefile.targ
-echo_file  usr/src/uts/Makefile.uts
-echo_file  usr/src/uts/sun/Makefile.files
-echo_file  usr/src/uts/sun/Makefile.rules
-echo_file  usr/src/uts/common/Makefile.files
-echo_file  usr/src/uts/common/Makefile.rules
-echo_file  usr/src/uts/sparc/Makefile
-echo_file  usr/src/uts/sparc/Makefile.files
-echo_file  usr/src/uts/sparc/Makefile.rules
-echo_file  usr/src/uts/sparc/Makefile.sparc
-echo_file  usr/src/uts/sparc/Makefile.sparc.shared
-echo_file  usr/src/uts/sparc/Makefile.targ
-echo_file  usr/src/uts/sparc/Makefile.targ.shared
-echo_file  usr/src/uts/sparc/req.flg
-echo_file  usr/src/uts/i86pc/Makefile.i86pc
-echo_file  usr/src/uts/i86pc/Makefile.i86pc.shared
-echo_file  usr/src/uts/i86pc/Makefile.files
-echo_file  usr/src/uts/i86pc/Makefile.rules
-echo_file  usr/src/uts/i86pc/Makefile.targ
-echo_file  usr/src/uts/i86pc/Makefile.targ.shared
-echo_file  usr/src/uts/i86pc/Makefile.workarounds
-echo_file  usr/src/uts/intel/Makefile
-echo_file  usr/src/uts/intel/Makefile.files
-echo_file  usr/src/uts/intel/Makefile.rules
-echo_file  usr/src/uts/intel/Makefile.intel
-echo_file  usr/src/uts/intel/Makefile.intel.shared
-echo_file  usr/src/uts/intel/Makefile.targ
-echo_file  usr/src/uts/intel/Makefile.targ.shared
diff --git a/usr/src/uts/common/io/audio/sada/amsrc2/README b/usr/src/uts/common/io/audio/sada/amsrc2/README
deleted file mode 100644
index ee89a89634..0000000000
--- a/usr/src/uts/common/io/audio/sada/amsrc2/README
+++ /dev/null
@@ -1,106 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-/*
- * Copyright 2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-Bandlimited Interpolation
-
-In bandlimited interpolation a windowed sinc function is used to
-interpolate input samples and implement sample rate conversion.
-More information on the algorithm can be found on the website at
-http://ccrma-www.stanford.edu/~jos/resample. Choices to be made
-during an implementation include (i) the type of window with which
-to truncate the infinitely long sinc function (ii) the length of
-the latter window (i.e. how many sinc zero-crossings to include)
-(iii) the density of sinc function sampling (iv) the number of bits
-used to encode each sinc coefficient (v) whether to interpolate
-sinc coefficients during sample rate conversion.
-
-In our implementation we (i) use a Kaiser window to truncate
-the sinc function (ii) sample up to the 10th zero crossing (iii)
-sample the interval between each zero crossing 256 times (iv)
-use an integer (32 bits) to encode each coefficient (v) do not,
-for performance reasons, perform coefficient interpolation during
-conversion.
-
-Using the above settings we end up with a conversion table which
-contains (2 * 10 + 1) * 256 = 5376 entries. (We add some more to
-simplify the implementation.) For performance reasons, we use two
-copies of the table, one for up and one for downsampling. Each is
-a reordering of the other. We also use separate functions for up
-and downsampling mono and stereo files giving 4 resampling functions.
-Each can handle, theoretically, any start/target sample rate
-configuration and is accurate to 1Hz.
-
-Upsampling
-
-During conversion the sinc window is centered over the current output
-sample location. We project upwards from input sample values under
-the sinc "umbrella" and multiply and sum corresponding input samples
-and sinc coefficients. The sinc function acts like a low pass filter
-with a cut off frequency of Fs/2 where Fs is the input sample rate.
-The number of multiply-adds is fixed, irrespective of the start and
-target sample rates. Often an input sample location will not intersect
-with a prestored coefficient value and as stated earlier we do not
-interpolate but multiply by the nearest coefficient value.
-
-Downsampling
-
-Downsampling is not as straightforward as upsampling. The stored
-sinc table must be traversed in such a way as to act like a low
-pass filter but this time with a cut off frequency equal to Fs'/2
-where Fs' is the target sample rate. To achieve this the sinc
-coefficients are stretched over a longer interval of the input
-signal (compared to upsampling) and traversed at a slower rate.
-Compared to upsampling there are then more multiply-adds to be
-performed but this is offset by the fact that there are fewer
-output samples to be generated.
-
-Notes
-
-When resampling we are performing an interpolation. This interpolation
-is imperfect and errors will occur. Furthermore, in our implementation
-there is often no direct mapping from filter coefficents to input
-samples and in this case we use the closest filter coefficient which
-introduces more error. These errors lead to unwanted frequencies in
-the audio output signal. Many of these could be removed or reduced by
-(i) employing a greater sampling density over the stored sinc function,
-thereby increasing the overall table size and lowering the error
-introduced by a lack of filter coefficient interpolation or (ii)
-maintaining the same overall table size but interpolating between
-filter coefficients for increased accuracy at a computational cost.
-Implementing either of these would have a significant effect on
-output audio quality.
-
-As is stands our current implementation should theoretically exhibit a
-stopband rejection of 60dB and a transition bandwidth of ~34% (these
-are theoretical values since the errors mentioned above drag quality
-downwards). Our transition band is centered over Fs/2 with the result
-that should there be significant energy in the upper ~17% of the spectrum
-there will be a considerable artifact introduced in the output signal.
-We take that chance. Stopband rejection can be increased at the expense
-of transition bandwidth and transition bandwidth can be lessened by
-using more sinc zero crossings (i.e. lengthening the filter kernel).
diff --git a/usr/src/uts/common/io/audio/sada/amsrc2/am_src2.c b/usr/src/uts/common/io/audio/sada/amsrc2/am_src2.c
deleted file mode 100644
index 1897eb6934..0000000000
--- a/usr/src/uts/common/io/audio/sada/amsrc2/am_src2.c
+++ /dev/null
@@ -1,1037 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-/*
- * Audio Mixer Sample Rate Conversion Routine 2
- *
- * This module is used by the audio mixer to perform sample rate conversion
- * of audio streams. The audio driver actually specifies which sample rate
- * conversion routine it wants to use, and then passes that information to
- * the audio mixer.
- *
- *	NOTE: This module depends on the misc/audiosup and misc/mixer
- *		modules being loaded 1st.
- *
- *	NOTE: We do NOT allocate the buffers used for sample rate conversion.
- *		That is the job of the audio mixer.
- */
-
-#include <sys/types.h>
-#include <sys/modctl.h>
-#include <sys/cmn_err.h>
-#include <sys/kmem.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/impl/am_src2_impl.h>
-#include <sys/audio/impl/am_src2_table.h>
-
-/*
- * Local sample rate conversion routines.
- */
-static int am_src2_adjust(srchdl_t, int, int);
-static int *am_src2_convert(srchdl_t, int, int, int *, int *, int *, int *);
-static void am_src2_exit(srchdl_t, int);
-static size_t am_src2_init(srchdl_t, int);
-static size_t am_src2_size(srchdl_t, audio_prinfo_t *, int, int, int);
-static int am_src2_update(srchdl_t, audio_prinfo_t *, audio_prinfo_t *,
-	void *, int);
-static int am_src2_up_s(am_src2_data_t *, int *, int *, int);
-static int am_src2_up_m(am_src2_data_t *, int *, int *, int);
-static int am_src2_dn_s(am_src2_data_t *, int *, int *, int);
-static int am_src2_dn_m(am_src2_data_t *, int *, int *, int);
-
-/*
- * Global variable to provide generic sample rate conversion facility.
- */
-am_ad_src_entry_t am_src2 = {
-	AM_SRC_VERSION,
-	am_src2_init,
-	am_src2_exit,
-	am_src2_update,
-	am_src2_adjust,
-	am_src2_convert,
-	am_src2_size
-};
-
-/*
- * Module Linkage Structures
- */
-/* Linkage structure for loadable drivers */
-static struct modlmisc amsrc2_modlmisc = {
-	&mod_miscops,			/* drv_modops */
-	AM_SRC2_MOD_NAME,		/* drv_linkinfo */
-};
-
-static struct modlinkage amsrc2_modlinkage =
-{
-	MODREV_1,			/* ml_rev */
-	(void*)&amsrc2_modlmisc,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-/*
- *  Loadable Module Configuration Entry Points
- *
- *
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-	ATRACE("in amsrc2 _init()", 0);
-
-	/* Standard linkage call */
-	if ((error = mod_install(&amsrc2_modlinkage)) != 0) {
-		ATRACE_32("amsrc2 _init() error 1", error);
-		return (error);
-	}
-
-	ATRACE("amsrc2 _init() successful", 0);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description
- *	Module de-initialization, called when driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int	error;
-
-	ATRACE("in amsrc2 _fini()", 0);
-
-	if ((error = mod_remove(&amsrc2_modlinkage)) != 0) {
-		ATRACE_32("amsrc2 _fini() mod_remove failed", error);
-		return (error);
-	}
-
-	ATRACE_32("amsrc2 _fini() successful", error);
-
-	return (error);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo	*modinfop	Pointer to an opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int		rc;
-
-	rc = mod_info(&amsrc2_modlinkage, modinfop);
-
-	ATRACE_32("amsrc2 _info() returning", rc);
-
-	return (rc);
-
-}	/* _info() */
-
-/*
- * am_src2_adjust()
- *
- * Description:
- *	This routine is used to adjust the number of hardware samples so we
- *	know how many channel samples were converted.
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *	int		samples		The number of hardware samples
- *
- * Returns:
- *	>= 0				The number of channel samples
- *	AUDIO_FAILURE			Couldn't adjust the size
- */
-static int
-am_src2_adjust(srchdl_t	handle, int dir, int samples)
-{
-	am_src2_data_t		*pptr;
-	int			value;
-
-	ATRACE("in am_src2_adjust()", handle);
-	ATRACE_32("am_src2_adjust() direction", dir);
-	ATRACE_32("am_src2_adjust() samples", samples);
-
-	/* Get the conversion info */
-	if ((pptr = (am_src2_data_t *)am_get_src_data(handle, dir)) == NULL) {
-		cmn_err(CE_WARN, "!amsrc2: src2_adjust() no pptr");
-		return (AUDIO_FAILURE);
-	}
-
-	/* Do the math */
-	mutex_enter(&pptr->src2_lock);
-	ATRACE_32("am_src2_adjust() inFs", pptr->src2_inFs);
-	ATRACE_32("am_src2_adjust() outFS", pptr->src2_outFs);
-	value = (samples * pptr->src2_inFs) / pptr->src2_outFs;
-	mutex_exit(&pptr->src2_lock);
-
-	ATRACE_32("am_src2_adjust() returning", value);
-
-	return (value);
-
-}       /* am_src2_adjust() */
-
-/*
- * am_src2_convert()
- *
- * Description:
- *	This routine manages the sample rate conversion process. It converts
- *	from src2_inFs to src2_outFs. The input stream must be 16-bit Linear
- *	PCM held as 32-bit integers.
- *
- *	The returned pointer, if valid, must be one of the two passed in
- *	pointers. Otherwise memory will become lost.
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	int		channels	The number of channels in conversion
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *	int		*src		Original data to convert
- *	int		*ptr1		Conversion buffer
- *	int		*ptr2		Conversion buffer (not used)
- *	int		*samples	Pointer to the number of samples to
- *					convert, and when we return, the number
- *					of samples converted.
- *
- * Returns:
- *	valid pointer			Pointer to the converted audio stream
- *	NULL				Conversion failed
- */
-/*ARGSUSED*/
-static int *
-am_src2_convert(srchdl_t handle, int channels, int dir, int *src,
-    int *ptr1, int *ptr2, int *samples)
-{
-	am_src2_data_t		*pdata;
-	int			converted;
-
-	ATRACE("in am_src2_convert()", handle);
-	ATRACE_32("am_src2_convert() direction", dir);
-	ATRACE_32("am_src2_convert() channels", channels);
-
-	/* Get the sample rate conversion data */
-	if ((pdata = (am_src2_data_t *)am_get_src_data(handle, dir)) == NULL) {
-		cmn_err(CE_WARN, "!amsrc2: src2_convert() no pdata");
-		return (NULL);
-	}
-
-	mutex_enter(&pdata->src2_lock);
-
-	/* Set conversion function depending on direction and channels */
-	if (dir == AUDIO_PLAY) {
-		if (channels == AUDIO_CHANNELS_MONO) {
-			pdata->src2_resample = am_src2_up_m;
-		} else {
-			ASSERT(channels == AUDIO_CHANNELS_STEREO);
-			pdata->src2_resample = am_src2_up_s;
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		if (channels == AUDIO_CHANNELS_MONO) {
-			pdata->src2_resample = am_src2_dn_m;
-		} else {
-			ASSERT(channels == AUDIO_CHANNELS_STEREO);
-			pdata->src2_resample = am_src2_dn_s;
-		}
-
-	}
-
-	/* Resample */
-	converted = pdata->src2_resample(pdata, src, ptr1, *samples);
-
-	ATRACE_32("am_src2_convert() final number converted", converted);
-	ATRACE("am_src2_convert() final conversions returning", ptr1);
-
-	*samples = converted;
-
-	mutex_exit(&pdata->src2_lock);
-
-	return (ptr1);
-
-}	/* am_src2_convert() */
-
-/*
- * am_src2_exit()
- *
- * Description:
- *	Free the private data structure allocated in am_src2_init()
- *
- *	NOTE: We do NOT free the buffers used for sample rate conversion.
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *
- * Returns:
- *	void
- */
-static void
-am_src2_exit(srchdl_t handle, int dir)
-{
-	am_src2_data_t		*pdata;
-
-	ATRACE_32("am_src2_exit() direction", dir);
-
-	/* Get pointers, based on which direction we are going */
-	pdata = (am_src2_data_t *)am_get_src_data(handle, dir);
-
-	ATRACE("am_src2_exit() data", pdata);
-
-	if (pdata == NULL) {
-		ATRACE("am_src2_exit() done", pdata);
-		return;
-	}
-
-	/* Get the mutex */
-	mutex_enter(&pdata->src2_lock);
-
-	/* Free memory for table pointers */
-	if (dir == AUDIO_PLAY) {
-		kmem_free((void *)pdata->src2_tables,
-			sizeof (*pdata->src2_tables) * (AM_SRC2_CPZC + 1));
-	}
-
-	/* Release and destroy the mutex */
-	mutex_exit(&pdata->src2_lock);
-
-	mutex_destroy(&pdata->src2_lock);
-
-	kmem_free(pdata, sizeof (*pdata));
-
-	am_set_src_data(handle, dir, NULL);
-
-	ATRACE("am_src2_exit() done", NULL);
-
-}	/* am_src2_exit() */
-
-/*
- * am_src2_init()
- *
- * Description:
- *	Allocate memory for the sample rate conversion private data
- *	structure and initialize the mutex that guarantees we don't
- *	mess with buffers and parameters in the middle of a conversion.
- *	Because the current output sample depends on the current and
- *	past input samples we require room to store past input samples.
- *	Here we return the amount of extra room required in bytes.
- *	We initialise all we can at this point in order to keep the
- *	work required of am_src2_update() to a minimum.
- *
- *	CAUTION: This routine may be called only once without calling
- *	am_src2_exit(), otherwise we'll have a memory leak.
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *
- * Returns:
- *	size_t		prebuffer	Prebuffer memory requirement
- */
-static size_t
-am_src2_init(srchdl_t handle, int dir)
-{
-	am_src2_data_t		*pdata;
-	size_t			prebuffer;
-	int			i;
-
-	ATRACE("in am_src2_init()", handle);
-	ATRACE_32("am_src2_init() direction", dir);
-
-	/* Allocate src data structure */
-	pdata = kmem_zalloc(sizeof (*pdata), KM_SLEEP);
-	ATRACE("am_src2_init() new src data structure", pdata);
-
-	/* Set up mutex */
-	mutex_init(&pdata->src2_lock, NULL, MUTEX_DRIVER, NULL);
-
-	/* Get the mutex */
-	mutex_enter(&pdata->src2_lock);
-
-	if (dir == AUDIO_PLAY) {
-		/* Scale for 32-bit ints */
-		prebuffer = (AM_SRC2_PBUFFER << AM_SRC2_SHIFT2);
-		pdata->src2_pbsize = AM_SRC2_PBUFFER;
-		pdata->src2_i_index = (pdata->src2_pbsize >> AM_SRC2_SHIFT1);
-		pdata->src2_delta_c = 0;
-		pdata->src2_cover = 0;
-		pdata->src2_tables = kmem_alloc(sizeof (*pdata->src2_tables) *
-			(AM_SRC2_CPZC + 1), KM_SLEEP);
-		for (i = 0; i < AM_SRC2_CPZC + 1; i++) {
-			pdata->src2_tables[i] =
-				&_amsrc2tables[i * AM_SRC2_COFFS];
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		/* Scale for 32-bit ints */
-		prebuffer = (AM_SRC2_RBUFFER << AM_SRC2_SHIFT2);
-		pdata->src2_pbsize = AM_SRC2_RBUFFER;
-		pdata->src2_i_index = (pdata->src2_pbsize >> AM_SRC2_SHIFT1);
-		pdata->src2_delta_c = 0;
-		pdata->src2_delta_n = 0;
-		pdata->src2_cover = 0;
-		pdata->src2_table = _amsrc2table;
-	}
-
-	/* Set the data */
-	am_set_src_data(handle, dir, pdata);
-
-	/* Release the mutex */
-	mutex_exit(&pdata->src2_lock);
-
-	return (prebuffer);
-
-}	/* am_src2_init() */
-
-/*
- * am_src2_size()
- *
- * Description:
- *	Determine the size of a buffer, in bytes, needed to hold the number
- *	of "samples" when they are converted. We adjust the size based on
- *	the number of source hardware channels.
- *
- *	NOTE: This size of the buffer is based on 32-bit per sample.
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	audio_prinfo_t	*prinfo		Ptr to the channel's information
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *	int		samples		The number of samples
- *	int		hw_channels	Number of hardware channels
- *
- * Returns:
- *	size			The max # of bytes any sample rate conversion
- *				step could need in buffer space
- *	AUDIO_FAILURE		Couldn't find this size
- */
-static size_t
-am_src2_size(srchdl_t handle, audio_prinfo_t *prinfo, int dir, int samples,
-    int hw_channels)
-{
-	am_src2_data_t		*pdata;
-	size_t			size;
-
-	ATRACE("in am_src2_size()", handle);
-	ATRACE_32("am_src2_size() direction", dir);
-	ATRACE_32("am_src2_size() samples", samples);
-	ATRACE_32("am_src2_size() hw_channels", hw_channels);
-
-	pdata = (am_src2_data_t *)am_get_src_data(handle, dir);
-	if (pdata == NULL) {
-		cmn_err(CE_WARN, "!amsrc2: src2_size() no pdata");
-		return ((size_t)AUDIO_FAILURE);
-	}
-
-	ASSERT(pdata);
-
-	mutex_enter(&pdata->src2_lock);
-	ASSERT(pdata->src2_inFs);
-	ASSERT(pdata->src2_outFs);
-	ATRACE_32("am_src2_size() inFs", pdata->src2_inFs);
-	ATRACE_32("am_src2_size() outFs", pdata->src2_outFs);
-
-	/* Round up to maximum frame size if any leftover */
-	size = samples * pdata->src2_outFs;
-	if (size % pdata->src2_inFs) {
-		size /= pdata->src2_inFs;
-		size++;
-	} else {
-		size /= pdata->src2_inFs;
-	}
-	if (size % AUDIO_CHANNELS_STEREO) {
-		size++;
-	}
-
-	/* Scale for 32-bit ints */
-	size <<= AM_SRC2_SHIFT2;
-
-	/* Now adjust for the number of channels */
-	if (dir == AUDIO_PLAY) {
-		if (prinfo->channels < hw_channels) {
-			size *= hw_channels;
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		if (prinfo->channels > hw_channels) {
-			size *= prinfo->channels;
-		}
-	}
-
-	mutex_exit(&pdata->src2_lock);
-
-	ATRACE_32("am_src2_size() returned size", size);
-
-	return (size);
-
-}	/* am_src2_size() */
-
-/*
- * am_src2_update()
- *
- * Description:
- *	Initialise the sample rate conversion private data structure. Here
- *	we (re-)initialise what we couldn't in am_src2_init().
- *
- * Arguments:
- *	srchdl_t	handle		Mixer/src handle
- *	audio_prinfo_t	*ch_prinfo	Ptr to the channel's information
- *	audio_prinfo_t	*hw_prinfo	Ptr to the Codec's information
- *	void		*src_info	Src information (not used)
- *	int		dir		Direction, AUDIO_PLAY or AUDIO_RECORD
- *
- * Returns:
- *	AUDIO_SUCCESS			Initialisation succeeded
- *	AUDIO_FAILURE			Initialisation failed
- */
-/*ARGSUSED*/
-static int
-am_src2_update(srchdl_t handle, audio_prinfo_t *ch_prinfo, audio_prinfo_t
-	*hw_prinfo, void *src_info, int dir)
-{
-	am_src2_data_t		*pdata;
-
-	ATRACE("in am_src2_update()", handle);
-	ATRACE_32("am_src2_update() direction", dir);
-	ATRACE("am_src2_update() ch_prinfo", ch_prinfo);
-	ATRACE("am_src2_update() hw_prinfo", hw_prinfo);
-
-	/* Get src data structure */
-	if ((pdata = (am_src2_data_t *)am_get_src_data(handle, dir)) == NULL) {
-		cmn_err(CE_WARN, "!amsrc2: src2_update() no pdata");
-		return (AUDIO_FAILURE);
-	}
-
-	mutex_enter(&pdata->src2_lock);
-
-	ATRACE("am_src2_update() pdata", pdata);
-
-	/*
-	 * Set direction dependent data. Note: we should probably do some
-	 * checking here to make sure sane samples rates are being used.
-	 * Not as strict as amsrc1 but there should be something.
-	 */
-	if (dir == AUDIO_PLAY) {
-		pdata->src2_inFs = ch_prinfo->sample_rate;
-		pdata->src2_outFs = hw_prinfo->sample_rate;
-		pdata->src2_csteps = ((pdata->src2_inFs << AM_SRC2_CPZC_SHIFT) /
-			pdata->src2_outFs);
-		pdata->src2_cmod = ((pdata->src2_inFs << AM_SRC2_CPZC_SHIFT) %
-			pdata->src2_outFs);
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		pdata->src2_inFs = hw_prinfo->sample_rate;
-		pdata->src2_outFs = ch_prinfo->sample_rate;
-		pdata->src2_csteps = (pdata->src2_inFs * ch_prinfo->channels
-			<< AM_SRC2_CPZC_SHIFT) / pdata->src2_outFs;
-		pdata->src2_cmod = ((pdata->src2_inFs * ch_prinfo->channels
-			<< AM_SRC2_CPZC_SHIFT) % pdata->src2_outFs);
-		pdata->src2_tsteps = ((pdata->src2_outFs << AM_SRC2_CPZC_SHIFT)
-			/ pdata->src2_inFs);
-		/*
-		 * Increment the filter step size if there is any left over.
-		 * Otherwise we can overrun the end of the input signal when
-		 * downsampling as the rounding errors accumulate.
-		 */
-		if ((pdata->src2_outFs << AM_SRC2_CPZC_SHIFT)
-			% pdata->src2_inFs) {
-			pdata->src2_tsteps++;
-		}
-	}
-
-	ATRACE("am_src2_update() ch_prinfo", ch_prinfo);
-	ATRACE_32("am_src2_update() src2_inFs", pdata->src2_inFs);
-	ATRACE_32("am_src2_update() src2_outFs", pdata->src2_outFs);
-
-	mutex_exit(&pdata->src2_lock);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_src2_update() */
-
-/*
- * am_src2_up_m()
- *
- * Description:
- *	Carry out upsampling on a mono buffer.
- *
- * Arguments:
- *	am_src2_data_t	*parms		Conversion parameters structure
- *	int		inbuf		The input buffer to convert
- *	int		outbuf		The converted audio buffer
- *	int		samples		The number of samples to convert
- *
- * Returns:
- *	>= 0				The number of samples after conversion
- */
-static int
-am_src2_up_m(am_src2_data_t *parms, int *inbuf, int *outbuf, int samples)
-{
-	long long	sum;		/* Output sample value */
-	int		k;		/* Coefficient counter */
-	int		*t;		/* Pointer into table */
-	int		count;		/* Output sample counter */
-	int		s_index;	/* Input sample index */
-	int		l_index;	/* Last sample index */
-	int		c_index;	/* Starting coefficient index */
-	int		rounding;	/* For rounding to nearest index */
-	int		outFs;		/* Local copies of variables */
-	int		cmod;
-	int		cover;
-	int		csteps;
-	int		i_index;
-	int		delta_c;
-	int		**tables;
-
-	/* Initialise */
-	count = 0;
-	cmod = parms->src2_cmod;
-	cover = parms->src2_cover;
-	tables = parms->src2_tables;
-	csteps = parms->src2_csteps;
-	i_index = parms->src2_i_index;
-	delta_c = parms->src2_delta_c;
-	i_index += (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-	outFs = parms->src2_outFs;
-	l_index = samples + (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-
-	ATRACE("in am_src2_up_m()", NULL);
-	ATRACE_32("in am_src2_up_m() cmod", cmod);
-	ATRACE_32("in am_src2_up_m() cover", cover);
-	ATRACE_32("in am_src2_up_m() csteps", csteps);
-	ATRACE_32("in am_src2_up_m() i_index", i_index);
-	ATRACE_32("in am_src2_up_m() delta_c", delta_c);
-	ATRACE_32("in am_src2_up_m() outFs", outFs);
-	ATRACE_32("in am_src2_up_m() l_index", l_index);
-	ATRACE_32("in am_src2_up_m() samples", samples);
-
-	/* Continue until the end */
-	while (i_index < l_index) {
-
-		/* Starting coefficient */
-		c_index = AM_SRC2_CPZC - delta_c;
-		t = tables[c_index];
-
-		/* Starting sample */
-		s_index = i_index + AM_SRC2_NZCS;
-
-		/*
-		 * Calculate output sample. Note we work from right to left
-		 * starting with the rightmost sample and going backwards.
-		 * We do it this way so a while loop can be used for better
-		 * efficiency. This relies on the fact that t[0] is always
-		 * zero.
-		 */
-		sum = 0;
-		k = AM_SRC2_COFFS - 1;
-		while (k) {
-			sum += t[k--] * inbuf[s_index--];
-		}
-
-		/* Write out */
-		outbuf[count++] = (sum >> AM_SRC2_COFF_SHIFT);
-
-		/* Increment counters and pointers */
-		cover += cmod;
-		rounding = (cover << AM_SRC2_SHIFT1) / outFs;
-		cover -= rounding * outFs;
-		delta_c += rounding + csteps;
-		i_index += (delta_c >> AM_SRC2_CPZC_SHIFT);
-		delta_c &= AM_SRC2_CPZC_MASK;
-	}
-
-	/* Put back */
-	parms->src2_i_index = i_index - l_index;
-	parms->src2_delta_c = delta_c;
-	parms->src2_cover = cover;
-
-	/* Copy to front */
-	bcopy(&inbuf[samples], inbuf, parms->src2_pbsize * sizeof (*inbuf));
-
-	return (count);
-
-}	/* am_src2_up_m() */
-
-/*
- * am_src2_up_s()
- *
- * Description:
- *	Carry out upsampling on a stereo buffer.
- *
- * Arguments:
- *	am_src2_data_t	*parms		Conversion parameters structure
- *	int		inbuf		The input buffer to convert
- *	int		outbuf		The converted audio buffer
- *	int		samples		The number of samples to convert
- *
- * Returns:
- *	> 0				The number of samples after conversion
- */
-static int
-am_src2_up_s(am_src2_data_t *parms, int *inbuf, int *outbuf, int samples)
-{
-	long long	sum_l;		/* Output sample value */
-	long long	sum_r;		/* Output sample value */
-	int		c;		/* Coefficient value */
-	int		*t;		/* Pointer into table */
-	int		k;		/* Coefficient counter */
-	int		count;		/* Output sample counter */
-	int		s_index;	/* Input sample index */
-	int		l_index;	/* Last sample index */
-	int		c_index;	/* Starting coefficient index */
-	int		rounding;	/* For rounding to nearest index */
-	int		outFs;		/* Local copies of variables */
-	int		cmod;
-	int		cover;
-	int		csteps;
-	int		i_index;
-	int		delta_c;
-	int		**tables;
-
-	/* Initialise */
-	count = 0;
-	cmod = parms->src2_cmod;
-	cover = parms->src2_cover;
-	tables = parms->src2_tables;
-	csteps = parms->src2_csteps;
-	i_index = parms->src2_i_index;
-	delta_c = parms->src2_delta_c;
-	i_index += (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-	outFs = parms->src2_outFs;
-	l_index = samples + (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-
-	ATRACE("in am_src2_up_s()", NULL);
-	ATRACE_32("in am_src2_up_s() cmod", cmod);
-	ATRACE_32("in am_src2_up_s() cover", cover);
-	ATRACE_32("in am_src2_up_s() csteps", csteps);
-	ATRACE_32("in am_src2_up_s() i_index", i_index);
-	ATRACE_32("in am_src2_up_s() delta_c", delta_c);
-	ATRACE_32("in am_src2_up_s() outFs", outFs);
-	ATRACE_32("in am_src2_up_s() l_index", l_index);
-	ATRACE_32("in am_src2_up_s() samples", samples);
-
-	/* Continue until the end */
-	while (i_index < l_index) {
-
-		/* Starting coefficient */
-		c_index = AM_SRC2_CPZC - delta_c;
-		t = tables[c_index];
-
-		/* Starting sample */
-		s_index = i_index + (AM_SRC2_NZCS << AM_SRC2_SHIFT1) + 1;
-
-		/*
-		 * Calculate output sample. Note we work from right to left
-		 * starting with the rightmost sample and going backwards.
-		 * We do it this way so a while loop can be used for better
-		 * efficiency. This relies on the fact that t[0] is always
-		 * zero.
-		 */
-		sum_l = sum_r = 0;
-		k = AM_SRC2_COFFS - 1;
-		while (k) {
-			c = t[k--];
-			sum_r += c * inbuf[s_index--];
-			sum_l += c * inbuf[s_index--];
-		}
-
-		/* Write out */
-		outbuf[count++] = (sum_l >> AM_SRC2_COFF_SHIFT);
-		outbuf[count++] = (sum_r >> AM_SRC2_COFF_SHIFT);
-
-		/* Increment counters and pointers */
-		cover += cmod;
-		rounding = (cover << AM_SRC2_SHIFT1) / outFs;
-		cover -= rounding * outFs;
-		delta_c += rounding + csteps;
-		i_index += (delta_c >> AM_SRC2_CPZC_SHIFT);
-		i_index += (i_index & AM_SRC2_STEREO_MASK);
-		delta_c &= AM_SRC2_CPZC_MASK;
-	}
-
-	/* Put back */
-	parms->src2_i_index = i_index - l_index;
-	parms->src2_delta_c = delta_c;
-	parms->src2_cover = cover;
-
-	/* Copy to front */
-	bcopy(&inbuf[samples], inbuf, parms->src2_pbsize * sizeof (*inbuf));
-
-	return (count);
-
-}	/* am_src2_up_s() */
-
-/*
- * am_src2_dn_m()
- *
- * Description:
- *	Carry out downsampling on a mono buffer.
- *
- * Arguments:
- *	am_src2_data_t	*parms		Conversion parameters structure
- *	int		inbuf		The input buffer to convert
- *	int		outbuf		The converted audio buffer
- *	int		samples		The number of samples to convert
- *
- * Returns:
- *	> 0				The number of samples after conversion
- */
-static int
-am_src2_dn_m(am_src2_data_t *parms, int *inbuf, int *outbuf, int samples)
-{
-	long long	sum;		/* Output sample value */
-	int		count;		/* Output sample counter */
-	int		s_index;	/* Input sample index */
-	int		l_index;	/* Last sample index */
-	int		c_index;	/* Starting coefficient index */
-	int		rounding;	/* For rounding to nearest index */
-	int		outFs;		/* Local copies of variables */
-	int		inFs;
-	int		cmod;
-	int		cover;
-	int		csteps;
-	int		tsteps;
-	int		*table;
-	int		i_index;
-	int		delta_c;
-	int		delta_n;
-
-	/* Initialise */
-	count = 0;
-	cmod = parms->src2_cmod;
-	table = parms->src2_table;
-	cover = parms->src2_cover;
-	tsteps = parms->src2_tsteps;
-	csteps = parms->src2_csteps;
-	i_index = parms->src2_i_index;
-	delta_c = parms->src2_delta_c;
-	delta_n = parms->src2_delta_n;
-	i_index += (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-	outFs = parms->src2_outFs;
-	inFs = parms->src2_inFs;
-	l_index = samples + (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-
-	ATRACE("in am_src2_dn_m()", NULL);
-	ATRACE_32("in am_src2_dn_m() cmod", cmod);
-	ATRACE_32("in am_src2_dn_m() cover", cover);
-	ATRACE_32("in am_src2_dn_m() csteps", csteps);
-	ATRACE_32("in am_src2_dn_m() tsteps", tsteps);
-	ATRACE_32("in am_src2_dn_m() i_index", i_index);
-	ATRACE_32("in am_src2_dn_m() delta_c", delta_c);
-	ATRACE_32("in am_src2_dn_m() delta_n", delta_n);
-	ATRACE_32("in am_src2_dn_m() outFs", outFs);
-	ATRACE_32("in am_src2_dn_m() inFs", inFs);
-	ATRACE_32("in am_src2_dn_m() l_index", l_index);
-	ATRACE_32("in am_src2_dn_m() samples", samples);
-
-	/* Continue until the end */
-	while (i_index < l_index) {
-
-		/* Starting coefficient and sample */
-		c_index = AM_SRC2_MIDDLE - delta_n;
-		s_index = i_index - ((c_index - AM_SRC2_START) / tsteps);
-		c_index -= tsteps * ((c_index - AM_SRC2_START) / tsteps);
-
-		/* Calculate output sample */
-		sum = 0;
-		while (c_index <= AM_SRC2_END) {
-			sum += (table[c_index] * inbuf[s_index++]);
-			c_index += tsteps;
-		}
-
-		/* Write out */
-		outbuf[count++] = ((sum * outFs / inFs) >> AM_SRC2_COFF_SHIFT);
-
-		/* Increment counters and pointers */
-		cover += cmod;
-		rounding = (cover << AM_SRC2_SHIFT1) / outFs;
-		cover -= rounding * outFs;
-		delta_c += rounding + csteps;
-		i_index += (delta_c >> AM_SRC2_CPZC_SHIFT);
-		delta_c &= AM_SRC2_CPZC_MASK;
-		delta_n = delta_c * outFs / inFs;
-	}
-
-	/* Put back */
-	parms->src2_i_index = i_index - l_index;
-	parms->src2_delta_c = delta_c;
-	parms->src2_delta_n = delta_n;
-	parms->src2_cover = cover;
-
-	/* Copy to front */
-	bcopy(&inbuf[samples], inbuf, parms->src2_pbsize * sizeof (*inbuf));
-
-	return (count);
-
-}	/* am_src2_dn_m() */
-
-/*
- * am_src2_dn_s()
- *
- * Description:
- *	Carry out downsampling on a stereo buffer.
- *
- * Arguments:
- *	am_src2_data_t	*parms		Conversion parameters structure
- *	int		inbuf		The input buffer to convert
- *	int		outbuf		The converted audio buffer
- *	int		samples		The number of samples to convert
- *
- * Returns:
- *	> 0				The number of samples after conversion
- */
-static int
-am_src2_dn_s(am_src2_data_t *parms, int *inbuf, int *outbuf, int samples)
-{
-	long long	sum_l;		/* Output sample value */
-	long long	sum_r;		/* Output sample value */
-	int		c;		/* Coefficient value */
-	int		count;		/* Output sample counter */
-	int		s_index;	/* Input sample index */
-	int		l_index;	/* Last sample index */
-	int		c_index;	/* Starting coefficient index */
-	int		rounding;	/* For rounding to nearest index */
-	int		outFs;		/* Local copies of variables */
-	int		inFs;
-	int		cmod;
-	int		cover;
-	int		csteps;
-	int		*table;
-	int		tsteps;
-	int		i_index;
-	int		delta_c;
-	int		delta_n;
-
-	/* Initialise */
-	count = 0;
-	cmod = parms->src2_cmod;
-	table = parms->src2_table;
-	cover = parms->src2_cover;
-	tsteps = parms->src2_tsteps;
-	csteps = parms->src2_csteps;
-	i_index = parms->src2_i_index;
-	delta_c = parms->src2_delta_c;
-	delta_n = parms->src2_delta_n;
-	i_index += (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-	outFs = parms->src2_outFs;
-	inFs = parms->src2_inFs;
-	l_index = samples + (parms->src2_pbsize >> AM_SRC2_SHIFT1);
-
-	ATRACE("in am_src2_dn_s()", NULL);
-	ATRACE_32("in am_src2_dn_s() cmod", cmod);
-	ATRACE_32("in am_src2_dn_s() cover", cover);
-	ATRACE_32("in am_src2_dn_s() csteps", csteps);
-	ATRACE_32("in am_src2_dn_s() tsteps", tsteps);
-	ATRACE_32("in am_src2_dn_s() i_index", i_index);
-	ATRACE_32("in am_src2_dn_s() delta_c", delta_c);
-	ATRACE_32("in am_src2_dn_s() delta_n", delta_n);
-	ATRACE_32("in am_src2_dn_s() outFs", outFs);
-	ATRACE_32("in am_src2_dn_s() inFs", inFs);
-	ATRACE_32("in am_src2_dn_s() l_index", l_index);
-	ATRACE_32("in am_src2_dn_s() samples", samples);
-
-	/* Continue until the end */
-	while (i_index < l_index) {
-
-		/* Starting coefficient and sample */
-		c_index = AM_SRC2_MIDDLE - delta_n;
-		s_index = i_index - (i_index & AM_SRC2_STEREO_MASK);
-		s_index -= (((c_index - AM_SRC2_START) / tsteps) <<
-			AM_SRC2_SHIFT1);
-		c_index -= tsteps * ((c_index - AM_SRC2_START) / tsteps);
-
-		sum_l = sum_r = 0;
-		while (c_index <= AM_SRC2_END) {
-			c = table[c_index];
-			sum_l += c * inbuf[s_index++];
-			sum_r += c * inbuf[s_index++];
-			c_index += tsteps;
-		}
-
-		/* Write out */
-		outbuf[count++] = (sum_l * outFs / inFs) >> AM_SRC2_COFF_SHIFT;
-		outbuf[count++] = (sum_r * outFs / inFs) >> AM_SRC2_COFF_SHIFT;
-
-		/* Increment counters and pointers */
-		cover += cmod;
-		rounding = (cover << AM_SRC2_SHIFT1) / outFs;
-		cover -= rounding * outFs;
-		delta_c += rounding + csteps;
-		i_index += (delta_c >> AM_SRC2_CPZC_SHIFT);
-		delta_c &= AM_SRC2_CPZC_MASK;
-		delta_n = ((i_index & AM_SRC2_STEREO_MASK) <<
-			(AM_SRC2_CPZC_SHIFT - 1)) + (delta_c >> AM_SRC2_SHIFT1);
-		delta_n = delta_n * outFs / inFs;
-	}
-
-	/* Put back */
-	parms->src2_i_index = i_index - l_index;
-	parms->src2_delta_c = delta_c;
-	parms->src2_delta_n = delta_n;
-	parms->src2_cover = cover;
-
-	/* Copy to front */
-	bcopy(&inbuf[samples], inbuf, parms->src2_pbsize * sizeof (*inbuf));
-
-	return (count);
-
-}	/* am_src2_dn_s() */
diff --git a/usr/src/uts/common/io/audio/sada/drv/audio810/README b/usr/src/uts/common/io/audio/sada/drv/audio810/README
deleted file mode 100644
index c3cfcd773d..0000000000
--- a/usr/src/uts/common/io/audio/sada/drv/audio810/README
+++ /dev/null
@@ -1,383 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-# Design Document for Metropolis (Amd8111/ALC655) Audio Driver
-
-1. Technical overview
-
-SADA (Solaris audio driver architecture) is the native audio framework of
-Solaris. It supports quickly writing new STREAMS audio drivers by providing
-a number of library modules. There are two types of library modules, the
-audio support module and audio personality modules. The unique audio support
-module provides driver instance, buffer and state management routines. Audio
-personality modules implement the semantics of the device interface. The mixer
-module, as a personality module, implements the semantics of the audio(7I) and
-mixer(7I) man pages. And the another personality module, amsrc2, provides the
-sample rate conversion for mixer. So a SADA driver with mixer function would
-communicate with audio support module, mixer module, and amsrc2 module or
-another conversion module.
-
-
-1.1 Audio support module
-
-The audio support module provides a number of common routines for managing the
-instances and state information for the audio driver. Audio drivers that don't
-need special open(), close(), put(), svc(), and getinfo() routines set their
-qinit and dev_ops data structures as follows:
-
-	/* read queue */
-	static struct qinit xx_rqueue = {
-		audio_sup_rput,		/* put procedure */
-		audio_sup_rsvc,		/* service procedure */
-		audio_sup_open,		/* open procedure */
-		audio_sup_close,	/* close procedure */
-		NULL,			/* unused */
-		&xx_modinfo,		/* module parameters */
-		NULL			/* module statistics */
-	};
-
-	/* write queue */
-	static struct qinit xx_wqueue = {
-		audio_sup_wput,		/* put procedure */
-		audio_sup_wsvc,		/* service procedure */
-		NULL,			/* open procedure */
-		NULL,			/* close procedure */
-		NULL,			/* unused */
-		&xx_modinfo,		/* module parameters */
-		NULL			/* module statistics */
-	};
-
-	/* Device operations structure */
-	static struct dev_ops audiots_dev_ops = {
-		DEVO_REV,		/* devo_rev */
-		0,			/* devo_refcnt */
-		audio_sup_getinfo,	/* devo_getinfo */
-		nulldev,		/* devo_identify - obsolete */
-		nulldev,		/* devo_probe */
-		xx_attach,		/* devo_attach */
-		xx_detach,		/* devo_detach */
-		nodev,			/* devo_reset */
-		&xx_cb_ops,		/* devi_cb_ops */
-		NULL,			/* devo_bus_ops */
-		xx_power		/* devo_power */
-	};
-
-
-	The audio support module manages the device minor nodes for each
-	instance of an audio device. Minor nodes are mapped as follows
-
-		0 AUDIO	/dev/audio, /dev/sound/?
-		1 AUDIOCTL	/dev/audioctl, /dev/sound/?ctl
-		2 WTABLE	future wave table synthesizer
-		3 MIDI		future hardware MIDI ports
-		4 ATIME	future audio timer, probably something like
-				SMPTE, which is used to sync audio and video,
-				there are a lot of MIDI devices that also
-				provide SMPTE information
-		5 - 7		unassigned
-		8 USER1	available for any use
-		9 USER2	available for any use
-		10 USER3	available for any use
-
-		11 - 127	used when the open is cloned
-
-	For example, if an app opens /dev/audio that means that minor node 0
-	was opened. audiosup sees this and knows that it needs to call
-	am_open() in the mixer, which it does. am_open() enforces the open()
-	semantics for /dev/audio. Assuming all is legal it checks looks for
-	the first clone minor node that isn't being used. It then constructs
-	a new device number based on this new minor number and this is
-	returned to the kernel. Thus when open() returns you have a cloned
-	node. Since the mixer doesn't make the node you don't see an entry
-	in /devices, it is invisible.
-
-	The number of minor numbers at the beginning is tunable, as is the
-	total minor numbers per instance.
-
-
-1.2 Audio Mixer Module
-
-	The mixer implements the semantics of the audio(7I) and mixer(7I)
-	man pages. Doing so it is the module that provides /dev/audio(ctl).
-
-	The mixer supports two modes of operation, mixer on and mixer off.
-	When the mixer is off it supports the audio(7I) semantics and just
-	a few new ioctl()s from mixer(7I). Essentially it enforces only one
-	play and one record stream. Also, the audioctl devices represents
-	the hardware.
-
-	When the mixer is enabled it takes an audio stream, converts it to
-	32 bit signed linear PCM and saves this converted data as well as
-	the original data. When the audio driver needs audio to play the
-	mixer scans all clone channels looking for AUDIO play channels. When
-	one is found the audio data is retrieved and summed. When all audio
-	data is summed it is clipped and converted to the format that the
-	audio device is programmed to use. All semantics of audio(7I) and
-	mixer(7I) are available. One of the biggest differences is that
-	the audioctl channel may now represent either the audio device or the
-	clone channel. The rules for this is very complicated and are
-	explained elsewhere.
-
-	The mixer also supports record, which is essentially the play in
-	reverse. The biggest difference is that record doesn't save any
-	messages. As for play, the mixer may be on or off. If off then the
-	recorded audio is passed straight to the application. If on then
-	the mixer scans all clone channels looking for AUDIO record channels.
-	When one is found the audio is converted to the right format and
-	sent to the application. For both mixer modes the recorded audio is
-	queued up in a buffer before it is sent to the application. The size
-	of this buffer is set by the record buffer size.
-
-1.3 Amsrc Module
-
-	This is the sample rate converted that the mixer uses when the mixer
-	is enabled. Currently there are just two sample rate converter modules.
-	However the mixer supports the audio driver author providing his/her
-	own. This can be for performance or audio quality reasons. In this
-	project we use the amsrc2 module.
-
-
-2. Hardware Configuration
-
-	This driver is for the low end audio configuration of Metropols
-	workstation. This configuration consists of two components, controller
-	and codec. The controller is integrated in the southbridge, amd8111, as
-	a function of amd8111. And the ALC655 codec is an AC97 2.3 compatible
-	codec. It is a 16-bit full-duplex six-channel audio codec with fixed
-	48khz sampling rate.
-
-
-3. Software Environment
-
-	Because SADA framework supports only mono/stereo at most, which means
-	SADA does not support multi-channel(3, 4, 6 channel), this driver cannot
-	support multi-channel unless we modify SADA framework. We plan to use
-	Solaris 10&9 as the target operating system for the driver.
-
-
-4. The features/functions
-
-	The driver will be placed in /kernel/drv. Like other audio drivers,
-	the driver can be added/removed by users via "add_drv/rem_drv" tools.
-
-5. The source code structure
-
-	The driver consists of three files, audio810.c, audio810_impl.h and
-	audio810.h. The audio810.c is the C source file which contains all
-	the C codes of the driver. The audio810.h is the header file containing
-	only some baud rate definitions. The audio810_impl.h also a header file
-	which contains hardware specific information, such as data structures.
-
-	In audio810.c file, we implement the entry functions defined by mixer
-	module as follows:
-
-	1) audio810_ad_set_config,	/* ad_set_config() */
-	2) audio810_ad_set_format,	/* ad_set_format() */
-	3) audio810_ad_start_play,	/* ad_start_play() */
-	4) audio810_ad_pause_play,	/* ad_pause_play() */
-	5) audio810_ad_stop_play,	/* ad_stop_play() */
-	6) audio810_ad_start_record,	/* ad_start_record() */
-	7) audio810_ad_stop_record,	/* ad_stop_record() */
-
-6. The implementations
-	In this section, we describe the design of the driver in detail.
-
-6.1 Main data structure
-
-6.1.1 Buffer descriptor list
-	According to the datasheet of amd8111, the buffer descriptor list
-	(BDL) allows device drivers to program DMA transfer using by the
-	audio controller. The BDL is an array of up to 32 entry, each of
-	which describes a data buffer. So we define the BDL entry structure
-	as follows:
-
-	struct i810_bd_entry {
-		uint32_t	buf_base;	/* the address of the buffer */
-		uint16_t	buf_len;	/* the number of samples */
-		uint16_t	reserved:14;
-		uint8_t	cmd_bup:1;	/* interrupt if underflows */
-		uint8_t	cmd_ioc:1;	/* interrupt if completing this entry */
-	};
-
-	AMD8111 audio function has separated DMA engine for PCM in/out, so
-	we must set the two BDLs. For simplification, we allocate continuous
-	DMA memory for all BDLs, and define the BDL as follows,
-
-	struct i810_bd_list {
-		i810_bd_entry_t		pcm_in[I810_BD_NUMS];
-		i810_bd_entry_t		pcm_out[I810_BD_NUMS];
-	};
-
-
-6.1.2 Data buffer for BDL entry
-
-	For simplification, we allocate DMA buffer for each BDL entry.
-
-	struct i810_bdlist_chunk {
-	caddr_t			data_buf;	/* virtual address of buffer */
-	caddr_t			addr_phy;	/* physical address of buffer */
-	ddi_dma_handle_t	dma_handle;	/* dma handle */
-	ddi_acc_handle_t	acc_handle;	/* access handle */
-	size_t			real_len;	/* real len */
-	};
-
-
-6.1.3 Sample buffer
-
-	struct i810_sample_buf {
-	boolean_t	io_started;	/* start/stop state for play/record */
-	int		avail;		/* the number of available chunk(s) */
-	uint8_t		tail;		/* For CPU, 1st available BD entry */
-	uint8_t		head;		/* For CPU, 1st BD entry to reclaim */
-	i810_bdlist_chunk_t	chunk[2];	/* 2 chunks for each buffers */
-	};
-
-
-6.1.4 Soft state
-
-	Every instance of the hardware should have a soft state to save
-	instance-wide data in it.
-
-	struct audio810_state {
-	kmutex_t		inst_lock;	/* state protection lock */
-	ddi_iblock_cookie_t	intr_iblock;
-	dev_info_t		*dip;		/* used by audio810_getinfo() */
-	audiohdl_t		audio_handle;	/* audio handle */
-	am_ad_info_t		ad_info;	/* audio device info state */
-	uint16_t		codec_shadow[64]; /* shadow of AC97 registers */
-
-	boolean_t		var_sr;		/* variable sample rate ? */
-	ddi_acc_handle_t	pci_conf_handle; /* pci configuration space */
-	ddi_acc_handle_t	am_regs_handle; /* for audio mixer register */
-	ddi_acc_handle_t	bm_regs_handle; /* for bus master register */
-	caddr_t			am_regs_base;	/* base of audio mixer regs */
-	caddr_t			bm_regs_base;	/* base of bus master regs */
-
-	ddi_dma_handle_t	bdl_dma_handle; /* for buffer descriptor list */
-	ddi_acc_handle_t	bdl_acc_handle; /* access handle of bdlist */
-	i810_bd_list_t		*bdl_virtual;	/* virtual address of BDL */
-	i810_bd_list_t		*bdl_phys;	/* Physical address of BDL */
-	size_t			bdl_size;	/* real len of BDL */
-
-	i810_sample_buf_t	play_buf;	/* buffer for playback */
-	i810_sample_buf_t	record_buf;	/* buffer for record */
-	int			play_buf_size;	/* the size of play buffer */
-	int			record_buf_size;	/* size of in buffer */
-
-	audio_info_t		i810_defaults;	/* default state for dev */
-	audio_device_t		i810_dev_info;	/* audio device info state */
-	uint16_t		vol_bits_mask;	/* bits used to ctrl volume */
-
-	kstat_t			*i810_ksp;	/* kernel statistics */
-	uint32_t		flags;		/* state flags */
-
-	uint_t			i810_psample_rate;	/* play sample rate */
-	uint_t			i810_pchannels;		/* play channels */
-	uint_t			i810_pprecision;	/* play precision */
-	uint_t			i810_csample_rate;	/* record sample rate */
-	uint_t			i810_cchannels;		/* record channels */
-	uint_t			i810_cprecision;	/* record precision */
-	uint_t			i810_output_port;	/* current out port */
-	uint_t			i810_input_port;	/* current input port */
-	uint_t			i810_monitor_gain;	/* monitor gain */
-	int			i810_csamples;	/* pcm-in samples/interrupt */
-	int			i810_psamples;	/* pcm-out samples/intr */
-
-	uint32_t		i810_res_flags;		/* resource flags */
-	};
-
-6.2 Interrupt handler
-
-	We use the BD list (buffer descriptor list) as a round-robin FIFO.
-	Both the software and hardware loop around the BD list. For playback,
-	the software writes to the buffers pointed by the BD entries of BD
-	list, and the hardware sends the data in the buffers out. For record,
-	the process is reversed. The structure i810_sample_buf (Refer to 6.1.3)
-	is defined for the sample buffer. This structure has two chunks(data
-	buffers). The software uses the head, tail and avail fields of this
-	structure to manipulate the FIFO. Head field indicates the first valid
-	BD hardware can manipulate, and tail indicates the BD after the last
-	valid BD, avail indicates how many buffers (chunks) are available.
-	There're also two hardware registers to index the FIFO, the CIV (current
-	index value) indicates the current BD the hardware is transferring, the
-	LVI (last valid index) indicates the last valid BD that contains proper
-	data, which the hardware should not pass over. Each time a BD is
-	processed, the hardware will update the CIV (plus 1), and issue an
-	interrupt if the cmd_ioc field of this processed BD entry is set. If
-	the system is busy, there can be more than one BD to be processed when
-	the OS has chance to handle the interrupt. Except for the initialization
-	at the beginning of playback/record, the driver do not modify the
-	content of CIV register.
-
-
-	CIV=3  LVI=3 (CIV is updated by hardware)
-	----------------------------------------------------------------
-	  0   1   2   3                                               31
-	----------------------------------------------------------------
-	          head=2 --> (reclaim, from head to CIV)
-	      	      tail=3 --> (fill, from tail to head)
-	
-		(a) before reclaiming
-
-
-	CIV=3  LVI=3
-	----------------------------------------------------------------
-	  0   1   2   3                                               31
-	----------------------------------------------------------------
-	              head=3 --> (reclaim, from head to CIV)
-	      	      tail=3 --> (fill, from tail to head)
-
-		 (b) after reclaiming/before filling
-
-	
-	CIV=3  LVI=5(LVI is updated by software)
-	----------------------------------------------------------------
-	  0   1   2   3   4    5                                      31
-	----------------------------------------------------------------
-	              head=3 --> (reclaim, from head to CIV)
-	      	               tail=5 --> (fill, from tail to head)
-	
-		(c) after filling
-
-		Fig 1. Interrupt handler(one pass)
-
-	 Note:
-		Here, we use only two BDs at any time, so when we "fill" BD,
-		we can't "fill" all BDs from tail to head.
-			tail - head <=2 , when tail>=head;
-			tail + 32 - head <=2, when tail<head;
-
-	Now, let us explain what the interrupt handler does. It will first
-	reclaim the BD(s) which had been transferred, which ranges from
-	"head" to (CIV - 1). This job is done in the routine
-	audio810_reclaim_play_buf(...). Then it invokes audio810_fill_play_buf
-	(...), fills audio samples into buffers pointed by the available BDs
-	beginning from "tail", updates the LVI to to the last processed BD
-	and update tail to LVI + 1, and update the avail to 0. we use 2 buffers
-	and let the 0, 2, 4, 6, ..., 28, 30 BDs pointing to the first buffer,
-	let the 1, 3, 5, ..., 29, 31 BDs pointing to the second buffer, so the
-	hardware and software will manipulate the two buffer in turn.
diff --git a/usr/src/uts/common/io/audio/sada/drv/audio810/audio810.c b/usr/src/uts/common/io/audio/sada/drv/audio810/audio810.c
deleted file mode 100644
index 815d48c2e5..0000000000
--- a/usr/src/uts/common/io/audio/sada/drv/audio810/audio810.c
+++ /dev/null
@@ -1,3490 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-
-/*
- * audio810 Audio Driver
- *
- * The driver is primarily targeted at providing audio support for
- * the W1100z and W2100z systems, which use the AMD 8111 audio core
- * and the Realtek ALC 655 codec. The ALC 655 chip supports only
- * fixed 48k sample rate. However, the audio core of AMD 8111 is
- * completely compatible to the Intel ICHx chips (Intel 8x0 chipsets),
- * so the driver can work for the ICHx. In order to support more
- * chipsets, the driver supports variable sample rates, rather than
- * fixed 48k, but it does not support the rates below 8k because some
- * codec chips do not support the sample rates in that scope. Therefore
- * the option of loading the driver in compat mode through the .conf
- * file on the W1100z and W2100z systems is not supported and the
- * "mixer-mode" property has been removed from that file.
- *
- * This driver uses the mixer Audio Personality Module to implement
- * audio(7I) and mixer(7I) semantics. Both play and record are single
- * streaming.
- *
- * The AMD 8111 audio core, as an AC'97 controller, has independent
- * channels for PCM in, PCM out, mic in, modem in, and modem out.
- * The AC'97 controller is a PCI bus master with scatter/gather
- * support. Each channel has a DMA engine. Currently, we use only
- * the PCM in and PCM out channels. Each DMA engine uses one buffer
- * descriptor list. And the buffer descriptor list is an array of up
- * to 32 entries, each of which describes a data buffer. Each entry
- * contains a pointer to a data buffer, control bits, and the length
- * of the buffer being pointed to, where the length is expressed as
- * the number of samples. This, combined with the 16-bit sample size,
- * gives the actual physical length of the buffer.
- *
- * We use the BD list (buffer descriptor list) as a round-robin FIFO.
- * Both the software and hardware loop around the BD list. For playback,
- * the software writes to the buffers pointed by the BD entries of BD
- * list, and the hardware sends the data in the buffers out. For record,
- * the process is reversed. So we define the struct, i810_sample_buf,
- * to handle BD. The software uses the head, tail and avail fields of
- * this structure to manipulate the FIFO. The head field indicates the
- * first valid BD hardware can manipulate. The tail field indicates the
- * BD after the last valid BD. And the avail field indicates how many
- * buffers are available. There're also two hardware registers to index
- * the FIFO, the CIV (current index value) indicating the current BD the
- * hardware is transferring, and the LVI (last valid index) indicating
- * the last valid BD that contains proper data which the hardware should
- * not pass over. Each time a BD is processed, the hardware will issue an
- * interrupt. If the system is busy, there can be more than one BD to be
- * processed when the OS have chance to handle the interrupt. When an
- * interrupt generated, the interrupt handler will first reclaim the BD(s)
- * which had been transferred, which will be the limit [head, CIV-1], then
- * update the value of the head field to CIV, update the value of avail to
- * CIV - head. And then it will process avail BDs from tail, and set the
- * LVI to the last processed BD and update tail to LVI + 1, and update the
- * avail to 0.
- *
- * We allocate only 2 blocks of DMA memory, say A and B, for every DMA
- * engine, and bind the first, Block A, to the even entries of BDL,
- * while bind the second to the odd entries. That's say, for each buffer
- * descriptor list of DMA engine, entry 0, 2,,, 30 would be bound to Block
- * A, and entry 1, 3,,, 31 would be bound to Block B. Take the playback as
- * an example. At the beginning of playback, we set the entry 0 and 1 to
- * point to block A and B separately, and tell the DMA engine that the last
- * valid entry is entry 1. So the DMA engine doesn't access the entries after
- * entry 1. When the first playback interrupt generated, we reclaim entry
- * 0, and fill the BD entry 2 with the address of Block A, then set the
- * entry 2 to be the last valid entry, and so on. So at any time there are
- * at most two entries available for per DMA engine.
- *
- * Every time we program AC97 codec, we save the value in codec_shadow[].
- * This means that register state information is saved for power management
- * shutdown (we'll support this later). When the codec is started back up
- * we use this saved state to restore codec's state in audio810_chip_init()
- *
- * A workaround for the AD1980 and AD1985 codec:
- *	Most vendors connect the surr-out of the codecs to the line-out jack.
- *	So far we haven't found which vendors don't do that. So we assume that
- *	all vendors swap the surr-out and the line-out outputs. So we need swap
- *	the two outputs. But we still internally process the
- *	"ad198x-swap-output" property. If someday some vendors do not swap the
- *	outputs, we would set "ad198x-swap-output = 0" in the
- *	/kernel/drv/audio810.conf file, and unload and reload the audio810
- *	driver (or reboot).
- *
- * System power management is not yet supported by the driver.
- *
- * 	NOTE:
- * 	This driver depends on the misc/audiosup, misc/amsrc2 and
- * 	misc/mixer modules being loaded first.
- */
-#include <sys/types.h>
-#include <sys/modctl.h>
-#include <sys/kmem.h>
-#include <sys/conf.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/pci.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/ac97.h>
-#include <sys/audio/impl/audio810_impl.h>
-#include <sys/audio/audio810.h>
-
-/*
- * Module linkage routines for the kernel
- */
-static int audio810_getinfo(dev_info_t *, ddi_info_cmd_t, void*, void**);
-static int audio810_attach(dev_info_t *, ddi_attach_cmd_t);
-static int audio810_detach(dev_info_t *, ddi_detach_cmd_t);
-
-/*
- * Entry point routine prototypes
- */
-static int audio810_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int audio810_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int audio810_ad_start_play(audiohdl_t, int);
-static void audio810_ad_pause_play(audiohdl_t, int);
-static void audio810_ad_stop_play(audiohdl_t, int);
-static int audio810_ad_start_record(audiohdl_t, int);
-static void audio810_ad_stop_record(audiohdl_t, int);
-
-/*
- * interrupt handler
- */
-static uint_t	audio810_intr(caddr_t);
-
-/*
- * Local Routine Prototypes
- */
-static void audio810_set_busy(audio810_state_t *);
-static void audio810_set_idle(audio810_state_t *);
-static int audio810_codec_sync(audio810_state_t *);
-static int audio810_write_ac97(audio810_state_t *, int, uint16_t);
-static int audio810_read_ac97(audio810_state_t *, int, uint16_t *);
-static int audio810_and_ac97(audio810_state_t *, int, uint16_t);
-static int audio810_or_ac97(audio810_state_t *, int, uint16_t);
-static int audio810_reset_ac97(audio810_state_t *);
-static int audio810_init_state(audio810_state_t *, dev_info_t *);
-static int audio810_map_regs(dev_info_t *, audio810_state_t *);
-static void audio810_unmap_regs(audio810_state_t *);
-static int audio810_alloc_sample_buf(audio810_state_t *, int, int);
-static void audio810_free_sample_buf(audio810_state_t *, i810_sample_buf_t *);
-static void audio810_stop_dma(audio810_state_t *);
-static int audio810_chip_init(audio810_state_t *, int);
-static int audio810_fill_play_buf(audio810_state_t *);
-static int audio810_prepare_record_buf(audio810_state_t *);
-static void audio810_reclaim_play_buf(audio810_state_t *);
-static void audio810_reclaim_record_buf(audio810_state_t *);
-static int audio810_set_gain(audio810_state_t *, int, int, int);
-static int audio810_set_port(audio810_state_t *, int, int);
-static int audio810_set_monitor_gain(audio810_state_t *, int);
-
-/*
- * Global variables, but used only by this file.
- */
-
-/* anchor for soft state structures */
-static void	*audio810_statep;
-
-/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
-static char	*audio810_name = I810_NAME;
-
-
-/*
- * STREAMS structures
- */
-
-/* STREAMS driver id and limit value struct */
-static struct module_info audio810_modinfo = {
-	I810_IDNUM,		/* module ID number */
-	I810_NAME,		/* module name */
-	I810_MINPACKET,		/* minimum packet size */
-	I810_MAXPACKET,		/* maximum packet size */
-	I810_HIWATER,		/* high water mark */
-	I810_LOWATER,		/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit audio810_rqueue = {
-	audio_sup_rput,		/* put procedure */
-	audio_sup_rsvc,		/* service procedure */
-	audio_sup_open,		/* open procedure */
-	audio_sup_close,	/* close procedure */
-	NULL,			/* unused */
-	&audio810_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit audio810_wqueue = {
-	audio_sup_wput,		/* write procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&audio810_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab audio810_str_info = {
-	&audio810_rqueue,	/* read queue */
-	&audio810_wqueue,	/* write queue */
-	NULL,			/* mux lower read queue */
-	NULL,			/* mux lower write queue */
-};
-
-/*
- * DDI Structures
- */
-
-/* Entry points structure */
-static struct cb_ops audio810_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&audio810_str_info,	/* cb_str */
-	D_NEW | D_MP | D_64BIT, /* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_awrite */
-};
-
-/* Device operations structure */
-static struct dev_ops audio810_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	audio810_getinfo,	/* devo_getinfo */
-	nulldev,		/* devo_identify - obsolete */
-	nulldev,		/* devo_probe */
-	audio810_attach,	/* devo_attach */
-	audio810_detach,	/* devo_detach */
-	nodev,			/* devo_reset */
-	&audio810_cb_ops,	/* devi_cb_ops */
-	NULL,			/* devo_bus_ops */
-	NULL,			/* devo_power */
-	ddi_quiesce_not_supported,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv audio810_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	I810_MOD_NAME,		/* drv_linkinfo */
-	&audio810_dev_ops,	/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage audio810_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&audio810_modldrv,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-static uint_t audio810_mixer_srs[] = {
-	I810_SAMPR5510, I810_SAMPR48000, 0
-};
-
-static uint_t audio810_min_compat_srs[] = {
-	I810_SAMPR48000, 0
-};
-
-static uint_t audio810_compat_srs [] = {
-	I810_SAMPR8000, I810_SAMPR9600, I810_SAMPR11025,
-	I810_SAMPR16000, I810_SAMPR18900, I810_SAMPR22050,
-	I810_SAMPR27420, I810_SAMPR32000, I810_SAMPR33075,
-	I810_SAMPR37800, I810_SAMPR44100, I810_SAMPR48000,
-	0
-};
-
-static am_ad_sample_rates_t audio810_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	audio810_mixer_srs
-};
-
-static am_ad_sample_rates_t audio810_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audio810_compat_srs
-};
-
-/* Some codec, such as the ALC 655, only support 48K sample rate */
-static am_ad_sample_rates_t audio810_min_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audio810_min_compat_srs
-};
-
-/* now, only support stereo */
-static uint_t audio810_channels[] = {
-	AUDIO_CHANNELS_STEREO,
-	0
-};
-
-
-static am_ad_cap_comb_t audio810_combinations[] = {
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
-
-/*
- * device access attributes for register mapping
- */
-static struct ddi_device_acc_attr dev_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_LE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-/*
- * DMA attributes of buffer descriptor list
- */
-static ddi_dma_attr_t	bdlist_dma_attr = {
-	DMA_ATTR_V0,	/* version */
-	0,		/* addr_lo */
-	0xffffffff,	/* addr_hi */
-	0x0000ffff,	/* count_max */
-	8,		/* align, BDL must be aligned on a 8-byte boundary */
-	0x3c,		/* burstsize */
-	8,		/* minxfer, set to the size of a BDlist entry */
-	0x0000ffff,	/* maxxfer */
-	0x00000fff,	/* seg, set to the RAM pagesize of intel platform */
-	1,		/* sgllen, there's no scatter-gather list */
-	8,		/* granular, set to the value of minxfer */
-	0		/* flags, use virtual address */
-};
-
-/*
- * DMA attributes of buffers to be used to receive/send audio data
- */
-static ddi_dma_attr_t	sample_buf_dma_attr = {
-	DMA_ATTR_V0,
-	0,		/* addr_lo */
-	0xffffffff,	/* addr_hi */
-	0x0001fffe,	/* count_max */
-	2,		/* align, data buffer is aligned on a 2-byte boundary */
-	0x3c,		/* burstsize */
-	4,		/* minxfer, set to the size of a sample data */
-	0x0001ffff,	/* maxxfer */
-	0x0001ffff,	/* seg */
-	1,		/* sgllen, no scatter-gather */
-	4,		/* granular, set to the value of minxfer */
-	0,		/* flags, use virtual address */
-};
-
-static am_ad_entry_t audio810_entry = {
-	NULL,				/* ad_setup() */
-	NULL,				/* ad_teardown() */
-	audio810_ad_set_config,		/* ad_set_config() */
-	audio810_ad_set_format,		/* ad_set_format() */
-	audio810_ad_start_play,		/* ad_start_play() */
-	audio810_ad_pause_play,		/* ad_pause_play() */
-	audio810_ad_stop_play,		/* ad_stop_play() */
-	audio810_ad_start_record,	/* ad_start_record() */
-	audio810_ad_stop_record,	/* ad_stop_record() */
-	NULL,				/* ad_ioctl() */
-	NULL				/* ad_iocdata() */
-};
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *	This is how access is initially given to all the static structures.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-	ATRACE("in audio810 _init()", NULL);
-
-	if ((error = ddi_soft_state_init(&audio810_statep,
-	    sizeof (audio810_state_t), 1)) != 0) {
-		ATRACE("i810 ddi_soft_state_init() failed", audio810_statep);
-		return (error);
-	}
-
-	if ((error = mod_install(&audio810_modlinkage)) != 0) {
-		ddi_soft_state_fini(&audio810_statep);
-	}
-
-	ATRACE("audio810 _init() audio810_statep", audio810_statep);
-	ATRACE("audio810 _init() returning", error);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description:
- *	Module de-initialization, called when the driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int		error;
-
-	ATRACE("in audio810 _fini()", audio810_statep);
-
-	if ((error = mod_remove(&audio810_modlinkage)) != 0) {
-		return (error);
-	}
-
-	ddi_soft_state_fini(&audio810_statep);
-
-	ATRACE_32("audio810 _fini() returning", error);
-
-	return (0);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo		*modinfop	Pointer to the opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int	error;
-
-	ATRACE("in audio810 _info()", NULL);
-
-	error = mod_info(&audio810_modlinkage, modinfop);
-
-	ATRACE_32("audio810 _info() returning", error);
-
-	return (error);
-
-}	/* _info() */
-
-
-/* ******************* Driver Entry Points ********************************* */
-
-/*
- * audio810_getinfo()
- */
-static int
-audio810_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd, void *arg, void **result)
-{
-	audio810_state_t	*state;
-	int 			instance;
-	int 			error;
-
-	error = DDI_FAILURE;
-	ATRACE("in audio810_getinfo()", dip);
-
-	switch (cmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((state = ddi_get_soft_state(audio810_statep,
-		    instance)) != NULL) {
-			*result = state->dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
-		break;
-
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void*)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-
-	default:
-		break;
-	}
-
-	return (error);
-
-}	/* audio810_getinfo() */
-
-/*
- * audio810_attach()
- *
- * Description:
- *	Attach an instance of the audio810 driver. This routine does the
- * 	device dependent attach tasks. When it is completed, it calls
- *	audio_sup_register() and am_attach() so they may do their work.
- *
- *	NOTE: mutex_init() no longer needs a name string, so set
- *		to NULL to save kernel space.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd	Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-static int
-audio810_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	int 			instance;
-	uint16_t		cmdreg;
-	audio810_state_t	*statep;
-	audio_sup_reg_data_t	data;
-
-	ATRACE("in audio810_attach()", dip);
-
-	instance = ddi_get_instance(dip);
-
-	ATRACE("audio810_attach() audio810_statep", audio810_statep);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		break;
-	case DDI_RESUME:
-		ATRACE("I810_attach() DDI_RESUME", NULL);
-
-		if ((statep = ddi_get_soft_state(audio810_statep, instance)) ==
-		    NULL) {
-			audio_sup_log(NULL, CE_WARN,
-			    "!attach() DDI_RESUME get soft state failed");
-			return (DDI_FAILURE);
-		}
-
-		ASSERT(dip == statep->dip);
-
-		mutex_enter(&statep->inst_lock);
-
-		ASSERT(statep->i810_suspended == I810_SUSPENDED);
-
-		statep->i810_suspended = I810_NOT_SUSPENDED;
-
-		/* Restore the audio810 chip's state */
-		if (audio810_chip_init(statep, I810_INIT_RESTORE) !=
-		    AUDIO_SUCCESS) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!attach() DDI_RESUME failed to init chip");
-			mutex_exit(&statep->inst_lock);
-			return (DDI_FAILURE);
-		}
-
-		mutex_exit(&statep->inst_lock);
-
-		/* Resume playing and recording, if required */
-		if (audio_sup_restore_state(statep->audio_handle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!attach() DDI_RESUME audio restart failed");
-		}
-
-		mutex_enter(&statep->inst_lock);
-		cv_broadcast(&statep->i810_cv);	/* let entry points continue */
-		mutex_exit(&statep->inst_lock);
-
-		ATRACE("audio810_attach() DDI_RESUME done", NULL);
-
-		return (DDI_SUCCESS);
-	default:
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() unknown command: 0x%x",
-		    audio810_name, instance, cmd);
-		return (DDI_FAILURE);
-	}
-
-	/* we don't support high level interrupts in the driver */
-	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() unsupported high level interrupt",
-		    audio810_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	/* allocate the soft state structure */
-	if (ddi_soft_state_zalloc(audio810_statep, instance) != DDI_SUCCESS) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() soft state allocate failed",
-		    audio810_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	if ((statep = ddi_get_soft_state(audio810_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() soft state failed",
-		    audio810_name, instance);
-		goto error_state;
-	}
-
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-	if ((statep->audio_handle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() audio_sup_register() failed",
-		    audio810_name, instance);
-		goto error_state;
-	}
-
-	/* save private state */
-	audio_sup_set_private(statep->audio_handle, statep);
-
-	if ((audio810_init_state(statep, dip)) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() init state structure failed");
-		goto error_audiosup;
-	}
-
-	/* map in the registers, allocate DMA buffers, etc. */
-	if (audio810_map_regs(dip, statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() couldn't map registers");
-		goto error_destroy;
-	}
-
-	/* set PCI command register */
-	cmdreg = pci_config_get16(statep->pci_conf_handle, PCI_CONF_COMM);
-	pci_config_put16(statep->pci_conf_handle, PCI_CONF_COMM,
-	    cmdreg | PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
-
-	if ((audio810_alloc_sample_buf(statep, I810_DMA_PCM_OUT,
-	    statep->play_buf_size) == AUDIO_FAILURE) ||
-	    (audio810_alloc_sample_buf(statep, I810_DMA_PCM_IN,
-	    statep->record_buf_size) == AUDIO_FAILURE)) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() couldn't allocate sample buffers");
-		goto error_unmap;
-	}
-
-	/* initialize audio controller and AC97 codec */
-	if (audio810_chip_init(statep, I810_INIT_NO_RESTORE) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() failed to init chip");
-		goto error_dealloc;
-	}
-
-	/* call the mixer attach() routine */
-	if (am_attach(statep->audio_handle, cmd, &statep->ad_info) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() am_attach() failed");
-		goto error_dealloc;
-	}
-
-	/* set up kernel statistics */
-	if ((statep->i810_ksp = kstat_create(I810_NAME, instance,
-	    I810_NAME, "controller", KSTAT_TYPE_INTR, 1,
-	    KSTAT_FLAG_PERSISTENT)) != NULL) {
-		kstat_install(statep->i810_ksp);
-	}
-
-	/* set up the interrupt handler */
-	if (ddi_add_intr(dip, 0, &statep->intr_iblock,
-	    (ddi_idevice_cookie_t *)NULL, audio810_intr, (caddr_t)statep) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!attach() bad interrupt specification ");
-		goto error_kstat;
-	}
-	ddi_report_dev(dip);
-
-	return (DDI_SUCCESS);
-
-error_kstat:
-	if (statep->i810_ksp) {
-		kstat_delete(statep->i810_ksp);
-	}
-	(void) am_detach(statep->audio_handle, DDI_DETACH);
-
-error_dealloc:
-	audio810_free_sample_buf(statep, &statep->play_buf);
-	audio810_free_sample_buf(statep, &statep->record_buf);
-
-error_unmap:
-	audio810_unmap_regs(statep);
-
-error_destroy:
-	ATRACE("audio810_attach() error_destroy", statep);
-	mutex_destroy(&statep->inst_lock);
-	cv_destroy(&statep->i810_cv);
-
-error_audiosup:
-	ATRACE("audio810_attach() error_audiosup", statep);
-	(void) audio_sup_unregister(statep->audio_handle);
-
-error_state:
-	ATRACE("audio810_attach() error_state", statep);
-	ddi_soft_state_free(audio810_statep, instance);
-
-	ATRACE("audio810_attach() returning failure", NULL);
-
-	return (DDI_FAILURE);
-
-}	/* audio810_attach() */
-
-/*
- * audio810_detach()
- *
- * Description:
- *	Detach an instance of the audio810 driver. After the Codec is detached
- *	we call am_detach() and audio_sup_register() so they may do their work.
- *
- * Arguments:
- *	dev_info_t		*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t	cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS	The driver was detached
- *	DDI_FAILURE	The driver couldn't be detached
- */
-static int
-audio810_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
-{
-	audio810_state_t	*statep;
-	int			instance;
-
-	instance = ddi_get_instance(dip);
-
-	ATRACE_32("audio810_detach() instance", instance);
-	ATRACE("audio810_detach() audio810_statep", audio810_statep);
-
-	if ((statep = ddi_get_soft_state(audio810_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: detach() get soft state failed",
-		    audio810_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-	case DDI_SUSPEND:
-		ATRACE("i810_detach() SUSPEND", statep);
-
-		mutex_enter(&statep->inst_lock);
-
-		ASSERT(statep->i810_suspended == I810_NOT_SUSPENDED);
-
-		statep->i810_suspended = I810_SUSPENDED; /* stop new ops */
-
-		/* wait for current operations to complete */
-		while (statep->i810_busy_cnt != 0)
-			cv_wait(&statep->i810_cv, &statep->inst_lock);
-
-		/* stop DMA engines */
-		audio810_stop_dma(statep);
-
-		if (audio_sup_save_state(statep->audio_handle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!detach() DDI_SUSPEND audio save failed");
-		}
-
-		mutex_exit(&statep->inst_lock);
-
-		ATRACE("audio810_detach() DDI_SUSPEND successful", statep);
-
-		return (DDI_SUCCESS);
-	default:
-		ATRACE("i810_detach() unknown command", cmd);
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!detach() unknown command: 0x%x", cmd);
-		return (DDI_FAILURE);
-	}
-
-	/* stop DMA engines */
-	mutex_enter(&statep->inst_lock);
-	audio810_stop_dma(statep);
-	mutex_exit(&statep->inst_lock);
-
-	/* remove the interrupt handler */
-	ddi_remove_intr(dip, 0, statep->intr_iblock);
-
-	/* free DMA memory */
-	audio810_free_sample_buf(statep, &statep->play_buf);
-	audio810_free_sample_buf(statep, &statep->record_buf);
-
-	/* free the kernel statistics structure */
-	if (statep->i810_ksp) {
-		kstat_delete(statep->i810_ksp);
-	}
-
-	/* detach audio mixer */
-	(void) am_detach(statep->audio_handle, cmd);
-
-	/*
-	 * call the audio support module's detach routine to remove this
-	 * driver completely from the audio driver architecture.
-	 */
-	(void) audio_sup_unregister(statep->audio_handle);
-
-	mutex_destroy(&statep->inst_lock);
-	cv_destroy(&statep->i810_cv);
-
-	audio810_unmap_regs(statep);
-
-	ddi_soft_state_free(audio810_statep, instance);
-
-	return (DDI_SUCCESS);
-
-}	/* audio810_detach */
-
-/*
- * audio810_intr()
- *
- * Description:
- *	Interrupt service routine for both play and record. For play we
- *	get the next buffers worth of audio. For record we send it on to
- *	the mixer.
- *
- *	Each of buffer descriptor has a field IOC(interrupt on completion)
- *	When both this and the IOC bit of correspondent dma control register
- *	is set, it means that the controller should issue an interrupt upon
- *	completion of this buffer.
- *	(AMD 8111 hypertransport I/O hub data sheet. 3.8.3 page 71)
- *
- * Arguments:
- *	caddr_t		arg	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-audio810_intr(caddr_t arg)
-{
-	audio810_state_t	*statep;
-	uint16_t		gsr;
-
-	statep = (audio810_state_t *)arg;
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->i810_suspended == I810_SUSPENDED) {
-		ATRACE("audio810_intr() device suspended", NULL);
-		mutex_exit(&statep->inst_lock);
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	gsr = I810_BM_GET32(I810_REG_GSR);
-
-	/* check if device is interrupting */
-	if ((gsr & I810_GSR_USE_INTR) == 0) {
-		ATRACE_32("audio810_intr() not our interrupt", gsr);
-		mutex_exit(&statep->inst_lock);
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	/* PCM in interrupt */
-	if (gsr & I810_GSR_INTR_PIN) {
-		I810_BM_PUT8(I810_PCM_IN_SR,
-		    I810_BM_SR_LVBCI |
-		    I810_BM_SR_BCIS |
-		    I810_BM_SR_FIFOE);
-
-		if (statep->flags & I810_DMA_RECD_STARTED) {
-			audio810_reclaim_record_buf(statep);
-			(void) audio810_prepare_record_buf(statep);
-		}
-	}
-
-	/* PCM out interrupt */
-	if (gsr & I810_GSR_INTR_POUT) {
-		I810_BM_PUT8(I810_PCM_OUT_SR,
-		    I810_BM_SR_LVBCI |
-		    I810_BM_SR_BCIS |
-		    I810_BM_SR_FIFOE);
-
-		if (statep->flags & I810_DMA_PLAY_STARTED) {
-			audio810_reclaim_play_buf(statep);
-			(void) audio810_fill_play_buf(statep);
-		}
-	}
-
-	/* update the kernel interrupt statistics */
-	if (statep->i810_ksp) {
-		I810_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
-	}
-
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE("audio810_intr() done", statep);
-
-	return (DDI_INTR_CLAIMED);
-
-}	/* audio810_intr() */
-
-/*
- * audio810_set_busy()
- *
- * Description:
- *	This routine is called whenever a routine needs to guarantee
- *	that it will not be suspended.  It will also block any routine
- *	while a suspend is going on.
- *
- *	CAUTION: This routine cannot be called by routines that will
- *		block. Otherwise DDI_SUSPEND will be blocked for a
- *		long time. And that is the wrong thing to do.
- *
- * Arguments:
- *	audio810_state_t	*statep		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_set_busy(audio810_state_t *statep)
-{
-	ATRACE("in audio810_set_busy()", statep);
-
-	ASSERT(!mutex_owned(&statep->inst_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&statep->inst_lock);
-
-	/* block if we are suspended */
-	while (statep->i810_suspended == I810_SUSPENDED) {
-		cv_wait(&statep->i810_cv, &statep->inst_lock);
-	}
-
-	/*
-	 * Okay, we aren't suspended, so mark as busy.
-	 * This will keep us from being suspended when we release the lock.
-	 */
-	ASSERT(statep->i810_busy_cnt >= 0);
-	statep->i810_busy_cnt++;
-
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE("audio810_set_busy() done", statep);
-
-	ASSERT(!mutex_owned(&statep->inst_lock));
-
-}	/* audio810_set_busy() */
-
-/*
- * audio810_set_idle()
- *
- * Description:
- *	This routine reduces the busy count. It then does a cv_broadcast()
- *	if the count is 0 so a waiting DDI_SUSPEND will continue forward.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_set_idle(audio810_state_t *statep)
-{
-	ATRACE("in audio810_set_idle()", statep);
-
-	ASSERT(!mutex_owned(&statep->inst_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&statep->inst_lock);
-
-	ASSERT(statep->i810_suspended == I810_NOT_SUSPENDED);
-
-	/* decrement the busy count */
-	ASSERT(statep->i810_busy_cnt > 0);
-	statep->i810_busy_cnt--;
-
-	/* if no longer busy, then we wake up a waiting SUSPEND */
-	if (statep->i810_busy_cnt == 0) {
-		cv_broadcast(&statep->i810_cv);
-	}
-
-	/* we're done, so unlock */
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE("audio810_set_idle() done", statep);
-
-	ASSERT(!mutex_owned(&statep->inst_lock));
-
-}	/* audio810_set_idle() */
-
-/* *********************** Mixer Entry Point Routines ******************* */
-/*
- * audio810_ad_set_config()
- *
- * Description:
- *	This routine is used to set new Codec parameters, except the data
- *	format which has it's own routine. If the Codec doesn't support a
- *	particular parameter and it is asked to set it then we return
- *	AUDIO_FAILURE.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		command		The configuration to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		Argument #1
- *	int		arg2		Argument #2, not always needed
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set,
- *				or the parameter couldn't be set
- */
-static int
-audio810_ad_set_config(audiohdl_t ahandle, int stream, int command,
-    int dir, int arg1, int arg2)
-{
-	audio810_state_t	*statep;
-	int 			rc = AUDIO_SUCCESS;
-
-	ATRACE_32("i810_ad_set_config() stream", stream);
-	ATRACE_32("i810_ad_set_config() command", command);
-	ATRACE_32("i810_ad_set_config() dir", dir);
-	ATRACE_32("i810_ad_set_config() arg1", arg1);
-	ATRACE_32("i810_ad_set_config() arg2", arg2);
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, of balance on gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	arg1 --> gain
-		 * 	arg2 --> channel #, 0 == left, 1 == right
-		 */
-		rc = audio810_set_gain(statep, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of in ports, as well as which ports
-		 * are modifiable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 * 	arg1 --> port bit pattern
-		 * 	arg2 --> not used
-		 */
-		rc = audio810_set_port(statep, dir, arg1);
-		break;
-
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	dir ---> N/A
-		 *	arg1 --> gain
-		 * 	arg2 --> not used
-		 */
-		rc = audio810_set_monitor_gain(statep, arg1);
-		break;
-
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == mute, 0 == enable
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* mute */
-			(void) audio810_or_ac97(statep,
-			    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-			(void) audio810_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, HPVR_MUTE);
-			(void) audio810_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MMVR_MUTE);
-		} else {	/* not muted */
-
-			/* by setting the port we unmute only active ports */
-			(void) audio810_set_port(statep,
-			    AUDIO_PLAY, statep->i810_output_port);
-		}
-		break;
-
-	case AM_MIC_BOOST:
-		/*
-		 * Enable or disable the mic's 20 dB boost preamplifier.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == enable, 0 == disabled
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* enable */
-			(void) audio810_or_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
-			statep->ad_info.ad_add_mode |= AM_ADD_MODE_MIC_BOOST;
-		} else {	/* disable */
-			(void) audio810_and_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER,
-			    (uint16_t)~MICVR_20dB_BOOST);
-			statep->ad_info.ad_add_mode &=
-			    ~AM_ADD_MODE_MIC_BOOST;
-		}
-		break;
-
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 *
-		 * AM_SET_GAIN_BAL
-		 * AM_SET_MONO_MIC
-		 * AM_BASS_BOOST
-		 * AM_MID_BOOST
-		 * AM_TREBLE_BOOST
-		 * AM_LOUDNESS
-		 */
-		rc = AUDIO_FAILURE;
-		ATRACE_32("i810_ad_set_config() unsupported command",
-		    command);
-		break;
-	}
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-	ATRACE_32("i810_ad_set_config() returning", rc);
-
-	return (rc);
-
-}	/* audio810_ad_set_config() */
-
-/*
- * audio810_ad_set_format()
- *
- * Description:
- *	This routine is used to set a new audio control data format.
- *	We only support 16 bit signed linear.
- *
- * Arguments:
- * 	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		sample_rate	Data sample rate
- *	int		channels	Number of channels, 1 or 2
- *	int		precision	Bits per sample, 16
- *	int		encoding	Encoding method, linear
- *
- * Returns:
- *	AUDIO_SUCCESS	The Codec data format has been set
- *	AUDIO_FAILURE	The Codec data format has not been set, or the
- *			data format couldn't be set
- */
-static int
-audio810_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-    int sample_rate, int channels, int precision, int encoding)
-{
-	audio810_state_t	*statep;
-	uint16_t		val;
-	int			rc = AUDIO_FAILURE;
-
-	ASSERT(precision == AUDIO_PRECISION_16);
-	ASSERT(channels == AUDIO_CHANNELS_STEREO);
-
-	ATRACE_32("i810_ad_set_format() stream", stream);
-	ATRACE_32("i810_ad_set_format() dir", dir);
-	ATRACE_32("i810_ad_set_format() sample_rate", sample_rate);
-	ATRACE_32("i810_ad_set_format() channels", channels);
-	ATRACE_32("i810_ad_set_format() precision", precision);
-	ATRACE_32("i810_ad_set_format() encoding", encoding);
-
-	if (encoding != AUDIO_ENCODING_LINEAR) {
-		ATRACE("i810_ad_set_format() bad encoding", encoding);
-		return (AUDIO_FAILURE);
-	}
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->var_sr == B_FALSE) {
-		/* codec doesn't support variable sample rate */
-
-		if (sample_rate != I810_SAMPR48000) {
-			audio_sup_log(statep->audio_handle, CE_NOTE,
-			    "!ad_set_format() bad sample rate %d\n",
-			    sample_rate);
-			goto done;
-		}
-	} else {
-		switch (sample_rate) {
-		case I810_SAMPR8000:	break;
-		case I810_SAMPR9600:	break;
-		case I810_SAMPR11025:	break;
-		case I810_SAMPR16000:	break;
-		case I810_SAMPR18900:	break;
-		case I810_SAMPR22050:	break;
-		case I810_SAMPR27420:	break;
-		case I810_SAMPR32000:	break;
-		case I810_SAMPR33075:	break;
-		case I810_SAMPR37800:	break;
-		case I810_SAMPR44100:	break;
-		case I810_SAMPR48000:	break;
-		default:
-			ATRACE_32("i810_ad_set_format() bad SR", sample_rate);
-			goto done;
-		}
-	}
-
-	if (dir == AUDIO_PLAY) {
-
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, sample_rate);
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_SURROUND_DAC_RATE_REGISTER, sample_rate);
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_LFE_DAC_RATE_REGISTER, sample_rate);
-
-		/*
-		 * Some codecs before ac97 2.2, such as YMF753 produced by
-		 * Yamaha LSI, don't have the AC'97 registers indexed range
-		 * from 0x2c to 0x34. So we assume this kind of codec
-		 * supports fixed 48k sample rate.
-		 */
-		if (statep->var_sr == B_TRUE) {
-			(void) audio810_read_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, &val);
-			if (val != sample_rate) {
-				ATRACE_32("ad_set_format() bad out SR",
-				    sample_rate);
-				audio_sup_log(statep->audio_handle, CE_NOTE,
-				    "!set_format() bad output sample rate %d",
-				    sample_rate);
-				goto done;
-			}
-		}
-
-		statep->i810_psample_rate = sample_rate;
-		statep->i810_pchannels = channels;
-		statep->i810_pprecision = precision;
-	} else {
-
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_LR_DAC_RATE_REGISTER, sample_rate);
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_MIC_ADC_RATE_REGISTER, sample_rate);
-
-		/*
-		 * Some codecs before ac97 2.2, such as YMF753 produced by
-		 * Yamaha LSI, don't have the AC'97 registers indexed range
-		 * from 0x2c to 0x34. So we assume this kind of codec
-		 * supports fixed 48k sample rate.
-		 */
-		if (statep->var_sr == B_TRUE) {
-			(void) audio810_read_ac97(statep,
-			    AC97_EXTENDED_LR_DAC_RATE_REGISTER, &val);
-			if (val != sample_rate) {
-				ATRACE_32("ad_set_format() bad input SR",
-				    sample_rate);
-				audio_sup_log(statep->audio_handle, CE_NOTE,
-				    "!set_format() bad input sample rate %d",
-				    sample_rate);
-				goto done;
-			}
-		}
-
-		statep->i810_csample_rate = sample_rate;
-		statep->i810_cchannels = channels;
-		statep->i810_cprecision = precision;
-	}
-
-	rc = AUDIO_SUCCESS;
-done:
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-	ATRACE_32("i810_ad_set_format() returning", rc);
-
-	return (rc);
-
-}	/* audio810_ad_set_format() */
-
-/*
- * audio810_ad_start_play()
- *
- * Description:
- *	This routine starts the playback DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Play not started/restarted, no audio to play
- */
-static int
-audio810_ad_start_play(audiohdl_t ahandle, int stream)
-{
-	audio810_state_t	*statep;
-	uint8_t			cr;
-	int			rc = AUDIO_SUCCESS;
-
-	ATRACE_32("i810_ad_start_play() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->flags & I810_DMA_PLAY_PAUSED) {
-		statep->flags |= I810_DMA_PLAY_STARTED;
-		statep->flags &= ~I810_DMA_PLAY_PAUSED;
-		cr = I810_BM_GET8(I810_PCM_OUT_CR);
-		cr |= I810_BM_CR_RUN;
-		I810_BM_PUT8(I810_PCM_OUT_CR, cr);
-		goto done;
-	}
-
-	if (statep->flags & I810_DMA_PLAY_STARTED) {
-		goto done;
-	}
-
-	rc = audio810_fill_play_buf(statep);
-	if (rc == AUDIO_FAILURE) {
-		statep->flags &= ~I810_DMA_PLAY_STARTED;
-	} else {
-		statep->flags |= I810_DMA_PLAY_STARTED;
-	}
-
-done:
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-	return (rc);
-
-}	/* audio810_ad_start_play() */
-
-/*
- * audio810_ad_pause_play()
- *
- * Description:
- *	This routine pauses the play DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- * 	void
- */
-static void
-audio810_ad_pause_play(audiohdl_t ahandle, int stream)
-{
-	audio810_state_t	*statep;
-	uint8_t			cr;
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-	ATRACE("audio810_ad_pause_play() ", ahandle);
-	ATRACE_32("i810_ad_pause_play() stream", stream);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-	if ((statep->flags & I810_DMA_PLAY_STARTED) == 0)
-		goto done;
-	cr = I810_BM_GET8(I810_PCM_OUT_CR);
-	cr &= ~I810_BM_CR_RUN;
-	I810_BM_PUT8(I810_PCM_OUT_CR, cr);
-	statep->flags |= I810_DMA_PLAY_PAUSED;
-done:
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-}	/* audio810_ad_pause_play() */
-
-/*
- * audio810_ad_stop_play()
- *
- * Description:
- *	This routine stops the playback DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- *	void
- */
-static void
-audio810_ad_stop_play(audiohdl_t ahandle, int stream)
-{
-	audio810_state_t	*statep;
-	i810_sample_buf_t	*buf;
-
-	ATRACE("audio810_ad_stop_play() ", ahandle);
-	ATRACE_32("i810_ad_stop_play() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	/* pause bus master */
-	I810_BM_PUT8(I810_PCM_OUT_CR, I810_BM_CR_PAUSE);
-
-	/* reset registers */
-	I810_BM_PUT8(I810_PCM_OUT_CR, I810_BM_CR_RST);
-
-	buf = &statep->play_buf;
-	buf->io_started = B_FALSE;
-	statep->flags &= ~(I810_DMA_PLAY_STARTED
-	    |I810_DMA_PLAY_PAUSED | I810_DMA_PLAY_EMPTY);
-
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-}	/* audio810_ad_stop_play() */
-
-/*
- * audio810_ad_start_record()
- *
- * Description:
- *	This routine starts the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which isn't going to apply for record
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Record not started
- */
-static int
-audio810_ad_start_record(audiohdl_t ahandle, int stream)
-{
-	audio810_state_t	*statep;
-	int			rc = AUDIO_SUCCESS;
-
-	ATRACE("audio810_ad_start_record() ", ahandle);
-	ATRACE_32("i810_ad_start_record() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-	if (statep->flags & I810_DMA_RECD_STARTED)
-		goto done;
-
-	rc = audio810_prepare_record_buf(statep);
-	if (rc == AUDIO_SUCCESS) {
-		statep->flags |= I810_DMA_RECD_STARTED;
-	}
-done:
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-	return (rc);
-
-}	/* audio810_ad_start_record() */
-
-/*
- * audio810_ad_stop_record()
- *
- * Description:
- *	This routine stops the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream
- *					Codecs, which isn't going to apply
- *					for record
- *
- * Returns:
- *	void
- */
-static void
-audio810_ad_stop_record(audiohdl_t ahandle, int stream)
-{
-	audio810_state_t	*statep;
-	i810_sample_buf_t	*buf;
-
-	ATRACE("audio810_ad_stop_record() ", ahandle);
-	ATRACE_32("i810_ad_stop_record() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	audio810_set_busy(statep);
-
-	mutex_enter(&statep->inst_lock);
-	statep->flags &= ~I810_DMA_RECD_STARTED;
-
-	buf = &statep->record_buf;
-	buf->io_started = B_FALSE;
-
-	/* pause bus master */
-	I810_BM_PUT8(I810_PCM_IN_CR, I810_BM_CR_PAUSE);
-
-	/* reset registers */
-	I810_BM_PUT8(I810_PCM_IN_CR, I810_BM_CR_RST);
-
-	mutex_exit(&statep->inst_lock);
-
-	audio810_set_idle(statep);
-
-}	/* audio810_ad_stop_record() */
-
-/* *********************** Local Routines *************************** */
-
-/*
- * audio810_init_state()
- *
- * Description:
- *	This routine initializes the audio driver's state structure
- *
- *	CAUTION: This routine cannot allocate resources, unless it frees
- *		them before returning for an error. Also, error_destroy:
- *		in audio810_attach() would need to be fixed as well.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	dev_info_t		*dip		Pointer to the device's
- *						dev_info struct
- *
- * Returns:
- *	AUDIO_SUCCESS		State structure initialized
- *	AUDIO_FAILURE		State structure not initialized
- */
-static int
-audio810_init_state(audio810_state_t *statep, dev_info_t *dip)
-{
-	int rints;
-	int pints;
-	int cdrom;
-	int mode;
-
-	ATRACE("audio810_init_state()", NULL);
-
-	statep->dip = dip;
-	statep->vol_bits_mask = 5;
-
-	cdrom = ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "cdrom", 0);
-
-	/* get the mode from the .conf file */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "mixer-mode", 1)) {
-		mode = AM_MIXER_MODE;
-	} else {
-		mode = AM_COMPAT_MODE;
-	}
-
-	pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "play-interrupts", I810_INTS);
-	if (pints > I810_MAX_INTS) {
-		ATRACE_32("i810_init_state() "
-		    "play interrupt rate too high, resetting", pints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "init_state() "
-		    "play interrupt rate set too high, %d, resetting to %d",
-		    pints, I810_INTS);
-		pints = I810_INTS;
-	} else if (pints < I810_MIN_INTS) {
-		ATRACE_32("i810_init_state() "
-		    "play interrupt rate too low, resetting", pints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "init_state() "
-		    "play interrupt rate set too low, %d, resetting to %d",
-		    pints, I810_INTS);
-		pints = I810_INTS;
-	}
-	rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "record-interrupts", I810_INTS);
-	if (rints > I810_MAX_INTS) {
-		ATRACE_32("i810_init_state() "
-		    "record interrupt rate too high, resetting", rints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "init_state() "
-		    "record interrupt rate set too high, %d, resetting to %d",
-		    rints, I810_INTS);
-		rints = I810_INTS;
-	} else if (rints < I810_MIN_INTS) {
-		ATRACE_32("i810_init_state() "
-		    "record interrupt rate too low, resetting", rints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "init_state() "
-		    "record interrupt rate set too low, %d, resetting to %d",
-		    rints, I810_INTS);
-		rints = I810_INTS;
-	}
-
-	/* fill in the device default state */
-	statep->i810_defaults.play.sample_rate = I810_DEFAULT_SR;
-	statep->i810_defaults.play.channels = I810_DEFAULT_CH;
-	statep->i810_defaults.play.precision = I810_DEFAULT_PREC;
-	statep->i810_defaults.play.encoding = I810_DEFAULT_ENC;
-	statep->i810_defaults.play.gain = I810_DEFAULT_PGAIN;
-	statep->i810_defaults.play.port = AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->i810_defaults.play.avail_ports = AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->i810_defaults.play.mod_ports = AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->i810_defaults.play.buffer_size = I810_BSIZE;
-	statep->i810_defaults.play.balance = I810_DEFAULT_BAL;
-
-	statep->i810_defaults.record.sample_rate = I810_DEFAULT_SR;
-	statep->i810_defaults.record.channels = I810_DEFAULT_CH;
-	statep->i810_defaults.record.precision = I810_DEFAULT_PREC;
-	statep->i810_defaults.record.encoding = I810_DEFAULT_ENC;
-	statep->i810_defaults.record.gain = I810_DEFAULT_PGAIN;
-	statep->i810_defaults.record.port = AUDIO_MICROPHONE;
-	statep->i810_defaults.record.avail_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->i810_defaults.record.mod_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->i810_defaults.record.buffer_size = I810_BSIZE;
-	statep->i810_defaults.record.balance = I810_DEFAULT_BAL;
-
-	statep->i810_defaults.monitor_gain = I810_DEFAULT_MONITOR_GAIN;
-	statep->i810_defaults.output_muted = B_FALSE;
-	statep->i810_defaults.ref_cnt = B_FALSE;
-	statep->i810_defaults.hw_features =
-	    AUDIO_HWFEATURE_DUPLEX | AUDIO_HWFEATURE_PLAY |
-	    AUDIO_HWFEATURE_IN2OUT | AUDIO_HWFEATURE_RECORD;
-	statep->i810_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	if (cdrom) {
-		statep->i810_defaults.record.avail_ports |= AUDIO_CD;
-		statep->i810_defaults.record.mod_ports |= AUDIO_CD;
-	}
-
-	statep->i810_psample_rate = statep->i810_defaults.play.sample_rate;
-	statep->i810_pchannels = statep->i810_defaults.play.channels;
-	statep->i810_pprecision = statep->i810_defaults.play.precision;
-	statep->i810_csample_rate = statep->i810_defaults.record.sample_rate;
-	statep->i810_cchannels = statep->i810_defaults.record.channels;
-	statep->i810_cprecision = statep->i810_defaults.record.precision;
-
-	/*
-	 * fill in the ad_info structure
-	 */
-	statep->ad_info.ad_mode = mode;
-	statep->ad_info.ad_int_vers = AM_VERSION;
-	statep->ad_info.ad_add_mode = NULL;
-	statep->ad_info.ad_codec_type = AM_TRAD_CODEC;
-	statep->ad_info.ad_defaults = &statep->i810_defaults;
-	statep->ad_info.ad_play_comb = audio810_combinations;
-	statep->ad_info.ad_rec_comb = audio810_combinations;
-	statep->ad_info.ad_entry = &audio810_entry;
-	statep->ad_info.ad_dev_info = &statep->i810_dev_info;
-	statep->ad_info.ad_diag_flags = AM_DIAG_INTERNAL_LOOP;
-	statep->ad_info.ad_diff_flags =
-	    AM_DIFF_SR | AM_DIFF_CH | AM_DIFF_PREC | AM_DIFF_ENC;
-	statep->ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	statep->ad_info.ad_misc_flags = AM_MISC_RP_EXCL | AM_MISC_MONO_DUP;
-	statep->ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	statep->ad_info.ad_play.ad_mixer_srs = audio810_mixer_sample_rates;
-	statep->ad_info.ad_play.ad_compat_srs = audio810_compat_sample_rates;
-	statep->ad_info.ad_play.ad_conv = &am_src2;
-	statep->ad_info.ad_play.ad_sr_info = NULL;
-	statep->ad_info.ad_play.ad_chs = audio810_channels;
-	statep->ad_info.ad_play.ad_int_rate = pints;
-	statep->ad_info.ad_play.ad_max_chs = I810_MAX_OUT_CHANNELS;
-	statep->ad_info.ad_play.ad_bsize = I810_BSIZE;
-
-	/* record capabilities */
-	statep->ad_info.ad_record.ad_mixer_srs = audio810_mixer_sample_rates;
-	statep->ad_info.ad_record.ad_compat_srs =
-	    audio810_compat_sample_rates;
-	statep->ad_info.ad_record.ad_conv = &am_src2;
-	statep->ad_info.ad_record.ad_sr_info = NULL;
-	statep->ad_info.ad_record.ad_chs = audio810_channels;
-	statep->ad_info.ad_record.ad_int_rate = rints;
-	statep->ad_info.ad_record.ad_max_chs = I810_MAX_CHANNELS;
-	statep->ad_info.ad_record.ad_bsize = I810_BSIZE;
-
-	if (ddi_get_iblock_cookie(dip, (uint_t)0, &statep->intr_iblock) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!init_state() cannot get iblock cookie");
-		return (AUDIO_FAILURE);
-	}
-	mutex_init(&statep->inst_lock, NULL, MUTEX_DRIVER, statep->intr_iblock);
-	cv_init(&statep->i810_cv, NULL, CV_DRIVER, NULL);
-
-	/* fill in device info strings */
-	(void) strcpy(statep->i810_dev_info.name, I810_DEV_NAME);
-	(void) strcpy(statep->i810_dev_info.config, I810_DEV_CONFIG);
-	(void) strcpy(statep->i810_dev_info.version, I810_DEV_VERSION);
-
-	statep->play_buf_size = I810_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / pints;
-	statep->play_buf_size += I810_MOD_SIZE -
-	    (statep->play_buf_size % I810_MOD_SIZE);
-	statep->record_buf_size = I810_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / rints;
-	statep->record_buf_size += I810_MOD_SIZE -
-	    (statep->record_buf_size % I810_MOD_SIZE);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_init_state */
-
-
-/*
- * audio810_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. Finally, the registers are mapped in.
- *
- *	CAUTION: Make sure all errors call audio_sup_log().
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-audio810_map_regs(dev_info_t *dip, audio810_state_t *statep)
-{
-	ddi_dma_cookie_t	cookie;
-	uint_t			count;
-	uint_t			nregs = 0;
-	int			*regs_list;
-	int			i;
-	int			pciBar1 = 0;
-	int			pciBar2 = 0;
-	int			pciBar3 = 0;
-	int			pciBar4 = 0;
-
-	ATRACE("audio810_map_regs()", statep);
-
-	statep->i810_res_flags = 0;
-
-	/* map PCI config space */
-	if (pci_config_setup(statep->dip, &statep->pci_conf_handle) ==
-	    DDI_FAILURE) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() configuration memory mapping failed");
-		goto error;
-	}
-	statep->i810_res_flags |= I810_RS_PCI_REGS;
-
-	/* check the "reg" property to get the length of memory-mapped I/O */
-	if (ddi_prop_lookup_int_array(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "reg", (int **)&regs_list, &nregs) != DDI_PROP_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() inquire regs property failed");
-		goto error;
-	}
-	/*
-	 * Some hardwares, such as Intel ICH0/ICH and AMD 8111, use PCI 0x10
-	 * and 0x14 BAR separately for native audio mixer BAR and native bus
-	 * mastering BAR. More advanced hardwares, such as Intel ICH4 and ICH5,
-	 * support PCI memory BAR, via PCI 0x18 and 0x1C BAR, that allows for
-	 * higher performance access to the controller register. All features
-	 * can be accessed via this BAR making the I/O BAR (PCI 0x10 and 0x14
-	 * BAR) capabilities obsolete. However, these controller maintain the
-	 * I/O BAR capability to allow for the reuse of legacy code maintaining
-	 * backward compatibility. The I/O BAR is disabled unless system BIOS
-	 * enables the simultaneous backward compatible capability on the 0x41
-	 * register.
-	 *
-	 * When I/O BAR is enabled, the value of "reg" property should be like
-	 * this,
-	 *	phys_hi   phys_mid  phys_lo   size_hi   size_lo
-	 * --------------------------------------------------------
-	 *	0000fd00  00000000  00000000  00000000  00000000
-	 *	0100fd10  00000000  00000000  00000000  00000100
-	 *	0100fd14  00000000  00000000  00000000  00000040
-	 *	0200fd18  00000000  00000000  00000000  00000200
-	 *	0200fd1c  00000000  00000000  00000000  00000100
-	 *
-	 * When I/O BAR is disabled, the "reg" property of the device node does
-	 * not consist of the description for the I/O BAR. The following example
-	 * illustrates the vaule of "reg" property,
-	 *
-	 *	phys_hi   phys_mid  phys_lo   size_hi   size_lo
-	 * --------------------------------------------------------
-	 *	0000fd00  00000000  00000000  00000000  00000000
-	 *	0200fd18  00000000  00000000  00000000  00000200
-	 *	0200fd1c  00000000  00000000  00000000  00000100
-	 *
-	 * If the hardware has memory-mapped I/O access, first try to use
-	 * this facility, otherwise we will try I/O access.
-	 */
-	for (i = 1; i < nregs/I810_INTS_PER_REG_PROP; i++) {
-		switch (regs_list[I810_INTS_PER_REG_PROP * i] & 0x000000ff) {
-			case 0x10:
-				pciBar1 = i;
-				break;
-			case 0x14:
-				pciBar2 = i;
-				break;
-			case 0x18:
-				pciBar3 = i;
-				break;
-			case 0x1c:
-				pciBar4 = i;
-				break;
-			default:	/* we don't care others */
-				break;
-		}
-	}
-
-	if ((pciBar3 != 0) && (pciBar4 != 0)) {
-		/* map audio mixer registers */
-		if ((ddi_regs_map_setup(dip, pciBar3,
-		    (caddr_t *)&statep->am_regs_base, 0,
-		    regs_list[I810_INTS_PER_REG_PROP * pciBar3 +
-		    I810_REG_PROP_ADDR_LEN_IDX], &dev_attr,
-		    &statep->am_regs_handle)) != DDI_SUCCESS) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!map_regs() memory am mapping failed, len=0x%08x",
-			    regs_list[I810_INTS_PER_REG_PROP * pciBar3 +
-			    I810_REG_PROP_ADDR_LEN_IDX]);
-			goto error;
-		}
-		statep->i810_res_flags |= I810_RS_AM_REGS;
-
-		/* map bus master register */
-		if ((ddi_regs_map_setup(dip, pciBar4,
-		    (caddr_t *)&statep->bm_regs_base, 0,
-		    regs_list[I810_INTS_PER_REG_PROP * pciBar4 +
-		    I810_REG_PROP_ADDR_LEN_IDX], &dev_attr,
-		    &statep->bm_regs_handle)) != DDI_SUCCESS) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!map_regs() memory bm mapping failed, len=0x%08x",
-			    regs_list[I810_INTS_PER_REG_PROP * pciBar4 +
-			    I810_REG_PROP_ADDR_LEN_IDX]);
-			goto error;
-		}
-		statep->i810_res_flags |= I810_RS_BM_REGS;
-
-	} else if ((pciBar1 != 0) && (pciBar2 != 0)) {
-		/* map audio mixer registers */
-		if ((ddi_regs_map_setup(dip, pciBar1,
-		    (caddr_t *)&statep->am_regs_base, 0,
-		    regs_list[I810_INTS_PER_REG_PROP * pciBar1 +
-		    I810_REG_PROP_ADDR_LEN_IDX], &dev_attr,
-		    &statep->am_regs_handle)) != DDI_SUCCESS) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!map_regs() I/O am mapping failed, len=0x%08x",
-			    regs_list[I810_INTS_PER_REG_PROP * pciBar1 +
-			    I810_REG_PROP_ADDR_LEN_IDX]);
-			goto error;
-		}
-		statep->i810_res_flags |= I810_RS_AM_REGS;
-
-		/* map bus master register */
-		if ((ddi_regs_map_setup(dip, pciBar2,
-		    (caddr_t *)&statep->bm_regs_base, 0,
-		    regs_list[I810_INTS_PER_REG_PROP * pciBar2 +
-		    I810_REG_PROP_ADDR_LEN_IDX], &dev_attr,
-		    &statep->bm_regs_handle)) != DDI_SUCCESS) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!map_regs() I/O bm mapping failed, len=: 0x%08x",
-			    regs_list[I810_INTS_PER_REG_PROP * pciBar2 +
-			    I810_REG_PROP_ADDR_LEN_IDX]);
-			goto error;
-		}
-		statep->i810_res_flags |= I810_RS_BM_REGS;
-	} else {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_reg() pci BAR error");
-		goto error;
-	}
-
-	/*
-	 * now, from here we allocate DMA memory for buffer descriptor list.
-	 * we allocate adjacent DMA memory for all DMA engines.
-	 */
-	if (ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &statep->bdl_dma_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() ddi_dma_alloc_handle(bdlist) failed ");
-		goto error;
-	}
-	statep->i810_res_flags |= I810_RS_DMA_BDL_HANDLE;
-
-	/*
-	 * we allocate all buffer descriptors lists in continuous dma memory.
-	 */
-	if (ddi_dma_mem_alloc(statep->bdl_dma_handle,
-	    sizeof (i810_bd_entry_t) * I810_BD_NUMS * 2,
-	    &dev_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    (caddr_t *)&statep->bdl_virtual, &statep->bdl_size,
-	    &statep->bdl_acc_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() ddi_dma_mem_alloc(bdlist) failed");
-		goto error;
-	}
-	statep->i810_res_flags |= I810_RS_DMA_BDL_MEM;
-
-	if (ddi_dma_addr_bind_handle(statep->bdl_dma_handle, NULL,
-	    (caddr_t)statep->bdl_virtual, statep->bdl_size,
-	    DDI_DMA_RDWR|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &cookie,
-	    &count) != DDI_DMA_MAPPED) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() addr_bind_handle failed");
-		goto error;
-	}
-
-	/*
-	 * there some bugs in the DDI framework and it is possible to
-	 * get multiple cookies
-	 */
-	if (count != 1) {
-		(void) ddi_dma_unbind_handle(statep->bdl_dma_handle);
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!map_regs() addr_bind_handle failed, cookies > 1");
-		goto error;
-	}
-
-	statep->bdl_virt_pin = (i810_bd_entry_t *)(statep->bdl_virtual);
-	statep->bdl_virt_pout = statep->bdl_virt_pin + I810_BD_NUMS;
-	statep->bdl_phys_pin = (uint32_t)(cookie.dmac_address);
-	statep->bdl_phys_pout = statep->bdl_phys_pin +
-	    sizeof (i810_bd_entry_t) * I810_BD_NUMS;
-
-	statep->i810_res_flags |= I810_RS_DMA_BDL_BIND;
-
-	ddi_prop_free(regs_list);
-
-	return (AUDIO_SUCCESS);
-
-error:
-	if (nregs > 0) {
-		ddi_prop_free(regs_list);
-	}
-	audio810_unmap_regs(statep);
-
-	return (AUDIO_FAILURE);
-
-}	/* audio810_map_regs() */
-
-/*
- * audio810_unmap_regs()
- *
- * Description:
- *	This routine unbinds the play and record DMA handles, frees
- *	the DMA buffers and the unmaps control registers.
- *
- * Arguments:
- *	audio810_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_unmap_regs(audio810_state_t *statep)
-{
-	if (statep->i810_res_flags & I810_RS_DMA_BDL_BIND) {
-		statep->i810_res_flags &= ~I810_RS_DMA_BDL_BIND;
-		(void) ddi_dma_unbind_handle(statep->bdl_dma_handle);
-	}
-
-	if (statep->i810_res_flags & I810_RS_DMA_BDL_MEM) {
-		statep->i810_res_flags &= ~I810_RS_DMA_BDL_MEM;
-		ddi_dma_mem_free(&statep->bdl_acc_handle);
-	}
-
-	if (statep->i810_res_flags & I810_RS_DMA_BDL_HANDLE) {
-		statep->i810_res_flags &= ~I810_RS_DMA_BDL_HANDLE;
-		ddi_dma_free_handle(&statep->bdl_dma_handle);
-	}
-
-	if (statep->i810_res_flags & I810_RS_BM_REGS) {
-		statep->i810_res_flags &= ~I810_RS_BM_REGS;
-		ddi_regs_map_free(&statep->bm_regs_handle);
-	}
-
-	if (statep->i810_res_flags & I810_RS_AM_REGS) {
-		statep->i810_res_flags &= ~I810_RS_AM_REGS;
-		ddi_regs_map_free(&statep->am_regs_handle);
-	}
-
-	if (statep->i810_res_flags & I810_RS_PCI_REGS) {
-		statep->i810_res_flags &= ~I810_RS_PCI_REGS;
-		pci_config_teardown(&statep->pci_conf_handle);
-	}
-
-}	/* audio810_unmap_regs() */
-
-/*
- * audio810_alloc_sample_buf()
- *
- * Description:
- *	This routine allocates DMA buffers for the sample buffer. It
- *	allocates two DMA chunks (buffers) to the specified DMA engine
- *	(sample buffer structure). The two data chunks will be bound
- *	to the buffer descriptor entries of corresponding buffer
- *	descriptor list, and be used to transfer audio sample data to
- *	and from the audio controller.
- *
- * Arguments:
- *	audio810_state_t	*state	The device's state structure
- *	int			which	Which sample buffer, PCM in or PCM out
- *					I810_DMA_PCM_IN ---PCM in DMA engine
- *					I810_DMA_PCM_OUT---PCM out DMA engine
- *	int			len	The length of the DMA buffers
- *
- * Returns:
- *	AUDIO_SUCCESS	 Allocating DMA buffers successfully
- *	AUDIO_FAILURE	 Failed to allocate dma buffers
- */
-
-static int
-audio810_alloc_sample_buf(audio810_state_t *statep, int which, int len)
-{
-	i810_sample_buf_t	*buf;
-	i810_bdlist_chunk_t	*chunk;
-	ddi_dma_cookie_t	cookie;
-	uint_t			count;
-	int			i;
-
-	if (which == I810_DMA_PCM_OUT) {
-		buf = &statep->play_buf;
-	} else {
-		ASSERT(which == I810_DMA_PCM_IN);
-		buf = &statep->record_buf;
-	}
-
-	for (i = 0; i < 2; i++) {
-		chunk = &(buf->chunk[i]);
-
-		if (ddi_dma_alloc_handle(statep->dip, &sample_buf_dma_attr,
-		    DDI_DMA_SLEEP, NULL, &chunk->dma_handle) !=
-		    DDI_SUCCESS) {
-			goto error;
-		}
-
-		if (ddi_dma_mem_alloc(chunk->dma_handle, len, &dev_attr,
-		    DDI_DMA_STREAMING, DDI_DMA_SLEEP, NULL, &chunk->data_buf,
-		    &chunk->real_len, &chunk->acc_handle) != DDI_SUCCESS) {
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		if (ddi_dma_addr_bind_handle(chunk->dma_handle, NULL,
-		    chunk->data_buf, chunk->real_len, DDI_DMA_WRITE,
-		    DDI_DMA_SLEEP, NULL, &cookie, &count) !=
-		    DDI_DMA_MAPPED) {
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		/*
-		 * there some bugs in the DDI framework and it is possible to
-		 * get multiple cookies
-		 */
-		if (count != 1) {
-			(void) ddi_dma_unbind_handle(chunk->dma_handle);
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		chunk->addr_phy = (uint32_t)cookie.dmac_address;
-	}
-
-	return (AUDIO_SUCCESS);
-
-error:
-	if (i != 0) {
-		(void) ddi_dma_unbind_handle((buf->chunk[0].dma_handle));
-		ddi_dma_mem_free(&(buf->chunk[0].acc_handle));
-		ddi_dma_free_handle(&(buf->chunk[0].dma_handle));
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audio810_alloc_sample_buf() */
-
-/*
- * audio810_free_sample_buf()
- *
- * Description:
- *	This routine frees the DMA buffers of the sample buffer. The DMA
- *	buffers were allocated by calling audio810_alloc_sample_buf().
- *
- * Arguments:
- *	audio810_state_t	*state	The device's state structure
- *	i810_sample_buf_t	*buf	The sample buffer structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_free_sample_buf(audio810_state_t *statep, i810_sample_buf_t *buf)
-{
-	i810_bdlist_chunk_t	 *chunk;
-	int 	i;
-
-	ATRACE("audio810_free_sample_buf() audio810_statep", statep);
-
-	for (i = 0; i < 2; i++) {
-		chunk = &(buf->chunk[i]);
-		(void) ddi_dma_unbind_handle(chunk->dma_handle);
-		ddi_dma_mem_free(&chunk->acc_handle);
-		chunk->acc_handle = 0;
-		ddi_dma_free_handle(&chunk->dma_handle);
-	}
-
-}	/* audio810_free_sample_buf() */
-
-/*
- * audio810_reclaim_play_buf()
- *
- * Description:
- *	When the audio controller finishes fetching the data from DMA
- *	buffers, this routine will be called by interrupt handler to
- *	reclaim the DMA buffers.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_reclaim_play_buf(audio810_state_t *statep)
-{
-	i810_sample_buf_t	*buf;
-	int16_t		bmciv;
-
-	ASSERT(mutex_owned(&statep->inst_lock));
-
-	buf = &statep->play_buf;
-	bmciv = I810_BM_GET8(I810_PCM_OUT_CIV);
-	while (buf->head != bmciv) {
-		buf->avail++;
-		buf->head++;
-		if (buf->head >= I810_BD_NUMS) {
-			buf->head = 0;
-		}
-	}
-
-}	/* audio810_reclaim_play_buf() */
-
-/*
- * audio810_chip_init()
- *
- * Description:
- *	This routine initializes the AMD 8111 audio controller and the AC97
- *	codec.  The AC97 codec registers are programmed from codec_shadow[].
- *	If we are not doing a restore, we initialize codec_shadow[], otherwise
- *	we use the current values of shadow
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			restore		If I810_INIT_RESTORE then
- *						restore	from codec_shadow[]
- * Returns:
- *	AUDIO_SUCCESS	The hardware was initialized properly
- *	AUDIO_FAILURE	The hardware couldn't be initialized properly
- */
-static int
-audio810_chip_init(audio810_state_t *statep, int restore)
-{
-	uint32_t	gcr;
-	uint32_t	gsr;
-	uint32_t	codec_ready;
-	uint16_t	*shadow;
-	int 		loop;
-	int 		i;
-	int		j;
-	uint16_t	sr;
-	uint16_t	vid1;
-	uint16_t	vid2;
-	uint16_t	tmp;
-	clock_t		ticks;
-
-	gcr = I810_BM_GET32(I810_REG_GCR);
-	ticks = drv_usectohz(100);
-
-	/*
-	 * SADA only supports stereo, so we set the channel bits
-	 * to "00" to select 2 channels.
-	 */
-	gcr &= ~(I810_GCR_ACLINK_OFF | I810_GCR_CHANNELS_MASK);
-
-	/*
-	 * Datasheet(ICH5, document number of Intel: 252751-001):
-	 * 3.6.5.5(page 37)
-	 * 	if reset bit(bit1) is "0", driver must set it
-	 * 	to "1" to de-assert the AC_RESET# signal in AC
-	 * 	link, thus completing a cold reset. But if the
-	 * 	bit is "1", then a warm reset is required.
-	 */
-	gcr |= (gcr & I810_GCR_COLD_RST) == 0 ?
-	    I810_GCR_COLD_RST:I810_GCR_WARM_RST;
-	I810_BM_PUT32(I810_REG_GCR, gcr);
-
-	/* according AC'97 spec, wait for codec reset */
-	for (loop = 6000; --loop >= 0; ) {
-		delay(ticks);
-		gcr = I810_BM_GET32(I810_REG_GCR);
-		if ((gcr & I810_GCR_WARM_RST) == 0) {
-			break;
-		}
-	}
-
-	/* codec reset failed */
-	if (loop < 0) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!Failed to reset codec");
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Wait for codec ready. The hardware can provide the state of
-	 * codec ready bit on SDATA_IN[0], SDATA_IN[1] or SDATA_IN[2]
-	 */
-	codec_ready =
-	    I810_GSR_PRI_READY | I810_GSR_SEC_READY | I810_GSR_TRI_READY;
-	for (loop = 7000; --loop >= 0; ) {
-		delay(ticks);
-		gsr = I810_BM_GET32(I810_REG_GSR);
-		if ((gsr & codec_ready) != 0) {
-			break;
-		}
-	}
-	if (loop < 0) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!No codec ready signal received");
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * put the audio controller into quiet state, everything off
-	 */
-	audio810_stop_dma(statep);
-
-	/* AC97 register reset */
-	if (audio810_reset_ac97(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	shadow = statep->codec_shadow;
-
-	if (restore == I810_INIT_NO_RESTORE) {
-		for (i = 0; i < I810_LAST_AC_REG; i += 2) {
-			(void) audio810_read_ac97(statep, i,
-			    &(shadow[I810_CODEC_REG(i)]));
-		}
-
-		/* 02h - set master line out volume, muted, 0dB */
-		shadow[I810_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] = MVR_MUTE;
-
-		/* 04h - set alternate line out volume, muted, 0dB */
-		shadow[I810_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-
-		/* 06h - set master mono volume, muted, 0dB */
-		shadow[I810_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE;
-
-		/* 08h - set master tone control to no modification */
-		shadow[I810_CODEC_REG(AC97_MASTER_TONE_CONTROL_REGISTER)] =
-		    MTCR_BASS_BYPASS|MTCR_TREBLE_BYPASS;
-
-		/* 0ah - open pc beep, 0dB */
-		shadow[I810_CODEC_REG(AC97_PC_BEEP_REGISTER)] = PCBR_0dB_ATTEN;
-
-		/* 0ch - set phone input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_PHONE_VOLUME_REGISTER)] =
-		    PVR_MUTE|PVR_0dB_GAIN;
-
-		/* 0eh - set mic input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_MIC_VOLUME_REGISTER)] =
-		    MICVR_MUTE|MICVR_0dB_GAIN;
-
-		/* 10h - set line input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_LINE_IN_VOLUME_REGISTER)] =
-		    LIVR_MUTE|LIVR_RIGHT_0dB_GAIN|LIVR_LEFT_0dB_GAIN;
-
-		/* 12h - set cd input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_CD_VOLUME_REGISTER)] =
-		    CDVR_MUTE|CDVR_RIGHT_0dB_GAIN|CDVR_LEFT_0dB_GAIN;
-
-		/* 14h - set video input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_VIDEO_VOLUME_REGISTER)] =
-		    VIDVR_MUTE|VIDVR_RIGHT_0dB_GAIN|VIDVR_LEFT_0dB_GAIN;
-
-		/* 16h - set aux input, mute, 0dB attenuation */
-		shadow[I810_CODEC_REG(AC97_AUX_VOLUME_REGISTER)] =
-		    AUXVR_MUTE|AUXVR_RIGHT_0dB_GAIN|AUXVR_LEFT_0dB_GAIN;
-
-		/* 18h - set PCM out input, NOT muted, 0dB gain */
-		shadow[I810_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_RIGHT_0dB_GAIN|PCMOVR_LEFT_0dB_GAIN;
-
-		/* 1ah - set input device as mic */
-		shadow[I810_CODEC_REG(AC97_RECORD_SELECT_CTRL_REGISTER)] =
-		    RSCR_R_MIC|RSCR_L_MIC;
-
-		/* 1ch - set record gain to 0dB and not muted */
-		shadow[I810_CODEC_REG(AC97_RECORD_GAIN_REGISTER)] =
-		    RGR_RIGHT_0db_GAIN|RGR_LEFT_0db_GAIN;
-
-		/* 1eh - set record mic gain to 0dB and not muted */
-		shadow[I810_CODEC_REG(AC97_RECORD_GAIN_MIC_REGISTER)] =
-		    RGMR_0db_GAIN;
-
-		/* 20h - set GP register, mic 1, everything else off */
-		shadow[I810_CODEC_REG(AC97_GENERAL_PURPOSE_REGISTER)] =
-		    GPR_MS_MIC1|GPR_MONO_MIX_IN;
-
-		/* 22h - set 3D control to NULL */
-		shadow[I810_CODEC_REG(AC97_THREE_D_CONTROL_REGISTER)] =
-		    TDCR_NULL;
-
-		/*
-		 * 26h - set EAPD to 1 for devices with ac97-invert-amp
-		 * property.
-		 *
-		 * According to AC'97 spec, EAPD (PR7) independently controls
-		 * an output pin that manages an optional external audio
-		 * amplifier. AC'97 compliance requires the implementation of
-		 * a dedicated output pin for external audio amplifier control.
-		 * The pin is controlled via the ¡°EAPD¡±(External Amplifier
-		 * Powerdown) bit in Powerdown Ctrl/Stat Register, bit 15
-		 * (formerly PR7). EAPD = 0 places a 0 on the output pin,
-		 * enabling an external audio amplifier, EAPD = 1 shuts it
-		 * down. Audio amplifier devices that operate with reverse
-		 * polarity may require an external inverter. By default,
-		 * EAPD = 0 is to enable external audio amplifier, but for
-		 * some Sony Vaio laptops, we need to revert polarity to
-		 * enable external amplifier.
-		 */
-		switch (ddi_prop_get_int(DDI_DEV_T_ANY, statep->dip,
-		    DDI_PROP_DONTPASS, "ac97-invert-amp", -1)) {
-		case -1:
-			/* not attempt to flip EAPD */
-			break;
-
-		case 0:
-			/* set EAPD to 0 */
-			shadow[I810_CODEC_REG(
-			    AC97_POWERDOWN_CTRL_STAT_REGISTER)] &= ~PCSR_EAPD;
-			break;
-
-		case 1:
-			/* set EAPD to 1 */
-			shadow[I810_CODEC_REG(
-			    AC97_POWERDOWN_CTRL_STAT_REGISTER)] |= PCSR_EAPD;
-			break;
-
-		default:
-			/* invalid */
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!Invalid value for ac97-invert-amp property");
-			break;
-		}
-
-		/*
-		 * The rest we ignore, most are reserved.
-		 */
-
-	}
-
-	if (restore == I810_INIT_RESTORE) {
-		/* Restore from saved values */
-		shadow[I810_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE;
-		shadow[I810_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-		shadow[I810_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE;
-		shadow[I810_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_MUTE;
-	}
-
-	/* Now we set the AC97 codec registers to the saved values */
-	for (i = 2; i <= I810_LAST_AC_REG; i += 2)
-		(void) audio810_write_ac97(statep, i,
-		    shadow[I810_CODEC_REG(i)]);
-
-	(void) audio810_read_ac97(statep, AC97_RESET_REGISTER, &tmp);
-	if (tmp & RR_HEADPHONE_SUPPORT) {
-		statep->i810_defaults.play.port |= AUDIO_HEADPHONE;
-		statep->i810_defaults.play.avail_ports |= AUDIO_HEADPHONE;
-		statep->i810_defaults.play.mod_ports |= AUDIO_HEADPHONE;
-	}
-
-	/*
-	 * Most vendors connect the surr-out of ad1980/ad1985 codecs to the
-	 * line-out jack. So far we haven't found which vendors don't
-	 * do that. So we assume that all vendors swap the surr-out
-	 * and the line-out outputs. So we need swap the two outputs.
-	 * But we still internally process the "ad198x-swap-output"
-	 * property. If someday some vendors do not swap the outputs,
-	 * we would set "ad198x-swap-output = 0" in the
-	 * /kernel/drv/audio810.conf file, and unload and reload the
-	 * audio810 driver (or reboot).
-	 */
-	(void) audio810_read_ac97(statep, AC97_VENDOR_ID1_REGISTER, &vid1);
-	(void) audio810_read_ac97(statep, AC97_VENDOR_ID2_REGISTER, &vid2);
-	if (vid1 == AD1980_VID1 &&
-	    (vid2 == AD1980_VID2 || vid2 == AD1985_VID2)) {
-		if (ddi_prop_get_int(DDI_DEV_T_ANY, statep->dip,
-		    DDI_PROP_DONTPASS, "ad198x-swap-output", 1) == 1) {
-			statep->swap_out = B_TRUE;
-			(void) audio810_read_ac97(statep, CODEC_AD_REG_MISC,
-			    &tmp);
-			(void) audio810_write_ac97(statep, CODEC_AD_REG_MISC,
-			    tmp | AD1980_MISC_LOSEL | AD1980_MISC_HPSEL);
-		}
-	}
-
-	/* check if the codec implements 6 bit volume register */
-	(void) audio810_write_ac97(statep, AC97_MASTER_VOLUME_REGISTER,
-	    MVR_MUTE | MVR_RIGHT_OPTIONAL_MASK | MVR_LEFT_OPTIONAL_MASK);
-	(void) audio810_read_ac97(statep, AC97_MASTER_VOLUME_REGISTER, &tmp);
-	if ((tmp & 0x7fff) != (MVR_RIGHT_MASK | MVR_LEFT_MASK)) {
-		statep->vol_bits_mask = 6;
-	}
-	/* resume the master volume to the max */
-	(void) audio810_write_ac97(statep, AC97_MASTER_VOLUME_REGISTER,
-	    MVR_MUTE);
-
-	/*
-	 * if the codec chip does not support variable sample rate,
-	 * we set the sample rate to 48K
-	 */
-	(void) audio810_read_ac97(statep, AC97_EXTENDED_AUDIO_REGISTER, &tmp);
-	audio_sup_log(statep->audio_handle, CE_NOTE,
-	    "!%s%d: xid=0x%04x, vid1=0x%04x, vid2=0x%04x",
-	    audio810_name,  ddi_get_instance(statep->dip), tmp, vid1, vid2);
-	if (!(tmp & EAR_VRA)) {
-		statep->var_sr = B_FALSE;
-		statep->ad_info.ad_record.ad_compat_srs =
-		    audio810_min_compat_sample_rates;
-		statep->ad_info.ad_play.ad_compat_srs =
-		    audio810_min_compat_sample_rates;
-		statep->i810_defaults.play.sample_rate =
-		    I810_SAMPR48000;
-		statep->i810_defaults.record.sample_rate =
-		    I810_SAMPR48000;
-	} else {	/* variable sample rate supported */
-		statep->var_sr = B_TRUE;
-
-		/* set variable rate mode */
-		(void) audio810_write_ac97(statep,
-		    AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER, EASCR_VRA);
-
-		/* check the sample rates supported */
-		for (i = 0, j = 0; audio810_compat_srs[i] != 0; i++) {
-			(void) audio810_write_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER,
-			    audio810_compat_srs[i]);
-			(void) audio810_read_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, &sr);
-
-			if (sr == audio810_compat_srs[i]) {
-				if (i != j) {
-					audio810_compat_srs[j] =
-					    audio810_compat_srs[i];
-				}
-				j++;
-			}
-		}
-
-		if (j < 1) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!No standard sample rate is supported");
-			return (AUDIO_FAILURE);
-		}
-		audio810_compat_srs[j] = 0;
-
-		/*
-		 * if the configuration doesn't support 8K sample rate,
-		 * we modify the default value to the first.
-		 */
-		if (audio810_compat_srs[0] != I810_SAMPR8000) {
-			statep->i810_defaults.play.sample_rate =
-			    audio810_compat_srs[0];
-			statep->i810_defaults.record.sample_rate =
-			    audio810_compat_srs[0];
-		}
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_chip_init() */
-
-/*
- * audio810_stop_dma()
- *
- * Description:
- *	This routine is used to put each DMA engine into the quiet state.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_stop_dma(audio810_state_t *statep)
-{
-	/* pause bus master (needed for the following reset register) */
-	I810_BM_PUT8(I810_PCM_IN_CR, 0x0);
-	I810_BM_PUT8(I810_PCM_OUT_CR, 0x0);
-	I810_BM_PUT8(I810_MIC_CR, 0x0);
-
-	/* and then reset the bus master registers for a three DMA engines */
-	I810_BM_PUT8(I810_PCM_IN_CR, I810_BM_CR_RST);
-	I810_BM_PUT8(I810_PCM_OUT_CR, I810_BM_CR_RST);
-	I810_BM_PUT8(I810_MIC_CR, I810_BM_CR_RST);
-
-	statep->flags = 0;
-
-/*
- * XXXX Not sure what these declarations are for, but I brought them from
- * the PM gate.
- */
-	statep->play_buf.io_started = B_FALSE;
-
-	statep->record_buf.io_started = B_FALSE;
-
-}	/* audio810_stop_dma() */
-
-/*
- * audio810_set_gain()
- *
- * Description:
- *	Set the play/record gain.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD, if
- *						direction is important
- *	int			arg1		The gain to set
- *	int			arg2		The channel, 0 == left
- *						or 1 == right
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audio810_set_gain(audio810_state_t *statep, int dir, int gain, int channel)
-{
-	uint16_t	tmp;
-	uint16_t	channel_gain;
-	int		regidx;
-	uint16_t	mask;
-
-	if (gain > AUDIO_MAX_GAIN) {
-		gain = AUDIO_MAX_GAIN;
-	} else if (gain < AUDIO_MIN_GAIN) {
-		gain = AUDIO_MIN_GAIN;
-	}
-
-	if (statep->vol_bits_mask == 6)
-		mask = 0x3f;
-	else
-		mask = 0x1f;
-
-	channel_gain = AUDIO_MAX_GAIN - gain;
-	channel_gain = ((channel_gain << statep->vol_bits_mask) -
-	    channel_gain) / AUDIO_MAX_GAIN;
-
-	if (dir == AUDIO_PLAY) {
-		if (statep->swap_out == B_TRUE) {
-			regidx = AC97_EXTENDED_LRS_VOLUME_REGISTER;
-		} else {
-			regidx = AC97_PCM_OUT_VOLUME_REGISTER;
-		}
-		(void) audio810_read_ac97(statep, regidx, &tmp);
-
-		if (channel == 0) { /* left channel */
-			tmp &= mask;
-			tmp |= (channel_gain << 8);
-		} else {	/* right channel */
-			tmp &= (mask << 8);
-			tmp |= channel_gain;
-		}
-
-		(void) audio810_write_ac97(statep, regidx, tmp);
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-
-		(void) audio810_read_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, &tmp);
-
-		if (channel == 0) {	/* left channel */
-			tmp &= ~RGR_LEFT_MASK;
-			tmp |= gain & RGR_LEFT_MASK;
-		} else {
-			/* right channel */
-			ASSERT(channel == 1);
-			tmp &= ~RGR_RIGHT_MASK;
-			tmp |= gain & RGR_RIGHT_MASK;
-		}
-		(void) audio810_write_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, tmp);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_set_gain() */
-
-/*
- * audio810_set_port()
- *
- * Description:
- *	Set the play/record port.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *						which is not how we program
- *						the device for now.
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD,
- *						if direction is important
- *	int			port		The port to set
- *				AUDIO_SPEAKER	output to built-in speaker
- *
- *				AUDIO_MICROPHONE	input from microphone
- *				AUDIO_LINE_IN		input from line in
- *				AUDIO_CODEC_LOOPB_IN	input from Codec
- *							internal loopback
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The port could not been set
- */
-static int
-audio810_set_port(audio810_state_t *statep, int dir, int port)
-{
-	uint16_t	tmp;
-
-	if (dir == AUDIO_PLAY) {	/* output port */
-		tmp = 0;
-		if (port == I810_PORT_UNMUTE) {
-			port = statep->i810_output_port;
-		}
-
-		if (port & AUDIO_SPEAKER) {
-			(void) audio810_and_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_SPEAKER;
-		} else {
-			(void) audio810_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MVR_MUTE);
-		}
-
-		if (port & AUDIO_LINE_OUT) {
-			if (statep->swap_out == B_FALSE) {
-				(void) audio810_and_ac97(statep,
-				    AC97_MASTER_VOLUME_REGISTER,
-				    (uint16_t)~MVR_MUTE);
-			} else {
-				(void) audio810_and_ac97(statep,
-				    AC97_EXTENDED_LRS_VOLUME_REGISTER,
-				    (uint16_t)~AD1980_SURR_MUTE);
-			}
-			tmp |= AUDIO_LINE_OUT;
-		} else {
-			if (statep->swap_out == B_FALSE) {
-				(void) audio810_or_ac97(statep,
-				    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-			} else {
-				(void) audio810_or_ac97(statep,
-				    AC97_EXTENDED_LRS_VOLUME_REGISTER,
-				    AD1980_SURR_MUTE);
-			}
-		}
-
-		if (port & AUDIO_HEADPHONE) {
-			(void) audio810_and_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_HEADPHONE;
-		} else {
-			(void) audio810_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, MVR_MUTE);
-		}
-
-		ATRACE_32("810_set_port() out port", tmp);
-		statep->i810_output_port = tmp;
-		if (tmp != port) {
-			ATRACE_32("810_set_port() bad out port", port);
-			return (AUDIO_FAILURE);
-		}
-
-	} else {		/* input port */
-		ASSERT(dir == AUDIO_RECORD);
-
-		switch (port) {
-		case AUDIO_NONE:
-			/* set to an unused input */
-			tmp = RSCR_R_PHONE | RSCR_L_PHONE;
-
-			/* mute the master record input */
-			(void) audio810_or_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER, RGR_MUTE);
-
-			if (statep->i810_monitor_gain) {
-				if (statep->i810_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio810_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audio810_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_CD) {
-					(void) audio810_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-
-		case AUDIO_MICROPHONE:
-			/* set to the mic input */
-			tmp = RSCR_R_MIC | RSCR_L_MIC;
-
-			if (statep->i810_monitor_gain) {
-				if (statep->i810_input_port == AUDIO_LINE_IN) {
-					(void) audio810_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_CD) {
-					(void) audio810_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				(void) audio810_write_ac97(statep,
-				    AC97_MIC_VOLUME_REGISTER,
-				    statep->i810_monitor_gain);
-			}
-			break;
-
-		case AUDIO_LINE_IN:
-			/* set to the line in input */
-			tmp = RSCR_R_LINE_IN | RSCR_L_LINE_IN;
-
-			/* see if we need to update monitor loopback */
-			if (statep->i810_monitor_gain) {
-				if (statep->i810_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio810_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_CD) {
-					(void) audio810_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				(void) audio810_write_ac97(statep,
-				    AC97_LINE_IN_VOLUME_REGISTER,
-				    statep->i810_monitor_gain);
-			}
-			break;
-
-		case AUDIO_CD:
-			/* set to the line in input */
-			tmp = RSCR_R_CD|RSCR_L_CD;
-
-			/* see if we need to update monitor loopback */
-			if (statep->i810_monitor_gain) {
-				if (statep->i810_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio810_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audio810_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				}
-				(void) audio810_write_ac97(statep,
-				    AC97_CD_VOLUME_REGISTER,
-				    statep->i810_monitor_gain);
-			}
-			break;
-
-		case AUDIO_CODEC_LOOPB_IN:
-			/* set to the loopback input */
-			tmp = RSCR_R_STEREO_MIX | RSCR_L_STEREO_MIX;
-
-			if (statep->i810_monitor_gain) {
-				if (statep->i810_input_port == AUDIO_LINE_IN) {
-					(void) audio810_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio810_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->i810_input_port ==
-				    AUDIO_CD) {
-					(void) audio810_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-
-		default:
-			ATRACE_32("810_set_port bad in port", port);
-			return (AUDIO_FAILURE);
-		}
-
-		/* select the input */
-		(void) audio810_write_ac97(statep,
-		    AC97_RECORD_SELECT_CTRL_REGISTER, tmp);
-		if ((port != AUDIO_NONE) &&
-		    (statep->codec_shadow[I810_CODEC_REG(
-		    AC97_RECORD_GAIN_REGISTER)] & RGR_MUTE)) {
-			(void) audio810_and_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER,
-			    (uint16_t)~RGR_MUTE);
-		}
-		statep->i810_input_port = port;
-	}
-
-	ATRACE_32("810_set_port() returning", 0);
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_set_port() */
-
-/*
- * audio810_set_monitor_gain()
- *
- * Description:
- *	Set the monitor gain.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			gain		The gain to set
- *
- * Returns:
- * 	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audio810_set_monitor_gain(audio810_state_t *statep, int gain)
-{
-	uint16_t	tmp_short;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("in audio810_set_monitor_gain()", statep);
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (gain == 0) {
-		/* disable loopbacks when gain == 0 */
-		tmp_short = MVR_MUTE;
-	} else {
-		/* Adjust the value of gain to the requirement of AC'97 */
-		tmp_short = AUDIO_MAX_GAIN - gain;
-		tmp_short = ((tmp_short << statep->vol_bits_mask) - tmp_short) /
-		    AUDIO_MAX_GAIN;
-		tmp_short |= (((tmp_short << statep->vol_bits_mask) -
-		    tmp_short) / AUDIO_MAX_GAIN) << 8;
-	}
-
-	switch (statep->i810_input_port) {
-	case AUDIO_NONE:
-		/*
-		 * It is possible to set the value of gain before any input
-		 * is selected. So, we just save the gain and then return
-		 * SUCCESS.
-		 */
-		break;
-
-	case AUDIO_MICROPHONE:
-		/*
-		 * MIC input has 20dB boost, we just preserve it
-		 */
-		tmp_short |= statep->codec_shadow[I810_CODEC_REG(
-		    AC97_MIC_VOLUME_REGISTER)] & MICVR_20dB_BOOST;
-		(void) audio810_write_ac97(statep,
-		    AC97_MIC_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_LINE_IN:
-		(void) audio810_write_ac97(statep,
-		    AC97_LINE_IN_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CD:
-		(void) audio810_write_ac97(statep,
-		    AC97_CD_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CODEC_LOOPB_IN:
-		/* we already are getting the loopback, so done */
-		rc = AUDIO_SUCCESS;
-		goto done;
-
-	default:
-		/* this should never happen! */
-		ATRACE("i810_ad_set_config() monitor gain bad device", NULL);
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-
-	if (gain == 0) {
-		statep->i810_monitor_gain = 0;
-	} else {
-		statep->i810_monitor_gain = tmp_short;
-	}
-
-done:
-	ATRACE_32("audio810_set_monitor_gain()", rc);
-
-	return (rc);
-
-}	/* audio810_set_monitor_gain() */
-
-/*
- * audio810_codec_sync()
- *
- * Description:
- *	Serialize access to the AC97 audio mixer registers.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Ready for an I/O access to the codec
- *	AUDIO_FAILURE		An I/O access is currently in progress, can't
- *				perform another I/O access.
- */
-static int
-audio810_codec_sync(audio810_state_t *statep)
-{
-	int 	i;
-	uint16_t	casr;
-
-	for (i = 0; i < 300; i++) {
-		casr = I810_BM_GET8(I810_REG_CASR);
-		if ((casr & 1) == 0) {
-			return (AUDIO_SUCCESS);
-		}
-		drv_usecwait(10);
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audio810_codec_sync() */
-
-/*
- * audio810_and_ac97()
- *
- * Description:
- *	Logically AND the value with the specified ac97 codec register
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to AND
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio810_and_ac97(audio810_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t	tmp;
-
-	if (audio810_codec_sync(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	I810_AM_PUT16(reg, data & statep->codec_shadow[I810_CODEC_REG(reg)]);
-
-	(void) audio810_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_and_ac97() */
-
-/*
- * audio810_or_ac97()
- *
- * Description:
- *	Logically OR the value with the specified ac97 codec register
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to OR
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio810_or_ac97(audio810_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t	tmp;
-
-	if (audio810_codec_sync(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	I810_AM_PUT16(reg, data | statep->codec_shadow[I810_CODEC_REG(reg)]);
-
-	(void) audio810_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_or_ac97() */
-
-/*
- * audio810_write_ac97()
- *
- * Description:
- *	Set the specific AC97 Codec register.
- *
- * Arguments:
- *	audio810_state_t	 *state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The data want to be set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio810_write_ac97(audio810_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t tmp;
-
-	if (audio810_codec_sync(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-	I810_AM_PUT16(reg, data);
-
-	(void) audio810_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_write_ac97() */
-
-/*
- * audio810_read_ac97()
- *
- * Description:
- *	Get the specific AC97 Codec register. It also updates codec_shadow[]
- *	with the register value.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		*data		The data to be returned
- *
- * Returns:
- *	AUDIO_SUCCESS		Reading the codec register successfully
- *	AUDIO_FAILURE		Failed to read the register
- */
-static int
-audio810_read_ac97(audio810_state_t *statep, int reg, uint16_t *data)
-{
-	if (audio810_codec_sync(statep) != AUDIO_SUCCESS) {
-		*data = 0xffff;
-		return (AUDIO_FAILURE);
-	}
-	*data = I810_AM_GET16(reg);
-	statep->codec_shadow[I810_CODEC_REG(reg)] = *data;
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_read_ac97() */
-
-/*
- * audio810_reset_ac97()
- *
- * Description:
- *	Reset AC97 Codec register.
- *
- * Arguments:
- *	audio810_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Reset the codec successfully
- *	AUDIO_FAILURE		Failed to reset the codec
- */
-static int
-audio810_reset_ac97(audio810_state_t *statep)
-{
-	uint16_t	tmp;
-
-	if (audio810_read_ac97(statep,
-	    AC97_POWERDOWN_CTRL_STAT_REGISTER, &tmp) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	if (audio810_write_ac97(statep, AC97_RESET_REGISTER, 1) !=
-	    AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	if (audio810_read_ac97(statep, AC97_RESET_REGISTER, &tmp) !=
-	    AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_reset_ac97() */
-
-/*
- * audio810_fill_play_buf()
- *
- * Description:
- *	This routine is called by i810_ad_start_play() and the interrupt
- *	handler. It fills playback samples into the DMA memory, sets the
- *	BDL entries, and starts the playback DMA engine.
- *
- * Arguments:
- *	audio810_state_t *statep	The device's state structure
- *
- * Returns:
- * 	AUDIO_SUCCESS		Starting PCM out engine successfully
- * 	AUDIO_FAILURE		Failed to start PCM out engine.
- */
-static int
-audio810_fill_play_buf(audio810_state_t *statep)
-{
-	i810_bdlist_chunk_t	*chunk;
-	i810_sample_buf_t	*buf;
-	i810_bd_entry_t		*bdesc;
-	int			samples;
-	int			rs;
-	uint8_t			cr;
-
-	buf = &statep->play_buf;
-
-	if (!buf->io_started) {
-		/*
-		 * ready to start PCM out engine
-		 */
-		I810_BM_PUT8(I810_PCM_OUT_CR, 0);
-		I810_BM_PUT8(I810_PCM_OUT_CR, I810_BM_CR_RST |I810_BM_CR_IOCE);
-		I810_BM_PUT8(I810_PCM_OUT_LVI, 0);
-		I810_BM_PUT32(I810_PCM_OUT_BD_BASE, statep->bdl_phys_pout);
-		buf->head = 0;
-		buf->tail = 0;
-		buf->avail = 2;	/* have only two buffers for playback */
-	}
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	samples = statep->i810_psample_rate * statep->i810_pchannels /
-	    statep->ad_info.ad_play.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	while (buf->avail > 0) {
-		chunk = &(buf->chunk[buf->tail & 1]);
-		mutex_exit(&statep->inst_lock);
-		rs = am_get_audio(statep->audio_handle,
-		    (char *)(chunk->data_buf), AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&statep->inst_lock);
-
-		if (((statep->flags & I810_DMA_PLAY_STARTED) == 0) &&
-		    (buf->io_started)) {
-			return (AUDIO_FAILURE);
-		}
-
-		if (rs <= 0) {
-			if (statep->flags & I810_DMA_PLAY_EMPTY) {
-
-				/*
-				 * Clear the flag so if audio is restarted while
-				 * in am_play_shutdown() we can detect it and
-				 * not mess things up.
-				 */
-				statep->flags &= ~I810_DMA_PLAY_STARTED;
-
-				/* shutdown the mixer */
-				mutex_exit(&statep->inst_lock);
-				am_play_shutdown(statep->audio_handle, NULL);
-				mutex_enter(&statep->inst_lock);
-
-				/*
-				 * Make sure playing wasn't restarted when lock
-				 * lost if reopened, should return success
-				 */
-				if (statep->flags & I810_DMA_PLAY_STARTED) {
-					return (AUDIO_SUCCESS);
-				}
-
-				/* Finished playing, then stop it */
-				I810_BM_PUT8(I810_PCM_OUT_CR, 0);
-				I810_BM_PUT8(I810_PCM_OUT_CR, I810_BM_CR_RST);
-				buf->io_started = B_FALSE;
-
-				/* clr the flags getting ready for next start */
-				statep->flags &= ~(I810_DMA_PLAY_PAUSED |
-				    I810_DMA_PLAY_EMPTY);
-
-				/* return the value for i810_ad_start_play() */
-				return (AUDIO_FAILURE);
-			} else {
-				/*
-				 * this time, we use one BD entry with empty
-				 * buffer next time we shut down, if no sound
-				 * again
-				 */
-				statep->flags |= I810_DMA_PLAY_EMPTY;
-			}
-		} else {
-			/* we got at least one sample */
-			statep->flags &= ~I810_DMA_PLAY_EMPTY;
-			(void) ddi_dma_sync(chunk->dma_handle, 0, rs << 1,
-			    DDI_DMA_SYNC_FORDEV);
-		}
-
-		/* put the samples into buffer descriptor list entry */
-		bdesc = &(statep->bdl_virt_pout[buf->tail]);
-		bdesc->buf_base = chunk->addr_phy;
-		bdesc->buf_len = (uint16_t)rs;
-		bdesc->cmd_bup = 0;
-		bdesc->reserved = 0;
-		bdesc->cmd_ioc = 1;
-		I810_BM_PUT8(I810_PCM_OUT_LVI, buf->tail);
-		buf->tail++;
-		buf->avail--;
-
-		if (buf->tail >= I810_BD_NUMS) {
-			buf->tail = 0;
-		}
-	}
-
-	cr = I810_BM_GET8(I810_PCM_OUT_CR);
-
-	if (!buf->io_started) {
-		cr &= ~(I810_BM_CR_FEIE|I810_BM_CR_LVBIE);
-		cr |= I810_BM_CR_IOCE;
-		buf->io_started = B_TRUE;
-	}
-
-	/* start PCM out engine */
-	cr |= I810_BM_CR_RUN;
-	I810_BM_PUT8(I810_PCM_OUT_CR, cr);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_fill_play_buf() */
-
-/*
- * audio810_prepare_record_buf()
- *
- * Description:
- *	This routine is called by audio810_ad_start_record(). It prepares DMA
- *	memory for PCM in engine, sets the buffer descriptor entries for PCM
- *	in engine, and starts PCM in engine for recording.
- *
- * Arguments:
- *	audio810_state_t	*statep		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Started PCM in  engine successfully
- *	AUDIO_FAILURE		Failed to start PCM in engine.
- *
- */
-static int
-audio810_prepare_record_buf(audio810_state_t *statep)
-{
-	i810_bdlist_chunk_t	*chunk;
-	i810_sample_buf_t	*buf;
-	i810_bd_entry_t		*bdesc;
-	int			samples;
-	uint8_t			cr;
-
-	buf = &statep->record_buf;
-
-	if (!buf->io_started) {
-		/* pause PCM in DMA engine */
-		I810_BM_PUT8(I810_PCM_IN_CR, I810_BM_CR_PAUSE);
-
-		/* reset PCM in DMA engine */
-		I810_BM_PUT8(I810_PCM_IN_CR, I810_BM_CR_RST |I810_BM_CR_IOCE);
-
-		/* set last valid index to 0 */
-		I810_BM_PUT8(I810_PCM_IN_LVI, 0);
-
-		/* buffer base */
-		I810_BM_PUT32(I810_PCM_IN_BD_BASE, statep->bdl_phys_pin);
-		buf->head = 0;
-		buf->tail = 0;
-		buf->avail = 2;
-	}
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	samples = statep->i810_csample_rate * statep->i810_cchannels /
-	    statep->ad_info.ad_record.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	statep->i810_csamples = samples;
-	while (buf->avail > 0) {
-		chunk = &buf->chunk[buf->tail & 1];
-		bdesc = &(statep->bdl_virt_pin[buf->tail]);
-		bdesc->buf_len = (uint16_t)samples;
-		bdesc->cmd_bup = 0;
-		bdesc->reserved = 0;
-		bdesc->cmd_ioc = 1;
-		bdesc->buf_base = chunk->addr_phy;
-		I810_BM_PUT8(I810_PCM_IN_LVI, buf->tail);
-		buf->tail++;
-		buf->avail--;
-		if (buf->tail >= I810_BD_NUMS) {
-			buf->tail = 0;
-		}
-	}
-	cr = I810_BM_GET8(I810_PCM_IN_CR);
-
-	if (!buf->io_started) {
-		cr &= ~(I810_BM_CR_FEIE|I810_BM_CR_LVBIE);
-		cr |= I810_BM_CR_IOCE;
-		buf->io_started = B_TRUE;
-	}
-
-	if (buf->avail < 2) {
-		cr |= I810_BM_CR_RUN;
-	}
-
-	I810_BM_PUT8(I810_PCM_IN_CR, cr);
-	cr = I810_BM_GET8(I810_PCM_IN_CR);
-
-	if ((cr & (I810_BM_CR_RUN | I810_BM_CR_IOCE)) !=
-	    (I810_BM_CR_RUN | I810_BM_CR_IOCE)) {
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio810_prepare_record_buf() */
-
-/*
- * audio810_reclaim_record_buf()
- *
- * Description:
- *	This routine is called by the interrupt handler. It sends the PCM
- *	samples (record data) up to the mixer module by calling am_send_audio(),
- *	and reclaims the buffer descriptor entries for PCM in engine.
- *
- * Arguments:
- *	audio810_state_t	*statep		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio810_reclaim_record_buf(audio810_state_t *statep)
-{
-	i810_bdlist_chunk_t 	*chunk;
-	i810_sample_buf_t	*buf;
-	int16_t		bmciv;
-	int		samples;
-
-	buf = &statep->record_buf;
-	bmciv = I810_BM_GET8(I810_PCM_IN_CIV);
-	samples = statep->i810_csamples;
-
-	while ((buf->head != bmciv) && (buf->avail < 2)) {
-		chunk = &buf->chunk[buf->head & 1];
-		(void) ddi_dma_sync(chunk->dma_handle, 0,
-		    chunk->real_len, DDI_DMA_SYNC_FORCPU);
-		mutex_exit(&statep->inst_lock);
-		am_send_audio(statep->audio_handle, chunk->data_buf,
-		    AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&statep->inst_lock);
-		buf->avail++;
-		buf->head++;
-
-		if (buf->head >= I810_BD_NUMS) {
-			buf->head = 0;
-		}
-		if ((statep->flags & I810_DMA_RECD_STARTED) == 0) {
-			break;
-		}
-	}
-
-}	/* audio810_reclaim_record_buf() */
diff --git a/usr/src/uts/common/io/audio/sada/drv/audioixp/audioixp.c b/usr/src/uts/common/io/audio/sada/drv/audioixp/audioixp.c
deleted file mode 100644
index 743a9dec05..0000000000
--- a/usr/src/uts/common/io/audio/sada/drv/audioixp/audioixp.c
+++ /dev/null
@@ -1,3389 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-/*
- * audioixp Audio Driver
- *
- * This driver supports audio hardware integrated in ATI IXP400 chipset.
- *
- * This driver uses the mixer Audio Personality Module to implement
- * audio(7I) and mixer(7I) semantics. Both play and record are single
- * streaming.
- *
- * The IXP400 audio core is an AC'97 controller, which has independent
- * channels for PCM in, PCM out. The AC'97 controller is a PCI bus master
- * with scatter/gather support. Each channel has a DMA engine. Currently,
- * we use only the PCM in and PCM out channels. Each DMA engine uses one
- * buffer descriptor list.  And the buffer descriptor list is an array
- * of up to 32 entries, each of which describes a data buffer. You dont need
- * to use all these entries. Each entry contains a pointer to a data buffer,
- * status, length of the buffer being pointed to and the pointer to the next
- * entry. Length of the buffer is in number of bytes. Interrupt will be
- * triggered each time a entry is processed by hardware.
- *
- * We use the BD list (buffer descriptor list) as a round-robin FIFO.
- * Both the software and hardware loop around the BD list. For playback,
- * the software writes to the buffers pointed by the BD entries of BD
- * list, and the hardware sends the data in the buffers out. For record,
- * the process is reversed. So we define the struct, audioixp_sample_buf,
- * to handle BD. The software uses the head, tail and avail fields of
- * this structure to manipulate the FIFO. The head field indicates the
- * first valid BD hardware can manipulate. The tail field indicates the
- * BD after the last valid BD. And the avail field indicates how many
- * buffers are available. Two DMA buffers are allocated for both playback
- * and record, and two BD entries are used. When processing interrupt,
- * the current hardware pointer will be check to tell which buffer is
- * being processed. It's possible for the hardware to interrupt twice
- * for one buffer, this logic is handled in the routine
- * audioixp_chunk_processed.
- *
- * Every time we program AC97 codec, we save the value in codec_shadow[].
- * This means that register state information is saved for power management
- * shutdown (we'll support this later). When the codec is started back up
- * we use this saved state to restore codec's state in audioixp_chip_init().
- *
- * System power management is not yet supported by the driver.
- *
- * 	NOTE:
- * 	This driver depends on the misc/audiosup, misc/amsrc2 and
- * 	misc/mixer modules being loaded first.
- */
-#include <sys/types.h>
-#include <sys/modctl.h>
-#include <sys/kmem.h>
-#include <sys/conf.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/pci.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/ac97.h>
-#include <sys/audio/impl/audioixp_impl.h>
-#include <sys/audio/audioixp.h>
-
-/*
- * Module linkage routines for the kernel
- */
-static int audioixp_getinfo(dev_info_t *, ddi_info_cmd_t, void*, void**);
-static int audioixp_attach(dev_info_t *, ddi_attach_cmd_t);
-static int audioixp_detach(dev_info_t *, ddi_detach_cmd_t);
-
-/*
- * Entry point routine prototypes
- */
-static int audioixp_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int audioixp_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int audioixp_ad_start_play(audiohdl_t, int);
-static void audioixp_ad_pause_play(audiohdl_t, int);
-static void audioixp_ad_stop_play(audiohdl_t, int);
-static int audioixp_ad_start_record(audiohdl_t, int);
-static void audioixp_ad_stop_record(audiohdl_t, int);
-
-/*
- * interrupt handler
- */
-static uint_t	audioixp_intr(caddr_t);
-
-/*
- * Local Routine Prototypes
- */
-static int audioixp_codec_sync(audioixp_state_t *);
-static int audioixp_write_ac97(audioixp_state_t *, int, uint16_t);
-static int audioixp_read_ac97(audioixp_state_t *, int, uint16_t *);
-static int audioixp_and_ac97(audioixp_state_t *, int, uint16_t);
-static int audioixp_or_ac97(audioixp_state_t *, int, uint16_t);
-static int audioixp_reset_ac97(audioixp_state_t *);
-static int audioixp_init_state(audioixp_state_t *, dev_info_t *);
-static int audioixp_map_regs(dev_info_t *, audioixp_state_t *);
-static void audioixp_unmap_regs(audioixp_state_t *);
-static int audioixp_alloc_sample_buf(audioixp_state_t *, int, int);
-static void audioixp_free_sample_buf(audioixp_state_t *,
-						audioixp_sample_buf_t *);
-static void audioixp_setup_bdl(audioixp_state_t *);
-static void audioixp_start_dma(audioixp_state_t *, int);
-static void audioixp_stop_dma(audioixp_state_t *, int);
-static int audioixp_chip_init(audioixp_state_t *, int);
-static void audioixp_chip_fini(audioixp_state_t *);
-static int audioixp_chunk_processed(audioixp_state_t *, int);
-static int audioixp_fill_play_buf(audioixp_state_t *);
-static int audioixp_prepare_record_buf(audioixp_state_t *);
-static void audioixp_reclaim_play_buf(audioixp_state_t *);
-static void audioixp_reclaim_record_buf(audioixp_state_t *);
-static int audioixp_set_gain(audioixp_state_t *, int, int, int);
-static int audioixp_set_port(audioixp_state_t *, int, int);
-static int audioixp_set_monitor_gain(audioixp_state_t *, int);
-
-/*
- * Global variables, but used only by this file.
- */
-
-/* anchor for soft state structures */
-static void	*audioixp_statep;
-
-/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
-static char	*audioixp_name = IXP_NAME;
-
-/*
- * STREAMS structures
- */
-
-/* STREAMS driver id and limit value struct */
-static struct module_info audioixp_modinfo = {
-	IXP_IDNUM,		/* module ID number */
-	IXP_NAME,		/* module name */
-	IXP_MINPACKET,		/* minimum packet size */
-	IXP_MAXPACKET,		/* maximum packet size */
-	IXP_HIWATER,		/* high water mark */
-	IXP_LOWATER,		/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit audioixp_rqueue = {
-	audio_sup_rput,		/* put procedure */
-	audio_sup_rsvc,		/* service procedure */
-	audio_sup_open,		/* open procedure */
-	audio_sup_close,	/* close procedure */
-	NULL,			/* unused */
-	&audioixp_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit audioixp_wqueue = {
-	audio_sup_wput,		/* write procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&audioixp_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab audioixp_str_info = {
-	&audioixp_rqueue,	/* read queue */
-	&audioixp_wqueue,	/* write queue */
-	NULL,			/* mux lower read queue */
-	NULL,			/* mux lower write queue */
-};
-
-/*
- * DDI Structures
- */
-
-/* Entry points structure */
-static struct cb_ops audioixp_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&audioixp_str_info,	/* cb_str */
-	D_NEW | D_MP | D_64BIT, /* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_awrite */
-};
-
-/* Device operations structure */
-static struct dev_ops audioixp_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	audioixp_getinfo,	/* devo_getinfo */
-	nulldev,		/* devo_identify - obsolete */
-	nulldev,		/* devo_probe */
-	audioixp_attach,	/* devo_attach */
-	audioixp_detach,	/* devo_detach */
-	nodev,			/* devo_reset */
-	&audioixp_cb_ops,	/* devi_cb_ops */
-	NULL,			/* devo_bus_ops */
-	NULL,			/* devo_power */
-	ddi_quiesce_not_supported,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv audioixp_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	IXP_MOD_NAME,		/* drv_linkinfo */
-	&audioixp_dev_ops,	/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage audioixp_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&audioixp_modldrv,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-static uint_t audioixp_mixer_srs[] = {
-	IXP_SAMPR8000, IXP_SAMPR48000, 0
-};
-
-static uint_t audioixp_min_compat_srs[] = {
-	IXP_SAMPR48000, 0
-};
-
-static uint_t audioixp_compat_srs [] = {
-	IXP_SAMPR8000, IXP_SAMPR11025, IXP_SAMPR16000,
-	IXP_SAMPR22050, IXP_SAMPR24000, IXP_SAMPR32000,
-	IXP_SAMPR44100, IXP_SAMPR48000, 0
-};
-
-static am_ad_sample_rates_t audioixp_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	audioixp_mixer_srs
-};
-
-static am_ad_sample_rates_t audioixp_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audioixp_compat_srs
-};
-
-/* Some codec, only support 48K sample rate */
-static am_ad_sample_rates_t audioixp_min_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audioixp_min_compat_srs
-};
-
-static uint_t audioixp_channels[] = {
-	AUDIO_CHANNELS_STEREO,
-	0
-};
-
-static am_ad_cap_comb_t audioixp_combinations[] = {
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
-
-/*
- * device access attributes for register mapping
- */
-static struct ddi_device_acc_attr dev_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_LE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-/*
- * DMA attributes of buffer descriptor list
- */
-static ddi_dma_attr_t	bdlist_dma_attr = {
-	DMA_ATTR_V0,	/* version */
-	0,		/* addr_lo */
-	0xffffffff,	/* addr_hi */
-	0x0000ffff,	/* count_max */
-	8,		/* align, BDL must be aligned on a 8-byte boundary */
-	0x3c,		/* burstsize */
-	8,		/* minxfer, set to the size of a BDlist entry */
-	0x0000ffff,	/* maxxfer */
-	0x00000fff,	/* seg, set to the RAM pagesize of intel platform */
-	1,		/* sgllen, there's no scatter-gather list */
-	8,		/* granular, set to the value of minxfer */
-	0		/* flags, use virtual address */
-};
-
-/*
- * DMA attributes of buffers to be used to receive/send audio data
- */
-static ddi_dma_attr_t	sample_buf_dma_attr = {
-	DMA_ATTR_V0,
-	0,		/* addr_lo */
-	0xffffffff,	/* addr_hi */
-	0x0001fffe,	/* count_max */
-	2,		/* align, data buffer is aligned on a 2-byte boundary */
-	0x3c,		/* burstsize */
-	4,		/* minxfer, set to the size of a sample data */
-	0x0001ffff,	/* maxxfer */
-	0x0001ffff,	/* seg */
-	1,		/* sgllen, no scatter-gather */
-	4,		/* granular, set to the value of minxfer */
-	0,		/* flags, use virtual address */
-};
-
-static am_ad_entry_t audioixp_entry = {
-	NULL,				/* ad_setup() */
-	NULL,				/* ad_teardown() */
-	audioixp_ad_set_config,		/* ad_set_config() */
-	audioixp_ad_set_format,		/* ad_set_format() */
-	audioixp_ad_start_play,		/* ad_start_play() */
-	audioixp_ad_pause_play,		/* ad_pause_play() */
-	audioixp_ad_stop_play,		/* ad_stop_play() */
-	audioixp_ad_start_record,	/* ad_start_record() */
-	audioixp_ad_stop_record,	/* ad_stop_record() */
-	NULL,				/* ad_ioctl() */
-	NULL				/* ad_iocdata() */
-};
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *	This is how access is initially given to all the static structures.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-	ATRACE("in audioixp _init()", NULL);
-
-	if ((error = ddi_soft_state_init(&audioixp_statep,
-	    sizeof (audioixp_state_t), 1)) != 0) {
-		ATRACE("audioixp ddi_soft_state_init() failed",
-		    audioixp_statep);
-		return (error);
-	}
-
-	if ((error = mod_install(&audioixp_modlinkage)) != 0) {
-		ddi_soft_state_fini(&audioixp_statep);
-	}
-
-	ATRACE("audioixp _init() audioixp_statep", audioixp_statep);
-	ATRACE("audioixp _init() returning", error);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description:
- *	Module de-initialization, called when the driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int		error;
-
-	ATRACE("in audioixp _fini()", audioixp_statep);
-
-	if ((error = mod_remove(&audioixp_modlinkage)) != 0) {
-		return (error);
-	}
-
-	ddi_soft_state_fini(&audioixp_statep);
-
-	ATRACE_32("audioixp _fini() returning", error);
-
-	return (0);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo		*modinfop	Pointer to the opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int	error;
-
-	ATRACE("in audioixp _info()", NULL);
-
-	error = mod_info(&audioixp_modlinkage, modinfop);
-
-	ATRACE_32("audioixp _info() returning", error);
-
-	return (error);
-
-}	/* _info() */
-
-
-/* ******************* Driver Entry Points ********************************* */
-
-/*
- * audioixp_getinfo()
- */
-static int
-audioixp_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd,
-	void *arg, void **result)
-{
-	audioixp_state_t	*state;
-	int 			instance;
-	int 			error;
-
-	error = DDI_FAILURE;
-	ATRACE("in audioixp_getinfo()", dip);
-
-	switch (cmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((state = ddi_get_soft_state(audioixp_statep,
-		    instance)) != NULL) {
-			*result = state->dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
-		break;
-
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void*)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-
-	default:
-		break;
-	}
-
-	return (error);
-
-}	/* audioixp_getinfo() */
-
-/*
- * audioixp_attach()
- *
- * Description:
- *	Attach an instance of the audioixp driver. This routine does
- * 	the device dependent attach tasks. When it is completed, it calls
- *	audio_sup_register() and am_attach() so they may do their work.
- *
- *	NOTE: mutex_init() no longer needs a name string, so set
- *		to NULL to save kernel space.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd	Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-static int
-audioixp_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	int 			instance;
-	uint16_t		cmdeg;
-	audioixp_state_t	*statep;
-	audio_sup_reg_data_t	data;
-
-	ATRACE("in audioixp_attach()", dip);
-
-	instance = ddi_get_instance(dip);
-
-	ATRACE("audioixp_attach() audioixp_statep",
-	    audioixp_statep);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		break;
-
-	/*
-	 * now, no suspend/resume supported. we'll do it in the future.
-	 */
-	case DDI_RESUME:
-		ATRACE("audioixp_attach() DDI_RESUME", NULL);
-		audio_sup_log(NULL, CE_WARN,
-		    "%s%d: audioixp_attach() resume is not supported yet",
-		    audioixp_name, instance);
-		return (DDI_FAILURE);
-
-	default:
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_attach() unknown command: 0x%x",
-		    audioixp_name, instance, cmd);
-		return (DDI_FAILURE);
-	}
-
-	/* we don't support high level interrupts in the driver */
-	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_attach()"
-		    " unsupported high level interrupt",
-		    audioixp_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	/* allocate the soft state structure */
-	if (ddi_soft_state_zalloc(audioixp_statep, instance) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_attach() soft state allocate failed",
-		    audioixp_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	if ((statep = ddi_get_soft_state(audioixp_statep, instance)) ==
-	    NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_attach() soft state failed",
-		    audioixp_name, instance);
-		goto error_state;
-	}
-
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-	if ((statep->audio_handle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_attach() audio_sup_register() failed",
-		    audioixp_name, instance);
-		goto error_state;
-	}
-
-	/* save private state */
-	audio_sup_set_private(statep->audio_handle, statep);
-
-	if ((audioixp_init_state(statep, dip)) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() init state structure failed");
-		goto error_audiosup;
-	}
-
-	/* map in the registers, allocate DMA buffers, etc. */
-	if (audioixp_map_regs(dip, statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() couldn't map registers");
-		goto error_destroy;
-	}
-
-	/* set PCI command register */
-	cmdeg = pci_config_get16(statep->pci_conf_handle, PCI_CONF_COMM);
-	pci_config_put16(statep->pci_conf_handle, PCI_CONF_COMM,
-	    cmdeg | PCI_COMM_IO | PCI_COMM_MAE);
-
-	if (audioixp_alloc_sample_buf(statep, IXP_DMA_PCM_OUT,
-	    statep->play_buf_size) == AUDIO_FAILURE) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() couldn't allocate play sample "
-		    "buffers");
-		goto error_unmap;
-	}
-
-	if (audioixp_alloc_sample_buf(statep, IXP_DMA_PCM_IN,
-	    statep->record_buf_size) == AUDIO_FAILURE) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() couldn't allocate record sample "
-		    "buffers");
-		goto error_dealloc_play;
-	}
-
-	audioixp_setup_bdl(statep);
-
-	/* set up kernel statistics */
-	if ((statep->ixp_ksp = kstat_create(IXP_NAME, instance,
-	    IXP_NAME, "controller", KSTAT_TYPE_INTR, 1,
-	    KSTAT_FLAG_PERSISTENT)) != NULL) {
-		kstat_install(statep->ixp_ksp);
-	}
-
-	/* set up the interrupt handler */
-	if (ddi_add_intr(dip, 0, &statep->intr_iblock,
-	    (ddi_idevice_cookie_t *)NULL, audioixp_intr, (caddr_t)statep) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() bad interrupt specification ");
-		goto error_kstat;
-	}
-
-	if (audioixp_chip_init(statep, IXP_INIT_NO_RESTORE) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() failed to init chip");
-		goto error_intr;
-	}
-
-	/* call the mixer attach() routine */
-	if (am_attach(statep->audio_handle, cmd, &statep->ad_info) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_attach() am_attach() failed");
-		goto error_intr;
-	}
-
-	ddi_report_dev(dip);
-
-	return (DDI_SUCCESS);
-
-error_intr:
-	ddi_remove_intr(dip, 0, statep->intr_iblock);
-
-error_kstat:
-	if (statep->ixp_ksp) {
-		kstat_delete(statep->ixp_ksp);
-	}
-
-error_dealloc:
-	audioixp_free_sample_buf(statep, &statep->record_buf);
-
-error_dealloc_play:
-	audioixp_free_sample_buf(statep, &statep->play_buf);
-
-error_unmap:
-	audioixp_unmap_regs(statep);
-
-error_destroy:
-	ATRACE("audioixp_attach() error_destroy", statep);
-	mutex_destroy(&statep->inst_lock);
-
-error_audiosup:
-	ATRACE("audioixp_attach() error_audiosup", statep);
-	(void) audio_sup_unregister(statep->audio_handle);
-
-error_state:
-	ATRACE("audioixp_attach() error_state", statep);
-	ddi_soft_state_free(audioixp_statep, instance);
-
-	ATRACE("audioixp_attach() returning failure", NULL);
-
-	return (DDI_FAILURE);
-
-}	/* audioixp_attach() */
-
-/*
- * audioixp_detach()
- *
- * Description:
- *	Detach an instance of the audioixp driver. After the Codec is
- *	detached, we call am_detach() and audio_sup_register() so they may
- *	do their work.
- *
- * Arguments:
- *	dev_info_t		*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t	cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS	The driver was detached
- *	DDI_FAILURE	The driver couldn't be detached
- */
-static int
-audioixp_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
-{
-	audioixp_state_t	*statep;
-	int			instance;
-
-	instance = ddi_get_instance(dip);
-
-	ATRACE_32("audioixp_detach() instance", instance);
-	ATRACE("audioixp_detach() audioixp_statep",
-	    audioixp_statep);
-
-	if ((statep = ddi_get_soft_state(audioixp_statep, instance)) ==
-	    NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: audioixp_detach() get soft state failed",
-		    audioixp_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-
-	/*
-	 * now, no suspend/resume supported. we'll do it in the future.
-	 */
-	case DDI_SUSPEND:
-		ATRACE("audioixp_detach() SUSPEND", statep);
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "audioixp_detach() suspend is not supported yet");
-		return (DDI_FAILURE);
-
-	default:
-		ATRACE("audioixp_detach() unknown command", cmd);
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_detach() unknown command: 0x%x", cmd);
-		return (DDI_FAILURE);
-	}
-
-	audioixp_chip_fini(statep);
-
-	/* stop DMA engines */
-	mutex_enter(&statep->inst_lock);
-	audioixp_stop_dma(statep, AUDIO_PLAY);
-	audioixp_stop_dma(statep, AUDIO_RECORD);
-	mutex_exit(&statep->inst_lock);
-
-	/* remove the interrupt handler */
-	ddi_remove_intr(dip, 0, statep->intr_iblock);
-
-	/* free DMA memory */
-	audioixp_free_sample_buf(statep, &statep->play_buf);
-	audioixp_free_sample_buf(statep, &statep->record_buf);
-
-	/* free the kernel statistics structure */
-	if (statep->ixp_ksp) {
-		kstat_delete(statep->ixp_ksp);
-	}
-
-	/* detach audio mixer */
-	(void) am_detach(statep->audio_handle, cmd);
-
-	/*
-	 * call the audio support module's detach routine to remove this
-	 * driver completely from the audio driver architecture.
-	 */
-	(void) audio_sup_unregister(statep->audio_handle);
-
-	mutex_destroy(&statep->inst_lock);
-
-	audioixp_unmap_regs(statep);
-
-	ddi_soft_state_free(audioixp_statep, instance);
-
-	return (DDI_SUCCESS);
-
-}	/* audioixp_detach */
-
-/*
- * audioixp_intr()
- *
- * Description:
- *	Interrupt service routine for both play and record. For play we
- *	get the next buffers worth of audio. For record we send it on to
- *	the mixer.
- *
- *	There's a hardware pointer which indicate memory location where
- *	the hardware is processing. We check this pointer to decide whether
- *	to handle the buffer and how many buffers should be handled.
- *	Refer to ATI IXP400/450 Register Reference Manual, page 193,194.
- *
- * Arguments:
- *	caddr_t		arg	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-audioixp_intr(caddr_t arg)
-{
-	audioixp_state_t	*statep;
-	uint32_t		sr;
-	int			intr_claimed = DDI_INTR_UNCLAIMED;
-
-	statep = (audioixp_state_t *)arg;
-	mutex_enter(&statep->inst_lock);
-
-	sr = IXP_AM_GET32(IXP_AUDIO_INT);
-
-	/* PCM in interrupt */
-	if (sr & IXP_AUDIO_INT_IN_DMA) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_IN_DMA);
-
-		if (statep->flags & IXP_DMA_RECD_STARTED) {
-			audioixp_reclaim_record_buf(statep);
-		}
-	}
-
-	/* PCM out interrupt */
-	if (sr & IXP_AUDIO_INT_OUT_DMA) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_OUT_DMA);
-
-		if (statep->flags & IXP_DMA_PLAY_STARTED) {
-			audioixp_reclaim_play_buf(statep);
-			(void) audioixp_fill_play_buf(statep);
-		}
-	}
-
-	/* system is too busy to process the input stream, ignore it */
-	if (sr & IXP_AUDIO_INT_IN_DMA_OVERFLOW) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_IN_DMA_OVERFLOW);
-	}
-
-	/* System is too busy, ignore it */
-	if (sr & IXP_AUDIO_INT_OUT_DMA_UNDERFLOW) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_OUT_DMA_UNDERFLOW);
-	}
-
-	if (sr & IXP_AUDIO_INT_CODEC0_NOT_READY) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_CODEC0_NOT_READY);
-		statep -> ixp_codec_not_ready_bits |=
-		    IXP_AUDIO_INT_CODEC0_NOT_READY;
-	}
-
-	if (sr & IXP_AUDIO_INT_CODEC1_NOT_READY) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_CODEC1_NOT_READY);
-		statep -> ixp_codec_not_ready_bits |=
-		    IXP_AUDIO_INT_CODEC1_NOT_READY;
-	}
-
-	if (sr & IXP_AUDIO_INT_CODEC2_NOT_READY) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_CODEC2_NOT_READY);
-		statep -> ixp_codec_not_ready_bits |=
-		    IXP_AUDIO_INT_CODEC2_NOT_READY;
-	}
-
-	if (sr & IXP_AUDIO_INT_NEW_FRAME) {
-		intr_claimed = DDI_INTR_CLAIMED;
-		IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AUDIO_INT_NEW_FRAME);
-		statep -> ixp_codec_not_ready_bits |= IXP_AUDIO_INT_NEW_FRAME;
-	}
-
-	if (intr_claimed == DDI_INTR_UNCLAIMED) {
-		mutex_exit(&statep->inst_lock);
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	/* update the kernel interrupt statistics */
-	if (statep->ixp_ksp) {
-		IXP_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
-	}
-
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE("audioixp_intr() done", statep);
-
-	return (DDI_INTR_CLAIMED);
-
-}	/* audioixp_intr() */
-
-/* *********************** Mixer Entry Point Routines ******************* */
-/*
- * audioixp_ad_set_config()
- *
- * Description:
- *	This routine is used to set new Codec parameters, except the data
- *	format which has it's own routine. If the Codec doesn't support a
- *	particular parameter and it is asked to set it then we return
- *	AUDIO_FAILURE.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		command		The configuration to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		Argument #1
- *	int		arg2		Argument #2, not always needed
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set,
- *				or the parameter couldn't be set
- */
-static int
-audioixp_ad_set_config(audiohdl_t ahandle, int stream, int command,
-	int dir, int arg1, int arg2)
-{
-	audioixp_state_t	*statep;
-	int 			rc = AUDIO_SUCCESS;
-
-	ATRACE_32("audioixp_ad_set_config() stream", stream);
-	ATRACE_32("audioixp_ad_set_config() command", command);
-	ATRACE_32("audioixp_ad_set_config() dir", dir);
-	ATRACE_32("audioixp_ad_set_config() arg1", arg1);
-	ATRACE_32("audioixp_ad_set_config() arg2", arg2);
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, of balance on gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	arg1 --> gain
-		 * 	arg2 --> channel #, 0 == left, 1 == right
-		 */
-		rc = audioixp_set_gain(statep, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of in ports, as well as which ports
-		 * are modifiable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 * 	arg1 --> port bit pattern
-		 * 	arg2 --> not used
-		 */
-		rc = audioixp_set_port(statep, dir, arg1);
-		break;
-
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	dir ---> N/A
-		 *	arg1 --> gain
-		 * 	arg2 --> not used
-		 */
-		rc = audioixp_set_monitor_gain(statep, arg1);
-		break;
-
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == mute, 0 == enable
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* mute */
-			if (statep->swap_out == B_FALSE) {
-				(void) audioixp_or_ac97(statep,
-				    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-			} else {
-				(void) audioixp_or_ac97(statep,
-				    AC97_EXTENDED_LRS_VOLUME_REGISTER,
-				    AD1980_SURR_MUTE);
-			}
-			(void) audioixp_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, HPVR_MUTE);
-			(void) audioixp_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MMVR_MUTE);
-
-		} else {	/* not muted */
-
-			/* by setting the port we unmute only active ports */
-			(void) audioixp_set_port(statep,
-			    AUDIO_PLAY, statep->ixp_output_port);
-		}
-		break;
-
-	case AM_MIC_BOOST:
-		/*
-		 * Enable or disable the mic's 20 dB boost preamplifier.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == enable, 0 == disabled
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* enable */
-			(void) audioixp_or_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
-			statep->ad_info.ad_add_mode |= AM_ADD_MODE_MIC_BOOST;
-		} else {	/* disable */
-			(void) audioixp_and_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER,
-			    (uint16_t)~MICVR_20dB_BOOST);
-			statep->ad_info.ad_add_mode &=
-			    ~AM_ADD_MODE_MIC_BOOST;
-		}
-		break;
-
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 *
-		 * AM_SET_GAIN_BAL
-		 * AM_SET_MONO_MIC
-		 * AM_BASS_BOOST
-		 * AM_MID_BOOST
-		 * AM_TREBLE_BOOST
-		 * AM_LOUDNESS
-		 */
-		rc = AUDIO_FAILURE;
-		ATRACE_32("audioixp_ad_set_config() unsupported command",
-		    command);
-		break;
-	}
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE_32("audioixp_ad_set_config() returning", rc);
-
-	return (rc);
-
-}	/* audioixp_ad_set_config() */
-
-/*
- * audioixp_ad_set_format()
- *
- * Description:
- *	This routine is used to set a new audio control data format.
- *
- * Arguments:
- * 	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		sample_rate	Data sample rate
- *	int		channels	Number of channels, 2
- *	int		precision	Bits per sample, 16
- *	int		encoding	Encoding method, linear
- *
- * Returns:
- *	AUDIO_SUCCESS	The Codec data format has been set
- *	AUDIO_FAILURE	The Codec data format has not been set, or the
- *			data format couldn't be set
- */
-static int
-audioixp_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-	int sample_rate, int channels, int precision, int encoding)
-{
-	audioixp_state_t	*statep;
-	uint16_t		val;
-	uint32_t		slot;
-	uint32_t		cmd;
-
-	ASSERT(precision == AUDIO_PRECISION_16);
-	ASSERT(channels == AUDIO_CHANNELS_STEREO);
-	ASSERT(encoding == AUDIO_ENCODING_LINEAR);
-
-	ATRACE_32("audioixp_ad_set_format() stream", stream);
-	ATRACE_32("audioixp_ad_set_format() dir", dir);
-	ATRACE_32("audioixp_ad_set_format() sample_rate", sample_rate);
-	ATRACE_32("audioixp_ad_set_format() channels", channels);
-	ATRACE_32("audioixp_ad_set_format() precision", precision);
-	ATRACE_32("audioixp_ad_set_format() encoding", encoding);
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->var_sr == B_FALSE) {
-		/* codec doesn't support variable sample rate */
-
-		if (sample_rate != IXP_SAMPR48000) {
-			audio_sup_log(statep->audio_handle, CE_NOTE,
-			    "!audioixp_ad_set_format() bad sample"
-			    " rate %d\n", sample_rate);
-			mutex_exit(&statep->inst_lock);
-			return (AUDIO_FAILURE);
-		}
-	} else {
-		switch (sample_rate) {
-		case IXP_SAMPR8000:	break;
-		case IXP_SAMPR11025:	break;
-		case IXP_SAMPR16000:	break;
-		case IXP_SAMPR22050:	break;
-		case IXP_SAMPR24000:	break;
-		case IXP_SAMPR32000:	break;
-		case IXP_SAMPR44100:	break;
-		case IXP_SAMPR48000:	break;
-		default:
-			ATRACE_32("audioixp_ad_set_format() bad SR",
-			    sample_rate);
-			mutex_exit(&statep->inst_lock);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	if (dir == AUDIO_PLAY) {
-
-		(void) audioixp_write_ac97(statep,
-		    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, sample_rate);
-
-		/*
-		 * Some codecs before ac97 2.2, such as YMF753 produced by
-		 * Yamaha LSI, don't have the AC'97 registers indexed range
-		 * from 0x2c to 0x34. So we assume this kind of codec
-		 * supports fixed 48k sample rate.
-		 */
-		if (statep->var_sr == B_TRUE) {
-			(void) audioixp_read_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, &val);
-			if (val != sample_rate) {
-				ATRACE_32("audioixp_ad_set_format()"
-				    " bad out SR", sample_rate);
-				audio_sup_log(statep->audio_handle, CE_NOTE,
-				    "!audioixp_ad_set_format() bad out"
-				    " SR %d\n", sample_rate);
-				mutex_exit(&statep->inst_lock);
-				return (AUDIO_FAILURE);
-			}
-		}
-
-		slot = IXP_AM_GET32(IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD);
-		slot |= IXP_AUDIO_OUT_DMA_SLOT_3
-		    | IXP_AUDIO_OUT_DMA_SLOT_4;
-		slot &= ~ (IXP_AUDIO_OUT_DMA_SLOT_5
-		    |IXP_AUDIO_OUT_DMA_SLOT_6
-		    |IXP_AUDIO_OUT_DMA_SLOT_7
-		    |IXP_AUDIO_OUT_DMA_SLOT_8
-		    |IXP_AUDIO_OUT_DMA_SLOT_9
-		    |IXP_AUDIO_OUT_DMA_SLOT_10
-		    |IXP_AUDIO_OUT_DMA_SLOT_11
-		    |IXP_AUDIO_OUT_DMA_SLOT_12);
-
-		IXP_AM_PUT32(IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD, slot);
-
-		cmd = IXP_AM_GET32(IXP_AUDIO_CMD);
-		cmd |= IXP_AUDIO_CMD_INTER_OUT;
-		IXP_AM_PUT32(IXP_AUDIO_CMD, cmd);
-
-		statep->ixp_psample_rate = sample_rate;
-		statep->ixp_pchannels = channels;
-		statep->ixp_pprecision = precision;
-	} else {
-		(void) audioixp_write_ac97(statep,
-		    AC97_EXTENDED_LR_DAC_RATE_REGISTER, sample_rate);
-
-		/*
-		 * Some codecs before ac97 2.2, such as YMF753 produced by
-		 * Yamaha LSI, don't have the AC'97 registers indexed range
-		 * from 0x2c to 0x34. So we assume this kind of codec
-		 * supports fixed 48k sample rate.
-		 */
-		if (statep->var_sr == B_TRUE) {
-			(void) audioixp_read_ac97(statep,
-			    AC97_EXTENDED_LR_DAC_RATE_REGISTER, &val);
-			if (val != sample_rate) {
-				ATRACE_32("audioixp_ad_set_format() bad"
-				    " in SR", sample_rate);
-				audio_sup_log(statep->audio_handle, CE_NOTE,
-				    "!audioixp_ad_set_format() bad in"
-				    " SR %d\n", sample_rate);
-				mutex_exit(&statep->inst_lock);
-				return (AUDIO_FAILURE);
-			}
-		}
-
-		cmd = IXP_AM_GET32(IXP_AUDIO_CMD);
-		cmd |= IXP_AUDIO_CMD_INTER_IN;
-		IXP_AM_PUT32(IXP_AUDIO_CMD, cmd);
-
-		statep->ixp_csample_rate = sample_rate;
-		statep->ixp_cchannels = channels;
-		statep->ixp_cprecision = precision;
-	}
-
-done:
-	mutex_exit(&statep->inst_lock);
-
-	ATRACE_32("audioixp_ad_set_format() returning success", 0);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_ad_set_format() */
-
-/*
- * audioixp_ad_start_play()
- *
- * Description:
- *	This routine starts the playback DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Play not started/restarted, no audio to play
- */
-static int
-audioixp_ad_start_play(audiohdl_t ahandle, int stream)
-{
-	audioixp_state_t	*statep;
-	int			rc = AUDIO_SUCCESS;
-
-
-	ATRACE_32("audioixp_ad_start_play() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->flags & IXP_DMA_PLAY_PAUSED) {
-		statep->flags |= IXP_DMA_PLAY_STARTED;
-		statep->flags &= ~IXP_DMA_PLAY_PAUSED;
-		audioixp_start_dma(statep, AUDIO_PLAY);
-		IXP_AM_UPDATE32(
-		    IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_OUT,
-		    IXP_AUDIO_CMD_EN_OUT);
-		goto done;
-	}
-
-	if (statep->flags & IXP_DMA_PLAY_STARTED) {
-		goto done;
-	}
-
-	audioixp_start_dma(statep, AUDIO_PLAY);
-	rc = audioixp_fill_play_buf(statep);
-	if (rc == AUDIO_SUCCESS) {
-		statep->flags |= IXP_DMA_PLAY_STARTED;
-	}
-
-done:
-	mutex_exit(&statep->inst_lock);
-	return (rc);
-
-}	/* audioixp_ad_start_play() */
-
-/*
- * audioixp_ad_pause_play()
- *
- * Description:
- *	This routine pauses the play DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- * 	void
- */
-static void
-audioixp_ad_pause_play(audiohdl_t ahandle, int stream)
-{
-	audioixp_state_t	*statep;
-
-	statep = audio_sup_get_private(ahandle);
-
-	ASSERT(statep);
-	ATRACE("audioixp_ad_pause_play() ", ahandle);
-	ATRACE_32("audioixp_ad_pause_play() stream", stream);
-
-	mutex_enter(&statep->inst_lock);
-
-	if ((statep->flags & IXP_DMA_PLAY_STARTED) == 0) {
-		mutex_exit(&statep->inst_lock);
-		return;
-	}
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT, 0);
-	statep->flags |= IXP_DMA_PLAY_PAUSED;
-
-	mutex_exit(&statep->inst_lock);
-
-}	/* audioixp_ad_pause_play() */
-
-/*
- * audioixp_ad_stop_play()
- *
- * Description:
- *	This routine stops the playback DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- *	void
- */
-static void
-audioixp_ad_stop_play(audiohdl_t ahandle, int stream)
-{
-	audioixp_state_t		*statep;
-	audioixp_sample_buf_t		*buf;
-
-	ATRACE("audioixp_ad_stop_play() ", ahandle);
-	ATRACE_32("audioixp_ad_stop_play() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_OUT, 0);
-	audioixp_stop_dma(statep, AUDIO_PLAY);
-
-	buf = &statep->play_buf;
-	buf->io_started = B_FALSE;
-	statep->flags &= ~(IXP_DMA_PLAY_STARTED
-	    | IXP_DMA_PLAY_PAUSED | IXP_DMA_PLAY_EMPTY);
-
-	mutex_exit(&statep->inst_lock);
-
-}	/* audioixp_ad_stop_play() */
-
-/*
- * audioixp_ad_start_record()
- *
- * Description:
- *	This routine starts the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which isn't going to apply for record
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Record not started
- */
-static int
-audioixp_ad_start_record(audiohdl_t ahandle, int stream)
-{
-	audioixp_state_t	*statep;
-	int			rc;
-
-	ATRACE("audioixp_ad_start_record() ", ahandle);
-	ATRACE_32("audioixp_ad_start_record() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-
-	if (statep->flags & IXP_DMA_RECD_STARTED) {
-		mutex_exit(&statep->inst_lock);
-		return (AUDIO_SUCCESS);
-	}
-
-	audioixp_start_dma(statep, AUDIO_RECORD);
-	rc = audioixp_prepare_record_buf(statep);
-	if (rc == AUDIO_SUCCESS) {
-		statep->flags |= IXP_DMA_RECD_STARTED;
-	}
-
-	mutex_exit(&statep->inst_lock);
-
-	return (rc);
-
-}	/* audioixp_ad_start_record() */
-
-/*
- * audioixp_ad_stop_record()
- *
- * Description:
- *	This routine stops the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream
- *					Codecs, which isn't going to apply
- *					for record
- *
- * Returns:
- *	void
- */
-static void
-audioixp_ad_stop_record(audiohdl_t ahandle, int stream)
-{
-	audioixp_state_t		*statep;
-	audioixp_sample_buf_t		*buf;
-
-	ATRACE("audioixp_ad_stop_record() ", ahandle);
-	ATRACE_32("audioixp_ad_stop_record() stream", stream);
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->inst_lock);
-	statep->flags &= ~IXP_DMA_RECD_STARTED;
-
-	buf = &statep->record_buf;
-	buf->io_started = B_FALSE;
-
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_EN_IN, 0);
-	audioixp_stop_dma(statep, AUDIO_RECORD);
-
-	mutex_exit(&statep->inst_lock);
-
-}	/* audioixp_ad_stop_record() */
-
-/* *********************** Local Routines *************************** */
-
-/*
- * audioixp_init_state()
- *
- * Description:
- *	This routine initializes the audio driver's state structure
- *
- *	CAUTION: This routine cannot allocate resources, unless it frees
- *		them before returning for an error. Also, error_destroy:
- *		in audioixp_attach() would need to be fixed as well.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	dev_info_t		*dip		Pointer to the device's
- *						dev_info struct
- *
- * Returns:
- *	AUDIO_SUCCESS		State structure initialized
- *	AUDIO_FAILURE		State structure not initialized
- */
-static int
-audioixp_init_state(audioixp_state_t *statep, dev_info_t *dip)
-{
-	int rints;
-	int pints;
-	int cdrom;
-	int mode;
-
-	ATRACE("audioixp_init_state()", NULL);
-
-	statep->dip = dip;
-	statep->vol_bits_mask = 5;
-
-	cdrom = ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "cdrom", 0);
-
-	/* get the mode from the .conf file */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "mixer-mode", 1)) {
-		mode = AM_MIXER_MODE;
-	} else {
-		mode = AM_COMPAT_MODE;
-	}
-
-	pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "play-interrupts", IXP_INTS);
-	if (pints > IXP_MAX_INTS) {
-		ATRACE_32("audioixp_init_state() "
-		    "play interrupt rate too high, resetting", pints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "audioixp_init_state() "
-		    "play interrupt rate set too high, %d, resetting to %d",
-		    pints, IXP_INTS);
-		pints = IXP_INTS;
-	} else if (pints < IXP_MIN_INTS) {
-		ATRACE_32("audioixp_init_state() "
-		    "play interrupt rate too low, resetting", pints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "audioixp_init_state() "
-		    "play interrupt rate set too low, %d, resetting to %d",
-		    pints, IXP_INTS);
-		pints = IXP_INTS;
-	}
-	rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "record-interrupts", IXP_INTS);
-	if (rints > IXP_MAX_INTS) {
-		ATRACE_32("audioixp_init_state() "
-		    "record interrupt rate too high, resetting", rints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "audioixp_init_state() "
-		    "record interrupt rate set too high, %d, resetting to "
-		    "%d",
-		    rints, IXP_INTS);
-		rints = IXP_INTS;
-	} else if (rints < IXP_MIN_INTS) {
-		ATRACE_32("audioixp_init_state() "
-		    "record interrupt rate too low, resetting", rints);
-		audio_sup_log(statep->audio_handle, CE_NOTE,
-		    "audioixp_init_state() "
-		    "record interrupt rate set too low, %d, resetting to "
-		    "%d",
-		    rints, IXP_INTS);
-		rints = IXP_INTS;
-	}
-
-	/* fill in the device default state */
-	statep->ixp_defaults.play.sample_rate = IXP_DEFAULT_SR;
-	statep->ixp_defaults.play.channels = IXP_DEFAULT_CH;
-	statep->ixp_defaults.play.precision = IXP_DEFAULT_PREC;
-	statep->ixp_defaults.play.encoding = IXP_DEFAULT_ENC;
-	statep->ixp_defaults.play.gain = IXP_DEFAULT_PGAIN;
-	statep->ixp_defaults.play.port =
-	    AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->ixp_defaults.play.avail_ports =
-	    AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->ixp_defaults.play.mod_ports =
-	    AUDIO_SPEAKER | AUDIO_LINE_OUT;
-	statep->ixp_defaults.play.buffer_size = IXP_BSIZE;
-	statep->ixp_defaults.play.balance = IXP_DEFAULT_BAL;
-
-	statep->ixp_defaults.record.sample_rate = IXP_DEFAULT_SR;
-	statep->ixp_defaults.record.channels = IXP_DEFAULT_CH;
-	statep->ixp_defaults.record.precision = IXP_DEFAULT_PREC;
-	statep->ixp_defaults.record.encoding = IXP_DEFAULT_ENC;
-	statep->ixp_defaults.record.gain = IXP_DEFAULT_PGAIN;
-	statep->ixp_defaults.record.port = AUDIO_MICROPHONE;
-	statep->ixp_defaults.record.avail_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->ixp_defaults.record.mod_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->ixp_defaults.record.buffer_size = IXP_BSIZE;
-	statep->ixp_defaults.record.balance = IXP_DEFAULT_BAL;
-
-	statep->ixp_defaults.monitor_gain = IXP_DEFAULT_MONITOR_GAIN;
-	statep->ixp_defaults.output_muted = B_FALSE;
-	statep->ixp_defaults.ref_cnt = B_FALSE;
-	statep->ixp_defaults.hw_features =
-	    AUDIO_HWFEATURE_DUPLEX | AUDIO_HWFEATURE_PLAY |
-	    AUDIO_HWFEATURE_IN2OUT | AUDIO_HWFEATURE_RECORD;
-	statep->ixp_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	if (cdrom) {
-		statep->ixp_defaults.record.avail_ports |= AUDIO_CD;
-		statep->ixp_defaults.record.mod_ports |= AUDIO_CD;
-	}
-
-	statep->ixp_psample_rate =
-	    statep->ixp_defaults.play.sample_rate;
-	statep->ixp_pchannels =
-	    statep->ixp_defaults.play.channels;
-	statep->ixp_pprecision =
-	    statep->ixp_defaults.play.precision;
-	statep->ixp_csample_rate =
-	    statep->ixp_defaults.record.sample_rate;
-	statep->ixp_cchannels =
-	    statep->ixp_defaults.record.channels;
-	statep->ixp_cprecision =
-	    statep->ixp_defaults.record.precision;
-
-	/*
-	 * fill in the ad_info structure
-	 */
-	statep->ad_info.ad_mode = mode;
-	statep->ad_info.ad_int_vers = AM_VERSION;
-	statep->ad_info.ad_add_mode = NULL;
-	statep->ad_info.ad_codec_type = AM_TRAD_CODEC;
-	statep->ad_info.ad_defaults = &statep->ixp_defaults;
-	statep->ad_info.ad_play_comb = audioixp_combinations;
-	statep->ad_info.ad_rec_comb = audioixp_combinations;
-	statep->ad_info.ad_entry = &audioixp_entry;
-	statep->ad_info.ad_dev_info = &statep->ixp_dev_info;
-	statep->ad_info.ad_diag_flags = AM_DIAG_INTERNAL_LOOP;
-	statep->ad_info.ad_diff_flags =
-	    AM_DIFF_SR | AM_DIFF_CH | AM_DIFF_PREC | AM_DIFF_ENC;
-	statep->ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	statep->ad_info.ad_misc_flags = AM_MISC_RP_EXCL | AM_MISC_MONO_DUP;
-	statep->ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	statep->ad_info.ad_play.ad_mixer_srs =
-	    audioixp_mixer_sample_rates;
-	statep->ad_info.ad_play.ad_compat_srs =
-	    audioixp_compat_sample_rates;
-	statep->ad_info.ad_play.ad_conv = &am_src2;
-	statep->ad_info.ad_play.ad_sr_info = NULL;
-	statep->ad_info.ad_play.ad_chs = audioixp_channels;
-	statep->ad_info.ad_play.ad_int_rate = pints;
-	statep->ad_info.ad_play.ad_max_chs = IXP_MAX_OUT_CHANNELS;
-	statep->ad_info.ad_play.ad_bsize = IXP_BSIZE;
-
-	/* record capabilities */
-	statep->ad_info.ad_record.ad_mixer_srs =
-	    audioixp_mixer_sample_rates;
-	statep->ad_info.ad_record.ad_compat_srs =
-	    audioixp_compat_sample_rates;
-	statep->ad_info.ad_record.ad_conv = &am_src2;
-	statep->ad_info.ad_record.ad_sr_info = NULL;
-	statep->ad_info.ad_record.ad_chs = audioixp_channels;
-	statep->ad_info.ad_record.ad_int_rate = rints;
-	statep->ad_info.ad_record.ad_max_chs = IXP_MAX_CHANNELS;
-	statep->ad_info.ad_record.ad_bsize = IXP_BSIZE;
-
-	if (ddi_get_iblock_cookie(dip, (uint_t)0, &statep->intr_iblock) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_init_state() cannot get iblock cookie");
-		return (AUDIO_FAILURE);
-	}
-	mutex_init(&statep->inst_lock, NULL, MUTEX_DRIVER, statep->intr_iblock);
-
-	/* fill in device info strings */
-	(void) strcpy(statep->ixp_dev_info.name, IXP_DEV_NAME);
-	(void) strcpy(statep->ixp_dev_info.config, IXP_DEV_CONFIG);
-	(void) strcpy(statep->ixp_dev_info.version, IXP_DEV_VERSION);
-
-	statep->play_buf_size = IXP_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / pints;
-	statep->play_buf_size += IXP_MOD_SIZE -
-	    (statep->play_buf_size % IXP_MOD_SIZE);
-	statep->record_buf_size = IXP_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / rints;
-	statep->record_buf_size += IXP_MOD_SIZE - 1;
-	statep->record_buf_size -= statep->record_buf_size % IXP_MOD_SIZE;
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_init_state() */
-
-/*
- * audioixp_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. Finally, the registers are mapped in.
- *
- *	CAUTION: Make sure all errors call audio_sup_log().
- *
- * Arguments:
- *	dev_info_t		*dip		Pointer to the device's devinfo
- *	audioixp_state_t	*state		  The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-audioixp_map_regs(dev_info_t *dip, audioixp_state_t *statep)
-{
-	ddi_dma_cookie_t	cookie;
-	uint_t			count;
-
-	ATRACE("audioixp_map_regs()", statep);
-
-	statep->ixp_res_flags = 0;
-
-	/* map PCI config space */
-	if (pci_config_setup(statep->dip, &statep->pci_conf_handle) ==
-	    DDI_FAILURE) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() configuration "
-		    "memory mapping failed");
-		goto error;
-	}
-	statep->ixp_res_flags |= IXP_RS_PCI_REGS;
-
-	/* map audio mixer register */
-	if ((ddi_regs_map_setup(dip, IXP_IO_AM_REGS,
-	    (caddr_t *)&statep->am_regs_base, 0, 0,
-	    &dev_attr, &statep->am_regs_handle)) != DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() audio mixer "
-		    "memory mapping failed");
-		goto error;
-	}
-	statep->ixp_res_flags |= IXP_RS_AM_REGS;
-
-	/*
-	 * now, from here we allocate DMA memory for buffer descriptor list.
-	 * we allocate adjacent DMA memory for all DMA engines.
-	 */
-	if (ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &statep->bdl_dma_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() ddi_dma_alloc_handle(bdlist)"
-		    " failed");
-		goto error;
-	}
-	statep->ixp_res_flags |= IXP_RS_DMA_BDL_HANDLE;
-
-	if (ddi_dma_mem_alloc(statep->bdl_dma_handle,
-	    sizeof (audioixp_bd_list_t), &dev_attr, DDI_DMA_CONSISTENT,
-	    DDI_DMA_SLEEP, NULL, (caddr_t *)&statep->bdl_virtual,
-	    &statep->bdl_size, &statep->bdl_acc_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() ddi_dma_mem_alloc(bdlist) "
-		    "failed");
-		goto error;
-	}
-	statep->ixp_res_flags |= IXP_RS_DMA_BDL_MEM;
-
-	if (ddi_dma_addr_bind_handle(statep->bdl_dma_handle, NULL,
-	    (caddr_t)statep->bdl_virtual, statep->bdl_size,
-	    DDI_DMA_RDWR|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &cookie,
-	    &count) != DDI_DMA_MAPPED) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() addr_bind_handle failed");
-		goto error;
-	}
-
-	/*
-	 * there are some bugs in the DDI framework and it is possible to
-	 * get multiple cookies
-	 */
-	if (count != 1) {
-		(void) ddi_dma_unbind_handle(statep->bdl_dma_handle);
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_map_regs() addr_bind_handle failed,"
-		    " cookies > 1");
-		goto error;
-	}
-
-	statep->bdl_phys =
-	    (audioixp_bd_list_t *)(long)(cookie.dmac_address);
-	statep->ixp_res_flags |= IXP_RS_DMA_BDL_BIND;
-
-	return (AUDIO_SUCCESS);
-
-error:
-	audioixp_unmap_regs(statep);
-
-	return (AUDIO_FAILURE);
-
-}	/* audioixp_map_regs() */
-
-/*
- * audioixp_unmap_regs()
- *
- * Description:
- *	This routine unbinds the play and record DMA handles, frees
- *	the DMA buffers and then unmaps control registers.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_unmap_regs(audioixp_state_t *statep)
-{
-	if (statep->ixp_res_flags & IXP_RS_DMA_BDL_BIND) {
-		statep->ixp_res_flags &= ~IXP_RS_DMA_BDL_BIND;
-		(void) ddi_dma_unbind_handle(statep->bdl_dma_handle);
-	}
-
-	if (statep->ixp_res_flags & IXP_RS_DMA_BDL_MEM) {
-		statep->ixp_res_flags &= ~IXP_RS_DMA_BDL_MEM;
-		ddi_dma_mem_free(&statep->bdl_acc_handle);
-	}
-
-	if (statep->ixp_res_flags & IXP_RS_DMA_BDL_HANDLE) {
-		statep->ixp_res_flags &= ~IXP_RS_DMA_BDL_HANDLE;
-		ddi_dma_free_handle(&statep->bdl_dma_handle);
-	}
-
-	if (statep->ixp_res_flags & IXP_RS_AM_REGS) {
-		statep->ixp_res_flags &= ~IXP_RS_AM_REGS;
-		ddi_regs_map_free(&statep->am_regs_handle);
-	}
-
-	if (statep->ixp_res_flags & IXP_RS_PCI_REGS) {
-		statep->ixp_res_flags &= ~IXP_RS_PCI_REGS;
-		pci_config_teardown(&statep->pci_conf_handle);
-	}
-
-}	/* audioixp_unmap_regs() */
-
-/*
- * audioixp_alloc_sample_buf()
- *
- * Description:
- *	This routine allocates DMA buffers for the sample buffer. It
- *	allocates two DMA chunks (buffers) to the specified DMA engine
- *	(sample buffer structure). The two data chunks will be bound
- *	to the buffer descriptor entries of corresponding buffer
- *	descriptor list, and be used to transfer audio sample data to
- *	and from the audio controller.
- *
- * Arguments:
- *	audioixp_state_t	*state	The device's state structure
- *	int			which	Which sample buffer, PCM in or PCM out
- *					IXP_DMA_PCM_IN ---PCM in DMA engine
- *					IXP_DMA_PCM_OUT---PCM out DMA engine
- *	int			len	The length of the DMA buffers
- *
- * Returns:
- *	AUDIO_SUCCESS	 Allocating DMA buffers successfully
- *	AUDIO_FAILURE	 Failed to allocate dma buffers
- */
-
-static int
-audioixp_alloc_sample_buf(audioixp_state_t *statep, int which, int len)
-{
-	audioixp_sample_buf_t	*buf;
-	audioixp_bdlist_chunk_t	*chunk;
-	ddi_dma_cookie_t		cookie;
-	uint_t				count;
-	int				i;
-
-	if (which == IXP_DMA_PCM_OUT) {
-		buf = &statep->play_buf;
-	} else {
-		ASSERT(which == IXP_DMA_PCM_IN);
-		buf = &statep->record_buf;
-	}
-
-	for (i = 0; i < 2; i++) {
-		chunk = &(buf->chunk[i]);
-
-		if (ddi_dma_alloc_handle(statep->dip, &sample_buf_dma_attr,
-		    DDI_DMA_SLEEP, NULL, &chunk->dma_handle) !=
-		    DDI_SUCCESS) {
-			goto error;
-		}
-
-		if (ddi_dma_mem_alloc(chunk->dma_handle, len, &dev_attr,
-		    DDI_DMA_STREAMING, DDI_DMA_SLEEP, NULL,
-		    &chunk->data_buf, &chunk->real_len,
-		    &chunk->acc_handle) != DDI_SUCCESS) {
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		if (ddi_dma_addr_bind_handle(chunk->dma_handle, NULL,
-		    chunk->data_buf, chunk->real_len, DDI_DMA_WRITE,
-		    DDI_DMA_SLEEP, NULL, &cookie, &count) !=
-		    DDI_DMA_MAPPED) {
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		/*
-		 * there some bugs in the DDI framework and it is possible to
-		 * get multiple cookies
-		 */
-		if (count != 1) {
-			(void) ddi_dma_unbind_handle(chunk->dma_handle);
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-			goto error;
-		}
-
-		chunk->addr_phy = (uint32_t)cookie.dmac_address;
-	}
-
-	return (AUDIO_SUCCESS);
-
-error:
-	if (i != 0) {
-		(void) ddi_dma_unbind_handle((buf->chunk[0].dma_handle));
-		ddi_dma_mem_free(&(buf->chunk[0].acc_handle));
-		ddi_dma_free_handle(&(buf->chunk[0].dma_handle));
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audioixp_alloc_sample_buf() */
-
-/*
- * audioixp_free_sample_buf()
- *
- * Description:
- *	This routine frees the DMA buffers of the sample buffer. The DMA
- *	buffers were allocated by calling audioixp_alloc_sample_buf().
- *
- * Arguments:
- *	audioixp_state_t		*state	The device's state structure
- *	audioixp_sample_buf_t	*buf	The sample buffer structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_free_sample_buf(audioixp_state_t *statep,
-	audioixp_sample_buf_t *buf)
-{
-	audioixp_bdlist_chunk_t	*chunk;
-	int 				i;
-
-	ATRACE("audioixp_free_sample_buf() audioixp_statep", statep);
-
-	for (i = 0; i < 2; i++) {
-		chunk = &(buf->chunk[i]);
-		(void) ddi_dma_unbind_handle(chunk->dma_handle);
-		ddi_dma_mem_free(&chunk->acc_handle);
-		chunk->acc_handle = 0;
-		ddi_dma_free_handle(&chunk->dma_handle);
-	}
-
-}	/* audioixp_free_sample_buf() */
-
-
-/*
- * audioixp_setup_bdl()
- *
- * Description:
- * 	This routine setup the buf descriptor list.
- *
- * Arguments:
- *	audioixp_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-
-static void audioixp_setup_bdl(audioixp_state_t *statep)
-{
-	int i;
-	audioixp_bd_entry_t *bd_p;
-
-	/* setup playback bdlist */
-	for (i = 0; i < IXP_BD_NUMS; i ++) {
-		bd_p = &(((audioixp_bd_list_t *)(statep->bdl_virtual))
-		    ->pcm_out[i]);
-		bd_p->buf_base = statep->play_buf.chunk[i&IXP_CHUNK_MASK]
-		    .addr_phy;
-		bd_p->status = 0;
-		bd_p->buf_len = 0;
-		bd_p->next = (uintptr_t)&(((audioixp_bd_list_t *)
-		    (statep->bdl_phys))->pcm_out[(i+1)%IXP_BD_NUMS]);
-	}
-
-	/* setup record bdlist */
-	for (i = 0; i < IXP_BD_NUMS; i ++) {
-		bd_p = &(((audioixp_bd_list_t *)(statep->bdl_virtual))
-		    ->pcm_in[i]);
-		bd_p->buf_base = statep->record_buf.chunk[i&IXP_CHUNK_MASK]
-		    .addr_phy;
-		bd_p->status = 0;
-		bd_p->buf_len = 0;
-		bd_p->next = (uintptr_t)&(((audioixp_bd_list_t *)
-		    (statep->bdl_phys))->pcm_in[(i+1)%IXP_BD_NUMS]);
-	}
-}	/* audioixp_setup_bdl() */
-
-/*
- * audioixp_start_dma()
- *
- * Description:
- *	This routine is used to put each DMA engine into working state.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_start_dma(audioixp_state_t *statep, int dir)
-{
-
-	ASSERT(dir == AUDIO_PLAY || dir == AUDIO_RECORD);
-
-	if (dir == AUDIO_PLAY) {
-		IXP_AM_PUT32(IXP_AUDIO_FIFO_FLUSH, IXP_AUDIO_FIFO_FLUSH_OUT);
-		IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_OUT_DMA,
-		    IXP_AUDIO_CMD_EN_OUT_DMA);
-	} else {
-		IXP_AM_PUT32(IXP_AUDIO_FIFO_FLUSH, IXP_AUDIO_FIFO_FLUSH_IN);
-		IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_IN_DMA,
-		    IXP_AUDIO_CMD_EN_IN_DMA);
-	}
-
-}	/* audioixp_start_dma() */
-
-/*
- * audioixp_stop_dma()
- *
- * Description:
- *	This routine is used to put each DMA engine into the quiet state.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_stop_dma(audioixp_state_t *statep, int dir)
-{
-
-	ASSERT(dir == AUDIO_PLAY || dir == AUDIO_RECORD);
-
-	IXP_AM_PUT32(IXP_AUDIO_FIFO_FLUSH, IXP_AUDIO_FIFO_FLUSH_IN);
-
-	if (dir == AUDIO_PLAY) {
-		IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_OUT_DMA,
-		    0);
-	} else {
-		IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_IN_DMA,
-		    0);
-	}
-
-}	/* audioixp_stop_dma() */
-
-
-/*
- * audioixp_codec_ready ()
- *
- * Description:
- *	This routine checks the state of codecs. This routine is called by
- *	chip_init before interrupt is enabled. It enables interrupt first,
- *	then waits a moment for interrupt handler to set the flag according
- *	to the hardware configuration. Then it checks the flag to confirm
- *	that primary codec is ready. The original value of interrupt enable
- *	register is restored.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS	 codec is ready
- *	AUDIO_FAILURE	 codec is not ready
- */
-static int
-audioixp_codec_ready(audioixp_state_t *statep)
-{
-	uint32_t	old_reg;
-
-	old_reg = IXP_AM_GET32(IXP_AUDIO_INT_EN);
-
-	IXP_AM_UPDATE32(IXP_AUDIO_INT_EN,
-	    IXP_AUDIO_INT_EN_CODEC0_NOT_READY
-	    | IXP_AUDIO_INT_EN_CODEC1_NOT_READY
-	    | IXP_AUDIO_INT_EN_CODEC2_NOT_READY,
-	    IXP_AUDIO_INT_EN_CODEC0_NOT_READY
-	    | IXP_AUDIO_INT_EN_CODEC1_NOT_READY
-	    | IXP_AUDIO_INT_EN_CODEC2_NOT_READY);
-
-	drv_usecwait(1000);
-	IXP_AM_PUT32(IXP_AUDIO_INT_EN, old_reg);
-
-	if (statep->ixp_codec_not_ready_bits & IXP_AUDIO_INT_CODEC0_NOT_READY) {
-		audio_sup_log(NULL, CE_WARN, "primary codec not ready");
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-}
-
-/*
- * audioixp_codec_sync()
- *
- * Description:
- *	Serialize access to the AC97 audio mixer registers.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Ready for an I/O access to the codec
- *	AUDIO_FAILURE		An I/O access is currently in progress, can't
- *				perform another I/O access.
- */
-static int
-audioixp_codec_sync(audioixp_state_t *statep)
-{
-	int 		i;
-	uint32_t	cmd;
-
-	for (i = 0; i < 300; i++) {
-		cmd = IXP_AM_GET32(IXP_AUDIO_OUT_PHY_ADDR_DATA);
-		if (!(cmd & IXP_AUDIO_OUT_PHY_EN)) {
-			return (AUDIO_SUCCESS);
-		}
-		drv_usecwait(10);
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audioixp_codec_sync() */
-
-
-/*
- * audioixp_read_ac97()
- *
- * Description:
- *	Get the specific AC97 Codec register. It also updates codec_shadow[]
- *	with the register value.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		*data		The data to be returned
- *
- * Returns:
- *	AUDIO_SUCCESS		Reading the codec register successfully
- *	AUDIO_FAILURE		Failed to read the register
- */
-static int
-audioixp_read_ac97(audioixp_state_t *statep, int reg, uint16_t *data)
-{
-	uint32_t	value;
-	uint32_t	result;
-	int		i;
-
-	if (audioixp_codec_sync(statep) != AUDIO_SUCCESS) {
-		*data = 0xffff;
-		return (AUDIO_FAILURE);
-	}
-
-	value = IXP_AUDIO_OUT_PHY_PRIMARY_CODEC
-	    | IXP_AUDIO_OUT_PHY_READ
-	    | IXP_AUDIO_OUT_PHY_EN
-	    | ((reg << IXP_AUDIO_OUT_PHY_ADDR_SHIFT)
-	    & IXP_AUDIO_OUT_PHY_ADDR_MASK);
-	IXP_AM_PUT32(IXP_AUDIO_OUT_PHY_ADDR_DATA, value);
-
-	if (audioixp_codec_sync(statep) != AUDIO_SUCCESS) {
-		*data = 0xffff;
-		return (AUDIO_FAILURE);
-	}
-
-	for (i = 0; i < 300; i++) {
-		result = IXP_AM_GET32(IXP_AUDIO_IN_PHY_ADDR_DATA);
-		if (result & IXP_AUDIO_IN_PHY_READY)	{
-			*data = (result & IXP_AUDIO_IN_PHY_DATA_MASK)
-			    >> IXP_AUDIO_IN_PHY_DATA_SHIFT;
-			statep->codec_shadow[IXP_CODEC_REG(reg)] = *data;
-			return (AUDIO_SUCCESS);
-		}
-		drv_usecwait(10);
-	}
-
-	*data = 0xffff;
-	return (AUDIO_FAILURE);
-
-}	/* audioixp_read_ac97() */
-
-/*
- * audioixp_write_ac97()
- *
- * Description:
- *	Set the specific AC97 Codec register.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The data want to be set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audioixp_write_ac97(audioixp_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t	tmp;
-	uint32_t	value;
-
-	if (audioixp_codec_sync(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	value = IXP_AUDIO_OUT_PHY_PRIMARY_CODEC
-	    | IXP_AUDIO_OUT_PHY_WRITE
-	    | IXP_AUDIO_OUT_PHY_EN
-	    | ((reg << IXP_AUDIO_OUT_PHY_ADDR_SHIFT)
-	    & IXP_AUDIO_OUT_PHY_ADDR_MASK)
-	    | ((data << IXP_AUDIO_OUT_PHY_DATA_SHIFT)
-	    & IXP_AUDIO_OUT_PHY_DATA_MASK);
-	IXP_AM_PUT32(IXP_AUDIO_OUT_PHY_ADDR_DATA, value);
-
-	(void) audioixp_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_write_ac97() */
-
-/*
- * audioixp_and_ac97()
- *
- * Description:
- *	Logically AND the value with the specified ac97 codec register
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to AND
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audioixp_and_ac97(audioixp_state_t *statep, int reg, uint16_t data)
-{
-	data &= statep->codec_shadow[IXP_CODEC_REG(reg)];
-	if (audioixp_write_ac97(statep, reg, data) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_and_ac97() */
-
-/*
- * audioixp_or_ac97()
- *
- * Description:
- *	Logically OR the value with the specified ac97 codec register
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to OR
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audioixp_or_ac97(audioixp_state_t *statep, int reg, uint16_t data)
-{
-	data |= statep->codec_shadow[IXP_CODEC_REG(reg)];
-	if (audioixp_write_ac97(statep, reg, data) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_or_ac97() */
-
-/*
- * audioixp_reset_ac97()
- *
- * Description:
- *	Reset AC97 Codec register.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Reset the codec successfully
- *	AUDIO_FAILURE		Failed to reset the codec
- */
-static int
-audioixp_reset_ac97(audioixp_state_t *statep)
-{
-	uint32_t	cmd;
-	int i;
-
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD, IXP_AUDIO_CMD_POWER_DOWN, 0);
-	drv_usecwait(10);
-
-	/* register reset */
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-	    IXP_AUDIO_CMD_AC_SOFT_RESET,
-	    IXP_AUDIO_CMD_AC_SOFT_RESET);
-
-	drv_usecwait(10);
-	IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-	    IXP_AUDIO_CMD_AC_SOFT_RESET,
-	    0);
-
-	/* cold reset */
-	for (i = 0; i < 300; i++) {
-		cmd = IXP_AM_GET32(IXP_AUDIO_CMD);
-		if (cmd & IXP_AUDIO_CMD_AC_ACTIVE) {
-			cmd |= IXP_AUDIO_CMD_AC_RESET;
-			IXP_AM_PUT32(IXP_AUDIO_CMD, cmd);
-			return (AUDIO_SUCCESS);
-		}
-		cmd &= ~IXP_AUDIO_CMD_AC_RESET;
-		IXP_AM_PUT32(IXP_AUDIO_CMD, cmd);
-		(void) IXP_AM_GET32(IXP_AUDIO_CMD);
-		drv_usecwait(10);
-		cmd |= IXP_AUDIO_CMD_AC_RESET;
-		IXP_AM_PUT32(IXP_AUDIO_CMD, cmd);
-		drv_usecwait(10);
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audioixp_reset_ac97() */
-
-/*
- * audioixp_chunk_processed()
- *
- * Description:
- *	This routine returns the count of chunk processed. It's called by
- *	audioixp_reclaim_play_buf and audioixp_reclaim_record_buf
- *	This routine compares the current hw_point value with its last value,
- *	there're two cases:
- *	case 1: new pointer is bigger than the last one and smaller than the
- *		last one + len of the last chunk, which mean the current
- *		chunk has not been finished, return 0.
- *	case 2: the hw_pointer return to the old value, which means both chunks
- *		have been processed,return 2
- *	case 3: one chunk is processed, return 1.
- *
- * Arguments:
- *	audioixp_state_t 	*statep	The device's state structure
- *	int			dir	AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *
- * Returns:
- *	count of chunk processed
- */
-static int
-audioixp_chunk_processed(audioixp_state_t *statep, int dir)
-{
-	audioixp_sample_buf_t	*buf;
-	uint32_t 		hw_pointer;
-	int			result;
-	audioixp_bd_entry_t	*bd;
-	int 			i;
-	int			retry_count;
-
-	ASSERT(mutex_owned(&statep->inst_lock));
-
-	retry_count = 0;
-	while (++retry_count < 100) {
-		if (dir == AUDIO_PLAY) {
-			buf = &statep->play_buf;
-			hw_pointer = IXP_AM_GET32(IXP_AUDIO_OUT_DMA_DT_CUR);
-		} else {
-			buf = &statep->record_buf;
-			hw_pointer = IXP_AM_GET32(IXP_AUDIO_IN_DMA_DT_CUR);
-		}
-
-		for (i = 0; i < IXP_BD_NUMS; i ++) {
-			if (dir == AUDIO_PLAY)
-				bd = &statep->bdl_virtual->pcm_out[i];
-			else
-				bd = &statep->bdl_virtual->pcm_in[i];
-
-			if (hw_pointer >= bd->buf_base &&
-			    hw_pointer < bd->buf_base + bd->buf_len*4)
-				break;
-		}
-
-		if (i < IXP_BD_NUMS)
-			break;
-	}
-
-	/*
-	 * cannot get valid hw_pointer, return 0 without updating
-	 * last_hw_pointer
-	 */
-	if (retry_count == 100) {
-		cmn_err(CE_WARN, "!bad hw_pointer, hw_pointer=0x%08x",
-		    hw_pointer);
-		for (i = 0; i < IXP_BD_NUMS; i ++) {
-			if (dir == AUDIO_PLAY)
-				bd = &statep->bdl_virtual->pcm_out[i];
-			else
-				bd = &statep->bdl_virtual->pcm_in[i];
-
-			cmn_err(CE_WARN, "!bd[%d], base=0x%08x, len=0x%x",
-			    i, bd->buf_base, bd->buf_len);
-		}
-		return (0);
-	}
-
-	if (buf->last_hw_pointer >= bd->buf_base && /* case 1 */
-	    hw_pointer > buf->last_hw_pointer &&
-	    hw_pointer < bd->buf_base + bd->buf_len * 4)
-		result = 0;
-	else if (buf->last_hw_pointer == hw_pointer) /* case 2 */
-		result = 2;
-	else /* case 3 */
-		result = 1;
-
-	buf->last_hw_pointer = hw_pointer;
-
-	return (result);
-
-}	/* audioixp_chunk_processed() */
-
-/*
- * audioixp_fill_play_buf()
- *
- * Description:
- *	This routine is called by audioixp_ad_start_play() and the
- * 	interrupt handler. It fills playback samples into the DMA memory,
- *	sets the BDL entries, and starts the playback DMA engine.
- *
- * Arguments:
- *	audioixp_state_t 	*statep	The device's state structure
- *
- * Returns:
- * 	AUDIO_SUCCESS		Starting PCM out engine successfully
- * 	AUDIO_FAILURE		Failed to start PCM out engine.
- */
-static int
-audioixp_fill_play_buf(audioixp_state_t *statep)
-{
-	audioixp_bdlist_chunk_t	*chunk;
-	audioixp_sample_buf_t	*buf;
-	audioixp_bd_entry_t		*bdesc;
-	int				samples;
-	int				rs;
-
-	buf = &statep->play_buf;
-
-	if (!buf->io_started) {
-		/*
-		 * ready to start PCM out engine
-		 */
-		IXP_AM_PUT32(
-		    IXP_AUDIO_OUT_DMA_LINK_P,
-		    (uint32_t)(uintptr_t)statep->bdl_phys->pcm_out |
-		    IXP_AUDIO_OUT_DMA_LINK_P_EN);
-
-		buf->next = 0;
-		buf->avail = 2;	/* have only two buffers for playback */
-		buf->last_hw_pointer = statep->bdl_virtual->pcm_out[0].buf_base;
-	}
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	samples = statep->ixp_psample_rate * statep->ixp_pchannels /
-	    statep->ad_info.ad_play.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	while (buf->avail > 0) {
-		chunk = &(buf->chunk[buf->next & 1]);
-		mutex_exit(&statep->inst_lock);
-		rs = am_get_audio(statep->audio_handle,
-		    (char *)(chunk->data_buf), AUDIO_NO_CHANNEL, samples);
-
-		mutex_enter(&statep->inst_lock);
-
-		if (((statep->flags & IXP_DMA_PLAY_STARTED) == 0) &&
-		    (buf->io_started)) {
-			return (AUDIO_FAILURE);
-		}
-
-		if (rs <= 0) {
-			if (statep->flags & IXP_DMA_PLAY_EMPTY) {
-
-				/*
-				 * Clear the flag so if audio is restarted while
-				 * in am_play_shutdown() we can detect it and
-				 * not mess things up.
-				 */
-				statep->flags &= ~IXP_DMA_PLAY_STARTED;
-
-				/* shutdown the mixer */
-				mutex_exit(&statep->inst_lock);
-				am_play_shutdown(statep->audio_handle, NULL);
-				mutex_enter(&statep->inst_lock);
-
-				/*
-				 * Make sure playing wasn't restarted when lock
-				 * lost if reopened, should return success
-				 */
-				if (statep->flags & IXP_DMA_PLAY_STARTED) {
-					return (AUDIO_SUCCESS);
-				}
-
-				/* Finished playing, then stop it */
-				IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-				    IXP_AUDIO_CMD_EN_OUT,
-				    0);
-
-				buf->io_started = B_FALSE;
-
-				/* clr the flags getting ready for next start */
-				statep->flags &= ~(IXP_DMA_PLAY_PAUSED |
-				    IXP_DMA_PLAY_EMPTY);
-
-				return (AUDIO_FAILURE);
-			} else {
-				/*
-				 * this time, we use one BD entry with empty
-				 * buffer next time we shut down, if no sound
-				 * again
-				 */
-				statep->flags |= IXP_DMA_PLAY_EMPTY;
-			}
-		} else {
-			/* we got at least one sample */
-			statep->flags &= ~IXP_DMA_PLAY_EMPTY;
-			(void) ddi_dma_sync(chunk->dma_handle, 0, rs<<1,
-			    DDI_DMA_SYNC_FORDEV);
-		}
-
-		/* put the samples into buffer descriptor list entry */
-		bdesc = &(statep->bdl_virtual->pcm_out[buf->next]);
-		bdesc->buf_len = (uint16_t)rs>>1; /* in dword */
-
-		buf->avail --;
-		buf->next ++;
-		buf->next %= IXP_BD_NUMS;
-	}
-
-	/* start PCM out engine */
-	if (!buf->io_started) {
-		IXP_AM_UPDATE32(
-		    IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_OUT,
-		    IXP_AUDIO_CMD_EN_OUT);
-
-		buf->io_started = B_TRUE;
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_fill_play_buf() */
-
-/*
- * audioixp_reclaim_play_buf()
- *
- * Description:
- *	When the audio controller finishes fetching the data from DMA
- *	buffers, this routine will be called by interrupt handler to
- *	reclaim the DMA buffers.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_reclaim_play_buf(audioixp_state_t *statep)
-{
-
-	audioixp_sample_buf_t *buf;
-
-	buf = &statep->play_buf;
-	buf->avail += audioixp_chunk_processed(statep, AUDIO_PLAY);
-
-	return;
-
-}	/* audioixp_reclaim_play_buf() */
-
-/*
- * audioixp_prepare_record_buf()
- *
- * Description:
- *	This routine is called by audioixp_ad_start_record(). It prepares
- *	DMA memory for PCM in engine, sets the buffer descriptor entries for PCM
- *	in engine, and starts PCM in engine for recording.
- *
- * Arguments:
- *	audioixp_state_t	*statep		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Started PCM in  engine successfully
- *	AUDIO_FAILURE		Failed to start PCM in engine.
- *
- */
-static int
-audioixp_prepare_record_buf(audioixp_state_t *statep)
-{
-	audioixp_sample_buf_t	*buf;
-	audioixp_bd_entry_t		*bdesc;
-	int				samples;
-	int 				i;
-
-	buf = &statep->record_buf;
-
-	if (!buf->io_started) {
-
-		/* buffer base */
-		IXP_AM_PUT32(IXP_AUDIO_IN_DMA_LINK_P,
-		    (uint32_t)(uintptr_t)statep->bdl_phys->pcm_in |
-		    IXP_AUDIO_IN_DMA_LINK_P_EN);
-		buf->next = 0;
-		buf->avail = 2;
-		buf->last_hw_pointer = statep->bdl_virtual->pcm_in[0].buf_base;
-	}
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	samples = statep->ixp_csample_rate * statep->ixp_cchannels /
-	    statep->ad_info.ad_record.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	statep->ixp_csamples = samples;
-	for (i = 0; i < 2; i ++) {
-		samples = statep->ixp_csamples;
-		bdesc = &(statep->bdl_virtual->pcm_in[i]);
-		bdesc->buf_len = (uint16_t)samples >> 1; /* in dword */
-		buf->avail --;
-	}
-
-	if (!buf->io_started) {
-		buf->io_started = B_TRUE;
-		IXP_AM_UPDATE32(IXP_AUDIO_CMD,
-		    IXP_AUDIO_CMD_EN_IN,
-		    IXP_AUDIO_CMD_EN_IN);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_prepare_record_buf() */
-
-/*
- * audioixp_reclaim_record_buf()
- *
- * Description:
- *	This routine is called by the interrupt handler. It sends the PCM
- *	samples (record data) up to the mixer module by calling am_send_audio(),
- *	and reclaims the buffer descriptor entries for PCM in engine.
- *
- * Arguments:
- *	audioixp_state_t	*statep		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audioixp_reclaim_record_buf(audioixp_state_t *statep)
-{
-	audioixp_bdlist_chunk_t 	*chunk;
-	audioixp_sample_buf_t	*buf;
-	int				samples;
-
-	buf = &statep->record_buf;
-	samples = statep->ixp_csamples;
-
-	buf->avail += audioixp_chunk_processed(statep, AUDIO_RECORD);
-
-	while (buf->avail > 0) {
-		chunk = &buf->chunk[buf->next & 1];
-		(void) ddi_dma_sync(chunk->dma_handle, 0,
-		    samples<<1, DDI_DMA_SYNC_FORCPU);
-		mutex_exit(&statep->inst_lock);
-		am_send_audio(statep->audio_handle, chunk->data_buf,
-		    AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&statep->inst_lock);
-		buf->avail --;
-		buf->next ++;
-		buf->next %= IXP_BD_NUMS;
-
-		if ((statep->flags & IXP_DMA_RECD_STARTED) == 0) {
-			break;
-		}
-	}
-}	/* audioixp_reclaim_record_buf() */
-
-/*
- * audioixp_set_gain()
- *
- * Description:
- *	Set the play/record gain.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD, if
- *						direction is important
- *	int			arg1		The gain to set
- *	int			arg2		The channel, 0 == left
- *						or 1 == right
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audioixp_set_gain(audioixp_state_t *statep, int dir, int gain,
-	int channel)
-{
-	uint16_t	tmp;
-	uint16_t	channel_gain;
-	int		regidx;
-	int		max_channel_gain;
-
-	if (gain > AUDIO_MAX_GAIN) {
-		gain = AUDIO_MAX_GAIN;
-	} else if (gain < AUDIO_MIN_GAIN) {
-		gain = AUDIO_MIN_GAIN;
-	}
-
-	max_channel_gain = (1<<PCMOVR_GAIN_BITS)-1;
-	channel_gain = 31-gain*max_channel_gain/AUDIO_MAX_GAIN;
-
-	if (dir == AUDIO_PLAY) {
-		if (statep->swap_out == B_TRUE) {
-			regidx = AC97_EXTENDED_LRS_VOLUME_REGISTER;
-		} else {
-			regidx = AC97_PCM_OUT_VOLUME_REGISTER;
-		}
-		(void) audioixp_read_ac97(statep, regidx, &tmp);
-
-		if (channel == 0) { /* left channel */
-			tmp &= PCMOVR_RIGHT_GAIN_MASK;
-			tmp |= (channel_gain << 8);
-		} else {	/* right channel */
-			tmp &= PCMOVR_LEFT_GAIN_MASK;
-			tmp |= channel_gain;
-		}
-
-		(void) audioixp_write_ac97(statep, regidx, tmp);
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-
-		(void) audioixp_read_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, &tmp);
-
-		if (channel == 0) {	/* left channel */
-			tmp &= ~RGR_LEFT_MASK;
-			tmp |= gain & RGR_LEFT_MASK;
-		} else {
-			/* right channel */
-			ASSERT(channel == 1);
-			tmp &= ~RGR_RIGHT_MASK;
-			tmp |= gain & RGR_RIGHT_MASK;
-		}
-		(void) audioixp_write_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, tmp);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_set_gain() */
-
-/*
- * audioixp_set_port()
- *
- * Description:
- *	Set the play/record port.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *						which is not how we program
- *						the device for now.
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD,
- *						if direction is important
- *	int			port		The port to set
- *				AUDIO_SPEAKER	output to built-in speaker
- *
- *				AUDIO_MICROPHONE	input from microphone
- *				AUDIO_LINE_IN		input from line in
- *				AUDIO_CODEC_LOOPB_IN	input from Codec
- *							internal loopback
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The port could not been set
- */
-static int
-audioixp_set_port(audioixp_state_t *statep, int dir, int port)
-{
-	uint16_t	tmp;
-
-	if (dir == AUDIO_PLAY) {	/* output port */
-		tmp = 0;
-		if (port == IXP_PORT_UNMUTE) {
-			port = statep->ixp_output_port;
-		}
-
-		if (port & AUDIO_SPEAKER) {
-			(void) audioixp_and_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_SPEAKER;
-		} else {
-			(void) audioixp_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MVR_MUTE);
-		}
-
-		if (port & AUDIO_LINE_OUT) {
-			if (statep->swap_out == B_FALSE) {
-				(void) audioixp_and_ac97(statep,
-				    AC97_MASTER_VOLUME_REGISTER,
-				    (uint16_t)~MVR_MUTE);
-			} else {
-				(void) audioixp_and_ac97(statep,
-				    AC97_EXTENDED_LRS_VOLUME_REGISTER,
-				    (uint16_t)~AD1980_SURR_MUTE);
-			}
-			tmp |= AUDIO_LINE_OUT;
-		} else {
-			if (statep->swap_out == B_FALSE) {
-				(void) audioixp_or_ac97(statep,
-				    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-			} else {
-				(void) audioixp_or_ac97(statep,
-				    AC97_EXTENDED_LRS_VOLUME_REGISTER,
-				    AD1980_SURR_MUTE);
-			}
-		}
-
-		if (port & AUDIO_HEADPHONE) {
-			(void) audioixp_and_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_HEADPHONE;
-		} else {
-			(void) audioixp_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, MVR_MUTE);
-		}
-
-		ATRACE_32("audioixp_set_port() out port", tmp);
-		statep->ixp_output_port = tmp;
-		if (tmp != port) {
-			ATRACE_32("audioixp_set_port() bad out port", port);
-			return (AUDIO_FAILURE);
-		}
-
-	} else {		/* input port */
-		ASSERT(dir == AUDIO_RECORD);
-
-		switch (port) {
-		case AUDIO_NONE:
-			/* set to an unused input */
-			tmp = RSCR_R_PHONE | RSCR_L_PHONE;
-
-			/* mute the master record input */
-			(void) audioixp_or_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER, RGR_MUTE);
-
-			if (statep->ixp_monitor_gain) {
-				if (statep->ixp_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_CD) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-
-		case AUDIO_MICROPHONE:
-			/* set to the mic input */
-			tmp = RSCR_R_MIC | RSCR_L_MIC;
-
-			if (statep->ixp_monitor_gain) {
-				if (statep->ixp_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_CD) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				(void) audioixp_write_ac97(statep,
-				    AC97_MIC_VOLUME_REGISTER,
-				    statep->ixp_monitor_gain);
-			}
-			break;
-
-		case AUDIO_LINE_IN:
-			/* set to the line in input */
-			tmp = RSCR_R_LINE_IN | RSCR_L_LINE_IN;
-
-			/* see if we need to update monitor loopback */
-			if (statep->ixp_monitor_gain) {
-				if (statep->ixp_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_CD) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				(void) audioixp_write_ac97(statep,
-				    AC97_LINE_IN_VOLUME_REGISTER,
-				    statep->ixp_monitor_gain);
-			}
-			break;
-
-		case AUDIO_CD:
-			/* set to the line in input */
-			tmp = RSCR_R_CD|RSCR_L_CD;
-
-			/* see if we need to update monitor loopback */
-			if (statep->ixp_monitor_gain) {
-				if (statep->ixp_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				}
-				(void) audioixp_write_ac97(statep,
-				    AC97_CD_VOLUME_REGISTER,
-				    statep->ixp_monitor_gain);
-			}
-			break;
-
-		case AUDIO_CODEC_LOOPB_IN:
-			/* set to the loopback input */
-			tmp = RSCR_R_STEREO_MIX | RSCR_L_STEREO_MIX;
-
-			if (statep->ixp_monitor_gain) {
-				if (statep->ixp_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->ixp_input_port ==
-				    AUDIO_CD) {
-					(void) audioixp_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-
-		default:
-			ATRACE_32("audioixp_set_port bad in port", port);
-			return (AUDIO_FAILURE);
-		}
-
-		/* select the input */
-		(void) audioixp_write_ac97(statep,
-		    AC97_RECORD_SELECT_CTRL_REGISTER, tmp);
-		if ((port != AUDIO_NONE) &&
-		    (statep->codec_shadow[IXP_CODEC_REG(
-		    AC97_RECORD_GAIN_REGISTER)] & RGR_MUTE)) {
-			(void) audioixp_and_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER,
-			    (uint16_t)~RGR_MUTE);
-		}
-		statep->ixp_input_port = port;
-	}
-
-	ATRACE_32("audioixp_set_port() returning", 0);
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_set_port() */
-
-/*
- * audioixp_set_monitor_gain()
- *
- * Description:
- *	Set the monitor gain.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			gain		The gain to set
- *
- * Returns:
- * 	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audioixp_set_monitor_gain(audioixp_state_t *statep, int gain)
-{
-	uint16_t	tmp_short;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("in audioixp_set_monitor_gain()", statep);
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (gain == 0) {
-		/* disable loopbacks when gain == 0 */
-		tmp_short = MVR_MUTE;
-	} else {
-		/* Adjust the value of gain to the requirement of AC'97 */
-		tmp_short = AUDIO_MAX_GAIN - gain;
-		tmp_short = ((tmp_short << statep->vol_bits_mask) - tmp_short)
-		    / AUDIO_MAX_GAIN;
-		tmp_short |= (((tmp_short << statep->vol_bits_mask) -
-		    tmp_short) / AUDIO_MAX_GAIN) << 8;
-	}
-
-	switch (statep->ixp_input_port) {
-	case AUDIO_NONE:
-		/*
-		 * It is possible to set the value of gain before any input
-		 * is selected. So, we just save the gain and then return
-		 * SUCCESS.
-		 */
-		break;
-
-	case AUDIO_MICROPHONE:
-		/*
-		 * MIC input has 20dB boost, we just preserve it
-		 */
-		tmp_short |=
-		    statep->codec_shadow[IXP_CODEC_REG(
-		    AC97_MIC_VOLUME_REGISTER)] & MICVR_20dB_BOOST;
-		(void) audioixp_write_ac97(statep,
-		    AC97_MIC_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_LINE_IN:
-		(void) audioixp_write_ac97(statep,
-		    AC97_LINE_IN_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CD:
-		(void) audioixp_write_ac97(statep,
-		    AC97_CD_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CODEC_LOOPB_IN:
-		/* we already are getting the loopback, so done */
-		rc = AUDIO_SUCCESS;
-		goto done;
-
-	default:
-		/* this should never happen! */
-		ATRACE("audioixp_ad_set_config() monitor gain bad device",
-		    NULL);
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-
-	if (gain == 0) {
-		statep->ixp_monitor_gain = 0;
-	} else {
-		statep->ixp_monitor_gain = tmp_short;
-	}
-
-done:
-	ATRACE_32("audioixp_set_monitor_gain()", rc);
-
-	return (rc);
-
-}	/* audioixp_set_monitor_gain() */
-
-/*
- * audioixp_chip_init()
- *
- * Description:
- *	This routine initializes ATI IXP audio controller and the AC97
- *	codec.  The AC97 codec registers are programmed from codec_shadow[].
- *	If we are not doing a restore, we initialize codec_shadow[], otherwise
- *	we use the current values of shadow
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *	int			restore		If IXP_INIT_RESTORE then
- *						restore	from codec_shadow[]
- * Returns:
- *	AUDIO_SUCCESS	The hardware was initialized properly
- *	AUDIO_FAILURE	The hardware couldn't be initialized properly
- */
-static int
-audioixp_chip_init(audioixp_state_t *statep, int restore)
-{
-	uint16_t	*shadow;
-	int 		i;
-	int 		j;
-	uint16_t	xid;
-	uint16_t	vid1;
-	uint16_t	vid2;
-	uint16_t	sr;
-	uint16_t 	tmp;
-
-	/*
-	 * put the audio controller into quiet state, everything off
-	 */
-	audioixp_stop_dma(statep, AUDIO_PLAY);
-	audioixp_stop_dma(statep, AUDIO_RECORD);
-
-	/* AC97 reset */
-	if (audioixp_reset_ac97(statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_chip_init() AC97 codec reset failed");
-		return (AUDIO_FAILURE);
-	}
-
-	if (audioixp_codec_ready(statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->audio_handle, CE_WARN,
-		    "!audioixp_chip_init() AC97 codec not ready");
-		return (AUDIO_FAILURE);
-	}
-
-	shadow = statep->codec_shadow;
-
-	if (restore == IXP_INIT_NO_RESTORE) {
-		for (i = 0; i <= IXP_LAST_AC_REG; i += 2) {
-			(void) audioixp_read_ac97(statep, i,
-			    &(shadow[IXP_CODEC_REG(i)]));
-		}
-
-		/* 02h - set master line out volume, muted, 0dB */
-		shadow[IXP_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE;
-
-		/* 04h - set alternate line out volume, muted, 0dB */
-		shadow[IXP_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-
-		/* 06h - set master mono volume, muted, 0dB */
-		shadow[IXP_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE;
-
-		/* 08h - set master tone control to no modification */
-		shadow[IXP_CODEC_REG(AC97_MASTER_TONE_CONTROL_REGISTER)] =
-		    MTCR_BASS_BYPASS|MTCR_TREBLE_BYPASS;
-
-		/*
-		 * 0ah - turn pc beep mute off, 0dB
-		 *
-		 * AC'97 Spec does define the optional PC Beep support, that is,
-		 * the BIOS (dependent on hardware design) can use the audio
-		 * hardware for the beep, especially on some laptops, in order
-		 * to save cost. So we have to turn the pc_beep mute off, that
-		 * is, enable the PC Beep support.
-		 */
-		shadow[IXP_CODEC_REG(AC97_PC_BEEP_REGISTER)] =
-		    PCBR_0dB_ATTEN;
-
-		/* 0ch - set phone input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_PHONE_VOLUME_REGISTER)] =
-		    PVR_MUTE|PVR_0dB_GAIN;
-
-		/* 0eh - set mic input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_MIC_VOLUME_REGISTER)] =
-		    MICVR_MUTE|MICVR_0dB_GAIN;
-
-		/* 10h - set line input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_LINE_IN_VOLUME_REGISTER)] =
-		    LIVR_MUTE|LIVR_RIGHT_0dB_GAIN|LIVR_LEFT_0dB_GAIN;
-
-		/* 12h - set cd input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_CD_VOLUME_REGISTER)] =
-		    CDVR_MUTE|CDVR_RIGHT_0dB_GAIN|CDVR_LEFT_0dB_GAIN;
-
-		/* 14h - set video input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_VIDEO_VOLUME_REGISTER)] =
-		    VIDVR_MUTE|VIDVR_RIGHT_0dB_GAIN|VIDVR_LEFT_0dB_GAIN;
-
-		/* 16h - set aux input, mute, 0dB attenuation */
-		shadow[IXP_CODEC_REG(AC97_AUX_VOLUME_REGISTER)] =
-		    AUXVR_MUTE|AUXVR_RIGHT_0dB_GAIN|AUXVR_LEFT_0dB_GAIN;
-
-		/* 18h - set PCM out input, NOT muted, 0dB gain */
-		shadow[IXP_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_RIGHT_0dB_GAIN|PCMOVR_LEFT_0dB_GAIN;
-
-		/* 1ah - set input device as mic */
-		shadow[IXP_CODEC_REG(AC97_RECORD_SELECT_CTRL_REGISTER)] =
-		    RSCR_R_MIC|RSCR_L_MIC;
-
-		/* 1ch - set record gain to 0dB and not muted */
-		shadow[IXP_CODEC_REG(AC97_RECORD_GAIN_REGISTER)] =
-		    RGR_RIGHT_0db_GAIN|RGR_LEFT_0db_GAIN;
-
-		/* 1eh - set record mic gain to 0dB and not muted */
-		shadow[IXP_CODEC_REG(AC97_RECORD_GAIN_MIC_REGISTER)] =
-		    RGMR_0db_GAIN;
-
-		/* 20h - set GP register, mic 1, everything else off */
-		shadow[IXP_CODEC_REG(AC97_GENERAL_PURPOSE_REGISTER)] =
-		    GPR_MS_MIC1|GPR_MONO_MIX_IN;
-
-		/* 22h - set 3D control to NULL */
-		shadow[IXP_CODEC_REG(AC97_THREE_D_CONTROL_REGISTER)] =
-		    TDCR_NULL;
-
-		/*
-		 * The rest we ignore, most are reserved.
-		 */
-
-	}
-
-	if (restore == IXP_INIT_RESTORE) {
-		/* Restore from saved values */
-		shadow[IXP_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE;
-		shadow[IXP_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-		shadow[IXP_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE;
-		shadow[IXP_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_MUTE;
-	}
-
-	/* Now we set the AC97 codec registers to the saved values */
-	for (i = 2; i <= IXP_LAST_AC_REG; i += 2)
-		(void) audioixp_write_ac97(statep, i,
-		    shadow[IXP_CODEC_REG(i)]);
-
-	(void) audioixp_read_ac97(statep, AC97_RESET_REGISTER, &tmp);
-	if (tmp & RR_HEADPHONE_SUPPORT) {
-		statep->ixp_defaults.play.port |= AUDIO_HEADPHONE;
-		statep->ixp_defaults.play.avail_ports |= AUDIO_HEADPHONE;
-		statep->ixp_defaults.play.mod_ports |= AUDIO_HEADPHONE;
-	}
-
-	/*
-	 * Most vendors connect the surr-out of ad1980/ad1985 codecs to the
-	 * line-out jack. So far we haven't found which vendors don't
-	 * do that. So we assume that all vendors swap the surr-out
-	 * and the line-out outputs. So we need swap the two outputs.
-	 * But we still internally process the "ad198x-swap-output"
-	 * property. If someday some vendors do not swap the outputs,
-	 * we would set "ad198x-swap-output = 0" in the
-	 * /kernel/drv/audioixp.conf file, and unload and reload the
-	 * audioixp driver (or reboot).
-	 */
-	(void) audioixp_read_ac97(statep, AC97_VENDOR_ID1_REGISTER, &vid1);
-	(void) audioixp_read_ac97(statep, AC97_VENDOR_ID2_REGISTER, &vid2);
-	if (vid1 == AD1980_VID1 &&
-	    (vid2 == AD1980_VID2 || vid2 == AD1985_VID2)) {
-		if (ddi_prop_get_int(DDI_DEV_T_ANY, statep->dip,
-		    DDI_PROP_DONTPASS, "ad198x-swap-output", 1) == 1) {
-			statep->swap_out = B_TRUE;
-			(void) audioixp_read_ac97(statep, CODEC_AD_REG_MISC,
-			    &tmp);
-			(void) audioixp_write_ac97(statep,
-			    CODEC_AD_REG_MISC,
-			    tmp | AD1980_MISC_LOSEL | AD1980_MISC_HPSEL);
-		}
-	}
-
-	/*
-	 * check if the codec implements 6 bit volume register,
-	 * but the ALC202 does not strictly obey the AC'97 Spec
-	 * and it only supports 5 bit volume register, so we
-	 * skip the check for it as a workaround.
-	 */
-	if (!(vid1 == ALC202_VID1 && vid2 == ALC202_VID2)) {
-		(void) audioixp_write_ac97(statep,
-		    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE |
-		    MVR_RIGHT_OPTIONAL_MASK | MVR_LEFT_OPTIONAL_MASK);
-
-		(void) audioixp_read_ac97(statep,
-		    AC97_MASTER_VOLUME_REGISTER, &tmp);
-
-		if ((tmp & 0x7fff) != (MVR_RIGHT_MASK | MVR_LEFT_MASK)) {
-			statep->vol_bits_mask = 6;
-		}
-	}
-
-	/* resume the master volume to the max */
-	(void) audioixp_write_ac97(statep, AC97_MASTER_VOLUME_REGISTER,
-	    MVR_MUTE);
-
-	/*
-	 * if the codec chip does not support variable sample rate,
-	 * we set the sample rate to 48K
-	 */
-	(void) audioixp_read_ac97(statep, AC97_EXTENDED_AUDIO_REGISTER,
-	    &xid);
-	audio_sup_log(statep->audio_handle, CE_NOTE,
-	    "!%s%d: xid=0x%04x, vid1=0x%04x, vid2=0x%04x",
-	    audioixp_name,  ddi_get_instance(statep->dip), xid, vid1, vid2);
-	if (!(xid & EAR_VRA)) {
-		statep->var_sr = B_FALSE;
-		statep->ad_info.ad_record.ad_compat_srs =
-		    audioixp_min_compat_sample_rates;
-		statep->ad_info.ad_play.ad_compat_srs =
-		    audioixp_min_compat_sample_rates;
-		statep->ixp_defaults.play.sample_rate =
-		    IXP_SAMPR48000;
-		statep->ixp_defaults.record.sample_rate =
-		    IXP_SAMPR48000;
-	} else {	/* variable sample rate supported */
-		statep->var_sr = B_TRUE;
-
-		/* set variable rate mode */
-		(void) audioixp_write_ac97(statep,
-		    AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER, EASCR_VRA);
-
-		/* check the sample rates supported */
-		for (i = 0, j = 0; audioixp_compat_srs[i] != 0; i++) {
-			(void) audioixp_write_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER,
-			    audioixp_compat_srs[i]);
-			(void) audioixp_read_ac97(statep,
-			    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, &sr);
-
-			if (sr == audioixp_compat_srs[i]) {
-				if (i != j) {
-					audioixp_compat_srs[j] =
-					    audioixp_compat_srs[i];
-				}
-				j++;
-			}
-		}
-
-		if (j < 1) {
-			audio_sup_log(statep->audio_handle, CE_WARN,
-			    "!No standard sample rate is supported");
-			return (AUDIO_FAILURE);
-		}
-		audioixp_compat_srs[j] = 0;
-
-		/*
-		 * if the configuration doesn't support 8K sample rate,
-		 * we modify the default value to the first.
-		 */
-		for (i = 0; audioixp_compat_srs[i] != 0; i++) {
-			if (audioixp_compat_srs[i] == IXP_SAMPR8000) {
-				break;
-			}
-		}
-		if (audioixp_compat_srs[i] != IXP_SAMPR8000) {
-			statep->ixp_defaults.play.sample_rate =
-			    audioixp_compat_srs[0];
-			statep->ixp_defaults.record.sample_rate =
-			    audioixp_compat_srs[0];
-		}
-	}
-
-	/* enable interrupts */
-	IXP_AM_PUT32(IXP_AUDIO_INT, 0xffffffff);
-	IXP_AM_PUT32(
-	    IXP_AUDIO_INT_EN,
-	    IXP_AUDIO_INT_EN_IN_DMA_OVERFLOW |
-	    IXP_AUDIO_INT_EN_STATUS |
-	    IXP_AUDIO_INT_EN_OUT_DMA_UNDERFLOW);
-	return (AUDIO_SUCCESS);
-
-}	/* audioixp_chip_init() */
-
-/*
- * audioixp_chip_fini()
- *
- * Description:
- *	This routine disables hardware interrupts.
- *
- * Arguments:
- *	audioixp_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void audioixp_chip_fini(audioixp_state_t *statep)
-{
-	IXP_AM_PUT32(IXP_AUDIO_INT, IXP_AM_GET32(IXP_AUDIO_INT));
-	IXP_AM_PUT32(IXP_AUDIO_INT_EN, 0);
-}	/* audioixp_chip_fini() */
diff --git a/usr/src/uts/common/io/audio/sada/drv/audiots/audiots.c b/usr/src/uts/common/io/audio/sada/drv/audiots/audiots.c
deleted file mode 100644
index d1a69da820..0000000000
--- a/usr/src/uts/common/io/audio/sada/drv/audiots/audiots.c
+++ /dev/null
@@ -1,4264 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-
-/*
- * audiots Audio Driver
- *
- * This Audio Driver controls the T2 audio core in the ALI M1553
- * southbridge chip. This chip supports multiple play streams, but just
- * a single record stream. It also supports wave table synthesis and
- * hardware MIDI and joystick ports. Unfortunately the MIDI ports are
- * not available because their pins have been re-assigned to expose
- * interrupts. We also aren't going to do anything with the joystick
- * ports. The audio core controls an AC-97 V2.1 Codec.
- *
- * This driver uses the mixer Audio Personality Module to implement
- * audio(7I) and mixer(7I) semantics. Although the play side of the
- * audio core supports multiple streams we don't use that feature.
- * The mixer needs to be fixed up first. Thus we let the mixer do it's
- * thing for both directions.
- *
- * The DMA engine uses a single buffer which is large enough to hold
- * two interrupts worth of data. When it gets to the mid point an
- * interrupt is generated and data is either sent (for record) or
- * requested and put in that half of the buffer (for play). When the
- * second half is played we do the same, but the audio core loops the
- * pointer back to the beginning. For play we bzero() the half buffer
- * before we ask for more audio. That way if there isn't enough waiting
- * for us we just play silence. If more arrives later we'll keep going,
- * but after a slight pop.
- *
- * The audio core has a bug in silicon that doesn't let it read the AC-97
- * Codec's register. T2 has provided an algorithm that attempts to read the
- * the Codec several times. This is probably heuristic and thus isn't
- * absolutely guaranteed to work. However we do have to place a limit on
- * the looping, otherwise when we read a valid 0x00 we would never exit
- * the loop. Unfortunately there is also a problem with writing the AC-97
- * Codec's registers as well. Thus we read it back to verify the write.
- *
- * Every time we program the AC-97 Codec we save the value in ts_shadow[].
- * Thus every time we need to get a Codec register we don't have to do
- * a very long read. This also means that register state information is
- * saved for power management shutdown (CPR). When the Codec is started
- * back up we use this saved state to restore the Codec's state in
- * audiots_chip_init().
- *
- * We don't save any of the audio controller registers during normal
- * operation. When we need to save register state we only have to save
- * the aram and eram. The rest of the controller state is never modified
- * from the initial programming. Thus restoring the controller state
- * can be done from audiots_chip_init() as well.
- *
- *
- * WARNING: The SME birdsnest platform uses a PCI bridge chip between the
- *	CPU and the southbridge containing the audio core. There is
- *	a bug in silicon that causes a bogus parity error. With the mixer
- *	reimplementation project, Bug 4374774, the audio driver is always
- *	set to the best precision and number of channels. Thus when turning
- *	the mixer on and off the only thing that changes is the sample rate.
- *	This change in programming doesn't trigger the silicon error.
- *	Thus the supported channels must always be 2 and the precision
- *	must always be 16-bits. This will keep any future change in the
- *	mixer from exposing this bug.
- *
- * Due to a hardware bug, system power management is not supported by this
- * driver.
- *
- *	CAUTION: If audio controller state is changed outside of aram
- *		and eram then that information must be saved and restored
- *		during power management shutdown and bringup.
- *
- *	NOTE: The AC-97 Codec's reset pin is set to PCI reset, so we
- *		can't power down the Codec all the way.
- *
- *	NOTE: This driver depends on the misc/audiosup and misc/mixer
- *		modules being loaded first.
- *
- *	NOTE: Don't OR the ap_stop register to stop a play or record. This
- *		will just stop all active channels because a read of ap_stop
- *		returns ap_start. Just set the ap_stop register with the
- *		channels you want to stop. The same goes for ap_start.
- *
- *	NOTE: There is a hardware problem with P2 rev motherboards. After
- *		prolonged use, reading the AC97 register will always return
- *		busy. The AC97 register is now useless. Consequently, we are no
- *		longer able to program the Codec. This work around disables
- *		audio when this state is detected. It's not great, but its
- *		better than having audio blasting out at 100% all the time.
- *
- *	NOTE: Power Management testing has also exposed this AC97 timeout
- *		problem. Management has decided this is too risky for customers
- *		and hence they want power management support removed from the
- *		audio subsystem. All PM support is now removed.
- */
-
-#include <sys/modctl.h>
-#include <sys/kmem.h>
-#include <sys/stropts.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/audiots.h>
-#include <sys/audio/impl/audiots_impl.h>
-#include <sys/audio/ac97.h>
-
-/*
- * Module linkage routines for the kernel
- */
-static int audiots_getinfo(dev_info_t *, ddi_info_cmd_t, void *, void **);
-static int audiots_attach(dev_info_t *, ddi_attach_cmd_t);
-static int audiots_detach(dev_info_t *, ddi_detach_cmd_t);
-static int audiots_power(dev_info_t *, int, int);
-
-/*
- * Entry point routine prototypes
- */
-static int audiots_ad_setup(audiohdl_t, int, int);
-static void audiots_ad_pause_play(audiohdl_t, int);
-static int audiots_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int audiots_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int audiots_ad_start_play(audiohdl_t, int);
-static void audiots_ad_stop_play(audiohdl_t, int);
-static int audiots_ad_start_record(audiohdl_t, int);
-static void audiots_ad_stop_record(audiohdl_t, int);
-
-/*
- * Local Routine Prototypes
- */
-static void audiots_and_ac97(audiots_state_t *, int, uint16_t);
-static void audiots_chip_init(audiots_state_t *, int);
-static uint16_t audiots_get_ac97(audiots_state_t *, int);
-static int audiots_init_state(audiots_state_t *, dev_info_t *);
-static uint_t audiots_intr(caddr_t);
-static int audiots_map_regs(dev_info_t *, audiots_state_t *);
-static void audiots_or_ac97(audiots_state_t *, int, uint16_t);
-static void audiots_power_down(audiots_state_t *);
-static void audiots_power_up(audiots_state_t *);
-static uint16_t audiots_read_ac97(audiots_state_t *, int);
-static void audiots_save_controller(audiots_state_t *);
-static void audiots_set_ac97(audiots_state_t *, int, const uint16_t);
-static int audiots_set_busy(audiots_state_t *);
-static int audiots_set_gain(audiots_state_t *, int, int, int, int);
-static void audiots_set_idle(audiots_state_t *);
-static int audiots_set_monitor_gain(audiots_state_t *, int);
-static int audiots_set_port(audiots_state_t *, int, int);
-static int audiots_start_play(audiots_state_t *);
-static void audiots_stop_play(audiots_state_t *);
-static int audiots_start_record(audiots_state_t *);
-static void audiots_stop_record(audiots_state_t *);
-static void audiots_stop_everything(audiots_state_t *);
-static void audiots_unmap_regs(audiots_state_t *);
-
-/*
- * Global variables, but viewable only by this file.
- */
-
-/* anchor for soft state structures */
-static void *audiots_statep;
-
-/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
-static char *audiots_name = TS_NAME;
-
-static uint_t audiots_mixer_srs[] = {
-	TS_SAMPR5510, TS_SAMPR48000, 0
-};
-
-static uint_t audiots_compat_srs[] = {
-	TS_SAMPR5510, TS_SAMPR6620, TS_SAMPR8000,
-	TS_SAMPR9600, TS_SAMPR11025, TS_SAMPR16000,
-	TS_SAMPR18900, TS_SAMPR22050, TS_SAMPR27420,
-	TS_SAMPR32000, TS_SAMPR33075, TS_SAMPR37800,
-	TS_SAMPR44100, TS_SAMPR48000, 0
-};
-
-static am_ad_sample_rates_t audiots_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	audiots_mixer_srs
-};
-
-static am_ad_sample_rates_t audiots_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audiots_compat_srs
-};
-
-static uint_t audiots_channels[] = {
-	AUDIO_CHANNELS_STEREO, 0
-};
-
-static am_ad_cap_comb_t audiots_combinations[] = {
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
-
-static am_ad_entry_t audiots_entry = {
-	audiots_ad_setup,		/* ad_setup() */
-	NULL,				/* ad_teardown() */
-	audiots_ad_set_config,		/* ad_set_config() */
-	audiots_ad_set_format,		/* ad_set_format() */
-	audiots_ad_start_play,		/* ad_start_play() */
-	audiots_ad_pause_play,		/* ad_pause_play() */
-	audiots_ad_stop_play,		/* ad_stop_play() */
-	audiots_ad_start_record,	/* ad_start_record() */
-	audiots_ad_stop_record,		/* ad_stop_record() */
-	NULL,				/* ad_ioctl() */
-	NULL				/* ad_iocdata() */
-};
-
-/*
- * STREAMS Structures
- */
-
-/* STREAMS driver id and limit value structure */
-static struct module_info audiots_modinfo = {
-	TS_IDNUM,		/* module ID number */
-	TS_NAME,		/* module name */
-	TS_MINPACKET,		/* minimum packet size */
-	TS_MAXPACKET,		/* maximum packet size */
-	TS_HIWATER,		/* high water mark */
-	TS_LOWATER		/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit audiots_rqueue = {
-	audio_sup_rput,		/* put procedure */
-	audio_sup_rsvc,		/* service procedure */
-	audio_sup_open,		/* open procedure */
-	audio_sup_close,	/* close procedure */
-	NULL,			/* unused */
-	&audiots_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit audiots_wqueue = {
-	audio_sup_wput,		/* put procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&audiots_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab audiots_str_info = {
-	&audiots_rqueue,	/* read queue */
-	&audiots_wqueue,	/* write queue */
-	NULL,			/* mux lower read queue */
-	NULL,			/* mux lower write queue */
-};
-
-/*
- * DDI Structures
- */
-
-/* Entry points structure */
-static struct cb_ops audiots_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&audiots_str_info,	/* cb_str */
-	D_NEW|D_MP|D_64BIT,	/* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_arwite */
-};
-
-/* Device operations structure */
-static struct dev_ops audiots_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	audiots_getinfo,	/* devo_getinfo */
-	nulldev,		/* devo_identify - obsolete */
-	nulldev,		/* devo_probe */
-	audiots_attach,		/* devo_attach */
-	audiots_detach,		/* devo_detach */
-	nodev,			/* devo_reset */
-	&audiots_cb_ops,	/* devi_cb_ops */
-	NULL,			/* devo_bus_ops */
-	audiots_power,		/* devo_power */
-	ddi_quiesce_not_supported,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv audiots_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	TS_MOD_NAME,		/* drv_linkinfo */
-	&audiots_dev_ops	/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage audiots_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&audiots_modldrv,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-
-/*
- * NOTE: Grover OBP v4.0.166 and rev G of the ALI Southbridge chip force the
- * audiots driver to use the upper 2 GB DMA address range. However to maintain
- * backwards compatibility with older systems/OBP, we're going to try the full
- * 4 GB DMA range.
- *
- * Eventually, this will be set back to using the proper high 2 GB DMA range.
- */
-
-/* Device attribute structure - full 4 gig address range */
-static ddi_dma_attr_t audiots_attr = {
-	DMA_ATTR_VERSION,		/* version */
-	0x0000000000000000LL,		/* dlim_addr_lo */
-	0x00000000ffffffffLL,		/* dlim_addr_hi */
-	0x0000000000003fffLL,		/* DMA counter register - 16 bits */
-	0x0000000000000008LL,		/* DMA address alignment, 64-bit */
-	0x0000007f,			/* 1 through 64 byte burst sizes */
-	0x00000001,			/* min effective DMA size */
-	0x0000000000003fffLL,		/* maximum transfer size, 16k */
-	0x000000000000ffffLL,		/* segment boundary, 64k */
-	0x00000001,			/* s/g list length, no s/g */
-	0x00000001,			/* granularity of device, don't care */
-	0				/* DMA flags */
-};
-
-static ddi_device_acc_attr_t ts_acc_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_LE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-
-/* *******  Loadable Module Configuration Entry Points  ********************* */
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *	This is how access is initially given to all the static structures.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int		error;
-
-	ATRACE("in audiots _init()", NULL);
-
-	/* initialize the soft state */
-	if ((error = ddi_soft_state_init(&audiots_statep,
-	    sizeof (audiots_state_t), 1)) != 0) {
-		ATRACE("audiots ddi_soft_state_init() failed", audiots_statep);
-		return (error);
-	}
-
-	if ((error = mod_install(&audiots_modlinkage)) != 0) {
-		ddi_soft_state_fini(&audiots_statep);
-	}
-
-	ATRACE("audiots _init() audiots_statep", audiots_statep);
-
-	ATRACE_32("audiots _init() returning", error);
-
-	return (error);
-}
-
-/*
- * _fini()
- *
- * Description:
- *	Module de-initialization, called when the driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int		error;
-
-	ATRACE("in audiots _fini()", audiots_statep);
-
-	if ((error = mod_remove(&audiots_modlinkage)) != 0) {
-		return (error);
-	}
-
-	/* free the soft state internal structures */
-	ddi_soft_state_fini(&audiots_statep);
-
-	ATRACE_32("audiots _fini() returning", error);
-
-	return (0);
-}
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns infomation about the driver.
- *
- * Arguments:
- *	modinfo *modinfop	Pointer to the opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int		error;
-
-	ATRACE("in audiots _info()", NULL);
-
-	error = mod_info(&audiots_modlinkage, modinfop);
-
-	ATRACE_32("audiots _info() returning", error);
-
-	return (error);
-}
-
-
-/* *******  Driver Entry Points  ******************************************** */
-
-/*
- * audiots_getinfo()
- */
-/*ARGSUSED*/
-static int
-audiots_getinfo(dev_info_t *dip, ddi_info_cmd_t infocmd,
-	void *arg, void **result)
-{
-	audiots_state_t	*state;
-	int error = DDI_FAILURE;
-	int instance;
-
-	switch (infocmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((state = ddi_get_soft_state(audiots_statep,
-		    instance)) != NULL) {
-			*result = state->ts_dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
-		break;
-	case DDI_INFO_DEVT2INSTANCE:
-		*result =
-		    (void *)(uintptr_t)audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-	default:
-		break;
-	}
-
-	return (error);
-}
-
-/*
- * audiots_attach()
- *
- * Description:
- *	Attach an instance of the audiots driver. This routine does the
- *	device dependent attach tasks. When it is complete it calls
- *	audio_sup_register() and am_attach() so they may do their work.
- *
- *	NOTE: mutex_init() no longer needs a name string, so set
- *		to NULL to save kernel space.
- *
- *	NOTE: audiots_attach() assumes the Codec is powered down.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd	Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-static int
-audiots_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	audiots_state_t		*state;
-	audio_sup_reg_data_t	data;
-	char			*pm_comp[] = {
-		"NAME=audiots audio device",
-		"0=off",
-		"1=on"
-	};
-	int			instance;
-
-	ATRACE("in ts_attach()", dip);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("ts_attach() instance", instance);
-	ATRACE("ts_attach() audiots_statep", audiots_statep);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		break;
-	case DDI_RESUME:
-		ATRACE("ts_attach() DDI_RESUME", NULL);
-
-		/* we've already allocated the state structure so get ptr */
-		if ((state = ddi_get_soft_state(audiots_statep, instance)) ==
-		    NULL) {
-			audio_sup_log(NULL, CE_WARN,
-			    "!%s%d: attach() RESUME get soft state failed",
-			    audiots_name, instance);
-			return (DDI_FAILURE);
-		}
-
-		ASSERT(dip == state->ts_dip);
-		ASSERT(!mutex_owned(&state->ts_lock));
-
-		/* suspend/resume resets the chip, so we have no more faults */
-		if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-			ddi_dev_report_fault(state->ts_dip,
-			    DDI_SERVICE_RESTORED,
-			    DDI_DEVICE_FAULT,
-			    "check port, gain, balance, and mute settings");
-			/* and clear the fault state flags */
-			state->ts_flags &=
-			    ~(TS_AUDIO_READ_FAILED|TS_READ_FAILURE_PRINTED);
-		}
-
-		if (state->ts_flags & TS_PM_SUPPORTED) {
-			/*
-			 * power up the Codec, see comment
-			 * in audiots_set_busy()
-			 */
-			ASSERT(state->ts_powered == TS_PWR_OFF);
-			(void) pm_busy_component(state->ts_dip, TS_COMPONENT);
-			if (pm_raise_power(state->ts_dip, TS_COMPONENT,
-			    TS_PWR_ON) == DDI_FAILURE) {
-				/* match the busy call above */
-				(void) pm_idle_component(state->ts_dip,
-				    TS_COMPONENT);
-				audio_sup_log(state->ts_ahandle, CE_WARN,
-				    "!attach() DDI_RESUME failed");
-				return (DDI_FAILURE);
-			}
-		}
-		mutex_enter(&state->ts_lock);
-
-		state->ts_powered = TS_PWR_ON;
-
-		ASSERT(state->ts_suspended == TS_SUSPENDED);
-		state->ts_suspended = TS_NOT_SUSPENDED;
-
-		/* Restore the audiots chip's state */
-		audiots_chip_init(state, TS_INIT_RESTORE);
-
-		/*
-		 * Put the address engine interrupt enable register in a known
-		 * state - everything off.
-		 */
-
-		ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-		    TS_ALL_DMA_OFF);
-
-		mutex_exit(&state->ts_lock);
-
-		/*
-		 * Start playing and recording, if not needed they'll stop
-		 * on their own. But, we don't start them if the hardware has
-		 * failed.
-		 */
-		if (audio_sup_restore_state(state->ts_ahandle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(state->ts_ahandle, CE_WARN,
-			    "!attach() audio restart failed");
-		}
-
-		cv_broadcast(&state->ts_cv);	/* let entry points continue */
-
-		if (state->ts_flags & TS_PM_SUPPORTED) {
-			/* we're no longer busy */
-			ASSERT(state->ts_powered == TS_PWR_ON);
-			(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-		}
-
-		ATRACE("ts_attach() DDI_RESUME done", NULL);
-
-		ASSERT(!mutex_owned(&state->ts_lock));
-
-		return (DDI_SUCCESS);
-
-	default:
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() unknown command: 0x%x", audiots_name,
-		    instance, cmd);
-		return (DDI_FAILURE);
-	}
-
-	/* before we do anything make sure that we haven't had a h/w failure */
-	if (ddi_get_devstate(dip) == DDI_DEVSTATE_DOWN) {
-		audio_sup_log(NULL, CE_WARN, "%s%d: The audio hardware has "
-		    "been disabled.", audiots_name, instance);
-		audio_sup_log(NULL, CE_CONT, "Please reboot to restore audio.");
-		return (DDI_FAILURE);
-	}
-
-	/* we don't support high level interrupts in this driver */
-	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() unsupported high level interrupt",
-		    audiots_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	/* allocate the state structure */
-	if (ddi_soft_state_zalloc(audiots_statep, instance) == DDI_FAILURE) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() soft state allocate failed",
-		    audiots_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	/*
-	 * WARNING: From here on all errors require that we free memory,
-	 *	including the state structure.
-	 */
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(audiots_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() get soft state failed",
-		    audiots_name, instance);
-		goto error_mem;
-	}
-
-	/* call audiosup module registration routine */
-	ATRACE("ts_attach() calling audio_sup_register()", NULL);
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-	if ((state->ts_ahandle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() audio_sup_register() failed",
-		    audiots_name, instance);
-		goto error_mem;
-	}
-
-	/* initialize the audio state structures */
-	if (audiots_init_state(state, dip) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!attach() init state structure failed");
-		goto error_audiosup;
-	}
-
-	/* map in the registers, allocate DMA buffers, etc. */
-	if (audiots_map_regs(dip, state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!attach() couldn't map registers");
-		goto error_destroy;
-	}
-
-	if (state->ts_rev_id == AC_REV_ID2) {
-		/* set up the pm-components */
-		if (ddi_prop_update_string_array(DDI_DEV_T_NONE, dip,
-		    "pm-components", pm_comp, 3) != DDI_PROP_SUCCESS) {
-			audio_sup_log(state->ts_ahandle, CE_WARN,
-			    "!init_state() couldn't create component");
-			return (DDI_FAILURE);
-		}
-
-		/* Mark PM supported */
-		state->ts_flags |= TS_PM_SUPPORTED;
-
-		/* Mark as powering up at attach time */
-		state->ts_flags |= TS_ATTACH_PWR;
-
-		/* make sure the power framework knows the we are powered up */
-		ASSERT(state->ts_powered == TS_PWR_OFF);
-		(void) pm_busy_component(state->ts_dip, TS_COMPONENT);
-		if (pm_raise_power(state->ts_dip, TS_COMPONENT, TS_PWR_ON) ==
-		    DDI_FAILURE) {
-			(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-			ATRACE("ts_attach() pm_raise_power() failed", NULL);
-			audio_sup_log(state->ts_ahandle, CE_WARN,
-			    "!attach() power up failed");
-			goto error_destroy;
-		}
-		(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-
-		/* Clear the attach time powering flag */
-		state->ts_flags &= ~TS_ATTACH_PWR;
-	} else {
-		/* Mark on for CPR */
-		state->ts_powered = TS_PWR_ON;
-	}
-
-	/* initialize the audio controller and the AC-97 Codec */
-	audiots_chip_init(state, TS_INIT_NO_RESTORE);
-
-	/*
-	 * Put the address engine interrupt enable register in a known
-	 * state - everything off.
-	 */
-
-	ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-	    TS_ALL_DMA_OFF);
-
-	/* save private state */
-	audio_sup_set_private(state->ts_ahandle, state);
-
-	/* call the mixer attach() routine */
-	ATRACE("ts_attach() calling am_attach()", &state->ts_ad_info);
-	if (am_attach(state->ts_ahandle, cmd, &state->ts_ad_info) ==
-	    AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!attach() am_attach() failed");
-		goto error_unmap;
-	}
-
-	/* set up kernel statistics */
-	if ((state->ts_ksp = kstat_create(TS_NAME, instance, TS_NAME,
-	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT)) != NULL) {
-		kstat_install(state->ts_ksp);
-	}
-
-	/* set up the interrupt handler */
-	if (ddi_add_intr(dip, 0, (ddi_iblock_cookie_t *)NULL,
-	    (ddi_idevice_cookie_t *)NULL, audiots_intr,
-	    (caddr_t)state) != DDI_SUCCESS) {
-		ATRACE("ts_attach() bad interrupt spec", state);
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!attach() bad interrupt specification");
-		goto error_kstat;
-	}
-
-	/* everything worked out, so report the device */
-	ddi_report_dev(dip);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (DDI_SUCCESS);
-
-	/*
-	 * CAUTION: Make sure there's an audio_sup_log() call before jumping
-	 * here
-	 */
-
-error_kstat:
-	ATRACE("ts_attach() error_kstat", state);
-	if (state->ts_ksp) {
-		kstat_delete(state->ts_ksp);
-	}
-
-	(void) am_detach(state->ts_ahandle, DDI_DETACH);
-
-error_unmap:
-	ATRACE("ts_attach() error_unmap", state);
-	audiots_unmap_regs(state);
-
-error_destroy:
-	ATRACE("ts_attach() error_destroy", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	kmem_free(state->ts_tcbuf, state->ts_cbuf_size);
-
-	/* destroy the state mutexes and cvs */
-	mutex_destroy(&state->ts_lock);
-	cv_destroy(&state->ts_cv);
-
-error_audiosup:
-	ATRACE("ts_attach() error_audiosup", state);
-	(void) audio_sup_unregister(state->ts_ahandle);
-
-error_mem:
-	ATRACE("ts_attach() error_mem", state);
-	ddi_soft_state_free(audiots_statep, instance);
-
-	ATRACE("ts_attach() returning failure", NULL);
-
-	return (DDI_FAILURE);
-
-}	/* audiots_attach() */
-
-/*
- * audiots_detach()
- *
- * Description:
- *	Detach an instance of the audiots driver. After the Codec is detached
- *	we call am_detach() and audio_sup_unregister() so they may do their
- *	work.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was detached
- *	DDI_FAILURE		The driver couldn't be detached
- */
-static int
-audiots_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
-{
-	audiots_state_t		*state;
-	int			instance;
-
-	ATRACE_32("in ts_detach()", cmd);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("ts_detach() instance", instance);
-	ATRACE("ts_detach() audiots_statep", audiots_statep);
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(audiots_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: detach() get soft state failed",
-		    audiots_name, instance);
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-	case DDI_SUSPEND:
-		ATRACE("ts_detach() DDI_SUSPEND", NULL);
-
-		mutex_enter(&state->ts_lock);
-
-		ASSERT(state->ts_suspended == TS_NOT_SUSPENDED);
-
-		state->ts_suspended = TS_SUSPENDED;	/* stop new ops */
-
-		/* wait for current operations to complete */
-		while (state->ts_busy_cnt != 0) {
-			cv_wait(&state->ts_cv, &state->ts_lock);
-		}
-
-		/* we may already be powered down, so only save state if up */
-		if (state->ts_powered == TS_PWR_ON) {
-
-			/* stop playing and recording */
-			(void) audiots_stop_everything(state);
-
-			/*
-			 * Save the controller state. The Codec's state is
-			 * already in ts_shadow[]. audiots_power_down()
-			 * saves the state and gets us ready to be powered
-			 * back up when we resume.
-			 */
-			if (state->ts_flags & TS_PM_SUPPORTED) {
-				audiots_power_down(state);
-			} else {
-				audiots_save_controller(state);
-			}
-		}
-
-		if (audio_sup_save_state(state->ts_ahandle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(state->ts_ahandle, CE_WARN,
-			    "!detach() audio save failed");
-		}
-
-		mutex_exit(&state->ts_lock);
-
-		ATRACE("ts_detach() SUSPEND successful", state);
-
-		ASSERT(!mutex_owned(&state->ts_lock));
-
-		return (DDI_SUCCESS);
-
-	default:
-		ATRACE_32("ts_detach() unknown command failure", cmd);
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!detach() unknown command: 0x%x", cmd);
-		ASSERT(!mutex_owned(&state->ts_lock));
-		return (DDI_FAILURE);
-	}
-
-	/* Make sure play and record are stopped and disable all interrupts */
-	ATRACE("ts_detach() stopping all DMA engines before detaching", 0);
-	mutex_enter(&state->ts_lock);
-	audiots_stop_everything(state);
-	mutex_exit(&state->ts_lock);
-
-	if (state->ts_flags & TS_PM_SUPPORTED) {
-		/*
-		 * power down the device, no reason to waste power
-		 * without a driver
-		 */
-		(void) pm_lower_power(state->ts_dip, TS_COMPONENT, TS_PWR_OFF);
-	}
-
-	/* remove the interrupt handler */
-	ATRACE("ts_detach() removing interrupt handler", state);
-	ddi_remove_intr(dip, 0, NULL);
-
-	/* free the kernel statistics structure */
-	if (state->ts_ksp) {
-		kstat_delete(state->ts_ksp);
-	}
-
-	/*
-	 * Call the mixer detach routine to tear down the mixer before
-	 * we lose the hardware.
-	 */
-	ATRACE("ts_detach() calling am_detach()", dip);
-	(void) am_detach(state->ts_ahandle, cmd);
-
-	/*
-	 * Now call the audio support module's detach routine to remove this
-	 * driver completely from the audio driver architecture.
-	 */
-	ATRACE("ts_detach() calling audio_sup_unregister()", dip);
-	(void) audio_sup_unregister(state->ts_ahandle);
-
-	/* unmap the registers */
-	audiots_unmap_regs(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* free temporary capture buffer */
-	kmem_free(state->ts_tcbuf, state->ts_cbuf_size);
-
-	/* destroy the state mutexes and cvs */
-	mutex_destroy(&state->ts_lock);
-	cv_destroy(&state->ts_cv);
-
-	/* free the memory for the state pointer */
-	ddi_soft_state_free(audiots_statep, instance);
-
-	ATRACE("ts_detach() returning success", audiots_statep);
-
-	return (DDI_SUCCESS);
-
-}	/* audiots_detach() */
-
-
-/*
- * audiots_power()
- *
- * Description:
- *	This routine is used to turn the power to the Codec and audio core
- *	on and off. The Codec's registers are always saved, however, when
- *	we power down we have to save the audio core's state. When powering
- *	on we restore both Codec and core state via audiots_chip_init().
- *
- *	This routine doesn't worry about starting or stopping audio, other
- *	routines have that responsibility.
- *
- * Arguments:
- *	def_info_t	*dip		Ptr to the device's dev_info structure
- *	int		component	Which component to power up/down
- *	int		level		The power level for the component
- *
- * Returns:
- *	DDI_SUCCESS			Power level changed
- *	DDI_FAILURE			Power level didn't change
- */
-/*ARGSUSED*/
-static int
-audiots_power(dev_info_t *dip, int component, int level)
-{
-	audiots_state_t		*state;
-	int			instance;
-	int			rc = DDI_FAILURE;
-
-	ATRACE("in ts_power()", dip);
-	ATRACE("ts_power() audiots_statep", audiots_statep);
-	ATRACE_32("ts_power() component", component);
-	ATRACE_32("ts_power() level", level);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("ts_power() instance", instance);
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(audiots_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: power() get soft state failed", audiots_name,
-		    instance);
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-	ASSERT(component == 0);
-
-	mutex_enter(&state->ts_lock);
-
-	/* PM should be enabled */
-	ASSERT(state->ts_flags & TS_PM_SUPPORTED);
-
-	/* check the level change to see what we need to do */
-	if (level == TS_PWR_OFF && state->ts_powered == TS_PWR_ON) {
-		ATRACE("ts_power() powering down", NULL);
-
-		/* don't power off if we're busy */
-		if (ddi_get32(state->ts_handle,
-		    &state->ts_regs->aud_regs.ap_start) ||
-		    state->ts_busy_cnt) {
-			/* device is busy, so don't power off */
-			mutex_exit(&state->ts_lock);
-
-			/* reset the timer */
-			(void) pm_idle_component(dip, TS_COMPONENT);
-
-			ATRACE("ts_power() power off failed, busy",
-			    state->ts_busy_cnt);
-			ASSERT(rc == DDI_FAILURE);
-
-			goto done;
-		}
-
-		/* power down and save the state */
-		audiots_power_down(state);
-
-		ATRACE("ts_power() power down complete", NULL);
-	} else if (level == TS_PWR_ON && state->ts_powered == TS_PWR_OFF) {
-		ATRACE("ts_power() powering up", NULL);
-
-		audiots_power_up(state);
-
-		ATRACE("ts_power() power up complete", NULL);
-#ifdef DEBUG
-	} else {
-		ATRACE_32("ts_power() no change to make", level);
-#endif
-	}
-
-	mutex_exit(&state->ts_lock);
-
-	rc = DDI_SUCCESS;
-
-done:
-
-	ATRACE("ts_power() done", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_power() */
-
-/* *******  DDAudio Entry Point Routines ************************************ */
-
-/*
- * audiots_ad_setup()
- *
- * Description:
- *	This routine checks whether the audio hardware has failed. If so, it
- *	returns AUDIO_FAILURE and opens are blocked. Otherwise, it lets
- *	everything proceed as normal.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		dir		Direction of audio, we don't care here
- *
- * Returns:
- *	AUDIO_SUCCESS		The audio hardware is working
- *	AUDIO_FAILURE		The audio hardware has failed
- */
-/*ARGSUSED*/
-static int
-audiots_ad_setup(audiohdl_t ahandle, int stream, int dir)
-{
-	audiots_state_t		*state;
-	int			rc = AUDIO_SUCCESS;
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	ATRACE_32("in audiots_ad_setup(), instance", state->ts_instance);
-
-	/* Check if the hardware has failed */
-	mutex_enter(&state->ts_lock);
-	if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-		ATRACE_32("ts_ad_setup() h/w has failed", state->ts_flags);
-		rc = AUDIO_FAILURE;
-	}
-	mutex_exit(&state->ts_lock);
-
-	ATRACE_32("ts_ad_setup() returning", rc);
-
-	return (rc);
-
-}	/* audiots_ad_setup() */
-
-/*
- * audiots_ad_pause_play()
- *
- * Description:
- *	This routine pauses the play DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-audiots_ad_pause_play(audiohdl_t ahandle, int stream)
-{
-	audiots_state_t		*state;
-	ddi_acc_handle_t	handle;
-
-	ATRACE("in ts_ad_pause_play()", ahandle);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!pause_play() set_busy() failed");
-		return;
-	}
-
-	mutex_enter(&state->ts_lock);
-
-	handle = state->ts_handle;
-
-	/* we don't do anything if we aren't already running */
-	if (!(ddi_get32(handle, &state->ts_regs->aud_regs.ap_start) &
-	    TS_OUTPUT_CHANNEL)) {
-		ATRACE("ts_pause_play() DMA engine already stopped", state);
-		goto done;
-	}
-
-	ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_stop,
-	    TS_OUTPUT_CHANNEL);
-	state->ts_flags |= TS_DMA_ENGINE_PAUSED;
-
-done:
-	mutex_exit(&state->ts_lock);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE("ts_ad_pause_play() done", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-}	/* audiots_ad_pause_play() */
-
-/*
- * audiots_ad_set_config()
- *
- * Description:
- *	This routine is used to set new Codec parameters, except the data
- *	format which has it's own routine. If the Codec doesn't support a
- *	particular parameter and it is asked to set it then we return
- *	AUDIO_FAILURE.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		command		The configuration to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		Argument #1
- *	int		arg2		Argument #2, not always needed
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set, or the
- *				parameter couldn't be set
- */
-/*ARGSUSED*/
-static int
-audiots_ad_set_config(audiohdl_t ahandle, int stream, int command, int dir,
-    int arg1, int arg2)
-{
-	audiots_state_t		*state;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE_32("ts_ad_set_config() stream", stream);
-	ATRACE_32("ts_ad_set_config() command", command);
-	ATRACE_32("ts_ad_set_config() dir", dir);
-	ATRACE_32("ts_ad_set_config() arg1", arg1);
-	ATRACE_32("ts_ad_set_config() arg2", arg2);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* Check if the hardware has failed */
-	mutex_enter(&state->ts_lock);
-	if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-		mutex_exit(&state->ts_lock);
-		ATRACE_32("ts_ad_set_config() h/w has failed", state->ts_flags);
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!set_config() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * CAUTION: From here on we must goto done to exit.
-	 */
-
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, of balance on gain.
-		 *
-		 *	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 *	arg1 --> gain
-		 *	arg2 --> channel #, 0 == left, 1 == right
-		 */
-		rc = audiots_set_gain(state, stream, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of in ports, as well as which ports
-		 * are modifiable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 *	arg1 --> port bit pattern
-		 *	arg2 --> not used
-		 */
-		rc = audiots_set_port(state, dir, arg1);
-		break;
-
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 *	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> gain
-		 *	arg2 --> not used
-		 */
-		rc = audiots_set_monitor_gain(state, arg1);
-		break;
-
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> ~0 == mute, 0 == enable
-		 *	arg2 --> not used
-		 */
-
-		if (arg1) {	/* mute */
-			mutex_enter(&state->ts_lock);
-			audiots_or_ac97(state, AC97_MASTER_VOLUME_REGISTER,
-			    MVR_MUTE);
-			audiots_or_ac97(state, AC97_HEADPHONE_VOLUME_REGISTER,
-			    HPVR_MUTE);
-			audiots_or_ac97(state, AC97_MONO_MASTER_VOLUME_REGSITER,
-			    MMVR_MUTE);
-			mutex_exit(&state->ts_lock);
-		} else {	/* not muted */
-			/* by setting the port we unmute only active ports */
-			(void) audiots_set_port(state, AUDIO_PLAY,
-			    TS_PORT_UNMUTE);
-		}
-
-		rc = AUDIO_SUCCESS;
-		break;
-
-	case AM_MIC_BOOST:
-		/*
-		 * Enable or disable the mic's 20 dB boost preamplifier.
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> ~0 == enable, 0 == disabled
-		 *	arg2 --> not used
-		 */
-		mutex_enter(&state->ts_lock);
-
-		if (arg1) {	/* enable */
-			audiots_or_ac97(state, AC97_MIC_VOLUME_REGISTER,
-			    MICVR_20dB_BOOST);
-			state->ts_ad_info.ad_add_mode |= AM_ADD_MODE_MIC_BOOST;
-		} else {	/* disable */
-			audiots_and_ac97(state, AC97_MIC_VOLUME_REGISTER,
-			    ~MICVR_20dB_BOOST);
-			state->ts_ad_info.ad_add_mode &= ~AM_ADD_MODE_MIC_BOOST;
-		}
-
-		mutex_exit(&state->ts_lock);
-
-		rc = AUDIO_SUCCESS;
-		break;
-
-	case AM_SET_DIAG_MODE:
-		/*
-		 * Set the loopback diagnostics mode.
-		 *
-		 *	arg1 --> 1 == diagnostics on, 0 == diagnostics off
-		 *	arg2 --> not used
-		 */
-		mutex_enter(&state->ts_lock);
-
-		if (arg1) {
-			state->ts_ad_info.ad_add_mode |= AM_ADD_MODE_DIAG_MODE;
-		} else {
-			state->ts_ad_info.ad_add_mode &= ~AM_ADD_MODE_DIAG_MODE;
-		}
-
-		mutex_exit(&state->ts_lock);
-
-		rc = AUDIO_SUCCESS;
-		break;
-
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 */
-		ATRACE_32("ts_ad_set_config() unsupported command", command);
-		break;
-	}
-
-done:
-	ATRACE_32("ts_ad_set_config() done", rc);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE_32("ts_ad_set_config() returning", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_ad_set_config() */
-
-/*
- * audiots_ad_set_format()
- *
- * Description:
- *	This routine is used to set a new audio control data format.
- *	We only support 8 and 16 bit signed linear.
- *
- *	NOTE: We don't support mono or 8-bit. See the WARNING at the
- *		top of the file.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		sample_rate	Data sample rate
- *	int		channels	Number of channels, 1 or 2
- *	int		precision	Bits per sample, 8 or 16
- *	int		encoding	Encoding method, u-law, A-law and linear
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec data format has been set
- *	AUDIO_FAILURE		The Codec data format has not been set, or the
- *				data format couldn't be set
- */
-/*ARGSUSED*/
-static int
-audiots_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-    int sample_rate, int channels, int precision, int encoding)
-{
-	audiots_state_t		*state;
-	ddi_acc_handle_t	handle;
-	int			ch;
-	int			rc = AUDIO_FAILURE;
-	uint16_t		sign;
-	uint16_t		tmp_short;
-
-	ATRACE_32("ts_ad_set_format() stream", stream);
-	ATRACE_32("ts_ad_set_format() dir", dir);
-	ATRACE_32("ts_ad_set_format() sample rate", sample_rate);
-	ATRACE_32("ts_ad_set_format() channels", channels);
-	ATRACE_32("ts_ad_set_format() precision", precision);
-	ATRACE_32("ts_ad_set_format() encoding", encoding);
-
-	/*
-	 * first, check the encoding method
-	 */
-	if (encoding != AUDIO_ENCODING_LINEAR) {
-		ATRACE_32("ts_ad_set_format() bad encoding", encoding);
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * get the state structure
-	 */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* Check if the hardware has failed */
-	mutex_enter(&state->ts_lock);
-	if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-		mutex_exit(&state->ts_lock);
-		ATRACE_32("ts_ad_set_format() h/w has failed", state->ts_flags);
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!set_format() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * CAUTION: From here on we must goto done to exit.
-	 */
-
-	switch (sample_rate) {
-	case TS_SAMPR5510:		break;
-	case TS_SAMPR6620:		break;
-	case TS_SAMPR8000:		break;
-	case TS_SAMPR9600:		break;
-	case TS_SAMPR11025:		break;
-	case TS_SAMPR16000:		break;
-	case TS_SAMPR18900:		break;
-	case TS_SAMPR22050:		break;
-	case TS_SAMPR27420:		break;
-	case TS_SAMPR32000:		break;
-	case TS_SAMPR33075:		break;
-	case TS_SAMPR37800:		break;
-	case TS_SAMPR44100:		break;
-	case TS_SAMPR48000:		break;
-	default:
-		ATRACE_32("ts_ad_set_format() bad SR", sample_rate);
-		goto done;
-	}
-
-	/* can't fail, so update the saved format and implement */
-	mutex_enter(&state->ts_lock);
-	if (dir == AUDIO_PLAY) {
-		/* save for later use */
-		state->ts_psample_rate = sample_rate;
-		state->ts_pchannels = channels;
-		state->ts_pprecision = precision;
-
-		/*
-		 * first find the h/w channel with the stream
-		 */
-		ch = TS_OUTPUT_STREAM;
-
-		/*
-		 * convert the sample rate into 4.12 format
-		 */
-		sample_rate = (sample_rate << TS_SRC_SHIFT) / TS_SAMPR48000;
-
-		/* for play we always use signed */
-		sign = ERAM_SIGNED_PCM;
-
-		ATRACE_32("ts_ad_set_format() play SR", sample_rate);
-	} else {
-		/* save for later use */
-		state->ts_csample_rate = sample_rate;
-		state->ts_cchannels = channels;
-		state->ts_cprecision = precision;
-
-		/*
-		 * we set the only record stream
-		 */
-		ch = TS_INPUT_STREAM;
-
-		/*
-		 * convert the sample rate into 4.12 format
-		 */
-		sample_rate = (TS_SAMPR48000 << TS_SRC_SHIFT) / sample_rate;
-
-		ATRACE_32("ts_ad_set_format() record SR", sample_rate);
-
-		/* for 16-bit record we use signed */
-		sign = ERAM_SIGNED_PCM;
-	}
-
-	/* check for stereo only */
-	ASSERT(channels == AUDIO_CHANNELS_STEREO);
-	channels = ERAM_STEREO;
-
-	/* check for 16-bit only */
-	ASSERT(precision == AUDIO_PRECISION_16);
-	precision = ERAM_16_BITS;
-
-	/* always linear encoding */
-	ASSERT(encoding == AUDIO_ENCODING_LINEAR);
-
-	/* program the sample rate */
-	handle = state->ts_handle;
-	ddi_put16(handle, &state->ts_regs->aud_ram[ch].aram.aram_delta,
-	    (uint16_t)sample_rate);
-
-	/* program the precision and number of channels */
-	tmp_short = ddi_get16(handle,
-	    &state->ts_regs->aud_ram[ch].eram.eram_ctrl_ec) & ~(ERAM_CTRL_MASK);
-	tmp_short |= precision|channels|ERAM_LOOP_MODE|sign;
-	ddi_put16(handle, &state->ts_regs->aud_ram[ch].eram.eram_ctrl_ec,
-	    tmp_short);
-
-	mutex_exit(&state->ts_lock);
-
-	ATRACE("ts_ad_set_format() finished programming the device", 0);
-
-	rc = AUDIO_SUCCESS;
-
-done:
-	ATRACE_32("ts_ad_set_format() done", rc);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE_32("ts_ad_set_format() returning", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_ad_set_format() */
-
-/*
- * audiots_ad_start_play()
- *
- * Description:
- *	Wrapper to call audiots_start_play().
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Play not started/restarted, no audio to play
- */
-/*ARGSUSED*/
-static int
-audiots_ad_start_play(audiohdl_t ahandle, int stream)
-{
-	audiots_state_t		*state;
-	int			rc;
-
-	ATRACE("in ts_ad_start_play() handle", ahandle);
-	ATRACE_32("ts_ad_start_play() stream", stream);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* Check if the hardware has failed */
-	mutex_enter(&state->ts_lock);
-	if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-		mutex_exit(&state->ts_lock);
-		ATRACE_32("ts_ad_start_play() h/w has failed", state->ts_flags);
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!start_play() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	mutex_enter(&state->ts_lock);
-	rc = audiots_start_play(state);
-
-	if (rc == AUDIO_SUCCESS && (state->ts_flags & TS_PLAY_ACTIVE) == 0) {
-		state->ts_flags |= TS_PLAY_ACTIVE;
-
-		if (state->ts_flags & TS_PM_SUPPORTED) {
-			(void) pm_busy_component(state->ts_dip, TS_COMPONENT);
-		}
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE_32("ts_ad_start_play() returning", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_ad_start_play() */
-
-/*
- * audiots_ad_stop_play()
- *
- * Description:
- *	Wrapper to call audiots_stop_play().
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-audiots_ad_stop_play(audiohdl_t ahandle, int stream)
-{
-	audiots_state_t		*state;
-
-	ATRACE("in ts_ad_stop_play()", ahandle);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!stop_play() set_busy() failed");
-		return;
-	}
-
-	mutex_enter(&state->ts_lock);
-	audiots_stop_play(state);
-
-	if ((state->ts_flags & (TS_PM_SUPPORTED|TS_PLAY_ACTIVE)) ==
-	    (TS_PM_SUPPORTED|TS_PLAY_ACTIVE)) {
-		(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-	}
-
-	state->ts_flags &= ~TS_PLAY_ACTIVE;
-
-	mutex_exit(&state->ts_lock);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE("ts_ad_stop_play() returning", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-}	/* audiots_ad_stop_play() */
-
-/*
- * audiots_ad_start_record()
- *
- * Description:
- *	Wrapper to call audiots_start_record().
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which isn't going to apply for record
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Record not started
- */
-/*ARGSUSED*/
-static int
-audiots_ad_start_record(audiohdl_t ahandle, int stream)
-{
-	audiots_state_t		*state;
-	int			rc;
-
-	ATRACE("in ts_ad_start_record()", ahandle);
-	ATRACE_32("ts_ad_start_record() stream", stream);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* Check if the hardware has failed */
-	mutex_enter(&state->ts_lock);
-	if (state->ts_flags & TS_AUDIO_READ_FAILED) {
-		mutex_exit(&state->ts_lock);
-		ATRACE_32("ts_ad_start_record() h/w has failed",
-		    state->ts_flags);
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!start_record() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	mutex_enter(&state->ts_lock);
-	rc = audiots_start_record(state);
-	mutex_exit(&state->ts_lock);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ATRACE_32("ts_ad_start_record() returning", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_ad_start_record() */
-
-/*
- * audiots_ad_stop_record()
- *
- * Description:
- *	Wrapper to call audiots_stop_record().
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which isn't going to apply for record
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-audiots_ad_stop_record(audiohdl_t ahandle, int stream)
-{
-	audiots_state_t		*state;
-
-	ATRACE("in ts_ad_stop_record()", ahandle);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* wait on suspend, power up and mark as busy */
-	if (audiots_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!stop_record() set_busy() failed");
-		return;
-	}
-
-	mutex_enter(&state->ts_lock);
-	audiots_stop_record(state);
-
-	if (state->ts_flags & TS_PM_SUPPORTED) {
-		(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-	}
-	mutex_exit(&state->ts_lock);
-
-	/* we're no longer busy */
-	audiots_set_idle(state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-}	/* audiots_ad_stop_record() */
-
-
-/* ******* Local Routines *************************************************** */
-
-/*
- * audiots_and_ac97()
- *
- * Description:
- *	Logically AND a value with the specified AC-97 Codec register.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		reg		AC-97 register number
- *	uint16_t	data		The value to AND
- *
- * Returns:
- *	void
- */
-static void
-audiots_and_ac97(audiots_state_t *state, int reg, uint16_t data)
-{
-	audiots_set_ac97(state, reg,
-	    (data & state->ts_shadow[TS_CODEC_REG(reg)]));
-
-}	/* audiots_and_ac97() */
-
-/*
- * audiots_chip_init()
- *
- * Description:
- *	Initialize the audio core and the AC-97 Codec. The AC-97 Codec is
- *	always programmed from ts_shadow[]. If we aren't doing a restore
- *	we initialize ts_shadow[], otherwise we use the current values of
- *	ts_shadow[]. This allows this routine to be used for both attaching
- *	and for power management power up.
- *
- *	Speaker, line out and headphone out gain are not set to 0 gain.
- *	Thus their respective registers are never set to 0. Since the failed
- *	read and write over the AC-97 link results in a 0 we now know for
- *	sure if the write succeeded or not.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		restore		If TS_INIT_RESTORE then restore
- *					from ts_shadow[]
- *
- * Returns:
- *	void
- */
-static void
-audiots_chip_init(audiots_state_t *state, int restore)
-{
-	ddi_acc_handle_t	handle = state->ts_handle;
-	int			i;
-	int			str;
-
-	ATRACE("in ts_chip_init()", state);
-	ATRACE_32("ts_chip_init() restore", restore);
-
-	/* set global music and wave volume to 0dB */
-	ddi_put32(handle, &state->ts_regs->aud_regs.ap_volume, 0x0);
-
-	/*
-	 * Enable middle and end interrupts for all channels. Since
-	 * we always set these we don't have to save it as well.
-	 */
-	ddi_put32(handle, &state->ts_regs->aud_regs.ap_cir_gc,
-	    (AP_CIR_GC_ENDLP_IE|AP_CIR_GC_MIDLP_IE));
-
-	/* for each channel, set gain and enable interrupts for middle & end */
-	for (str = 0; str < TS_MAX_HW_CHANNELS; str++) {
-		/*
-		 * Set volume to 0dB attenuation, 1st left and then right.
-		 * These are never changed, so we don't have to save them.
-		 */
-		ddi_put16(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_gvsel_pan_vol,
-		    (ERAM_WAVE_VOL|ERAM_PAN_LEFT|ERAM_PAN_0dB|
-		    ERAM_VOL_DEFAULT));
-		ddi_put16(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_gvsel_pan_vol,
-		    (ERAM_WAVE_VOL|ERAM_PAN_RIGHT|ERAM_PAN_0dB|
-		    ERAM_VOL_DEFAULT));
-
-		/*
-		 * The envelope engine *MUST* remain in still mode (off).
-		 * Otherwise bad things like gain randomly disappearing might
-		 * happen. See bug #4332773.
-		 */
-
-		ddi_put32(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_ebuf1,
-		    ERAM_EBUF_STILL);
-		ddi_put32(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_ebuf2,
-		    ERAM_EBUF_STILL);
-
-		/* Set the eram and aram state */
-		ddi_put16(handle,
-		    &state->ts_regs->aud_ram[str].aram.aram_delta,
-		    state->ts_save_regs[str].aram_delta);
-		ddi_put16(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_ctrl_ec,
-		    state->ts_save_regs[str].eram_ctrl_ec);
-	}
-
-	/* program channel 31 for record */
-	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_global_control,
-	    (AP_CLOGAL_CTRL_E_PCMIN_CH31|AP_CLOGAL_CTRL_PCM_OUT_AC97|
-	    AP_CLOGAL_CTRL_MMC_FROM_MIXER|AP_CLOGAL_CTRL_PCM_OUT_TO_AC97));
-
-	/* do a warm reset, which powers up the Codec */
-	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_sctrl,
-	    AP_SCTRL_WRST_CODEC);
-	drv_usecwait(2);
-	AND_SET_WORD(handle, &state->ts_regs->aud_regs.ap_sctrl,
-	    ~AP_SCTRL_WRST_CODEC);
-
-	/* do a warm reset via the Codec, yes, I'm being paranoid! */
-	audiots_set_ac97(state, AC97_RESET_REGISTER, 0);
-
-	/* Make sure the Codec is powered up. */
-	i = TS_WAIT_CNT;
-	while ((audiots_get_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER) &
-	    PCSR_POWERD_UP) != PCSR_POWERD_UP && i--) {
-		drv_usecwait(1);
-	}
-
-	if ((restore == TS_INIT_RESTORE) && (state->ts_flags & TS_ATTACH_PWR)) {
-		/*
-		 * Here we're raising power at attach time, we need to
-		 * initialize the shadow array and then mute the outputs.
-		 * If we don't, there's a loud double pop on the headphones.
-		 */
-		for (str = 0; str < sizeof (ac97_v21_t); str += 2) {
-			(void) audiots_get_ac97(state, str);
-		}
-
-		/* set outputs muted */
-		state->ts_shadow[TS_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE;
-		state->ts_shadow[TS_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-		state->ts_shadow
-		    [TS_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE;
-		state->ts_shadow[TS_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_MUTE;
-	}
-
-	/* preload shadow registers if not restoring */
-	if (restore == TS_INIT_NO_RESTORE) {
-		for (str = 0; str < sizeof (ac97_v21_t); str += 2) {
-			/* read the Codec & save in the shadow register array */
-			(void) audiots_get_ac97(state, str);
-		}
-
-		/* 02h - set master line out volume, muted, 0dB */
-		state->ts_shadow[TS_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE|TS_AC97_ATTEN_LINE;
-
-		/* 04h - set alternate line out volume, muted, 0dB */
-		state->ts_shadow[TS_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE|TS_AC97_ATTEN_HP;
-
-		/* 06h - set master mono volume, muted, 0dB */
-		state->ts_shadow[
-		    TS_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE|TS_AC97_ATTEN_SPKR;
-
-		/* 08h - set master tone control to no modification */
-		state->ts_shadow[
-		    TS_CODEC_REG(AC97_MASTER_TONE_CONTROL_REGISTER)] =
-		    MTCR_BASS_BYPASS|MTCR_TREBLE_BYPASS;
-
-		/* 0ah - mute pc beep, 0dB */
-		state->ts_shadow[TS_CODEC_REG(AC97_PC_BEEP_REGISTER)] =
-		    PCBR_MUTE|PCBR_0dB_ATTEN;
-
-		/* 0ch - set phone input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_PHONE_VOLUME_REGISTER)] =
-		    PVR_MUTE|PVR_0dB_GAIN;
-
-		/* 0eh - set mic input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_MIC_VOLUME_REGISTER)] =
-		    MICVR_MUTE|MICVR_0dB_GAIN;
-
-		/* 10h - set line input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_LINE_IN_VOLUME_REGISTER)] =
-		    LIVR_MUTE|LIVR_RIGHT_0dB_GAIN|LIVR_LEFT_0dB_GAIN;
-
-		/* 12h - set cd input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_CD_VOLUME_REGISTER)] =
-		    CDVR_MUTE|CDVR_RIGHT_0dB_GAIN|CDVR_LEFT_0dB_GAIN;
-
-		/* 14h - set video input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_VIDEO_VOLUME_REGISTER)] =
-		    VIDVR_MUTE|VIDVR_RIGHT_0dB_GAIN|VIDVR_LEFT_0dB_GAIN;
-
-		/* 16h - set aux input, mute, 0dB attenuation */
-		state->ts_shadow[TS_CODEC_REG(AC97_AUX_VOLUME_REGISTER)] =
-		    AUXVR_MUTE|AUXVR_RIGHT_0dB_GAIN|AUXVR_LEFT_0dB_GAIN;
-
-		/* 18h - set PCM out input, NOT muted, 0dB gain */
-		state->ts_shadow[TS_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_RIGHT_0dB_GAIN|PCMOVR_LEFT_0dB_GAIN;
-
-		/* 1ah - set input device as mic */
-		state->ts_shadow[
-		    TS_CODEC_REG(AC97_RECORD_SELECT_CTRL_REGISTER)] =
-		    RSCR_R_MIC|RSCR_L_MIC;
-
-		/* 1ch - set record gain to 0dB and not muted */
-		state->ts_shadow[TS_CODEC_REG(AC97_RECORD_GAIN_REGISTER)] =
-		    RGR_RIGHT_0db_GAIN|RGR_LEFT_0db_GAIN;
-
-		/* 1eh - set record mic gain to 0dB and not muted */
-		state->ts_shadow[TS_CODEC_REG(AC97_RECORD_GAIN_MIC_REGISTER)] =
-		    RGMR_0db_GAIN;
-
-		/* 20h - set GP register, mic 1, everything else off */
-		state->ts_shadow[TS_CODEC_REG(AC97_GENERAL_PURPOSE_REGISTER)] =
-		    GPR_MS_MIC1|GPR_MONO_MIX_IN;
-
-		/* 22h - set 3D control to NULL */
-		state->ts_shadow[TS_CODEC_REG(AC97_THREE_D_CONTROL_REGISTER)] =
-		    TDCR_NULL;
-
-		/*
-		 * The rest we ignore, most are reserved.
-		 *
-		 * CAUTION: If we add to the list we need to fix the end
-		 *	of the loop below.
-		 */
-	}
-
-	/* now program the AC-97 Codec from ts_shadow[] */
-	for (i = 0; i < TS_LAST_AC_REG; i += TS_REG_SIZE) {
-		audiots_set_ac97(state, i, state->ts_shadow[TS_CODEC_REG(i)]);
-	}
-
-	ATRACE("ts_chip_init() returning", state);
-
-}	/* audiots_chip_init() */
-
-/*
- * audiots_get_ac97()
- *
- * Description:
- *	Get the value in the specified AC-97 Codec register. There is a
- *	bug in silicon which forces us to do multiple reads of the Codec's
- *	register. This algorithm was provided by T2 and is heuristic in
- *	nature. Unfortunately we have no guarantees that the real answer
- *	isn't 0x0000, which is what we get when a read fails. So we loop
- *	TS_LOOP_CNT times before we give up. We just have to hope this is
- *	sufficient to give us the correct value.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		reg		AC-97 register number
- *
- * Returns:
- *	unsigned short		The value in the specified register
- */
-static uint16_t
-audiots_get_ac97(audiots_state_t *state, int reg)
-{
-	ddi_acc_handle_t	handle = state->ts_handle;
-	uint16_t		*data;
-	int			count;
-	int			delay;
-	uint16_t		first;
-	uint16_t		next;
-
-	ATRACE_32("in ts_get_ac97()", reg);
-
-	if (state->ts_rev_id == AC_REV_ID1) {
-		data = &state->ts_regs->aud_regs.ap_acrd_35D_data;
-	} else {
-		data = &state->ts_regs->aud_regs.ap_acrdwr_data;
-	}
-
-	/* make sure the register is good */
-	reg &= AP_ACRD_INDEX_MASK;
-	for (count = TS_LOOP_CNT; count--; ) {
-		if ((first = audiots_read_ac97(state, reg)) != 0) {
-			next = first;
-			break;
-		}
-
-		delay = TS_DELAY_CNT;
-		while (delay--) {
-			(void) ddi_get16(handle, data);
-		}
-
-		if ((next = audiots_read_ac97(state, reg)) != 0) {
-			break;
-		}
-	}
-	/* save the value in the shadow register array */
-	state->ts_shadow[TS_CODEC_REG(reg)] = next;
-
-	ATRACE_16("ts_get_ac97() returning",
-	    state->ts_shadow[TS_CODEC_REG(reg)]);
-
-	/*
-	 * Arggg, if you let the next read happen too soon then it fails.
-	 * 12 usec fails, 13 usec succeeds. So set it to 20 for safety.
-	 */
-	drv_usecwait(TS_20US);
-
-	return (state->ts_shadow[TS_CODEC_REG(reg)]);
-
-}	/* audiots_get_ac97() */
-
-/*
- * audiots_init_state()
- *
- * Description:
- *	This routine initializes the audio driver's state structure.
- *	This includes reading the properties.
- *
- *	CAUTION: This routine cannot allocate resources, unless it frees
- *		them before returning for an error. Also, error_destroy:
- *		in audiots_attach() would need to be fixed as well.
- *
- *	NOTE: birdsnest supports CD ROM input. We check for the cdrom
- *		property. If there we turn it on.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	dev_info_t	*dip		Pointer to the device's dev_info struct
- *
- * Returns:
- *	AUDIO_SUCCESS			State structure initialized
- *	AUDIO_FAILURE			State structure not initialized
- */
-static int
-audiots_init_state(audiots_state_t *state, dev_info_t *dip)
-{
-	int			ts_pints;
-	int			ts_rints;
-	int			cdrom = 0;
-
-	ATRACE("in ts_init_state()", NULL);
-
-	/* get the number of play and record interrupts per second */
-	ts_pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "play-interrupts", TS_INTS);
-	ts_rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "record-interrupts", TS_INTS);
-
-	/* make sure the values are good */
-	if (ts_pints < TS_MIN_INTS) {
-		ATRACE_32("ts_init_state() "
-		    "play interrupt rate set too low, resetting", ts_pints);
-		audio_sup_log(state->ts_ahandle, CE_NOTE, "init_state() "
-		    "play interrupt rate set too low, %d, resetting to %d",
-		    ts_pints, TS_MIN_INTS);
-		ts_pints = TS_INTS;
-	} else if (ts_pints > TS_MAX_INTS) {
-		ATRACE_32("ts_init_state() "
-		    "play interrupt rate set too high, resetting", ts_pints);
-		audio_sup_log(state->ts_ahandle, CE_NOTE, "init_state() "
-		    "play interrupt rate set too high, %d, resetting to %d",
-		    ts_pints, TS_MAX_INTS);
-		ts_pints = TS_INTS;
-	}
-	if (ts_rints < TS_MIN_INTS) {
-		ATRACE_32("ts_init_state() "
-		    "record interrupt rate set too low, resetting", ts_rints);
-		audio_sup_log(state->ts_ahandle, CE_NOTE, "init_state() "
-		    "record interrupt rate set too low, %d, resetting to %d",
-		    ts_rints, TS_MIN_INTS);
-		ts_rints = TS_INTS;
-	} else if (ts_rints > TS_MAX_INTS) {
-		ATRACE_32("ts_init_state() "
-		    "record interrupt rate set too high, resetting", ts_rints);
-		audio_sup_log(state->ts_ahandle, CE_NOTE, "init_state() "
-		    "record interrupt rate set too high, %d, resetting to %d",
-		    ts_rints, TS_MAX_INTS);
-		ts_rints = TS_INTS;
-	}
-
-	/* get the mode from the .conf file */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "mixer-mode", AM_MIXER_MODE)) {
-		state->ts_ad_info.ad_mode = AM_MIXER_MODE;
-	} else {
-		state->ts_ad_info.ad_mode = AM_COMPAT_MODE;
-	}
-
-	ATRACE_32("ts_init_state() setting mode", state->ts_ad_info.ad_mode);
-
-	/* figure out the platform */
-	cdrom = ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "cdrom", 0);
-
-	/* save the device info pointer */
-	state->ts_dip = dip;
-
-	/* get the iblock cookie needed for interrupt context */
-	if (ddi_get_iblock_cookie(dip, (uint_t)0, &state->ts_iblock) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!init_state() cannot get iblock cookie");
-		return (AUDIO_FAILURE);
-	}
-
-	/* initialize the state mutexes and condition variables */
-	mutex_init(&state->ts_lock, NULL, MUTEX_DRIVER, state->ts_iblock);
-	cv_init(&state->ts_cv, NULL, CV_DRIVER, NULL);
-
-	/*
-	 * CAUTION: From here on we must destroy the mutexes if there's
-	 *	an error and we return failure.
-	 */
-	/* fill in the device default state */
-	state->ts_defaults.play.sample_rate = TS_DEFAULT_SR;
-	state->ts_defaults.play.channels = TS_DEFAULT_CH;
-	state->ts_defaults.play.precision = TS_DEFAULT_PREC;
-	state->ts_defaults.play.encoding = TS_DEFAULT_ENC;
-	state->ts_defaults.play.gain = TS_DEFAULT_PGAIN;
-	state->ts_defaults.play.port = AUDIO_SPEAKER|AUDIO_HEADPHONE;
-	state->ts_defaults.play.avail_ports = AUDIO_SPEAKER|AUDIO_HEADPHONE|\
-	    AUDIO_LINE_OUT;
-	state->ts_defaults.play.mod_ports = AUDIO_SPEAKER|AUDIO_LINE_OUT;
-	state->ts_defaults.play.buffer_size = TS_BSIZE;
-	state->ts_defaults.play.balance = TS_DEFAULT_BAL;
-	state->ts_defaults.record.sample_rate = TS_DEFAULT_SR;
-	state->ts_defaults.record.channels = TS_DEFAULT_CH;
-	state->ts_defaults.record.precision = TS_DEFAULT_PREC;
-	state->ts_defaults.record.encoding = TS_DEFAULT_ENC;
-	state->ts_defaults.record.gain = TS_DEFAULT_PGAIN;
-	state->ts_defaults.record.port = AUDIO_MICROPHONE;
-	state->ts_defaults.record.avail_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	state->ts_defaults.record.mod_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	state->ts_defaults.record.buffer_size = TS_BSIZE;
-	state->ts_defaults.record.balance = TS_DEFAULT_BAL;
-	state->ts_defaults.monitor_gain = TS_DEFAULT_MONITOR_GAIN;
-	state->ts_defaults.output_muted = B_FALSE;
-	state->ts_defaults.ref_cnt = B_FALSE;
-	state->ts_defaults.hw_features = AUDIO_HWFEATURE_DUPLEX|
-	    AUDIO_HWFEATURE_IN2OUT|AUDIO_HWFEATURE_PLAY|AUDIO_HWFEATURE_RECORD;
-	state->ts_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	/* add CD ROM for birdsnest */
-	if (cdrom) {
-		state->ts_defaults.record.avail_ports |= AUDIO_CD;
-		state->ts_defaults.record.mod_ports |= AUDIO_CD;
-	}
-
-	state->ts_psample_rate = state->ts_defaults.play.sample_rate;
-	state->ts_pchannels = state->ts_defaults.play.channels;
-	state->ts_pprecision = state->ts_defaults.play.precision;
-	state->ts_csample_rate = state->ts_defaults.record.sample_rate;
-	state->ts_cchannels = state->ts_defaults.record.channels;
-	state->ts_cprecision = state->ts_defaults.record.precision;
-
-	/* fill in the ad_info structure */
-	state->ts_ad_info.ad_int_vers = AM_VERSION;
-	state->ts_ad_info.ad_add_mode = NULL;
-	state->ts_ad_info.ad_codec_type = AM_TRAD_CODEC;
-	state->ts_ad_info.ad_defaults = &state->ts_defaults;
-	state->ts_ad_info.ad_play_comb = audiots_combinations;
-	state->ts_ad_info.ad_rec_comb = audiots_combinations;
-	state->ts_ad_info.ad_entry = &audiots_entry;
-	state->ts_ad_info.ad_dev_info = &state->ts_dev_info;
-	state->ts_ad_info.ad_diag_flags = AM_DIAG_INTERNAL_LOOP;
-	state->ts_ad_info.ad_diff_flags =
-	    AM_DIFF_SR|AM_DIFF_CH|AM_DIFF_PREC|AM_DIFF_ENC;
-	state->ts_ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	state->ts_ad_info.ad_misc_flags = AM_MISC_RP_EXCL|AM_MISC_MONO_DUP;
-	state->ts_ad_info.ad_translate_flags = AM_MISC_8_R_TRANSLATE;
-	state->ts_ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	state->ts_ad_info.ad_play.ad_mixer_srs = audiots_mixer_sample_rates;
-	state->ts_ad_info.ad_play.ad_compat_srs = audiots_compat_sample_rates;
-	state->ts_ad_info.ad_play.ad_conv = &am_src2;
-	state->ts_ad_info.ad_play.ad_sr_info = NULL;
-	state->ts_ad_info.ad_play.ad_chs = audiots_channels;
-	state->ts_ad_info.ad_play.ad_int_rate = ts_pints;
-	state->ts_ad_info.ad_play.ad_max_chs = TS_MAX_OUT_CHANNELS;
-	state->ts_ad_info.ad_play.ad_bsize = TS_BSIZE;
-
-	/* record capabilities */
-	state->ts_ad_info.ad_record.ad_mixer_srs = audiots_mixer_sample_rates;
-	state->ts_ad_info.ad_record.ad_compat_srs = audiots_compat_sample_rates;
-	state->ts_ad_info.ad_record.ad_conv = &am_src2;
-	state->ts_ad_info.ad_record.ad_sr_info = NULL;
-	state->ts_ad_info.ad_record.ad_chs = audiots_channels;
-	state->ts_ad_info.ad_record.ad_int_rate = ts_rints;
-	state->ts_ad_info.ad_record.ad_max_chs = TS_MAX_CHANNELS;
-	state->ts_ad_info.ad_record.ad_bsize = TS_BSIZE;
-
-	/* fill in device info strings */
-	(void) strcpy(state->ts_dev_info.name, TS_DEV_NAME);
-	(void) strcpy(state->ts_dev_info.config, TS_DEV_CONFIG);
-	if (!cdrom) {
-		(void) strcpy(state->ts_dev_info.version, TS_DEV_VERSION_A);
-	} else {
-		(void) strcpy(state->ts_dev_info.version, TS_DEV_VERSION_B);
-	}
-
-	/*
-	 * Figure out the largest transfer size for the DMA engine. We then
-	 * double the size because an interrupt happens in the middle and
-	 * the end of the buffer. This size must be modulo 16 to fit two 16-bit
-	 * stereo streams in on the interrupt boundaries, middle and end,
-	 * with an 8-byte boundary.
-	 */
-	state->ts_pbuf_size = TS_BUF_HALVES * TS_SAMPR48000 *
-	    AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / ts_pints;
-	state->ts_pbuf_size += TS_MOD_SIZE -
-	    (state->ts_pbuf_size % TS_MOD_SIZE);
-	ASSERT(state->ts_pbuf_size > 0);
-	ASSERT((state->ts_pbuf_size % TS_MOD_SIZE) == 0);
-
-	state->ts_cbuf_size = TS_BUF_HALVES * TS_SAMPR48000 *
-	    AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / ts_rints;
-	state->ts_cbuf_size += TS_MOD_SIZE -
-	    (state->ts_cbuf_size % TS_MOD_SIZE);
-	ASSERT(state->ts_cbuf_size > 0);
-	ASSERT((state->ts_cbuf_size % TS_MOD_SIZE) == 0);
-	state->ts_tcbuf = kmem_alloc(state->ts_cbuf_size, KM_SLEEP);
-
-	/* init power management state */
-	state->ts_suspended = TS_NOT_SUSPENDED;
-	state->ts_powered = TS_PWR_OFF;
-	state->ts_busy_cnt = 0;
-
-	ATRACE("ts_init_state() success", NULL);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audiots_init_state() */
-
-/*
- * audiots_intr()
- *
- * Description:
- *	Interrupt service routine for both play and record. For play we
- *	get the next buffers worth of audio. For record we send it on to
- *	the mixer.
- *
- *	NOTE: This device needs to make sure any PIO access required to clear
- *	its interrupt has made it out on the PCI bus before returning from its
- *	interrupt handler so that the interrupt has been deasserted. This is
- *	done by rereading the address engine interrupt register.
- *
- * Arguments:
- *	caddr_t		T	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-audiots_intr(caddr_t T)
-{
-	audiots_state_t		*state = (audiots_state_t *)T;
-	ddi_acc_handle_t	handle;
-	char			*bptr;
-	uint16_t		*sptr;
-	uint16_t		*tcbuf;
-	uint32_t		interrupts;
-	uint32_t		location;
-	int			i;
-	int			count;
-	int			samples;
-	int			rc = DDI_INTR_UNCLAIMED;
-
-	ATRACE("in ts_intr()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	mutex_enter(&state->ts_lock);
-
-	handle = state->ts_handle;
-
-	if (state->ts_flags & TS_INTR_PENDING) {
-		state->ts_flags &= ~TS_INTR_PENDING;
-		rc = DDI_INTR_CLAIMED;
-		mutex_exit(&state->ts_lock);
-		ATRACE("ts_intr() servicing with pending intr from stop()", 0);
-		goto done;
-	}
-
-	interrupts = ddi_get32(handle, &state->ts_regs->aud_regs.ap_aint);
-	ATRACE_32("ts_intr() interrupts", interrupts);
-	if (interrupts == 0) {
-		mutex_exit(&state->ts_lock);
-		/* no interrupts to process, so it's not us */
-		ATRACE_32("ts_intr() device didn't send interrupt", interrupts);
-		ASSERT(rc == DDI_INTR_UNCLAIMED);
-		goto done;
-	}
-
-	if (interrupts & TS_OUTPUT_CHANNEL) {	/* play interrupt */
-		ATRACE_32("ts_intr() play interrupt", interrupts);
-
-		/* is this in the 1st or 2nd half of the buffer? */
-		location = ddi_get32(handle, &state->ts_regs->aud_regs.ap_cspf);
-
-		/* clear the interrupt, should this be after getting loc? */
-		ddi_put32(handle, &state->ts_regs->aud_regs.ap_aint,
-		    TS_OUTPUT_CHANNEL);
-
-		/*
-		 * Reread the interrupt reg to ensure that
-		 * PIO write has completed.
-		 */
-		(void) ddi_get32(handle, &state->ts_regs->aud_regs.ap_aint);
-
-		/* set up for 1st or 2nd half */
-		if (location & TS_OUTPUT_CHANNEL) {	/* in the 2nd half */
-			bptr = &state->ts_pb[TS_1ST_HALF];
-			count = state->ts_pcnt[TS_1ST_HALF];
-			samples = state->ts_psamples[TS_1ST_HALF];
-		} else {				/* in 1st half */
-			/* pcnt[0] could be different from pcnt[TS_2ND_HALF] */
-			bptr = &state->ts_pb[state->ts_pcnt[TS_1ST_HALF]];
-			count = state->ts_pcnt[TS_2ND_HALF];
-			samples = state->ts_psamples[TS_2ND_HALF];
-		}
-
-		/* always zero for silence */
-		bzero(bptr, count);
-
-		/* get the next chunk of audio */
-		mutex_exit(&state->ts_lock);
-		if (am_get_audio(state->ts_ahandle, bptr, AUDIO_NO_CHANNEL,
-		    samples) == 0) {
-			mutex_enter(&state->ts_lock);
-			if (state->ts_flags & TS_DMA_ENGINE_EMPTY) {
-				/*
-				 * Clear the flag so if audio is restarted while
-				 * in am_play_shutdown() we can detect it and
-				 * not mess things up.
-				 */
-				state->ts_flags &= ~TS_DMA_ENGINE_INITIALIZED;
-
-				/* shutdown the mixer */
-				mutex_exit(&state->ts_lock);
-				am_play_shutdown(state->ts_ahandle, NULL);
-				mutex_enter(&state->ts_lock);
-
-				/*
-				 * Make sure playing wasn't restarted when
-				 * lock lost.
-				 */
-				if (state->ts_flags &
-				    TS_DMA_ENGINE_INITIALIZED) {
-					/* yes, it was, so we're done */
-					ATRACE("ts_intr() restart after"
-					    "shutdown", 0);
-					rc = DDI_INTR_CLAIMED;
-					goto rec_intr;
-				}
-
-				/* done playing, so stop playing */
-				ddi_put32(handle,
-				    &state->ts_regs->aud_regs.ap_stop,
-				    TS_OUTPUT_CHANNEL);
-
-				/* clr the flags getting ready for next start */
-				state->ts_flags &= ~(TS_DMA_ENGINE_PAUSED|
-				    TS_DMA_ENGINE_EMPTY);
-			} else {
-				/* next time we shut down, if no sound again */
-				state->ts_flags |= TS_DMA_ENGINE_EMPTY;
-			}
-		} else {
-			mutex_enter(&state->ts_lock);
-			/* we got at least one sample, so don't shutdown yet */
-			state->ts_flags &= ~TS_DMA_ENGINE_EMPTY;
-			(void) ddi_dma_sync(state->ts_ph, (off_t)bptr, count,
-			    DDI_DMA_SYNC_FORDEV);
-		}
-		rc = DDI_INTR_CLAIMED;
-	}
-
-rec_intr:
-
-	if (interrupts & TS_INPUT_CHANNEL) {	/* record interrupt */
-		ATRACE_32("ts_intr() record interrupt", interrupts);
-
-		/* is this in the 1st or 2nd half of the buffer? */
-		location = ddi_get32(handle, &state->ts_regs->aud_regs.ap_cspf);
-		ATRACE_32("ts_intr(c) location", location);
-
-		/* clear the interrupt, should this be after getting loc? */
-		ddi_put32(handle, &state->ts_regs->aud_regs.ap_aint,
-		    TS_INPUT_CHANNEL);
-
-		/*
-		 * Reread the interrupt reg to ensure that
-		 * PIO write has completed.
-		 */
-		(void) ddi_get32(handle, &state->ts_regs->aud_regs.ap_aint);
-
-		if (location & TS_INPUT_CHANNEL_MASK) {
-			/* 1st half just filled */
-			if (state->ts_flags & TS_DMA_RECORD_START) {
-			    /* skip the first interrupt, it is NULL */
-				state->ts_flags &= ~TS_DMA_RECORD_START;
-			} else {
-				if (ddi_dma_sync(state->ts_ch,
-				    (off_t)((state->ts_ccnt<<1) - TS_FIFO_SIZE),
-				    TS_FIFO_SIZE, DDI_DMA_SYNC_FORCPU) ==
-				    DDI_FAILURE) {
-					audio_sup_log(state->ts_ahandle,
-					    CE_NOTE,
-					    "!dma_sync(1) failed audio lost");
-					mutex_exit(&state->ts_lock);
-					goto done;
-				}
-				samples = TS_FIFO_SIZE /
-				    (state->ts_cprecision >>
-				    AUDIO_PRECISION_SHIFT);
-
-				sptr =  (uint16_t *)
-				    &state->ts_cb[(state->ts_ccnt<<1) -
-				    TS_FIFO_SIZE];
-				for (i = 0; i < (TS_FIFO_SIZE/2); i++) {
-					state->ts_tcbuf[i] = sptr[i];
-				}
-				ATRACE_32("ts_intr(c1) "
-				"calling am_send_audio() samples", samples);
-				tcbuf = state->ts_tcbuf;
-				mutex_exit(&state->ts_lock);
-				am_send_audio(state->ts_ahandle, tcbuf,
-				    AUDIO_NO_CHANNEL, samples);
-				mutex_enter(&state->ts_lock);
-
-				if (ddi_dma_sync(state->ts_ch, (off_t)0,
-				    state->ts_ccnt - TS_FIFO_SIZE,
-				    DDI_DMA_SYNC_FORCPU) == DDI_FAILURE) {
-					audio_sup_log(state->ts_ahandle,
-					    CE_NOTE,
-					    "!dma_sync(2) failed audio lost");
-					mutex_exit(&state->ts_lock);
-					goto done;
-				}
-				samples = (state->ts_ccnt - TS_FIFO_SIZE) /
-				    (state->ts_cprecision >>
-				    AUDIO_PRECISION_SHIFT);
-
-				sptr =  (uint16_t *)state->ts_cb;
-				count = (state->ts_ccnt - TS_FIFO_SIZE) >> 1;
-				for (i = 0; i < count; i++) {
-				state->ts_tcbuf[i] = sptr[i];
-				}
-				ATRACE_32("ts_intr(c2) "
-				"calling am_send_audio() samples", samples);
-				tcbuf = state->ts_tcbuf;
-				mutex_exit(&state->ts_lock);
-				am_send_audio(state->ts_ahandle, tcbuf,
-				    AUDIO_NO_CHANNEL, samples);
-				mutex_enter(&state->ts_lock);
-			}
-		} else {
-			/* 2nd half just filled */
-			if (ddi_dma_sync(state->ts_ch,
-			    (off_t)(state->ts_ccnt - TS_FIFO_SIZE),
-			    state->ts_ccnt, DDI_DMA_SYNC_FORCPU) ==
-			    DDI_FAILURE) {
-				audio_sup_log(state->ts_ahandle, CE_NOTE,
-				    "!dma_sync(2) failed audio lost");
-				mutex_exit(&state->ts_lock);
-				goto done;
-			}
-			samples = state->ts_ccnt / (state->ts_cprecision >>
-			    AUDIO_PRECISION_SHIFT);
-
-			sptr = (uint16_t *)&state->ts_cb[state->ts_ccnt -
-			    TS_FIFO_SIZE];
-			count = state->ts_ccnt >> 1;
-			for (i = 0; i < count; i++) {
-				state->ts_tcbuf[i] = sptr[i];
-			}
-			ATRACE_32("ts_intr(c3) "
-			    "calling am_send_audio() samples", samples);
-			tcbuf = state->ts_tcbuf;
-			mutex_exit(&state->ts_lock);
-			am_send_audio(state->ts_ahandle, tcbuf,
-			    AUDIO_NO_CHANNEL, samples);
-			mutex_enter(&state->ts_lock);
-		}
-
-		rc = DDI_INTR_CLAIMED;
-	}
-
-	interrupts &= ~(TS_INPUT_CHANNEL|TS_OUTPUT_CHANNEL);
-	if (interrupts) {
-		/*
-		 * handle, but don't service non play or record audiots
-		 * interrupts and shutdown the other DMA engines if they
-		 * somehow activated themselves - which is why we are here in
-		 * the first place.
-		 */
-		ATRACE_32("ts_intr() unused audiots interrupt", interrupts);
-		ddi_put32(handle, &state->ts_regs->aud_regs.ap_aint,
-		    interrupts);
-		/*
-		 * Reread the interrupt reg to ensure that
-		 * PIO write has completed.
-		 */
-		(void) ddi_get32(handle, &state->ts_regs->aud_regs.ap_aint);
-
-		AND_SET_WORD(handle, &state->ts_regs->aud_regs.ap_ainten,
-		    TS_INPUT_CHANNEL|TS_OUTPUT_CHANNEL);
-
-		ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_stop,
-		    ~(TS_INPUT_CHANNEL|TS_OUTPUT_CHANNEL));
-
-		rc = DDI_INTR_CLAIMED;
-	}
-
-	/* update the kernel interrupt statistics */
-	if (state->ts_ksp) {
-		if (rc == DDI_INTR_CLAIMED) {
-			TS_KIOP(state)->intrs[KSTAT_INTR_HARD]++;
-		}
-	}
-
-	mutex_exit(&state->ts_lock);
-
-done:
-	ATRACE("audiots_intr() done", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_intr() */
-
-/*
- * audiots_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. It then binds each of the buffers to its
- *	respective handle, getting a DMA cookie. Finally, the registers
- *	are mapped in.
- *
- *	Once the config space registers are mapped in we determine if the
- *	audio core may be power managed. It should, but if it doesn't,
- *	then trying to may cause the core to hang.
- *
- *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
- *		memory. This means these calls will almost always succeed.
- *
- *	NOTE: ddi_dma_alloc_handle() attempts to use the full 4 GB DMA address
- *		range. This is to work around Southbridge rev E/G OBP issues.
- *		(See Grover OBP note above)
- *
- *	CAUTION: Make sure all errors call audio_sup_log().
- *
- * Arguments:
- *	dev_info_t	*dip            Pointer to the device's devinfo
- *	audiots_state_t	*state          The device's state structure
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-audiots_map_regs(dev_info_t *dip, audiots_state_t *state)
-{
-	uint_t			dma_cookie_count;
-	uint32_t		rev_id;
-	int			rc;
-
-	ATRACE("in ts_map_regs()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* allocate one handle for play and one for record */
-	if ((rc = ddi_dma_alloc_handle(dip, &audiots_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->ts_ph)) != DDI_SUCCESS) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_alloc_handle(P) failed: %d", rc);
-		goto error;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &audiots_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->ts_ch)) != DDI_SUCCESS) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_alloc_handle(C) failed: %d", rc);
-		goto error_ph;
-	}
-
-	/* allocate the two DMA buffers, one for play and one for record */
-	ASSERT(state->ts_pbuf_size > 0);
-	ASSERT(state->ts_cbuf_size > 0);
-	if (ddi_dma_mem_alloc(state->ts_ph, state->ts_pbuf_size, &ts_acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &state->ts_pb,
-	    &state->ts_pml, &state->ts_pmh) == DDI_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_mem_alloc(P) failed");
-		goto error_ch;
-	}
-	if (ddi_dma_mem_alloc(state->ts_ch, state->ts_cbuf_size, &ts_acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &state->ts_cb,
-	    &state->ts_cml, &state->ts_cmh) == DDI_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_mem_alloc(C) failed");
-		goto error_pmh;
-	}
-
-	/* bind each of the buffers to a DMA handle */
-	if ((rc = ddi_dma_addr_bind_handle(state->ts_ph, (struct as *)0,
-	    state->ts_pb, state->ts_pbuf_size, DDI_DMA_WRITE|DDI_DMA_CONSISTENT,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->ts_pc, &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_addr_bind_handle(P) failed: %d", rc);
-		goto error_cmh;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->ts_ch, (struct as *)0,
-	    state->ts_cb, state->ts_cbuf_size, DDI_DMA_READ|DDI_DMA_CONSISTENT,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->ts_cc, &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_dma_addr_bind_handle(C) failed: %d", rc);
-		goto error_pc;
-	}
-	ASSERT(dma_cookie_count == 1);
-
-	/* map in the registers, the config and memory mapped registers */
-	if (ddi_regs_map_setup(dip, TS_CONFIG_REGS,
-	    (caddr_t *)&state->ts_config, 0, 0, &ts_acc_attr,
-	    &state->ts_chandle) != DDI_SUCCESS) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_regs_map_setup() failed: %d", rc);
-		goto error_cc;
-	}
-
-	/* make sure we can power manage the audio core */
-	if (ddi_get16(state->ts_chandle, &state->ts_config->tsc_cap_ptr) ==
-	    TS_CAP_PTR) {
-		state->ts_pm_core = TS_PWR_MANAGE;
-		ATRACE("Audio core is power manageable", 0);
-	} else {
-		state->ts_pm_core = TS_NO_PWR_MANAGE;
-		ATRACE("Audio core is not power manageable", 0);
-	}
-
-	if (ddi_regs_map_setup(dip, TS_MEM_MAPPED_REGS,
-	    (caddr_t *)&state->ts_regs, 0, 0, &ts_acc_attr,
-	    &state->ts_handle) != DDI_SUCCESS) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!map_regs() ddi_regs_map_setup() failed: %d", rc);
-		goto error_unmap;
-	}
-
-	/* Read the Audio Controller's revision ID */
-	rev_id = ddi_get32(state->ts_chandle,
-	    &state->ts_config->tsc_class_code_rev_id);
-	if (rev_id & AC_REV_ID1) {
-		state->ts_rev_id = AC_REV_ID1;
-		ATRACE("Old SB audio rev ID", rev_id);
-	} else if (rev_id & AC_REV_ID2) {
-		state->ts_rev_id = AC_REV_ID2;
-		ATRACE("New SB audio rev ID", rev_id);
-	} else {
-		/* Unknown rev, who knows what else has changed... */
-		ATRACE("Unknown SB rev - don't use", rev_id);
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "map_regs() unsupported SouthBridge Chip revision: %x",
-		    rev_id);
-		return (AUDIO_FAILURE);
-	}
-
-	/* let the state structure know about the attributes */
-	state->ts_dma_attr = &audiots_attr;
-
-	ATRACE("ts_map_regs() returning success", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (AUDIO_SUCCESS);
-
-error_unmap:
-	ddi_regs_map_free(&state->ts_chandle);
-error_cc:
-	(void) ddi_dma_unbind_handle(state->ts_ch);
-error_pc:
-	(void) ddi_dma_unbind_handle(state->ts_ph);
-error_cmh:
-	ddi_dma_mem_free(&state->ts_cmh);
-error_pmh:
-	ddi_dma_mem_free(&state->ts_pmh);
-error_ch:
-	ddi_dma_free_handle(&state->ts_ch);
-error_ph:
-	ddi_dma_free_handle(&state->ts_ph);
-error:
-	ATRACE("ts_map_regs() returning failure", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (AUDIO_FAILURE);
-
-}	/* audiots_map_regs() */
-
-/*
- * audiots_or_ac97()
- *
- * Description:
- *	Logically OR a value with the specified AC-97 Codec register.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		reg		AC-97 register number
- *	uint16_t	data		The value to OR
- *
- * Returns:
- *	void
- */
-static void
-audiots_or_ac97(audiots_state_t *state, int reg, uint16_t data)
-{
-
-	audiots_set_ac97(state, reg,
-	    (data | state->ts_shadow[TS_CODEC_REG(reg)]));
-
-}	/* audiots_or_ac97() */
-
-/*
- * audiots_power_down()
- *
- * Description:
- *	Power down the AC-97 Codec and then, if allowed, power down
- *	the audio core. The audio core implements PCI power management
- *	version 1.0. We don't support state D3 (cold) since we don't
- *	have an external device to drive reseting the core to get it
- *	back up and running. We just toggle between D0 and D3 since
- *	we need the device either all on or all off.
- *
- *	Minimum transition times between the power states:
- *		D0 <---> D1	no time delay
- *		D0 <---> D2	200 microseconds
- *		D0 <---> D3	10 milliseconds
- *
- *	To be safe I'm doubling that time because if you don't wait
- *	long enough you get a hard hang for the app opening audio.
- *
- *	AC-97 Codec's walk from power states PR0 through PR5 plus the
- *	optional PR6 and EAPD. We always set PR6 EAPD since they don't
- *	hurt anything. Then we walk to PR2|PR3. We don't go any further
- *	because this requires that we do a cold reset. Since this pin
- *	isn't connected in the 1553 we can't do this.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_power_down(audiots_state_t *state)
-{
-	ATRACE("in audiots_power_down() done", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	/* PM should be enabled */
-	ASSERT(state->ts_flags & TS_PM_SUPPORTED);
-
-	/* No interrupts should be pending before we power down */
-	ASSERT(ddi_get32(state->ts_handle,
-	    &state->ts_regs->aud_regs.ap_aint) == 0);
-	ASSERT(ddi_get32(state->ts_handle,
-	    &state->ts_regs->aud_regs.ap_eint) == 0);
-
-	/* powering down, save the audio core's state */
-	audiots_save_controller(state);
-
-	/* power down the Codec, also shut down external amps */
-	audiots_set_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER,
-	    (PCSR_PR6|PCSR_EAPD));
-
-	/* shut down ADC and inputs */
-	audiots_set_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER, PCSR_PR0);
-
-	/* shut down DAC */
-	audiots_set_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER, PCSR_PR1);
-
-	/* shut down analog mixer and Vref */
-	audiots_set_ac97(state, AC97_POWERDOWN_CTRL_STAT_REGISTER,
-	    (PCSR_PR2|PCSR_PR3));
-
-	/* MUST clear the shadow register so we can power up! */
-	state->ts_shadow[TS_CODEC_REG(AC97_POWERDOWN_CTRL_STAT_REGISTER)] = 0;
-
-	/* power down the core, if allowed */
-	if (state->ts_pm_core == TS_PWR_MANAGE) {
-		/* okay, drop to D3 for the best power reduction */
-		OR_SET_SHORT(state->ts_chandle,
-		    &state->ts_config->tsc_pmcsr, TS_PWR_D3);
-
-		/*
-		 * wait 20 milliseconds for state change,
-		 * __lock_lint tells warlock not to flag this delay()
-		 */
-#ifndef __lock_lint
-		delay(drv_usectohz(TS_20MS));
-#endif
-	}
-
-	state->ts_powered = TS_PWR_OFF;
-
-	ATRACE("audiots_power_down() done", NULL);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_power_down() */
-
-/*
- * audiots_power_up()
- *
- * Description:
- *	Power up the audio core if allowed, and then the AC-97 Codec.
- *	The state change timings documented in audiots_power_down()
- *	apply to powering up as well.
- *
- *	audiots_chip_init() does a soft reset, which powers up all of
- *	the sections of the Codec that were powered down in
- *	audiots_power_down().
- *
- *	NOTE: We don't worry about locking since the only routines that
- *		may call us are audiots_attach() and audiots_power().
- *		Both of which should be the only threads in the driver.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_power_up(audiots_state_t *state)
-{
-	ATRACE("in audiots_power_up()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	/* PM should be enabled */
-	ASSERT(state->ts_flags & TS_PM_SUPPORTED);
-
-	/* power up the core, if allowed */
-	if (state->ts_pm_core == TS_PWR_MANAGE) {
-		/* go from D3 to D0 */
-		AND_SET_SHORT(state->ts_chandle,
-		    &state->ts_config->tsc_pmcsr, ~TS_PWR_D3);
-
-		/*
-		 * wait 20 milliseconds for state change,
-		 * __lock_lint tells warlock not to flag this delay()
-		 */
-#ifndef __lock_lint
-		delay(drv_usectohz(TS_20MS));
-#endif
-		/* clear the PME# flag */
-		OR_SET_SHORT(state->ts_chandle,
-		    &state->ts_config->tsc_pmcsr, TS_PWR_PME);
-	}
-
-	/* restore the state, does reset to power up the Codec */
-	audiots_chip_init(state, TS_INIT_RESTORE);
-
-	state->ts_powered = TS_PWR_ON;
-
-	ATRACE("audiots_power_up() done", NULL);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_power_up() */
-
-/*
- * audiots_read_ac97()
- *
- * Description:
- *	This routine actually reads the AC-97 Codec's register. It may
- *	be called several times to succeed.
- *
- * NOTE:
- * 	Revision M1535D B1-C of the ALI SouthBridge includes a workaround for
- *	the broken busy flag. Resetting the busy flag requires a software tweak
- *	to go with the worked around hardware. When we detect failure, we make
- *	10 attempts to reset the chip before we fail. This should reset the new
- *	SB systems. On all SB systems, this will increse the read delay
- *	slightly, but shouldn't bother it otherwise.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		reg		AC-97 register number
- *
- * Returns:
- *	unsigned short		The value in the specified register
- */
-static uint16_t
-audiots_read_ac97(audiots_state_t *state, int reg)
-{
-	ddi_acc_handle_t	handle = state->ts_handle;
-	uint16_t		*addr;
-	uint16_t		*data;
-	uint32_t		*stimer = &state->ts_regs->aud_regs.ap_stimer;
-	uint32_t		chk1;
-	uint32_t		chk2;
-	int			resets = 0;
-	int			i;
-
-	if (state->ts_rev_id == AC_REV_ID1) {
-		addr = &state->ts_regs->aud_regs.ap_acrd_35D_reg;
-		data = &state->ts_regs->aud_regs.ap_acrd_35D_data;
-		ATRACE("Using OLD SB 35D addresses for reading", 0x44);
-	} else {
-		addr = &state->ts_regs->aud_regs.ap_acrdwr_reg;
-		data = &state->ts_regs->aud_regs.ap_acrdwr_data;
-		ATRACE("Using NEW SB 35D+ addresses for reading", 0x40);
-	}
-
-first_read:
-	/* wait for ready to send read request */
-	for (i = 0; i < TS_READ_TRIES; i++) {
-		if (!(ddi_get16(handle, addr) & AP_ACRD_R_READ_BUSY)) {
-			break;
-		}
-		/* don't beat on the bus */
-		drv_usecwait(1);
-	}
-	if (i >= TS_READ_TRIES) {
-		if (resets < TS_RESET_TRIES) {
-			/* Attempt to reset */
-			ATRACE("Attempting to reset the AC97 #1", resets);
-			drv_usecwait(TS_20US);
-			ddi_put16(handle, addr, TS_SB_RESET);
-			resets++;
-			goto first_read;
-		} else {
-			ATRACE("Reading the AC97 register has failed #1", 0);
-			state->ts_flags |= TS_AUDIO_READ_FAILED;
-			if (!(state->ts_flags & TS_READ_FAILURE_PRINTED)) {
-				ddi_dev_report_fault(state->ts_dip,
-				    DDI_SERVICE_LOST, DDI_DEVICE_FAULT,
-				    "Unable to communicate with AC97 CODEC");
-				audio_sup_log(state->ts_ahandle, CE_WARN,
-				    "The audio AC97 register has timed out.");
-				audio_sup_log(state->ts_ahandle, CE_CONT,
-				    "Audio is now disabled.\n");
-				audio_sup_log(state->ts_ahandle, CE_CONT,
-				    "Please reboot to restore audio.\n");
-
-				/* Don't flood the console */
-				state->ts_flags |= TS_READ_FAILURE_PRINTED;
-			}
-		}
-		return (0);
-	}
-
-	/* program the register to read */
-	ddi_put16(handle, addr, (reg|AP_ACRD_W_PRIMARY_CODEC|
-	    AP_ACRD_W_READ_MIXER_REG|AP_ACRD_W_AUDIO_READ_REQ&
-	    (~AP_ACWR_W_SELECT_WRITE)));
-
-	/* hardware bug work around */
-	chk1 = ddi_get32(handle, stimer);
-	chk2 = ddi_get32(handle, stimer);
-	i = TS_WAIT_CNT;
-	while (chk1 == chk2 && i) {
-		chk2 = ddi_get32(handle, stimer);
-		i--;
-	}
-	OR_SET_SHORT(handle, addr, AP_ACRD_W_READ_MIXER_REG);
-	resets = 0;
-
-second_read:
-	/* wait again for read to send read request */
-	for (i = 0; i < TS_READ_TRIES; i++) {
-		if (!(ddi_get16(handle, addr) & AP_ACRD_R_READ_BUSY)) {
-			break;
-		}
-		/* don't beat on the bus */
-		drv_usecwait(1);
-	}
-	if (i >= TS_READ_TRIES) {
-		if (resets < TS_RESET_TRIES) {
-			/* Attempt to reset */
-			ATRACE("Attempting to reset the AC97 #2", resets);
-			drv_usecwait(TS_20US);
-			ddi_put16(handle, addr, TS_SB_RESET);
-			resets++;
-			goto second_read;
-		} else {
-			ATRACE("Reading the AC97 register has failed #2", 0);
-			state->ts_flags |= TS_AUDIO_READ_FAILED;
-			if (!(state->ts_flags & TS_READ_FAILURE_PRINTED)) {
-				ddi_dev_report_fault(state->ts_dip,
-				    DDI_SERVICE_LOST, DDI_DEVICE_FAULT,
-				    "Unable to communicate with AC97 CODEC");
-				audio_sup_log(state->ts_ahandle, CE_WARN,
-				    "The audio AC97 register has timed out.");
-				audio_sup_log(state->ts_ahandle, CE_CONT,
-				    "Audio is now disabled.\n");
-				audio_sup_log(state->ts_ahandle, CE_CONT,
-				    "Please reboot to restore audio.\n");
-
-				/* Don't flood the console */
-				state->ts_flags |= TS_READ_FAILURE_PRINTED;
-			}
-		}
-		return (0);
-	}
-
-	return (ddi_get16(handle, data));
-
-}	/* audiots_read_ac97() */
-
-/*
- * audiots_save_controller()
- *
- * Description:
- *	Save the state of the audio controller.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_save_controller(audiots_state_t *state)
-{
-	ddi_acc_handle_t	handle;
-	int			str;
-
-	ATRACE("in ts_save_controller()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	/* save the controller state, Codec state is in ts_shadow[] */
-	handle = state->ts_handle;
-	for (str = 0; str < TS_MAX_HW_CHANNELS; str++) {
-		state->ts_save_regs[str].aram_delta = ddi_get16(handle,
-		    &state->ts_regs->aud_ram[str].aram.aram_delta);
-		state->ts_save_regs[str].eram_ctrl_ec = ddi_get16(handle,
-		    &state->ts_regs->aud_ram[str].eram.eram_ctrl_ec);
-	}
-
-	ATRACE("ts_save_controller() done", NULL);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_save_controller() */
-
-/*
- * audiots_set_ac97()
- *
- * Description:
- *	Set the value in the specified AC-97 Codec register. Just like
- *	reading the AC-97 Codec, it is possible there is a problem writing
- *	it as well. So we loop.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		reg		AC-97 register number
- *	uint16_t	value		The value to write
- *
- * Returns:
- *	void
- */
-static void
-audiots_set_ac97(audiots_state_t *state, int reg, const uint16_t data)
-{
-	ddi_acc_handle_t handle = state->ts_handle;
-	uint16_t	*data_addr = &state->ts_regs->aud_regs.ap_acrdwr_data;
-	uint16_t	*reg_addr = &state->ts_regs->aud_regs.ap_acrdwr_reg;
-	int		count;
-	int		i;
-	uint16_t	tmp_short;
-
-	reg &= AP_ACWR_INDEX_MASK;
-
-	/* Don't touch the reserved bits on the pre 35D+ SouthBridge */
-	if (state->ts_rev_id == AC_REV_ID1) {
-		reg |= AP_ACWR_W_PRIMARY_CODEC|AP_ACWR_W_WRITE_MIXER_REG;
-		ATRACE("Writing to 35D SB", reg);
-	} else {
-		reg |= AP_ACWR_W_PRIMARY_CODEC|AP_ACWR_W_WRITE_MIXER_REG|
-		    AP_ACWR_W_SELECT_WRITE;
-		ATRACE("Writing to 35D+ SB", reg);
-	}
-
-	for (count = TS_LOOP_CNT; count--; ) {
-		/* wait for ready to write */
-		for (i = 0; i < TS_WAIT_CNT; i++) {
-			if (!(ddi_get16(handle, reg_addr) &
-			    AP_ACWR_R_WRITE_BUSY)) {
-				/* ready to write */
-				ddi_put16(handle, reg_addr, reg);
-
-				/* Write the data */
-				ddi_put16(handle, data_addr, data);
-				break;
-			}
-		}
-		if (i >= TS_WAIT_CNT) {
-			/* try again */
-			continue;
-		}
-
-		/* wait for write to complete */
-		for (i = 0; i < TS_WAIT_CNT; i++) {
-			if (!(ddi_get16(handle, reg_addr) &
-			    AP_ACWR_R_WRITE_BUSY)) {
-				/* done writing */
-				break;
-			}
-		}
-
-		/* verify the value written and also update ts_shadow[] */
-		tmp_short = audiots_get_ac97(state, reg);
-		if (data == tmp_short) {
-			/* successfully loaded, so we can return */
-			return;
-		}
-	}
-
-}	/* audiots_set_ac97() */
-
-/*
- * audiots_set_busy()
- *
- * Description:
- *	This routine is called whenever a routine needs to guarantee
- *	that it will not be suspended or the power removed by the power
- *	manager. It will also block any routine while a suspend is
- *	going on.
- *
- *	CAUTION: This routine cannot be called by routines that will
- *		block. Otherwise DDI_SUSPEND will be blocked for a
- *		long time. And that is the wrong thing to do.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS			Set busy and powered up
- *	AUDIO_FAILURE			Couldn't power up, so not busy
- */
-static int
-audiots_set_busy(audiots_state_t *state)
-{
-	ATRACE("in audiots_set_busy()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&state->ts_lock);
-
-	/* Don't proceed if PM is not enabled */
-	if (!(state->ts_flags & TS_PM_SUPPORTED)) {
-		mutex_exit(&state->ts_lock);
-		return (AUDIO_SUCCESS);
-	}
-
-	/* block if we are going to be suspended */
-	while (state->ts_suspended == TS_SUSPENDED) {
-		cv_wait(&state->ts_cv, &state->ts_lock);
-	}
-
-	/*
-	 * Okay, we aren't going to be suspended yet, so mark as busy.
-	 * This will keep us from being suspended when we release the lock.
-	 */
-	ASSERT(state->ts_busy_cnt >= 0);
-	state->ts_busy_cnt++;
-
-	/* now can release the lock before we raise the power */
-	mutex_exit(&state->ts_lock);
-
-	/*
-	 * Mark as busy before we ask for power to be raised. This removes
-	 * the race condtion between the call to audiots_power() and our call
-	 * to raise power. After we raise power we immediately mark as idle
-	 * so the count is still good.
-	 */
-	(void) pm_busy_component(state->ts_dip, TS_COMPONENT);
-	if (pm_raise_power(state->ts_dip, TS_COMPONENT, TS_PWR_ON) ==
-	    DDI_FAILURE) {
-		/* match the busy call above */
-		(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-
-		ATRACE("ts_set_busy() pm_raise_power() failed", NULL);
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!set_busy() power up failed");
-
-		mutex_enter(&state->ts_lock);
-		state->ts_busy_cnt--;		/* restore busy count */
-		if (state->ts_busy_cnt == 0) {
-			/* let DDI_SUSPEND continue */
-			cv_broadcast(&state->ts_cv);
-		}
-		mutex_exit(&state->ts_lock);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* power is up and we are marked as busy, so we are done */
-
-	ATRACE("audiots_set_busy() done", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (AUDIO_SUCCESS);
-
-}	/* audiots_set_busy() */
-
-/*
- * audiots_set_gain()
- *
- * Description:
- *	Set the play/record gain.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		The gain to set
- *	int		arg2		The channel, 0 == left or 1 == right
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-/*ARGSUSED*/
-static int
-audiots_set_gain(audiots_state_t *state, int stream, int dir, int gain,
-    int channel)
-{
-	uint16_t	tmp_short;
-
-	ATRACE("in audiots_set_gain()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (dir == AUDIO_PLAY) {	/* play gain */
-		/*
-		 * For play we use PCM so all volumes change with just
-		 * one write. This way we get line out, headphone and
-		 * internal speaker in one shot.
-		 *
-		 * The AC-97 Codec goes from -34.5 dB (11111) to 0 dB
-		 * (01000) to +12.0 dB (00000). We turn gain into atten.
-		 */
-		mutex_enter(&state->ts_lock);
-		tmp_short = state->ts_shadow[
-		    TS_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)];
-		if (channel == 0) {	/* left channel */
-			tmp_short &= ~PCMOVR_LEFT_GAIN_MASK;
-			tmp_short |= (((AUDIO_MAX_GAIN - gain) >>
-			    TS_GAIN_SHIFT3) << TS_BYTE_SHIFT) &
-			    PCMOVR_LEFT_GAIN_MASK;
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			tmp_short &= ~PCMOVR_RIGHT_GAIN_MASK;
-			tmp_short |= ((AUDIO_MAX_GAIN - gain) >>
-			    TS_GAIN_SHIFT3) & PCMOVR_RIGHT_GAIN_MASK;
-		}
-		audiots_set_ac97(state, AC97_PCM_OUT_VOLUME_REGISTER,
-		    tmp_short);
-		mutex_exit(&state->ts_lock);
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		/*
-		 * For record we use the master record gain with all
-		 * of the inputs set to 0dB.
-		 *
-		 * The AC-97 Codec goes from 0 dB (0000) to +22.5 dB
-		 * (1111), so gain remains gain. We chop off the bottom
-		 * 4 bits and use the top for the gain.
-		 */
-		mutex_enter(&state->ts_lock);
-		tmp_short =
-		    state->ts_shadow[TS_CODEC_REG(AC97_RECORD_GAIN_REGISTER)];
-		if (channel == 0) {	/* left channel */
-			tmp_short &= ~RGR_LEFT_MASK;
-			tmp_short |= (gain << TS_GAIN_SHIFT4) & RGR_LEFT_MASK;
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			tmp_short &= ~RGR_RIGHT_MASK;
-			tmp_short |= (gain >> TS_GAIN_SHIFT4) & RGR_RIGHT_MASK;
-		}
-		audiots_set_ac97(state, AC97_RECORD_GAIN_REGISTER,
-		    tmp_short);
-		mutex_exit(&state->ts_lock);
-	}
-
-	ATRACE("audiots_set_gain() done", NULL);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (AUDIO_SUCCESS);
-
-}	/* audiots_set_gain() */
-
-/*
- * audiots_set_idle()
- *
- * Description:
- *	This routine reduces the busy count. It then does a cv_broadcast()
- *	if the count is 0 so a waiting DDI_SUSPEND will continue forward.
- *	It ends by resetting the power management timer.
- *
- *	We don't do anything with power because the routine that is no longer
- *	busy either doesn't need the hardware, or we are playing or recording
- *	so the power won't come down anyway.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_set_idle(audiots_state_t *state)
-{
-	ATRACE("in audiots_set_idle()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&state->ts_lock);
-
-	/* Don't proceed if PM is not enabled */
-	if (!(state->ts_flags & TS_PM_SUPPORTED)) {
-		mutex_exit(&state->ts_lock);
-		return;
-	}
-
-	ASSERT(state->ts_suspended == TS_NOT_SUSPENDED);
-
-	/* decrement the busy count */
-	state->ts_busy_cnt--;
-
-	/* if no longer busy, then we wake up a waiting SUSPEND */
-	if (state->ts_busy_cnt == 0) {
-		cv_broadcast(&state->ts_cv);
-	}
-
-	/* we're done, so unlock */
-	mutex_exit(&state->ts_lock);
-
-	/* reset the timer */
-	(void) pm_idle_component(state->ts_dip, TS_COMPONENT);
-
-	ATRACE("audiots_set_idle() done", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-}	/* audiots_set_idle() */
-
-/*
- * audiots_set_monitor_gain()
- *
- * Description:
- *	Set the monitor gain.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *	int		gain		The gain to set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-/*ARGSUSED*/
-static int
-audiots_set_monitor_gain(audiots_state_t *state, int gain)
-{
-	uint16_t	tmp_short;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("in audiots_set_monitor_gain()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (gain == 0) {
-		/* disable loopbacks when gain == 0 */
-		tmp_short = MVR_MUTE;
-	} else {
-		/* Adjust the value of gain to meet the requirement of AC'97 */
-		tmp_short = (((AUDIO_MAX_GAIN - gain) >> TS_GAIN_SHIFT3) <<
-		    TS_BYTE_SHIFT) & PCMOVR_LEFT_GAIN_MASK;
-		tmp_short |= ((AUDIO_MAX_GAIN - gain) >> TS_GAIN_SHIFT3) &
-		    PCMOVR_RIGHT_GAIN_MASK;
-	}
-
-	mutex_enter(&state->ts_lock);
-
-	switch (state->ts_input_port) {
-	case AUDIO_NONE:
-		/*
-		 * It is possible to set the value of gain before any input
-		 * is selected. So, we just save the gain and then return
-		 * SUCCESS.
-		 */
-		break;
-	case AUDIO_MICROPHONE:
-		/*
-		 * MIC input has 20dB boost, we just preserve it
-		 */
-		tmp_short |= state->ts_shadow[TS_CODEC_REG(
-		    AC97_MIC_VOLUME_REGISTER)] & MICVR_20dB_BOOST;
-		audiots_set_ac97(state,
-		    AC97_MIC_VOLUME_REGISTER, tmp_short);
-		break;
-	case AUDIO_LINE_IN:
-		audiots_set_ac97(state,
-		    AC97_LINE_IN_VOLUME_REGISTER, tmp_short);
-		break;
-	case AUDIO_CD:
-		audiots_set_ac97(state,
-		    AC97_CD_VOLUME_REGISTER, tmp_short);
-		break;
-	case AUDIO_CODEC_LOOPB_IN:
-		/* we already are getting the loopback, so done */
-		mutex_exit(&state->ts_lock);
-		rc = AUDIO_SUCCESS;
-		goto done;
-	default:
-		/* this should never happen! */
-		mutex_exit(&state->ts_lock);
-		ATRACE("ts_ad_set_config() monitor gain bad device",
-		    NULL);
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-
-	if (gain == 0) {
-		state->ts_monitor_gain = 0;
-	} else {
-		state->ts_monitor_gain = tmp_short;
-	}
-	mutex_exit(&state->ts_lock);
-
-done:
-	ATRACE_32("audiots_set_monitor_gain()", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_set_monitor_gain() */
-
-/*
- * audiots_set_port()
- *
- * Description:
- *	Set the play/record port.
- *
- *	We also use this routine to unmute the play ports. By passing
- *	TS_PORT_UNMUTE as the port we know to use the stored play port.
- *	This has to be done inside the lock, otherwise there is the
- *	possibility that the port passed in when this routine is called
- *	could be wrong by the time we execute the unmute.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *					which is not how we program the device
- *					for now.
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		port		The port to set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The port could not been set
- */
-static int
-audiots_set_port(audiots_state_t *state, int dir, int port)
-{
-	uint32_t	tmp_word;
-	uint16_t	tmp_short;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("in audiots_set_port()", state);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	if (dir == AUDIO_PLAY) {	/* output port(s) */
-		/* figure out which output port(s) to turn on */
-		tmp_word = 0;
-
-		mutex_enter(&state->ts_lock);
-
-		/*
-		 * CAUTION: We must get the saved output port after the
-		 *	lock so we know it is good.
-		 */
-		if (port == TS_PORT_UNMUTE) {
-			port = state->ts_output_port;
-		}
-
-		if (port & AUDIO_SPEAKER) {
-			audiots_and_ac97(state,
-			    AC97_MONO_MASTER_VOLUME_REGSITER,
-			    (uint16_t)~MMVR_MUTE);
-			tmp_word |= AUDIO_SPEAKER;
-		} else {
-			audiots_or_ac97(state,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MMVR_MUTE);
-		}
-
-		if (port & AUDIO_HEADPHONE) {
-			audiots_and_ac97(state,
-			    AC97_HEADPHONE_VOLUME_REGISTER,
-			    (uint16_t)~HPVR_MUTE);
-			tmp_word |= AUDIO_HEADPHONE;
-		} else {
-			audiots_or_ac97(state,
-			    AC97_HEADPHONE_VOLUME_REGISTER, HPVR_MUTE);
-		}
-
-		if (port & AUDIO_LINE_OUT) {
-			audiots_and_ac97(state, AC97_MASTER_VOLUME_REGISTER,
-			    (uint16_t)~MVR_MUTE);
-			tmp_word |= AUDIO_LINE_OUT;
-		} else {
-			audiots_or_ac97(state, AC97_MASTER_VOLUME_REGISTER,
-			    MVR_MUTE);
-		}
-
-		state->ts_output_port = tmp_word;
-
-		mutex_exit(&state->ts_lock);
-
-		ATRACE_32("ts_ad_set_config() set out port", tmp_word);
-
-		if (tmp_word != (port & TS_PORT_MASK)) {
-			ATRACE_32("ts_ad_set_config() bad out port", port);
-			rc = AUDIO_FAILURE;
-			goto done;
-		}
-
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-
-		/* figure out which input port to set */
-		mutex_enter(&state->ts_lock);
-		switch (port) {
-		case AUDIO_NONE:
-			/* set to an unused input */
-			tmp_short = RSCR_R_PHONE|RSCR_L_PHONE;
-
-			/* mute the master record input */
-			audiots_or_ac97(state, AC97_RECORD_GAIN_REGISTER,
-			    RGR_MUTE);
-
-			/* see if we need to update monitor loopback */
-			if (state->ts_monitor_gain) {
-				if (state->ts_input_port == AUDIO_MICROPHONE) {
-					audiots_or_ac97(state,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (state->ts_input_port ==
-				    AUDIO_LINE_IN) {
-					audiots_or_ac97(state,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (state->ts_input_port ==
-				    AUDIO_CD) {
-					audiots_or_ac97(state,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-		case AUDIO_MICROPHONE:
-			/* set to the mic input */
-			tmp_short = RSCR_R_MIC|RSCR_L_MIC;
-
-			/* see if we need to update monitor loopback */
-			if (state->ts_monitor_gain) {
-				if (state->ts_input_port == AUDIO_LINE_IN) {
-					audiots_or_ac97(state,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (state->ts_input_port == AUDIO_CD) {
-					audiots_or_ac97(state,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				audiots_set_ac97(state,
-				    AC97_MIC_VOLUME_REGISTER,
-				    state->ts_monitor_gain);
-			}
-			break;
-		case AUDIO_LINE_IN:
-			/* set to the line in input */
-			tmp_short = RSCR_R_LINE_IN|RSCR_L_LINE_IN;
-
-			/* see if we need to update monitor loopback */
-			if (state->ts_monitor_gain) {
-				if (state->ts_input_port == AUDIO_MICROPHONE) {
-					audiots_or_ac97(state,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (state->ts_input_port == AUDIO_CD) {
-					audiots_or_ac97(state,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				audiots_set_ac97(state,
-				    AC97_LINE_IN_VOLUME_REGISTER,
-				    state->ts_monitor_gain);
-			}
-			break;
-		case AUDIO_CD:
-			/* set to the line in input */
-			tmp_short = RSCR_R_CD|RSCR_L_CD;
-
-			/* see if we need to update monitor loopback */
-			if (state->ts_monitor_gain) {
-				if (state->ts_input_port == AUDIO_MICROPHONE) {
-					audiots_or_ac97(state,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (state->ts_input_port ==
-				    AUDIO_LINE_IN) {
-					audiots_or_ac97(state,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				}
-				audiots_set_ac97(state,
-				    AC97_CD_VOLUME_REGISTER,
-				    state->ts_monitor_gain);
-			}
-			break;
-		case AUDIO_CODEC_LOOPB_IN:
-			/* set to the loopback input */
-			tmp_short = RSCR_R_STEREO_MIX|RSCR_L_STEREO_MIX;
-
-			/* see if we need to update monitor loopback */
-			if (state->ts_monitor_gain) {
-				if (state->ts_input_port == AUDIO_LINE_IN) {
-					audiots_or_ac97(state,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (state->ts_input_port ==
-				    AUDIO_MICROPHONE) {
-					audiots_or_ac97(state,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (state->ts_input_port == AUDIO_CD) {
-					audiots_or_ac97(state,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			}
-			break;
-		default:
-			/* unknown or inclusive input ports */
-			mutex_exit(&state->ts_lock);
-			ATRACE_32("ts_ad_set_config() bad in port", port);
-			rc =  AUDIO_FAILURE;
-			goto done;
-		}
-		/* select the input */
-		audiots_set_ac97(state, AC97_RECORD_SELECT_CTRL_REGISTER,
-		    tmp_short);
-
-		/* if an input port then make sure we aren't muted */
-		if (port != AUDIO_NONE &&
-		    (state->ts_shadow[TS_CODEC_REG(AC97_RECORD_GAIN_REGISTER)] &
-		    RGR_MUTE)) {
-			audiots_and_ac97(state, AC97_RECORD_GAIN_REGISTER,
-			    (uint16_t)~RGR_MUTE);
-		}
-
-		state->ts_input_port = port;
-
-		mutex_exit(&state->ts_lock);
-	}
-
-done:
-	ATRACE_32("audiots_set_port()", rc);
-
-	ASSERT(!mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_set_port() */
-
-/*
- * audiots_start_play()
- *
- * Description:
- *	The audio core uses a single DMA buffer which is divided into two
- *	halves. An interrupt is generated when the middle of the buffer has
- *	been reached and at the end. The audio core resets the pointer back
- *	to the beginning automatically. After the interrupt the driver clears
- *	the buffer and asks the mixer for more audio samples. If there aren't
- *	enough then silence is played out.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Play not started/restarted, no audio to play
- */
-static int
-audiots_start_play(audiots_state_t *state)
-{
-	ddi_acc_handle_t	handle;
-	size_t			buf_size;
-	char			*bptr;
-	int			rc = AUDIO_SUCCESS;
-	int			rs;
-	int			samples;
-	int			str = TS_OUTPUT_STREAM;
-
-	ATRACE("in ts_start_play()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	handle = state->ts_handle;
-
-	/* see if we are already playing */
-	if ((ddi_get32(handle, &state->ts_regs->aud_regs.ap_start) &
-	    TS_OUTPUT_CHANNEL)) {
-		ATRACE("ts_start_play() DMA engine already running", state);
-		ASSERT(rc == AUDIO_SUCCESS);
-		goto done;
-	}
-
-	/* see if we are just paused */
-	if (state->ts_flags & TS_DMA_ENGINE_PAUSED) {
-		ATRACE_32("ts_start_play() DMA paused", state->ts_flags);
-		state->ts_flags &= ~TS_DMA_ENGINE_PAUSED;
-
-		/* make sure it starts playing */
-		ddi_put32(handle, &state->ts_regs->aud_regs.ap_start,
-		    TS_OUTPUT_CHANNEL);
-
-		ATRACE_32("ts_start_play() DMA engine restarted",
-		    state->ts_flags);
-		ASSERT(rc == AUDIO_SUCCESS);
-		goto done;
-	}
-
-	/*
-	 * Okay, we are starting from scratch, so get the first chunks of audio.
-	 * We ask for half of the buffer twice. That way we don't mess up the
-	 * mixer's paradigm of using two buffers. But before we get the audio
-	 * we zero the buffer, giving it silence, just in case there isn't
-	 * enough audio.
-	 */
-	bzero(state->ts_pb, state->ts_pbuf_size);
-
-	samples = state->ts_psample_rate * state->ts_pchannels /
-	    state->ts_ad_info.ad_play.ad_int_rate;
-	/* if not an even number of samples we panic! */
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples % AUDIO_CHANNELS_STEREO) != 0) {
-		ATRACE_32("ts_start_play() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-
-	buf_size = samples * (state->ts_pprecision >> AUDIO_PRECISION_SHIFT);
-	ASSERT((buf_size << 1) <= state->ts_pbuf_size);
-
-	ATRACE("ts_start_play() getting 1st audio", NULL);
-	bptr = state->ts_pb;
-
-	mutex_exit(&state->ts_lock);
-	rs = am_get_audio(state->ts_ahandle, &bptr[0], AUDIO_NO_CHANNEL,
-	    samples);
-	mutex_enter(&state->ts_lock);
-
-	if (rs == 0) {
-		/* there's nothing to play */
-		ATRACE("ts_start_play() nothing to play", NULL);
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-	ATRACE_32("ts_start_play() 1st am_get_audio()", rs);
-	state->ts_pcnt[TS_1ST_HALF] = buf_size;
-	state->ts_psamples[TS_1ST_HALF] = samples;
-
-	ATRACE("ts_start_play() getting 2nd audio", NULL);
-
-	mutex_exit(&state->ts_lock);
-	rs = am_get_audio(state->ts_ahandle, &bptr[buf_size], AUDIO_NO_CHANNEL,
-	    samples);
-	mutex_enter(&state->ts_lock);
-
-	ATRACE_32("ts_start_play() 2nd am_get_audio()", rs);
-	if (rs > 0) {
-		state->ts_pcnt[TS_2ND_HALF] = buf_size;
-	} else {
-		state->ts_pcnt[TS_2ND_HALF] = 0;
-	}
-	state->ts_psamples[TS_2ND_HALF] = samples;
-
-	/* sync the DMA buffer */
-	if (ddi_dma_sync(state->ts_ph, (off_t)0, (off_t)0,
-	    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-		audio_sup_log(state->ts_ahandle, CE_WARN,
-		    "!start_play() ddi_dma_sync() failed, audio lost");
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-
-	/* set ALPHA and FMS to 0 */
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_alpha_fms,
-	    0x0);
-
-	/* set CSO to 0 */
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_cso, 0x0);
-
-	/* set LBA */
-	ddi_put32(handle, &state->ts_regs->aud_ram[str].aram.aram_cptr_lba,
-	    state->ts_pc.dmac_address & ARAM_LBA_MASK);
-
-	/* set ESO */
-	if (state->ts_pchannels == AUDIO_CHANNELS_MONO) {
-		samples = (samples << 1) - 1;
-	} else {
-		samples--;
-	}
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_eso, samples);
-
-	/* enable interrupts */
-	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_ainten,
-	    TS_OUTPUT_CHANNEL);
-
-	/* make sure it starts playing */
-	ddi_put32(handle, &state->ts_regs->aud_regs.ap_start,
-	    TS_OUTPUT_CHANNEL);
-
-	state->ts_flags &= ~(TS_DMA_ENGINE_PAUSED|TS_DMA_ENGINE_EMPTY);
-	state->ts_flags |= TS_DMA_ENGINE_INITIALIZED;
-
-	ATRACE("ts_start_play() successful", state);
-	ASSERT(rc == AUDIO_SUCCESS);
-
-done:
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	return (rc);
-
-}	/* audiots_start_play() */
-
-/*
- * audiots_stop_play()
- *
- * Description:
- *	This routine stops the play DMA engine.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_stop_play(audiots_state_t *state)
-{
-	ATRACE("in ts_stop_play()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_stop,
-	    TS_OUTPUT_CHANNEL);
-
-	AND_SET_WORD(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-	    ~TS_OUTPUT_CHANNEL);
-
-	state->ts_flags &= ~(TS_DMA_ENGINE_INITIALIZED|TS_DMA_ENGINE_PAUSED|
-	    TS_DMA_ENGINE_EMPTY);
-
-	ATRACE("ts_stop_play() returning", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_stop_play() */
-
-/*
- * audiots_start_record()
- *
- * Description:
- *	The record DMA engine works the same way as the play DMA engine.
- *	Record cannot be multi-stream and must be on T2 channel 31.
- *
- *	There isn't a pause for record, only start and stop. So this code
- *	is a touch simpler than starting up play.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Record not started
- */
-/*ARGSUSED*/
-static int
-audiots_start_record(audiots_state_t *state)
-{
-	ddi_acc_handle_t	handle;
-	size_t			buf_size;
-	int			rc = AUDIO_SUCCESS;
-	int			samples;
-	int			str = TS_INPUT_STREAM;
-
-	ATRACE("in ts_start_record()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	handle = state->ts_handle;
-
-	/* see if we are already recording */
-	if ((ddi_get32(handle, &state->ts_regs->aud_regs.ap_start) &
-	    TS_INPUT_CHANNEL)) {
-		ATRACE("ts_start_record() DMA engine already running",
-		    state);
-		ASSERT(rc == AUDIO_SUCCESS);
-		return (rc);
-	}
-
-	/* okay, we start from scratch; start by figuring out sample sizes */
-	samples = state->ts_csample_rate * state->ts_cchannels /
-	    state->ts_ad_info.ad_record.ad_int_rate;
-	/* if not an even number of samples we panic! */
-	if ((samples % AUDIO_CHANNELS_STEREO) != 0) {
-		ATRACE_32("ts_start_record() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-
-	buf_size = samples * (state->ts_cprecision >> AUDIO_PRECISION_SHIFT);
-	ASSERT((buf_size << 1) <= state->ts_cbuf_size);
-
-	state->ts_ccnt = buf_size;
-
-	/* set ALPHA and FMS to 0 */
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_alpha_fms,
-	    0x0);
-
-	/* set CSO to 0 */
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_cso, 0x0);
-
-	/* set LBA */
-	ddi_put32(handle, &state->ts_regs->aud_ram[str].aram.aram_cptr_lba,
-	    state->ts_cc.dmac_address & ARAM_LBA_MASK);
-
-	/* set ESO */
-	if (state->ts_cchannels == AUDIO_CHANNELS_MONO) {
-		samples = (samples << 1) - 1;
-	} else {
-		samples--;
-	}
-	ddi_put16(handle, &state->ts_regs->aud_ram[str].aram.aram_eso, samples);
-
-	/* enable interrupts */
-	OR_SET_WORD(handle, &state->ts_regs->aud_regs.ap_ainten,
-	    TS_INPUT_CHANNEL);
-
-	/* make sure it starts recording */
-	ddi_put32(handle, &state->ts_regs->aud_regs.ap_start, TS_INPUT_CHANNEL);
-
-	state->ts_flags |= TS_DMA_RECORD_START;
-
-	ATRACE("ts_start_record() done", NULL);
-	ASSERT(rc == AUDIO_SUCCESS);
-
-done:
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	if (rc == AUDIO_SUCCESS && (state->ts_flags & TS_PM_SUPPORTED)) {
-		(void) pm_busy_component(state->ts_dip, TS_COMPONENT);
-	}
-
-	return (rc);
-
-}	/* audiots_start_record() */
-
-/*
- * audiots_stop_record()
- *
- * Description:
- *	This routine stops the record DMA engine. Any data in the buffer
- *	is just thrown away. After all, we're done recording so there aren't
- *	any apps that need audio.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-audiots_stop_record(audiots_state_t *state)
-{
-	ddi_acc_handle_t	handle = state->ts_handle;
-
-	ATRACE("in ts_stop_record()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	/* stop recording */
-	ddi_put32(handle, &state->ts_regs->aud_regs.ap_stop, TS_INPUT_CHANNEL);
-
-	AND_SET_WORD(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-	    ~TS_INPUT_CHANNEL);
-
-	state->ts_flags &= ~TS_DMA_RECORD_START;
-
-	ATRACE("ts_stop_record() done", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_stop_record() */
-
-/*
- * audiots_stop_everything()
- *
- * Description:
- *	This routine disables the address engine interrupt for all 32 DMA
- *	engines. Just to be sure, it then explicitly issues a stop command to
- *	the address engine and envelope engines for all 32 channels.
- *
- * NOTE:
- * 	There is a hardware bug that generates a spurious interrupt when the DMA
- *	engines are stopped. It's not consistent - it happens every 1 out of 6
- *	stops or so. It will show up as a record interrupt. The problem is that
- *	once the driver is detached or if the system goes into low power mode,
- *	nobody will service that interrupt. The system will eventually become
- *	unusable.
- *
- * Arguments:
- *	audiots_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audiots_stop_everything(audiots_state_t *state)
-{
-	uint_t		intr;
-
-	ATRACE("in ts_stop_everything()", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-	ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-	    TS_ALL_DMA_OFF);
-
-	ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_stop,
-	    TS_ALL_DMA_ENGINES);
-
-	intr = ddi_get32(state->ts_handle, &state->ts_regs->aud_regs.ap_aint);
-	if (intr != 0) {
-		ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_aint,
-		    intr);
-		ddi_put32(state->ts_handle, &state->ts_regs->aud_regs.ap_ainten,
-		    TS_ALL_DMA_OFF);
-		ATRACE("ts_stop_everything() clearing the bogus interrupt",
-		    intr);
-
-		/*
-		 * Flag this interrupt for the audiots_intr() routine to claim,
-		 * but this interrupt doesn't need to be processed.
-		 */
-		state->ts_flags |= TS_INTR_PENDING;
-	}
-
-	ATRACE("ts_stop_everything() done", state);
-
-	ASSERT(mutex_owned(&state->ts_lock));
-
-}	/* audiots_stop_everything() */
-
-/*
- * audiots_unmap_regs()
- *
- * Description:
- *	This routine unbinds the DMA cookies, frees the DMA buffers,
- *	deallocates the DMA handles, and finally unmaps the Codec's and
- *	DMA engine's registers.
- *
- * Arguments:
- *	audiots_state_t	*state	The device's state structure
- *
- * Returns:
- *	None
- */
-static void
-audiots_unmap_regs(audiots_state_t *state)
-{
-	ATRACE("in ts_unmap_regs()", state);
-
-	(void) ddi_dma_unbind_handle(state->ts_ph);
-	(void) ddi_dma_unbind_handle(state->ts_ch);
-
-	ddi_dma_mem_free(&state->ts_pmh);
-	ddi_dma_mem_free(&state->ts_cmh);
-
-	ddi_dma_free_handle(&state->ts_ph);
-	ddi_dma_free_handle(&state->ts_ch);
-
-	ddi_regs_map_free(&state->ts_handle);
-	ddi_regs_map_free(&state->ts_chandle);
-
-	ATRACE("ts_unmap_regs() returning", state);
-
-}	/* audiots_unmap_regs() */
diff --git a/usr/src/uts/common/io/audio/sada/framework/audio_support.c b/usr/src/uts/common/io/audio/sada/framework/audio_support.c
deleted file mode 100644
index 3d20b6a313..0000000000
--- a/usr/src/uts/common/io/audio/sada/framework/audio_support.c
+++ /dev/null
@@ -1,4209 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-/*
- * Audio Support Module
- *
- * This module is use by Audio Drivers that use the new audio driver
- * architecture. It provides common services to get and set data
- * structures used by Audio Drivers and Audio Personality Modules.
- * It provides the home for the audio_state_t structures, one per
- * audio device.
- *
- * Audio Drivers set their qinit structures to the open, close, info, put
- * and service routines in this module. Then this module determines
- * which Audio Personality Module to call to implement the read, write,
- * and ioctl semantics.
- *
- * This module supports persistent data across driver/module unloads
- * and reloads. The space_*() routines are used to save an main anchor
- * which points to a linked list of instance data structures. Each instance
- * data structure points to APM data structures for its instance. These
- * instance and APM persistent data structures are available only via the
- * main anchor.
- */
-
-#include <sys/note.h>
-#include <sys/varargs.h>
-#include <sys/stream.h>
-#include <sys/modctl.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/strsun.h>
-#include <sys/taskq.h>
-#include <sys/audio.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_apm.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/impl/audio_support_impl.h>
-
-/*
- * Solaris external defines.
- */
-extern pri_t minclsyspri;
-
-/*
- * External functions not declared in header files.
- */
-extern uintptr_t space_fetch(char *key);
-extern void space_free(char *key);
-extern int space_store(char *key, uintptr_t ptr);
-
-/* streams stuff */
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", copyreq))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", copyresp))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", datab))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", iocblk))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", msgb))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", queue))
-
-/* other unshared/stable or no lock needed stuff */
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", audio_channel))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", audio_i_state))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", audio_qptr))
-
-#ifdef DEBUG
-/*
- * Global audio tracing variables
- */
-audio_trace_buf_t audio_trace_buffer[AUDIO_TRACE_BUFFER_SIZE];
-kmutex_t	audio_tb_lock;
-size_t		audio_tb_siz = AUDIO_TRACE_BUFFER_SIZE;
-int		audio_tb_pos = 0;
-uint_t		audio_tb_seq = 0;
-#endif
-
-/*
- * Global hidden variables.
- */
-/* for persistent memory */
-static audio_inst_persist_t **audio_main_anchor = NULL;
-static char *audio_key_class = AUDIO_KEY_CLASS;
-
-static audio_inst_info_t *audio_drv_list_head;
-					/* list of all registered drivers */
-
-/* locks */
-static kmutex_t audio_drv_list_lock;	/* mutex to protect driver list */
-static kmutex_t audio_persist_lock;	/* mutex to protect persistent data */
-_NOTE(MUTEX_PROTECTS_DATA(audio_drv_list_lock, audio_inst_info))
-_NOTE(MUTEX_PROTECTS_DATA(audio_persist_lock, audio_main_anchor))
-_NOTE(MUTEX_PROTECTS_DATA(audio_persist_lock, audio_inst_persist))
-_NOTE(MUTEX_PROTECTS_DATA(audio_persist_lock, audio_apm_persist))
-
-/* console message logging */
-static kmutex_t audio_sup_log_lock;
-static char audio_sup_log_buf[256];
-_NOTE(MUTEX_PROTECTS_DATA(audio_sup_log_lock, audio_sup_log_buf))
-
-static audio_device_t audio_device_info = {
-	AUDIO_NAME,
-	AUDIO_VERSION,
-	AUDIO_CONFIGURATION
-};
-
-/*
- * Local Support Routine Prototypes For Audio Support Module
- */
-static audio_inst_info_t *audio_sup_create_drv_entry(dev_info_t *);
-static audio_inst_info_t *audio_sup_lookup_drv_entry(dev_info_t *);
-static void audio_sup_free_drv_entry(dev_info_t *);
-static void audio_sup_free_apm_persist(audio_state_t *, audio_apm_persist_t *);
-static void audio_sup_free_inst_persist(audio_state_t *,
-	audio_inst_persist_t *);
-static int audio_sup_persist(audio_state_t *, char *);
-
-static int audio_sup_wiocdata(queue_t *, mblk_t *, audio_ch_t *);
-static int audio_sup_wioctl(queue_t *, mblk_t *, audio_ch_t *);
-
-/*
- * Module Linkage Structures
- */
-/* Linkage structure for loadable drivers */
-static struct modlmisc audio_modlmisc = {
-	&mod_miscops,		/* drv_modops */
-	AUDIO_MOD_NAME,		/* drv_linkinfo */
-};
-
-static struct modlinkage audio_modlinkage =
-{
-	MODREV_1,		/* ml_rev */
-	(void*)&audio_modlmisc,	/* ml_linkage */
-	NULL			/* NULL terminates the list */
-};
-
-/*
- *  Standard loadable Module Configuration Entry Points
- */
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when module is first loaded.
- *	Global module locks are initialized here.
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-#ifdef DEBUG
-	/* initialize the trace lock */
-	mutex_init(&audio_tb_lock, NULL, MUTEX_DRIVER, NULL);
-#endif
-
-	/* standard linkage call */
-	if ((error = mod_install(&audio_modlinkage)) != 0) {
-		ATRACE_32("audiosup _init() error 1", error);
-#ifdef DEBUG
-		mutex_destroy(&audio_tb_lock);
-#endif
-		return (error);
-
-	}
-
-	/* init mutexes after we can't have any failures */
-
-	/* initialize the instance list lock */
-	mutex_init(&audio_drv_list_lock, NULL, MUTEX_DRIVER, NULL);
-	/* log buffer mutex */
-	mutex_init(&audio_sup_log_lock, NULL, MUTEX_DRIVER, NULL);
-
-	/* persistent data mutex */
-	mutex_init(&audio_persist_lock, NULL, MUTEX_DRIVER, NULL);
-
-	ATRACE("audiosup _init() successful", 0);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description
- *	Module de-initialization, called when driver is to be unloaded.
- *	Free resources that were allocated in _init().
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int	error;
-
-	ATRACE("in audiosup _fini()", 0);
-
-	/* all drivers must have unregistered */
-	ASSERT(audio_drv_list_head == NULL);
-
-	if ((error = mod_remove(&audio_modlinkage)) != 0) {
-		ATRACE_32("audiosup _fini() mod_remove failed", error);
-
-		return (error);
-
-	}
-
-	/* free instance list lock */
-	mutex_destroy(&audio_drv_list_lock);
-	/* free log lock */
-	mutex_destroy(&audio_sup_log_lock);
-	/* free persistent memory lock */
-	mutex_destroy(&audio_persist_lock);
-
-	ATRACE_32("audiosup _fini() successful", error);
-
-#ifdef DEBUG
-	mutex_destroy(&audio_tb_lock);
-#endif
-
-	return (0);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo	*modinfop	Pointer to an opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int		rc;
-
-	rc = mod_info(&audio_modlinkage, modinfop);
-
-	ATRACE_32("audiosup _info() returning", rc);
-
-	return (rc);
-
-}	/* _info() */
-
-/*
- * Public Audio Device Independent Driver Entry Points
- *
- * Standard Driver Entry Points
- */
-
-/*
- * audio_sup_attach() and audio_sup_detach() are being replaced in the
- * next minor release of Solaris. Audio drivers must be modified to use
- * the new interfaces.
- */
-/*ARGSUSED*/
-audiohdl_t
-audio_sup_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	audio_sup_reg_data_t	data;
-
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-
-	return (audio_sup_register(dip, &data));
-
-}       /* audio_sup_attach() */
-
-/*ARGSUSED*/
-int
-audio_sup_detach(audiohdl_t handle, ddi_detach_cmd_t cmd)
-{
-	return (audio_sup_unregister(handle));
-}
-
-/*
- * audio_sup_register()
- *
- * Description:
- *	This routine initializes structures used for an instance of an
- *	audio driver. Persistent state storage is allocated and used to
- *	store state when drivers are unloaded. This routine always allocates
- *	a very small chunk of memory for each instance which we never free.
- *
- * Arguments:
- *	dev_info_t		*dip	Ptr to the device's dev_info structure
- *	audio_sup_reg_data_t	*data	Registration data
- *
- * Returns:
- *	audiohdl_t			Handle to the audio device is successful
- *	NULL				Attach failed
- */
-audiohdl_t
-audio_sup_register(dev_info_t *dip, audio_sup_reg_data_t *data)
-{
-	audio_inst_info_t	*instp;		/* inst info pointer */
-	audio_state_t		*statep;	/* instance state pointer */
-	audio_ch_t		*chptr;		/* channel pointer */
-	int			i;
-	int			sup_chs;
-
-	ATRACE("in audio_sup_register() dip", dip);
-	ATRACE("audio_sup_register() version", data->asrd_version);
-
-	/* make sure we have a supported version */
-	if (data->asrd_version != AUDIOSUP_VERSION) {
-		ATRACE("audio_sup_register() bad version", data->asrd_version);
-		audio_sup_log(NULL, CE_WARN, "unsupported version: %d",
-		    data->asrd_version);
-		return (NULL);
-	}
-
-	/* minors per inst - reserved audio channels */
-	sup_chs = AUDIO_MINOR_PER_INST - AUDIO_NUM_DEVS;
-	ATRACE_32("audio_sup_register() supported channels per device",
-	    sup_chs);
-
-	/* register and get device instance number */
-	if ((instp = audio_sup_create_drv_entry(dip)) == NULL) {
-		ATRACE("audio_sup_register() "
-		    "audio_sup_create_drv_entry() failed", 0);
-
-		return (NULL);
-	}
-
-	statep = &instp->ail_state;
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*statep))
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*chptr))
-
-	/*
-	 * WARNING: From here on all error returns must worry about the
-	 *	driver list.
-	 *
-	 * Initialize the instance mutex and condition variables. Used to
-	 * allocate channels.
-	 */
-	mutex_init(&statep->as_lock, NULL, MUTEX_DRIVER, NULL);
-	cv_init(&statep->as_cv, NULL, CV_DRIVER, NULL);
-
-	/* initialize other state information */
-	statep->as_dip = dip;
-	statep->as_dev_instance = ddi_get_instance(dip);
-	statep->as_major = ddi_driver_major(dip);
-	statep->as_max_chs = sup_chs;
-	statep->as_minors_per_inst = AUDIO_MINOR_PER_INST;
-	statep->as_audio_reserved = AUDIO_NUM_DEVS;
-
-	/* sanity check that we can use the device */
-	ASSERT((sup_chs >= AUDIO_MIN_CLONE_CHS) &&
-	    (sup_chs <= AUDIO_CLONE_CHANLIM));
-
-	/* setup persistent memory */
-	if (audio_sup_persist(statep, data->asrd_key) == AUDIO_FAILURE) {
-		ATRACE("audio_sup_register() couldn't set up persist mem", 0);
-
-		goto error;
-	}
-
-	/*
-	 * WARNING: From here on we cannot fail. Otherwise we would have
-	 *	to loop through the channel structures and clear those CVs
-	 *	and locks.
-	 *
-	 * Initialize the channel structures.
-	 */
-	ATRACE("audio_sup_register() # sup channels", sup_chs);
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < sup_chs; i++, chptr++) {
-		/* most everything is zero, do it quickly */
-		bzero(chptr, sizeof (*chptr));
-
-		/* now do the non-zero members */
-		chptr->ch_statep =		statep;
-		chptr->ch_dev =			NODEV;
-		chptr->ch_info.ch_number =	i;
-		chptr->ch_info.dev_type =	UNDEFINED;
-
-		mutex_init(&chptr->ch_lock, NULL, MUTEX_DRIVER, NULL);
-		mutex_init(&chptr->ch_adata_lock, NULL, MUTEX_DRIVER, NULL);
-		cv_init(&chptr->ch_cv, NULL, CV_DRIVER, NULL);
-	}
-
-	ATRACE_32("audio_sup_register() returning", statep->as_dev_instance);
-
-done:
-	ATRACE("audio_sup_register() handle", statep);
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*statep))
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*chptr))
-
-	return (AUDIO_STATE2HDL(statep));
-
-error:
-	ATRACE("audio_sup_register() error", statep);
-
-	mutex_destroy(&statep->as_lock);
-	cv_destroy(&statep->as_cv);
-
-	audio_sup_free_drv_entry(statep->as_dip);
-
-	return (NULL);
-
-}	/* audio_sup_register() */
-
-/*
- * audio_sup_unregister()
- *
- * Description:
- *	This routine de-initializes structures used for an instance of the
- *	of an audio driver. It doesn't really do too much.
- *
- * Arguments:
- *	audiohdl_t		handle	Handle to the device
- *
- * Returns:
- *	AUDIO_SUCCESS			Detach successful
- *	AUDIO_FAILURE			Detach failed
- */
-int
-audio_sup_unregister(audiohdl_t handle)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_inst_persist_t	*persistp;
-
-	ATRACE("in audio_sup_unregister()", handle);
-
-	/*
-	 * WARNING: From here on all error returns must worry about the
-	 *	instance state structures.
-	 */
-
-	/* get the state pointer for this instance */
-	if ((statep = audio_sup_devinfo_to_state(statep->as_dip)) == NULL) {
-		ATRACE("audio_sup_unregister() "
-		    "audio_sup_devinfo_to_state() failed", 0);
-
-		return (AUDIO_FAILURE);
-
-	}
-
-	/*
-	 * Free the instance persistent data struct if it doesn't point to
-	 * any saved data.
-	 */
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&audio_persist_lock);
-	ASSERT(audio_main_anchor);
-
-	persistp = (audio_inst_persist_t *)statep->as_persistp;
-
-	if (persistp->amp_apmp == NULL) {
-		(void) audio_sup_free_inst_persist(statep, persistp);
-		statep->as_persistp = NULL;
-	}
-
-	mutex_exit(&audio_persist_lock);
-	mutex_exit(&statep->as_lock);
-
-	/* remove the dip from the instance list */
-	ATRACE("audio_sup_unregister() freeing instance", statep->as_dip);
-	audio_sup_free_drv_entry(statep->as_dip);
-
-	ATRACE("audio_sup_unregister() returning", 0);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_unregister() */
-
-/*
- * audio_sup_open()
- *
- * Description:
- *	This routine is called when the kernel wants to open an Audio Driver
- *	channel. It figures out what kind of device it is and calls the
- *	appropriate Audio Personality Module.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the read queue
- *	dev_t		*devp	Pointer to the device
- *	int		flag	Open flags
- *	int		sflag	STREAMS flags
- *	cred_t		*credp	Pointer to the user's credential struct.
- *
- * Returns:
- *	0			Successfully opened the device
- *	errno			Error number for failed open
- */
-int
-audio_sup_open(queue_t *q, dev_t *devp, int flag, int sflag, cred_t *credp)
-{
-	audio_state_t		*statep;
-	audio_apm_info_t	*apm_infop;
-	audio_ch_t		*chptr;
-	audio_device_type_e	type;
-	int			rc;
-
-	ATRACE("in audio_sup_open()", q);
-
-	/* get the state structure */
-	if ((statep = audio_sup_devt_to_state(*devp)) == NULL) {
-		ATRACE_32(
-		    "audio_sup_open() audio_sup_devt_to_state() failed", 0);
-
-		return (ENXIO);
-
-	}
-	ATRACE("audio_sup_open() statep", statep);
-
-	/* get the device type */
-	type = audio_sup_devt_to_ch_type(statep, *devp);
-	ATRACE_32("audio_sup_open() type", type);
-
-	/* get the APM info structure */
-	if ((apm_infop = audio_sup_get_apm_info(statep, type)) == NULL) {
-		ATRACE_32("audio_sup_open() audio_sup_get_apm_info() failed",
-		    type);
-
-		return (ENXIO);
-
-	}
-	ATRACE("audio_sup_open() apm_infop", apm_infop);
-
-	ASSERT(apm_infop->apm_open);
-
-	rc = (*apm_infop->apm_open)(q, devp, flag, sflag, credp);
-
-	if (rc == AUDIO_SUCCESS) {
-		/* open was successful, make sure we've got routines */
-		chptr = (audio_ch_t *)audio_sup_get_qptr_data(q);
-		if (chptr == NULL ||
-		    chptr->ch_wput == NULL || chptr->ch_wsvc == NULL ||
-		    chptr->ch_rput == NULL || chptr->ch_rsvc == NULL) {
-			ATRACE("audio_sup_open() bad open", chptr);
-			/* close the device */
-			(*apm_infop->apm_close)(q, flag, credp);
-			rc = EIO;
-		}
-	}
-
-	return (rc);
-
-}	/* audio_sup_open() */
-
-/*
- * audio_sup_close()
- *
- * Description:
- *	This routine is called when the kernel wants to close an Audio Driver
- *	channel. It figures out what kind of device it is and calls the
- *	appropriate Audio Personality Module.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the read queue
- *	int		flag	Open flags
- *	cred_t		*credp	Pointer to the user's credential struct.
- *
- * Returns:
- *	0			Successfully closed the device
- *	errno			Error number for failed close
- */
-int
-audio_sup_close(queue_t *q, int flag, cred_t *credp)
-{
-	audio_ch_t		*chptr;
-
-	ATRACE("in audio_sup_close()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_close() bad chptr", 0);
-		return (EIO);
-	}
-
-	ASSERT(chptr->ch_apm_infop);
-	ASSERT(chptr->ch_apm_infop->apm_close);
-	ATRACE("audio_sup_close() chptr->ch_apm_infop", chptr->ch_apm_infop);
-
-	return ((*chptr->ch_apm_infop->apm_close)(q, flag, credp));
-
-}	/* audio_sup_open() */
-
-/*
- * audio_sup_restore_state()
- *
- * Description:
- *	Restore the state of the hardware that the specified APM controls.
- *	The specified APM is called to do the restore. It is up to that
- *	APM's restore state function to restart the hardware or not. The
- *	APM also deals with the direction. If anything keeps the state from
- *	being restored then AUDIO_FAILURE is returned. If AUDIO_ALL_DEVICES
- *	is sent then if any one APM fails the whole call fails. It is also
- *	a failure to restore an APM that doesn't have an apm_restore_state()
- *	function, which is optional.
- *
- * Arguments:
- *	audiohdl_t		handle		Device handle
- *	audio_device_type_e	device		The device to restore
- *	int			dir		The direction to restore
- *
- * Returns:
- *	AUDIO_SUCCESS				State restored
- *	AUDIO_FAILURE				State not restored
- */
-int
-audio_sup_restore_state(audiohdl_t handle, audio_device_type_e device, int dir)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t	*apm_infop;
-	int			found;
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	ATRACE("in audio_sup_restore_state()", handle);
-	ATRACE_32("audio_sup_restore_state() device", device);
-
-	if (statep == NULL) {
-		ATRACE("audio_sup_restore_state() no statep", statep);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	if ((dir & AUDIO_BOTH) == 0) {
-		ATRACE_32("audio_sup_restore_state() no direction", dir);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	mutex_enter(&statep->as_lock);
-	if ((apm_infop = statep->as_apm_info_list) == NULL) {
-		/* nothing to restore, so this is an error */
-		mutex_exit(&statep->as_lock);
-		ATRACE("audio_sup_restore_state() no apm list", statep);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&statep->as_lock);
-
-	/* go through the list looking for APMs that match this device */
-	for (found = 0; apm_infop != NULL; apm_infop = apm_infop->apm_next) {
-		/* does this APM match? */
-		if (device != AUDIO_ALL_DEVICES &&
-		    device != apm_infop->apm_type) {
-			/* nope, it doesn't, so get the next one */
-			ATRACE_32("audio_sup_restore_state() not this one",
-			    apm_infop->apm_type);
-			continue;
-		}
-
-		/* does this APM have a restore function? */
-		if (apm_infop->apm_restore_state == NULL) {
-			/* if AUDIO_ALL_DEVICES then it's okay to be NULL */
-			if (device == AUDIO_ALL_DEVICES) {
-				ATRACE("audio_sup_restore_state() NULL", 0);
-				continue;
-			} else {
-				ATRACE("audio_sup_restore_state() fail NULL",
-				    0);
-				_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-				return (AUDIO_FAILURE);
-			}
-		}
-
-		/* this is ours, so call it */
-		ATRACE("audio_sup_restore_state() calling restore",
-		    apm_infop->apm_restore_state);
-		if ((*apm_infop->apm_restore_state)(statep, apm_infop, dir) ==
-		    AUDIO_FAILURE) {
-			ATRACE("audio_sup_restore_state() restore failed",
-			    apm_infop->apm_restore_state);
-			_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-			return (AUDIO_FAILURE);
-		}
-		ATRACE_32("audio_sup_restore_state() restore succeeded",
-		    apm_infop->apm_type);
-
-		/* are we done? */
-		if (device != AUDIO_ALL_DEVICES) {
-			/* yes, just a single device to restore */
-			ATRACE_32(
-			    "audio_sup_restore_state() single dev success",
-			    device);
-			_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-			return (AUDIO_SUCCESS);
-		}
-	}
-
-	/* if we specify the device then it's possible it wasn't there, so ck */
-	if (device != AUDIO_ALL_DEVICES && !found) {
-		ATRACE_32("audio_sup_restore_state() single device not found",
-		    device);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Either we found the single device or we looked at all the devices
-	 * and they succeeded. Even if there wasn't a single restore function
-	 * to call when AUDIO_ALL_DEVICES was specified we still succeeded.
-	 */
-
-	ATRACE_32("audio_sup_restore_state() all devices found", device);
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_restore_state() */
-
-/*
- * audio_sup_save_state()
- *
- * Description:
- *	Save the state of the hardware that the specified APM controls.
- *	The specified APM is called to do the save. It is up to that
- *	APM's save state function to define any interaction with the
- *	hardware. If anything keeps the state from being saved then
- *	AUDIO_FAILURE is returned. If AUDIO_ALL_DEVICES is sent then if
- *	any one APM fails the whole call fails. It is also a failure to
- *	save an APM that doesn't have an apm_save_state() function, which
- *	is optional.
- *
- * Arguments:
- *	audiohdl_t		handle		Device handle
- *	audio_device_type_e	device		The device to save
- *	int			dir		The direction to save
- *
- * Returns:
- *	AUDIO_SUCCESS				State save
- *	AUDIO_FAILURE				State not save
- */
-int
-audio_sup_save_state(audiohdl_t handle, audio_device_type_e device, int dir)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t	*apm_infop;
-	int			found;
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	ATRACE("in audio_sup_save_state()", handle);
-	ATRACE_32("audio_sup_save_state() device", device);
-
-	if (statep == NULL) {
-		ATRACE("audio_sup_save_state() no statep", statep);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	if ((dir & AUDIO_BOTH) == 0) {
-		ATRACE_32("audio_sup_save_state() no direction", dir);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	mutex_enter(&statep->as_lock);
-	if ((apm_infop = statep->as_apm_info_list) == NULL) {
-		/* nothing to save, so this is an error */
-		mutex_exit(&statep->as_lock);
-		ATRACE("audio_sup_save_state() no apm list", statep);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-	mutex_exit(&statep->as_lock);
-
-	/* go through the list looking for APMs that match this device */
-	for (found = 0; apm_infop != NULL; apm_infop = apm_infop->apm_next) {
-		/* does this APM match? */
-		if (device != AUDIO_ALL_DEVICES &&
-		    device != apm_infop->apm_type) {
-			/* nope, it doesn't, so get the next one */
-			ATRACE_32("audio_sup_save_state() not this one",
-			    apm_infop->apm_type);
-			continue;
-		}
-
-		/* does this APM have a save function? */
-		if (apm_infop->apm_save_state == NULL) {
-			/* if AUDIO_ALL_DEVICES then it's okay to be NULL */
-			if (device == AUDIO_ALL_DEVICES) {
-				ATRACE("audio_sup_save_state() NULL", 0);
-				continue;
-			} else {
-				ATRACE("audio_sup_save_state() fail NULL",
-				    0);
-				_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-				return (AUDIO_FAILURE);
-			}
-		}
-
-		/* this is ours, so call it */
-		ATRACE("audio_sup_save_state() calling save",
-		    apm_infop->apm_save_state);
-		if ((*apm_infop->apm_save_state)(statep, apm_infop, dir) ==
-		    AUDIO_FAILURE) {
-			ATRACE("audio_sup_save_state() save failed",
-			    apm_infop->apm_save_state);
-			_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-			return (AUDIO_FAILURE);
-		}
-		ATRACE_32("audio_sup_save_state() save succeeded",
-		    apm_infop->apm_type);
-
-		/* are we done? */
-		if (device != AUDIO_ALL_DEVICES) {
-			/* yes, just a single device to save */
-			ATRACE_32(
-			    "audio_sup_save_state() single dev success",
-			    device);
-			_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-			return (AUDIO_SUCCESS);
-		}
-	}
-
-	/* if we specify the device then it's possible it wasn't there, so ck */
-	if (device != AUDIO_ALL_DEVICES && !found) {
-		ATRACE_32("audio_sup_save_state() single device not found",
-		    device);
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Either we found the single device or we looked at all the devices
-	 * and they succeeded. Even if there wasn't a single save function
-	 * to call when AUDIO_ALL_DEVICES was specified we still succeeded.
-	 */
-
-	ATRACE_32("audio_sup_save_state() all devices found", device);
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_save_state() */
-
-/*
- * audio_sup_getinfo()
- *
- * XXX this function is incorrect and should be obsoleted
- *
- * Description:
- *	Get driver information.
- *
- * Arguments:
- *	def_info_t	*dip	Pointer to the device's dev_info structure
- *				WARNING: Don't use this dev_info structure
- *	ddi_info_cmd_t	infocmd	Getinfo command
- *	void		*arg	Command specific argument
- *	void		**result Pointer to the requested information
- *
- * Returns:
- *	DDI_SUCCESS		The information could be returned
- *	DDI_FAILURE		The information couldn't be returned
- */
-/*ARGSUSED*/
-int
-audio_sup_getinfo(dev_info_t *dip, ddi_info_cmd_t infocmd, void *arg,
-    void **result)
-{
-	int			error;
-	int			instance = 0;
-	audio_state_t		*statep;
-
-	switch (infocmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		statep = audio_sup_devt_to_state((dev_t)arg);
-		if ((statep == NULL) || (statep->as_dip == NULL)) {
-			error = DDI_FAILURE;
-		} else {
-			*result = (void *)statep->as_dip;
-			error = DDI_SUCCESS;
-		}
-		break;
-
-	case DDI_INFO_DEVT2INSTANCE:
-		instance = AUDIO_MINOR_TO_INST((dev_t)arg);
-		*result = (void *)(uintptr_t)instance;
-
-		if (*result != NULL) {
-			error = DDI_SUCCESS;
-		} else {
-			error = DDI_FAILURE;
-		}
-		break;
-
-	default:
-		error = DDI_FAILURE;
-		break;
-	}
-
-	return (error);
-
-}	/* audio_sup_getinfo() */
-
-/*
- * audio_sup_rput()
- *
- * Description:
- *	Make sure we have a valid function pointer. If we do we make the call.
- *
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *	mblk_t		*mp	Ptr to the msg block being passed to the queue
- *
- * Returns:
- *	0			Always returns 0
- */
-int
-audio_sup_rput(queue_t *q, mblk_t *mp)
-{
-	audio_ch_t		*chptr;
-	int			rc;
-
-	ATRACE("in audio_sup_rput()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_rput() bad chptr", 0);
-		return (0);
-	}
-	ASSERT(chptr->ch_rput);
-
-	ATRACE("audio_sup_rput() calling ch_rput()", chptr);
-
-	rc = chptr->ch_rput(q, mp);
-
-	ATRACE_32("audio_sup_rput() ch_rput() returned", rc);
-
-	return (rc);
-
-}	/* audio_sup_rput() */
-
-/*
- * audio_sup_rsvc()
- *
- * Description:
- *	Make sure we have a valid function pointer. If we do we make the call.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *
- * Returns:
- *	0			Always returns 0
- */
-int
-audio_sup_rsvc(queue_t *q)
-{
-	audio_ch_t		*chptr;
-	int			rc;
-
-	ATRACE("in audio_sup_rsvc()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_rsvc() null chptr", 0);
-		return (0);
-	}
-	ASSERT(chptr->ch_rsvc);
-
-	ATRACE("audio_sup_rsvc() calling ch_rsvc()", chptr);
-
-	rc = chptr->ch_rsvc(q);
-
-	ATRACE_32("audio_sup_rsvc() ch_rsvc() returned", rc);
-
-	return (rc);
-
-}	/* audio_sup_rsvc() */
-
-/*
- * audio_sup_wput()
- *
- * Description:
- *	Make sure we have a valid function pointer. If we do we make the call.
- *
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *	mblk_t		*mp	Ptr to the msg block being passed to the queue
- *
- * Returns:
- *	0			Always returns 0
- */
-int
-audio_sup_wput(queue_t *q, mblk_t *mp)
-{
-	audio_ch_t		*chptr;
-	audio_i_state_t		*cmd;
-	struct iocblk		*iocbp;
-	struct copyresp		*csp;
-	int			rc = 0;
-
-	ATRACE("in audio_sup_wput()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_close() bad chptr", 0);
-		return (0);
-	}
-	ASSERT(chptr->ch_wput);
-
-	/* pick off the audio support ioctls, there aren't very many */
-	ATRACE_32("audio_sup_wput() type", mp->b_datap->db_type);
-	switch (mp->b_datap->db_type) {
-
-	case M_IOCTL:
-		ATRACE("audio_sup_wput() IOCTL ", chptr);
-		iocbp = (struct iocblk *)mp->b_rptr;	/* ptr to ioctl info */
-
-		switch (iocbp->ioc_cmd) {
-		case AUDIO_GET_CH_NUMBER:
-		case AUDIO_GET_CH_TYPE:
-		case AUDIO_GET_NUM_CHS:
-		case AUDIO_GET_AD_DEV:
-		case AUDIO_GET_APM_DEV:
-		case AUDIO_GET_AS_DEV:
-			ATRACE_32("audio_sup_wput() "
-			    "IOCTL calling audio_sup_wioctl()", iocbp->ioc_cmd);
-			rc = audio_sup_wioctl(q, mp, chptr);
-			break;
-		default:
-			ATRACE("audio_sup_wput() IOCTL calling ch_wput()",
-			    chptr);
-			rc = chptr->ch_wput(q, mp);
-		}
-		break;
-	case M_IOCDATA:
-		ATRACE("audio_sup_wput() IOCDATA ", chptr);
-		iocbp = (struct iocblk *)mp->b_rptr;	/* ptr to ioctl info */
-		csp = (struct copyresp *)mp->b_rptr;	/* copy response ptr */
-		cmd = (audio_i_state_t *)csp->cp_private; /* get state info */
-
-		if (cmd == NULL) {
-			ATRACE("audio_sup_wput() M_IOCDATA NULL cmd calling "
-			    "ch_wput()", chptr);
-			csp->cp_rval = 0;
-			rc = chptr->ch_wput(q, mp);
-			break;
-		}
-
-		switch (cmd->ais_command) {
-		case AUDIO_COPY_OUT_CH_NUMBER:
-		case AUDIO_COPY_OUT_CH_TYPE:
-		case AUDIO_COPY_OUT_NUM_CHS:
-		case AUDIO_COPY_OUT_AD_DEV:
-		case AUDIO_COPY_OUT_APM_DEV:
-		case AUDIO_COPY_OUT_AS_DEV:
-			ATRACE_32("audio_sup_wput() "
-			    "IOCDATA calling audio_sup_wiocdata()",
-			    cmd->ais_command);
-			rc = audio_sup_wiocdata(q, mp, chptr);
-			break;
-		default:
-			ATRACE("audio_sup_wput() IOCDATA calling ch_wput()",
-			    chptr);
-			rc = chptr->ch_wput(q, mp);
-		}
-		break;
-	default:
-		ATRACE("audio_sup_wput() calling ch_wput()", chptr);
-		rc = chptr->ch_wput(q, mp);
-	}
-
-	ATRACE_32("audio_sup_wput() ch_wput() returned", rc);
-
-	return (rc);
-
-}	/* audio_sup_wput() */
-
-/*
- * audio_sup_wsvc()
- *
- * Description:
- *	Make sure we have a valid function pointer. If we do we make the call.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *
- * Returns:
- *	0			Always returns 0
- */
-int
-audio_sup_wsvc(queue_t *q)
-{
-	audio_ch_t		*chptr;
-	int			rc;
-
-	ATRACE("in audio_sup_wsvc()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_wsvc() bad chptr", 0);
-		return (EIO);
-	}
-	ASSERT(chptr->ch_wsvc);
-
-	ATRACE("audio_sup_wsvc() calling ch_wsvc()", chptr);
-
-	rc = chptr->ch_wsvc(q);
-
-	ATRACE_32("audio_sup_wsvc() ch_wsvc() returned", rc);
-
-	return (rc);
-
-}	/* audio_sup_wsvc() */
-
-/*
- * Public Audio Personality Module Support Routines
- *
- * Channel Routines
- */
-
-/*
- * audio_sup_alloc_ch()
- *
- * Description:
- *	Go through the list of channels. Find the first one that isn't
- *	assigned and take it. If there aren't any channels then depending
- *	on the flags either wait for a channel to become free or return
- *	NULL
- *
- *	CAUTION: Make sure *error is always set before returning.
- *
- *	NOTE: This routine expects the Audio Personality Module to fill in
- *		all the members of the audio_ch_t structure.
- *
- * Arguments:
- *	audio_state_t		*statep	The device state structure
- *	int			*error	Error code
- *	audio_device_type_e	type	The device type
- *	int			flags	AUDIO_NO_SLEEP or AUDIO_SLEEP
- *
- * Returns:
- *	valid pointer to ch		Channel allocated
- *	NULL				Channel not allocated
- */
-audio_ch_t *
-audio_sup_alloc_ch(audio_state_t *statep, int *error, audio_device_type_e type,
-    int flags)
-{
-	audio_ch_t		*chptr;
-	audio_apm_info_t	*apm_infop;
-	int			i;
-	int			max_chs = statep->as_max_chs;
-	int			rc;
-
-	ATRACE("in audio_sup_alloc_ch()", statep);
-
-	ASSERT((flags & (AUDIO_NO_SLEEP|AUDIO_SLEEP)) !=
-	    (AUDIO_NO_SLEEP|AUDIO_SLEEP));
-
-	/* make sure there's an apm_infop for the type */
-	if ((apm_infop = audio_sup_get_apm_info(statep, type)) == NULL) {
-		ATRACE("audio_sup_alloc_ch() audio_sup_get_apm_info() failed",
-		    statep);
-		*error = EIO;
-
-		return (NULL);
-
-	}
-
-	/* find the first unused channel */
-	mutex_enter(&statep->as_lock);
-
-	/* cv_broadcast() means we may need to try many times */
-	while (statep->as_ch_inuse >= max_chs) {
-		/* no channels available right now, do we wait? */
-		if (flags & AUDIO_NO_SLEEP) {
-			/* don't wait for a channel to become free */
-			mutex_exit(&statep->as_lock);
-			ATRACE("audio_sup_alloc_ch() no ch return", 0);
-			*error = EBUSY;
-
-			return (NULL);
-
-		}
-
-		/* wait for a channel to become free */
-		rc = cv_wait_sig(&statep->as_cv, &statep->as_lock);
-		if (rc <= 0) {
-			ATRACE("audio_sup_alloc_ch() max chs signal wakeup",
-			    statep);
-			/*
-			 * This channel may have had a signal, but that doesn't
-			 * mean any of the other channels may proceed. So make
-			 * sure every channel gets another go.
-			 */
-			mutex_exit(&statep->as_lock);
-			ATRACE("audio_sup_alloc_ch() no ch sig return", 0);
-			*error = EINTR;
-
-			return (NULL);
-
-		}
-	}
-
-	/* we've got a channel, so find it */
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < max_chs; i++, chptr++) {
-		mutex_enter(&chptr->ch_lock);
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED)) {
-			/* found it! */
-			mutex_exit(&chptr->ch_lock);
-			break;
-		}
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*chptr))
-	ASSERT(i <= max_chs);
-	ASSERT(i == chptr->ch_info.ch_number);
-	ASSERT(statep == chptr->ch_statep);
-
-	ATRACE("audio_sup_alloc_ch() found channel", chptr);
-
-	chptr->ch_info.dev_type = type;
-	chptr->ch_apm_infop = apm_infop;
-	chptr->ch_flags |= AUDIO_CHNL_ALLOCATED;
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*chptr))
-
-	/* just to be sure */
-	mutex_enter(&chptr->ch_adata_lock);
-	chptr->ch_adata = NULL;
-	chptr->ch_adata_end = NULL;
-	chptr->ch_adata_cnt = 0;
-	mutex_exit(&chptr->ch_adata_lock);
-
-	statep->as_ch_inuse++;		/* inc. the number of allocated chs */
-
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_alloc_ch() returning", chptr);
-
-	*error = 0;	/* no errors */
-
-	return (chptr);
-
-}	/* audio_sup_alloc_ch() */
-
-/*
- * audio_sup_free_ch()
- *
- * Description:
- *	This routine returns a channel structure to the device instance's
- *	pool of channel structures. All the various pointers must be freed
- *	and NULLed before this routine is called. It then does a cv_broadcast()
- *	to wake up any cv_wait()/cv_wait_sig() calls which might be waiting
- *	for a channel to be freed.
- *
- * Arguments:
- *	audio_ch_t	*chptr	The channel structure to free
- *
- * Returns:
- *	AUDIO_SUCCESS		No error
- *	AUDIO_FAILURE		One of the pointers is not set to NULL or
- *				chptr is not valid
- */
-int
-audio_sup_free_ch(audio_ch_t *chptr)
-{
-	audio_state_t		*statep;
-	audio_apm_info_t	*apm_infop;
-
-	ATRACE("in audio_sup_free_ch()", chptr);
-
-	if (chptr == NULL) {
-		ATRACE("audio_sup_free_ch() chptr == NULL", chptr);
-
-		return (AUDIO_FAILURE);
-	}
-
-	statep = chptr->ch_statep;
-	apm_infop = chptr->ch_apm_infop;
-
-	ASSERT(chptr->ch_statep == statep);
-	ASSERT(apm_infop != NULL);
-
-	/* finally, clear the channel structure and make available for reuse */
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&chptr->ch_lock);
-
-#ifdef DEBUG
-	mutex_enter(&chptr->ch_adata_lock);
-	ASSERT(chptr->ch_adata_cnt == 0);
-	mutex_exit(&chptr->ch_adata_lock);
-#endif
-
-	ATRACE("audio_sup_free_ch() chptr", chptr);
-	ATRACE("audio_sup_free_ch() ch_private", chptr->ch_private);
-	ATRACE("audio_sup_free_ch() ch_info.info", chptr->ch_info.info);
-
-	if (chptr->ch_private) {
-		ATRACE("audio_sup_free_ch() chptr->ch_private != NULL",
-		    chptr->ch_private);
-		mutex_exit(&chptr->ch_lock);
-		mutex_exit(&statep->as_lock);
-		return (AUDIO_FAILURE);
-
-	}
-	if (chptr->ch_info.info) {
-		ATRACE("audio_sup_free_ch() chptr->ch_info.info != NULL",
-		    chptr->ch_info.info);
-		mutex_exit(&chptr->ch_lock);
-		mutex_exit(&statep->as_lock);
-		return (AUDIO_FAILURE);
-
-	}
-
-	/* free the message list */
-	ATRACE("audio_sup_free_ch() freeing saved messages", chptr);
-	audio_sup_flush_audio_data(chptr);
-
-	chptr->ch_flags =		0;
-	chptr->ch_info.pid =		0;
-
-	statep->as_ch_inuse--;
-
-	ATRACE("audio_sup_free_ch() resetting channel info", chptr);
-
-	chptr->ch_qptr =		NULL;
-	mutex_enter(&chptr->ch_adata_lock);
-	chptr->ch_adata =		NULL;
-	chptr->ch_adata_end =		NULL;
-	chptr->ch_adata_cnt =		0;
-	mutex_exit(&chptr->ch_adata_lock);
-	chptr->ch_apm_infop =		NULL;
-	chptr->ch_wput =		NULL;
-	chptr->ch_wsvc =		NULL;
-	chptr->ch_rput =		NULL;
-	chptr->ch_rsvc =		NULL;
-	chptr->ch_dir =			0;
-	chptr->ch_dev =			NODEV;
-	chptr->ch_info.dev_type =	UNDEFINED;
-
-	mutex_exit(&chptr->ch_lock);
-
-	/* the channel is freed, so send the broadcast */
-
-	cv_broadcast(&statep->as_cv);
-
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_free_ch() returning success", statep);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_free_ch() */
-
-/*
- * Persistent Memory Routines
- */
-
-/*
- * audio_sup_get_persist_state()
- *
- * Description:
- *	Search through the linked list of audio_apm_persist data structures
- *	for this driver instance's persistent data for the specifed audio
- *	personality module.
- *
- * Arguments:
- *	audio_state_t		*statep		Device state structure
- *	audio_device_type_e	dev_type	APM to set data for
- *
- * Returns:
- *	valid pointer			Pointer to the saved data
- *	NULL				Couldn't find the data
- */
-void *
-audio_sup_get_persist_state(audio_state_t *statep, audio_device_type_e dev_type)
-{
-	audio_inst_persist_t	*instp;
-	audio_apm_persist_t	*tmp;
-	void			*ret_ptr;
-
-	ATRACE("in audio_sup_get_persist_state()", statep);
-	ATRACE_32("audio_sup_get_persist_state() dev_type", dev_type);
-
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&audio_persist_lock);
-
-	ASSERT(statep->as_persistp);
-	instp = statep->as_persistp;
-
-	for (tmp = instp->amp_apmp; tmp; tmp = tmp->ap_next) {
-		if (tmp->ap_apm_type == dev_type) {
-			ret_ptr = tmp->ap_data;	/* make warlock happy */
-			mutex_exit(&audio_persist_lock);
-			mutex_exit(&statep->as_lock);
-			ATRACE("audio_sup_get_persist_state() found state",
-			    ret_ptr);
-			return (ret_ptr);
-		}
-	}
-	mutex_exit(&audio_persist_lock);
-	mutex_exit(&statep->as_lock);
-
-	return (NULL);
-
-}	/* audio_sup_get_persist_state() */
-
-/*
- * audio_sup_free_persist_state()
- *
- * Description:
- *	Search through the linked list of audio_apm_persist data structures
- *	for this driver instance's persistent data for the specifed audio
- *	personality module. When found remove it from the linked list and
- *	free it.
- *
- *	If the dev_type is set to AUDIO_ALL_DEVICES then we free all of
- *	the entries.
- *
- * Arguments:
- *	audio_state_t		*statep		Device state structure
- *	audio_device_type_e	dev_type	APM to free data for
- *
- * Returns:
- *	AUDIO_SUCCESS			Memory freed
- *	AUDIO_FAILURE			Memory not found to free
- */
-int
-audio_sup_free_persist_state(audio_state_t *statep,
-	audio_device_type_e dev_type)
-{
-	audio_apm_persist_t	**plist;
-	audio_apm_persist_t	*list;
-	audio_apm_persist_t	*tmp;
-
-	ATRACE("in audio_sup_free_persist_state()", statep);
-	ATRACE_32("audio_sup_free_persist_state() dev_type", dev_type);
-
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&audio_persist_lock);
-	plist = &((audio_inst_persist_t *)statep->as_persistp)->amp_apmp;
-
-	if (*plist == NULL) {
-		/* no items on the list */
-		mutex_exit(&audio_persist_lock);
-		mutex_exit(&statep->as_lock);
-		ATRACE("audio_sup_free_persist_state() empty list", plist);
-		return (AUDIO_FAILURE);
-	}
-
-	list = *plist;
-
-	/* see if we clear all, or just one that matches dev_type */
-	if (dev_type == AUDIO_ALL_DEVICES) {
-		/* clear all of them */
-		ATRACE("audio_sup_free_persist_state() clear all", plist);
-		while (list) {
-			tmp = list;
-			list = list->ap_next;
-			kmem_free(tmp->ap_data, tmp->ap_size);
-			kmem_free(tmp, sizeof (*tmp));
-		}
-		((audio_inst_persist_t *)statep->as_persistp)->amp_apmp = NULL;
-
-		mutex_exit(&audio_persist_lock);
-		mutex_exit(&statep->as_lock);
-
-		ATRACE("audio_sup_free_persist_state() free all done", statep);
-
-		return (AUDIO_SUCCESS);
-
-	} else {
-		/* clear just the match */
-		ATRACE("audio_sup_free_persist_state() clear match", dev_type);
-		while (list) {
-			if (list->ap_apm_type == dev_type) {
-				/* remove from the linked list */
-				*plist = list->ap_next;
-
-				ATRACE("audio_sup_free_persist_state() "
-				    "freeing state", list);
-
-				kmem_free(list->ap_data, list->ap_size);
-				kmem_free(list, sizeof (*list));
-
-				mutex_exit(&audio_persist_lock);
-				mutex_exit(&statep->as_lock);
-				return (AUDIO_SUCCESS);
-			}
-			plist = &list->ap_next;
-			list = list->ap_next;
-		}
-	}
-
-	mutex_exit(&audio_persist_lock);
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_free_persist_state() not found, failed", 0);
-
-	return (AUDIO_FAILURE);
-
-}	/* audio_sup_free_persist_state() */
-
-/*
- * audio_sup_set_persist_state()
- *
- * Description:
- *	Search through the list to see if we already have data for this
- *	APM type. If so then free it, then update with the new data. If
- *	the APM type isn't found then allocate an audio_apm_persist_t
- *	structure and add to the list.
- *
- * Arguments:
- *	audio_state_t		*statep		Device state structure
- *	audio_device_type_e	dev_type	APM to set data for
- *	void			*state_data	The data to save
- *	size_t			state_size	Size of the persistent data
- *
- * Returns:
- *	AUDIO_SUCCESS			Successfully set persistent state info
- *	AUDIO_FAILURE			Couldn't set persistent state info
- */
-int
-audio_sup_set_persist_state(audio_state_t *statep, audio_device_type_e dev_type,
-	void *state_data, size_t state_size)
-{
-	audio_apm_persist_t	*anchor;
-	audio_apm_persist_t	*tmp;
-
-	ATRACE("in audio_sup_set_persist_state() statep", statep);
-	ATRACE_32("audio_sup_set_persist_state() dev_type", dev_type);
-	ATRACE("audio_sup_set_persist_state() data", state_data);
-	ATRACE("audio_sup_set_persist_state() size", state_size);
-
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&audio_persist_lock);
-
-	ASSERT(statep->as_persistp);
-
-	/* get the anchor */
-	anchor = ((audio_inst_persist_t *)statep->as_persistp)->amp_apmp;
-
-	/* look for a matching device type */
-	for (tmp = anchor; tmp; tmp = tmp->ap_next) {
-		if (tmp->ap_apm_type == dev_type) {
-			ATRACE("audio_sup_set_persist_state() found tmp", tmp);
-			break;
-		}
-	}
-
-	/*
-	 * If not found or first on list then allocate a new structure and
-	 * place on the list.
-	 */
-	if (tmp == NULL) {
-		/* allocate an audio_apm_persist struct */
-		tmp = kmem_alloc(sizeof (*tmp), KM_SLEEP);
-		tmp->ap_apm_type = dev_type;
-
-		/* and place at the beginning of the list */
-		tmp->ap_next = anchor;
-		((audio_inst_persist_t *)statep->as_persistp)->amp_apmp = tmp;
-	} else {
-		/* we found on the list, so free old data */
-		kmem_free(tmp->ap_data, tmp->ap_size);
-	}
-
-	/* update the rest of the new data structure */
-	tmp->ap_data = state_data;
-	tmp->ap_size = state_size;
-
-	mutex_exit(&audio_persist_lock);
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_set_persist_state() new return", tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_set_persist_state() */
-
-/*
- * Device Independent Driver Registration Routines
- */
-
-/*
- * audio_sup_register_apm()
- *
- * Description:
- *	Register the Audio Personality Module with this instance of the
- *	Audio Driver. This provides a place to store state information
- *	for the APM.
- *
- *	We only allow one instance of an APM present for each instance of
- *	an Audio Driver.
- *
- *	NOTE: Instance and type are mandatory.
- *
- *	NOTE: It is okay for memory allocation to sleep.
- *
- * Arguments:
- *	audio_state_t	*statep		Device state structure
- *	audio_device_type_e type	APM type
- *	audio_apm_reg_t	*reg_info	Ptr to APM registration information
- *
- * Returns:
- *	valid pointer			The audio_apm_info structure registered
- *	NULL				Couldn't register the APM
- */
-audio_apm_info_t *
-audio_sup_register_apm(audio_state_t *statep, audio_device_type_e type,
-	audio_apm_reg_t *reg_info)
-{
-	audio_apm_info_t	*apm_infop;
-	audio_apm_info_t	*tmp_apm_infop;
-
-	ATRACE_32("in audio_sup_register_apm()", type);
-
-	/* make sure the registration information data structure is okay */
-	if (reg_info->aar_version != AM_AAR_VERSION) {
-		ATRACE("audio_sup_register_apm() bad version",
-		    reg_info->aar_version);
-		return (NULL);
-	}
-
-	/* we must have an open() and close() routine */
-	if (reg_info->aar_apm_open == NULL || reg_info->aar_apm_close == NULL) {
-		ATRACE("audio_sup_register_apm() aar_apm_open()",
-		    reg_info->aar_apm_open);
-		ATRACE("audio_sup_register_apm() apm_close()",
-		    reg_info->aar_apm_close);
-
-		return (NULL);
-
-	}
-
-	/* allocate the structure now so we won't sleep with as_lock held */
-	tmp_apm_infop = kmem_zalloc(sizeof (*tmp_apm_infop), KM_SLEEP);
-
-	/* first make sure we haven't already registered this type before */
-	mutex_enter(&statep->as_lock);
-
-	for (apm_infop = statep->as_apm_info_list; apm_infop != NULL;
-	    apm_infop = apm_infop->apm_next) {
-		if (apm_infop->apm_type == type) {
-			mutex_exit(&statep->as_lock);
-			kmem_free(tmp_apm_infop, sizeof (*tmp_apm_infop));
-			ATRACE("audio_sup_register_apm() "
-			    "duplicate diaudio type", 0);
-
-			return (NULL);
-
-		}
-	}
-	apm_infop = tmp_apm_infop;
-
-	ATRACE("audio_sup_register_apm() not a duplicate, ok to continue", 0);
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	mutex_init(&apm_infop->apm_lock, NULL, MUTEX_DRIVER, NULL);
-
-	apm_infop->apm_open =		reg_info->aar_apm_open;
-	apm_infop->apm_close =		reg_info->aar_apm_close;
-	apm_infop->apm_restore_state =	reg_info->aar_apm_restore_state;
-	apm_infop->apm_save_state =	reg_info->aar_apm_save_state;
-	apm_infop->apm_info =		reg_info->aar_dev_info;
-	apm_infop->apm_type =		type;
-	apm_infop->apm_private =	reg_info->aar_private;
-	apm_infop->apm_ad_infop =	reg_info->aar_info;
-	apm_infop->apm_ad_state =	reg_info->aar_state;
-
-	/* put at the head of the list */
-	apm_infop->apm_next = statep->as_apm_info_list;
-	statep->as_apm_info_list = apm_infop;
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*apm_infop))
-
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_register_apm() returning successful", apm_infop);
-
-	return (apm_infop);
-
-}	/* audio_sup_register_apm() */
-
-/*
- *
- * Description:
- *	Unregister the Audio Personality Module from this instance of the
- *	Audio Driver. If the APM wasn't registered we silently and
- *	successfully fail and return. Otherwise we free the structure which
- *	represents the APM. However, if the private data wasn't freed we
- *	fail.
- *
- * Arguments:
- *	audio_state_t		*statep		device state structure
- *	audio_device_type_e	type		APM type
- *
- * Returns:
- *	AUDIO_SUCCESS			APM unregistered
- *	AUDIO_FAILURE			APM private data not freed
- */
-int
-audio_sup_unregister_apm(audio_state_t *statep, audio_device_type_e type)
-{
-	audio_apm_info_t	*apm_infop;
-	audio_apm_info_t	**papm_infop;
-
-	ATRACE_32("in audio_sup_unregister_apm()", type);
-
-	/* protect the personality module list */
-	mutex_enter(&statep->as_lock);
-
-	papm_infop = &statep->as_apm_info_list;
-	apm_infop = *papm_infop;
-
-	while (apm_infop) {
-		if (apm_infop->apm_type == type) {
-			break;
-		}
-		papm_infop = &apm_infop->apm_next;
-		apm_infop = *papm_infop;
-	}
-	ATRACE("audio_sup_unregister_apm() while done", apm_infop);
-
-	/* type not found on the list or the list is empty */
-	if (apm_infop == NULL) {
-		mutex_exit(&statep->as_lock);
-		ATRACE_32("audio_sup_unregister_apm() not found on list", type);
-
-		return (AUDIO_SUCCESS);
-
-	}
-
-	/* make sure the private data has been freed */
-	if (apm_infop->apm_private) {
-		mutex_exit(&statep->as_lock);
-		ATRACE("audio_sup_unregister_apm() private data not cleared",
-		    apm_infop->apm_private);
-
-		return (AUDIO_FAILURE);
-
-	}
-	ATRACE("audio_sup_unregister_apm() ok to unreregister", apm_infop);
-
-	/* remove the item by bypassing it */
-	*papm_infop = apm_infop->apm_next;
-
-	/* APM off the list, so okay to release the lock */
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("audio_sup_unregister_apm() freeing apm_infop", apm_infop);
-	mutex_destroy(&apm_infop->apm_lock);
-	kmem_free(apm_infop, sizeof (*apm_infop));
-
-	ATRACE("audio_sup_unregister_apm() done", 0);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_unregister_apm() */
-
-/*
- * Audio task queue routines
- */
-
-/*
- * audio_sup_taskq_create()
- *
- * Description:
- *	Wrapper to abstract the creation of a kernel taskq. We pick some
- *	defaults that make sense for audio, such as single threaded.
- *
- * Arguments:
- *	const char	*q_name		Name of the task queue to create
- *
- * Returns:
- *	handle				Queue successfully created
- *	NULL				Couldn't allocate structs, failed
- */
-audio_taskq_t
-audio_sup_taskq_create(const char *q_name)
-{
-	taskq_t		*tq;
-
-	ATRACE("in audio_sup_taskq_create()", q_name);
-
-	tq = taskq_create(q_name, AUDIO_SUP_TASKQ_NTHREADS, minclsyspri,
-	    AUDIO_SUP_TASKQ_MINALLOC, AUDIO_SUP_TASKQ_MAXALLOC,
-	    TASKQ_PREPOPULATE);
-
-	ATRACE("audio_sup_taskq_create() returning", tq);
-
-	return (AUDIO_TQHDL2AUDIOTQHDL(tq));
-
-}	/* audio_sup_taskq_create() */
-
-/*
- * audio_sup_taskq_destroy()
- *
- * Description:
- *	Destroy the taskq.
- *
- *	CAUTION: If the taskq is used after destroying then the system
- *		will probably panic
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to destroy
- *
- * Returns:
- *	void
- */
-void
-audio_sup_taskq_destroy(audio_taskq_t tq_handle)
-{
-	ATRACE("in audio_sup_taskq_destroy()", tq_handle);
-
-	taskq_destroy(AUDIO_AUDIOTQHDL2TQHDL(tq_handle));
-
-	ATRACE("audio_sup_taskq_destroy() returning", tq_handle);
-
-}	/* audio_sup_taskq_destroy() */
-
-/*
- * audio_sup_taskq_dispatch()
- *
- * Description:
- *	Dispatch a task. The task_function pointer must not be NULL.
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to destroy
- *	void (*)()	task_function	Ptr to the function to execute
- *	void		*arg		Ptr to argument for task_function
- *	int		sleep		KM_SLEEP or KM_NOSLEEP
- *
- * Returns:
- *	AUDIO_SUCCESS			Task scheduled
- *	AUDIO_FAILURE			Task not scheduled or bad task_function
- *					or bad sleep flags
- */
-int
-audio_sup_taskq_dispatch(audio_taskq_t tq_handle,
-    void (*task_function)(void *arg), void *arg, int sleep)
-{
-	taskq_t		*tq;
-	taskqid_t	tid;
-
-	ATRACE("in audio_sup_taskq_dispatch()", tq_handle);
-
-	if (task_function == NULL) {
-		ATRACE("audio_sup_taskq_dispatch() NULL function", tq_handle);
-		return (AUDIO_FAILURE);
-	}
-
-	if (sleep != KM_SLEEP && sleep != KM_NOSLEEP) {
-		ATRACE_32("audio_sup_taskq_dispatch() bad sleep", sleep);
-		return (AUDIO_FAILURE);
-	}
-
-	tq = AUDIO_AUDIOTQHDL2TQHDL(tq_handle);
-
-	if ((tid = taskq_dispatch(tq, task_function, arg, sleep))) {
-		ATRACE("audio_sup_taskq_dispatch() successful", tid);
-		return (AUDIO_SUCCESS);
-	} else {
-		ATRACE("audio_sup_taskq_dispatch() failed", tid);
-		return (AUDIO_FAILURE);
-	}
-
-}	/* audio_sup_taskq_dispatch() */
-
-/*
- * audio_sup_taskq_resume()
- *
- * Description:
- *	Resume task execution.
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to wait on
- *
- * Returns:
- *	void
- */
-void
-audio_sup_taskq_resume(audio_taskq_t tq_handle)
-{
-	ATRACE("in audio_sup_taskq_resume()", tq_handle);
-
-	taskq_resume(AUDIO_AUDIOTQHDL2TQHDL(tq_handle));
-
-	ATRACE("audio_sup_taskq_resume() returning", tq_handle);
-
-}	/* audio_sup_taskq_resume() */
-
-/*
- * audio_sup_taskq_suspended()
- *
- * Description:
- *	Determine if the taskq is running or suspended.
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to wait on
- *
- * Returns:
- *	AUDIO_TASKQ_RUNNING		The taskq is running
- *	AUDIO_TASKQ_SUSPENDED		The taskq is not running (suspended)
- */
-int
-audio_sup_taskq_suspended(audio_taskq_t tq_handle)
-{
-	ATRACE("in audio_sup_taskq_suspend()", tq_handle);
-
-	if (taskq_suspended(AUDIO_AUDIOTQHDL2TQHDL(tq_handle))) {
-		ATRACE("audio_sup_taskq_suspend() returning suspended",
-		    tq_handle);
-		return (AUDIO_TASKQ_SUSPENDED);
-	} else {
-		ATRACE("audio_sup_taskq_suspend() returning running",
-		    tq_handle);
-		return (AUDIO_TASKQ_RUNNING);
-	}
-
-}	/* audio_sup_taskq_suspended() */
-
-/*
- * audio_sup_taskq_suspend()
- *
- * Description:
- *	Tasks on the taskq are suspended when this routine returns.
- *	Running tasks will continue to execute, but all new tasks will
- *	be suspended.
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to wait on
- *
- * Returns:
- *	void
- */
-void
-audio_sup_taskq_suspend(audio_taskq_t tq_handle)
-{
-	ATRACE("in audio_sup_taskq_suspend()", tq_handle);
-
-	taskq_suspend(AUDIO_AUDIOTQHDL2TQHDL(tq_handle));
-
-	ATRACE("audio_sup_taskq_suspend() returning", tq_handle);
-
-}	/* audio_sup_taskq_suspend() */
-
-/*
- * audio_sup_taskq_wait()
- *
- * Description:
- *	Wait for all pending tasks to complete
- *
- * Arguments:
- *	audio_taskq_t	tq_handle	Handle to the taskq to wait on
- *
- * Returns:
- *	void
- */
-void
-audio_sup_taskq_wait(audio_taskq_t tq_handle)
-{
-	ATRACE("in audio_sup_taskq_wait()", tq_handle);
-
-	taskq_wait(AUDIO_AUDIOTQHDL2TQHDL(tq_handle));
-
-	ATRACE("audio_sup_taskq_wait() returning", tq_handle);
-
-}	/* audio_sup_taskq_wait() */
-
-
-/*
- * Audio Data Handling Routines
- */
-
-/*
- * audio_sup_flush_audio_data()
- *
- * Description:
- *	Flush all the data queued up for a channel. We remain locked at
- *	all times so no one else can sneak in and grab a data structure.
- *
- * Arguments:
- *	audio_ch_t	*chptr	Pointer to the channel structure
- *
- * Returns:
- *	void
- */
-void
-audio_sup_flush_audio_data(audio_ch_t *chptr)
-{
-	audio_data_t	*tmp;
-
-	ATRACE("in audio_sup_flush_audio_data()", chptr);
-
-	ASSERT(mutex_owned(&chptr->ch_lock));
-
-	mutex_enter(&chptr->ch_adata_lock);
-
-	while ((tmp = chptr->ch_adata) != 0) {
-		/* set up for next loop */
-		chptr->ch_adata = tmp->adata_next;
-
-		ATRACE("audio_sup_flush_audio_data() freeing data", tmp);
-		audio_sup_free_audio_data(tmp);
-
-		chptr->ch_adata_cnt--;
-	}
-
-	ASSERT(chptr->ch_adata_cnt == 0);
-
-	chptr->ch_adata = NULL;
-	chptr->ch_adata_end = NULL;
-
-	mutex_exit(&chptr->ch_adata_lock);
-
-	ATRACE("audio_sup_flush_audio_data() finished", chptr);
-
-}	/* audio_sup_flush_audio_data() */
-
-/*
- * audio_sup_free_audio_data()
- *
- * Description:
- *	Free the audio data.
- *
- *	NOTE: The audio data structure must already be off the list, so there
- *		isn't a need to lock the data list.
- *
- * Arguments:
- *	audio_data_t	*adata		The audio data structure to free
- *
- * Returns:
- *	void
- */
-void
-audio_sup_free_audio_data(audio_data_t *adata)
-{
-	ATRACE("in audio_sup_free_audio_data()", adata);
-
-	if (adata == NULL) {
-		ATRACE("audio_sup_free_audio_data() nothing to free", adata);
-
-		return;
-
-	}
-
-	ATRACE("audio_sup_free_audio_data() adata differ, orig",
-	    adata->adata_orig);
-	ATRACE("audio_sup_free_audio_data() adata differ, proc",
-	    adata->adata_proc);
-	if (adata->adata_orig) {
-		ATRACE("audio_sup_free_audio_data() freeing original data",
-		    adata->adata_orig);
-		kmem_free(adata->adata_orig, adata->adata_osize);
-	}
-	if (adata->adata_proc) {
-		ATRACE("audio_sup_free_audio_data() freeing processed data",
-		    adata->adata_proc);
-		kmem_free(adata->adata_proc, adata->adata_psize);
-	}
-
-	ATRACE("audio_sup_free_audio_data() freeing adata structure", adata);
-	kmem_free(adata, sizeof (*adata));
-
-	ATRACE("audio_sup_free_audio_data() done", 0);
-
-}	/* audio_sup_free_audio_data() */
-
-/*
- * audio_sup_get_audio_data()
- *
- * Description:
- *	Get the oldest audio data structure off the channel's data list, which
- *	would be the first message.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *
- * Returns:
- *	Valid audio_data_t pointer	The audio data structure
- *	NULL				No audio data available
- */
-audio_data_t *
-audio_sup_get_audio_data(audio_ch_t *chptr)
-{
-	audio_data_t	*tmp;
-
-	ATRACE("in audio_sup_get_audio_data()", chptr);
-
-	mutex_enter(&chptr->ch_adata_lock);
-
-	tmp = chptr->ch_adata;
-
-	if (tmp) {
-		/* set up for next time */
-		chptr->ch_adata = tmp->adata_next;
-
-		chptr->ch_adata_cnt--;
-
-		ASSERT(chptr->ch_adata_cnt >= 0);
-
-		mutex_exit(&chptr->ch_adata_lock);
-
-		ATRACE("audio_sup_get_audio_data() found data to return", tmp);
-
-		return (tmp);
-
-	}
-	ASSERT(chptr->ch_adata_cnt == 0);
-
-	mutex_exit(&chptr->ch_adata_lock);
-
-	ATRACE("audio_sup_get_audio_data() NO data found", chptr);
-
-	return (NULL);
-
-}	/* audio_sup_get_audio_data() */
-
-/*
- * audio_sup_get_audio_data_cnt()
- *
- * Description:
- *	Get the number of data structures currently queued on the data list.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *
- * Returns:
- *	>= 0				The number of queued data structures
- */
-int
-audio_sup_get_audio_data_cnt(audio_ch_t *chptr)
-{
-	int		tmp;
-
-	ATRACE("in audio_sup_get_audio_data_cnt()", chptr);
-
-	mutex_enter(&chptr->ch_adata_lock);
-	ASSERT(chptr->ch_adata_cnt >= 0);
-	tmp = chptr->ch_adata_cnt;
-	mutex_exit(&chptr->ch_adata_lock);
-
-	ATRACE("audio_sup_get_audio_data_cnt() returning", tmp);
-
-	return (tmp);
-
-}	/* audio_sup_get_audio_data_cnt() */
-
-/*
- * audio_sup_get_audio_data_size()
- *
- * Description:
- *	Get the number of bytes stored in data structures that are currently
- *	queued on the data list. Look at the proc_size if it's there and
- *	otherwise look at the orig_size.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *
- * Returns:
- *	>= 0				The number of bytes queued
- */
-int
-audio_sup_get_audio_data_size(audio_ch_t *chptr)
-{
-	audio_data_t	*adata;
-	int		tmp = 0;
-
-	ATRACE("in audio_sup_get_audio_data_size()", chptr);
-
-	/* lock the structure */
-	mutex_enter(&chptr->ch_adata_lock);
-
-	ASSERT(chptr->ch_adata_cnt >= 0);
-
-	adata = chptr->ch_adata;
-	while (adata != 0) {
-		if (adata->adata_proc) {
-			tmp += adata->adata_psize;
-		} else {
-			tmp += adata->adata_osize;
-		}
-		adata = adata->adata_next;
-	}
-
-
-	mutex_exit(&chptr->ch_adata_lock);
-
-	ATRACE("audio_sup_get_audio_data_size() returning", tmp);
-
-	return (tmp);
-
-}	/* audio_sup_get_audio_data_size() */
-
-/*
- * audio_sup_putback_audio_data()
- *
- * Description:
- *	Put the audio data structure back onto the list. It will be the first
- *	to be removed the next time audio_sup_get_audio_data() is called.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *	audio_data_t	*adata		The message to put back on the list
- *
- * Returns:
- *	void
- */
-void
-audio_sup_putback_audio_data(audio_ch_t *chptr, audio_data_t *adata)
-{
-	ATRACE("in audio_sup_putback_audio_data()", chptr);
-
-	if (adata == 0) {
-		ATRACE("audio_sup_putback_audio_data() bad message pointer",
-		    adata);
-
-		return;
-
-	}
-
-	ATRACE("audio_sup_putback_audio_data() putting data back", adata);
-
-	/* lock the data list */
-	mutex_enter(&chptr->ch_adata_lock);
-
-	adata->adata_next = chptr->ch_adata;
-
-	chptr->ch_adata = adata;
-
-	chptr->ch_adata_cnt++;
-
-	ASSERT(chptr->ch_adata_cnt >= 1);
-
-	mutex_exit(&chptr->ch_adata_lock);
-
-	ATRACE("audio_sup_putback_audio_data() done", chptr);
-
-}	/* audio_sup_putback_audio_data() */
-
-/*
- * audio_sup_save_audio_data()
- *
- * Description:
- *	Save audio data on the channel's data list. New data is placed at
- *	the end of the list.
- *
- *	CAUTION: This routine may be called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *	void		*adata_orig	Pointer to the original data to save
- *	size_t		adata_osize	Size of the original data
- *	void		*adata_proc	Pointer to the processed data to save
- *	size_t		adata_psize	Size of the processed data
- *
- * Returns:
- *	AUDIO_SUCCESS		The message was successfully saved
- *	AUDIO_FAILURE		The message was not successfully saved
- */
-int
-audio_sup_save_audio_data(audio_ch_t *chptr, void *adata_orig,
-    size_t adata_osize, void *adata_proc, size_t adata_psize)
-{
-	audio_data_t	*new;
-
-	ATRACE("in audio_sup_save_audio_data()", chptr);
-
-	/* first we allocate an audio_data_t structure (zeros out next field) */
-	if ((new = kmem_zalloc(sizeof (*new), KM_NOSLEEP)) == NULL) {
-		ATRACE("audio_sup_save_audio_data() kmem_zalloc() failed", 0);
-
-		return (AUDIO_FAILURE);
-
-	}
-
-	if (adata_orig) {
-		new->adata_orig = adata_orig;	/* orig data from app */
-		new->adata_optr = adata_orig;
-		new->adata_oeptr = (char *)adata_orig + adata_osize;
-		new->adata_osize = adata_osize;
-	}
-	if (adata_proc) {
-		new->adata_proc = adata_proc;	/* the processed data */
-		new->adata_pptr = adata_proc;
-		new->adata_peptr = (char *)adata_proc + adata_psize;
-		new->adata_psize = adata_psize;
-	}
-
-	/* now we save the message */
-	mutex_enter(&chptr->ch_adata_lock);
-
-	/* see if this is the first message */
-	if (chptr->ch_adata == NULL) {		/* it is */
-		ASSERT(chptr->ch_adata_cnt == 0);
-		chptr->ch_adata = new;		/* next is already set to 0 */
-		chptr->ch_adata_cnt = 1;
-		chptr->ch_adata_end = new;
-		mutex_exit(&chptr->ch_adata_lock);
-		ATRACE("audio_sup_save_audio_data() first message", new);
-
-		return (AUDIO_SUCCESS);
-
-	}
-
-	ATRACE("audio_sup_save_audio_data() saving message", new);
-
-	/* append new message to list */
-	chptr->ch_adata_end->adata_next = new;
-	chptr->ch_adata_end = new;
-
-	chptr->ch_adata_cnt++;
-
-	ASSERT(chptr->ch_adata_cnt >= 1);
-
-	mutex_exit(&chptr->ch_adata_lock);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_save_audio_data() */
-
-/*
- * Minor <--> Channel Routines:
- *
- * audio_sup_ch_to_minor()
- *
- * Description:
- *	Return the minor number of the channel.
- *
- * Arguments:
- *	audio_state_t	*statep		The device state structure
- *	int		channel		The device channel
- *
- * Returns:
- *	>= 0			The minor number of the channel
- */
-int
-audio_sup_ch_to_minor(audio_state_t *statep, int channel)
-{
-	ATRACE("in audio_sup_ch_to_minor(): statep", statep);
-	ATRACE_32("audio_sup_ch_to_minor(): channel #", channel);
-
-	ATRACE_32("audio_sup_ch_to_minor() returning",
-	    ((statep->as_dev_instance * statep->as_minors_per_inst) +
-	    channel + statep->as_audio_reserved));
-
-	return ((statep->as_dev_instance * statep->as_minors_per_inst) +
-	    channel + statep->as_audio_reserved);
-
-}	/* audio_sup_ch_to_minor() */
-
-/*
- * audio_sup_get_max_chs()
- *
- * Description:
- *	Get the maximum number of supported channels per instance.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *
- * Returns:
- *	> 0			The number of minor numbers per instance
- */
-int
-audio_sup_get_max_chs(audiohdl_t handle)
-{
-	audio_state_t	*statep = AUDIO_HDL2STATE(handle);
-
-	ATRACE_32("in audio_sup_get_max_chs() returning",
-	    statep->as_max_chs);
-
-	return (statep->as_max_chs);
-
-}	/* audio_sup_get_max_chs() */
-
-/*
- * audio_sup_get_minors_per_inst()
- *
- * Description:
- *	Get the number of minor numbers allowed per instance.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *
- * Returns:
- *	> 0			The number of minor numbers per instance
- */
-int
-audio_sup_get_minors_per_inst(audiohdl_t handle)
-{
-	audio_state_t	*statep = AUDIO_HDL2STATE(handle);
-
-	ATRACE_32("in audio_sup_get_minors_per_inst() returning",
-	    statep->as_minors_per_inst);
-
-	return (statep->as_minors_per_inst);
-
-}	/* audio_sup_get_minors_per_inst() */
-
-/*
- * audio_sup_construct_minor()
- *
- * Description:
- *	construct a minor number for this dip and device type
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	device_type_e	device_type	type of audio device the channel is
- *					associated with
- *
- * Returns:
- *	>= 0				minor node number
- */
-int
-audio_sup_construct_minor(audiohdl_t handle, audio_device_type_e device_type)
-{
-	audio_state_t	*statep = AUDIO_HDL2STATE(handle);
-	dev_info_t	*dip = statep->as_dip;
-	int		minors_per_inst = statep->as_minors_per_inst;
-	int		minor;
-
-	ATRACE_32("in audio_sup_construct_minor()", device_type);
-
-	minor = (ddi_get_instance(dip) * minors_per_inst) +
-	    audio_sup_type_to_minor(device_type);
-
-	ATRACE_32("audio_sup_construct_minor() returning", minor);
-
-	return (minor);
-
-}	/* audio_sup_construct_minor() */
-
-/*
- * audio_sup_minor_to_ch()
- *
- * Description:
- *	Convert a minor number to a channel number.
- *
- * Arguments:
- *	audio_state_t	*statep		The device state structure
- *	minor_t		minor		The minor number to convert
- *
- * Returns:
- *	>= 0			The channel number
- */
-int
-audio_sup_minor_to_ch(audio_state_t *statep, minor_t minor)
-{
-	int	minors_per_inst = statep->as_minors_per_inst;
-	int	audio_reserved = statep->as_audio_reserved;
-
-	ATRACE_32("in audio_sup_minor_to_ch()", minor);
-	ATRACE_32("audio_sup_minor_to_ch() returning",
-	    ((minor % minors_per_inst) - audio_reserved));
-
-	return ((minor % minors_per_inst) - audio_reserved);
-
-}	/* audio_sup_minor_to_ch() */
-
-/*
- * audio_sup_type_to_minor()
- *
- * Description:
- *	Normally a macro would be used to figure out the minor number. But
- *	we don't want the Audio Driver using the Audio Support Module's
- *	macros which might change. So we provide a call that will let us
- *	change what we are doing later on if we wish.
- *
- * Arguments:
- *	audio_device_type_e	type	The device type we want the minor # of
- *
- * Returns:
- *	The device type
- *	AUDIO_FAILURE		Unrecognized audio device
- */
-int
-audio_sup_type_to_minor(audio_device_type_e type)
-{
-	int		minor;
-
-	ATRACE_32("in audio_sup_type_to_minor()", type);
-
-	switch (type) {
-	case AUDIO:
-		minor = AUDIO_MINOR_AUDIO;
-		break;
-	case AUDIOCTL:
-		minor = AUDIO_MINOR_AUDIOCTL;
-		break;
-	case WTABLE:
-		minor = AUDIO_MINOR_WAVE_TABLE;
-		break;
-	case MIDI:
-		minor = AUDIO_MINOR_MIDI_PORT;
-		break;
-	case ATIME:
-		minor = AUDIO_MINOR_TIME;
-		break;
-	case USER1:
-		minor = AUDIO_MINOR_USER1;
-		break;
-	case USER2:
-		minor = AUDIO_MINOR_USER2;
-		break;
-	case USER3:
-		minor = AUDIO_MINOR_USER3;
-		break;
-	case UNDEFINED:
-		/*FALLTHROUGH*/
-	default:
-		minor = AUDIO_FAILURE;
-		break;
-	}
-
-	ATRACE_32("audio_sup_type_to_minor() returning minor", minor);
-
-	return (minor);
-
-}	/* audio_sup_type_to_minor() */
-
-/*
- * audio_sup_devt_to_instance()
- *
- * Description:
- *	Convert a dev_t to instance
- *
- * Arguments:
- *	dev_t		dev	The device we are getting the instance for
- *
- * Returns:
- *	>= 0			The instance number
- */
-int
-audio_sup_devt_to_instance(dev_t devt)
-{
-	return (AUDIO_MINOR_TO_INST(devt));
-
-}	/* audio_sup_devt_to_instance() */
-
-/*
- * Miscellaneous Routines
- */
-
-/*
- * audio_sup_devt_to_ch_type()
- *
- * Description:
- *	Given a channel's minor number figure out what kind of channel
- *	it is. This works for both the reserved minor nodes as well as
- *	the clone channels.
- *
- * Arguments:
- *	audio_state_t	*statep		The device state structure
- *	dev_t		dev		The device we are getting the type of
- *
- * Returns:
- *	The device type
- *	AUDIO_FAILURE		Couldn't get the state structure, so failed
- */
-audio_device_type_e
-audio_sup_devt_to_ch_type(audio_state_t *statep, dev_t dev)
-{
-	audio_device_type_e	type;
-	minor_t			minor;
-	int			minors_per_inst = statep->as_minors_per_inst;
-	int			audio_reserved = statep->as_audio_reserved;
-
-	ATRACE("in audio_sup_devt_to_ch_type()", dev);
-
-	/* figure out the minor number given an instance */
-	minor = getminor(dev) % minors_per_inst;
-
-	if (minor < audio_reserved) {
-		ATRACE_32("audio_sup_devt_to_ch_type() reserved minor",
-		    minor);
-
-		switch (minor) {
-		case AUDIO_MINOR_AUDIO:
-			type = AUDIO;
-			break;
-		case AUDIO_MINOR_AUDIOCTL:
-			type = AUDIOCTL;
-			break;
-		case AUDIO_MINOR_WAVE_TABLE:
-			type = WTABLE;
-			break;
-		case AUDIO_MINOR_MIDI_PORT:
-			type = MIDI;
-			break;
-		case AUDIO_MINOR_TIME:
-			type = TIME;
-			break;
-		case AUDIO_MINOR_USER1:
-			type = USER1;
-			break;
-		case AUDIO_MINOR_USER2:
-			type = USER2;
-			break;
-		case AUDIO_MINOR_USER3:
-			type = USER3;
-			break;
-		default:
-			type = UNDEFINED;
-			break;
-		}
-
-		ATRACE_32("audio_sup_devt_to_ch_type() reserved, returning",
-		    type);
-
-		return (type);
-
-	} else {
-		audio_state_t	*statep;
-		audio_ch_t	*chptr;
-
-		ATRACE_32("audio_sup_devt_to_ch_type() allocated channel",
-		    minor);
-
-		if ((statep = audio_sup_devt_to_state(dev)) == NULL) {
-			ATRACE("audio_sup_devt_to_ch_type() "
-			    "audio_sup_devt_to_ch_type() failed", 0);
-
-			return ((audio_device_type_e)AUDIO_FAILURE);
-
-		}
-
-		ATRACE("audio_sup_devt_to_ch_type() statep", statep);
-
-		chptr = &statep->as_channels[audio_sup_minor_to_ch(
-		    statep, minor)];
-
-		ATRACE("audio_sup_devt_to_ch_type() chptr", chptr);
-
-		ATRACE_32("audio_sup_devt_to_ch_type() returning type",
-		    chptr->ch_info.dev_type);
-
-		return (chptr->ch_info.dev_type);
-
-	}
-
-}	/* audio_sup_devt_to_ch_type() */
-
-/*
- * audio_sup_get_channel_number()
- *
- * Description:
- *	Get the channel number for the audio queue.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue structure
- *
- * Returns:
- *	channel number		The channel number for the audio queue.
- *	AUDIO_FAILURE		Bad q_ptr
- */
-int
-audio_sup_get_channel_number(queue_t *q)
-{
-	audio_ch_t		*chptr;
-
-	ATRACE("in audio_sup_get_channel_number()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_get_channel_number() bad chptr", 0);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("audio_sup_get_channel_number() returning",
-	    chptr->ch_info.ch_number);
-
-	return (chptr->ch_info.ch_number);
-
-}	/* audio_sup_get_channel_number() */
-
-/*
- * audio_sup_get_apm_info()
- *
- * Description:
- *	Get the audio_apm_info structure for the audio instance and
- *	type passed in.
- *
- *	NOTE: Since the apm_info list is created when the driver is
- *		attached it should never change during normal operation
- *		of the audio device. Therefore we don't need to lock
- *		the list while we traverse it.
- *
- * Arguments:
- *	audio_state_t		*statep		device state structure
- *	audio_device_type_e	type		APM type
- *
- * Returns:
- *	valid pointer		Ptr to the returned audio_apm_info struct
- *	NULL			audio_apm_info struct not found
- */
-audio_apm_info_t *
-audio_sup_get_apm_info(audio_state_t *statep, audio_device_type_e type)
-{
-	audio_apm_info_t	*apm_infop;
-
-	ATRACE("in audio_sup_get_apm_info()", statep);
-
-	/* sanity check */
-	if (type == UNDEFINED) {
-		ATRACE("audio_sup_get_apm_info() returning NULL (fail)", 0);
-
-		return (NULL);
-
-	}
-
-	mutex_enter(&statep->as_lock);
-	apm_infop = statep->as_apm_info_list;
-
-	while (apm_infop != NULL) {
-		if (apm_infop->apm_type == type) {
-			ATRACE_32("audio_sup_get_apm_info() found type", type);
-			break;
-		}
-		apm_infop = apm_infop->apm_next;
-	}
-
-	mutex_exit(&statep->as_lock);
-
-	/* make sure we got a structure */
-	if (apm_infop == NULL) {
-		ATRACE_32("audio_sup_get_apm_info() didn't find type", type);
-
-		return (NULL);
-
-	}
-
-	ATRACE("audio_sup_get_apm_info() returning", apm_infop);
-
-	return (apm_infop);
-
-}	/* audio_sup_get_apm_info() */
-
-/*
- * audio_sup_get_dip()
- *
- * Description:
- *	Get the dev_info_t pointer for the audio handle.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *
- * Returns:
- *	dev_info_t *			The dip for the handle, always returned
- */
-dev_info_t *
-audio_sup_get_dip(audiohdl_t handle)
-{
-
-	return (AUDIO_HDL2STATE(handle)->as_dip);
-
-}	/* audio_sup_get_dip() */
-
-/*
- * audio_sup_get_info()
- *
- * Description:
- *	Get the info structure for the audio queue.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue structure
- *
- * Returns:
- *	valid pointer		Ptr to the returned audio_apm_info struct
- */
-void *
-audio_sup_get_info(queue_t *q)
-{
-	audio_ch_t		*chptr;
-
-	ATRACE("in audio_sup_get_info()", q);
-
-	if ((chptr = (audio_ch_t *)audio_sup_get_qptr_data(q)) == NULL) {
-		ATRACE("audio_sup_close() bad chptr", 0);
-		return (NULL);
-	}
-
-	ATRACE("audio_sup_get_info() returning", chptr->ch_info.info);
-
-	return (chptr->ch_info.info);
-
-}	/* audio_sup_get_info() */
-
-
-/*
- * audio_sup_mblk_alloc()
- *
- * Description:
- *	Allocate a STREAMS message block if the current block isn't there or
- *	is too small. This is placed into the continuation pointer for the
- *	passed in message block.
- *
- *	When we return the b_wptr is set to b_rptr + size;
- *
- * Arguments:
- *	mblk_t		*mp		STREAMS message block to add to
- *	size_t		size		Size of the message to allocate
- *
- * Returns:
- *	AUDIO_SUCCESS			Message block allocated
- *	AUDIO_FAILURE			Message block not allocated
- */
-int
-audio_sup_mblk_alloc(mblk_t *mp, size_t size)
-{
-	/* first the easy case, the buffer is already big enough */
-	if (msgdsize(mp->b_cont) >= size) {
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr + size;
-		return (AUDIO_SUCCESS);
-	}
-
-	/* the old message either isn't there or it's too small */
-	if (mp->b_cont) {
-		/* here, but too small */
-		freemsg(mp->b_cont);
-	}
-
-	/* no memory leak, time to allocate the new message */
-	mp->b_cont = allocb((size), BPRI_HI);
-	if (mp->b_cont == NULL) {
-		return (AUDIO_FAILURE);
-	}
-
-	mp->b_cont->b_wptr = mp->b_cont->b_rptr + size;
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_mblk_alloc() */
-
-/*
- * audio_sup_get_private()
- *
- * Description:
- *	Return the pointer to the audio driver's private state information.
- *
- * Arguments:
- *	audiohdl_t		handle		Handle to the device
- *
- * Returns:
- *	Pointer to private data, even if it's set to NULL
- */
-void *
-audio_sup_get_private(audiohdl_t handle)
-{
-
-	return (AUDIO_HDL2STATE(handle)->as_private);
-
-}	/* audio_sup_get_private() */
-
-/*
- * audio_sup_set_private()
- *
- * Description:
- *	Set the audio driver's private state in the audio handle. It may
- *	be reset at any time.
- *
- * Arguments:
- *	audiohdl_t		handle		Handle to the device
- *	void			*private	Audio driver's private data
- *
- * Returns:
- *	void
- */
-void
-audio_sup_set_private(audiohdl_t handle, void *private)
-{
-	AUDIO_HDL2STATE(handle)->as_private = private;
-
-}	/* audio_sup_set_private() */
-
-/*
- * audio_sup_log()
- *
- * Description:
- *	similar to cmn_err but prefixes the message with
- *	<drivername><instance>:
- *
- * Arguments:
- *	audiohdl_t	handle		May be NULL in which case it behaves as
- *					cmn_err()
- *	uint_t		level		A constant indicating the severity
- *	char		*format		The message to be displayed
- *
- * NOTE: ? ! ^ should work as expected
- */
-void
-audio_sup_log(audiohdl_t handle, uint_t level, char *fmt, ...)
-{
-	va_list		ap;
-	audio_state_t	*statep = AUDIO_HDL2STATE(handle);
-	int		n = 0;
-	int		skip = 0;
-
-	mutex_enter(&audio_sup_log_lock);
-
-	if (statep) {
-		dev_info_t 	*dip = statep->as_dip;
-
-		switch (fmt[0]) {
-		case '?':
-		case '!':
-		case '^':
-			audio_sup_log_buf[0] = fmt[n++];
-			skip++;
-			break;
-		default:
-			break;
-		}
-
-		(void) sprintf(&audio_sup_log_buf[n], "%s%d: ",
-		    ddi_driver_name(dip), ddi_get_instance(dip));
-
-		n = strlen(audio_sup_log_buf);
-	}
-
-	va_start(ap, fmt);
-	(void) vsprintf(&audio_sup_log_buf[n], &fmt[skip], ap);
-	va_end(ap);
-	cmn_err(level, audio_sup_log_buf);
-
-	ATRACE(audio_sup_log_buf, 0);
-
-	mutex_exit(&audio_sup_log_lock);
-
-}	/* audio_sup_log() */
-
-/*
- * audio_sup_update_persist_key()
- *
- * Description:
- *	Update the persistent key to a new value. Memory is freed from the old
- *	key and allocated for the new key.
- *
- * Arguments:
- *	dev_info_t	*dip		dev_info_t pointer for the device
- *	char		*key		Pointer to the new key string
- *	int		sleep_okay	Non-zero if okay to sleep on mem. alloc.
- *
- * Returns:
- *	AUDIO_SUCCESS			Key successfully updated to new key
- *	AUDIO_FAILURE			Memory allocation failed, dip not found
- */
-int
-audio_sup_update_persist_key(dev_info_t *dip, char *key, int sleep_okay)
-{
-	audio_inst_persist_t	*tptr;
-	char			*new_key;
-	size_t			key_length;
-	major_t			major = ddi_driver_major(dip);
-	int			instance = ddi_get_instance(dip);
-
-	ATRACE("in audio_sup_update_persist_key()", key);
-
-	/* see audio_sup_persist() for an explaination on keys */
-	if (key == NULL) {
-		ATRACE("audio_sup_update_persist_key() NULL key", key);
-		return (AUDIO_FAILURE);
-	}
-
-	key_length = strlen(audio_key_class);
-	key_length += strlen(key);
-	key_length++;	/* make room for NULL at end of string */
-
-	if (sleep_okay) {
-		new_key = kmem_alloc(key_length, KM_SLEEP);
-	} else {
-		new_key = kmem_alloc(key_length, KM_NOSLEEP);
-		if (new_key == NULL) {
-			ATRACE("audio_sup_update_persist_key() no memory", key);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	(void) sprintf(new_key, "%s%s", audio_key_class, key);
-	ATRACE("audio_sup_update_persist_key() key string", new_key);
-
-	/*
-	 * Find our dip, we do this after getting the key so we hold the
-	 * locks for as short a time as possible.
-	 */
-	mutex_enter(&audio_persist_lock);
-	ASSERT(audio_main_anchor);
-	for (tptr = *audio_main_anchor; tptr; tptr = tptr->amp_next) {
-		if (tptr->amp_major == major &&
-		    tptr->amp_instance == instance) {
-			break;
-		}
-	}
-	if (tptr == NULL) {
-		kmem_free(new_key, key_length);
-		mutex_exit(&audio_persist_lock);
-		ATRACE("audio_sup_update_persist_key() dip not found", dip);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("audio_sup_update_persist_key() dip found", dip);
-	kmem_free(tptr->amp_key, strlen(tptr->amp_key)+1);
-	tptr->amp_key = new_key;
-
-	mutex_exit(&audio_persist_lock);
-
-	ATRACE("audio_sup_update_persist_key() done", new_key);
-
-	return (AUDIO_SUCCESS);
-
-
-}	/* audio_sup_update_persist_key() */
-
-/*
- * STREAMS Private Data Routines
- */
-
-/*
- * audio_sup_set_qptr()
- *
- * Description:
- *	Allocate an audio_qptr structure to hold the private STREAM data
- *	structure. Then set the values and point both the RD() and WR()
- *	STREAMS queues to this data structure. That way it'll be available
- *	regardless of which direction the queue may be pointed.
- *
- * NOTE: open() and close() use the read queue, which is why we put it on
- *	both the read and write side.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	dev_t		dev	Device name
- *	void		*data	Pointer to the private data
- *
- * Returns:
- *	void
- */
-void
-audio_sup_set_qptr(queue_t *q, dev_t dev, void *data)
-{
-	audio_qptr_t		*tmp;
-
-	ATRACE("in audio_sup_set_qptr()", q);
-
-	tmp = (audio_qptr_t *)kmem_alloc(sizeof (audio_qptr_t), KM_SLEEP);
-
-	tmp->aq_dev = dev;
-	tmp->aq_data = data;
-
-	RD(q)->q_ptr = (caddr_t)tmp;
-	WR(q)->q_ptr = (caddr_t)tmp;
-
-	ATRACE("audio_sup_set_qptr() done", tmp);
-
-}	/* audio_sup_set_qptr() */
-
-/*
- * audio_sup_free_qptr()
- *
- * Description:
- *	Free the private STREAMS data structure.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *
- * Returns:
- *	void
- */
-void
-audio_sup_free_qptr(queue_t *q)
-{
-	ATRACE("in audio_sup_free_qptr()", q);
-
-	kmem_free(q->q_ptr, sizeof (audio_qptr_t));
-
-	RD(q)->q_ptr = NULL;
-	WR(q)->q_ptr = NULL;
-
-	ATRACE("audio_sup_free_qptr() done", q);
-
-}	/* audio_sup_free_qptr() */
-
-/*
- * audio_sup_get_qptr_dev()
- *
- * Description:
- *	Get the device info from the private data.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *
- * Returns:
- *	The dev_t that was saved
- */
-dev_t
-audio_sup_get_qptr_dev(queue_t *q)
-{
-	audio_qptr_t		*tmp;
-
-	ATRACE("in audio_sup_get_qptr_dev()", q);
-
-	/* make sure the q_ptr is valid */
-	if (q->q_ptr == NULL) {
-		ATRACE("audio_sup_get_qptr_dev() null q_ptr", NULL);
-		return (NULL);
-	}
-
-	tmp = (audio_qptr_t *)q->q_ptr;
-
-	ATRACE("audio_sup_get_qptr_dev() dev_t", tmp->aq_dev);
-
-	return (tmp->aq_dev);
-
-}	/* audio_sup_get_qptr_dev() */
-
-/*
- * audio_sup_get_qptr_data()
- *
- * Description:
- *	Get the data info from the private data.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *
- * Returns:
- *	Pointer to the private data.
- */
-void *
-audio_sup_get_qptr_data(queue_t *q)
-{
-	audio_qptr_t		*tmp;
-
-	ATRACE("in audio_sup_get_qptr_data()", q);
-
-	/* make sure the q_ptr is valid */
-	if (q->q_ptr == NULL) {
-		ATRACE("audio_sup_get_qptr_data() null q_ptr", NULL);
-		return (NULL);
-	}
-
-	tmp = (audio_qptr_t *)q->q_ptr;
-
-	ATRACE("audio_sup_get_qptr_data() done, data", tmp->aq_data);
-
-	return (tmp->aq_data);
-
-}	/* audio_sup_get_qptr_data() */
-
-/*
- * audio_sup_get_qptr_instance()
- *
- * Description:
- *	Get the instance number from the private data's dev_t.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *
- * Returns:
- *	>= 0			The instance number
- */
-int
-audio_sup_get_qptr_instance(queue_t *q)
-{
-	dev_t		dev = audio_sup_get_qptr_dev(q);
-	audio_state_t	*statep = audio_sup_devt_to_state(dev);
-	int		minor = getminor(dev);
-	int		instance;
-
-	instance = minor / audio_sup_get_minors_per_inst(
-	    AUDIO_STATE2HDL(statep));
-
-	ATRACE("in audio_sup_get_qptr_instance()", q);
-	ATRACE_32("in audio_sup_get_qptr_instance() returning", instance);
-
-	return (instance);
-
-}	/* audio_sup_get_qptr_instance() */
-
-/*
- * State structure handling:
- */
-
-/*
- * audio_sup_devt_to_state()
- *
- * Description:
- *	This routine is used to associate the device number with the
- *	dev_info_t pointer. It returns the device's state structure when
- *	it is done.
- *
- * Arguments:
- *	dev_t		dev	Device name
- *
- * Returns:
- *	Valid pointer			Pointer to the state
- *	NULL				State pointer not found
- */
-audio_state_t *
-audio_sup_devt_to_state(dev_t dev)
-{
-	audio_inst_info_t	*list;	/* for walking the list */
-	audio_inst_info_t	**plist	/* for walking the list */;
-	dev_info_t		*my_dip;
-	audio_state_t		*statep;
-	int			instance;
-	major_t			major = getmajor(dev);
-	major_t			my_major;
-
-	ATRACE("in audio_sup_devt_to_state()", dev);
-
-	/* protect the driver list */
-	mutex_enter(&audio_drv_list_lock);
-
-	plist = &audio_drv_list_head;
-	list = *plist;
-	while (list) {
-		statep = &list->ail_state;
-		mutex_enter(&statep->as_lock);
-		my_dip = statep->as_dip;
-		my_major = statep->as_major;
-		instance = getminor(dev) /
-		    audio_sup_get_minors_per_inst(AUDIO_STATE2HDL(statep));
-
-		if ((my_major == major) &&
-		    (instance == ddi_get_instance(my_dip))) {
-			mutex_exit(&statep->as_lock);
-			mutex_exit(&audio_drv_list_lock);
-
-			ATRACE("audio_sup_devt_to_state() returning", statep);
-
-			return (statep);
-
-		}
-		plist = &list->ail_next;
-		list = list->ail_next;
-		mutex_exit(&statep->as_lock);
-	}
-	mutex_exit(&audio_drv_list_lock);
-
-	ATRACE("audio_sup_devt_to_state() returning NULL", 0);
-
-	return (NULL);
-
-}	/* audio_sup_devt_to_state() */
-
-/*
- * audio_sup_devinfo_to_state()
- *
- * Description:
- *	Get the state pointer for the audio device given a devinfo_t *.
- *
- * Arguments:
- *	dev_info_t	*dip	dev_info_t pointer for the device
- *
- * Returns:
- *	Valid pointer			Pointer to the state
- *	NULL				State pointer not found
- */
-audio_state_t *
-audio_sup_devinfo_to_state(dev_info_t *dip)
-{
-	audio_inst_info_t	*instp;
-
-	ATRACE("in audio_sup_devinfo_to_state(), dip", dip);
-
-	mutex_enter(&audio_drv_list_lock);
-	instp = audio_sup_lookup_drv_entry(dip);
-
-	if (instp && (instp->ail_state.as_dip == dip)) {
-		mutex_exit(&audio_drv_list_lock);
-
-		/* verify, if given */
-		ATRACE("audio_sup_devinfo_to_state() "
-		    "found dip match, returning", &instp->ail_state);
-
-		return (&instp->ail_state);
-
-	} else {
-		mutex_exit(&audio_drv_list_lock);
-		ATRACE("audio_sup_devinfo_to_state() no dip match", dip);
-
-		return (NULL);
-
-	}
-
-}	/* audio_sup_devinfo_to_state() */
-
-/*
- * audio_sup_create_drv_entry()
- *
- * Description:
- *	Create & add an entry in the instance list (audio_drv_list_head).
- *	The whole linked list is scanned
- *	to make sure we don't ever get a duplicate entry.
- *
- * Arguments:
- *	dev_info_t	*dip	dev_info_t pointer for the device, what we use
- *				to find the instance
- *
- * Returns:
- *	Valid pointer		Valid instance
- *	NULL			Instance already registered
- */
-static audio_inst_info_t *
-audio_sup_create_drv_entry(dev_info_t *dip)
-{
-	audio_inst_info_t	*list;		/* for walking the list */
-	audio_inst_info_t	**plist;	/* for walking the list */
-	audio_inst_info_t	*entry;		/* new entry */
-
-	ATRACE("in audio_sup_create_drv_entry()", dip);
-
-	/* protect the driver list */
-	mutex_enter(&audio_drv_list_lock);
-	plist = &audio_drv_list_head;
-	list = *plist;
-	while (list) {
-		if (list->ail_state.as_dip == dip) {
-			mutex_exit(&audio_drv_list_lock);
-
-			ATRACE("audio_sup_create_drv_entry() "
-			    "instance already registered", dip);
-			audio_sup_log(NULL, CE_NOTE,
-			    "%s%d already registered",
-			    ddi_driver_name(dip), ddi_get_instance(dip));
-
-			return (NULL);
-
-		}
-		plist = &list->ail_next;
-		list = list->ail_next;
-	}
-
-	/* "dip" is not registered, create new one and add to list */
-	ATRACE("audio_sup_create_drv_entry() dip not registered", dip);
-
-	entry = kmem_zalloc(sizeof (*entry), KM_SLEEP);
-	*plist = entry;
-	mutex_exit(&audio_drv_list_lock);
-
-	return (entry);
-
-}	/* audio_sup_create_drv_entry() */
-
-/*
- * audio_sup_free_drv_entry()
- *
- * Description:
- *	This routine is used to clear entries in the driver list that
- *	were added in by audio_sup_create_drv_entry().
- *
- * Arguments:
- *	dev_info_t	*dip		dev_info_t pointer for the device.
- *					if NULL, delete all entries
- *
- * Returns:
- *	void
- */
-static void
-audio_sup_free_drv_entry(dev_info_t *dip)
-{
-	audio_inst_info_t	*list;	/* for walking the list */
-	audio_inst_info_t	**plist	/* for walking the list */;
-	audio_ch_t		*chptr;
-	audio_state_t		*statep;
-	int			i;
-	int			num_chs;
-
-	ATRACE("in audio_sup_free_drv_entry()", dip);
-
-	/* protect the driver list */
-	mutex_enter(&audio_drv_list_lock);
-	plist = &audio_drv_list_head;
-	list = *plist;
-	while (list) {
-		if ((dip == NULL) || (list->ail_state.as_dip == dip)) {
-			/* found it */
-			*plist = list->ail_next;
-			list->ail_next = NULL;
-
-			statep = &list->ail_state;
-
-			/* de-initialize the channel structures */
-			num_chs = statep->as_max_chs;
-			for (i = 0, chptr = &statep->as_channels[0];
-			    i < num_chs; i++, chptr++) {
-				/*
-				 * All we have to worry about is locks and
-				 * condition variables.
-				 */
-				mutex_destroy(&chptr->ch_lock);
-				mutex_destroy(&chptr->ch_adata_lock);
-				cv_destroy(&chptr->ch_cv);
-			}
-
-			/* destroy mutex, cv */
-			mutex_destroy(&statep->as_lock);
-			cv_destroy(&statep->as_cv);
-
-			/* free the structure */
-			kmem_free(list, sizeof (audio_inst_info_t));
-
-			/* see if we need to go again */
-			if (dip) {
-				/* nope, done */
-				break;
-			}
-		}
-		plist = &list->ail_next;
-		list = list->ail_next;
-	}
-	mutex_exit(&audio_drv_list_lock);
-
-	ATRACE("audio_sup_free_drv_entry() done", 0);
-
-}	/* audio_sup_free_drv_entry() */
-
-/*
- * audio_sup_free_apm_persist()
- *
- * Description:
- *	Free the audio_apm_persist list that is associated with the instance.
- *
- * Arguments:
- *	audio_state_t		*statep		Pointer to device state info
- *	audio_apm_persist_t	*instp		Pointer to instance data
- *
- * Returns:
- *	void					Silently fails if key useless
- */
-static void
-audio_sup_free_apm_persist(audio_state_t *statep, audio_apm_persist_t *instp)
-{
-	audio_apm_persist_t	*tmp;
-
-	ATRACE("in audio_sup_free_apm_persist() statep", statep);
-	ATRACE("audio_sup_free_apm_persist() instp", instp);
-
-	ASSERT(mutex_owned(&audio_persist_lock));
-
-	tmp = instp;
-	while (tmp) {
-		tmp = tmp->ap_next;
-		kmem_free(instp->ap_data, instp->ap_size);
-		kmem_free(instp, sizeof (*instp));
-	}
-
-	ATRACE("audio_sup_free_apm_persist() done", statep);
-
-}	/* audio_sup_free_apm_persist() */
-
-/*
- * audio_sup_free_inst_persist()
- *
- * Description:
- *	Free the audio_inst_persist_t structure for this dip.
- *
- * Arguments:
- *	audio_state_t		*statep		Pointer to device state info
- *	audio_inst_persist_t	*persistp	Pointer to the struct to remove
- *
- * Returns:
- *	void
- */
-static void
-audio_sup_free_inst_persist(audio_state_t *statep,
-	audio_inst_persist_t *persistp)
-{
-	audio_inst_persist_t	**plist;
-	audio_inst_persist_t	*list;
-
-	ATRACE("in audio_sup_free_inst_persist() statep", statep);
-	ATRACE("audio_sup_free_inst_persist() persistp", persistp);
-
-	ASSERT(mutex_owned(&audio_persist_lock));
-	ASSERT(audio_main_anchor);
-
-	plist = audio_main_anchor;
-	list = *plist;
-
-	while (list) {
-		if (list == persistp) {
-			/* found it, free the APM list and key */
-			if (list->amp_key) {
-				kmem_free(list->amp_key,
-				    strlen(list->amp_key)+1);
-			}
-			if (list->amp_apmp) {
-				audio_sup_free_apm_persist(statep,
-				    list->amp_apmp);
-			}
-			/* remove from the list */
-			*plist = list->amp_next;
-			ATRACE("audio_sup_free_inst_persist() "
-			    "freeing dip & return", list);
-			kmem_free(list, sizeof (*list));
-			return;
-		}
-
-		plist = &list->amp_next;
-		list = list->amp_next;
-	}
-
-	ATRACE("audio_sup_free_inst_persist() returning, not found", persistp);
-
-}	/* audio_sup_free_inst_persist() */
-
-/*
- * audio_sup_lookup_drv_entry()
- *
- * Description:
- *	Lookup audio_inst_info_t pointer corresponding to dip
- *
- * Arguments:
- *	dev_info_t	*dip		dev_info_t pointer for the device
- *
- * Returns:
- *	audio_inst_info_t pointer or NULL
- */
-static audio_inst_info_t *
-audio_sup_lookup_drv_entry(dev_info_t *dip)
-{
-	audio_inst_info_t	*list;	/* for walking the list */
-	audio_inst_info_t	**plist	/* for walking the list */;
-
-	ATRACE("in audio_sup_lookup_drv_entry()", dip);
-	ASSERT(mutex_owned(&audio_drv_list_lock));
-
-	plist = &audio_drv_list_head;
-	list = *plist;
-	while (list) {
-		if (list->ail_state.as_dip == dip) {
-			break;
-		}
-		plist = &list->ail_next;
-		list = list->ail_next;
-	}
-
-	ATRACE("audio_sup_lookup_drv_entry() done", list);
-
-	return (list);
-
-}	/* audio_sup_lookup_drv_entry() */
-
-/*
- * audio_sup_persist()
- *
- * Description:
- *	Routine to initialize and find persistent state and its unique key.
- *
- *	There are two types of persistent state, per instance and per APM
- *	state. Per instance saves the key and dip. Thus the correct instance
- *	may be found based on the key or the dip. It is also possible to
- *	detect, given a driver provided unique key, when a device moves
- *	from one port to another. Right now a linear search is used to find
- *	the matching dip. If we get to the point of supporting 100s or 1000s
- *	of instances then this will need to be changed to something more
- *	sophisticated.
- *
- *	The second type is used to save the state for each APM for the
- *	instance. It is not a requirement for any APM to save persistent
- *	state, thus we could be allocating the per instance state structure
- *	for nothing. The APM is responsible for allocating the memory and
- *	providing a pointer to that memory and the size.
- *	audio_sup_free_persist_state() will free this memory for the APM.
- *
- *	The per instance state structures are found by retreiving the
- *	main anchor using the AUDIO_SUP_KEY unique key.
- *
- *	If the reset-configuration property is set to non-zero then any
- *	saved state is cleared and the APMs rebuild the persistent data
- *	from scratch. This effectively resets them to their defaults.
- *
- *	NOTE: All allocated memory is owned by the main anchor.
- *
- * Arguments:
- *	audio_state_t		*statep	The device state structure
- *	char			*key	Unique key to for finding memory
- *
- * Returns:
- *	AUDIO_SUCCESS			Persistent state allocated/found
- *	AUDIO_FAILURE			Couldn't allocate/find persistent state
- */
-static int
-audio_sup_persist(audio_state_t *statep, char *key)
-{
-	audio_inst_persist_t	*key_persistp = NULL;
-	audio_inst_persist_t	*inst_persistp;
-	audio_inst_persist_t	*main_anchor;
-	audio_inst_persist_t	*tmp_persistp;
-	dev_info_t		*dip = statep->as_dip;
-	size_t			key_length;
-	major_t			major = ddi_driver_major(dip);
-	char			*anchor_key = AUDIO_SUP_KEY;
-	char			*new_key;
-	char			pathbuf[MAXPATHLEN];
-	int			do_reset = 0;
-	int			instance = statep->as_dev_instance;
-
-	ATRACE("in audio_sup_persist()", statep);
-
-	/*
-	 * The 1st step is to make sure we have the anchor. It could have
-	 * been allocated on a previous audio framework load, or it could be
-	 * unallocated.
-	 */
-	mutex_enter(&audio_persist_lock);
-	ATRACE("audio_sup_persist() audio_main_anchor", audio_main_anchor);
-	if (audio_main_anchor == NULL) {
-		/* anchor not loaded, so find it */
-		main_anchor = (audio_inst_persist_t *)space_fetch(anchor_key);
-		ATRACE("audio_sup_persist() space_fetch() main_anchor",
-		    main_anchor);
-		if (main_anchor == NULL) {
-			/* 1st load, need to allocate the anchor memory */
-			main_anchor = kmem_zalloc(sizeof (main_anchor),
-			    KM_SLEEP);
-			if (space_store(anchor_key,
-			    (uintptr_t)main_anchor) != 0) {
-				mutex_exit(&audio_persist_lock);
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "!cannot create persistent anchor");
-				kmem_free(main_anchor, sizeof (main_anchor));
-				return (AUDIO_FAILURE);
-			}
-			ATRACE("audio_sup_persist() space_store() main_anchor",
-			    main_anchor);
-		}
-		/* set this after the check above! */
-		audio_main_anchor = (audio_inst_persist_t **)main_anchor;
-		ATRACE("audio_sup_persist() new audio_main_anchor",
-		    audio_main_anchor);
-	}
-	mutex_exit(&audio_persist_lock);
-
-	/*
-	 * Set up the instance key. Following the comments in space.c this
-	 * key is in one of two formats:
-	 *
-	 *	AUDIO:key_string_from_driver
-	 *
-	 * or
-	 *
-	 *	AUDIO:pathname
-	 *
-	 * The former is when the driver passes a string for the key. This
-	 * string MUST be unique. The later is when the driver passes a
-	 * NULL key string.
-	 */
-	if (key == NULL) {
-		key = ddi_pathname(dip, pathbuf);
-		ATRACE("audio_sup_persist() NULL key string", key);
-	} else {
-		ATRACE("audio_sup_persist() provided key string", key);
-	}
-
-	key_length = strlen(audio_key_class);
-	key_length += strlen(key);
-	key_length++;	/* make room for NULL at end of string */
-
-	new_key = kmem_alloc(key_length, KM_SLEEP);
-	(void) sprintf(new_key, "%s%s", audio_key_class, key);
-	ATRACE("audio_sup_register() key string", new_key);
-
-	/*
-	 * In order to determine if this device has moved to a different port
-	 * or if the key has changed we need to search through all of the dev
-	 * structures. Once this loop ends we'll know if the current key and
-	 * dev are present. This is one of the simple search algorithms that
-	 * would need to be changed if we need to support a huge number of
-	 * instances.
-	 */
-	mutex_enter(&audio_persist_lock);
-	for (tmp_persistp = *audio_main_anchor; tmp_persistp;
-	    tmp_persistp = tmp_persistp->amp_next) {
-		if (strcmp(new_key, tmp_persistp->amp_key) == 0) {
-			ATRACE("audio_sup_persist() dev: found key",
-			    tmp_persistp);
-			ASSERT(!key_persistp);
-			key_persistp = tmp_persistp;
-			break;
-		}
-	}
-
-	if (key_persistp) {
-		key_persistp->amp_major = major;
-		key_persistp->amp_instance = instance;
-
-		ATRACE("audio_sup_persist() match", key_persistp);
-		inst_persistp = key_persistp;
-
-		/* free new_key so we don't have a memory leak */
-		kmem_free(new_key, key_length);
-	} else {
-		ATRACE("audio_sup_persist() no match", statep);
-		mutex_exit(&audio_persist_lock);
-		inst_persistp = kmem_zalloc(sizeof (*inst_persistp), KM_SLEEP);
-		mutex_enter(&audio_persist_lock);
-
-		/* don't free new_key because we need it */
-		inst_persistp->amp_key = new_key;
-		inst_persistp->amp_major = major;
-		inst_persistp->amp_instance = instance;
-
-		/* put on list */
-		inst_persistp->amp_next = *audio_main_anchor;
-		*audio_main_anchor = inst_persistp;
-	}
-
-	/*
-	 * CAUTION: From here new_key must have been freed or placed in an
-	 *	audio_inst_persist_t structure. Otherwise we'll have a memory
-	 *	leak.
-	 *
-	 * We are done!
-	 */
-
-	ASSERT(inst_persistp);
-	mutex_exit(&audio_persist_lock);
-
-	/* have statep point to instance persist info */
-	mutex_enter(&statep->as_lock);
-	statep->as_persistp = (void *)inst_persistp;
-	ATRACE("audio_sup_register() as_persistp", statep->as_persistp);
-	mutex_exit(&statep->as_lock);
-
-	/*
-	 * The final piece of information we need is whether we need to
-	 * do a reset or not.
-	 */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "reset-configuration", 0)) {
-		ATRACE("audio_sup_persist() need reset", statep);
-		do_reset++;
-	}
-
-	/*
-	 * Now that we have good persistence data see if we need to reset it.
-	 * We don't block the attach() in this case because the driver is still
-	 * useful.
-	 */
-	if (do_reset && audio_sup_free_persist_state(statep,
-	    AUDIO_ALL_DEVICES) == AUDIO_FAILURE) {
-		audio_sup_log(NULL, CE_NOTE, "!state reset failed");
-	}
-
-	ATRACE("audio_sup_persist() returning, tmp_persistp", tmp_persistp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio_sup_persist() */
-
-/*
- * audio_sup_wiocdata()
- *
- * Description:
- *	This routine is called by audio_sup_wput() to process the IOCDATA
- *	messages that belong to the Audio Support Module's routines.
- *
- *	We only support transparent ioctls.
- *
- *	WARNING: Don't forget to free the mblk_t struct used to hold private
- *		data. The done: jump point takes care of this.
- *
- *	WARNING: Don't free the private mblk_t structure if the command is
- *		going to call qreply(). This frees the private date that will
- *		be needed for the next M_IOCDATA message.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *
- * Returns:
- *	0			Always returns a 0, becomes on return for
- *				audio_sup_wput()
- */
-static int
-audio_sup_wiocdata(queue_t *q, mblk_t *mp, audio_ch_t *chptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	struct copyreq		*cqp;
-	struct copyresp		*csp;
-	audio_i_state_t		*cmd;
-	int			error = 0;
-
-	ATRACE("in audio_sup_wiocdata()", chptr);
-
-	ASSERT(statep);
-	ASSERT(chptr);
-
-	csp = (struct copyresp *)mp->b_rptr;	/* set up copy response ptr */
-	cmd = (audio_i_state_t *)csp->cp_private;	/* get state info */
-	cqp =  (struct copyreq *)mp->b_rptr;	/* set up copy request ptr */
-
-	/* make sure we've got a good return value */
-	if (csp->cp_rval) {
-		ATRACE("audio_sup_wiocdata() bad return value", csp->cp_rval);
-		error = EINVAL;
-		goto done;
-	}
-
-	/* find the command */
-	ATRACE_32("audio_sup_wiocdata() command", cmd->ais_command);
-	switch (cmd->ais_command) {
-
-	case AUDIO_COPY_OUT_CH_NUMBER:	/* AUDIO_GET_CH_NUMBER */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_CH_NUMBER", chptr);
-		if (csp->cp_cmd != AUDIO_GET_CH_NUMBER) {
-			error = EINVAL;
-		}
-		break;
-
-	case AUDIO_COPY_OUT_CH_TYPE:		/* AUDIO_GET_CH_TYPE */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_CH_TYPE", chptr);
-		if (csp->cp_cmd != AUDIO_GET_CH_TYPE) {
-			error = EINVAL;
-		}
-		break;
-
-	case AUDIO_COPY_OUT_NUM_CHS:		/* AUDIO_GET_NUM_CHS */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_CHANNELS", chptr);
-		if (csp->cp_cmd != AUDIO_GET_NUM_CHS) {
-			error = EINVAL;
-		}
-		break;
-
-	case AUDIO_COPY_OUT_AD_DEV:		/* AUDIO_GET_AD_DEV */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_AD_DEV", chptr);
-		if (csp->cp_cmd != AUDIO_GET_AD_DEV) {
-			error = EINVAL;
-		}
-		break;
-
-	case AUDIO_COPY_OUT_APM_DEV:		/* AUDIO_GET_APM_DEV */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_APM_DEV", chptr);
-		if (csp->cp_cmd != AUDIO_GET_APM_DEV) {
-			error = EINVAL;
-		}
-		break;
-
-	case AUDIO_COPY_OUT_AS_DEV:		/* AUDIO_GET_AS_DEV */
-		ATRACE("audio_sup_wiocdata() AUDIO_COPY_OUT_AS_DEV", chptr);
-		if (csp->cp_cmd != AUDIO_GET_AS_DEV) {
-			error = EINVAL;
-		}
-		break;
-
-	default:
-		ATRACE("audio_sup_wiocdata() default", chptr);
-		error = EINVAL;
-		break;
-	}
-
-	/* we always either ack or nack depending on if error is set or not */
-	ATRACE_32("audio_sup_wiocdata() switch done", error);
-
-done:
-	if (csp->cp_private) {
-		ATRACE("audio_sup_wiocdata() freeing csp->cp_private",
-		    csp->cp_private);
-		kmem_free(csp->cp_private, sizeof (audio_i_state_t));
-		csp->cp_private = NULL;
-	}
-	if (cqp->cq_private) {
-		ATRACE("audio_sup_wiocdata() freeing cqp->cq_private",
-		    cqp->cq_private);
-		kmem_free(cqp->cq_private, sizeof (audio_i_state_t));
-		cqp->cq_private = NULL;
-	}
-
-	if (error) {
-		miocnak(q, mp, 0, error);
-	} else {
-		miocack(q, mp, 0, 0);
-	}
-
-	ATRACE_32("audio_sup_wiocdata() returning", error);
-
-	return (0);
-
-}	/* audio_sup_wiocdata() */
-
-/*
- * audio_sup_wioctl()
- *
- * Description:
- *	This routine is called by audio_sup_wput() to process all M_IOCTL
- *	messages that the Audio Support Module provides.
- *
- *	We only support transparent ioctls. Since this is a driver we
- *	nack unrecognized ioctls.
- *
- *	The following ioctls are supported:
- *		AUDIO_GET_CH_NUMBER
- *		AUDIO_GET_CH_TYPE
- *		AUDIO_GET_NUM_CHS
- *		AUDIO_GET_AD_DEV
- *		AUDIO_GET_APM_DEV
- *		AUDIO_GET_AS_DEV
- *		unknown		nack back up the queue
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- *	WARNING: There cannot be any locks owned by calling routines.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *
- * Returns:
- *	0			Always returns a 0, becomes a return for
- *				audio_sup_wput()
- */
-static int
-audio_sup_wioctl(queue_t *q, mblk_t *mp, audio_ch_t *chptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_device_t		*devp;
-	struct iocblk		*iocbp;
-	audio_i_state_t		*state = NULL;
-	audio_device_type_e	type = chptr->ch_info.dev_type;
-	int			error;
-
-	ATRACE("in audio_sup_wioctl()", chptr);
-
-	ASSERT(statep);
-	ASSERT(chptr);
-
-	ASSERT(!mutex_owned(&statep->as_lock));
-
-	iocbp = (struct iocblk *)mp->b_rptr;	/* pointer to ioctl info */
-
-	/* make sure this is a transparent ioctl */
-	if (iocbp->ioc_count != TRANSPARENT) {
-		ATRACE_32("audio_sup_wioctl() not TRANSPARENT",
-		    iocbp->ioc_count);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* get a buffer for private data */
-	if ((state = kmem_alloc(sizeof (*state), KM_NOSLEEP)) == NULL) {
-		ATRACE("audio_sup_wioctl() state kmem_alloc() failed", 0);
-		error = ENOMEM;
-		goto nack;
-	}
-
-	ATRACE_32("audio_sup_wioctl() command", iocbp->ioc_cmd);
-	switch (iocbp->ioc_cmd) {
-
-	case AUDIO_GET_CH_NUMBER:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_CH_NUMBER", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_CH_NUMBER;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* only an int, so reuse the data block without checks */
-		*(int *)mp->b_cont->b_rptr = chptr->ch_info.ch_number;
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr + sizeof (int);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (int), state->ais_address,
-		    NULL);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_CH_NUMBER returning",
-		    chptr);
-
-		return (0);
-
-		/* end AUDIO_GET_CH_NUMBER */
-
-	case AUDIO_GET_CH_TYPE:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_CH_TYPE", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_CH_TYPE;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* only an int, so reuse the data block without checks */
-		*(int *)mp->b_cont->b_rptr = type;
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr + sizeof (AUDIO);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (int), state->ais_address,
-		    NULL);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_CH_TYPE returning", chptr);
-
-		return (0);
-
-		/* end AUDIO_GET_CH_TYPE */
-
-	case AUDIO_GET_NUM_CHS:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_NUM_CHS", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_NUM_CHS;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* only an int, so reuse the data block without checks */
-		*(int *)mp->b_cont->b_rptr = statep->as_max_chs;
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr + sizeof (int);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (int), state->ais_address,
-		    NULL);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_NUM_CHS returning",
-		    chptr);
-
-		return (0);
-
-		/* end AUDIO_GET_NUM_CHS */
-
-	case AUDIO_GET_AD_DEV:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_AD_DEV", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_AD_DEV;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (*chptr->ch_dev_info),
-		    state->ais_address, NULL);
-
-		/* put the data in the buffer, but try to reuse it first */
-		if (MBLKSIZE(mp->b_cont) < sizeof (audio_device_t)) {
-			freemsg(mp->b_cont);
-			mp->b_cont = allocb(
-			    sizeof (*chptr->ch_dev_info), BPRI_MED);
-			if (mp->b_cont == NULL) {
-				error = EAGAIN;
-				goto nack;
-			}
-		}
-
-		/*
-		 * We don't bother to lock the state structure because this
-		 * is static data.
-		 */
-
-		devp = (audio_device_t *)mp->b_cont->b_rptr;
-
-		bcopy(chptr->ch_dev_info, devp, sizeof (*chptr->ch_dev_info));
-
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr +
-		    sizeof (*chptr->ch_dev_info);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_AD_DEV returning", chptr);
-
-		return (0);
-
-	case AUDIO_GET_APM_DEV:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_APM_DEV", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_APM_DEV;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (*chptr->ch_dev_info),
-		    state->ais_address, NULL);
-
-		/* put the data in the buffer, but try to reuse it first */
-		if (MBLKSIZE(mp->b_cont) < sizeof (*chptr->ch_dev_info)) {
-			freemsg(mp->b_cont);
-			mp->b_cont = allocb(
-			    sizeof (*chptr->ch_dev_info), BPRI_MED);
-			if (mp->b_cont == NULL) {
-				error = EAGAIN;
-				goto nack;
-			}
-		}
-
-		/*
-		 * We don't bother to lock the state structure because this
-		 * is static data.
-		 */
-
-		devp = (audio_device_t *)mp->b_cont->b_rptr;
-
-		bcopy(chptr->ch_apm_infop->apm_info, devp,
-		    sizeof (*chptr->ch_dev_info));
-
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr +
-		    sizeof (*chptr->ch_dev_info);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_APM_DEV returning", chptr);
-
-		return (0);
-
-	case AUDIO_GET_AS_DEV:
-		ATRACE("audio_sup_wioctl() AUDIO_GET_AS_DEV", chptr);
-
-		/* save state for M_IOCDATA processing */
-		state->ais_command = AUDIO_COPY_OUT_AS_DEV;
-			/* user space addr */
-		state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-		/* Setup for copyout */
-		mcopyout(mp, (mblk_t *)state, sizeof (*chptr->ch_dev_info),
-		    state->ais_address, NULL);
-
-		/* put the data in the buffer, but try to reuse it first */
-		if (MBLKSIZE(mp->b_cont) < sizeof (*chptr->ch_dev_info)) {
-			freemsg(mp->b_cont);
-			mp->b_cont = allocb(
-			    sizeof (*chptr->ch_dev_info), BPRI_MED);
-			if (mp->b_cont == NULL) {
-				error = EAGAIN;
-				goto nack;
-			}
-		}
-
-		/*
-		 * We don't bother to lock the state structure because this
-		 * is static data.
-		 */
-
-		devp = (audio_device_t *)mp->b_cont->b_rptr;
-
-		bcopy(&audio_device_info, devp, sizeof (*chptr->ch_dev_info));
-
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr +
-		    sizeof (*chptr->ch_dev_info);
-
-		/* send the copy out request */
-		qreply(q, mp);
-
-		ATRACE("audio_sup_wioctl() AUDIO_GET_AS_DEV returning", chptr);
-
-		return (0);
-
-	default:	/* this should never happen */
-		ATRACE_32("audio_sup_wioctl() default", iocbp->ioc_cmd);
-		error = EINVAL;
-		break;
-	}
-
-	/* we always nack */
-	ATRACE_32("audio_sup_wioctl() switch done", error);
-
-nack:
-	/* we always nack if we break out of the switch() */
-	ATRACE("audio_sup_wioctl() nack", chptr);
-
-	if (state) {		/* free allocated state memory */
-		ATRACE("audio_sup_wioctl() nack freeing state", state);
-		kmem_free(state, sizeof (audio_i_state_t));
-	}
-
-	miocnak(q, mp, 0, error);
-
-	ATRACE("audio_sup_wioctl() returning failure", chptr);
-
-	return (0);
-
-}	/* audio_sup_wioctl() */
diff --git a/usr/src/uts/common/io/audio/sada/mixer/am_ad.c b/usr/src/uts/common/io/audio/sada/mixer/am_ad.c
deleted file mode 100644
index 7d39aee5a7..0000000000
--- a/usr/src/uts/common/io/audio/sada/mixer/am_ad.c
+++ /dev/null
@@ -1,5155 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * This file contains the code that the Audio Driver calls into and the
- * private routines that support these calls.
- *
- * NOTE: When allocating conversion buffers for recording we have to be
- *	careful that the recording device isn't mono. Otherwise we won't
- *	have enough memory for converting.
- *
- * These entry points are defined in audio_mixer.h:
- *	am_attach()
- *	am_detach()
- *	am_get_audio()
- *	am_play_shutdown()
- *	am_send_audio()
- *	am_hw_state_change()
- *	am_get_src_data()
- *	am_set_src_data()
- *
- * These additional functions are also provided:
- *	am_ad_pause_play()
- *	am_ad_set_config()
- *	am_ad_set_format()
- *	am_ad_setup()
- *	am_ad_start_play()
- *	am_ad_start_record()
- *	am_ad_stop_play()
- *	am_ad_stop_record()
- *	am_ad_teadown()
- *	am_ck_channels()
- *	am_ck_combinations()
- *	am_ck_sample_rate()
- *	am_safe_putnext()
- *	am_test_canputnext()
- */
-
-#include <sys/debug.h>
-#include <sys/kmem.h>
-#include <sys/stat.h>
-#include <sys/errno.h>
-#include <sys/stropts.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/file.h>
-#include <sys/fcntl.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_apm.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/g711.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/impl/audio_mixer_impl.h>
-
-/*
- * Local routine prototypes used only by this file.
- */
-static int am_attach_check(audio_state_t *,
-	am_ad_info_t *, am_apm_private_t *, int);
-static int am_ck_sample_rate_sanity(am_ad_ch_cap_t *, int);
-static void am_convert_to_format(int *, void *, int, int, int, int);
-static int am_get_audio_multi(audio_state_t *, void *, int, int);
-static int am_get_audio_trad_compat(audio_state_t *, audio_apm_info_t *,
-	void *, int);
-static int am_get_audio_trad_mixer(audio_state_t *, audio_apm_info_t *,
-	void *, int);
-static int am_get_samples(audio_ch_t *, int, void *, int);
-static void am_play_shutdown_multi(audiohdl_t, int);
-static void am_play_shutdown_trad(audiohdl_t, audio_apm_info_t *);
-static void am_release_ad_access(am_apm_private_t *);
-static void am_send_audio_common(audio_ch_t *, void *, int);
-static void am_send_audio_multi(audio_state_t *, am_ad_info_t *, void *,
-	int, int);
-static void am_send_audio_trad_compat(audio_state_t *, audio_apm_info_t *,
-	int *, int);
-static void am_send_audio_trad_mixer(audio_state_t *, audio_apm_info_t *,
-	int *, const int);
-static void am_serialize_ad_access(am_apm_private_t *);
-
-/*
- * Taskq callbacks.
- */
-static void am_hw_task(void *);
-
-/*
- * Module global hidden variables
- */
-static audio_device_t mixer_device_info = {
-	MIXER_NAME,
-	MIXER_VERSION,
-	MIXER_CONFIGURATION
-};
-
-static int mixer_bufsize = AM_DEFAULT_MIX_BUFSIZE;
-
-/* XXX this is a lie */
-_NOTE(SCHEME_PROTECTS_DATA("this is a lie", mixer_bufsize))
-
-/*
- * The main routines for this file. These are the Audio Driver entry points.
- */
-
-/*
- * am_attach()
- *
- * TODO:	Check for PLINK
- *
- * Description:
- *	Attach an instance of the mixer. We initialize all the data structures
- *	and register the APM for both AUDIO and AUDIOCTL. Two audio_apm_info
- *	data structures are created, however they share the same private data
- *	structure.
- *
- *	We check both MIXER and COMPAT modes because we can switch between
- *	them and thus we need to make sure everything is okay for both modes.
- *	Plus we need the initial condition for both modes to support switching.
- *
- *	NOTE: mutex_init() and cv_init() no longer needs a name string, so set
- *	      to NULL to save kernel space.
- *
- *	NOTE: It is okay for memory allocation to sleep.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	ddi_attach_cmd_t cmd		Attach command
- *	am_ad_info_t	*ad_infop	Ptr to the device's capabilities struct
- *
- * Returns:
- *	AUDIO_SUCCESS		If the mixer was initialized properly
- *	AUDIO_FAILURE		If the mixer couldn't be initialized properly
- */
-int
-am_attach(audiohdl_t handle, ddi_attach_cmd_t cmd, am_ad_info_t *ad_infop)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_info_t		*hw_info;
-	audio_apm_info_t	*apm_infop1;
-	audio_apm_info_t	*apm_infop2;
-	am_apm_persist_t	*persistp;
-	am_apm_private_t	*stpptr;
-	audio_apm_reg_t		reg_info;
-	minor_t			minor;
-	int			mpgain = 0;
-	int			mpbal = 0;
-	int			mrgain = 0;
-	int			mrbal = 0;
-	uint_t			mgain = 0;
-	uint_t			pbalance = AUDIO_MID_BALANCE;
-	uint_t			pgain = 0;
-	uint_t			rbalance = AUDIO_MID_BALANCE;
-	uint_t			rgain = 0;
-	int			doread;		/* device supports record */
-	int			dowrite;	/* device supports play */
-	int			mode;
-	int			psr;
-	int			rsr;
-
-	ATRACE("in am_attach()", ad_infop);
-	ASSERT(statep);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		break;
-	default:
-		ATRACE_32("am_attach() unknown command failure", cmd);
-		return (AUDIO_FAILURE);
-	}
-
-	/* before we do anything else make sure the interface versions are ok */
-	if (ad_infop->ad_int_vers != AM_VERSION) {
-		ATRACE_32("am_attach() unsupported interface versions",
-		    ad_infop->ad_int_vers);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() interface version not supported: %d",
-		    ad_infop->ad_int_vers);
-		return (AUDIO_FAILURE);
-	}
-
-	/* get the properties from the .conf file */
-	if ((mixer_bufsize = ddi_prop_get_int(DDI_DEV_T_ANY, statep->as_dip,
-	    DDI_PROP_DONTPASS, "mixer_bufsize", AM_DEFAULT_MIX_BUFSIZE)) ==
-	    AM_DEFAULT_MIX_BUFSIZE) {
-		ATRACE_32("am_attach() setting mix buffer size",
-		    AM_DEFAULT_MIX_BUFSIZE);
-#ifdef DEBUG
-	} else {
-		ATRACE_32("am_attach() setting mix buffer size from .conf",
-		    mixer_bufsize);
-#else
-		/*EMPTY*/
-#endif
-	}
-	if (mixer_bufsize < AM_DEFAULT_MIX_BUFSIZE) {
-		ATRACE_32("am_attach() mix buffer too small", mixer_bufsize);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN, "am_attach() "
-		    "ddi_prop_get_int() mix buffer too small, setting to %d",
-		    AM_DEFAULT_MIX_BUFSIZE);
-		mixer_bufsize = AM_DEFAULT_MIX_BUFSIZE;
-		ATRACE_32("am_attach() setting new mix buffer size",
-		    mixer_bufsize);
-	}
-
-	/* figure out the direction and sanity check */
-	doread = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD;
-	dowrite = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY;
-	if (!doread && !dowrite) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() audio device doesn't play or record");
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_attach() doread", doread);
-	ATRACE_32("am_attach() dowrite", dowrite);
-
-	/* check the sample rate converter */
-	if (ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		/* make sure the version is okay */
-		if (dowrite &&
-		    ad_infop->ad_play.ad_conv->ad_version != AM_SRC_VERSION) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_attach() bad play src version: %d",
-			    ad_infop->ad_play.ad_conv->ad_version);
-			return (AUDIO_FAILURE);
-		}
-		if (doread &&
-		    ad_infop->ad_record.ad_conv->ad_version != AM_SRC_VERSION) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_attach() bad record src version: %d",
-			    ad_infop->ad_record.ad_conv->ad_version);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	/* get the state pointer for this instance */
-	if ((statep = audio_sup_devinfo_to_state(statep->as_dip)) == NULL) {
-		ATRACE("am_attach() couldn't get state structure", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() couldn't get state structure");
-		return (AUDIO_FAILURE);
-	}
-
-	/* allocate the audio mixer private data */
-	stpptr = kmem_zalloc(sizeof (*stpptr), KM_SLEEP);
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*stpptr));
-
-	hw_info = &stpptr->am_hw_info;
-
-	/* make sure we won't free the device state structure */
-	hw_info->ref_cnt = 1;
-
-	/* we assume the device is on line when we attach */
-	stpptr->am_flags |= AM_PRIV_ON_LINE;
-
-	/* initialize mutexes and cvs */
-	mutex_init(&stpptr->am_mode_lock, NULL, MUTEX_DRIVER, NULL);
-	mutex_init(&stpptr->am_ad_lock, NULL, MUTEX_DRIVER, NULL);
-	cv_init(&stpptr->am_mode_cv, NULL, CV_DRIVER, NULL);
-	cv_init(&stpptr->am_ad_cv, NULL, CV_DRIVER, NULL);
-
-	/*
-	 * WARNING: From here on all error returns must be through one
-	 *	of the error_? labels. Otherwise we'll have a memory leak.
-	 */
-
-	/* start off with allocating the signal message buffer */
-	if ((stpptr->am_sig_mp = allocb(sizeof (int8_t), BPRI_HI)) == NULL) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach() couldn't allocate message block to send signal");
-		goto error_free_private;
-	}
-	ASSERT(stpptr->am_sig_mp->b_cont == 0);
-	/* turn it into a signal */
-	stpptr->am_sig_mp->b_datap->db_type = M_PCSIG;
-	*stpptr->am_sig_mp->b_wptr++ = SIGPOLL;
-
-	/* register the mixer with the Audio Support Module */
-	reg_info.aar_version =		AM_AAR_VERSION;
-	reg_info.aar_apm_open =		am_open_audio;
-	reg_info.aar_apm_close =	am_close_audio;
-	reg_info.aar_apm_save_state =	am_save_state;
-	reg_info.aar_apm_restore_state = am_restore_state;
-	reg_info.aar_private =		stpptr;
-	reg_info.aar_info =		ad_infop;
-	reg_info.aar_state =		hw_info;
-	reg_info.aar_dev_info =		&mixer_device_info;
-
-	if ((apm_infop1 = audio_sup_register_apm(statep, AUDIO,
-	    &reg_info)) == NULL) {
-		ATRACE_32("am_attach() couldn't register AUDIO", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN, "am_attach() "
-		    "couldn't register the AUDIO device with audiosup");
-		goto error_free_msg;
-	}
-
-	reg_info.aar_apm_open =		am_open_audioctl;
-	reg_info.aar_apm_close =	am_close_audioctl;
-	reg_info.aar_apm_save_state =	NULL;
-	reg_info.aar_apm_restore_state = NULL;
-
-	if ((apm_infop2 = audio_sup_register_apm(statep, AUDIOCTL,
-	    &reg_info)) == NULL) {
-		ATRACE_32("am_attach() couldn't register AUDIOCTL", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN, "am_attach() "
-		    "couldn't register the AUDIOCTL device with audiosup");
-		goto error_unregister;
-	}
-
-	/* set the max input and output channels, making sure they are sane */
-	if (ad_infop->ad_record.ad_max_chs > statep->as_max_chs) {
-		stpptr->am_max_in_chs = statep->as_max_chs;
-	} else {
-		stpptr->am_max_in_chs = ad_infop->ad_record.ad_max_chs;
-	}
-	if (ad_infop->ad_play.ad_max_chs > statep->as_max_chs) {
-		stpptr->am_max_out_chs = statep->as_max_chs;
-	} else {
-		stpptr->am_max_out_chs = ad_infop->ad_play.ad_max_chs;
-	}
-	ATRACE_32("am_attach() am_max_in_chs", stpptr->am_max_in_chs);
-	ATRACE_32("am_attach() am_max_out_chs", stpptr->am_max_out_chs);
-
-	/* get the persistent state, if not there then first time so allocate */
-	persistp = audio_sup_get_persist_state(statep, AUDIO);
-	if (persistp == NULL) {
-		ATRACE("am_attach() 1st time persist or reset state", persistp);
-
-		/* allocate the structure and save */
-		persistp = kmem_alloc(sizeof (*persistp), KM_SLEEP);
-		if (audio_sup_set_persist_state(statep, AUDIO,
-		    persistp, sizeof (*persistp)) == AUDIO_FAILURE) {
-			ATRACE("am_attach() cannot set persist", 0)
-			goto error_unregister;
-		}
-
-		/* get the mode from the driver */
-		mode = ad_infop->ad_mode;
-
-		/* and also get the monitor gain */
-		mgain = ad_infop->ad_defaults->monitor_gain;
-
-		/* get the gains from the drive, based on mode */
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			if (dowrite) {
-				pgain = AM_DEFAULT_MIXER_GAIN;
-				pbalance = AUDIO_MID_BALANCE;
-				mpgain = pgain;
-				mpbal = pbalance;
-			}
-			if (doread) {
-				rgain = AUDIO_MID_GAIN;
-				rbalance = AUDIO_MID_BALANCE;
-				mrgain = rgain;
-				mrbal = rbalance;
-			}
-		} else {
-			if (dowrite) {
-				pgain = ad_infop->ad_defaults->play.gain;
-				pbalance = ad_infop->ad_defaults->play.balance;
-			}
-			if (doread) {
-				rgain = ad_infop->ad_defaults->record.gain;
-				rbalance =
-				    ad_infop->ad_defaults->record.balance;
-			}
-		}
-
-		persistp->apm_mpgain = pgain;
-		persistp->apm_mpbal = pbalance;
-		persistp->apm_mrgain = rgain;
-		persistp->apm_mrbal = rbalance;
-		persistp->apm_mode = mode;
-		persistp->apm_pgain = pgain;
-		persistp->apm_pbal = pbalance;
-		persistp->apm_rgain = rgain;
-		persistp->apm_rbal = rbalance;
-		persistp->apm_mgain = mgain;
-
-		if (dowrite) {
-			ad_infop->ad_defaults->play.port |=
-			    ad_infop->ad_defaults->play.avail_ports ^
-			    ad_infop->ad_defaults->play.mod_ports;
-		}
-		if (doread) {
-			ad_infop->ad_defaults->record.port |=
-			    ad_infop->ad_defaults->record.avail_ports ^
-			    ad_infop->ad_defaults->record.mod_ports;
-		}
-	} else {
-		/* the mode comes from the peristent data */
-		mode = persistp->apm_mode;
-		mpgain = persistp->apm_mpgain;
-		mpbal = persistp->apm_mpbal;
-		mrgain = persistp->apm_mrgain;
-		mrbal = persistp->apm_mrbal;
-		pgain = persistp->apm_pgain;
-		pbalance = persistp->apm_pbal;
-		ad_infop->ad_defaults->play.port = persistp->apm_pport;
-		ad_infop->ad_defaults->output_muted = persistp->apm_pmute;
-		rgain = persistp->apm_rgain;
-		rbalance = persistp->apm_rbal;
-		mgain = persistp->apm_mgain;
-		ad_infop->ad_defaults->record.port = persistp->apm_rport;
-	}
-
-	/* save persistent state in private data structure */
-	stpptr->am_pstate = persistp;
-
-	/*
-	 * CAUTION: Keep the calls in am_restore_state() the same as in
-	 *	this routine. There may be order dependencies and once the
-	 *	audio driver works we don't want to break it if we change
-	 *	the order.
-	 *
-	 * Check the write capabilities, if the device supports it.
-	 */
-	ASSERT(stpptr->am_pflags == NULL);
-	ASSERT(stpptr->am_rflags == NULL);
-	if (dowrite) {
-		if (am_attach_check(statep, ad_infop, stpptr, AUDIO_PLAY) ==
-		    AUDIO_FAILURE) {
-			ATRACE("am_attach() am_attach_check() failed", 0);
-			goto error_unregister_both;
-		}
-
-		/* get initial conditions */
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			psr = stpptr->am_save_psr;
-		} else {
-			psr = ad_infop->ad_defaults->play.sample_rate;
-		}
-
-		/* see if we need to translate to unsigned */
-		if (ad_infop->ad_translate_flags & AM_MISC_16_P_TRANSLATE) {
-			stpptr->am_pflags |= AM_PRIV_16_TRANS;
-		}
-		if (ad_infop->ad_translate_flags & AM_MISC_8_P_TRANSLATE) {
-			stpptr->am_pflags |= AM_PRIV_8_TRANS;
-		}
-
-		/* now that we've got the h/w info we set the initial format */
-		if (am_set_format(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-		    AUDIO_PLAY, psr, stpptr->am_hw_pchs, stpptr->am_hw_pprec,
-		    stpptr->am_hw_penc, AM_NO_FORCE, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(AUDIO_STATE2HDL(statep),
-			    CE_WARN, "am_attach() "
-			    "couldn't set play data format: %d %d %d %d",
-			    psr, stpptr->am_hw_pchs, stpptr->am_hw_pprec,
-			    stpptr->am_hw_penc);
-			goto error_unregister_both;
-		}
-
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			if (am_set_gain(statep, apm_infop1,
-			    hw_info->play.channels, mpgain, mpbal, AUDIO_PLAY,
-			    AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-			    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "am_attach() couldn't set mixer play gain");
-				goto error_unregister_both;
-			}
-		} else {
-			if (am_set_gain(statep, apm_infop1,
-			    hw_info->play.channels, pgain, pbalance, AUDIO_PLAY,
-			    AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-			    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "am_attach() "
-				    "couldn't set compat play gain");
-				goto error_unregister_both;
-			}
-		}
-	} else {
-		ASSERT(hw_info->play.sample_rate == 0);
-		pgain = AUDIO_MIN_GAIN;
-		pbalance = AUDIO_MID_BALANCE;
-	}
-	hw_info->play.gain = pgain;
-	hw_info->play.balance = pbalance;
-
-	/* now check the read capabilities, if the device supports it */
-	if (doread) {
-		if (am_attach_check(statep, ad_infop, stpptr, AUDIO_RECORD) ==
-		    AUDIO_FAILURE) {
-			ATRACE("am_attach() am_attach_check() failed", 0);
-			goto error_unregister_both;
-		}
-
-		/* get initial conditions */
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			rsr = stpptr->am_save_rsr;
-		} else {
-			rsr = ad_infop->ad_defaults->record.sample_rate;
-		}
-
-		/* see if we need to translate to unsigned */
-		if (ad_infop->ad_translate_flags & AM_MISC_16_R_TRANSLATE) {
-			stpptr->am_rflags |= AM_PRIV_16_TRANS;
-		}
-		if (ad_infop->ad_translate_flags & AM_MISC_8_R_TRANSLATE) {
-			stpptr->am_rflags |= AM_PRIV_8_TRANS;
-		}
-
-		/* now that we've got the h/w info we set the initial format */
-		if (am_set_format(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-		    AUDIO_RECORD, rsr, stpptr->am_hw_rchs, stpptr->am_hw_rprec,
-		    stpptr->am_hw_renc, AM_NO_FORCE, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_attach() "
-			    "couldn't set record data format: %d %d %d %d",
-			    rsr, stpptr->am_hw_rchs, stpptr->am_hw_rprec,
-			    stpptr->am_hw_renc);
-			goto error_unregister_both;
-		}
-
-		/* set the gains */
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			if (am_set_gain(statep, apm_infop1,
-			    hw_info->record.channels, mrgain, mrbal,
-			    AUDIO_RECORD, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-			    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "am_attach() "
-				    "couldn't set mixer record gain");
-				goto error_unregister_both;
-			}
-		} else {
-			if (am_set_gain(statep, apm_infop1,
-			    hw_info->record.channels, rgain, rbalance,
-			    AUDIO_RECORD, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-			    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "am_attach() "
-				    "couldn't set compat record gain");
-				goto error_unregister_both;
-			}
-		}
-	} else {
-		ASSERT(hw_info->record.sample_rate == 0);
-		rgain = AUDIO_MIN_GAIN;
-		rbalance = AUDIO_MID_BALANCE;
-	}
-	hw_info->record.gain = rgain;
-	hw_info->record.balance = rbalance;
-
-	/* starting point for unmute */
-	stpptr->am_save_hw_rgain = AUDIO_MID_GAIN;
-
-	/*
-	 * Allocate mix and send buffers, but only if TRAD Codec and doing
-	 * this direction.
-	 */
-	ASSERT(stpptr->am_mix_size == 0);
-	ASSERT(stpptr->am_mix_buf == NULL);
-	ASSERT(stpptr->am_send_size == 0);
-	ASSERT(stpptr->am_send_buf == NULL);
-	if (ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		/* allocate the mix buffer */
-		if (dowrite) {
-			stpptr->am_mix_buf = kmem_alloc(mixer_bufsize,
-			    KM_SLEEP);
-			stpptr->am_mix_size = mixer_bufsize;
-		}
-
-		/* allocate the send buffer */
-		if (doread) {
-			stpptr->am_send_buf = kmem_alloc(mixer_bufsize,
-			    KM_SLEEP);
-			stpptr->am_send_size = mixer_bufsize;
-		}
-	}
-
-	/* create the devices for successive open()'s to clone off of */
-	if ((minor = audio_sup_type_to_minor(AUDIO)) == AUDIO_FAILURE) {
-		ATRACE("am_attach() sound,audio get minor failure", minor);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() create audio minor node failure");
-		goto error_unregister_both;
-	}
-	minor += statep->as_dev_instance *
-		audio_sup_get_minors_per_inst(AUDIO_STATE2HDL(statep));
-	if (ddi_create_minor_node(statep->as_dip, "sound,audio", S_IFCHR,
-	    minor, DDI_NT_AUDIO, 0) == DDI_FAILURE) {
-		ATRACE("am_attach() sound,audio minor dev failure", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() create audio minor node failure");
-		goto error_unregister_both;
-	}
-
-	if ((minor = audio_sup_type_to_minor(AUDIOCTL)) == AUDIO_FAILURE) {
-		ATRACE("am_attach() sound,audioctl get minor failure", minor);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() create audioctl minor node failure");
-		goto error_rem_minor;
-	}
-	minor += statep->as_dev_instance *
-		audio_sup_get_minors_per_inst(AUDIO_STATE2HDL(statep));
-	if (ddi_create_minor_node(statep->as_dip, "sound,audioctl", S_IFCHR,
-	    minor, DDI_NT_AUDIO, 0) == DDI_FAILURE) {
-		ATRACE("am_attach() sound,audioctl minor dev failure", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() create audioctl minor node failure");
-		goto error_rem_minor;
-	}
-
-	ASSERT(hw_info->play.port == 0);
-	ASSERT(hw_info->play.avail_ports == 0);
-	ASSERT(hw_info->play.mod_ports == 0);
-	/* turn on in the state structure those ports that will always be on */
-	if (dowrite) {
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_PORT, AUDIO_PLAY,
-		    ad_infop->ad_defaults->play.port, NULL,
-		    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_attach() couldn't set play port: 0x%x",
-			    ad_infop->ad_defaults->play.port);
-			goto error_rem_minor;
-		} else {
-			hw_info->play.port =
-			    ad_infop->ad_defaults->play.port;
-			hw_info->play.avail_ports =
-			    ad_infop->ad_defaults->play.avail_ports;
-			hw_info->play.mod_ports =
-			    ad_infop->ad_defaults->play.mod_ports;
-		}
-	}
-	ASSERT(hw_info->record.port == 0);
-	ASSERT(hw_info->record.avail_ports == 0);
-	ASSERT(hw_info->record.mod_ports == 0);
-	/* turn on in the state structure those ports that will always be on */
-	if (doread) {
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_PORT, AUDIO_RECORD,
-		    ad_infop->ad_defaults->record.port, NULL,
-		    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_attach() couldn't set record port: 0x%x",
-			    ad_infop->ad_defaults->record.port);
-			goto error_rem_minor;
-		} else {
-			hw_info->record.port =
-			    ad_infop->ad_defaults->record.port;
-			hw_info->record.avail_ports =
-			    ad_infop->ad_defaults->record.avail_ports;
-			hw_info->record.mod_ports =
-			    ad_infop->ad_defaults->record.mod_ports;
-		}
-	}
-
-	if ((ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_IN2OUT) &&
-	    am_ad_set_config(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-	    AM_SET_MONITOR_GAIN, AUDIO_BOTH, mgain, NULL, AM_NO_SERIALIZE) ==
-	    AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() couldn't set monitor gain: 0x%x", mgain);
-		goto error_rem_minor;
-	}
-
-	if (dowrite && am_ad_set_config(statep, stpptr, ad_infop,
-	    AM_SET_CONFIG_BOARD, AM_OUTPUT_MUTE, AUDIO_PLAY,
-	    ad_infop->ad_defaults->output_muted, NULL, AM_NO_SERIALIZE) ==
-	    AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() couldn't set output muted: 0x%x",
-		    ad_infop->ad_defaults->output_muted);
-		goto error_rem_minor;
-	}
-
-	if (doread && (ad_infop->ad_assist_flags & AM_ASSIST_MIC) &&
-	    am_ad_set_config(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-	    AM_MIC_BOOST, AUDIO_RECORD,
-	    (ad_infop->ad_add_mode & AM_ADD_MODE_MIC_BOOST), NULL,
-	    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() couldn't set mic boost: 0x%x",
-		    ad_infop->ad_add_mode);
-		goto error_rem_minor;
-	}
-
-	/* set misc hardware state */
-	if (dowrite) {
-		hw_info->play.buffer_size =
-		    ad_infop->ad_defaults->play.buffer_size;
-	} else {
-		hw_info->play.buffer_size = 0;
-	}
-	ASSERT(hw_info->play.samples == 0);
-	ASSERT(hw_info->play.eof == 0);
-	ASSERT(hw_info->play.pause == 0);
-	ASSERT(hw_info->play.error == 0);
-	ASSERT(hw_info->play.waiting == 0);
-	ASSERT(hw_info->play.minordev == 0);
-	ASSERT(hw_info->play.open == 0);
-	ASSERT(hw_info->play.active == 0);
-	if (doread) {
-		hw_info->record.buffer_size =
-		    ad_infop->ad_defaults->record.buffer_size;
-	} else {
-		hw_info->record.buffer_size = 0;
-	}
-	ASSERT(hw_info->record.samples == 0);
-	ASSERT(hw_info->record.eof == 0);
-	ASSERT(hw_info->record.pause == 0);
-	ASSERT(hw_info->record.error == 0);
-	ASSERT(hw_info->record.waiting == 0);
-	ASSERT(hw_info->record.minordev == 0);
-	ASSERT(hw_info->record.open == 0);
-	ASSERT(hw_info->record.active == 0);
-
-	hw_info->monitor_gain = mgain;
-	hw_info->output_muted = ad_infop->ad_defaults->output_muted;
-	hw_info->hw_features = ad_infop->ad_defaults->hw_features;
-	hw_info->sw_features = ad_infop->ad_defaults->sw_features;
-
-	if (mode == AM_MIXER_MODE) {
-		hw_info->sw_features_enabled = AUDIO_SWFEATURE_MIXER;
-	} else {
-		hw_info->sw_features_enabled = 0;
-	}
-
-	/* create a single threaded task queue, this always succeeds */
-	stpptr->am_taskq = audio_sup_taskq_create(AM_PRIV_TASKQ_NAME);
-	if (stpptr->am_taskq == NULL) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() couldn't create the taskq, out of memory");
-		goto error_rem_minor;
-	}
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*stpptr));
-
-	ATRACE_32("am_attach() returning", statep->as_dev_instance);
-
-	return (AUDIO_SUCCESS);
-
-error_rem_minor:
-	ATRACE("am_attach() failure, removing minor nodes", 0);
-	/*
-	 * We don't use NULL because other APMs may own other minor devices
-	 * and we don't want to remove them from under them.
-	 */
-	ddi_remove_minor_node(statep->as_dip, "sound,audio");
-	ddi_remove_minor_node(statep->as_dip, "sound,audioctl");
-
-error_unregister_both:
-	mutex_enter(&apm_infop2->apm_lock);
-	apm_infop2->apm_private = NULL;
-	mutex_exit(&apm_infop2->apm_lock);
-	if (audio_sup_unregister_apm(statep, AUDIOCTL) == AUDIO_FAILURE) {
-		ATRACE_32("am_attach() audio_sup_unregister_apm() "
-		    "AUDIOCTL failed", statep->as_dev_instance);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() audio_sup_unregister_apm() "
-		    "AUDIOCTL failed");
-	}
-
-error_unregister:
-	mutex_enter(&apm_infop1->apm_lock);
-	apm_infop1->apm_private = NULL;
-	mutex_exit(&apm_infop1->apm_lock);
-	if (audio_sup_unregister_apm(statep, AUDIO) == AUDIO_FAILURE) {
-		ATRACE_32("am_attach() audio_sup_unregister_apm() "
-		    "AUDIO failed", statep->as_dev_instance);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-		    "am_attach() audio_sup_unregister_apm() "
-		    "AUDIO failed");
-	}
-
-error_free_msg:
-	freemsg(stpptr->am_sig_mp);
-
-error_free_private:
-	/* destroy mutexes and cvs */
-	mutex_destroy(&stpptr->am_mode_lock);
-	mutex_destroy(&stpptr->am_ad_lock);
-	cv_destroy(&stpptr->am_mode_cv);
-	cv_destroy(&stpptr->am_ad_cv);
-
-	ATRACE("am_attach() failure, freeing private structure", 0);
-	if (stpptr->am_mix_buf) {
-		kmem_free(stpptr->am_mix_buf, stpptr->am_mix_size);
-	}
-	if (stpptr->am_send_buf) {
-		kmem_free(stpptr->am_send_buf, stpptr->am_send_size);
-	}
-	kmem_free(stpptr, sizeof (*stpptr));
-
-fail:
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*stpptr));
-	ATRACE("am_attach() returning failure", 0);
-
-	return (AUDIO_FAILURE);
-
-}	/* am_attach() */
-
-/*
- * am_detach()
- *
- * Description:
- *	Detach an instance of the mixer. Free up all data structures and
- *	unregister the APM for both AUDIO and AUDIOCTL. We also remove
- *	the device nodes. However it is possible another APM may still be
- *	attached, so we are careful to only remove the audio and audioctl
- *	devices.
- *
- *	NOTE: This routine will never be called in the audio device
- *		has any channels in use, so we don't need to check
- *		for this.
- *
- *
- * Arguments:
- *	audiohdl_t	handle	Handle to the device
- *	ddi_detach_cmd_t cmd	Detach command
- *
- * Returns:
- *	AUDIO_SUCCESS		If the mixer was detached
- *	AUDIO_FAILURE		If the mixer couldn't be detached
- */
-int
-am_detach(audiohdl_t handle, ddi_detach_cmd_t cmd)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t	*apm_infop1;
-	audio_apm_info_t	*apm_infop2;
-	am_apm_private_t	*stpptr;
-	am_apm_persist_t	*persistp;
-	audio_info_t		*hw_info;
-
-	ATRACE_32("in am_detach()", cmd);
-	ATRACE("am_detach() handle", handle);
-	ASSERT(statep);
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-	default:
-		ATRACE_32("am_detach() unknown command failure", cmd);
-		return (AUDIO_FAILURE);
-	}
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*stpptr));
-
-	/*
-	 * Remove only the AUDIO and AUDIOCTL minor nodes for this
-	 * dev_info. We don't want to free the nodes other APMs are
-	 * responsible for.
-	 */
-	ddi_remove_minor_node(statep->as_dip, "sound,audio");
-	ddi_remove_minor_node(statep->as_dip, "sound,audioctl");
-
-	/* get rid of the private data structure */
-	if ((apm_infop1 = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_detach() audio_sup_get_apm_info() AUDIO failed",
-		    statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "detach() audio_sup_get_apm_info() AUDIO failed");
-
-		goto fail;
-	}
-	if ((apm_infop2 =
-	    audio_sup_get_apm_info(statep, AUDIOCTL)) == NULL) {
-		ATRACE("am_detach() audio_sup_get_apm_info() AUDIOCTL failed",
-		    statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "detach() audio_sup_get_apm_info() AUDIOCTL failed");
-
-		goto fail;
-	}
-	/*
-	 * Both apm_info pointers use the same apm_private structure, so we
-	 * only need to clear it once.
-	 */
-	stpptr = apm_infop1->apm_private;
-	ASSERT(stpptr->am_channels == 0);
-	ASSERT(stpptr->am_in_chs == 0);
-	ASSERT(stpptr->am_out_chs == 0);
-
-	/* save the current h/w state in persistent memory */
-	hw_info = &stpptr->am_hw_info;
-	persistp = stpptr->am_pstate;
-	if (persistp->apm_mode == AM_MIXER_MODE) {
-		persistp->apm_mpgain = hw_info->play.gain;
-		persistp->apm_mpbal = hw_info->play.balance;
-		persistp->apm_mrgain = hw_info->record.gain;
-		persistp->apm_mrbal = hw_info->record.balance;
-	}
-	persistp->apm_pgain = hw_info->play.gain;
-	persistp->apm_pbal = hw_info->play.balance;
-	persistp->apm_pport = hw_info->play.port;
-	persistp->apm_pmute = hw_info->output_muted;
-	persistp->apm_rgain = hw_info->record.gain;
-	persistp->apm_rbal = hw_info->record.balance;
-	persistp->apm_rport = hw_info->record.port;
-	persistp->apm_mgain = hw_info->monitor_gain;
-
-	/* wait for the taskq to empty and then destroy it */
-	audio_sup_taskq_wait(stpptr->am_taskq);
-	audio_sup_taskq_destroy(stpptr->am_taskq);
-
-	/* destroy mutexes and cvs */
-	mutex_destroy(&stpptr->am_mode_lock);
-	mutex_destroy(&stpptr->am_ad_lock);
-	cv_destroy(&stpptr->am_mode_cv);
-	cv_destroy(&stpptr->am_ad_cv);
-
-	ASSERT(stpptr->am_sig_mp);
-	freemsg(stpptr->am_sig_mp);
-
-	if (stpptr->am_mix_buf) {
-		ASSERT(stpptr->am_mix_size);
-		kmem_free(stpptr->am_mix_buf, stpptr->am_mix_size);
-	}
-	if (stpptr->am_send_buf) {
-		ASSERT(stpptr->am_send_size);
-		kmem_free(stpptr->am_send_buf, stpptr->am_send_size);
-	}
-	kmem_free(stpptr, sizeof (*stpptr));
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*stpptr));
-
-	mutex_enter(&apm_infop1->apm_lock);
-	apm_infop1->apm_private = NULL;
-	mutex_exit(&apm_infop1->apm_lock);
-	mutex_enter(&apm_infop2->apm_lock);
-	apm_infop2->apm_private = NULL;
-	mutex_exit(&apm_infop2->apm_lock);
-
-	if (audio_sup_unregister_apm(statep, AUDIO) == AUDIO_FAILURE) {
-		ATRACE("am_detach() audio_sup_unregister_apm() "
-			"AUDIO failed", statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "detach() audio_sup_unregister_apm() "
-		    "AUDIO failed");
-	}
-
-	if (audio_sup_unregister_apm(statep, AUDIOCTL) == AUDIO_FAILURE) {
-		ATRACE("am_detach() audio_sup_unregister_apm() "
-			"AUDIOCTL failed", statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "detach() audio_sup_unregister_apm() "
-		    "AUDIOCTL failed");
-	}
-
-	ATRACE("am_detach() done", statep);
-
-	return (AUDIO_SUCCESS);
-
-fail:
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*stpptr));
-	return (AUDIO_FAILURE);
-
-}	/* am_detach() */
-
-/*
- * am_get_audio()
- *
- * Description:
- *	This routine directs the call to get audio depending on whether or
- *	not the Codec is a traditional or multi-channel Codec. It also does
- *	some error checking to make sure the call is valid.
- *
- *	It is an error for the number of samples to not be modulo the
- *	hardware's number of channels. The old diaudio driver would hang
- *	the channel until it was closed. We throw away enough of the damaged
- *	sample and press on.
- *
- *	We support devices that accept only unsigned linear PCM by translating
- *	the final data, if required.
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		wants. So it is samples at the hardware's sample rate.
- *
- * Arguments:
- *	audiohdl_t	handle	Handle to the device
- *	void		*buf	The buffer to place the audio into
- *	int		channel	For multi-channel Codecs this is the channel
- *				that will be playing the sound.
- *	int		samples	The number of samples to get
- *
- * Returns
- *	>= 0			The number of samples transferred to the buffer
- *	AUDIO_FAILURE		An error has occurred
- */
-int
-am_get_audio(audiohdl_t handle, void *buf, int channel, int samples)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t	*apm_infop;
-	am_apm_private_t	*stpptr;
-	am_ad_info_t		*ad_infop;
-	uint_t			hw_channels;
-	int			mode;
-	int			rc;
-
-	ATRACE_32("in am_get_audio() samples requested", samples);
-	ASSERT(statep);
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_get_audio() audio_sup_get_apm_info() AUDIO failed",
-		    statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep),
-		    CE_NOTE, "get_audio() "
-		    "audio_sup_get_apm_info() AUDIO failed, audio lost");
-		return (AUDIO_FAILURE);
-	}
-
-	ad_infop = apm_infop->apm_ad_infop;
-	stpptr = apm_infop->apm_private;
-
-	/*
-	 * Make sure we don't ask for more data than the mixer can provide.
-	 * The mixer uses mixer_bufsize to allocate buffers so we must insure
-	 * that we don't try to mix too much and thus get into trouble.
-	 */
-	if ((samples << AM_INT32_SHIFT) > mixer_bufsize) {
-		samples = mixer_bufsize >> AM_INT32_SHIFT;
-		ATRACE("am_get_audio() asking for too many samples, resetting",
-		    samples);
-	}
-
-	/* deal with multi-channel Codecs or regular Codecs */
-	if (ad_infop->ad_codec_type == AM_MS_CODEC) {
-		ATRACE("am_get_audio() calling am_get_audio_multi()", statep);
-		ASSERT(channel != AUDIO_NO_CHANNEL);
-		rc = am_get_audio_multi(statep, buf, channel, samples);
-		ATRACE_32("am_get_audio() am_get_audio_multi() returning", rc);
-	} else {
-		ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-		ASSERT(channel == AUDIO_NO_CHANNEL);
-
-		/* make sure the # of samples is modulo the # of H/W channels */
-		hw_channels = stpptr->am_hw_pchs;
-		if (hw_channels != AUDIO_CHANNELS_MONO &&
-		    (samples % hw_channels) != 0) {
-			ATRACE_32("am_get_audio() bad sample size", samples);
-			samples -= samples % hw_channels;
-		}
-
-		mode = stpptr->am_pstate->apm_mode;
-		if (mode == AM_MIXER_MODE) {
-			ATRACE("am_get_audio() "
-			    "calling am_get_audio_trad_mixer()", statep);
-			rc = am_get_audio_trad_mixer(statep, apm_infop, buf,
-			    samples);
-			ATRACE_32("am_get_audio() "
-			    "am_get_audio_trad_mixer() returning", rc);
-		} else {
-			ASSERT(mode == AM_COMPAT_MODE);
-			ATRACE("am_get_audio() "
-			    "calling am_get_audio_trad_compat()", statep);
-			rc = am_get_audio_trad_compat(statep, apm_infop, buf,
-			    samples);
-			ATRACE_32("am_get_audio() "
-			    "am_get_audio_trad_compat() returning", rc);
-		}
-	}
-
-	return (rc);
-
-}	/* am_get_audio() */
-
-/*
- * am_hw_state_change()
- *
- * Description:
- *	This routine provides feedback from the audio driver to the mixer
- *	when a user has caused a state change. Usually this is by a button
- *	or knob that the user presses or turns on a speaker or mic. Currently
- *	support for this feedback is somewhat limited as we only support
- *	gain, balance, and mute.
- *
- *	NOTE: We don't include a channel argument because user input is
- *		fairly limited, it is oriented towards the master hardware
- *		device.
- *
- *	NOTE: Unlike the ioctl()s, we don't increment acp_tq_cnt. acp_tq_cnt
- *		is on a per channel basis and we aren't doing this work for any
- *		one channel, but for the whole device. This means that there
- *		aren't any close()s that are going to wait for these tasks to
- *		complete.
- *
- *	NOTE: Mute is a toggle from the current setting.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	int		cmd		Command for update
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		value		The value to scale
- *	int		sleep		AUDIO_NO_SLEEP or AUDIO_SLEEP
- *
- * Returns:
- *	AUDIO_SUCCESS			Command queued up on the taskq
- *	AUDIO_FAILURE			Bad argument, can't allocate memory, etc
- */
-int
-am_hw_state_change(audiohdl_t handle, int cmd, int dir, int value, int sleep)
-{
-	audio_state_t		*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t	*apm_infop;
-	am_ad_info_t		*ad_infop;
-	am_apm_private_t	*stpptr;
-	am_state_ch_args_t	*arg;
-
-	ATRACE("in am_hw_state_change()", handle);
-
-	/* make sure the handle is good */
-	if (handle == NULL) {
-		ATRACE("am_hw_state_change() bad handle", handle);
-		return (AUDIO_FAILURE);
-	}
-
-	/* don't allow both play and record */
-	if ((dir & AUDIO_BOTH) == AUDIO_BOTH) {
-		ATRACE_32("am_hw_state_change() bad direction", dir);
-		return (AUDIO_FAILURE);
-	}
-
-	/* don't allow both sleep and no sleep */
-	if (sleep == (AUDIO_NO_SLEEP|AUDIO_SLEEP)) {
-		ATRACE_32("am_hw_state_change() bad sleep", sleep);
-		return (AUDIO_FAILURE);
-	}
-	if (sleep == AUDIO_SLEEP) {
-		sleep = KM_SLEEP;
-	} else {
-		ASSERT(sleep == AUDIO_NO_SLEEP);
-		sleep = KM_NOSLEEP;
-	}
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_hw_state_change() "
-		    "audio_sup_get_apm_info() AUDIO failed", statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep),
-		    CE_NOTE, "am_hw_state_change() "
-		    "audio_sup_get_apm_info() AUDIO failed");
-		return (AUDIO_FAILURE);
-	}
-	stpptr = apm_infop->apm_private;
-
-	/* Check if we have a valid taskq */
-	if (stpptr->am_taskq == NULL) {
-		ATRACE("am_hw_state_change() taskq not setup", stpptr);
-		return (AUDIO_FAILURE);
-	}
-
-	switch (cmd) {
-	case AM_HWSC_SET_GAIN_ABS:
-		/*FALLTHROUGH*/
-	case AM_HWSC_SET_BAL_ABS:
-		/*FALLTHROUGH*/
-	case AM_HWSC_SET_GAIN_DELTA:
-		/*FALLTHROUGH*/
-	case AM_HWSC_SET_BAL_DELTA:
-		/*FALLTHROUGH*/
-	case AM_HWSC_MUTE_TOGGLE:
-		/* does the h/w support the direction? */
-		ad_infop = apm_infop->apm_ad_infop;
-		if ((dir == AUDIO_PLAY &&
-		    !(ad_infop->ad_defaults->hw_features &
-		    AUDIO_HWFEATURE_PLAY)) ||
-		    (dir == AUDIO_RECORD &&
-		    !(ad_infop->ad_defaults->hw_features &
-		    AUDIO_HWFEATURE_RECORD))) {
-			ATRACE_32("am_hw_state_change() dir !supported", dir);
-			return (AUDIO_FAILURE);
-		}
-
-		/* allocate argument memory for taskq */
-		if ((arg = kmem_alloc(sizeof (*arg), sleep)) == NULL) {
-			ATRACE("am_hw_state_change() alloc failed", statep);
-			return (AUDIO_FAILURE);
-		}
-		arg->asca_statep = statep;
-		arg->asca_apm_infop = apm_infop;
-		arg->asca_cmd = cmd;
-		arg->asca_dir = dir;
-		arg->asca_value = value;
-
-		break;
-	case AM_HWSC_ONLINE:
-		mutex_enter(&apm_infop->apm_lock);
-		stpptr->am_flags |= AM_PRIV_ON_LINE;
-		mutex_exit(&apm_infop->apm_lock);
-
-		return (AUDIO_SUCCESS);
-	case AM_HWSC_OFFLINE:
-		mutex_enter(&apm_infop->apm_lock);
-		stpptr->am_flags &= ~AM_PRIV_ON_LINE;
-		mutex_exit(&apm_infop->apm_lock);
-
-		return (AUDIO_SUCCESS);
-	default:
-		ATRACE_32("am_hw_state_change() bad command", cmd);
-		return (AUDIO_FAILURE);
-	}
-
-	/* schedule the task */
-	if (audio_sup_taskq_dispatch(stpptr->am_taskq, am_hw_task,
-	    arg, sleep) == AUDIO_FAILURE) {
-		kmem_free(arg, sizeof (*arg));
-		ATRACE("am_hw_state_change() taskq sched failed", statep);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("am_hw_state_change() done", handle);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_hw_state_change() */
-
-/*
- * am_play_shutdown()
- *
- * Description:
- *	This routine is used to clean things up when the Audio Driver will
- *	no longer be servicing it's play interrupts. I.e., play interrupts
- *	have been turned off.
- *
- *	This routine makes sure that any DRAINs waiting for an interrupt are
- *	cleared.
- *
- *	It is also used to coordinate shutting down play so that we can
- *	switch between MIXER and COMPAT modes.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	int		channel		For multi-stream Codecs this is the
- *					stream to shutdown.
- *
- * Returns:
- *	void
- */
-void
-am_play_shutdown(audiohdl_t handle, int channel)
-{
-	audio_state_t			*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t		*apm_infop;
-	am_ad_info_t			*ad_infop;
-
-	ATRACE("in am_play_shutdown()", handle);
-	ASSERT(statep);
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_play_shutdown() audio_sup_get_apm_info() failed", 0);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "play_shutdown() audio_sup_get_apm_info() failed");
-		return;
-	}
-	ad_infop = apm_infop->apm_ad_infop;
-
-	/* deal with multi-channel Codecs vs. regular Codecs */
-	if (ad_infop->ad_codec_type == AM_MS_CODEC) {
-		ATRACE("am_play_shutdown() calling shutdown_multi()", ad_infop);
-		am_play_shutdown_multi(handle, channel);
-		ATRACE("am_play_shutdown() shutdown_multi() done", ad_infop);
-	} else {
-		ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-		ATRACE("am_play_shutdown() calling shutdown_trad()", ad_infop);
-		am_play_shutdown_trad(handle, apm_infop);
-		ATRACE("am_play_shutdown() shutdown_trad() done", ad_infop);
-	}
-
-	ATRACE("am_play_shutdown() returning", statep);
-
-	return;
-
-}	/* am_play_shutdown() */
-
-/*
- * am_send_audio()
- *	This routine directs the call to send audio depending on whether or
- *	not the Codec is a traditional or multi-channel Codec. It also does
- *	some error checking to make sure the call is valid.
- *
- *	It is an error for the number of samples to not be modulo the
- *	hardware's number of channels. The old diaudio driver would hang
- *	the channel until it was closed. We throw away enough of the damaged
- *	sample and press on.
- *
- *	We support devices that provide only unsigned linear PCM by translating
- *	the data to signed before it is used.
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		sends. So it is samples at the hardware's sample rate.
- *
- * Description:
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	void		*buf		The buffer the audio is in
- *	int		channel		For multi-channel Codecs this is the
- *					channel that will be receiving the audio
- *	int		samples		The number of samples to send
- *
- * Returns
- *	void
- */
-void
-am_send_audio(audiohdl_t handle, void *buf, int channel, int samples)
-{
-	audio_state_t			*statep = AUDIO_HDL2STATE(handle);
-	audio_apm_info_t		*apm_infop;
-	am_apm_private_t		*stpptr;
-	am_ad_info_t			*ad_infop;
-	size_t				size;
-	uint_t				hw_channels;
-	int				mode;
-
-	ATRACE("in am_send_audio()", handle);
-	ATRACE("am_send_audio() buf", buf);
-	ATRACE("am_send_audio() channel", channel);
-	ATRACE("am_send_audio() samples", samples);
-	ASSERT(statep);
-
-	/* reject processing 0 or less samples */
-	if (samples <= 0 || samples > AM_MAX_SAMPLES) {
-		ATRACE_32("am_send_audio() bad samples arg #1, returning",
-		    samples);
-		return;
-	}
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_send_audio() audio_sup_get_apm_info() AUDIO failed",
-		    statep);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "send_audio() audio_sup_get_apm_info() "
-		    "AUDIO failed, recorded audio lost");
-		return;
-	}
-
-	ad_infop = apm_infop->apm_ad_infop;
-
-	/* deal with multi-channel Codecs or regular Codecs */
-	if (ad_infop->ad_codec_type == AM_MS_CODEC) {
-		ATRACE("am_send_audio() calling am_send_audio_multi()", statep);
-		ASSERT(channel != AUDIO_NO_CHANNEL);
-		am_send_audio_multi(statep, apm_infop->apm_ad_infop, buf,
-		    channel, samples);
-		ATRACE_32("am_send_audio() am_send_audio_multi() returning", 0);
-	} else {
-		ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-		ASSERT(channel == AUDIO_NO_CHANNEL);
-
-		/* make sure the # of samples is modulo the # of H/W channels */
-		stpptr = apm_infop->apm_private;
-		hw_channels = stpptr->am_hw_rchs;
-		if (hw_channels != AUDIO_CHANNELS_MONO &&
-		    (samples % hw_channels) != 0) {
-			ATRACE_32("am_send_audio() bad sample size", samples);
-			samples -= samples % hw_channels;
-			ATRACE("am_send_audio() adjusted samples", samples);
-			if (samples <= 0) {
-				ATRACE_32("am_send_audio() bad samples arg #2",
-				    samples);
-				return;
-			}
-		}
-
-		/* make sure the send buffer is large enough */
-		size = ((size_t)samples) << AM_TIMES_4_SHIFT;
-		ATRACE("am_send_audio() size", size);
-
-		mutex_enter(&apm_infop->apm_lock);
-
-		if (stpptr->am_send_size < size) {
-			ATRACE("am_send_audio() old am_send_buf",
-			    stpptr->am_send_buf);
-			ATRACE("am_send_audio() old am_send_size",
-			    stpptr->am_send_size);
-			if (stpptr->am_send_size) {
-				kmem_free(stpptr->am_send_buf,
-				    stpptr->am_send_size);
-			}
-			stpptr->am_send_buf = kmem_alloc(size, KM_NOSLEEP);
-			if (stpptr->am_send_buf == NULL) {
-				stpptr->am_send_size = 0;
-				mutex_exit(&apm_infop->apm_lock);
-				audio_sup_log(AUDIO_STATE2HDL(statep),
-				    CE_NOTE, "!mixer: send_audio_trad() "
-				    "couldn't allocate send buffer, audio "
-				    "lost");
-				return;
-			}
-			stpptr->am_send_size = size;
-			ATRACE("am_send_audio() new am_send_buf",
-			    stpptr->am_send_buf);
-			ATRACE("am_send_audio() new am_send_size",
-			    stpptr->am_send_size);
-		}
-		mutex_exit(&apm_infop->apm_lock);
-
-		/* convert to canonical format */
-		am_convert_to_int(buf, stpptr->am_send_buf, samples,
-		    stpptr->am_hw_rprec, stpptr->am_hw_renc, stpptr->am_rflags);
-
-		mode = stpptr->am_pstate->apm_mode;
-		if (mode == AM_MIXER_MODE) {
-			ATRACE("am_send_audio() "
-			    "calling am_send_audio_trad_mixer()", statep);
-			am_send_audio_trad_mixer(statep, apm_infop,
-			    stpptr->am_send_buf, samples);
-			ATRACE_32("am_send_audio() "
-			    "am_send_audio_trad_mixer() returning", 0);
-		} else {
-			ASSERT(mode == AM_COMPAT_MODE);
-			ATRACE("am_send_audio() "
-			    "calling am_send_audio_trad_compat()", statep);
-			am_send_audio_trad_compat(statep, apm_infop,
-			    stpptr->am_send_buf, samples);
-			ATRACE_32("am_send_audio() "
-			    "am_send_audio_trad_compat() returning", 0);
-		}
-	}
-
-}	/* am_send_audio() */
-
-/*
- * am_get_src_data()
- *
- * Description:
- *	This routine returns the PLAY or RECORD sample rate conversion
- *	data structure that is saved in the channel's private data structure.
- *
- * Arguments:
- *	srchdl_t	handle		SRC handle
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *
- * Returns:
- *	void *				Pointer to the sample rate conversion
- *					structure
- */
-void *
-am_get_src_data(srchdl_t handle, int dir)
-{
-	audio_ch_t		*chptr = AM_SRC_HDL2CHPTR(handle);
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	void			*data;
-
-	ATRACE("in am_get_src_data()", chptr);
-	ATRACE("am_get_src_data() chpptr", chpptr);
-	ATRACE_32("am_get_src_data() dir", dir);
-
-	if (dir == AUDIO_PLAY) {
-		ATRACE("am_get_src_data() PLAY returning",
-		    chpptr->acp_play_src_data);
-		data = chpptr->acp_play_src_data;
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		ATRACE("am_get_src_data() CAPTURE returning",
-		    chpptr->acp_rec_src_data);
-		data = chpptr->acp_rec_src_data;
-	}
-
-	return (data);
-
-}	/* am_get_src_data() */
-
-/*
- * am_set_src_data()
- *
- * Description:
- *	This routine sets the PLAY or RECORD sample rate conversion
- *	data structure pointer with the pointer passed in.
- *
- * Arguments:
- *	srchdl_t	handle		SRC handle
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	void		*data		The sample rate conversion data
- *
- * Returns:
- *	void
- */
-void
-am_set_src_data(srchdl_t handle, int dir, void *data)
-{
-	audio_ch_t		*chptr = AM_SRC_HDL2CHPTR(handle);
-	am_ch_private_t		*chpptr = chptr->ch_private;
-
-	ATRACE("in am_set_src_data()", chptr);
-	ATRACE_32("am_set_src_data() dir", dir);
-	ATRACE("am_set_src_data() data", data);
-
-	if (dir == AUDIO_PLAY) {
-		ATRACE("am_set_src_data() setting PLAY", data);
-		chpptr->acp_play_src_data = data;
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		ATRACE("am_set_src_data() setting CAPTURE", data);
-		chpptr->acp_rec_src_data = data;
-	}
-
-}	/* am_set_src_data() */
-
-/*
- * Private utilities used by this and other audio mixer files.
- */
-
-/*
- * am_ad_pause_play()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	We further restrict this call such that if there aren't any open
- *	play channels we don't allow this call to succeed. This implies that
- *	there has been at least one call to ad_setup() before play is started
- *	and there is still an ad_teardown() call to come. This situation is
- *	possible if only AUDIOCTL and no AUDIO channels are open and one of
- *	the AUDIOCTL channels does an AUDIO_SETINFO to unpause.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to pause
- *
- * Returns:
- *	void
- */
-void
-am_ad_pause_play(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream)
-{
-	ATRACE("in am_ad_pause_play()", statep);
-	ATRACE_32("am_ad_pause_play() stream", stream);
-	ASSERT(statep);
-
-	/* make sure we can play */
-	mutex_enter(&statep->as_lock);
-	if (stpptr->am_out_chs == 0) {
-		ATRACE_32("am_ad_pause_play() no playing channels",
-		    stpptr->am_out_chs);
-		mutex_exit(&statep->as_lock);
-		return;
-	}
-	ATRACE_32("am_ad_pause_play() playing channels", stpptr->am_out_chs);
-	mutex_exit(&statep->as_lock);
-
-	/* wait for all other calls into the audio driver to return */
-	am_serialize_ad_access(stpptr);
-
-	ATRACE("am_ad_pause_play() calling ad_pause_play()", ad_infop);
-	ad_infop->ad_entry->ad_pause_play(AUDIO_STATE2HDL(statep), stream);
-	ATRACE("am_ad_pause_play() ad_pause_play() done", ad_infop);
-
-	/* we're done, so release any waiting threads */
-	am_release_ad_access(stpptr);
-
-	ATRACE("am_ad_pause_play() returning", 0);
-
-}	/* am_ad_pause_play() */
-
-/*
- * am_ad_set_config()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to set config
- *	int		command		The configuration command
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		arg1		Command argument #1
- *	int		arg2		Command argument #2
- *	int		serialize	Serialize access to the audio driver
- *
- * Returns:
- *	AUDIO_SUCCESS			Configuration set
- *	AUDIO_FAILURE			Configuration not set
- */
-int
-am_ad_set_config(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int command, int dir, int arg1,
-	int arg2, int serialize)
-{
-	int		rc;
-
-	ATRACE("in am_ad_set_config()", statep);
-	ATRACE_32("am_ad_set_config() stream", stream);
-	ASSERT(statep);
-
-	/* wait for all other calls into the audio driver to return */
-	if (serialize) {
-		am_serialize_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_set_config() calling ad_set_config()", ad_infop);
-	rc = ad_infop->ad_entry->ad_set_config(AUDIO_STATE2HDL(statep), stream,
-	    command, dir, arg1, arg2);
-	ATRACE_32("am_ad_set_config() ad_set_config() done", rc);
-
-	/* we're done, so release any waiting threads */
-	if (serialize) {
-		am_release_ad_access(stpptr);
-	}
-
-	ATRACE_32("am_ad_set_config() returning", rc);
-
-	return (rc);
-
-}	/* am_ad_set_config() */
-
-/*
- * am_ad_set_format()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	Furthermore, if the device was active we call the proper start
- *	routine. This kicks things off if the driver had to stop things
- *	to change the format, but has a hard time restarting afterwards.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to set format
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		sample_rate	Sample rate to set
- *	int		channels	Number of channels to set
- *	int		precision	Precision to set
- *	int		encoding	Encoding to set
- *	int		serialize	Serialize access to audio driver
- *
- * Returns:
- *	AUDIO_SUCCESS			Format set
- *	AUDIO_FAILURE			Format not set
- */
-int
-am_ad_set_format(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int dir, int sample_rate,
-	int channels, int precision, int encoding, int serialize)
-{
-	audio_info_t	*hw_info = &stpptr->am_hw_info;
-	int		rc;
-
-	ATRACE("in am_ad_set_format()", statep);
-	ATRACE_32("am_ad_set_format() stream", stream);
-	ASSERT(statep);
-
-	/* wait for all other calls into the audio driver to return */
-	if (serialize) {
-		am_serialize_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_set_format() calling ad_set_format()", ad_infop);
-	rc = ad_infop->ad_entry->ad_set_format(AUDIO_STATE2HDL(statep), stream,
-	    dir, sample_rate, channels, precision, encoding);
-	ATRACE_32("am_ad_set_format() ad_set_format() done", rc);
-
-	if (dir == AUDIO_PLAY) {
-		mutex_enter(&statep->as_lock);
-		if (stpptr->am_out_chs != 0 && hw_info->play.active) {
-			mutex_exit(&statep->as_lock);
-			ATRACE("am_ad_set_format() restart play", hw_info);
-			(void) ad_infop->ad_entry->ad_start_play(
-			    AUDIO_STATE2HDL(statep), stream);
-			ATRACE("am_ad_set_format() restart play done", 0);
-		} else {
-			mutex_exit(&statep->as_lock);
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		mutex_enter(&statep->as_lock);
-		if (stpptr->am_in_chs != 0 && hw_info->record.active) {
-			mutex_exit(&statep->as_lock);
-			ATRACE("am_ad_set_format() restart record", hw_info);
-			(void) ad_infop->ad_entry->ad_start_record(
-			    AUDIO_STATE2HDL(statep), stream);
-			ATRACE("am_ad_set_format() restart record done", 0);
-		} else {
-			mutex_exit(&statep->as_lock);
-		}
-	}
-
-	/* we're done, so release any waiting threads */
-	if (serialize) {
-		am_release_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_set_format() returning", rc);
-
-	return (rc);
-
-}	/* am_ad_set_format() */
-
-/*
- * am_ad_setup()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to set format
- *	int		flags		Setup flags
- *
- * Returns:
- *	AUDIO_SUCCESS			Setup successful
- *	AUDIO_FAILURE			Setup failed
- */
-int
-am_ad_setup(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int flags)
-{
-	int		rc;
-
-	ATRACE("in am_ad_setup()", statep);
-	ATRACE_32("am_ad_setup() stream", stream);
-	ASSERT(statep);
-
-	/* if there isn't an ad_setup() routine then we just return success */
-	if (ad_infop->ad_entry->ad_setup == NULL) {
-		ATRACE("am_ad_setup() not defined, returning", statep);
-		return (AUDIO_SUCCESS);
-	}
-
-	/* wait for all other calls into the audio driver to return */
-	am_serialize_ad_access(stpptr);
-
-	ATRACE("am_ad_setup() calling ad_setup()", ad_infop);
-	rc = ad_infop->ad_entry->ad_setup(AUDIO_STATE2HDL(statep), stream,
-	    flags);
-	ATRACE_32("am_ad_setup() ad_setup() done", rc);
-
-	/* we're done, so release any waiting threads */
-	am_release_ad_access(stpptr);
-
-	ATRACE("am_ad_setup() returning", rc);
-
-	return (rc);
-
-}	/* am_ad_setup() */
-
-/*
- * am_ad_start_play()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	We further restrict this call such that if there aren't any open
- *	play channels we don't allow this call to succeed. This implies that
- *	there has been at least one call to ad_setup() before play is started
- *	and there is still an ad_teardown() call to come. This situation is
- *	possible if only AUDIOCTL and no AUDIO channels are open and one of
- *	the AUDIOCTL channels does an AUDIO_SETINFO to unpause.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to start play
- *	int		serialize	Serialize access to the audio driver
- *
- * Returns:
- *	AUDIO_SUCCESS			Start play successful
- *	AUDIO_FAILURE			Start play not successful
- */
-int
-am_ad_start_play(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int serialize)
-{
-	int		rc;
-
-	ATRACE("in am_ad_start_play()", statep);
-	ATRACE_32("am_ad_start_play() stream", stream);
-	ASSERT(statep);
-
-	/* make sure we can play */
-	mutex_enter(&statep->as_lock);
-	if (stpptr->am_out_chs == 0) {
-		ATRACE_32("am_ad_start_play() no playing channels",
-		    stpptr->am_out_chs);
-		mutex_exit(&statep->as_lock);
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_ad_start_play() playing channels", stpptr->am_out_chs);
-	mutex_exit(&statep->as_lock);
-
-	/* wait for all other calls into the audio driver to return */
-	if (serialize) {
-		am_serialize_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_start_play() calling ad_start_play()", ad_infop);
-	rc = ad_infop->ad_entry->ad_start_play(AUDIO_STATE2HDL(statep), stream);
-	ATRACE_32("am_ad_start_play() ad_start_play() done", rc);
-
-	/* we're done, so release any waiting threads */
-	if (serialize) {
-		am_release_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_start_play() returning", rc);
-
-	return (rc);
-
-}	/* am_ad_start_play() */
-
-/*
- * am_ad_start_record()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	We further restrict this call such that if there aren't any open
- *	record channels we don't allow this call to succeed. This implies that
- *	there has been at least one call to ad_setup() before record is started
- *	and there is still an ad_teardown() call to come. This situation is
- *	possible if only AUDIOCTL and no AUDIO channels are open and one of
- *	the AUDIOCTL channels does an AUDIO_SETINFO to unpause.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to start record
- *	int		serialize	Serialize access to the audio driver
- *
- * Returns:
- *	AUDIO_SUCCESS			Start record successful
- *	AUDIO_FAILURE			Start record not successful
- */
-int
-am_ad_start_record(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int serialize)
-{
-	int		rc;
-
-	ATRACE("in am_ad_start_record()", statep);
-	ATRACE_32("am_ad_start_record() stream", stream);
-	ASSERT(statep);
-
-	/* make sure we can record */
-	mutex_enter(&statep->as_lock);
-	if (stpptr->am_in_chs == 0) {
-		ATRACE_32("am_ad_start_record() no record channels",
-		    stpptr->am_in_chs);
-		mutex_exit(&statep->as_lock);
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_ad_start_record() record channels", stpptr->am_in_chs);
-	mutex_exit(&statep->as_lock);
-
-	/* wait for all other calls into the audio driver to return */
-	if (serialize) {
-		am_serialize_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_start_record() calling ad_start_record()", ad_infop);
-	rc = ad_infop->ad_entry->ad_start_record(AUDIO_STATE2HDL(statep),
-	    stream);
-	ATRACE_32("am_ad_start_record() ad_start_record() done", rc);
-
-	/* we're done, so release any waiting threads */
-	if (serialize) {
-		am_release_ad_access(stpptr);
-	}
-
-	ATRACE("am_ad_start_record() returning", rc);
-
-	return (rc);
-
-}	/* am_ad_start_record() */
-
-/*
- * am_ad_stop_play()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	We further restrict this call such that if there aren't any open
- *	play channels we don't allow this call to succeed. This implies that
- *	there has been at least one call to ad_setup() before play is started
- *	and there is still an ad_teardown() call to come. This situation is
- *	possible if only AUDIOCTL and no AUDIO channels are open and one of
- *	the AUDIOCTL channels does an AUDIO_SETINFO to unpause.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to stop playing
- *
- * Returns:
- *	void
- */
-void
-am_ad_stop_play(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream)
-{
-	ATRACE("in am_ad_stop_play()", statep);
-	ATRACE_32("am_ad_stop_play() stream", stream);
-	ASSERT(statep);
-
-	/* make sure we can play */
-	mutex_enter(&statep->as_lock);
-	if (stpptr->am_out_chs == 0) {
-		ATRACE_32("am_ad_stop_play() no playing channels",
-		    stpptr->am_out_chs);
-		mutex_exit(&statep->as_lock);
-		return;
-	}
-	ATRACE_32("am_ad_stop_play() playing channels", stpptr->am_out_chs);
-	mutex_exit(&statep->as_lock);
-
-	/* wait for all other calls into the audio driver to return */
-	am_serialize_ad_access(stpptr);
-
-	ATRACE("am_ad_stop_play() calling ad_stop_play()", ad_infop);
-	ad_infop->ad_entry->ad_stop_play(AUDIO_STATE2HDL(statep), stream);
-	ATRACE_32("am_ad_stop_play() ad_stop_play() done", stream);
-
-	/* we're done, so release any waiting threads */
-	am_release_ad_access(stpptr);
-
-	ATRACE("am_ad_stop_play() returning", stream);
-
-}	/* am_ad_stop_play() */
-
-/*
- * am_ad_stop_record()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- *	We further restrict this call such that if there aren't any open
- *	record channels we don't allow this call to succeed. This implies that
- *	there has been at least one call to ad_setup() before record is started
- *	and there is still an ad_teardown() call to come. This situation is
- *	possible if only AUDIOCTL and no AUDIO channels are open and one of
- *	the AUDIOCTL channels does an AUDIO_SETINFO to unpause.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to stop recording
- *
- * Returns:
- *	void
- */
-void
-am_ad_stop_record(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream)
-{
-	ATRACE("in am_ad_stop_record()", statep);
-	ATRACE_32("am_ad_stop_record() stream", stream);
-	ASSERT(statep);
-
-	/* make sure we can record */
-	mutex_enter(&statep->as_lock);
-	if (stpptr->am_in_chs == 0) {
-		ATRACE_32("am_ad_stop_record() no record channels",
-		    stpptr->am_in_chs);
-		mutex_exit(&statep->as_lock);
-		return;
-	}
-	ATRACE_32("am_ad_stop_record() record channels", stpptr->am_in_chs);
-	mutex_exit(&statep->as_lock);
-
-	/* wait for all other calls into the audio driver to return */
-	am_serialize_ad_access(stpptr);
-
-	ATRACE("am_ad_stop_record() calling ad_stop_record()", ad_infop);
-	ad_infop->ad_entry->ad_stop_record(AUDIO_STATE2HDL(statep), stream);
-	ATRACE_32("am_ad_stop_record() ad_stop_record() done", stream);
-
-	/* we're done, so release any waiting threads */
-	am_release_ad_access(stpptr);
-
-	ATRACE("am_ad_stop_record() returning", stream);
-
-}	/* am_ad_stop_record() */
-
-/*
- * am_ad_teardown()
- *
- * Description:
- *	The official way to call into the audio driver. This uses a CV
- *	to ensure that only one call into the audio driver can be made
- *	at any one time. Thus the audio driver doesn't have to worry about
- *	serialization of the entry points.
- *
- * Arguments:
- *	audio_state_t	*statep		Ptr to the dev instance's state
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	int		stream		Which stream to set format
- *	int		dir		AUDIO_PLAY and/or AUDIO_RECORD
- *
- * Returns:
- *	void
- */
-void
-am_ad_teardown(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int dir)
-{
-	ATRACE("in am_ad_teardown()", statep);
-	ATRACE_32("am_ad_teardown() stream", stream);
-	ASSERT(statep);
-
-	/* if there isn't an ad_teardown() routine then just return success */
-	if (ad_infop->ad_entry->ad_teardown == NULL) {
-		ATRACE("am_ad_teardown() not defined, returning", statep);
-		return;
-	}
-
-	/* wait for all other calls into the audio driver to return */
-	am_serialize_ad_access(stpptr);
-
-	ATRACE("am_ad_teardown() calling ad_teardown()", ad_infop);
-	ad_infop->ad_entry->ad_teardown(AUDIO_STATE2HDL(statep), stream, dir);
-	ATRACE_32("am_ad_teardown() ad_teardown() done", stream);
-
-	/* we're done, so release any waiting threads */
-	am_release_ad_access(stpptr);
-
-	ATRACE("am_ad_teardown() returning", stream);
-
-}	/* am_ad_teardown() */
-
-/*
- * am_ck_channels()
- *
- * Description:
- *	This routine checks to see if the number of channels passed is one of
- *	the supported number of channels. If hw is set to B_TRUE then we check
- *	against what the hardware can do. If it is set to B_FALSE then we check
- *	against what we know how to translate.
- *
- * Arguments:
- *	am_ad_ch_cap_t	*cptr	Pointer to the play/record capability struct
- *	uint_t		ch	Number of channels to check
- *	boolean_t	hw	If B_TRUE report the true H/W capability
- *
- * Returns:
- *	AUDIO_SCCESS		Valid number of channels
- *	AUDIO_FAILURE		Invalid number of channels
- */
-int
-am_ck_channels(am_ad_ch_cap_t *cptr, uint_t ch, boolean_t hw)
-{
-	uint_t		*iptr = cptr->ad_chs;
-	int		i;
-
-	ATRACE("in am_ck_channels()", cptr);
-
-	if (hw) {
-		/* check against the hardware */
-		for (i = 0; *iptr != 0; i++, iptr++) {
-			if (*iptr == ch) {
-				ATRACE("am_ck_channels() hw true succeeded",
-				    iptr);
-				return (AUDIO_SUCCESS);
-			}
-		}
-		ATRACE("am_ck_channels() hw true failed", cptr->ad_chs);
-		return (AUDIO_FAILURE);
-	}
-
-	ASSERT(hw == B_FALSE);
-
-	/* check against all legal number of channels */
-	if (ch == AUDIO_CHANNELS_MONO || ch == AUDIO_CHANNELS_STEREO) {
-		ATRACE_32("am_ck_channels() hw false succeeded", ch);
-		return (AUDIO_SUCCESS);
-	}
-
-	ATRACE_32("am_ck_channels() hw false failed", hw);
-	return (AUDIO_FAILURE);
-
-}	/* am_ck_channels */
-
-/*
- * am_ck_combinations()
- *
- * Description:
- *	This routine makes sure that the combination of encoding and
- *	precision are legal. If hw is set to B_TRUE then we check against
- *	what the hardware can do. If it is set to B_FALSE then we check
- *	against what we know how to translate. We can translate between all
- *	supported encoding and precision combinations.
- *
- * Arguments:
- *	am_ad_cap_comb_t	*comb	Ptr to the play/rec legal combinations
- *	int			enc	The encoding to check
- *	int			prec	The precision to check
- *	boolean_t		hw	If B_TRUE report the true H/W capability
- *
- * Returns:
- *	AUDIO_SUCCESS		It is a legal combination or value
- *	AUDIO_FAILURE		It is not a legal combination or value
- */
-int
-am_ck_combinations(am_ad_cap_comb_t *comb, int enc, int prec, boolean_t hw)
-{
-	am_ad_cap_comb_t	*ptr;
-
-	ATRACE("in am_ck_combinations()", comb);
-	ATRACE("am_ck_combinations() enc", enc);
-	ATRACE("am_ck_combinations() prec", prec);
-
-	if (hw) {
-		/* check against the hardware */
-		for (ptr = comb; ptr->ad_prec != 0; ptr++) {
-			ATRACE_32("am_ck_combinations() enc", ptr->ad_enc);
-			ATRACE_32("am_ck_combinations() prec", ptr->ad_prec);
-			if (ptr->ad_prec == prec && ptr->ad_enc == enc) {
-				ATRACE("am_ck_combinations() "
-				    "found a legal combination", ptr);
-				return (AUDIO_SUCCESS);
-			}
-		}
-		ATRACE("am_ck_combinations() not in combination array", 0);
-		return (AUDIO_FAILURE);
-	}
-
-	ASSERT(hw == B_FALSE);
-
-	/* check against all legal combinations */
-	switch (prec) {
-	case AUDIO_PRECISION_16:
-		/* the only thing it can be is PCM */
-		switch (enc) {
-		case AUDIO_ENCODING_LINEAR:
-			return (AUDIO_SUCCESS);
-		default:
-			return (AUDIO_FAILURE);
-		}
-	case AUDIO_PRECISION_8:
-		/* we have more choices for 8-bit */
-		switch (enc) {
-		case AUDIO_ENCODING_LINEAR8:
-			/*FALLTHROUGH*/
-		case AUDIO_ENCODING_LINEAR:
-			/*FALLTHROUGH*/
-		case AUDIO_ENCODING_ULAW:
-			/*FALLTHROUGH*/
-		case AUDIO_ENCODING_ALAW:
-			return (AUDIO_SUCCESS);
-		default:
-			return (AUDIO_FAILURE);
-		}
-	default:
-		ATRACE_32("am_ck_combinations() illegal precision", prec);
-		return (AUDIO_FAILURE);
-	}
-
-}	/* am_ck_combinations() */
-
-/*
- * am_ck_sample_rate()
- *
- * Description:
- *	The sample rate information list is searched for the sample rate.
- *
- * Arguments:
- *	am_ad_ch_cap_t	*cptr	Pointer to the play/record capability struct
- *	int		mode	AM_MIXER_MODE or AM_COMPAT_MODE
- *	int		sr	Sample rate to check
- *
- * Returns:
- *	AUDIO_SUCCESS		Sample rate found
- *	AUDIO_FAILURE		Invalid sample rate
- */
-int
-am_ck_sample_rate(am_ad_ch_cap_t *cptr, int mode, int sr)
-{
-	am_ad_sample_rates_t	*srs;
-	uint_t			*ptr;
-
-	ATRACE("in am_ck_sample_rate()", cptr);
-	ATRACE_32("am_ck_sample_rate() mode", mode);
-	ATRACE_32("am_ck_sample_rate() sample rate", sr);
-
-	if (mode == AM_MIXER_MODE) {
-		srs = &cptr->ad_mixer_srs;
-	} else {
-		srs = &cptr->ad_compat_srs;
-	}
-	ptr = srs->ad_srs;
-
-	/* check the passed in sample rate against the list */
-	if (srs->ad_limits & MIXER_SRS_FLAG_SR_LIMITS) {
-		/*
-		 * We only check the limits and because we've already done the
-		 * sanity check in am_attach(). Therefore position 0 must be
-		 * the bottom limit and position 1 must be the top limit.
-		 */
-		ATRACE_32("am_ck_sample_rate() limits, min sr", ptr[0]);
-		ATRACE_32("am_ck_sample_rate() limits, max sr", ptr[1]);
-		if (sr < ptr[0] || sr > ptr[1]) {
-			ATRACE("am_ck_sample_rate() limit failed", srs);
-			return (AUDIO_FAILURE);
-		}
-
-		ATRACE_32("am_ck_sample_rate() found in limit", sr);
-		return (AUDIO_SUCCESS);
-	}
-
-	for (; *ptr != NULL; ptr++) {
-		ATRACE_32("am_ck_sample_rate() not limits, test sr", *ptr);
-		if (*ptr == sr) {
-			ATRACE_32("am_ck_sample_rate() found", sr);
-			return (AUDIO_SUCCESS);
-		}
-		if (*ptr > sr) {
-			ATRACE_32("am_ck_sample_rate() past", sr);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	ATRACE("am_ck_sample_rate() failed", cptr->ad_sr_info);
-
-	return (AUDIO_FAILURE);
-
-}	/* am_ck_sample_rate */
-
-/*
- * am_safe_putnext()
- *
- * Description:
- *	The usual reason for holding a lock across putnext() is we don't
- *	want a channel to call qprocsoff() while we are building a STREAMS
- *	message. This can happen because we have many different threads,
- *	including kernel threads, that are running at the same time.
- *
- *	If the channel has called qprocsoff() then we just return.
- *
- *	CAUTION: This is only for the read queue.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	mblk_t		*mp		Pointer to the STREAMS message to send
- *
- * Returns:
- *	void
- */
-void
-am_safe_putnext(audio_ch_t *chptr, mblk_t *mp)
-{
-	am_ch_private_t		*chpptr = chptr->ch_private;
-
-	ATRACE("in am_safe_putnext()", chptr);
-
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-	ASSERT(!MUTEX_HELD(&chptr->ch_statep->as_lock));
-
-	/* make sure qprocsoff() hasn't been called */
-	if (chpptr->acp_flags & AM_CHNL_QPROCSOFF) {
-		/* it has, so cleanup and return */
-		freemsg(mp);
-
-		ATRACE("am_safe_putnext() msg not sent, qprocsoff()", chptr);
-
-		return;
-	}
-
-	/* increment the number of putnext() calls outstanding */
-	chpptr->acp_busy_cnt++;
-	ATRACE_32("am_safe_putnext() inc acp_busy_cnt", chpptr->acp_busy_cnt);
-
-	/* it's okay to release the lock and do the putnext() */
-	mutex_exit(&chptr->ch_lock);
-
-	putnext(RD(chptr->ch_qptr), mp);
-	ATRACE("am_safe_putnext() putnext() done", chptr);
-
-	/* now we can reaquire the lock and check to see if need to signal */
-	mutex_enter(&chptr->ch_lock);
-
-	chpptr->acp_busy_cnt--;
-	ATRACE_32("am_safe_putnext() dec acp_busy_cnt", chpptr->acp_busy_cnt);
-
-	if ((chpptr->acp_busy_cnt == 0) &&
-	    (chpptr->acp_flags & AM_CHNL_SIGNAL_NEEDED)) {
-		ATRACE("am_safe_putnext() sending cv_signal()", chptr);
-		cv_signal(&chptr->ch_cv);
-	}
-
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-	ASSERT(!MUTEX_HELD(&chptr->ch_statep->as_lock));
-
-}	/* am_safe_putnext() */
-
-/*
- * am_test_canputnext()
- *
- * Description:
- *	The usual reason for holding a lock across putnext() is we don't
- *	want a channel to call qprocsoff() while we are building a STREAMS
- *	message. This can happen because we have many different threads,
- *	including kernel threads, that are running at the same time.
- *
- *	If the channel has called qprocsoff() then we just return.
- *
- *	CAUTION: This is only for the read queue.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *
- * Returns:
- *	AUDIO_SUCCESS		There is room in the next module's queue
- *	AUDIO_FAILURE		There is no room in the next module's queue
- *	AM_CHNL_CLOSED		The channel has been closed
- */
-int
-am_test_canputnext(audio_ch_t *chptr)
-{
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	int			room;
-
-	ATRACE("in am_test_canputnext()", chptr);
-
-	ASSERT(!MUTEX_HELD(&chptr->ch_lock));
-	mutex_enter(&chptr->ch_lock);
-
-	/* make sure qprocsoff() hasn't been called */
-	if (chpptr->acp_flags & AM_CHNL_QPROCSOFF) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_test_canputnext() channel closed", chptr);
-		return (AM_CHNL_CLOSED);
-	}
-
-	/*
-	 * Increment the number of putnext() calls outstanding,
-	 * treat canputnext() like putnext().
-	 */
-	chpptr->acp_busy_cnt++;
-	ATRACE_32("am_test_canputnext() inc acp_busy_cnt",
-	    chpptr->acp_busy_cnt);
-
-	mutex_exit(&chptr->ch_lock);
-
-	room = canputnext(RD(chptr->ch_qptr));
-
-	mutex_enter(&chptr->ch_lock);
-	chpptr->acp_busy_cnt--;
-
-	if ((chpptr->acp_busy_cnt == 0) &&
-	    (chpptr->acp_flags & AM_CHNL_SIGNAL_NEEDED)) {
-		ATRACE("am_test_canputnext() sending cv_signal()", chptr);
-		cv_signal(&chptr->ch_cv);
-	}
-	mutex_exit(&chptr->ch_lock);
-
-	ASSERT(!MUTEX_HELD(&chptr->ch_lock));
-
-	if (!room) {
-		ATRACE("am_test_canputnext() returning failure", chptr);
-		return (AUDIO_FAILURE);
-	} else {
-		ATRACE("am_test_canputnext() returning success", chptr);
-		return (AUDIO_SUCCESS);
-	}
-
-}	/* am_test_canputnext() */
-
-
-/*
- * Private utilities used only by this file.
- */
-
-/*
- * am_attach_check()
- *
- * Description:
- *	Verify all of the read/write parameters for the mixer. This includes
- *	making sure the format configuration is legal. We also figure out all
- *	the channel, encoding, and precision specifications and set flags. This
- *	makes things a lot faster later on.
- *
- *	NOTE: We don't set the am_hw_* members here so they'll still be set
- *		to 0 when we return. Thus the first time am_set_format() is
- *		called the hardware will be forced to be programmed.
- *
- * Arguments:
- *	am_ad_info_t	*ad_infop	Ptr to the device's capabilities struct
- *	am_apm_private_t *stpptr	Pointer to private APM data
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *
- * Returns:
- *	AUDIO_SUCCESS			The checks succeed
- *	AUDIO_FAILURE			The checks failed, device can't load
- */
-static int
-am_attach_check(audio_state_t *statep, am_ad_info_t *ad_infop,
-	am_apm_private_t *stpptr, int dir)
-{
-	am_ad_ch_cap_t		*cptr = &ad_infop->ad_record;
-	am_ad_cap_comb_t	*comb;
-	audio_info_t		*hw_info = &stpptr->am_hw_info;
-	audio_prinfo_t		*prinfo;
-	uint_t			flags = 0;
-	uint_t			mch = 0;	/* mixer channels */
-	uint_t			menc = 0;	/* mixer encoding */
-	uint_t			mprec = 0;	/* mixer precision */
-	int			csr;		/* compat sample rate */
-	int			mode;
-
-	ATRACE("in am_attach_check()", ad_infop);
-	ASSERT(statep);
-
-	if (dir == AUDIO_PLAY) {
-		ATRACE("am_attach_check() dir PLAY", 0);
-		cptr = &ad_infop->ad_play;
-		comb = ad_infop->ad_play_comb;
-		prinfo = &ad_infop->ad_defaults->play;
-	} else if (dir == AUDIO_RECORD) {
-		ATRACE("am_attach_check() dir RECORD", 0);
-		cptr = &ad_infop->ad_record;
-		comb = ad_infop->ad_rec_comb;
-		prinfo = &ad_infop->ad_defaults->record;
-	} else {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() illegal direction: %d", dir);
-		return (AUDIO_FAILURE);
-	}
-
-	mode = stpptr->am_pstate->apm_mode;
-
-	/*
-	 * Check the Audio Driver capabilities for good sample rate info. For
-	 * MIXER mode we get the default H/W sample rate. For COMPAT mode we
-	 * verify that the default is okay.
-	 *
-	 * Sanity check the record COMPAT mode sample rates.
-	 */
-	if ((csr = am_ck_sample_rate_sanity(cptr, AM_COMPAT_MODE)) ==
-	    AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad COMPAT mode sample rate list");
-		return (AUDIO_FAILURE);
-	}
-
-	/* make sure the default record SR for COMPAT mode is good */
-	if (am_ck_sample_rate(cptr, AM_COMPAT_MODE, prinfo->sample_rate) ==
-	    AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad COMPAT sample rate: %d",
-		    prinfo->sample_rate);
-		return (AUDIO_FAILURE);
-	}
-
-	/* sanity check & find the best record MIXER mode sample rate */
-	if (am_ck_sample_rate_sanity(cptr, AM_MIXER_MODE) == AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad MIXER mode sample rate list");
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_attach_check() COMPAT sample rate", prinfo->sample_rate);
-	ATRACE_32("am_attach_check() COMPAT max sample rate", csr);
-
-	/*
-	 * Figure out how many channel combinations we support. And also figure
-	 * out the best setting for MIXER mode. We try for stereo and then
-	 * settle for mono.
-	 */
-	if (am_ck_channels(cptr, AUDIO_CHANNELS_STEREO, B_TRUE) ==
-	    AUDIO_SUCCESS) {
-		flags |= AM_PRIV_CH_STEREO;
-		mch = AUDIO_CHANNELS_STEREO;
-	}
-	if (am_ck_channels(cptr, AUDIO_CHANNELS_MONO, B_TRUE) ==
-	    AUDIO_SUCCESS) {
-		flags |= AM_PRIV_CH_MONO;
-		if (mch == 0) {
-			mch = AUDIO_CHANNELS_MONO;
-		}
-	}
-	ATRACE_32("am_attach_check() MIXER record channels", mch);
-
-	/* make sure we have at least one of these set */
-	if (!(flags & (AM_PRIV_CH_STEREO|AM_PRIV_CH_MONO))) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad channels list");
-		return (AUDIO_FAILURE);
-	}
-
-	/* make sure the COMPAT mode default channel is legal */
-	if (prinfo->channels != AUDIO_CHANNELS_MONO &&
-	    prinfo->channels != AUDIO_CHANNELS_STEREO) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad COMPAT record channels list");
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_attach_check() COMPAT channels", prinfo->channels);
-
-	/*
-	 * Now check all possible encoding and precision combinations and
-	 * set flags to make translations easier later on. We also figure
-	 * out the best settings for the mixer. We prefer 16-bit linear PCM,
-	 * but we'll take whatever we can get.
-	 */
-	if (am_ck_combinations(comb, AUDIO_ENCODING_LINEAR, AUDIO_PRECISION_16,
-	    B_TRUE) == AUDIO_SUCCESS) {
-		flags |= AM_PRIV_16_PCM;
-		mprec = AUDIO_PRECISION_16;
-		menc = AUDIO_ENCODING_LINEAR;
-	}
-	if (am_ck_combinations(comb, AUDIO_ENCODING_LINEAR, AUDIO_PRECISION_8,
-	    B_TRUE) == AUDIO_SUCCESS) {
-		flags |= AM_PRIV_8_PCM;
-		if (mprec == 0) {
-			mprec = AUDIO_PRECISION_8;
-			menc = AUDIO_ENCODING_LINEAR;
-		}
-	}
-	if (am_ck_combinations(comb, AUDIO_ENCODING_LINEAR8, AUDIO_PRECISION_8,
-	    B_TRUE) == AUDIO_SUCCESS) {
-		flags |= AM_PRIV_8_PCM;
-		if (mprec == 0) {
-			mprec = AUDIO_PRECISION_8;
-			menc = AUDIO_ENCODING_LINEAR8;
-		}
-	}
-	if (am_ck_combinations(comb, AUDIO_ENCODING_ULAW, AUDIO_PRECISION_8,
-	    B_TRUE) == AUDIO_SUCCESS) {
-		flags |= AM_PRIV_8_ULAW;
-		if (mprec == 0) {
-			mprec = AUDIO_PRECISION_8;
-			menc = AUDIO_ENCODING_ULAW;
-		}
-	}
-	if (am_ck_combinations(comb, AUDIO_ENCODING_ALAW, AUDIO_PRECISION_8,
-	    B_TRUE) == AUDIO_SUCCESS) {
-		flags |= AM_PRIV_8_ALAW;
-		if (mprec == 0) {
-			mprec = AUDIO_PRECISION_8;
-			menc = AUDIO_ENCODING_ALAW;
-		}
-	}
-	ATRACE_32("am_attach_check() MIXER precision", mprec);
-	ATRACE_32("am_attach_check() MIXER encoding", menc);
-
-	/* make sure we got the mixer settings, if we didn't then bad list */
-	if (mprec == 0) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad combinations list");
-		return (AUDIO_FAILURE);
-	}
-
-	/* now double check a default COMPAT mode configuration */
-	if (am_ck_combinations(comb, prinfo->encoding, prinfo->precision,
-	    B_FALSE) == AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "attach_check() bad COMPAT combinations defaults");
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_attach_check() COMPAT precision", prinfo->precision);
-	ATRACE_32("am_attach_check() COMPAT encoding", prinfo->encoding);
-
-	/*
-	 * Initialize the Codec's state. The sample rate is always set to
-	 * 0 which forces am_set_format() to program the Codec the first
-	 * time it is called for play and record.
-	 */
-	if (mode == AM_MIXER_MODE && ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		if (dir == AUDIO_PLAY) {
-			hw_info->play.sample_rate = 0;
-			hw_info->play.channels = mch;
-			hw_info->play.precision = mprec;
-			hw_info->play.encoding = menc;
-		} else {
-			hw_info->record.sample_rate = 0;
-			hw_info->record.channels = mch;
-			hw_info->record.precision = mprec;
-			hw_info->record.encoding = menc;
-		}
-	} else {
-		if (dir == AUDIO_PLAY) {
-			hw_info->play.sample_rate = 0;
-			hw_info->play.channels = prinfo->channels;
-			hw_info->play.precision = prinfo->precision;
-			hw_info->play.encoding = prinfo->encoding;
-		} else {
-			hw_info->record.sample_rate = 0;
-			hw_info->record.channels = prinfo->channels;
-			hw_info->record.precision = prinfo->precision;
-			hw_info->record.encoding = prinfo->encoding;
-		}
-	}
-
-	/* initialize the saved state for when we go from COMPAT->MIXER mode */
-	if (dir == AUDIO_PLAY) {
-		/* once these get set they are set the same for life */
-		stpptr->am_pflags |= flags;
-		stpptr->am_save_psr = csr;
-		stpptr->am_hw_pchs = mch;
-		stpptr->am_hw_pprec = mprec;
-		stpptr->am_hw_penc = menc;
-	} else {
-		/* once these get set they are set the same for life */
-		stpptr->am_rflags |= flags;
-		stpptr->am_save_rsr = csr;
-		stpptr->am_hw_rchs = mch;
-		stpptr->am_hw_rprec = mprec;
-		stpptr->am_hw_renc = menc;
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_attach_check() */
-
-/*
- * am_ck_sample_rate_sanity()
- *
- * Description:
- *	Sanity check the sample rate list to make sure it is good. That
- *	means it is ordered from smallest to largest. The largest sample
- *	rate is returned if the list is okay.
- *	AUDIO_FAILURE is returned.
- *
- *	NOTE: The sample rate conversion information is specific to the routines
- *		that perform the conversions. Therefore the audio mixer doesn't
- *		check them for sanity. It could be done by creating another
- *		call in am_ad_src_entry_t, but thorough testing is all that
- *		is really needed.
- *
- * Arguments:
- *	am_ad_ch_cap_t	*cptr	Pointer to the play/record capability struct
- *	int		mode	AM_MIXER_MODE or AM_COMPAT_MODE
- *
- * Returns:
- *	AUDIO_SUCCESS		Sample rate list okay
- *	AUDIO_FAILURE		Bad sample rate list
- */
-static int
-am_ck_sample_rate_sanity(am_ad_ch_cap_t *cptr, int mode)
-{
-	am_ad_sample_rates_t	*srs;
-	uint_t			*ptr;
-	uint_t			big = 0;
-	int			i;
-
-	ATRACE("in am_ck_sample_rate_sanity()", cptr);
-	ATRACE_32("am_ck_sample_rate_sanity() mode", mode);
-
-	if (mode == AM_MIXER_MODE) {
-		srs = &cptr->ad_mixer_srs;
-	} else {
-		srs = &cptr->ad_compat_srs;
-	}
-	ptr = srs->ad_srs;
-
-
-	/* do a sanity check on the list, it must be in increasing order */
-	for (i = 0; *ptr != NULL; ptr++, i++) {
-		if (*ptr > big) {
-			big = *ptr;
-		} else {
-			ATRACE_32("am_ck_sample_rate_sanity() bad order, big",
-			    big);
-			ATRACE_32("am_ck_sample_rate_sanity() *ptr", *ptr);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	/* if limits then there should be only two samples */
-	if ((srs->ad_limits & MIXER_SRS_FLAG_SR_LIMITS) && i != 2) {
-		ATRACE_32("am_ck_sample_rate_sanity() "
-		    "too many samples for limits", i);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE_32("am_ck_sample_rate_sanity() found highest sample rate", big);
-
-	return (big);
-
-}	/* am_ck_sample_rate_sanity() */
-
-/*
- * am_convert_to_format()
- *
- * Description:
- *	This routine takes the source buffer, which is 32-bit integers,
- *	and converts it to whatever format the destination buffer is.
- *	While the source buffer is a 32-bit buffer, the data is really
- *	16-bits. We use 32-bits to make sure we can sum audio streams
- *	without loosing bits. When this routine is called we clip and
- *	then convert the data. This includes doing any required translations.
- *
- *	Other routines deal with the channel <--> channel conversion.
- *	In addition the audio is already sample rate converted, if it is
- *	needed. If the result needs to be unsigned PCM then we do that
- *	conversion.
- *
- *	The supported conversions are:
- *		32-bit signed linear	->	16-bit clipped signed linear
- *		32-bit signed linear	->	8-bit u-law
- *		32-bit signed linear	->	8-bit A-law
- *		32-bit signed linear	->	8-bit signed linear
- *
- *	We also support translating to unsigned PCM for 8-bit and 16-bit linear.
- *
- * Arguments:
- *	int		*src		Ptr to the src buffer, data to convert
- *	void		*dest		Ptr to the dest buffer, for converted
- *					data
- *	int		samples		The number of samples to convert
- *	int		precision	The precision of the output buffer
- *	int		encoding	The encoding of the output buffer
- *	int		flags		Flags, including AM_PRIV_8/16_TRANS
- *
- * Returns:
- *	none
- */
-static void
-am_convert_to_format(int *src, void *dest, int samples, int precision,
-	int encoding, int flags)
-{
-	int		val;
-
-	ATRACE_32("in am_convert_to_format()", samples);
-	ATRACE_32("am_convert_to_format() precision", precision);
-	ATRACE_32("am_convert_to_format() encoding", encoding);
-
-	/* this should have been stopped earlier */
-	ASSERT(precision != AUDIO_PRECISION_16 ||
-	    encoding == AUDIO_ENCODING_LINEAR);
-
-	/* make sure we have work to do */
-	if (samples == 0) {
-		ATRACE("am_convert_to_format() "
-		    "no samples to convert, returning", 0);
-		return;
-	}
-
-	ATRACE_32("am_convert_to_format() NON-Linearize Audio", samples);
-
-	if (precision == AUDIO_PRECISION_16) {
-		int16_t	*dptr = (int16_t *)dest;
-
-		ASSERT(encoding == AUDIO_ENCODING_LINEAR);
-
-		if (flags & AM_PRIV_16_TRANS) {
-			/* convert for unsigned int hardware */
-			for (; samples--; ) {
-				val = *src++;
-				if (val > INT16_MAX) {
-					val = INT16_MAX;
-				} else if (val < INT16_MIN) {
-					val  = INT16_MIN;
-				}
-				val += -INT16_MIN;
-				*dptr++ = (int16_t)val;
-			}
-		} else {
-			/* hardware is signed */
-			for (; samples--; ) {
-				val = *src++;
-				if (val > INT16_MAX) {
-					*dptr++ = INT16_MAX;
-				} else if (val < INT16_MIN) {
-					*dptr++ = INT16_MIN;
-				} else {
-					*dptr++ = (int16_t)val;
-				}
-			}
-		}
-		ATRACE("am_convert_to_format() 16-bit linear done", 0);
-	} else {	/* end 16-bit, begin 8-bit */
-		uint8_t		*dptr = (uint8_t *)dest;
-
-		ASSERT(precision == AUDIO_PRECISION_8);
-
-		if (encoding == AUDIO_ENCODING_LINEAR8 ||
-		    (flags & AM_PRIV_8_TRANS)) {
-			ATRACE_32(
-			    "am_convert_to_format() 8-bit unsigned linear",
-			    samples);
-
-			for (; samples--; ) {
-				val = *src++;
-				if (val > INT16_MAX) {
-					val = INT16_MAX;
-				} else if (val < INT16_MIN) {
-					val = INT16_MIN;
-				}
-				*dptr++ = (uint8_t)((val >> AM_256_SHIFT) -
-				    INT8_MAX);
-			}
-		} else if (encoding == AUDIO_ENCODING_LINEAR) {
-			ATRACE_32("am_convert_to_format() 8-bit signed linear",
-			    samples);
-
-			for (; samples--; ) {
-				val = *src++;
-				if (val > INT16_MAX) {
-					val = INT16_MAX;
-				} else if (val < INT16_MIN) {
-					val = INT16_MIN;
-				}
-				*dptr++ = (uint8_t)(val >> AM_256_SHIFT);
-			}
-		} else {	/* 8-bit U/A-Law */
-			uint8_t		*cptr;
-			int		shift;
-
-			if (encoding == AUDIO_ENCODING_ULAW) {
-				ATRACE("am_convert_to_format() 8-bit U-Law", 0);
-				cptr = &_14linear2ulaw8[G711_ULAW_MIDPOINT];
-				shift = 2;
-			} else {
-				ATRACE("am_convert_to_format() 8-bit A-Law", 0);
-				ASSERT(encoding == AUDIO_ENCODING_ALAW);
-				cptr = &_13linear2alaw8[G711_ALAW_MIDPOINT];
-				shift = 3;
-			}
-
-			ATRACE_32("am_convert_to_format() 8-bit U/A-Law",
-			    encoding);
-
-			for (; samples--; ) {
-				val = *src++;
-				if (val > INT16_MAX) {
-					val = INT16_MAX;
-				} else if (val < INT16_MIN) {
-					val = INT16_MIN;
-				}
-				*dptr++ = (uint8_t)cptr[val >> shift];
-			}
-		}
-		ATRACE("am_convert_to_format() 8-bit done", 0);
-	}
-
-	ATRACE_32("am_convert_to_format() done", encoding);
-
-}	/* am_convert_to_format() */
-
-/*
- * am_get_audio_multi()
- *
- * Description:
- *	This routine is used by multi-channel Codecs to get a single stream
- *	of audio data for an individual channel.
- *
- * Arguments:
- *	audio_state_t	*statep		Pointer to the device instance's state
- *	void		*buf		The buffer to place the audio into
- *	int		channel		The device channel number.
- *	int		samples		The number of samples to get
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		wants. So it is samples at the hardware's sample rate.
- *
- * Returns:
- *	>= 0			The number of samples transferred to the buffer
- *	AUDIO_FAILURE		An error has occurred
- */
-static int
-am_get_audio_multi(audio_state_t *statep, void *buf, int channel, int samples)
-{
-	audio_ch_t		*chptr = &statep->as_channels[channel];
-	am_ch_private_t		*chpptr;
-	audio_info_t		*info;
-	int			hw_channels;
-	int			ret_val;
-
-	ATRACE("in am_get_audio_multi()", statep);
-
-	ASSERT(statep);
-	ASSERT(chptr->ch_info.ch_number == channel);
-
-	/* lock the channel before we check it out */
-	mutex_enter(&chptr->ch_lock);
-
-	/*
-	 * The channel may have been closed while we waited on the mutex.
-	 * So once we get it we make sure the channel is still valid. We also
-	 * make sure it's an AUDIO channel.
-	 */
-	chpptr = chptr->ch_private;
-	info = chptr->ch_info.info;
-
-	if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    chptr->ch_info.pid == 0 || chptr->ch_info.dev_type != AUDIO ||
-	    chpptr == NULL || ((chpptr->acp_flags & AM_CHNL_OPEN) == 0) ||
-	    (chpptr->acp_flags & AM_CHNL_QPROCSOFF) || info == NULL) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_get_audio_multi() channel closed", chptr);
-		return (AUDIO_FAILURE);
-	}
-
-	/* skip if the channel is paused */
-	if (info->play.pause) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_get_audio_multi() channel paused", statep);
-		return (0);
-	}
-
-	/* make sure the number of samples is module the # of channels */
-	hw_channels = info->play.channels;
-	if (hw_channels != AUDIO_CHANNELS_MONO &&
-	    (samples % hw_channels) != 0) {
-		ATRACE_32("am_get_audio_multi() bad sample size", samples);
-		samples -= samples % hw_channels;
-	}
-
-	/* get "samples" worth of data */
-	ATRACE("am_get_audio_multi() calling am_get_samples()", chptr);
-	chpptr->acp_busy_cnt++;
-	ret_val = am_get_samples(chptr, samples, buf, AM_COMPAT_MODE);
-	chpptr->acp_busy_cnt--;
-	if (ret_val == AUDIO_FAILURE || ret_val == 0) {
-		am_audio_drained(chptr);
-		ATRACE_32("am_get_audio_multi() am_get_samples() failed",
-		    ret_val);
-		goto done;
-	}
-
-	ATRACE_32("am_get_audio_multi() am_get_samples() succeeded", ret_val);
-
-	/* XXX we need to convert to the hardware format */
-
-done:
-	if ((chpptr->acp_busy_cnt == 0) &&
-	    (chpptr->acp_flags & AM_CHNL_SIGNAL_NEEDED)) {
-		ATRACE("am_get_audio_multi() sending cv_signal()", chptr);
-		cv_signal(&chptr->ch_cv);
-	}
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_get_audio_multi() done", buf);
-
-	return (ret_val);
-
-}	/* am_get_audio_multi() */
-
-/*
- * am_get_audio_trad_compat()
- *
- * Description:
- *	This routine is used by traditional Codecs in COMPAT mode. The
- *	audio samples are placed directly into the buffer provided by the
- *	Audio Driver. Once one playing channel is found the search ends,
- *	no reason to waste more time.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instances' state
- *	audio_apm_info_t	*apm_infop	Personality module data struct
- *	void			*buf		The buf to place the audio into
- *	int			samples		The number of samples to get
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		wants. So it is samples at the hardware's sample rate.
- *
- * Returns:
- *	>= 0			The number of samples transferred to the buffer
- *	AUDIO_FAILURE		An error has occurred
- */
-static int
-am_get_audio_trad_compat(audio_state_t *statep, audio_apm_info_t *apm_infop,
-	void *buf, int samples)
-{
-	audio_ch_t		*chptr;
-	audio_info_t		*info;
-	am_ch_private_t		*chpptr;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	int			i;
-	int			max_chs;
-	int			ret_val;
-	size_t			size = samples << AM_INT32_SHIFT;
-
-	ATRACE("in am_get_audio_trad_compat()", statep);
-
-	ASSERT(statep);
-
-	/* get the number of chs for this instance */
-	max_chs = statep->as_max_chs;
-
-	/* go through the chs looking for the only playing AUDIO ch */
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < max_chs; i++, chptr++) {
-
-		/* lock the channel before we check it out */
-		mutex_enter(&chptr->ch_lock);
-
-		/* skip non-AUDIO and unallocated channels */
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    chptr->ch_info.dev_type != AUDIO ||
-		    chptr->ch_info.pid == 0) {
-
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* make sure this channel is valid */
-		chpptr = chptr->ch_private;
-		info = chptr->ch_info.info;
-		if (chpptr == NULL || info == NULL ||
-		    (chpptr->acp_flags & AM_CHNL_OPEN) == 0) {
-			mutex_exit(&chptr->ch_lock);
-			ATRACE("am_get_audio_trad_compat() channel closed",
-			    chptr);
-			continue;
-		}
-
-		/* make sure this channel is writing */
-		if (!chpptr->acp_writing) {
-			ATRACE("am_get_audio_trad_compat() not playing",
-			    chpptr);
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* skip paused AUDIO channels */
-		if (info->play.pause) {
-			mutex_exit(&chptr->ch_lock);
-			return (0);
-		}
-
-		mutex_exit(&chptr->ch_lock);
-
-		ATRACE_32("am_get_audio_trad_compat() found channel", i);
-		break;
-	}
-	if (i >= max_chs) {
-		ATRACE("am_get_audio_trad_compat() done, no play channel", buf);
-		return (0);
-	}
-
-	mutex_enter(&chptr->ch_lock);
-
-	/*
-	 * We had to free the lock above to make warlock happy. Unfortunately
-	 * it's possible that the channel closed between releasing and
-	 * reacquiring the lock. So we have to check again.
-	 */
-	if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    chptr->ch_info.dev_type != AUDIO || chptr->ch_info.pid == 0 ||
-	    chpptr == NULL || (chpptr->acp_flags & AM_CHNL_QPROCSOFF)) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_get_audio_trad_compat() ch closed on us", chptr);
-		return (0);
-	}
-
-	/* make sure the buffer is big enough */
-	if (chpptr->acp_psb_size < size) {
-		ATRACE_32("am_get_audio_trad_compat() freeing buffer",
-		    chpptr->acp_psb_size);
-		if (chpptr->acp_play_samp_buf) {
-			/* free the old buffer */
-			kmem_free(chpptr->acp_play_samp_buf,
-			    chpptr->acp_psb_size);
-		}
-		chpptr->acp_play_samp_buf = kmem_alloc(size, KM_NOSLEEP);
-		if (chpptr->acp_play_samp_buf == NULL) {
-			ATRACE_32("am_get_audio_trad_compat() "
-			    "kmem_alloc() play_samp_buf failed", i);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-			    "am_get_audio_trad_compat() "
-			    "sample buffer %d not allocated", i);
-			chpptr->acp_psb_size = 0;
-			mutex_exit(&chptr->ch_lock);
-			return (0);
-		}
-		chpptr->acp_psb_size = size;
-	}
-
-	ATRACE_32("am_get_audio_trad_compat() calling am_get_samples()",
-	    samples);
-
-	chpptr->acp_busy_cnt++;
-	ret_val = am_get_samples(chptr, samples, chpptr->acp_play_samp_buf,
-	    AM_COMPAT_MODE);
-	chpptr->acp_busy_cnt--;
-
-	ATRACE_32("am_get_audio_trad_compat() am_get_samples() succeeded",
-	    ret_val);
-
-	/* now we can see how well the am_get_samples() call did */
-	if (ret_val == AUDIO_FAILURE || ret_val == 0) {
-		am_audio_drained(chptr);
-		ATRACE_32("am_get_audio_trad_compat() am_get_samples() failed",
-		    ret_val);
-		goto done;
-	}
-
-	/* now convert to the format the audio device uses */
-	am_convert_to_format(chpptr->acp_play_samp_buf, buf, ret_val,
-	    stpptr->am_hw_pprec, stpptr->am_hw_penc, stpptr->am_pflags);
-
-done:
-	if ((chpptr->acp_busy_cnt == 0) &&
-	    (chpptr->acp_flags & AM_CHNL_SIGNAL_NEEDED)) {
-		ATRACE("am_get_audio_trad_compat() sending cv_signal()", chptr);
-		cv_signal(&chptr->ch_cv);
-	}
-
-	/* we can free the lock now */
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_get_audio_trad_compat() done", buf);
-
-	return (ret_val);
-
-}	/* am_get_audio_trad_compat() */
-
-/*
- * am_get_audio_trad_mixer()
- *
- * Description:
- *	This routine is used by traditional Codecs in MIXER mode to get
- *	multiple streams of audio data and mixing them down into one stream
- *	for the Codec. play_samp_buf is used to get audio samples. These
- *	samples are then mixed into the mix buffer. When all playing audio
- *	channels are mixed we convert to the proper output format, along with
- *	applying gain and balance, if needed.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instances' state
- *	audio_apm_info_t	*apm_infop	Personality module data struct
- *	void			*buf		The buf to place the audio into
- *	int			samples		The number of samples to get
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		wants. So it is samples at the hardware's sample rate.
- *
- * Returns:
- *	>= 0			The number of samples transferred to the buffer
- *	AUDIO_FAILURE		An error has occurred
- */
-static int
-am_get_audio_trad_mixer(audio_state_t *statep, audio_apm_info_t *apm_infop,
-	void *buf, int samples)
-{
-	audio_ch_t		*chptr;
-	audio_info_t		*info;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	audio_info_t		*hw_info;
-	am_ch_private_t		*chpptr;
-	int			*mix_dest;
-	int			*mix_src;
-	size_t			size = samples << AM_INT32_SHIFT;
-	uint_t			hw_channels = stpptr->am_hw_pchs;
-	int			balance;
-	int			i;
-	int			l_gain;
-	int			r_gain;
-	int			max_chs;
-	int			max_ret_val = 0;
-	int			ret_val;
-
-	ATRACE("in am_get_audio_trad_mixer()", statep);
-	ATRACE_32("am_get_audio_trad_mixer() samples", samples);
-	ASSERT(statep);
-
-	/* get the number of chs for this instance */
-	max_chs = statep->as_max_chs;
-
-	hw_info = apm_infop->apm_ad_state;
-
-	/* make sure the mix buffer is large enough */
-	if (stpptr->am_mix_size < size) {
-		/* mix buffer too small, adjust sample request */
-		ATRACE_32("am_get_audio_trad_mixer() mix buffer too small",
-		    stpptr->am_mix_size);
-		ATRACE_32("am_get_audio_trad_mixer() adjust num samples from",
-		    samples);
-		samples = stpptr->am_mix_size >> AM_INT32_SHIFT;
-		ATRACE_32("am_get_audio_trad_mixer() num samples now set to",
-		    samples);
-	}
-
-	/* zero the mix buffer, no reason to zero the whole buffer */
-	bzero(stpptr->am_mix_buf, size);
-
-	/* go through the chs looking for each AUDIO ch */
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < max_chs; i++, chptr++) {
-
-		/* lock the channel before we check it out */
-		mutex_enter(&chptr->ch_lock);
-
-		/* skip non-AUDIO and unallocated channels */
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    (chptr->ch_info.dev_type != AUDIO) ||
-		    (chptr->ch_info.pid == 0)) {
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* make sure this channel is valid */
-		chpptr = chptr->ch_private;
-		info = chptr->ch_info.info;
-
-		if (chpptr == NULL || info == NULL ||
-		    (chpptr->acp_flags & AM_CHNL_QPROCSOFF) ||
-		    (!chpptr->acp_writing)) {
-			mutex_exit(&chptr->ch_lock);
-			ATRACE("am_get_audio_trad_mixer() not valid", chpptr);
-			continue;
-		}
-
-		/* skip paused AUDIO channels */
-		if (info->play.pause) {
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		ATRACE_32("am_get_audio_trad_mixer() found channel", i);
-
-		info = chptr->ch_info.info;
-
-		/* make sure the buffer is big enough */
-		if (chpptr->acp_psb_size < size) {
-
-			ATRACE_32("am_get_audio_trad_mixer() freeing buffer",
-			    chpptr->acp_psb_size);
-			if (chpptr->acp_play_samp_buf) {
-				/* free the old buffer */
-				kmem_free(chpptr->acp_play_samp_buf,
-				    chpptr->acp_psb_size);
-			}
-			chpptr->acp_play_samp_buf =
-			    kmem_alloc(size, KM_NOSLEEP);
-			if (chpptr->acp_play_samp_buf == NULL) {
-				ATRACE_32("am_get_audio_trad_mixer() "
-				    "kmem_alloc() play_samp_buf failed", i);
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_WARN,
-				    "am_get_audio_trad_mixer() "
-				    "sample buffer %d not allocated", i);
-				chpptr->acp_psb_size = 0;
-				mutex_exit(&chptr->ch_lock);
-				continue;
-			}
-			chpptr->acp_psb_size = size;
-		}
-
-		/* get "samples" worth of audio */
-		ATRACE_32("am_get_audio_trad_mixer() calling am_get_samples()",
-		    samples);
-
-		chpptr->acp_busy_cnt++;
-		ret_val = am_get_samples(chptr, samples,
-		    chpptr->acp_play_samp_buf, AM_MIXER_MODE);
-		chpptr->acp_busy_cnt--;
-
-		ATRACE_32("am_get_audio_trad_mixer() "
-		    "am_get_samples() succeeded", ret_val);
-
-		/* now we can see how well the am_get_samples() call did */
-		if (ret_val == AUDIO_FAILURE || ret_val == 0) {
-			am_audio_drained(chptr);
-			mutex_exit(&chptr->ch_lock);
-			ATRACE_32("am_get_audio_trad_mixer() "
-			    "am_get_samples() failed", ret_val);
-			continue;
-		}
-
-
-		/* we return the maximum # of samples found & processed */
-		if (ret_val > max_ret_val) {
-			/* update to a new value */
-			max_ret_val = ret_val;
-			ATRACE_32("am_get_audio_trad_mixer() "
-			    "updated max_ret_val", max_ret_val);
-		}
-
-		/* mix this channel into the mix buffer */
-		mix_src = chpptr->acp_play_samp_buf;
-		mix_dest = stpptr->am_mix_buf;
-		ATRACE("am_get_audio_trad_mixer() mix_src before", mix_src);
-		ATRACE("am_get_audio_trad_mixer() mix_dest before", mix_dest);
-		/* apply gain and balance while summing */
-		if (hw_channels == AUDIO_CHANNELS_MONO) {
-			l_gain = info->play.gain;
-
-			for (; ret_val; ret_val--) {
-				*mix_dest++ +=
-				    (*mix_src++ * l_gain) >> AM_MAX_GAIN_SHIFT;
-			}
-		} else {
-			ASSERT(hw_channels == AUDIO_CHANNELS_STEREO);
-
-			l_gain = r_gain = info->play.gain;
-			balance = info->play.balance;
-
-			if (balance < AUDIO_MID_BALANCE) {
-				/* leave l gain alone and scale down r gain */
-				r_gain = (r_gain * balance) >> 5;
-			} else if (balance > AUDIO_MID_BALANCE) {
-				/* leave r gain alone and scale down l gain */
-				l_gain = (l_gain * (64 - balance)) >> 5;
-			}
-
-			for (; ret_val; ret_val -= 2) {
-				*mix_dest++ +=
-				    (*mix_src++ * l_gain) >> AM_MAX_GAIN_SHIFT;
-				*mix_dest++ +=
-				    (*mix_src++ * r_gain) >> AM_MAX_GAIN_SHIFT;
-			}
-		}
-
-		if ((chpptr->acp_busy_cnt == 0) &&
-		    (chpptr->acp_flags & AM_CHNL_SIGNAL_NEEDED)) {
-			ATRACE("am_get_audio_trad_mixer() sending cv_signal()",
-			    chptr);
-			cv_signal(&chptr->ch_cv);
-		}
-
-		/* we can free the lock now */
-		mutex_exit(&chptr->ch_lock);
-
-		ATRACE("am_get_audio_trad_mixer() mix_src after", mix_src);
-		ATRACE("am_get_audio_trad_mixer() mix_dest after", mix_dest);
-
-		ATRACE_32("am_get_audio_trad_mixer() ret_val", ret_val);
-		ATRACE_32("am_get_audio_trad_mixer() max_ret_val", max_ret_val);
-
-		ATRACE("am_get_audio_trad_mixer() going again", chptr);
-	}
-
-	/* now convert into the format the hardware needs */
-	ATRACE("am_get_audio_trad_mixer() calling am_convert_to_format()",
-	    stpptr->am_mix_buf);
-	am_convert_to_format(stpptr->am_mix_buf, buf, max_ret_val,
-	    stpptr->am_hw_pprec, stpptr->am_hw_penc, stpptr->am_pflags);
-
-	/* update hardware sample count */
-	hw_info->play.samples += max_ret_val / stpptr->am_hw_pchs;
-
-	ATRACE("am_get_audio_trad_mixer() done", buf);
-
-	return (max_ret_val);
-
-}	/* am_get_audio_trad_mixer() */
-
-/*
- * am_get_samples()
- *
- * Description:
- *	This routine takes the first message off the channel's queue. It
- *	then takes audio data until the requested number of samples has
- *	been reached or there are no more samples. If the message isn't
- *	empty it is put back onto the message queue.
- *
- *	If the channel is muted then the requested number of samples is
- *	updated in the buffer, the pointer for the message is advanced and
- *	the buffer is zeroed, filling it with silence.
- *
- *	If the "mode" argument is set to AM_COMPAT_MODE then the data size
- *	is set by the channel's info structure. Otherwise it is set to the
- *	size of an integer. Because multi-stream devices get raw data,
- *	am_get_audio_multi() calls this with AM_COMPAT_MODE set.
- *
- *	Keeping track of samples is complicated. When a complete message has
- *	been played info->play.samples is incremented by the number of sample
- *	frames in the original message. That way we don't build up errors by
- *	approximating the sample rate converted sample frames. As each chunk
- *	of samples is retrieved to be played we add this amount to
- *	acp_psamples_c, the running count. Then the next time that this routine
- *	is called the amount in acp_psamples_c is added to acp_psamples_p,
- *	the partial count. This causes the delay between when the samples
- *	are placed into the buffer and when it is actually played.
- *	acp_psamples_p is used by am_fix_info() to adjust info->play.samples
- *	in between full messages, including for sample rate conversion.
- *
- *	No conversions are done in this routine.
- *
- *	CAUTION: The channel must be locked before this routine is called.
- *
- *	NOTE: The variable "samples" is the number of samples returned
- *		ultimately to the hardware.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel's state structure
- *	int		samples		The number of samples to get
- *	void		*buf		The buffer to put the samples into
- *	int		mode		Mixer mode
- *
- * Returns:
- *	0 -> samples			The number of samples retrieved
- *	AUDIO_FAILURE			There was an error getting samples
- */
-static int
-am_get_samples(audio_ch_t *chptr, int samples, void *buf, int mode)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_data_t		*data;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	audio_info_t		*hw_info = &stpptr->am_hw_info;
-	audio_info_t		*info = chptr->ch_info.info;
-#ifdef FLOW_CONTROL
-	queue_t			*q;
-#endif
-	size_t			orig_size;
-	int			*bptr = buf;
-	int			*eptr;
-	int			*pstart;
-	int			*sptr;
-	boolean_t		active_signal = B_FALSE;
-	boolean_t		empty = B_FALSE;
-	boolean_t		mute;
-	boolean_t		EOF_processed = B_FALSE;
-	uint_t			ret_samples = 0;
-	int			bytes_needed;
-	int			count;
-	int			EOF_count;
-	int			hw_channels = stpptr->am_hw_pchs;
-
-	ATRACE("in am_get_samples()", chptr);
-	ATRACE_32("am_get_samples() samples", samples);
-	ATRACE("am_get_samples() buffer", buf);
-
-	ASSERT(statep);
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	/* do this now where it's easier */
-	mute = info->output_muted;
-	if (mute) {
-		/* we return zeros */
-		bytes_needed = samples << AM_INT32_SHIFT;
-		ATRACE("am_get_samples() bzero", buf);
-		bzero(buf, bytes_needed);
-	}
-
-	/* update played samples */
-	if (chpptr->acp_psamples_f) {
-		chpptr->acp_psamples_p = 0;
-	}
-
-	info->play.samples += chpptr->acp_psamples_f;
-	chpptr->acp_psamples_p += chpptr->acp_psamples_c;
-
-	chpptr->acp_psamples_f = 0;
-	chpptr->acp_psamples_c = 0;
-
-	/* go through as many buffers as we need to get the samples */
-	for (; samples > 0; ) {
-		if (!(chpptr->acp_flags & AM_CHNL_OPEN)) {
-			ATRACE_32("am_get_samples() not open",
-			    chpptr->acp_flags);
-			return (0);
-		}
-
-		/* get the data off the list */
-		data = audio_sup_get_audio_data(chptr);
-
-#ifdef FLOW_CONTROL
-		/*
-		 * See if we should re-enable the queue. We do this now
-		 * because we always need to do it and the code branches below.
-		 */
-		ASSERT(MUTEX_HELD(&chptr->ch_lock));
-		if ((chpptr->acp_flags & AM_CHNL_PFLOW) &&
-		    (audio_sup_get_audio_data_size(chptr) <
-		    (AM_MIN_QUEUED_MSGS_SIZE) &&
-		    audio_sup_get_audio_data_cnt(chptr) <
-		    AM_MIN_QUEUED_MSGS_CNT)) {
-			/* yes, re-enable the q */
-			chpptr->acp_flags &= ~AM_CHNL_PFLOW;
-
-			q = WR(chptr->ch_qptr);
-
-			ATRACE("am_get_samples() flow control disabled, q on",
-			    q);
-
-			enableok(q);
-			qenable(q);
-		}
-#endif
-
-		/* now we see if we got any data */
-		if (data == NULL) {
-			ATRACE("am_get_samples() no data", chptr);
-			/* we underflowed, so up error count and send signal */
-
-			mutex_enter(&chptr->ch_apm_infop->apm_lock);
-
-			if (!(chpptr->acp_flags &
-			    (AM_CHNL_EMPTY|AM_CHNL_ALMOST_EMPTY1)) &&
-			    !(stpptr->am_flags & AM_PRIV_SW_MODES)) {
-				/* but only send it one time */
-				info->play.error = 1;
-				empty = B_TRUE;
-			}
-			mutex_exit(&chptr->ch_apm_infop->apm_lock);
-
-			/* did we get any audio at all? */
-			ATRACE_32("am_get_samples() ret_samples", ret_samples);
-			if (ret_samples == 0) {
-			    ATRACE("am_get_samples() no data returning", chptr);
-
-				/*
-				 * Marking the channel as empty is a two step
-				 * process because data is queued up and still
-				 * being played the first time we determine we
-				 * are empty. So the first time we set
-				 * ALMOST_EMPTY. The second time we set EMPTY.
-				 */
-			    if (!(chpptr->acp_flags & (AM_CHNL_ALMOST_EMPTY1|
-				AM_CHNL_ALMOST_EMPTY2|AM_CHNL_EMPTY))) {
-
-				chpptr->acp_flags |= AM_CHNL_ALMOST_EMPTY1;
-				ATRACE("am_get_samples() no data empty1",
-				    chpptr->acp_flags);
-			    } else if
-				((chpptr->acp_flags & AM_CHNL_ALMOST_EMPTY1)) {
-
-				chpptr->acp_flags &= ~AM_CHNL_ALMOST_EMPTY1;
-				chpptr->acp_flags |= AM_CHNL_ALMOST_EMPTY2;
-				ATRACE("am_get_samples() no data empty2",
-				    chpptr->acp_flags);
-			    } else {
-				ASSERT(chpptr->acp_flags &
-				    (AM_CHNL_ALMOST_EMPTY2|AM_CHNL_EMPTY));
-				chpptr->acp_flags &= ~(AM_CHNL_ALMOST_EMPTY1|\
-				    AM_CHNL_ALMOST_EMPTY2|\
-				    AM_CHNL_DRAIN_NEXT_INT|AM_CHNL_DRAIN);
-				chpptr->acp_flags |= AM_CHNL_EMPTY;
-				ATRACE("am_get_samples() no data empty",
-				    chpptr->acp_flags);
-				if (info->play.active) {
-					active_signal = B_TRUE;
-					info->play.active = 0;
-				}
-			    }
-			    goto done_getting_samples;
-			}
-			ATRACE_32("am_get_samples() no data but samps",
-			    ret_samples);
-			goto done_getting_samples;
-		}
-
-		mutex_enter(&chptr->ch_apm_infop->apm_lock);
-		/*
-		 * See if we need to change modes. The following code makes
-		 * sure that this switch happens on a message boundary.
-		 *
-		 * It should be noted that it is possible to pause the
-		 * channel and then switch modes. Because pause may last
-		 * a long time we don't block the mode switch. When paused
-		 * it is very likely that we will not be on a message
-		 * boundary. If setting to mixer mode it is the responsibility
-		 * of the mode switch code to perform the sample rate
-		 * conversion of canonical audio and then set the pointers
-		 * to the approximate location for restarting, if setting
-		 * to mixer mode. It is also likely that there are additional
-		 * unprocessed messages. Code below will process each of
-		 * these as they come up.
-		 */
-		if (stpptr->am_flags & AM_PRIV_SW_MODES &&
-		    ((mode == AM_MIXER_MODE && data->adata_proc &&
-		    data->adata_pptr == data->adata_proc) ||
-		    (mode == AM_COMPAT_MODE && data->adata_orig &&
-		    data->adata_optr == data->adata_orig))) {
-			mutex_exit(&chptr->ch_apm_infop->apm_lock);
-
-			ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-
-			audio_sup_putback_audio_data(chptr, data);
-
-			ATRACE("am_get_samples() AM_PRIV_SW_MODES return", 0);
-
-			goto done_getting_samples;
-		}
-		mutex_exit(&chptr->ch_apm_infop->apm_lock);
-
-		/* check for EOF message, i.e., zero length buffer */
-		if (data->adata_osize == 0) {
-			ATRACE_32("am_get_samples() EOF",
-			    chpptr->acp_EOF[chpptr->acp_EOF_toggle]);
-
-			chpptr->acp_EOF[chpptr->acp_EOF_toggle]++;
-
-			EOF_processed = B_TRUE;
-
-			ATRACE_32("am_get_samples() EOF, new count",
-			    chpptr->acp_EOF[chpptr->acp_EOF_toggle]);
-
-			audio_sup_free_audio_data(data);
-			continue;
-		}
-
-		ATRACE("am_get_samples() got data", data);
-
-		/*
-		 * We may have changed modes or a previous attempt to process
-		 * the audio data failed. This gives us a second chance before
-		 * we have to throw the audio away.
-		 */
-		if (mode == AM_MIXER_MODE && data->adata_proc == NULL &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			ATRACE("am_get_samples(M) calling am_reprocess()",
-			    data);
-
-			/* if set to 0 then we've got a problem */
-			if (data->adata_osize == 0) {
-				ATRACE_32("am_get_samples(M) bad osize",
-				    data->adata_osize);
-				audio_sup_free_audio_data(data);
-				continue;
-			}
-
-			/* process the original data into src data */
-			if (am_reprocess(chptr, data) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep),
-				    CE_NOTE, "get_samples() "
-				    "couldn't process message, data lost");
-				audio_sup_free_audio_data(data);
-				continue;
-			}
-			ATRACE("am_get_samples() process successful", data);
-		}
-
-		/*
-		 * Get the size of the original data, in sample frames.
-		 * First convert from bytes to samples, then adjust for
-		 * the number of channels.
-		 */
-		orig_size = (data->adata_osize >> AM_INT32_SHIFT) / hw_channels;
-
-		/* get the right buffer */
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			pstart = data->adata_pptr;
-			eptr = data->adata_peptr;
-		} else {
-			ASSERT(mode == AM_COMPAT_MODE ||
-			    ad_infop->ad_codec_type == AM_MS_CODEC);
-			pstart = data->adata_optr;
-			eptr = data->adata_oeptr;
-		}
-
-		ATRACE("am_get_samples() beginning eptr", eptr);
-		ATRACE_32("am_get_samples() samples needed", samples);
-
-		/* get the data from the message and put into the buf */
-
-		sptr = pstart;
-		ASSERT(sptr >= pstart);
-		ATRACE("am_get_samples() beginning sptr", sptr);
-
-		if (mute == B_TRUE) {
-			/* we already zeroed the buffer above */
-			if ((eptr - sptr) < samples) {
-				count = eptr - sptr;
-				samples -= count;
-				sptr = eptr;
-				ASSERT(samples > 0);
-			} else {
-				count = samples;
-				samples = 0;
-				sptr += count;
-				ASSERT(sptr <= eptr);
-			}
-		} else {
-			/* copy into the buffer */
-			for (count = 0; sptr < eptr && samples > 0;
-			    samples--, count++) {
-				*bptr++ = *sptr++;
-			}
-		}
-
-		ATRACE("am_get_samples() ending sptr", sptr);
-		ATRACE_32("am_get_samples() ending samples needed", samples);
-		ATRACE_32("am_get_samples() ending count", count);
-
-		ASSERT(sptr <= eptr);
-		ret_samples += count;
-
-		ATRACE_32("am_get_samples() ret_samples #2", ret_samples);
-
-		/* see if we need to go again */
-		if (samples == 0) {	/* nope */
-			/* see if we're done with this message */
-			if (sptr >= eptr) {
-				ASSERT(sptr == eptr);
-				/* update sample counts */
-				chpptr->acp_psamples_f += orig_size;
-				chpptr->acp_psamples_c = 0;
-
-				/* end of data, so free */
-				audio_sup_free_audio_data(data);
-				data = NULL;
-			} else {	/* nope, use again next time */
-				pstart = sptr;
-				chpptr->acp_psamples_c += count / hw_channels;
-			}
-			break;
-		} else {
-			chpptr->acp_psamples_f += orig_size;
-			chpptr->acp_psamples_c = 0;
-		}
-		/* we need to go again, but free data first */
-		audio_sup_free_audio_data(data);
-		buf = bptr;	/* save for next go around */
-
-	}	/* get buffers for() loop */
-
-	/* update pointers, if partial buffer used */
-	if (data) {
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			/* update the processed data pointer */
-			data->adata_pptr = pstart;
-		} else {
-			/* update the original data pointer */
-			data->adata_optr = pstart;
-		}
-		audio_sup_putback_audio_data(chptr, data);
-	}
-
-done_getting_samples:
-
-	/* see if we need to send any EOF signals */
-	AUDIO_TOGGLE(chpptr->acp_EOF_toggle);
-	EOF_count = 0;
-	if (chpptr->acp_EOF[chpptr->acp_EOF_toggle]) {
-		EOF_count += chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-		chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-	}
-
-	/*
-	 * If all we have are EOFs then we need to flush the EOF count.
-	 * This is done in am_wsvc() as well. This way the EOF count is
-	 * incremented as soon as it can be detected.
-	 */
-	if (audio_sup_get_audio_data_cnt(chptr) == 0 && !hw_info->play.active) {
-		AUDIO_TOGGLE(chpptr->acp_EOF_toggle);
-		if (chpptr->acp_EOF[chpptr->acp_EOF_toggle]) {
-			EOF_count += chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-			chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-		}
-
-		/* if all we had was EOFs then we really are empty */
-		if (EOF_processed && ret_samples == 0) {
-			chpptr->acp_flags &= ~(AM_CHNL_ALMOST_EMPTY1|\
-			    AM_CHNL_ALMOST_EMPTY2);
-			chpptr->acp_flags |= AM_CHNL_EMPTY;
-		}
-	}
-	mutex_exit(&chptr->ch_lock);
-
-	/* Underflowed */
-	if (empty == B_TRUE) {
-		am_send_signal(chptr->ch_statep, stpptr);
-	}
-
-	/* EOF markers */
-	for (; EOF_count; EOF_count--) {
-		info->play.eof++;
-		am_send_signal(chptr->ch_statep, stpptr);
-	}
-
-	ATRACE("am_get_samples() done_getting_samples", chptr);
-
-	/* now we are done, so return how many samples we have */
-
-	ATRACE_32("am_get_samples() normal return", ret_samples);
-
-	/* make sure virtual channels are still active */
-	if (mode == AM_MIXER_MODE && ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		if (ret_samples) {
-			if (info->play.active == 0) {
-				active_signal = B_TRUE;
-				info->play.active = 1;
-			}
-			hw_info->play.active = 1;
-		}
-		if (empty == B_TRUE) {
-			if (info->play.active == 1) {
-				active_signal = B_TRUE;
-				info->play.active = 0;
-			}
-			/* we don't turn off the h/w active flag here */
-		}
-	}
-	/* send the signals for active change */
-	if (active_signal == B_TRUE) {
-		am_send_signal(chptr->ch_statep, stpptr);
-	}
-
-	mutex_enter(&chptr->ch_lock);
-
-	return (ret_samples);
-
-}	/* am_get_samples() */
-
-/*
- * am_play_shutdown_multi()
- *
- * Description:
- *	This routine is used to clean things up when the Audio Driver will
- *	no longer be servicing it's play interrupts. I.e., play interrupts
- *	have been turned off.
- *
- *	This routine makes sure that any DRAINs waiting for an interrupt are
- *	cleared.
- *
- *	It is also used to coordinate shutting down play so that we can
- *	switch between MIXER and COMPAT modes.
- *
- *	NOTE: We use the EOF_count to also determine if we have an active
- *		flag change and thus need a signal sent.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	int		channel		For multi-stream Codecs this is the
- *					stream to shutdown.
- *
- * Returns:
- *	void
- */
-static void
-am_play_shutdown_multi(audiohdl_t handle, int channel)
-{
-	audio_state_t			*statep = AUDIO_HDL2STATE(handle);
-	audio_ch_t			*chptr;
-	am_apm_private_t		*stpptr;
-	am_ch_private_t			*chpptr;
-	audio_info_t			*info;
-	int				EOF_count = 0;
-
-	ATRACE("in am_play_shutdown_multi()", handle);
-	ASSERT(statep);
-
-	chptr = &statep->as_channels[channel];
-	stpptr = chptr->ch_apm_infop->apm_private;
-
-	mutex_enter(&chptr->ch_lock);
-	/*
-	 * The channel may have been closed while we waited on the mutex.
-	 * So once we get it we make sure the channel is still valid.
-	 */
-	chpptr = chptr->ch_private;
-	if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    chptr->ch_info.pid == 0 ||
-	    (chpptr->acp_flags & AM_CHNL_OPEN) == 0) {
-		ATRACE("am_play_shutdown_multi() channel closed", chptr);
-		mutex_exit(&chptr->ch_lock);
-		return;
-	} else if (chpptr->acp_flags &
-	    (AM_CHNL_DRAIN|AM_CHNL_DRAIN_NEXT_INT|AM_CHNL_EMPTY|\
-	    AM_CHNL_ALMOST_EMPTY1|AM_CHNL_ALMOST_EMPTY2|AM_CHNL_CLOSING)) {
-		/* clear a bunch of flags */
-		chpptr->acp_flags &= ~(AM_CHNL_DRAIN|AM_CHNL_DRAIN_NEXT_INT|
-		    AM_CHNL_ALMOST_EMPTY1|AM_CHNL_ALMOST_EMPTY2);
-
-		/* are we empty? */
-		if (audio_sup_get_audio_data_cnt(chptr) == 0) {
-			/* yes, so mark as empty */
-			chpptr->acp_flags |= AM_CHNL_EMPTY;
-		}
-
-		/* turn off the channel, but only if not paused */
-		info = chptr->ch_info.info;
-		if (!info->play.pause && info->play.active) {
-			EOF_count++;
-			info->play.active = 0;
-		}
-
-		am_audio_drained(chptr);
-
-		/* make sure we send all pending signals */
-		EOF_count += chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-		chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-
-		AUDIO_TOGGLE(chpptr->acp_EOF_toggle);
-
-		EOF_count += chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-		chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-
-		mutex_exit(&chptr->ch_lock);
-
-		for (; EOF_count; EOF_count--) {
-			info->play.eof++;
-			am_send_signal(statep, stpptr);
-		}
-	} else {
-		/* we can free the lock now */
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	ATRACE("am_play_shutdown_multi() returning", statep);
-
-	return;
-
-}	/* am_play_shutdown_multi() */
-
-/*
- * am_play_shutdown_trad()
- *
- * Description:
- *	This routine is used to clean things up when the Audio Driver will
- *	no longer be servicing it's play interrupts. I.e., play interrupts
- *	have been turned off.
- *
- *	This routine makes sure that any DRAINs waiting for an interrupt are
- *	cleared.
- *
- *	It is also used to coordinate shutting down play so that we can
- *	switch between MIXER and COMPAT modes.
- *
- * Arguments:
- *	audiohdl_t	handle		Handle to the device
- *	audio_apm_info_t *apm_infop	Pointer to APM info
- *
- * Returns:
- *	void
- */
-static void
-am_play_shutdown_trad(audiohdl_t handle, audio_apm_info_t *apm_infop)
-{
-	audio_state_t			*statep = AUDIO_HDL2STATE(handle);
-	audio_ch_t			*chptr;
-	am_ch_private_t			*chpptr;
-	audio_info_t			*info;
-	audio_info_t			*hw_info;
-	am_apm_private_t		*stpptr;
-	int				EOF_count;
-	int				i;
-	int				max_chs;
-
-	ATRACE("in am_play_shutdown_trad()", handle);
-	ASSERT(statep);
-
-	max_chs = statep->as_max_chs;
-	stpptr = apm_infop->apm_private;
-	hw_info = apm_infop->apm_ad_state;
-
-	mutex_enter(&apm_infop->apm_lock);
-	if (stpptr->am_flags & AM_PRIV_SW_MODES) {
-		mutex_exit(&apm_infop->apm_lock);
-
-		ATRACE("am_play_shutdown_trad() change mode shutdown", stpptr);
-
-		/* don't worry about signal, the mode switch will send one */
-		hw_info->play.active = 0;
-
-		/* let the mode change proceed */
-		mutex_enter(&stpptr->am_mode_lock);
-		cv_signal(&stpptr->am_mode_cv);
-		mutex_exit(&stpptr->am_mode_lock);
-
-		return;
-	}
-	mutex_exit(&apm_infop->apm_lock);
-
-	/* go through all the channels */
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < max_chs; i++, chptr++) {
-		/* skip non-AUDIO and unallocated channels */
-		mutex_enter(&chptr->ch_lock);
-
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    chptr->ch_info.dev_type != AUDIO ||
-		    chptr->ch_info.pid == 0) {
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		chpptr = chptr->ch_private;
-
-		ATRACE_32("am_play_shutdown_trad() checking flags",
-		    chpptr->acp_flags);
-
-		info = chptr->ch_info.info;
-		if (chpptr->acp_flags &
-		    (AM_CHNL_DRAIN|AM_CHNL_DRAIN_NEXT_INT|\
-		    AM_CHNL_EMPTY|AM_CHNL_ALMOST_EMPTY1|\
-		    AM_CHNL_ALMOST_EMPTY2|AM_CHNL_CLOSING)) {
-			/* clear a bunch of flags */
-			chpptr->acp_flags &=
-			    ~(AM_CHNL_DRAIN|AM_CHNL_DRAIN_NEXT_INT|\
-			    AM_CHNL_ALMOST_EMPTY1|AM_CHNL_ALMOST_EMPTY2);
-
-			/* are we empty? */
-			if (audio_sup_get_audio_data_cnt(chptr) == 0) {
-				/* yes, so mark as empty */
-				chpptr->acp_flags |= AM_CHNL_EMPTY;
-			}
-
-			/* turn off the ch, but only if not paused */
-			if (!info->play.pause) {
-				info->play.active = 0;
-			}
-
-			ATRACE("am_play_shutdown_trad() regular drain", chptr);
-
-			am_audio_drained(chptr);
-		} else {
-			/* turn off the ch, but only if not paused */
-			if (!info->play.pause) {
-				info->play.active = 0;
-			}
-		}
-
-		/* make sure we send all pending signals */
-		chpptr = chptr->ch_private;
-
-		EOF_count = chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-		chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-
-		AUDIO_TOGGLE(chpptr->acp_EOF_toggle);
-
-		EOF_count += chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-		chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-
-		mutex_exit(&chptr->ch_lock);
-
-		for (; EOF_count; EOF_count--) {
-			info->play.eof++;
-
-			am_send_signal(statep, stpptr);
-		}
-	}
-
-	/* turn off the hardware active flag, always send a signal */
-	hw_info->play.active = 0;
-	am_send_signal(statep, stpptr);
-
-	ATRACE("am_play_shutdown_trad() returning", statep);
-
-	return;
-
-}	/* am_play_shutdown_trad() */
-
-/*
- * am_release_ad_access()
- *
- * Description:
- *	This routine is used to release the serial access to the audio driver
- *	entry points.
- *
- * Arguments:
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *
- * Returns:
- *	Void
- */
-static void
-am_release_ad_access(am_apm_private_t *stpptr)
-{
-	ATRACE("in am_release_ad_access()", stpptr);
-	ATRACE("am_release_ad_access() curthread", curthread);
-
-	/* wait for all other calls into the audio driver to return */
-	mutex_enter(&stpptr->am_ad_lock);
-
-	ASSERT(stpptr->am_ad_in & AM_APM_IN_DRIVER);
-
-	/* we're done, so release any waiting threads */
-	stpptr->am_ad_in &= ~AM_APM_IN_DRIVER;
-
-	/* wake up one thread waiting on this CV */
-	cv_signal(&stpptr->am_ad_cv);
-
-	mutex_exit(&stpptr->am_ad_lock);
-
-	ATRACE("am_release_ad_access() done", curthread);
-
-}	/* am_release_ad_access() */
-
-/*
- * am_set_record_streams()
- *
- * Description:
- *	Set the STREAMS high and low water marks for the record channel. We try
- *	this when the audio channel is initially opened. If that fails,
- *	am_send_audio_common() will also make an attempt to set the water marks.
- *	Otherwise, left alone the STREAMS default settings overflow almost
- *	immediately at the higher sample rates.
- *
- * Arguments:
- *      audio_ch_t	*chptr		Pointer to the channel structure
- *
- * Returns:
- *	AUDIO_SUCCESS		High and Low water marks increased
- *	AUDIO_FAILURE		High and Low water marks not increased
- */
-int
-am_set_record_streams(audio_ch_t *chptr)
-{
-	mblk_t			*mop;
-	struct stroptions	*sop;
-
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*sop));
-
-	if (mop = allocb(sizeof (struct stroptions), BPRI_HI)) {
-		mop->b_datap->db_type = M_SETOPTS;
-		mop->b_wptr += sizeof (struct stroptions);
-		sop = (struct stroptions *)mop->b_rptr;
-		sop->so_flags = SO_HIWAT | SO_LOWAT;
-		sop->so_hiwat = AM_MAX_QUEUED_MSGS_SIZE;
-		sop->so_lowat = AM_MIN_QUEUED_MSGS_SIZE;
-
-		am_safe_putnext(chptr, mop);
-
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*sop));
-
-		ATRACE("am_set_record_streams() adjustment successful", 0);
-
-		return (AUDIO_SUCCESS);
-	}
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*sop));
-
-	ATRACE("am_set_record_streams() adjustment failed", 0);
-
-	return (AUDIO_FAILURE);
-
-}	/* am_set_record_streams() */
-
-/*
- * am_send_audio_common()
- *	Common code between am_send_audio_multi() and am_send_audio_trad().
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to the channel structure
- *	void		*buf		The buffer to get audio from
- *	int		samples		The number of samples the Codec sent
- *
- * Returns:
- *	Void
- */
-static void
-am_send_audio_common(audio_ch_t *chptr, void *buf, int samples)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_info_t		*info = chptr->ch_info.info;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	mblk_t			*mp;
-	queue_t			*q = RD(chptr->ch_qptr);
-	size_t			size;
-	int			bytes_per_samplef;
-	int			channels = info->record.channels;
-	int			mp_size;
-	int			precision = info->record.precision;
-	int			remaining;
-
-	ATRACE("in am_send_audio_common()", chptr);
-
-	ASSERT(statep);
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	/*
-	 * Check if the STREAMS water marks for the record channel have been
-	 * properly set. If not, try and set them here.
-	 */
-	if (!(chpptr->acp_flags & AM_CHNL_RSTREAMS_SET)) {
-		ATRACE("am_send_audio_common() adjusting streams queue",
-		    chpptr);
-		if (am_set_record_streams(chptr) == AUDIO_SUCCESS) {
-			ATRACE("am_send_audio_common() streams queue adjustment"
-			    " successful", chpptr);
-			chpptr->acp_flags |= AM_CHNL_RSTREAMS_SET;
-		}
-	}
-
-	/* figure out how many bytes there are in a sample frame */
-	bytes_per_samplef = channels * (precision >> AUDIO_PRECISION_SHIFT);
-	ATRACE_32("am_send_audio_common() bytes_per_samplef",
-	    bytes_per_samplef);
-
-	/* figure out how many bytes we've got */
-	size = samples * (precision >> AUDIO_PRECISION_SHIFT);
-	ATRACE_32("am_send_audio_common() size", size);
-
-	/* first we see if we have a partial buffer waiting to be sent */
-	if (chpptr->acp_rec_mp) {	/* yup, we need to fill it first */
-		ATRACE("am_send_audio_common() filling partial buffer",
-		    chpptr->acp_rec_mp);
-
-		mp = chpptr->acp_rec_mp;
-
-		/*
-		 * Figure out how much of the buffer is remaining. We don't
-		 * use the record buffer size because it may have changed
-		 * since the message was allocated.
-		 */
-		remaining = chpptr->acp_rec_remaining;
-		ATRACE_32("am_send_audio_common() remaining", remaining);
-
-		/* make sure we've got enough to fill this buffer */
-		if (remaining > size) {
-			/* we don't, so use what we have and return */
-			bcopy(buf, mp->b_wptr, size);
-			mp->b_wptr += size;
-			chpptr->acp_rec_remaining -= size;
-			info->record.samples += size / bytes_per_samplef;
-			ATRACE("am_send_audio_common() not enough", mp);
-
-			/* make sure the channel is still active */
-			info->record.active = 1;
-
-			return;
-		}
-
-		/* we do, so fill and then go on to get a new buffer */
-		bcopy(buf, mp->b_wptr, remaining);
-		mp->b_wptr += remaining;
-		info->record.samples += remaining / bytes_per_samplef;
-		ASSERT(q == RD(q));
-
-		am_safe_putnext(chptr, mp);
-
-		chpptr->acp_rec_mp = NULL;
-		chpptr->acp_rec_remaining = 0;
-		buf = (char *)buf + remaining;
-		size -= remaining;
-		ATRACE_32("am_send_audio_common() partial buffer filled", size);
-		ASSERT(chpptr->acp_rec_remaining == 0);
-	}
-	ASSERT(chpptr->acp_rec_mp == NULL);
-
-	/* start with a full buffer */
-	remaining = info->record.buffer_size;
-
-	/* now place remaining data into new buffers */
-	while (size) {
-
-		/* buffer_size may change during loop */
-		mp_size = info->record.buffer_size;
-
-		if ((mp = allocb(mp_size, BPRI_HI)) == NULL) {
-			/*
-			 * Often times when one allocb() fails we get many
-			 * more that fail. We don't want this error message
-			 * to spew uncontrolled. So we set this flag and the
-			 * next time if it's set we don't display the message.
-			 * Once we get a success we clear the flags, thus one
-			 * message per continuous set of failures.
-			 */
-			if (!(chpptr->acp_flags & AM_CHNL_CLIMIT)) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-				    "mixer: send_audio_c() "
-				    "allocb() failed, recorded audio lost");
-				chpptr->acp_flags |= AM_CHNL_CLIMIT;
-			}
-
-			ATRACE("am_send_audio_common() allocb() failed",
-				chptr);
-			info->record.error = 1;
-			return;
-		}
-		chpptr->acp_flags &= ~AM_CHNL_CLIMIT;
-
-		mp->b_datap->db_type = M_DATA;
-
-		if (mp_size > size) {
-			/* partial buffer */
-			bcopy(buf, mp->b_rptr, size);
-			mp->b_wptr += size;
-			chpptr->acp_rec_remaining = mp_size - size;
-			info->record.samples += size / bytes_per_samplef;
-			chpptr->acp_rec_mp = mp;
-			ATRACE("am_send_audio_common() new not enough", mp);
-			break;
-		}
-		/* full buffer */
-		bcopy(buf, mp->b_rptr, mp_size);
-		mp->b_wptr += mp_size;
-		info->record.samples += mp_size / bytes_per_samplef;
-		ASSERT(q == RD(q));
-
-		am_safe_putnext(chptr, mp);
-
-		ASSERT(chpptr->acp_rec_mp == NULL);
-		chpptr->acp_rec_remaining = 0;
-		buf = (char *)buf + mp_size;
-		size -= mp_size;
-
-		/* start with a full buffer for the next loop */
-		remaining = info->record.buffer_size;
-	}
-
-	/* make sure the channel is still active */
-	info->record.active = 1;
-
-	ATRACE("am_send_audio_common() done", buf);
-
-}	/* am_send_audio_common() */
-
-/*
- * am_send_audio_multi()
- *
- * Description:
- *	This routine is used by multi-channel Codecs to send a single stream
- *	of audio data to an individual channel.
- *
- * Arguments:
- *	audio_state_t	*statep		Pointer to the device instance's state
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	void		*buf		The buffer to get audio from
- *	int		channel		The device channel number
- *	int		samples		The number of samples the Codec sent
- *
- *	NOTE: The variable "samples" is the number of samples the hardware
- *		sent. So it is samples at the hardware's sample rate.
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-am_send_audio_multi(audio_state_t *statep, am_ad_info_t *ad_infop, void *buf,
-	int channel, int samples)
-{
-	audio_ch_t		*chptr = &statep->as_channels[channel];
-	audio_info_t		*info = chptr->ch_info.info;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	size_t			size;
-	size_t			size_plus_rb;
-	int			channels = info->record.channels;
-
-	ASSERT(statep);
-
-	/* skip paused AUDIO channels */
-	if (info->record.pause) {
-		return;
-	}
-
-	/* make sure we can put the audio before we waste time converting it */
-	switch (am_test_canputnext(chptr)) {
-	case AM_CHNL_CLOSED:
-		/* Channel closed on us */
-		ATRACE("am_send_audio_multi() channel closed", chptr);
-		return;
-	case AUDIO_FAILURE:
-		info->record.active = 0;
-		info->record.error = 1;
-		ATRACE_32("am_send_audio_multi() ch flow controlled",
-		    chptr->ch_info.ch_number);
-		return;
-	case AUDIO_SUCCESS:
-				/* FALLTHROUGH */
-	default:
-		break;
-	}
-
-	/* make sure the # of samples is modulo the # of H/W channels */
-	if (channels != AUDIO_CHANNELS_MONO && (samples % channels) != 0) {
-		ATRACE_32("am_send_audio_multi() bad sample size", samples);
-		samples -= samples % channels;
-	}
-
-	mutex_enter(&chptr->ch_lock);
-
-	/* make sure the conversion buffer is large enough */
-	size = samples << AM_TIMES_8_SHIFT;
-	size_plus_rb = size + chpptr->acp_ch_rbuf_size;
-	if (am_update_conv_buffer(chptr, size_plus_rb, AUDIO_RECORD) ==
-		AUDIO_FAILURE) {
-		mutex_exit(&chptr->ch_lock);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "!mixer: send_audio_multi() couldn't allocate "
-		    "conversion buffer, audio lost");
-		return;
-	}
-
-	/* convert to canonical format */
-	am_convert_to_int(buf, chpptr->acp_ch_rconv1, samples,
-	    info->record.precision, info->record.encoding, stpptr->am_rflags);
-
-	/* send to the application */
-	am_send_audio_common(chptr, chpptr->acp_ch_rconv1, samples);
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_send_audio_multi() done", buf);
-
-}	/* am_send_audio_multi() */
-
-/*
- * am_send_audio_trad_compat()
- *
- * Description:
- *	This routine is used by traditional Codecs to send a single recorded
- *	audio stream up to applications in COMPAT mode. The data is converted
- *	to the applications format, except the sample rate, which the hardware
- *	must match.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instance's state
- *	audio_apm_info_t	*apm_infop	Personality module data struct
- *	int			*buf		The buffer to get audio from
- *	int			samples		The # of samples the Codec sent
- *
- * Returns:
- *	void
- */
-static void
-am_send_audio_trad_compat(audio_state_t *statep, audio_apm_info_t *apm_infop,
-	int *buf, int samples)
-{
-	audio_ch_t		*chptr;
-	am_ch_private_t		*chpptr;
-	audio_info_t		*info;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	size_t			size;
-	size_t			size_plus_rb;
-	uint_t			channels;
-	uint_t			hw_channels;
-	int			already_active;
-	int			i;
-	int			max_chs;
-
-	ATRACE("in am_send_audio_trad_compat()", statep);
-	ASSERT(statep);
-
-	/* get the number of chs for this instance */
-	max_chs = statep->as_max_chs;
-
-	/* go through the chs looking for the only recording AUDIO ch */
-	for (i = 0, chptr = &statep->as_channels[0]; i < max_chs;
-	    i++, chptr++) {
-
-		/* lock the channel before we check it out */
-		mutex_enter(&chptr->ch_lock);
-
-		/* skip non-AUDIO and unallocated channels */
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    chptr->ch_info.dev_type != AUDIO ||
-		    chptr->ch_info.pid == 0) {
-
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* make sure this channel is valid */
-		chpptr = chptr->ch_private;
-		info = chptr->ch_info.info;
-		if (chpptr == NULL || info == NULL ||
-		    (chpptr->acp_flags & AM_CHNL_OPEN) == 0) {
-			mutex_exit(&chptr->ch_lock);
-			ATRACE("am_get_audio_trad_compat() channel closed",
-			    chptr);
-			continue;
-		}
-
-		/* make sure this channel is reading */
-		if (!chpptr->acp_reading) {
-			ATRACE("am_send_audio_trad_compat() not recording",
-			    chpptr);
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* skip paused AUDIO channels */
-		if (info->record.pause) {
-			mutex_exit(&chptr->ch_lock);
-			return;
-		}
-
-		mutex_exit(&chptr->ch_lock);
-
-		ATRACE_32("am_send_audio_trad_compat() found channel", i);
-		break;
-	}
-	if (i >= max_chs) {
-		ATRACE("am_send_audio_trad_compat() done, no rec channel", buf);
-		return;
-	}
-
-	/* make sure we can put the audio before we waste time converting it */
-	switch (am_test_canputnext(chptr)) {
-	case AM_CHNL_CLOSED:
-		/* Channel closed on us */
-		ATRACE("am_send_audio_trad_compat() channel closed", chptr);
-		return;
-	case AUDIO_FAILURE:
-		am_send_signal(statep, stpptr);
-		info->record.active = 0;
-		info->record.error = 1;
-		ATRACE_32("am_send_audio_trad_compat() ch flow controlled",
-		    chptr->ch_info.ch_number);
-		return;
-	case AUDIO_SUCCESS:
-				/* FALLTHROUGH */
-	default:
-		break;
-	}
-
-	mutex_enter(&chptr->ch_lock);
-
-	/*
-	 * We had to free the lock above to make warlock happy. Unfortunately
-	 * it's possible that the channel closed between releasing and
-	 * reacquiring the lock. So we have to check again.
-	 */
-	if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    chptr->ch_info.dev_type != AUDIO || chptr->ch_info.pid == 0) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_send_audio_trad_compat() ch closed on us", chptr);
-		return;
-	}
-
-	hw_channels = stpptr->am_hw_rchs;
-	channels = info->record.channels;
-
-	size = (samples + AM_EXTRA_SAMPLES) << AM_TIMES_8_SHIFT;
-	size_plus_rb = size + chpptr->acp_ch_rbuf_size;
-
-	if (am_update_conv_buffer(chptr, size_plus_rb, AUDIO_RECORD) ==
-		AUDIO_FAILURE) {
-		mutex_exit(&chptr->ch_lock);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "!mixer: send_audio_trad(C) couldn't allocate "
-		    "conversion buffer, audio lost");
-		return;
-	}
-
-	/* convert between mono <---> stereo */
-	am_convert_int_mono_stereo(buf, chpptr->acp_ch_rconv1, &samples,
-	    hw_channels, channels);
-
-	/* now convert to the application's format */
-	am_convert_to_format(chpptr->acp_ch_rconv1, chpptr->acp_ch_rconv2,
-	    samples, info->record.precision, info->record.encoding, 0);
-
-	/* update recorded sample count */
-	hw_info->record.samples += samples / hw_channels;
-
-	/* send to the application */
-	already_active = info->record.active;
-	am_send_audio_common(chptr, chpptr->acp_ch_rconv2, samples);
-
-	mutex_exit(&chptr->ch_lock);
-
-	if (already_active != info->record.active) {
-		am_send_signal(statep, stpptr);
-	}
-
-	ATRACE("am_send_audio_trad_compat() done", buf);
-
-}	/* am_send_audio_trad_compat() */
-
-/*
- * am_send_audio_trad_mixer()
- *
- * Description:
- *	This routine is used by traditional Codecs to send multiple recorded
- *	audio stream up to applications in MIXER mode. The data is converted
- *	to the applications format, including sample rate.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instance's state
- *	audio_apm_info_t	*apm_infop	Personality module data struct
- *	int			*buf		The buffer to get audio from
- *	int			samples		The # of samples the Codec sent
- *
- * Returns:
- *	void
- */
-static void
-am_send_audio_trad_mixer(audio_state_t *statep, audio_apm_info_t *apm_infop,
-	int *buf, const int samples)
-{
-	audio_ch_t		*chptr;
-	am_ch_private_t		*chpptr;
-	audio_info_t		*info;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ad_src_entry_t	*rsrs = ad_infop->ad_record.ad_conv;
-	int			*conv_data;
-	size_t			size;
-	size_t			size_plus_rb;
-	uint_t			channels;
-	uint_t			hw_channels;
-	int			already_active;
-	int			i;
-	int			max_chs;
-	int			tmp_samples;
-
-	ATRACE("in am_send_audio_trad_mixer()", statep);
-	ASSERT(statep);
-
-	/* get the number of chs for this instance */
-	max_chs = statep->as_max_chs;
-
-	hw_channels = stpptr->am_hw_rchs;
-	ATRACE_32("am_send_audio_trad_mixer() am_in_chs", stpptr->am_in_chs);
-
-	/* go through the chs looking for each AUDIO ch */
-	for (i = 0, chptr = &statep->as_channels[0];
-	    i < max_chs; i++, chptr++) {
-
-		/* lock the channel before we check it out */
-		mutex_enter(&chptr->ch_lock);
-
-		/* skip non-AUDIO and unallocated channels */
-		if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    chptr->ch_info.dev_type != AUDIO ||
-		    chptr->ch_info.pid == 0) {
-
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* make sure this channel is valid */
-		chpptr = chptr->ch_private;
-		info = chptr->ch_info.info;
-		if (chpptr == NULL || info == NULL ||
-		    (chpptr->acp_flags & AM_CHNL_OPEN) == 0) {
-			mutex_exit(&chptr->ch_lock);
-			ATRACE("am_get_audio_trad_mixer() channel closed",
-			    chptr);
-			continue;
-		}
-
-		/* make sure this channel is reading */
-		if (!chpptr->acp_reading) {
-			ATRACE("am_send_audio_trad_mixer() not recording",
-			    chpptr);
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* skip paused AUDIO channels */
-		if (info->record.pause) {
-			mutex_exit(&chptr->ch_lock);
-			continue;
-		}
-
-		/* make sure we can put the audio before we waste time */
-		mutex_exit(&chptr->ch_lock);
-
-		switch (am_test_canputnext(chptr)) {
-		case AM_CHNL_CLOSED:
-			/* Channel closed on us */
-			ATRACE("am_send_audio_trad_mixer() channel closed",
-			    chptr);
-			return;
-		case AUDIO_FAILURE:
-			am_send_signal(statep, stpptr);
-			info->record.active = 0;
-			info->record.error = 1;
-				ATRACE_32(
-				    "am_send_audio_trad_mixer() ch flow "
-				    "controlled", chptr->ch_info.ch_number);
-			continue;
-		case AUDIO_SUCCESS:
-					/* FALLTHROUGH */
-		default:
-			break;
-		}
-		mutex_enter(&chptr->ch_lock);
-
-		ATRACE_32("am_send_audio_trad_mixer() found channel", i);
-
-		/* don't let samples get modified */
-		tmp_samples = samples;
-
-		channels = info->record.channels;
-
-		/* use the mode lock to keep the mode from switching */
-		mutex_enter(&stpptr->am_mode_lock);
-
-		/*
-		 * Make sure there's enough memory to convert to int. We
-		 * ask for more so that after SRC we hopefully don't need
-		 * to free and allocate again. We also need to make sure
-		 * we've got enough to go from mono to stereo.
-		 */
-		size = (tmp_samples + AM_EXTRA_SAMPLES) << AM_TIMES_8_SHIFT;
-		size_plus_rb = size + chpptr->acp_ch_rbuf_size;
-
-		if (am_update_conv_buffer(chptr, size_plus_rb, AUDIO_RECORD) ==
-		    AUDIO_FAILURE) {
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!mixer: send_audio_trad(C) couldn't "
-			    "allocate conversion buffer, audio lost");
-			return;
-		}
-
-		/* convert between mono <---> stereo */
-		am_convert_int_mono_stereo(buf,
-			(int *)((char *)chpptr->acp_ch_rconv1 +
-			chpptr->acp_ch_rbuf_size), &tmp_samples, hw_channels,
-			channels);
-
-		/*
-		 * Make sure we have the buffers to perform sample rate
-		 * conversion, then do the sample rate conversion.
-		 */
-		if (am_update_src_buffer(chptr, samples, hw_channels,
-		    AUDIO_RECORD) == AUDIO_FAILURE) {
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!mixer: send_audio_trad(M) couldn't "
-			    "allocate src buffer, audio lost");
-			continue;
-		}
-
-		conv_data = rsrs->ad_src_convert(AM_SRC_CHPTR2HDL(chptr),
-		    info->record.channels, AUDIO_RECORD, chpptr->acp_ch_rconv1,
-		    chpptr->acp_ch_rsrc1, chpptr->acp_ch_rsrc2, &tmp_samples);
-
-		ASSERT(tmp_samples <= (chpptr->acp_ch_rsrc_siz >>
-			AM_TIMES_4_SHIFT));
-
-		if (!conv_data) {
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!mixer: send_audio_trad(M) sample "
-			    "rate conversion failed, audio lost");
-			continue;
-		}
-
-		/*
-		 * Make sure there's enough memory to convert to format.
-		 * We also need to make sure we've got enough to go from
-		 * mono to stereo
-		 */
-		size = tmp_samples << AM_TIMES_8_SHIFT;
-		size_plus_rb = size + chpptr->acp_ch_rbuf_size;
-		if (am_update_conv_buffer(chptr, size_plus_rb, AUDIO_RECORD) ==
-		    AUDIO_FAILURE) {
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!mixer: send_audio_trad(C) couldn't "
-			    "allocate conversion buffer, audio lost");
-			return;
-		}
-
-		/* apply gain to the sample rate converted audio */
-		am_apply_gain_balance(conv_data, tmp_samples, channels,
-		    info->record.gain, info->record.balance);
-
-		/* now convert to the application's format */
-		am_convert_to_format(conv_data, chpptr->acp_ch_rconv2,
-		    tmp_samples, info->record.precision, info->record.encoding,
-		    0);
-
-		mutex_exit(&stpptr->am_mode_lock);
-
-		/* send to the application */
-		already_active = info->record.active;
-		am_send_audio_common(chptr, chpptr->acp_ch_rconv2, tmp_samples);
-
-		ATRACE("am_send_audio_trad_mixer() done with channel", chptr);
-
-		mutex_exit(&chptr->ch_lock);
-
-		if (already_active != info->record.active) {
-			am_send_signal(statep, stpptr);
-		}
-
-	}
-
-	/* update recorded sample count */
-	hw_info->record.samples += samples / hw_channels;
-
-	ATRACE("am_send_audio_trad_mixer() done", buf);
-
-}	/* am_send_audio_trad_mixer() */
-
-/*
- * am_serialize_ad_access()
- *
- * Description:
- *	This routine is used to guarantee that all of the calls into the
- *	audio driver, except for ad_ioctl() and ad_iocdata(), are serialized.
- *
- * Arguments:
- *	am_apm_private_t *stpptr	Ptr to APM private data
- *
- * Returns:
- *	Void
- */
-static void
-am_serialize_ad_access(am_apm_private_t *stpptr)
-{
-	ATRACE("in am_serialize_ad_access()", stpptr);
-	ATRACE("am_serialize_ad_access() curthread", curthread);
-
-	/* wait for all other calls into the audio driver to return */
-	mutex_enter(&stpptr->am_ad_lock);
-
-	while (stpptr->am_ad_in & (AM_APM_IN_DRIVER|AM_APM_FREEZE)) {
-		ATRACE("am_serialize_ad_access() in cv_wait()", curthread);
-		cv_wait(&stpptr->am_ad_cv, &stpptr->am_ad_lock);
-		ATRACE("am_serialize_ad_access() cv_wait() returned",
-		    curthread);
-	}
-
-	ASSERT(!(stpptr->am_ad_in & AM_APM_IN_DRIVER));
-
-	/* we have permission to enter, so flag as busy */
-	stpptr->am_ad_in |= AM_APM_IN_DRIVER;
-
-	mutex_exit(&stpptr->am_ad_lock);
-
-	ATRACE("am_serialize_ad_access() done", curthread);
-
-}	/* am_serialize_ad_access() */
-
-/*
- * Task queue callbacks.
- */
-
-/*
- * am_hw_task()
- *
- * Description:
- *	Called by the task queue to update the hardware state based on
- *	input from the audio driver. This is usually driven by the user
- *	pressing a button, like volume up/down, on the audio device.
- *
- *	NOTE: There isn't an input mute like there is an output mute, so we
- *		set the input gain to 0 and save the old gain for unmute. If
- *		another app raised the input gain then the input gain isn't
- *		0 and thus we are no longer muted.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_hw_task(void *arg)
-{
-	audio_state_t		*statep =
-				    ((am_state_ch_args_t *)arg)->asca_statep;
-	audio_apm_info_t	*apm_infop =
-				    ((am_state_ch_args_t *)arg)->asca_apm_infop;
-	audio_info_t		*hw_info =
-				    (audio_info_t *)apm_infop->apm_ad_state;
-	am_ad_info_t		*ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	int			cmd = ((am_state_ch_args_t *)arg)->asca_cmd;
-	int			dir = ((am_state_ch_args_t *)arg)->asca_dir;
-	int			value = ((am_state_ch_args_t *)arg)->asca_value;
-	int			balance;
-	int			gain;
-	int			send_signal = 0;
-	uint_t			channels;
-
-	ATRACE("in am_hw_task() statep", statep);
-	ATRACE_32("am_hw_task() cmd", cmd);
-	ATRACE_32("am_hw_task() dir", dir);
-	ATRACE_32("am_hw_task() value", value);
-	ASSERT(statep);
-
-	switch (cmd) {
-	case AM_HWSC_SET_GAIN_ABS:
-		gain = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.gain : hw_info->record.gain);
-		balance = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.balance : hw_info->record.balance);
-		channels = (dir == AUDIO_PLAY) ?
-		    stpptr->am_hw_pchs : stpptr->am_hw_rchs;
-
-		/*
-		 * Check limits, jump out if we are already at the max or min
-		 * value
-		 */
-		if (gain >= AUDIO_MAX_GAIN) {
-			ATRACE_32("am_hw_task() ABS gain already at max", gain);
-			break;
-		} else if (gain <= AUDIO_MIN_GAIN) {
-			ATRACE_32("am_hw_task() ABS gain already at min", gain);
-			break;
-		}
-
-		/* apply the new absolute gain */
-		gain = value;
-
-		/* make sure the result isn't too big or small */
-		if (gain > AUDIO_MAX_GAIN) {
-			gain = AUDIO_MAX_GAIN;
-		} else if (gain < AUDIO_MIN_GAIN) {
-			gain = AUDIO_MIN_GAIN;
-		}
-		ATRACE_32("am_hw_task() new absolute gain", gain);
-
-		if (am_set_gain(statep, apm_infop, channels, (uint_t)gain,
-		    (uint_t)balance, dir, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_SUCCESS) {
-			if (dir == AUDIO_PLAY) {
-				hw_info->play.gain = gain;
-			} else {
-				hw_info->record.gain = gain;
-			}
-		}
-
-		send_signal++;
-
-		break;
-	case AM_HWSC_SET_BAL_ABS:
-		gain = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.gain : hw_info->record.gain);
-		balance = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.balance : hw_info->record.balance);
-		channels = (dir == AUDIO_PLAY) ?
-		    stpptr->am_hw_pchs : stpptr->am_hw_rchs;
-
-		/*
-		 * Check limits, jump out if we are already at the max or min
-		 * value
-		 */
-		if (balance >= AUDIO_RIGHT_BALANCE) {
-			ATRACE_32("am_hw_task() ABS balance"
-			    " already right", balance);
-			break;
-		} else if (balance <= AUDIO_LEFT_BALANCE) {
-			ATRACE_32("am_hw_task() ABS balance"
-			    " already left", balance);
-			break;
-		}
-
-		/* apply the new absolute balance */
-		balance = value;
-
-		/* make sure the result isn't too big or small */
-		if (balance > AUDIO_RIGHT_BALANCE) {
-			balance = AUDIO_RIGHT_BALANCE;
-		} else if (balance < AUDIO_LEFT_BALANCE) {
-			balance = AUDIO_LEFT_BALANCE;
-		}
-		ATRACE_32("am_hw_task() new absolute balance", balance);
-
-		if (am_set_gain(statep, apm_infop, channels, (uint_t)gain,
-		    (uint_t)balance, dir, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_SUCCESS) {
-			if (dir == AUDIO_PLAY) {
-				hw_info->play.balance = (uchar_t)balance;
-			} else {
-				hw_info->record.balance = (uchar_t)balance;
-			}
-		}
-
-		send_signal++;
-
-		break;
-	case AM_HWSC_SET_GAIN_DELTA:
-		gain = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.gain : hw_info->record.gain);
-		balance = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.balance : hw_info->record.balance);
-		channels = (dir == AUDIO_PLAY) ?
-		    stpptr->am_hw_pchs : stpptr->am_hw_rchs;
-
-		/*
-		 * Check limits, jump out if we are already at the max or min
-		 * value
-		 */
-		if (value > 0 && gain >= AUDIO_MAX_GAIN) {
-			ATRACE_32("am_hw_task() gain already at max", gain);
-			break;
-		} else if (value < 0 && gain <= AUDIO_MIN_GAIN) {
-			ATRACE_32("am_hw_task() gain already at min", gain);
-			break;
-		}
-
-		/* apply delta */
-		gain += value;
-
-		/* make sure the result isn't too big or small */
-		if (gain > AUDIO_MAX_GAIN) {
-			gain = AUDIO_MAX_GAIN;
-		} else if (gain < AUDIO_MIN_GAIN) {
-			gain = AUDIO_MIN_GAIN;
-		}
-
-		if (am_set_gain(statep, apm_infop, channels, (uint_t)gain,
-		    (uint_t)balance, dir, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_SUCCESS) {
-			if (dir == AUDIO_PLAY) {
-				hw_info->play.gain = gain;
-			} else {
-				hw_info->record.gain = gain;
-			}
-		}
-
-		send_signal++;
-
-		break;
-	case AM_HWSC_SET_BAL_DELTA:
-		gain = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.gain : hw_info->record.gain);
-		balance = (int)((dir == AUDIO_PLAY) ?
-		    hw_info->play.balance : hw_info->record.balance);
-		channels = (dir == AUDIO_PLAY) ?
-		    stpptr->am_hw_pchs : stpptr->am_hw_rchs;
-
-		/*
-		 * Check limits, jump out if we are already at the max or min
-		 * value
-		 */
-		if (value > 0 && balance >= AUDIO_RIGHT_BALANCE) {
-			ATRACE_32("am_hw_task() bal. already right", balance);
-			break;
-		} else if (value < 0 && balance <= AUDIO_LEFT_BALANCE) {
-			ATRACE_32("am_hw_task() bal. already left", balance);
-			break;
-		}
-
-		/* apply delta */
-		balance += value;
-
-		/* make sure the result isn't too big or small */
-		if (balance > AUDIO_RIGHT_BALANCE) {
-			balance = AUDIO_RIGHT_BALANCE;
-		} else if (balance < AUDIO_LEFT_BALANCE) {
-			balance = AUDIO_LEFT_BALANCE;
-		}
-
-		if (am_set_gain(statep, apm_infop, channels, (uint_t)gain,
-		    (uint_t)balance, dir, AM_SET_CONFIG_BOARD, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_SUCCESS) {
-			if (dir == AUDIO_PLAY) {
-				hw_info->play.balance = (uchar_t)balance;
-			} else {
-				hw_info->record.balance = (uchar_t)balance;
-			}
-		}
-
-		send_signal++;
-
-		break;
-	case AM_HWSC_MUTE_TOGGLE:
-		if (dir == AUDIO_PLAY) {
-			/* toggle mute */
-			if (hw_info->output_muted) {
-				value = AM_NOT_MUTED;
-			} else {
-				value = AM_MUTED;
-			}
-
-			/* set new mute */
-			ad_infop = apm_infop->apm_ad_infop;
-			if (am_ad_set_config(statep, stpptr, ad_infop,
-			    AM_SET_CONFIG_BOARD, AM_OUTPUT_MUTE, AUDIO_PLAY,
-			    value, NULL, AM_SERIALIZE) == AUDIO_SUCCESS) {
-				hw_info->output_muted = (uchar_t)value;
-			}
-		} else {
-			if (hw_info->record.gain) {
-				stpptr->am_save_hw_rgain = hw_info->record.gain;
-				value = AUDIO_MIN_GAIN;
-			} else {
-				value = stpptr->am_save_hw_rgain;
-			}
-
-			if (am_set_gain(statep, apm_infop,
-			    stpptr->am_hw_rchs, (uint_t)value,
-			    hw_info->record.balance, AUDIO_RECORD,
-			    AM_SET_CONFIG_BOARD, AM_NO_FORCE, AM_SERIALIZE) ==
-			    AUDIO_SUCCESS) {
-				hw_info->record.gain = value;
-			}
-		}
-
-		send_signal++;
-
-		break;
-	default:
-		ATRACE_32("am_hw_task() unrecognized command", cmd);
-		break;
-	}
-
-	/* free the argument memory */
-
-	kmem_free(arg, sizeof (am_state_ch_args_t));
-
-	if (send_signal) {
-		am_send_signal(statep, stpptr);
-	}
-
-	ATRACE("am_hw_task() done", statep);
-
-}	/* am_hw_task() */
diff --git a/usr/src/uts/common/io/audio/sada/mixer/am_ioctl.c b/usr/src/uts/common/io/audio/sada/mixer/am_ioctl.c
deleted file mode 100644
index 86d849f9b1..0000000000
--- a/usr/src/uts/common/io/audio/sada/mixer/am_ioctl.c
+++ /dev/null
@@ -1,5376 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * This file contains the code for implementing the ioctl()s that the audio.7I
- * and mixer.7I man pages define. This file also contains private routines to
- * support these calls.
- *
- * am_wput() in am_main.c just grabs the M_IOCTL and M_IOCDATA messages and
- * sends them to either am_wioctl() or am_wiocdata(), where all processing
- * takes place. am_svc() doesn't do any ioctl() processing at all.
- *
- * The following ioctl()s are supported:
- *	AUDIO_DIAG_LOOPBACK
- *	AUDIO_DRAIN
- *	AUDIO_GETINFO
- *	AUDIO_SETINFO
- *	AUDIO_GETDEV
- *	AUDIO_MIXER_MULTIPLE_OPEN
- *	AUDIO_MIXER_SINGLE_OPEN
- *	AUDIO_MIXER_GET_SAMPLE_RATES
- *	AUDIO_MIXERCTL_GETINFO
- *	AUDIO_MIXERCTL_SETINFO
- *	AUDIO_MIXERCTL_GET_CHINFO
- *	AUDIO_MIXERCTL_SET_CHINFO
- *	AUDIO_MIXERCTL_GET_MODE
- *	AUDIO_MIXERCTL_SET_MODE
- *
- * Most of the ioctl()s copy in a data structure for use by the ioctl().
- * am_wioctl() or am_wiocdata() will request the data. Based on the ioctl()
- * it then creates a data structure and enqueues a task request to execute
- * that ioctl() in a separate thread. This allows am_wput() and am_wsvc() to
- * continue working while the ioctl() is processed. When the ioctl() in it's
- * own thread is complete it creates the appropriate message and sends it
- * back up the queue. Further processing to copy out and ack/nack is done by
- * am_wiocdata().
- *
- * A task queue is used to serialize all access to the hardware state
- * structures and the hardware. This greatly simplifies the locking model.
- * When closing we wait for all of the tasks to complete. This may introduce
- * a delay in closing, but tests with 40 playing channels shows no noticable
- * delays.
- *
- * Two ioctl()s are not placed on the task queue. They are:
- *
- *	AUDIO_GETINFO - Returns static data and thus the hardware state is
- *		irrelevant.
- *	AUDIO_DRAIN - This ioctl() doesn't change or get the device state.
- *		It is also a very long lived ioctl() and is dependent on how
- *		much audio is queued up. We don't want to block other channels
- *		from being open()ed or their ioctl()s. Because this is a long
- *		lived ioctl() it is handled in am_wsvc() instead of am_wput().
- *		Otherwise it would block am_wput() and this is against the
- *		rules.
- *
- * Signals are generated when the hardware is modified. The signal is sent
- * after the ioctl()'s ack or nack has been sent. That way we don't get an
- * interrupted system call.
- *
- * These routines are provided for use by the other mixer source code files:
- *	am_wiocdata()
- *	am_wioctl()
- *	am_audio_drained()
- *	am_audio_set_info()
- *	am_set_format()
- *	am_set_gain()
- */
-
-#include <sys/modctl.h>
-#include <sys/debug.h>
-#include <sys/kmem.h>
-#include <sys/errno.h>
-#include <sys/stropts.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/strsun.h>
-#include <sys/file.h>
-#include <sys/fcntl.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_apm.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/impl/audio_mixer_impl.h>
-
-/*
- * Local routine prototypes used only by this file.
- */
-static int am_ck_bits_set32(uint_t);
-static void am_exit_task(audio_ch_t *);
-static void am_fix_info(audio_ch_t *, audio_info_t *);
-static void am_fix_play_pause(audio_ch_t *);
-static void am_mixer_task_acknack(audio_i_state_t *, audio_ch_t *,
-	queue_t *, mblk_t *, am_ioctl_args_t *, int);
-static void am_restart(audio_state_t *, audio_info_t *);
-static int am_sched_task(queue_t *, mblk_t *, audio_ch_t *,
-	void (*func)(void *));
-static int am_set_compat_mode(audio_ch_t *, am_ad_info_t *, audio_ch_t *,
-	audio_ch_t *);
-static int am_set_mixer_mode(audio_ch_t *, am_ad_info_t *, am_apm_private_t *,
-	audio_ch_t *, audio_ch_t *);
-static int am_wiocdata_mixerctl_chinfo(queue_t *, mblk_t *, audio_i_state_t *);
-static int am_wiocdata_mixerctl_get_chinfo(queue_t *, mblk_t *,
-	audio_i_state_t *);
-static int am_wiocdata_sr(queue_t *, mblk_t *, struct copyreq *,
-	audio_i_state_t *);
-static int am_wioctl_copyin(queue_t *, mblk_t *, audio_ch_t *,
-	audio_i_state_t *, int);
-static void am_wioctl_drain(queue_t *, mblk_t *, audio_ch_t *,
-	struct copyreq *);
-static int am_wioctl_getdev(queue_t *, mblk_t *, audio_ch_t *,
-	audio_i_state_t *);
-
-/*
- * Taskq callbacks.
- */
-static void am_diag_loopback_task(void *);
-static void am_get_chinfo_task(void *);
-static void am_get_mode_task(void *);
-static void am_getinfo_task(void *);
-static void am_mixerctl_getinfo_task(void *);
-static void am_mixerctl_setinfo_task(void *);
-static void am_multiple_open_task(void *);
-static void am_sample_rate_task(void *);
-static void am_set_chinfo_task(void *);
-static void am_set_mode_task(void *);
-static void am_setinfo_task(void *);
-static void am_single_open_task(void *);
-
-/* this simulates the rw lock handling by taskq framework */
-#ifdef __lock_lint
-extern krwlock_t q_lock;
-
-static void
-am_enter_rwlock()
-{
-	_NOTE(READ_LOCK_ACQUIRED_AS_SIDE_EFFECT(&q_lock));
-	rw_enter(&q_lock, RW_READER);
-}
-
-static void
-am_release_rwlock()
-{
-	_NOTE(LOCK_RELEASED_AS_SIDE_EFFECT(&q_lock));
-	rw_exit(&q_lock);
-}
-#else
-#define	am_enter_rwlock()
-#define	am_release_rwlock()
-#endif
-
-/*
- * The main routines for this file.
- */
-
-/*
- * am_wiocdata()
- *
- * Description:
- *	This routine is called by am_wput() to process all M_IOCDATA
- *	messages.
- *
- *	We only support transparent ioctls.
- *
- *	This routine also is used to return a IOCNAK if the state pointer
- *	or the channel pointer, setup in am_wsvc(), are invalid.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- *	WARNING: Don't forget to free the mblk_t struct used to hold private
- *		data. The ack: and nack: jump points take care of this.
- *
- *	WARNING: Don't free the private mblk_t structure if the command is
- *		going to call qreply(). This frees the private data that will
- *		be needed for the next M_IOCDATA message.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *
- * Returns:
- *	0			Always returns a 0, becomes a return for
- *				am_wsvc()
- */
-int
-am_wiocdata(queue_t *q, mblk_t *mp, audio_ch_t *chptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	struct copyreq		*cqp;
-	struct copyresp		*csp;
-	audio_i_state_t		*cmd;
-	int			error = 0;
-	int			send_sig = 0;
-
-	ATRACE("in am_wiocdata()", chptr);
-	ATRACE_32("am_wiocdata() channel type", chptr->ch_info.dev_type);
-
-	ASSERT(statep);
-	ASSERT(!MUTEX_HELD(&apm_infop->apm_lock));
-
-	csp = (struct copyresp *)mp->b_rptr;	/* setup copy response ptr */
-	cqp = (struct copyreq *)mp->b_rptr;	/* setup copy request ptr */
-	cmd = (audio_i_state_t *)csp->cp_private;	/* get state info */
-
-	/* make sure we've got a good return value */
-	if (csp->cp_rval) {
-		ATRACE("am_wiocdata() bad return value", csp->cp_rval);
-		error = EINVAL;
-		goto done;
-	}
-
-	/*
-	 * Work through the iocdata messages. These are arranged so that
-	 * the messages that need to do further work are ordered first and
-	 * then the ACKs.
-	 */
-	if (cmd != NULL) {
-		ATRACE_32("am_wiocdata() command", cmd->ais_command);
-		switch (cmd->ais_command) {
-
-		case AM_COPY_IN_AUDIOINFO:
-			/* AUDIO_SETINFO */
-			ATRACE("am_wiocdata() AM_COPY_IN_AUDIOINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_SETINFO);
-
-			error = am_sched_task(q, mp, chptr, am_setinfo_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_DIAG_LOOPB:
-			/* AUDIO_DIAG_LOOPBACK */
-			ATRACE("am_wiocdata() AM_COPY_IN_DIAG_LOOPB", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_DIAG_LOOPBACK);
-
-			error = am_sched_task(q, mp, chptr,
-			    am_diag_loopback_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_SAMP_RATES:
-			/* AUDIO_MIXER_GET_SAMPLE_RATES */
-			ATRACE("am_wiocdata() AM_COPY_IN_SAMP_RATES", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXER_GET_SAMPLE_RATES);
-
-			error = am_wiocdata_sr(q, mp, cqp, cmd);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_SAMP_RATES2:
-			/* AUDIO_MIXER_GET_SAMPLE_RATES */
-			ATRACE("am_wiocdata() AM_COPY_IN_SAMP_RATES2", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXER_GET_SAMPLE_RATES);
-
-			error = am_sched_task(q, mp, chptr,
-			    am_sample_rate_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_MIXCTLINFO:
-			/* AUDIO_MIXERCTL_SETINFO */
-			ATRACE("am_wiocdata() AM_COPY_IN_MIXCTLINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_SETINFO);
-
-			error = am_sched_task(q, mp, chptr,
-			    am_mixerctl_setinfo_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_MIXCTL_GET_CHINFO:
-			/* AUDIO_MIXERCTL_GET_CHINFO */
-			ATRACE("am_wiocdata() AM_COPY_IN_MIXCTL_CHINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_GET_CHINFO);
-
-			error = am_sched_task(q, mp, chptr, am_get_chinfo_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_OUT_MIXCTL_GET_CHINFO:
-			/* AUDIO_MIXERCTL_GET/SET_CHINFO */
-			ATRACE("am_wiocdata() AM_COPY_OUT_MIXCTL_GET_CHINFO",
-			    chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_GET_CHINFO ||
-			    csp->cp_cmd == AUDIO_MIXERCTL_SET_CHINFO);
-
-			error = am_wiocdata_mixerctl_get_chinfo(q, mp, cmd);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_MIXCTL_SET_CHINFO:
-			/* AUDIO_MIXERCTL_SET_CHINFO */
-			ATRACE("am_wiocdata() AM_COPY_IN_MIXCTL_CHINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_SET_CHINFO);
-
-			error = am_wiocdata_mixerctl_chinfo(q, mp, cmd);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_MIXCTL_SET_CHINFO2:
-			/* AUDIO_MIXERCTL_SET_CHINFO */
-			ATRACE("am_wiocdata() AM_COPY_IN_MIXCTL_CHINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_SET_CHINFO);
-
-			error = am_sched_task(q, mp, chptr, am_set_chinfo_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_IN_MIXCTL_MODE:
-			/* AUDIO_MIXERCTL_SET_MODE */
-			ATRACE("am_wiocdata() AM_COPY_IN_MIXCTL_MODE", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_SET_MODE);
-
-			error = am_sched_task(q, mp, chptr, am_set_mode_task);
-			if (error) {
-				goto done;
-			}
-			return (0);
-
-		case AM_COPY_OUT_AUDIOINFO:
-			/* AUDIO_GETINFO */
-			ATRACE("am_wiocdata() AM_COPY_OUT_AUDIOINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_GETINFO);
-
-			goto done;
-
-		case AM_COPY_OUT_AUDIOINFO2:
-			/* AUDIO_SETINFO */
-			ATRACE("am_wiocdata() AM_COPY_OUT_AUDIOINFO2", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_SETINFO);
-
-			send_sig++;
-
-			goto done;
-
-		case AM_COPY_OUT_GETDEV:
-			/* AUDIO_GETDEV */
-			ATRACE("am_wiocdata() AM_COPY_OUT_GETDEV", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_GETDEV);
-
-			goto done;
-
-		case AM_COPY_OUT_SAMP_RATES:
-			/* AUDIO_MIXER_GET_SAMPLE_RATES */
-			ATRACE("am_wiocdata() AM_COPY_OUT_SAMP_RATES", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXER_GET_SAMPLE_RATES);
-
-			goto done;
-
-		case AM_COPY_OUT_MIXCTLINFO:
-			/* AUDIO_MIXERCTL_GET/SETINFO */
-			ATRACE("am_wiocdata() AM_COPY_OUT_MIXCTLINFO", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_GETINFO ||
-			    csp->cp_cmd == AUDIO_MIXERCTL_SETINFO);
-
-			/* generate a signal ONLY when we set the info */
-			if (csp->cp_cmd == AUDIO_MIXERCTL_SETINFO) {
-				send_sig++;
-			}
-
-			goto done;
-
-		case AM_COPY_OUT_MIXCTL_GET_CHINFO2:
-			/* AUDIO_MIXERCTL_GET/SET_CHINFO */
-			ATRACE("am_wiocdata() AM_COPY_OUT_MIXCTL_GET_CHINFO2",
-			    chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_GET_CHINFO ||
-			    csp->cp_cmd == AUDIO_MIXERCTL_SET_CHINFO);
-
-			/* generate a signal ONLY when we set the info */
-			if (csp->cp_cmd == AUDIO_MIXERCTL_SET_CHINFO) {
-				send_sig++;
-			}
-
-			goto done;
-
-		case AM_COPY_OUT_MIXCTL_MODE:
-			/* AUDIO_MIXERCTL_GET_MODE */
-			ATRACE("am_wiocdata() AM_COPY_OUT_MIXCTL_MODE", chptr);
-			ASSERT(csp->cp_cmd == AUDIO_MIXERCTL_GET_MODE);
-
-			goto done;
-
-		default:
-			ATRACE("am_wiocdata() No framework cmds found, "
-			    "check driver entry points next", chptr);
-			break;
-		}
-	}
-
-	/* see if we have an entry pt in the Audio Driver */
-	if (ad_infop->ad_entry->ad_iocdata) {
-		/* we do, so call it */
-		ATRACE("am_wiocdata(): "
-		    "calling Audio Driver iocdata() routine",
-		    ad_infop->ad_entry);
-
-		switch (ad_infop->ad_entry->ad_iocdata(
-		    AUDIO_STATE2HDL(statep), chptr->ch_info.ch_number,
-		    q, mp, &error)) {
-		case AM_WIOCDATA:
-			return (0);
-		case AM_ACK:
-			goto done;
-		case AM_NACK:
-			goto done;
-		default:
-			break;
-		}
-	}
-
-	/* no driver entry, so we nack unrecognized iocdata cmds */
-	ATRACE("am_wiocdata() no entry", chptr);
-	error = EINVAL;
-
-	/* Done checking */
-	ATRACE("am_wiocdata() switch & driver check done", chptr);
-
-done:
-	ATRACE("am_wiocdata() done", chptr);
-
-	if (csp->cp_private) {
-		kmem_free(csp->cp_private, sizeof (audio_i_state_t));
-		csp->cp_private = NULL;
-	}
-	if (cqp->cq_private) {
-		kmem_free(cqp->cq_private, sizeof (audio_i_state_t));
-		cqp->cq_private = NULL;
-	}
-
-	if (error) {
-		miocnak(q, mp, 0, error);
-	} else {
-		miocack(q, mp, 0, 0);
-	}
-
-	if (send_sig) {
-		am_send_signal(statep, stpptr);
-	}
-
-	ATRACE("am_wiocdata() returning success", chptr);
-
-	return (0);
-
-}	/* am_wiocdata() */
-
-/*
- * am_wioctl()
- *
- * Description:
- *	This routine is called by am_wput() to process all M_IOCTL
- *	messages.
- *
- *	We only support transparent ioctls. Since this is a driver we
- *	nack unrecognized ioctls.
- *
- *	This routine also is used to return a IOCNAK if the state pointer
- *	or the channel pointer, setup in am_wsvc(), are invalid.
- *
- *	The following ioctls are supported:
- *		AUDIO_DIAG_LOOPBACK		special diagnostics mode
- *		AUDIO_DRAIN
- *		AUDIO_GETDEV
- *		AUDIO_GETINFO
- *		AUDIO_SETINFO
- *		AUDIO_MIXER_MULTIPLE_OPEN
- *		AUDIO_MIXER_SINGLE_OPEN
- *		AUDIO_MIXER_GET_SAMPLE_RATES
- *		AUDIO_MIXERCTL_GETINFO
- *		AUDIO_MIXERCTL_SETINFO
- *		AUDIO_MIXERCTL_GET_CHINFO
- *		AUDIO_MIXERCTL_SET_CHINFO
- *		AUDIO_MIXERCTL_GET_MODE
- *		AUDIO_MIXERCTL_SET_MODE
- *		unknown		call Audio Driver ioctl() routine
- *
- *	WARNING: There cannot be any locks owned by calling routines.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *
- * Returns:
- *	0			Always returns a 0, becomes a return for
- *				am_wsvc()
- */
-int
-am_wioctl(queue_t *q, mblk_t *mp, audio_ch_t *chptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	struct iocblk		*iocbp;
-	struct copyreq		*cqp;
-	audio_i_state_t		*state = NULL;
-	audio_device_type_e	type = chptr->ch_info.dev_type;
-	int			command;
-	int			error = 0;
-
-	ATRACE("in am_wioctl()", chptr);
-	ATRACE_32("am_wioctl() channel type", chptr->ch_info.dev_type);
-
-	ASSERT(statep);
-	ASSERT(type == AUDIO || type == AUDIOCTL);
-	ASSERT(!MUTEX_HELD(&apm_infop->apm_lock));
-
-	iocbp = (struct iocblk *)mp->b_rptr;	/* pointer to ioctl info */
-	cqp = (struct copyreq *)mp->b_rptr;	/* setup copyreq ptr */
-
-	command = iocbp->ioc_cmd;
-
-	/* make sure this is a transparent ioctl */
-	if (iocbp->ioc_count != TRANSPARENT) {
-		ATRACE_32("am_wioctl() not TRANSPARENT", iocbp->ioc_count);
-		error = EINVAL;
-		goto done;
-	}
-
-	/* get a buffer for priv. data, but only if this isn't an AUDIO_DRAIN */
-	if (command != AUDIO_DRAIN) {
-		if ((state = kmem_zalloc(sizeof (*state), KM_NOSLEEP)) ==
-		    NULL) {
-			ATRACE("am_wioctl() state kmem_zalloc() failed", 0);
-			error = ENOMEM;
-			goto done;
-		}
-	}
-
-	ATRACE_32("am_wioctl() command", iocbp->ioc_cmd);
-	switch (command) {
-
-	case AUDIO_DIAG_LOOPBACK:
-		ATRACE("am_wioctl() AUDIO_DIAG_LOOPBACK", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_DIAG_LOOPBACK);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_DRAIN:
-		ATRACE("am_wioctl() AUDIO_DRAIN", chptr);
-
-		cqp->cq_private = NULL;
-		am_wioctl_drain(q, mp, chptr, cqp);
-		return (0);
-
-	case AUDIO_GETDEV:
-		ATRACE("am_wioctl() AUDIO_GETDEV", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_getdev(q, mp, chptr, state);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_GETINFO:
-		ATRACE("am_wioctl() AUDIO_GETINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_sched_task(q, mp, chptr, am_getinfo_task);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_SETINFO:
-		ATRACE("am_wioctl() AUDIO_SETINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_SETINFO);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXER_MULTIPLE_OPEN:
-		ATRACE("am_wioctl() AUDIO_MIXER_MULTIPLE_OPEN", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-
-		if (chptr->ch_info.dev_type != AUDIO) {
-			error = EINVAL;
-			goto done;
-		}
-
-		error = am_sched_task(q, mp, chptr, am_multiple_open_task);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXER_SINGLE_OPEN:
-		ATRACE("am_wioctl() AUDIO_MIXER_SINGLE_OPEN", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-
-		if (chptr->ch_info.dev_type != AUDIO) {
-			error = EINVAL;
-			goto done;
-		}
-
-		error = am_sched_task(q, mp, chptr, am_single_open_task);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXER_GET_SAMPLE_RATES:
-		ATRACE("am_wioctl() AUDIO_MIXER_GET_SAMPLE_RATES", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_MIXER_GET_SAMPLE_RATES);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_GETINFO:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_GETINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_sched_task(q, mp, chptr, am_mixerctl_getinfo_task);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_SETINFO:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_SETINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_MIXERCTL_SETINFO);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_GET_CHINFO:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_GET_CHINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_MIXERCTL_GET_CHINFO);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_SET_CHINFO:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_SET_CHINFO", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_MIXERCTL_SET_CHINFO);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_GET_MODE:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_GET_MODE", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-
-		/* allowed only on AUDIOCTL channels */
-		if (chptr->ch_info.dev_type != AUDIOCTL) {
-			error = EINVAL;
-			goto done;
-		}
-
-		error = am_sched_task(q, mp, chptr, am_get_mode_task);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	case AUDIO_MIXERCTL_SET_MODE:
-		ATRACE("am_wioctl() AUDIO_MIXERCTL_SET_MODE", chptr);
-
-		cqp->cq_private = (mblk_t *)state;
-		error = am_wioctl_copyin(q, mp, chptr, state,
-		    AUDIO_MIXERCTL_SET_MODE);
-		if (error) {
-			goto done;
-		}
-		return (0);
-
-	default:	/* see if we have an entry pt in the Audio Driver */
-		if (ad_infop->ad_entry->ad_ioctl) {
-			/* we do, so call it */
-			ATRACE("am_wioctl(): "
-			    "calling Audio Driver ioctl() routine",
-			    ad_infop->ad_entry);
-			ASSERT(ad_infop->ad_entry->ad_iocdata);
-
-			switch (ad_infop->ad_entry->ad_ioctl(
-			    AUDIO_STATE2HDL(statep), chptr->ch_info.ch_number,
-			    q, mp, &error)) {
-			case AM_WIOCDATA:
-				return (0);
-			case AM_ACK:
-				goto done;
-			case AM_NACK:
-				goto done;
-			default:
-				break;
-			}
-		}
-
-		/* no - we're a driver, so we nack unrecognized ioctls */
-		ATRACE_32("am_wioctl() default", iocbp->ioc_cmd);
-
-		audio_sup_log(AUDIO_STATE2HDL(statep),
-		    CE_NOTE, "wioctl() unrecognized ioc_cmd: 0x%x",
-		    iocbp->ioc_cmd);
-		error = EINVAL;
-		break;
-	}
-
-	/* we always either ack or nack depending on if error is set or not */
-	ATRACE_32("am_wioctl() switch done", error);
-
-done:
-	ATRACE("am_wioctl() done", chptr);
-
-	/* free allocated state memory */
-	if (state) {
-		kmem_free(state, sizeof (*state));
-	}
-
-	if (error) {
-		miocnak(q, mp, 0, error);
-	} else {
-		miocack(q, mp, 0, 0);
-	}
-
-	ATRACE_32("am_wioctl() returning", error);
-
-	return (0);
-
-}	/* am_wioctl() */
-
-/*
- * Private utilities used by this and other audio mixer files.
- */
-
-/*
- * am_audio_drained()
- *
- * Description:
- *	There's an AUDIO_DRAIN ioctl() that is waiting. When the channel
- *	goes empty this routine is called to send the ack back to the
- *	STREAMS head, which lets the ioctl() return.
- *
- *	We also generate a CV signal so that waiting close()s will wakeup.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *
- * Returns:
- *	void
- */
-void
-am_audio_drained(audio_ch_t *chptr)
-{
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = (am_ch_private_t *)chptr->ch_private;
-	mblk_t			*mp;
-
-	ATRACE("in am_audio_drained()", chptr);
-
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	/* make sure the channel is empty, while locked */
-	if (!(chpptr->acp_flags & AM_CHNL_EMPTY)) {
-		ATRACE("am_audio_drained() returned, not empty", chptr);
-		return;
-	}
-
-	chpptr->acp_flags &= ~(AM_CHNL_ALMOST_EMPTY1|AM_CHNL_ALMOST_EMPTY2|\
-	    AM_CHNL_DRAIN|AM_CHNL_DRAIN_NEXT_INT);
-	mp = chpptr->acp_drain_mp;
-	chpptr->acp_drain_mp = NULL;
-	cv_signal(&chptr->ch_cv);
-	ATRACE("am_audio_drained() MP", mp);
-
-	/*
-	 * By definition we don't have any audio to play thus if we need to
-	 * switch modes we can go ahead and do it now.
-	 */
-	mutex_enter(&stpptr->am_mode_lock);
-	cv_signal(&stpptr->am_mode_cv);
-	mutex_exit(&stpptr->am_mode_lock);
-
-	/* ack only if we have an mblk_t */
-	if (mp) {
-		miocack(WR(chptr->ch_qptr), mp, 0, 0);
-		ATRACE("am_audio_drained() AUDIO_DRAIN acked", mp);
-	}
-
-	ATRACE("am_audio_drained() returning", chptr);
-
-}	/* am_audio_drained() */
-
-/*
- * am_audio_set_info()
- *
- * Description:
- *	This routine double checks the passed in audio_info_t structure to
- *	make sure the values are legal. If they are then they are used to
- *	update the audio hardware. In COMPAT mode all the hardware is updated,
- *	as it is for a multi-stream Codec. However traditional Codecs in MIXER
- *	mode don't update the data format or gain. Everything else can be
- *	updated.
- *
- *	After the checks are completed and the hardware has been updated
- *	the reti pointer is checked. If NULL we are done. Otherwise the
- *	structure pointed to by reti is filled in with the new hardware
- *	configuration.
- *
- *	The mixer only supports a few formats, 16-bit linear and 8-bit
- *	u-law, A-law and linear. Any other format will cause the check to
- *	fail.
- *
- *	We don't bother checking the read only members, silently ignoring any
- *	modifications.
- *
- *	XXX Need to set hardware to original state if error, especially
- *	if src_update() fails. Also, maybe move src_update() up higher so it
- *	can fail before we change hardware. Plus, it's easier to undo
- *	src_update().
- *
- *	NOTE: The Codec's lock must NOT be held when calling this routine.
- *
- *	NOTE: reti will be NULL only when this routine is being called by
- *		am_open().
- *
- *	NOTE: The calling routine is responsible for sending the hardware
- *		change signal.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	audio_info_t	*newi		Pointer to the struct with new values
- *	audio_info_t	*reti		Pointer to the updated struct that is
- *					returned
- *
- * Returns:
- *	AUDIO_SUCCESS			Successful
- *	AUDIO_FAILURE			Failed
- */
-int
-am_audio_set_info(audio_ch_t *chptr, audio_info_t *newi, audio_info_t *reti)
-{
-	audio_info_t		*curi;		/* current state */
-	audio_state_t		*statep = chptr->ch_statep;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	audio_info_t		tempi;		/* approved values in here */
-	audio_device_type_e	type = chptr->ch_info.dev_type;
-	boolean_t		ch_ctl = B_FALSE;
-	boolean_t		new_play_samples = B_FALSE;
-	boolean_t		new_record_samples = B_FALSE;
-	boolean_t		play = B_FALSE;
-	boolean_t		record = B_FALSE;
-	boolean_t		start_play = B_FALSE;
-	boolean_t		start_record = B_FALSE;
-	boolean_t		stop_play = B_FALSE;
-	boolean_t		stop_record = B_FALSE;
-	int			codec_type = ad_infop->ad_codec_type;
-	int			doread;
-	int			dowrite;
-	int			mode = stpptr->am_pstate->apm_mode;
-	int			stream;
-
-	ATRACE("in am_audio_set_info()", chptr);
-	ATRACE_32("am_audio_set_info() mode", mode);
-
-	ASSERT(!mutex_owned(&statep->as_lock));
-	ASSERT(apm_infop);
-
-	/*
-	 * Are we playing and/or recording? For AUDIOCTL channels we
-	 * force play and record, thus checking to see if it is changing
-	 * the format. Since AUDIOCTL channels can't change the format
-	 * we fail if the format isn't the same.
-	 */
-	curi = chptr->ch_info.info;
-	if ((chptr->ch_dir & AUDIO_PLAY) || type == AUDIOCTL) {
-		play = B_TRUE;
-	}
-	if ((chptr->ch_dir & AUDIO_RECORD) || type == AUDIOCTL) {
-		record = B_TRUE;
-	}
-
-	/*
-	 * If hardware supports both play and record then we need to do the
-	 * play vs. record checks.
-	 */
-	doread = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD;
-	dowrite = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY;
-
-	/*
-	 * The AUDIO_MIXERCTL_SETINFO ioctl() creates a pseudo channel that
-	 * has it's ch_info.info set to hw_info. When in mixer mode this is
-	 * the only time this happens. For this ioctl() we set only a small
-	 * number of the h/w specific entries. So we fake out compat mode
-	 * so that the h/w does get set.
-	 */
-	if (mode == AM_MIXER_MODE && chptr->ch_info.info == hw_info) {
-		ATRACE("am_audio_set_info() AUDIO_MIXERCTL_SETINFO ioctl()",
-		    NULL);
-		mode = AM_COMPAT_MODE;
-	}
-
-	/* we use curi to get info to check against */
-	if (mode == AM_COMPAT_MODE) {
-		curi = hw_info;
-#ifdef DEBUG
-	} else {
-		/* this was set above, just be a bit paranoid */
-		ASSERT(mode == AM_MIXER_MODE);
-		ASSERT(curi == chptr->ch_info.info);
-#endif
-	}
-
-	/* first make sure the new data format is legal */
-	if (play && Modify(newi->play.sample_rate) &&
-	    newi->play.sample_rate != curi->play.sample_rate) {
-		if (type != AUDIO || am_ck_sample_rate(&ad_infop->ad_play,
-		    mode, newi->play.sample_rate) == AUDIO_FAILURE) {
-			goto error;
-		}
-		tempi.play.sample_rate = newi->play.sample_rate;
-	} else {
-		tempi.play.sample_rate = curi->play.sample_rate;
-	}
-	if (record && Modify(newi->record.sample_rate) &&
-	    newi->record.sample_rate != curi->record.sample_rate) {
-		if (type != AUDIO || am_ck_sample_rate(&ad_infop->ad_record,
-		    mode, newi->record.sample_rate) == AUDIO_FAILURE) {
-			goto error;
-		}
-		tempi.record.sample_rate = newi->record.sample_rate;
-	} else {
-		tempi.record.sample_rate = curi->record.sample_rate;
-	}
-	mutex_enter(&statep->as_lock);
-	if (doread && dowrite) {
-		if (mode == AM_COMPAT_MODE &&
-		    !(ad_infop->ad_diff_flags & AM_DIFF_SR) &&
-		    tempi.play.sample_rate != tempi.record.sample_rate) {
-			/* if only play or record we can fix this */
-			if (stpptr->am_in_chs && stpptr->am_out_chs == 0) {
-				/* set play to capture sample rate */
-				tempi.play.sample_rate =
-				    tempi.record.sample_rate;
-			} else if (stpptr->am_in_chs == 0 &&
-			    stpptr->am_out_chs) {
-				/* set capture to play sample rate */
-				tempi.record.sample_rate =
-				    tempi.play.sample_rate;
-			} else {
-				mutex_exit(&statep->as_lock);
-				goto error;
-			}
-		} else {
-			/*
-			 * There's a bug in audiotool which after doing an
-			 * AUDIO_SETINFO it updates the state in AudioDevice.cc
-			 * SetState() it uses the record side to get the new
-			 * sample rate! So work around if write only. Who knows,
-			 * perhaps other apps are as stupid!
-			 */
-			if (stpptr->am_out_chs != 0 && stpptr->am_in_chs == 0) {
-				/* set to the same sample rate, gads! */
-				tempi.record.sample_rate =
-				    tempi.play.sample_rate;
-			}
-		}
-	}
-
-	ATRACE_32("am_audio_set_info() PLAY sample rate set",
-	    tempi.play.sample_rate);
-	ATRACE_32("am_audio_set_info() RECORD sample rate set",
-	    tempi.record.sample_rate);
-
-	if (play && Modify(newi->play.channels) &&
-	    newi->play.channels != curi->play.channels) {
-		if (type != AUDIO || am_ck_channels(&ad_infop->ad_play,
-		    newi->play.channels, B_FALSE) == AUDIO_FAILURE) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.play.channels = newi->play.channels;
-	} else {
-		tempi.play.channels = curi->play.channels;
-	}
-	if (record && Modify(newi->record.channels) &&
-	    newi->record.channels != curi->record.channels) {
-		if (type != AUDIO || am_ck_channels(&ad_infop->ad_record,
-		    newi->record.channels, B_FALSE) == AUDIO_FAILURE) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.record.channels = newi->record.channels;
-	} else {
-		tempi.record.channels = curi->record.channels;
-	}
-	if (doread && dowrite) {
-		if (mode == AM_COMPAT_MODE &&
-		    !(ad_infop->ad_diff_flags & AM_DIFF_CH) &&
-		    tempi.play.channels != tempi.record.channels) {
-			/* if only play or record we can fix this */
-			if (stpptr->am_in_chs && stpptr->am_out_chs == 0) {
-				/* set play to capture sample rate */
-				tempi.play.channels = tempi.record.channels;
-			} else if (stpptr->am_in_chs != 0 &&
-			    stpptr->am_out_chs) {
-				/* set capture to play sample rate */
-				tempi.record.channels = tempi.play.channels;
-			} else {
-				mutex_exit(&statep->as_lock);
-				goto error;
-			}
-		} else {
-			/* see audiotool bug description above */
-			if (stpptr->am_out_chs != 0 && stpptr->am_in_chs == 0) {
-				/* set to the same channels, gads! */
-				tempi.record.channels = tempi.play.channels;
-			}
-		}
-	}
-	ATRACE_32("am_audio_set_info() PLAY channels set",
-	    tempi.play.channels);
-	ATRACE_32("am_audio_set_info() RECORD channels set",
-	    tempi.record.channels);
-
-	if (play && Modify(newi->play.precision) &&
-	    newi->play.precision != curi->play.precision) {
-		if (type != AUDIO) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.play.precision = newi->play.precision;
-	} else {
-		tempi.play.precision = curi->play.precision;
-	}
-	if (record && Modify(newi->record.precision) &&
-	    newi->record.precision != curi->record.precision) {
-		if (type != AUDIO) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.record.precision = newi->record.precision;
-	} else {
-		tempi.record.precision = curi->record.precision;
-	}
-	if (doread && dowrite) {
-		if (mode == AM_COMPAT_MODE &&
-		    !(ad_infop->ad_diff_flags & AM_DIFF_PREC) &&
-		    tempi.play.precision != tempi.record.precision) {
-			/* if only play or record we can fix this */
-			if (stpptr->am_in_chs && stpptr->am_out_chs == 0) {
-				/* set play to capture sample rate */
-				tempi.play.precision = tempi.record.precision;
-			} else if (stpptr->am_in_chs == 0 &&
-			    stpptr->am_out_chs) {
-				/* set capture to play sample rate */
-				tempi.record.precision = tempi.play.precision;
-			} else {
-				mutex_exit(&statep->as_lock);
-				goto error;
-			}
-		} else {
-			/* see audiotool bug description above */
-			if (stpptr->am_out_chs != 0 && stpptr->am_in_chs == 0) {
-				/* set to the same precision, gads! */
-				tempi.record.precision = tempi.play.precision;
-			}
-		}
-	}
-	ATRACE_32("am_audio_set_info() PLAY precision set",
-	    tempi.play.precision);
-	ATRACE_32("am_audio_set_info() RECORD precision set",
-	    tempi.record.precision);
-
-	if (play && Modify(newi->play.encoding) &&
-	    newi->play.encoding != curi->play.encoding) {
-		if (type != AUDIO) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.play.encoding = newi->play.encoding;
-	} else {
-		tempi.play.encoding = curi->play.encoding;
-	}
-	if (record && Modify(newi->record.encoding) &&
-	    newi->record.encoding != curi->record.encoding) {
-		if (type != AUDIO) {
-			mutex_exit(&statep->as_lock);
-			goto error;
-		}
-		tempi.record.encoding = newi->record.encoding;
-	} else {
-		tempi.record.encoding = curi->record.encoding;
-	}
-	if (doread && dowrite) {
-		if (mode == AM_COMPAT_MODE &&
-		    !(ad_infop->ad_diff_flags & AM_DIFF_ENC) &&
-		    tempi.play.encoding != tempi.record.encoding) {
-			/* if only play or record we can fix this */
-			if (stpptr->am_in_chs && stpptr->am_out_chs == 0) {
-				/* set play to capture sample rate */
-				tempi.play.encoding = tempi.record.encoding;
-			} else if (stpptr->am_in_chs == 0 &&
-			    stpptr->am_out_chs) {
-				/* set capture to play sample rate */
-				tempi.record.encoding = tempi.play.encoding;
-			} else {
-				mutex_exit(&statep->as_lock);
-				goto error;
-			}
-		} else {
-			/* see audiotool bug description above */
-			if (stpptr->am_out_chs != 0 && stpptr->am_in_chs == 0) {
-				/* set to the same encoding, gads! */
-				tempi.record.encoding = tempi.play.encoding;
-			}
-		}
-	}
-	mutex_exit(&statep->as_lock);
-
-	ATRACE_32("am_audio_set_info() PLAY encoding set",
-	    tempi.play.encoding);
-	ATRACE_32("am_audio_set_info() RECORD encoding set",
-	    tempi.record.encoding);
-
-	/*
-	 * In COMPAT mode or with multi-channel Codecs we check against
-	 * what the hardware allows. Otherwise, we check against what the
-	 * mixer can deal with. But only if an AUDIO channel.
-	 */
-	if (type == AUDIO) {
-		if (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC) {
-			if (dowrite && am_ck_combinations(
-			    ad_infop->ad_play_comb, tempi.play.encoding,
-			    tempi.play.precision, B_FALSE) == AUDIO_FAILURE) {
-				goto error;
-			}
-			if (doread && am_ck_combinations(ad_infop->ad_rec_comb,
-			    tempi.record.encoding, tempi.record.precision,
-			    B_FALSE) == AUDIO_FAILURE) {
-				goto error;
-			}
-
-		} else {	/* AM_MIXER_MODE */
-			/* make sure the mixer can deal with the combinations */
-			ASSERT(mode == AM_MIXER_MODE);
-
-			switch ((int)tempi.play.channels) {
-			case -1:		/* no change to channel */
-			case AUDIO_CHANNELS_MONO:
-			case AUDIO_CHANNELS_STEREO:
-				break;
-			default:
-				goto error;
-			}
-			switch ((int)tempi.record.channels) {
-			case -1:		/* no change to channel */
-			case AUDIO_CHANNELS_MONO:
-			case AUDIO_CHANNELS_STEREO:
-				break;
-			default:
-				goto error;
-			}
-
-			switch ((int)tempi.play.encoding) {
-			case -1:		/* no change to encoding */
-			    break;
-			case AUDIO_ENCODING_LINEAR:	/* signed */
-			    /* we support 8 & 16-bit linear */
-			    if (tempi.play.precision != AUDIO_PRECISION_16 &&
-				tempi.play.precision != AUDIO_PRECISION_8) {
-				    goto error;
-			    }
-			    break;
-			case AUDIO_ENCODING_LINEAR8:	/* unsigned */
-			case AUDIO_ENCODING_ULAW:
-			case AUDIO_ENCODING_ALAW:
-			    /* support 8-bit unsigned linear, u-law & A-law */
-			    if (tempi.play.precision != AUDIO_PRECISION_8) {
-				    goto error;
-			    }
-			    break;
-			default:
-			    goto error;
-			}
-			switch ((int)tempi.record.encoding) {
-			case -1:		/* no change to encoding */
-			    break;
-			case AUDIO_ENCODING_LINEAR:	/* signed */
-			    /* we support 8 & 16-bit linear */
-			    if (tempi.record.precision != AUDIO_PRECISION_16 &&
-				tempi.record.precision != AUDIO_PRECISION_8) {
-				    goto error;
-			    }
-			    break;
-			case AUDIO_ENCODING_LINEAR8:	/* unsigned */
-			case AUDIO_ENCODING_ULAW:
-			case AUDIO_ENCODING_ALAW:
-			    /* support 8-bit unsigned linear, u-law & A-law */
-			    if (tempi.record.precision != AUDIO_PRECISION_8) {
-				    goto error;
-			    }
-			    break;
-			default:
-			    goto error;
-			}
-		}
-	}
-	ATRACE("am_audio_set_info() precision/encoding checked OK", &tempi);
-
-	if (Modify(newi->play.gain)) {
-		if (newi->play.gain > AUDIO_MAX_GAIN) {
-			goto error;
-		}
-		tempi.play.gain = newi->play.gain;
-	} else {
-		tempi.play.gain = curi->play.gain;
-	}
-	if (Modify(newi->record.gain)) {
-		if (newi->record.gain > AUDIO_MAX_GAIN) {
-			goto error;
-		}
-		tempi.record.gain = newi->record.gain;
-	} else {
-		tempi.record.gain = curi->record.gain;
-	}
-	ATRACE_32("am_audio_set_info() PLAY gain set", tempi.play.gain);
-	ATRACE_32("am_audio_set_info() RECORD gain set", tempi.record.gain);
-
-	if (Modify(newi->play.port)) {
-		tempi.play.port = newi->play.port;
-	} else {
-		tempi.play.port = hw_info->play.port;
-	}
-	if (tempi.play.port & ~hw_info->play.avail_ports) { /* legal port? */
-		goto error;
-	}
-	/* always turn on un-modifiable ports */
-	tempi.play.port |= hw_info->play.avail_ports & ~hw_info->play.mod_ports;
-	if (ad_infop->ad_misc_flags & AM_MISC_PP_EXCL) { /* check exclusivity */
-		if (am_ck_bits_set32(tempi.play.port) > 1) {
-			goto error;
-		}
-	}
-	if (Modify(newi->record.port)) {
-		tempi.record.port = newi->record.port;
-	} else {
-		tempi.record.port = hw_info->record.port;
-	}
-	if (tempi.record.port & ~hw_info->record.avail_ports) {	/* legal ? */
-		goto error;
-	}
-	/* always turn on un-modifiable ports */
-	tempi.record.port |=
-	    hw_info->record.avail_ports & ~hw_info->record.mod_ports;
-	/* check exclusivity */
-	if (ad_infop->ad_misc_flags & AM_MISC_RP_EXCL) {
-		if (am_ck_bits_set32(tempi.record.port) > 1) {
-			goto error;
-		}
-	}
-	ATRACE_32("am_audio_set_info() PLAY ports set", tempi.play.port);
-	ATRACE_32("am_audio_set_info() RECORD ports set", tempi.record.port);
-
-	if (Modifyc(newi->play.balance)) {
-		if (newi->play.balance > AUDIO_RIGHT_BALANCE) {
-			goto error;
-		}
-		tempi.play.balance = newi->play.balance;
-	} else {
-		tempi.play.balance = curi->play.balance;
-	}
-	if (Modifyc(newi->record.balance)) {
-		if (newi->record.balance > AUDIO_RIGHT_BALANCE) {
-			goto error;
-		}
-		tempi.record.balance = newi->record.balance;
-	} else {
-		tempi.record.balance = curi->record.balance;
-	}
-
-	ATRACE_32("am_audio_set_info() PLAY balance set", tempi.play.balance);
-	ATRACE_32("am_audio_set_info() REC balance set", tempi.record.balance);
-
-	if (Modify(newi->monitor_gain)) {
-		if (ad_infop->ad_defaults->hw_features &
-		    AUDIO_HWFEATURE_IN2OUT) {
-			if (newi->monitor_gain > AUDIO_MAX_GAIN) {
-				goto error;
-			}
-			tempi.monitor_gain = newi->monitor_gain;
-		} else {
-			if (newi->monitor_gain != hw_info->monitor_gain) {
-				ATRACE("am_audio_set_info() "
-				    "monitor gain cannot be set", 0);
-				goto error;
-			}
-			tempi.monitor_gain = hw_info->monitor_gain;
-		}
-	} else {
-		tempi.monitor_gain = hw_info->monitor_gain;
-	}
-	ATRACE_32("am_audio_set_info() monitor gain set", tempi.monitor_gain);
-
-	if (dowrite && Modifyc(newi->output_muted)) {
-		tempi.output_muted = newi->output_muted;
-	} else {
-		tempi.output_muted = curi->output_muted;
-	}
-	ATRACE_32("am_audio_set_info() output muted set", tempi.output_muted);
-
-	/*
-	 * Now that we've got the new values verified we need to update the
-	 * hardware. The following is updated:
-	 *   COMPAT Mode, All Devices
-	 *	play.minordev (H/W)		record.minordev (H/W)
-	 *   COMPAT Mode, AUDIO Device
-	 *	play.sample_rate (H/W)		record.sample_rate (H/W)
-	 *	play.channels (H/W)		record.channels (H/W)
-	 *	play.precision (H/W)		record.precision (H/W)
-	 *	play.encoding (H/W)		record.encoding (H/W)
-	 *	play.gain (H/W)			record.gain (H/W)
-	 *	play.balance (H/W)		record.balance (H/W)
-	 *	output_muted (H/W)
-	 *   COMPAT Mode, AUDIOCTL Device
-	 *	play.gain (H/W)			record.gain (H/W)
-	 *	play.balance (H/W)		record.balance (H/W)
-	 *	output_muted (H/W)
-	 *   MIXER Mode, All Devices
-	 *	play.minordev (CH)		record.minordev (CH)
-	 *   MIXER Mode, AUDIO Device, Traditional Codec
-	 *	play.sample_rate (CH)		record.sample_rate (CH)
-	 *	play.channels (CH)		record.channels (CH)
-	 *	play.precision (CH)		record.precision (CH)
-	 *	play.encoding (CH)		record.encoding (CH)
-	 *	play.gain (CH)			record.gain (CH)
-	 *	play.balance (CH)		record.balance (CH)
-	 *	output_muted (CH)
-	 *   MIXER Mode, AUDIOCTL Device, Traditional Codec, Same Process As
-	 *   An AUDIO Channel, ch_ctl == TRUE
-	 *	play.gain (CH)			record.gain (CH)
-	 *	play.balance (CH)		record.balance (CH)
-	 *	output_muted (CH)
-	 *   MIXER Mode, AUDIOCTL Device, Traditional Codec, Different Proc.
-	 *   From An AUDIO Channel, ch_ctl == FALSE
-	 *	play.gain (H/W)			record.gain (H/W)
-	 *	play.balance (H/W)		record.balance (H/W)
-	 *	output_muted (H/W)
-	 *   MIXER Mode, AUDIO Device, Multi-Channel Codec
-	 *	play.sample_rate (CH H/W)	record.sample_rate (CH H/W)
-	 *	play.channels (CH H/W)		record.channels (CH H/W)
-	 *	play.precision (CH H/W)		record.precision (CH H/W)
-	 *	play.encoding (CH H/W)		record.encoding (CH H/W)
-	 *	play.gain (CH H/W)		record.gain (CH H/W)
-	 *	play.balance (CH H/W)		record.balance (CH H/W)
-	 *	output_muted (CH H/W)
-	 *   MIXER Mode, AUDIOCTL Device, Multi-Channel Codec, Same Proc. As
-	 *   An AUDIO Channel, ch_ctl == TRUE
-	 *	play.gain (CH H/W)		record.gain (CH H/W)
-	 *	play.balance (CH H/W)		record.balance (CH H/W)
-	 *	output_muted (CH H/W)
-	 *   MIXER Mode, AUDIOCTL Device, Multi-Channel Codec, Different
-	 *   Process From An AUDIO, ch_ctl == FALSE
-	 *	play.gain (H/W)			record.gain (H/W)
-	 *	play.balance (H/W)		record.balance (H/W)
-	 *	output_muted (H/W)
-	 *   All May Modify These Fields
-	 *	play.port (H/W)			record.port (H/W)
-	 *	monitor_gain (H/W)
-	 *
-	 * If we are in AM_COMPAT_MODE then output_muted controls the hardware,
-	 * otherwise it just affects the channel, if it is a ch_ctl.
-	 */
-
-	/* only AUDIO channels can affect the data format */
-	if (type == AUDIO) {
-		/* figure out our "stream number" */
-		if (codec_type == AM_MS_CODEC) {
-			stream = chptr->ch_info.ch_number;
-		} else {
-			stream = AM_SET_CONFIG_BOARD;
-		}
-
-		if (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC) {
-			/*
-			 * We only set the format if there's been a change.
-			 * Otherwise we risk introducing noise, pops, etc.,
-			 * for little good reason.
-			 */
-			if (dowrite && (hw_info->play.sample_rate !=
-				tempi.play.sample_rate ||
-			    hw_info->play.channels !=
-				tempi.play.channels ||
-			    hw_info->play.precision !=
-				tempi.play.precision ||
-			    hw_info->play.encoding !=
-				tempi.play.encoding)) {
-				if (am_set_format(statep, stpptr, ad_infop,
-				    stream, AUDIO_PLAY,
-				    tempi.play.sample_rate,
-				    stpptr->am_hw_pchs, stpptr->am_hw_pprec,
-				    stpptr->am_hw_penc, AM_NO_FORCE,
-				    AM_SERIALIZE) == AUDIO_FAILURE) {
-					goto error;
-				}
-			}
-			if (doread && (hw_info->record.sample_rate !=
-				tempi.record.sample_rate ||
-			    hw_info->record.channels !=
-				tempi.record.channels ||
-			    hw_info->record.precision !=
-				tempi.record.precision ||
-			    hw_info->record.encoding !=
-				tempi.record.encoding)) {
-				if (am_set_format(statep, stpptr, ad_infop,
-				    stream, AUDIO_RECORD,
-				    tempi.record.sample_rate,
-				    stpptr->am_hw_rchs, stpptr->am_hw_rprec,
-				    stpptr->am_hw_renc, AM_NO_FORCE,
-				    AM_SERIALIZE) == AUDIO_FAILURE) {
-					goto error;
-				}
-			}
-		}
-		/* lock state while updating so ISR calls will be okay */
-		mutex_enter(&chptr->ch_lock);
-		if (mode == AM_MIXER_MODE) {
-			curi->play.sample_rate = tempi.play.sample_rate;
-			curi->play.channels = tempi.play.channels;
-			curi->play.precision = tempi.play.precision;
-			curi->play.encoding = tempi.play.encoding;
-			curi->record.sample_rate = tempi.record.sample_rate;
-			curi->record.channels = tempi.record.channels;
-			curi->record.precision = tempi.record.precision;
-			curi->record.encoding = tempi.record.encoding;
-		} else {
-			hw_info->play.sample_rate = tempi.play.sample_rate;
-			hw_info->play.channels = tempi.play.channels;
-			hw_info->play.precision = tempi.play.precision;
-			hw_info->play.encoding = tempi.play.encoding;
-			hw_info->record.sample_rate = tempi.record.sample_rate;
-			hw_info->record.channels = tempi.record.channels;
-			hw_info->record.precision = tempi.record.precision;
-			hw_info->record.encoding = tempi.record.encoding;
-		}
-
-		/* see if we need to update the sample rate conv. routines */
-		if (mode == AM_MIXER_MODE && codec_type == AM_TRAD_CODEC) {
-			audio_apm_info_t *apm_infop = chptr->ch_apm_infop;
-
-			if (chpptr->acp_writing) {
-				ATRACE("am_audio_set_info() PLAY, "
-				    "calling src update", chpptr);
-				if (ad_infop->ad_play.ad_conv->ad_src_update(
-				    AM_SRC_CHPTR2HDL(chptr),
-				    &((audio_info_t *)chptr->ch_info.info)->
-				    play,
-				    &((audio_info_t *)apm_infop->apm_ad_state)->
-				    play,
-				    ((am_ad_info_t *)apm_infop->apm_ad_infop)->
-				    ad_play.ad_sr_info,
-				    AUDIO_PLAY) == AUDIO_FAILURE) {
-					ATRACE("am_audio_set_info() "
-					    "play src_update() failed", 0);
-					mutex_exit(&chptr->ch_lock);
-					goto error;
-				}
-			}
-			if (chpptr->acp_reading) {
-				ATRACE("am_audio_set_info() RECORD, "
-				    "calling src update", chpptr);
-				if (ad_infop->ad_record.ad_conv->ad_src_update(
-				    AM_SRC_CHPTR2HDL(chptr),
-				    &((audio_info_t *)chptr->ch_info.info)->
-				    record,
-				    &((audio_info_t *)apm_infop->apm_ad_state)->
-				    record,
-				    ((am_ad_info_t *)apm_infop->apm_ad_infop)->
-				    ad_record.ad_sr_info,
-				    AUDIO_RECORD) == AUDIO_FAILURE) {
-					ATRACE("am_audio_set_info() "
-					    "record src_update() failed", 0);
-					mutex_exit(&chptr->ch_lock);
-					goto error;
-				}
-			}
-		}
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	/* is this an AUDIOCTL ch with the PID as another AUDIO ch? */
-	mutex_enter(&chptr->ch_lock);
-	ch_ctl = (chpptr->acp_flags & AM_CHNL_CONTROL) ? B_TRUE : B_FALSE;
-	mutex_exit(&chptr->ch_lock);
-
-	/* re-figure out our "stream number" */
-	if (mode == AM_COMPAT_MODE || (mode == AM_MIXER_MODE && !ch_ctl)) {
-		stream = AM_SET_CONFIG_BOARD;
-	} else {
-		stream = chptr->ch_info.ch_number;
-	}
-
-	/*
-	 * AUDIO and AUDIOCTL can affect gains, ports, etc. If in COMPAT
-	 * mode or a MS Codec we affect hardware. Otherwise this is a
-	 * virtual ch. and only that channel's parameters are affected,
-	 * i.e., no hardware update. Also, if a MS Codec and an AUDIOCTL
-	 * channel isn't associated with a particular stream then we don't
-	 * muck with hardware either.
-	 */
-	if (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC ||
-	    (type == AUDIOCTL && !ch_ctl)) {
-		if (dowrite && am_set_gain(statep, apm_infop,
-		    stpptr->am_hw_pchs, (tempi.play.gain & 0x0ff),
-		    tempi.play.balance, AUDIO_PLAY, stream, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			goto error;
-		}
-		if (doread && am_set_gain(statep, apm_infop,
-		    stpptr->am_hw_rchs, (tempi.record.gain & 0x0ff),
-		    tempi.record.balance, AUDIO_RECORD, stream, AM_NO_FORCE,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			goto error;
-		}
-		/* only if output_muted actually changed */
-		if (hw_info->output_muted != tempi.output_muted) {
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_OUTPUT_MUTE, AUDIO_PLAY, tempi.output_muted,
-			    NULL, AM_SERIALIZE) == AUDIO_FAILURE) {
-				goto error;
-			}
-		}
-	}
-	if (mode == AM_MIXER_MODE) {
-		curi->play.gain = tempi.play.gain;
-		curi->play.balance = tempi.play.balance;
-		curi->record.gain = tempi.record.gain;
-		curi->record.balance = tempi.record.balance;
-		curi->output_muted = tempi.output_muted;
-		tempi.play.minordev = curi->play.minordev;
-		tempi.record.minordev = curi->record.minordev;
-	} else {
-		hw_info->play.gain = tempi.play.gain;
-		hw_info->play.balance = tempi.play.balance;
-		hw_info->record.gain = tempi.record.gain;
-		hw_info->record.balance = tempi.record.balance;
-		hw_info->output_muted = tempi.output_muted;
-		tempi.play.minordev = hw_info->play.minordev;
-		tempi.record.minordev = hw_info->record.minordev;
-	}
-
-	/* now we can set the ports and monitor gain, since all can set them */
-	if (tempi.play.port != hw_info->play.port) {
-		/* only if the play port actually changed */
-		if (hw_info->play.port != tempi.play.port) {
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_PORT, AUDIO_PLAY, tempi.play.port, NULL,
-			    AM_SERIALIZE) == AUDIO_FAILURE) {
-				goto error;
-			}
-		}
-		hw_info->play.port = tempi.play.port;
-	}
-	if (tempi.record.port != hw_info->record.port) {
-		/* only if the record port actually changed */
-		if (hw_info->record.port != tempi.record.port) {
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_PORT, AUDIO_RECORD, tempi.record.port,
-			    NULL, AM_SERIALIZE) == AUDIO_FAILURE) {
-				goto error;
-			}
-		}
-		hw_info->record.port = tempi.record.port;
-	}
-	if (tempi.monitor_gain != hw_info->monitor_gain) {
-		/* only if the monitor gain actually changed */
-		if (hw_info->monitor_gain != tempi.monitor_gain) {
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_MONITOR_GAIN, AUDIO_BOTH, tempi.monitor_gain,
-			    NULL, AM_SERIALIZE) == AUDIO_FAILURE) {
-				goto error;
-			}
-		}
-		hw_info->monitor_gain = tempi.monitor_gain;
-	}
-
-	/* we need to update the virtual channel, if we have one */
-	if (mode == AM_MIXER_MODE) {
-		curi->play.port = tempi.play.port;
-		curi->record.port = tempi.record.port;
-		curi->monitor_gain = tempi.monitor_gain;
-	}
-
-	/* now fix virtual channel parameters */
-	if (Modify(newi->play.buffer_size)) {
-		tempi.play.buffer_size = newi->play.buffer_size;
-	} else {
-		tempi.play.buffer_size = curi->play.buffer_size;
-	}
-	if (Modify(newi->record.buffer_size)) {
-		tempi.record.buffer_size = newi->record.buffer_size;
-	} else {
-		tempi.record.buffer_size = curi->record.buffer_size;
-	}
-	ATRACE("am_audio_set_info() buffer size set", &tempi);
-
-	if (Modify(newi->play.samples)) {
-		tempi.play.samples = newi->play.samples;
-		new_play_samples = B_TRUE;
-	} else {
-		tempi.play.samples = curi->play.samples;
-	}
-	if (Modify(newi->record.samples)) {
-		tempi.record.samples = newi->record.samples;
-		new_record_samples = B_TRUE;
-	} else {
-		tempi.record.samples = curi->record.samples;
-	}
-	ATRACE("am_audio_set_info() samples updated", &tempi);
-
-	if (Modify(newi->play.eof)) {
-		tempi.play.eof = newi->play.eof;
-	} else {
-		tempi.play.eof = curi->play.eof;
-	}
-	ATRACE("am_audio_set_info() eof updated", &tempi);
-
-	if (Modifyc(newi->play.pause)) {
-		tempi.play.pause = newi->play.pause;
-	} else {
-		tempi.play.pause = curi->play.pause;
-	}
-	if (Modifyc(newi->record.pause)) {
-		tempi.record.pause = newi->record.pause;
-	} else {
-		tempi.record.pause = curi->record.pause;
-	}
-	/* if we unpaused we need to make sure we start up again */
-	if (!tempi.play.pause && curi->play.pause) {
-		start_play = B_TRUE;
-	} else if (tempi.play.pause && !curi->play.pause &&
-	    (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC)) {
-		stop_play = B_TRUE;
-	}
-	if (!tempi.record.pause && curi->record.pause) {
-		start_record = B_TRUE;
-	} else if (tempi.record.pause && !curi->record.pause &&
-	    (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC)) {
-		stop_record = B_TRUE;
-	}
-	ATRACE("am_audio_set_info() pause set", &tempi);
-
-	if (Modifyc(newi->play.error)) {
-		tempi.play.error = newi->play.error;
-	} else {
-		tempi.play.error = curi->play.error;
-	}
-	if (Modifyc(newi->record.error)) {
-		tempi.record.error = newi->record.error;
-	} else {
-		tempi.record.error = curi->record.error;
-	}
-	ATRACE("am_audio_set_info() error updated", &tempi);
-
-	if (Modifyc(newi->play.waiting)) {
-		tempi.play.waiting = newi->play.waiting;
-	} else {
-		tempi.play.waiting = curi->play.waiting;
-	}
-	if (Modifyc(newi->record.waiting)) {
-		tempi.record.waiting = newi->record.waiting;
-	} else {
-		tempi.record.waiting = curi->record.waiting;
-	}
-	ATRACE("am_audio_set_info() waiting updated", &tempi);
-
-	/*
-	 * For MIXER mode we must update virtual channel parameters, because
-	 * as soon as we restart the DMA engine(s) it's going to ask for audio.
-	 * If the pause is still in effect then no data is going to be
-	 * transferred.
-	 */
-	if (mode == AM_MIXER_MODE) {
-		mutex_enter(&chptr->ch_lock);
-		curi->play.buffer_size = tempi.play.buffer_size;
-		curi->play.pause = tempi.play.pause;
-		curi->play.eof = tempi.play.eof;
-		curi->play.error = tempi.play.error;
-		curi->play.waiting = tempi.play.waiting;
-		curi->record.buffer_size = tempi.record.buffer_size;
-		curi->record.pause = tempi.record.pause;
-		curi->record.error = tempi.record.error;
-		curi->record.waiting = tempi.record.waiting;
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	/* before we leave, we need to restart the DMA engines, or ... */
-	if (start_play == B_TRUE) {
-		/* make sure the play DMA engine is running */
-		ASSERT(stop_play == B_FALSE);
-		ATRACE("am_audio_set_info() start play", chptr);
-		curi->play.pause = 0;	/* must be before the call */
-		curi->play.active = 1;	/* set before start for mode switch */
-		hw_info->play.active = 1;
-		if (am_ad_start_play(statep, stpptr, ad_infop, stream,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			/* we don't change pause flag if failed to start */
-			curi->play.active = 0;
-			hw_info->play.active = 0;
-			/*
-			 * Since we are serialized we don't worry about the
-			 * mode switch CV like am_wsvc() has to.
-			 */
-		} else {
-			am_send_signal(statep, stpptr);
-		}
-	} else if (stop_play == B_TRUE) {
-		/* make sure the play DMA engine is paused */
-		ATRACE("am_audio_set_info() pause play", chptr);
-		am_ad_pause_play(statep, stpptr, ad_infop, stream);
-		curi->play.active = 0;
-		hw_info->play.active = 0;
-		curi->play.pause = 1;
-		am_send_signal(statep, stpptr);
-	}
-	if (start_record == B_TRUE) {
-		/* make sure the record DMA engine is running */
-		ASSERT(stop_record == B_FALSE);
-		ATRACE("am_audio_set_info() start record", chptr);
-		curi->record.pause = 0;	/* must be before the call */
-		curi->record.active = 1;  /* set before start for mode switch */
-		hw_info->record.active = 1;
-		if (am_ad_start_record(statep, stpptr, ad_infop, stream,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			curi->record.active = 0;
-			hw_info->record.active = 0;
-			/* we don't change pause flag if failed to start */
-		} else {
-			am_send_signal(statep, stpptr);
-		}
-	} else if (stop_record == B_TRUE) {
-		/* make sure the record DMA engine is stopped */
-		ATRACE("am_audio_set_info() stop record", chptr);
-		am_ad_stop_record(statep, stpptr, ad_infop, stream);
-		curi->record.pause = 1;
-		curi->record.active = 0;
-		hw_info->record.active = 0;
-		am_send_signal(statep, stpptr);
-	}
-
-	/*
-	 * For COMPAT mode we are dealing with the hardware, not a virtual
-	 * channel. So the true state of the hardware can't be modified before
-	 * starting or stopping the DMA engine(s).
-	 */
-	if (mode == AM_COMPAT_MODE) {
-		mutex_enter(&chptr->ch_lock);
-		hw_info->play.buffer_size = tempi.play.buffer_size;
-		hw_info->play.pause = tempi.play.pause;
-		hw_info->play.eof = tempi.play.eof;
-		hw_info->play.error = tempi.play.error;
-		hw_info->play.waiting = tempi.play.waiting;
-		hw_info->record.buffer_size = tempi.record.buffer_size;
-		hw_info->record.pause = tempi.record.pause;
-		hw_info->record.error = tempi.record.error;
-		hw_info->record.waiting = tempi.record.waiting;
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	/* everything passed so we can update the samples count */
-	if (new_play_samples) {
-		curi->play.samples = tempi.play.samples;
-		mutex_enter(&chptr->ch_lock);
-		chpptr->acp_psamples_c = 0;
-		chpptr->acp_psamples_f = 0;
-		chpptr->acp_psamples_p = 0;
-		mutex_exit(&chptr->ch_lock);
-	}
-	if (new_record_samples) {
-		curi->record.samples = tempi.record.samples;
-	}
-
-	/*
-	 * If we don't have a reti pointer we ignore the R/O members. If we
-	 * need them we get them directly from the channel that is active.
-	 * So if reti == NULL we are done. Otherwise copy tempi into the
-	 * memory pointed to by reti and then copy over the R/O members.
-	 *
-	 * We pass the reserved members, just in case.
-	 *	play._xxx[1]			record._xxx[1]
-	 *	_xxx[1]
-	 *	_xxx[2]
-	 */
-	if (reti != NULL) {
-		ATRACE("am_audio_set_info() reti succeeded", chptr);
-
-		/* do a quick copy, and then fill in the special fields */
-		bcopy(curi, reti, sizeof (*curi));
-
-		mutex_enter(&chptr->ch_lock);
-		reti->play.avail_ports =	hw_info->play.avail_ports;
-		reti->record.avail_ports =	hw_info->record.avail_ports;
-		reti->play.mod_ports =		hw_info->play.mod_ports;
-		reti->record.mod_ports =	hw_info->record.mod_ports;
-		reti->record.eof =		0;
-		reti->monitor_gain =		hw_info->monitor_gain;
-		reti->output_muted =		hw_info->output_muted;
-		reti->hw_features =		hw_info->hw_features;
-		reti->sw_features =		hw_info->sw_features;
-		reti->sw_features_enabled =	hw_info->sw_features_enabled;
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	ATRACE("am_audio_set_info() succeeded", chptr);
-
-	return (AUDIO_SUCCESS);
-
-error:
-	ATRACE("am_audio_set_info() failed", chptr);
-
-	return (AUDIO_FAILURE);
-
-}	/* am_audio_set_info() */
-
-/*
- * am_set_format()
- *
- * Description:
- *	Set the hardware to the desired format. If the format is not
- *	supported we set it to the next best thing. Then the audio is
- *	translated to the desired format during playback or record.
- *
- *	NOTE: All setting of the hardware format MUST be done via this
- *		routine. Thus this is the only place the am_hw_* members
- *		are updated.
- *
- *	NOTE: We don't worry about checking the sample rate. This routine
- *		won't be called with a sample rate that isn't legal according
- *		to the audio driver's configuration tables.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instance's state
- *	am_apm_private_t	*stpptr		Ptr to APM private data
- *	am_ad_info_t		*ad_infop	Ptr to the AD's config info
- *	int			stream		Audio stream
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD
- *	int			sample_rate	Sample rate to set
- *	int			channels	The number of channels to set
- *	int			precision	The sample precision
- *	int			encoding	The encoding method
- *	int			force		Force the format to be set
- *	int			serialize	Serialize calls into driver
- *
- * Returns:
- *	AUDIO_SUCCESS		The format was successfully set
- *	AUDIO_FAILURE		The format wasn't set
- */
-int
-am_set_format(audio_state_t *statep, am_apm_private_t *stpptr,
-	am_ad_info_t *ad_infop, int stream, int dir, int sample_rate,
-	int channels, int precision, int encoding, int force, int serialize)
-{
-	uint_t		hw_sr;
-	uint_t		hw_chs;
-	uint_t		hw_enc;
-	uint_t		hw_prec;
-	uint_t		new_chs;
-	uint_t		new_enc;
-	uint_t		new_prec;
-	int		flags;
-
-	ATRACE("in am_set_format()", statep);
-	ATRACE_32("am_set_format() passed sample rate", sample_rate);
-
-	if (dir == AUDIO_PLAY) {
-		flags = stpptr->am_pflags;
-		hw_sr = stpptr->am_hw_info.play.sample_rate;
-		hw_chs = stpptr->am_hw_pchs;
-		hw_enc = stpptr->am_hw_penc;
-		hw_prec = stpptr->am_hw_pprec;
-	} else if (dir == AUDIO_RECORD) {
-		flags = stpptr->am_rflags;
-		hw_sr = stpptr->am_hw_info.record.sample_rate;
-		hw_chs = stpptr->am_hw_rchs;
-		hw_enc = stpptr->am_hw_renc;
-		hw_prec = stpptr->am_hw_rprec;
-	} else {
-		audio_sup_log(AUDIO_STATE2HDL(statep),
-		    CE_NOTE, "set_format() bad direction: %d", dir);
-		return (AUDIO_FAILURE);
-	}
-
-	/* start with channels */
-	switch (channels) {
-	case AUDIO_CHANNELS_MONO:
-		/* try to set the same 1st */
-		if (flags & AM_PRIV_CH_MONO) {
-			new_chs = AUDIO_CHANNELS_MONO;
-		} else {
-			/* we have to use stereo, so will need to translate */
-			ASSERT(flags & AM_PRIV_CH_STEREO);
-			new_chs = AUDIO_CHANNELS_STEREO;
-		}
-		break;
-	case AUDIO_CHANNELS_STEREO:
-		/* try to set the same 1st */
-		if (flags & AM_PRIV_CH_STEREO) {
-			new_chs = AUDIO_CHANNELS_STEREO;
-		} else {
-			/* we have to use mono, so will need to translate */
-			ASSERT(flags & AM_PRIV_CH_MONO);
-			new_chs = AUDIO_CHANNELS_MONO;
-		}
-		break;
-	default:
-		audio_sup_log(AUDIO_STATE2HDL(statep),
-		    CE_NOTE, "set_format() bad channels: %d",
-		    channels);
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_set_format() passed channels", channels);
-	ATRACE_32("am_set_format() derived channels", new_chs);
-
-	/* check the precision */
-	if (precision == AUDIO_PRECISION_16) {
-		/* see if the hardware supports what we want */
-		if (flags & AM_PRIV_16_PCM) {
-			new_prec = AUDIO_PRECISION_16;
-			new_enc = AUDIO_ENCODING_LINEAR;
-		} else {
-			/* the h/w doesn't, so pick an alternative */
-			new_prec = AUDIO_PRECISION_8;
-			if (flags & AM_PRIV_8_ULAW) {
-				new_enc = AUDIO_ENCODING_ULAW;
-			} else if (flags & AM_PRIV_8_ALAW) {
-				new_enc = AUDIO_ENCODING_ALAW;
-			} else {
-				ASSERT(flags & AM_PRIV_8_PCM);
-				new_enc = AUDIO_ENCODING_LINEAR;
-			}
-		}
-	} else {
-		ASSERT(precision == AUDIO_PRECISION_8);
-		/* just like above go through the list to see what we can use */
-		new_prec = AUDIO_PRECISION_8;
-		if (encoding == AUDIO_ENCODING_LINEAR) {
-			if (flags & AM_PRIV_8_PCM) {
-				new_enc = AUDIO_ENCODING_LINEAR;
-			} else {
-				if (flags & AM_PRIV_8_ULAW) {
-					new_enc = AUDIO_ENCODING_ULAW;
-				} else if (flags & AM_PRIV_8_ALAW) {
-					new_enc = AUDIO_ENCODING_ALAW;
-				} else {
-					ASSERT(flags & AM_PRIV_16_PCM);
-					new_prec = AUDIO_PRECISION_16;
-					new_enc = AUDIO_ENCODING_LINEAR;
-				}
-			}
-		} else if (encoding == AUDIO_ENCODING_ULAW) {
-			if (flags & AM_PRIV_8_ULAW) {
-				new_enc = AUDIO_ENCODING_ULAW;
-			} else {
-				if (flags & AM_PRIV_8_ALAW) {
-					new_enc = AUDIO_ENCODING_ALAW;
-				} else if (flags & AM_PRIV_8_PCM) {
-					new_enc = AUDIO_ENCODING_LINEAR;
-				} else {
-					ASSERT(flags & AM_PRIV_16_PCM);
-					new_prec = AUDIO_PRECISION_16;
-					new_enc = AUDIO_ENCODING_LINEAR;
-				}
-			}
-		} else {
-			ASSERT(encoding == AUDIO_ENCODING_ALAW);
-			if (flags & AM_PRIV_8_ALAW) {
-				new_enc = AUDIO_ENCODING_ALAW;
-			} else {
-				if (flags & AM_PRIV_8_ULAW) {
-					new_enc = AUDIO_ENCODING_ULAW;
-				} else if (flags & AM_PRIV_8_PCM) {
-					new_enc = AUDIO_ENCODING_LINEAR;
-				} else {
-					ASSERT(flags & AM_PRIV_16_PCM);
-					new_prec = AUDIO_PRECISION_16;
-					new_enc = AUDIO_ENCODING_LINEAR;
-				}
-			}
-		}
-	}
-	ATRACE_32("am_set_format() passed precision", precision);
-	ATRACE_32("am_set_format() derived precision", new_prec);
-	ATRACE_32("am_set_format() passed encoding", encoding);
-	ATRACE_32("am_set_format() derived encoding", new_enc);
-
-	/*
-	 * We now have the best possible h/w configuration. We see if
-	 * it matches what we already have. If so then there's nothing
-	 * to do. Otherwise the driver is called to set the hardware.
-	 * If we do call the h/w we use the derived format, not the
-	 * format passed to this routine.
-	 */
-	if (hw_sr != sample_rate || hw_chs != new_chs ||
-	    hw_enc != new_enc || hw_prec != new_prec || force) {
-		ATRACE("am_set_format() calling am_ad_set_format()", 0);
-		if (am_ad_set_format(statep, stpptr, ad_infop, stream, dir,
-		    sample_rate, new_chs, new_prec, new_enc, serialize) ==
-		    AUDIO_FAILURE) {
-			ATRACE("am_set_format() am_ad_set_format() failed", 0);
-			return (AUDIO_FAILURE);
-		}
-		/* update the true hardware image */
-		/* XXX this is probably not right! */
-		_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*stpptr))
-		if (dir == AUDIO_PLAY) {
-			stpptr->am_hw_info.play.sample_rate = sample_rate;
-			stpptr->am_hw_pchs = new_chs;
-			stpptr->am_hw_pprec = new_prec;
-			stpptr->am_hw_penc = new_enc;
-		} else {
-			ASSERT(dir == AUDIO_RECORD);
-			stpptr->am_hw_info.record.sample_rate = sample_rate;
-			stpptr->am_hw_rchs = new_chs;
-			stpptr->am_hw_rprec = new_prec;
-			stpptr->am_hw_renc = new_enc;
-		}
-		_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*stpptr))
-	}
-
-	/* there's no hardware change so just return */
-	ATRACE("am_set_format() just returning", 0);
-	return (AUDIO_SUCCESS);
-
-}	/* am_set_format() */
-
-/*
- * am_set_gain()
- *
- * Description:
- *	This routine is used to set the gain of all channels in the Codec.
- *	The gain is modified by balance. We try two different methods, the
- *	first with the gain and balance, and the second with gain and balance
- *	mixed down into left and right gain. This lets the Audio Driver accept
- *	whichever format it prefers. If the Audio Driver doesn't like the
- *	first method it returns AUDIO_FAILURE and the second is tried.
- *
- *	Some Codecs, like the Crystal 4231, will copy a mono input signal
- *	over to the 2nd channel. If this is the case then we apply balance
- *	to the left and right channels. Otherwise we adjust the only left
- *	gain.
- *
- *	NOTE: We change the gain only if it actually did change.
- *
- * Arguments:
- *	audio_state_t	*statep		Pointer to the device instance's state
- *	audio_apm_info_t *apm_infop	Ptr to driver's audio_apm_info structure
- *	uint_t		channels	The number of h/w channels
- *	uint_t		gain		The gain to set
- *	uint_t		balance		The balance to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		stream		The hardware stream to set gain on
- *	int		force		Force the gain to be set
- *	int		serialize	Serialize access to the audio driver
- *
- * Returns:
- *	AUDIO_SUCCESS			The gain was successfully set
- *	AUDIO_FAILURE			The gain was not successfully set
- */
-int
-am_set_gain(audio_state_t *statep, audio_apm_info_t *apm_infop, uint_t channels,
-	uint_t gain, uint_t balance, int dir, int stream, int force,
-	int serialize)
-{
-	audio_prinfo_t		*hw_pinfo;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	uint_t			g;
-
-	ATRACE("in am_set_gain()", apm_infop);
-	ATRACE("am_set_gain() channels", channels);
-	ATRACE("am_set_gain() gain", gain);
-	ATRACE("am_set_gain() balance", balance);
-
-	if (dir == AUDIO_PLAY) {
-		hw_pinfo = &((audio_info_t *)apm_infop->apm_ad_state)->play;
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		hw_pinfo = &((audio_info_t *)apm_infop->apm_ad_state)->record;
-	}
-
-	/* 1st try the gain and balance method since it's the easiest for us */
-	if (am_ad_set_config(statep, stpptr, ad_infop, stream, AM_SET_GAIN_BAL,
-	    dir, gain, balance, serialize) == AUDIO_SUCCESS) {
-		ATRACE("am_set_gain() AM_SET_GAIN_BAL successful", 0);
-		return (AUDIO_SUCCESS);
-	}
-
-	if (channels == 1 && !(ad_infop->ad_misc_flags & AM_MISC_MONO_DUP)) {
-		/* make sure there was a change */
-		if (!force && (hw_pinfo->gain == gain)) {
-			ATRACE_32("am_set_gain() mono, the same gain", gain);
-			return (AUDIO_SUCCESS);
-		}
-
-		/* we always set left gain */
-		if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-		    AM_SET_GAIN, dir, gain, 0, serialize) == AUDIO_FAILURE) {
-			return (AUDIO_FAILURE);
-		}
-		return (AUDIO_SUCCESS);
-	} else {
-		/* make sure there was a change */
-		if (!force &&
-		    (hw_pinfo->gain == gain) &&
-		    (hw_pinfo->balance == balance)) {
-			ATRACE_32("am_set_gain() stereo, the same gain", gain);
-			ATRACE_32("am_set_gain() stereo, the same balance",
-			    balance);
-			return (AUDIO_SUCCESS);
-		}
-
-		/*
-		 * Balance adjusts gain. If balance < 32 then left is
-		 * enhanced by attenuating right. If balance > 32 then
-		 * right is enhanced by attenuating left.
-		 */
-		if (balance == AUDIO_MID_BALANCE) {	/* no adj. */
-			/* left channel */
-			ATRACE_32("am_set_gain() L1 gain", gain);
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_GAIN, dir, gain, 0, serialize) ==
-			    AUDIO_FAILURE) {
-				return (AUDIO_FAILURE);
-			}
-			/* right channel */
-			ATRACE_32("am_set_gain() R1 gain", gain);
-			return (am_ad_set_config(statep, stpptr, ad_infop,
-			    stream, AM_SET_GAIN, dir, gain, 1, serialize));
-		} else if (balance < AUDIO_MID_BALANCE) {
-			/*
-			 * l = gain
-			 * r = (gain * balance) / 32
-			 */
-			g = (gain * balance) >> AM_BALANCE_SHIFT;
-			/* left channel */
-			ATRACE_32("am_set_gain() L2 gain", gain);
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_GAIN, dir, gain, 0, serialize) ==
-			    AUDIO_FAILURE) {
-				return (AUDIO_FAILURE);
-			}
-			/* right channel */
-			ATRACE_32("am_set_gain() R2 gain", g);
-			return (am_ad_set_config(statep, stpptr, ad_infop,
-			    stream, AM_SET_GAIN, dir, g, 1, serialize));
-		} else {
-			/*
-			 * l = (gain * (64 - balance)) / 32
-			 * r = gain
-			 */
-			g = (gain * (AUDIO_RIGHT_BALANCE - balance)) >>
-			    AM_BALANCE_SHIFT;
-			/* left channel */
-			ATRACE_32("am_set_gain() L3 gain", g);
-			if (am_ad_set_config(statep, stpptr, ad_infop, stream,
-			    AM_SET_GAIN, dir, g, 0, serialize) ==
-			    AUDIO_FAILURE) {
-				return (AUDIO_FAILURE);
-			}
-			/* right channel */
-			ATRACE_32("am_set_gain() R3 gain", gain);
-			return (am_ad_set_config(statep, stpptr, ad_infop,
-			    stream, AM_SET_GAIN, dir, gain, 1, serialize));
-		}
-	}
-
-}	/* am_set_gain() */
-
-/*
- * Private utilities used only by this file.
- */
-
-/*
- * am_ck_bits_set32()
- *
- * Description:
- *	This routine figures out how many bits are set in the passed in val.
- *
- * Arguments:
- *	uint	val		The argument to test
- *
- * Returns:
- *	0 - 32			The number of bits set
- */
-int
-am_ck_bits_set32(uint_t val)
-{
-	uint_t		mask = 0x00000001u;
-	int		count;
-	int		i;
-
-	ATRACE_32("in am_ck_bits_set32()", val);
-
-	for (i = 0, count = 0; i < 32; i++) {
-		if (mask & val) {
-			count++;
-		}
-		mask <<= 1;
-	}
-
-	ATRACE_32("am_ck_bits_set32() done", count);
-
-	return (count);
-
-}	/* am_ck_bits_set32() */
-
-/*
- * am_exit_task()
- *
- * Description:
- *	Exit from a task. This means decrementing the task counter so a
- *	blocked close() may continue.
- *
- * Arguments:
- *	audio_ch_t		*chptr		Ptr to the channel's struct
- *
- * Returns:
- *	void
- */
-static void
-am_exit_task(audio_ch_t *chptr)
-{
-	am_ch_private_t		*chpptr = chptr->ch_private;
-
-	ATRACE("in am_exit_task()", chptr);
-
-	mutex_enter(&chptr->ch_lock);
-
-	/* clear the taskq flag */
-	chpptr->acp_flags &= ~AM_CHNL_IOCTL_TASK;
-
-	ATRACE_32("am_exit_task() flag cleared", chpptr->acp_flags);
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_exit_task() done", chpptr);
-
-}	/* am_exit_task() */
-
-/*
- * am_fix_info()
- *
- * Description:
- *	When in mixer mode we usually play at a different sample rate
- *	than the data stream from the application. Therefore the sample
- *	count from the Codec is meaningless. This routine adjusts for the
- *	difference in sample rates.
- *
- *	We only adjust the play sample count because when recording you send
- *	x samples so you always know how many samples you sent so you don't
- *	have to adjust.
- *
- *	If this is an AUDIOCTL channel and it is associated with the H/W
- *	we don't do anything.
- *
- *	We also fix port and pause info, as well as other H/W related info,
- *	depending on the mixer mode.
- *
- * Arguments:
- *	audio_ch_t		*chptr	Ptr to the channel's state structure
- *	audio_info_t		*info	Ptr to the info structure to update
- *
- * Returns:
- *	void
- */
-static void
-am_fix_info(audio_ch_t *chptr, audio_info_t *info)
-{
-	audio_device_type_e	type = chptr->ch_info.dev_type;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	int			mode = stpptr->am_pstate->apm_mode;
-
-	ATRACE("in am_fix_info()", chptr);
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	/* first, update the features */
-	info->hw_features =		hw_info->hw_features;
-	info->sw_features =		hw_info->sw_features;
-	info->sw_features_enabled =	hw_info->sw_features_enabled;
-
-	/* now fix various other things */
-	if (mode == AM_MIXER_MODE) {
-		ATRACE("am_fix_info() fixing other things", 0);
-		if (info->play.pause) {
-			info->play.active = 0;
-		}
-		info->play.port = hw_info->play.port;
-		if (info->record.pause) {
-			info->record.active = 0;
-
-		}
-		info->record.port = hw_info->record.port;
-		info->monitor_gain = hw_info->monitor_gain;
-	}
-
-	/*
-	 * CAUTION: Don't place anything related to record below this
-	 *	point. Otherwise it may not execute.
-	 *
-	 * Finally, fix play samples, if we need to.
-	 */
-	if (!chpptr->acp_writing || (type == AUDIOCTL &&
-	    hw_info == chptr->ch_info.info)) {
-
-		ATRACE_32("am_fix_info() not writing, returning",
-		    chpptr->acp_writing);
-
-		return;
-	} else {
-		if (mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			ATRACE("am_fix_info() sample conversion", 0);
-
-			info->play.samples +=
-			    ad_infop->ad_play.ad_conv->ad_src_adjust(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_PLAY,
-			    chpptr->acp_psamples_p);
-		} else {
-			ATRACE("am_fix_info() NO sample conversion",
-			    info->play.samples);
-
-			info->play.samples += chpptr->acp_psamples_p;
-		}
-	}
-
-	ATRACE("am_fix_info() done", info);
-
-}	/* am_fix_info() */
-
-/*
- * am_fix_play_pause()
- *
- * Description:
- *	Convenience routine to clean up the code for switching to
- *	mixer mode while paused.
- *
- * Arguments:
- *	audio_ch_t		*chptr		Ptr to the channel
- *
- * Returns:
- *	void
- */
-static void
-am_fix_play_pause(audio_ch_t *chptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_info_t		*info = chptr->ch_info.info;
-	audio_data_t		*data;
-
-	ATRACE("in am_fix_play_pause()", chptr);
-
-	/*
-	 * It is possible that the channel was paused and then the mode
-	 * was switched. Thus we are most likely changing modes not on
-	 * a message boundary. Thus we need to make a best guess as to
-	 * where to start playing.
-	 */
-	if (info->play.pause &&
-	    (data = audio_sup_get_audio_data(chptr)) != NULL) {
-		/* it's remotely possible that we happen to be at the end */
-		if (data->adata_optr >= data->adata_oeptr) {
-			ATRACE("am_fix_play_pause() end of message", statep);
-
-			/* don't let it be played again! */
-			audio_sup_free_audio_data(data);
-			return;
-		} else if (data->adata_orig == data->adata_optr) {
-			/* or that we are just about to use a new msg */
-			ATRACE("am_fix_play_pause() new message", statep);
-
-			/* put it back to use next */
-			audio_sup_putback_audio_data(chptr, data);
-			return;
-		}
-
-		/* see if we need to process the data */
-		mutex_enter(&chptr->ch_lock);
-		if (data->adata_proc == NULL && am_reprocess(chptr, data) ==
-		    AUDIO_FAILURE) {
-			mutex_exit(&chptr->ch_lock);
-			/*
-			 * For some reason we can't convert the message and
-			 * there isn't much we can do, so just blow it away
-			 * and live with the gap in audio. We don't fail
-			 * changing modes because that went okay.
-			 */
-			ATRACE("am_fix_play_pause() am_reprocess() failed",
-			    statep);
-			return;
-		}
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_fix_play_pause() process successful", data);
-
-		/* put it back to use next */
-		audio_sup_putback_audio_data(chptr, data);
-	}
-
-	ATRACE("am_fix_play_pause() successful", statep);
-
-}	/* am_fix_play_pause() */
-
-/*
- * am_mixer_task_acknack()
- *
- * Description:
- *	Sometimes when a taskq thread is done the ioctl() is also done.
- *	To finish off an ioctl() an M_IOCACK or M_IOCNAK message must be sent
- *	up to the STREAMS head. This routine performs that task.
- *
- * Arguments:
- *	audio_i_state_t	*state		Pointer to the ioctl() state structure
- *	audio_ch_t	*chptr		Ptr 2 the channel's state structure
- *	queue_t		*q		Pointer to the STREAMS queue
- *	mblk_t		*mp		Pointer to the STREAMS message to use
- *	am_ioctl_args_t	*arg		Argument structure
- *	int		error		0 if no error, errno otherwise
- *
- * Returns:
- *	void
- */
-static void
-am_mixer_task_acknack(audio_i_state_t *state, audio_ch_t *chptr,
-    queue_t *q, mblk_t *mp, am_ioctl_args_t *arg, int error)
-{
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-
-	ATRACE("in am_mixer_task_acknack()", state);
-
-	/* no memory leaks allowed */
-	if (state->ais_address2) {
-		freemsg((mblk_t *)state->ais_address2);
-	}
-
-	if (cqp->cq_private) {
-		kmem_free(cqp->cq_private, sizeof (audio_i_state_t));
-		cqp->cq_private = NULL;
-	}
-
-	ATRACE_32("am_mixer_task_acknack() error", error);
-
-	if (error) {
-		miocnak(q, mp, 0, error);
-	} else {
-		miocack(q, mp, 0, 0);
-	}
-
-	/* let am_close() proceed */
-	am_exit_task(chptr);
-
-	kmem_free(arg, sizeof (*arg));
-
-	ATRACE("am_mixer_task_acknack() done", mp);
-
-}	/* am_mixer_task_acknack() */
-
-/*
- * am_restart()
- *
- * Description:
- *	This routine is used to restart playing and recording audio when
- *	they have been stopped to switch mixer modes.
- *
- *	NOTE: This can only be for traditional Codecs, multi-stream Codecs
- *		aren't stopped to changed modes.
- *
- * Arguments:
- *	audio_state_t	*statep		Pointer to the device instance's state
- *	audio_info_t	*hw_info	Pointer to the hardware state
- *
- * Returns:
- *	void
- */
-static void
-am_restart(audio_state_t *statep, audio_info_t *hw_info)
-{
-	audio_ch_t		*tchptr;
-	am_ch_private_t		*chpptr;
-	audio_apm_info_t	*apm_infop;
-	am_apm_private_t	*stpptr;
-	am_ad_info_t		*ad_infop;
-	audio_info_t		*tinfo;
-	int			i;
-	int			max_chs = statep->as_max_chs;
-
-	ATRACE("in am_restart()", statep);
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "am_restart() failed");
-		return;
-	}
-	stpptr = apm_infop->apm_private;
-	ad_infop = apm_infop->apm_ad_infop;
-
-	for (i = 0, tchptr = &statep->as_channels[0]; i < max_chs;
-	    i++, tchptr++) {
-
-		/* skip non-AUDIO and unallocated channels */
-		mutex_enter(&tchptr->ch_lock);
-		if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    tchptr->ch_info.dev_type != AUDIO ||
-		    tchptr->ch_info.pid == 0) {
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-
-		chpptr = tchptr->ch_private;
-		tinfo = tchptr->ch_info.info;
-
-		if (chpptr->acp_writing) {
-			/* turn the Q back on */
-			enableok(WR(tchptr->ch_qptr));
-			qenable(WR(tchptr->ch_qptr));
-
-			/* make sure we'll be flow controlled in am_wsvc() */
-			chpptr->acp_flags &= ~AM_CHNL_PFLOW;
-
-			/*
-			 * It is possible we switched modes right when the last
-			 * message was played and there's no way we're going
-			 * to get the three calls to am_get_samples() we need
-			 * to call cv_signal() for AUDIO_DRAIN. Therefore we
-			 * set AM_CHNL_EMPTY, which means the next time
-			 * am_get_samples() is called, which will happen when
-			 * we start playing below, it does the cv_signal().
-			 * We can subvert the number of times am_get_samples()
-			 * needs to be called because we know the DMA buffers
-			 * have been drained.
-			 */
-			if (audio_sup_get_audio_data_cnt(tchptr) == 0 &&
-			    chpptr->acp_flags & AM_CHNL_DRAIN) {
-				chpptr->acp_flags &= ~(AM_CHNL_ALMOST_EMPTY1|\
-				    AM_CHNL_ALMOST_EMPTY2|AM_CHNL_DRAIN|\
-				    AM_CHNL_DRAIN_NEXT_INT);
-				chpptr->acp_flags |= AM_CHNL_EMPTY;
-				/* signal any DRAIN situation */
-				am_audio_drained(tchptr);
-			}
-			ATRACE("am_restart() starting playing again", tchptr);
-
-			mutex_exit(&tchptr->ch_lock);
-			if (!tinfo->play.pause) {
-				/* set before start for mode switch */
-				if (tinfo->play.active == 0) {
-					am_send_signal(statep, stpptr);
-				}
-				tinfo->play.active = 1;
-				hw_info->play.active = 1;
-				if (am_ad_start_play(statep, stpptr, ad_infop,
-				    tchptr->ch_info.ch_number, AM_SERIALIZE) ==
-				    AUDIO_FAILURE) {
-					/*
-					 * We don't change pause if failed to
-					 * start.
-					 */
-					tinfo->play.active = 0;
-					hw_info->play.active = 0;
-					/*
-					 * Since we are part of the mode switch
-					 * we don't have to worry about the
-					 * mode switch CV like am_wsvc() has to.
-					 */
-				}
-			}
-			mutex_enter(&tchptr->ch_lock);
-		}
-
-		if (chpptr->acp_reading) {
-			ATRACE("am_restart() starting recording again", tchptr);
-			if (!tinfo->record.pause) {
-				/* set before start for mode switch */
-				mutex_exit(&tchptr->ch_lock);
-				if (tinfo->record.active == 0) {
-					am_send_signal(statep, stpptr);
-				}
-				tinfo->record.active = 1;
-				hw_info->record.active = 1;
-				if (am_ad_start_record(statep, stpptr, ad_infop,
-				    tchptr->ch_info.ch_number, AM_SERIALIZE) ==
-				    AUDIO_FAILURE) {
-					/*
-					 * We don't change pause if failed to
-					 * start.
-					 */
-					tinfo->record.active = 0;
-					hw_info->record.active = 0;
-				}
-			} else {
-				mutex_exit(&tchptr->ch_lock);
-			}
-		} else {
-			mutex_exit(&tchptr->ch_lock);
-		}
-	}
-
-}	/* am_restart() */
-
-/*
- * am_sched_task()
- *
- * Description:
- *	Common routine called to place a task on the taskq.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *	void		(*func)(void *)	Pointer to the task to schedule
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_sched_task(queue_t *q, mblk_t *mp, audio_ch_t *chptr,
-	void (*func)(void *))
-{
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ioctl_args_t		*arg;
-
-	ATRACE("in am_sched_task()", q);
-
-	/* get the arg structure and fill it in */
-	if ((arg = kmem_alloc(sizeof (*arg), KM_NOSLEEP)) == NULL) {
-		return (ENOMEM);
-	}
-
-	arg->aia_q = q;
-	arg->aia_mp = mp;
-
-	/* schedule the task */
-	if (audio_sup_taskq_dispatch(stpptr->am_taskq, func, arg, KM_NOSLEEP) ==
-	    AUDIO_FAILURE) {
-		/* let am_close() proceed and free the arg structure */
-		kmem_free(arg, sizeof (*arg));
-		am_exit_task(chptr);
-		return (EIO);
-	}
-
-	ATRACE("am_sched_task() returning", chptr);
-
-	return (0);
-
-}	/* am_sched_task() */
-
-/*
- * am_set_compat_mode()
- *
- * Description:
- *	This routine is used to convert the mixer from MIXER mode to COMPAT
- *	mode. Any playing and recording channels should have been stopped
- *	before this routine is called.
- *
- *	When this routine is called there may be one playing and one recording
- *	channel.
- *
- *	We don't have to worry about resetting psamples_f after calling
- *	am_audio_set_info() because am_get_samples() has been called twice
- *	while we wait to shutdown. Thus it has already been added into the
- *	sample count.
- *
- *	NOTE: Only traditional Codecs will use this code.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Ptr to the channel changing the mode
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	audio_ch_t	*pchptr		Ptr to the play channel
- *	audio_ch_t	*pchptr		Ptr to the record channel
- *
- * Returns:
- *	AUDIO_SUCCESS		Mode change completed successfully.
- *	AUDIO_FAILURE		Mode change failed.
- */
-static int
-am_set_compat_mode(audio_ch_t *chptr, am_ad_info_t *ad_infop,
-	audio_ch_t *pchptr, audio_ch_t *rchptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop;
-	audio_data_t		*data;
-	audio_info_t		*hw_info;
-	am_apm_persist_t	*persistp;
-	am_apm_private_t	*stpptr;
-	audio_ch_t		nchptr;
-	audio_info_t		new_info;
-	am_ch_private_t		ch_private;
-	long			tmp;
-	int			tmp_pgain;
-	int			tmp_pbal;
-	int			tmp_rgain;
-	int			tmp_rbal;
-	uchar_t			popen = 0;
-	uchar_t			ropen = 0;
-
-	ATRACE("in am_set_compat_mode()", chptr);
-	ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_set_compat_mode() audio_sup_get_apm_info() failed",
-		    statep);
-		return (AUDIO_FAILURE);
-	}
-	stpptr = apm_infop->apm_private;
-	hw_info = &stpptr->am_hw_info;
-	tmp_pgain = hw_info->play.gain;
-	tmp_pbal = hw_info->play.balance;
-	tmp_rgain = hw_info->record.gain;
-	tmp_rbal = hw_info->record.balance;
-
-	/* copy the original channel structure to the temp, just in case */
-	bcopy(chptr, &nchptr, sizeof (nchptr));
-
-	/* we only reset the hardware if we are playing or recording */
-	AUDIO_INIT(&new_info, sizeof (new_info));
-	bzero(&ch_private, sizeof (ch_private));
-
-	nchptr.ch_dir = 0;
-
-	if (pchptr) {
-		audio_info_t *p_info = pchptr->ch_info.info;
-
-		new_info.play.sample_rate = p_info->play.sample_rate;
-		new_info.play.channels = p_info->play.channels;
-		new_info.play.precision = p_info->play.precision;
-		new_info.play.encoding = p_info->play.encoding;
-		new_info.play.gain = p_info->play.gain;
-		new_info.play.balance = p_info->play.balance;
-		new_info.play.samples = p_info->play.samples;
-		new_info.play.eof = p_info->play.eof;
-		popen = p_info->play.open;
-		ch_private.acp_writing = 1;
-		nchptr.ch_dir |= AUDIO_PLAY;
-	}
-	if (rchptr) {
-		audio_info_t *r_info = rchptr->ch_info.info;
-
-		new_info.record.sample_rate = r_info->record.sample_rate;
-		new_info.record.channels = r_info->record.channels;
-		new_info.record.precision = r_info->record.precision;
-		new_info.record.encoding = r_info->record.encoding;
-		new_info.record.gain = r_info->record.gain;
-		new_info.record.balance = r_info->record.balance;
-		new_info.record.samples = r_info->record.samples;
-		ropen = r_info->record.open;
-		ch_private.acp_reading = 1;
-		nchptr.ch_dir |= AUDIO_RECORD;
-	}
-
-	/* we always save the hardware state, even if no play/rec channels */
-	stpptr->am_pstate->apm_mode = AM_COMPAT_MODE;
-	hw_info->sw_features_enabled &= ~AUDIO_SWFEATURE_MIXER;
-
-	/* change the hardware, if we had active play/record channels */
-	if (pchptr || rchptr) {
-		nchptr.ch_qptr = chptr->ch_qptr;
-		nchptr.ch_statep = chptr->ch_statep;
-		nchptr.ch_info.dev_type = AUDIO;
-		nchptr.ch_apm_infop = apm_infop;
-		nchptr.ch_private = &ch_private;
-		nchptr.ch_info.info = &new_info;
-		/*
-		 * It's possible that when the bcopy above happens,
-		 * the ch_lock is held by someone else. It should be
-		 * cleared. Or, later when we try to hold it and find
-		 * it's held, deadlock may happen.
-		 */
-		mutex_init(&nchptr.ch_lock, NULL, MUTEX_DRIVER, NULL);
-
-		if (am_audio_set_info(&nchptr, &new_info, NULL) ==
-		    AUDIO_FAILURE) {
-			ATRACE("am_set_compat_mode() am_audio_set_info()"
-			    "failed", &nchptr);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "set_compat() couldn't reconfigure hardware");
-			stpptr->am_pstate->apm_mode = AM_MIXER_MODE;
-			hw_info->sw_features_enabled |= AUDIO_SWFEATURE_MIXER;
-			mutex_destroy(&nchptr.ch_lock);
-			return (AUDIO_FAILURE);
-		}
-		mutex_destroy(&nchptr.ch_lock);
-	}
-
-	/* update the hardware open flags */
-	hw_info->play.open = popen;
-	hw_info->record.open = ropen;
-
-	/* update persistent memory */
-	persistp = stpptr->am_pstate;
-	persistp->apm_mpgain = tmp_pgain;
-	persistp->apm_mpbal = tmp_pbal;
-	persistp->apm_mrgain = tmp_rgain;
-	persistp->apm_mrbal = tmp_rbal;
-
-	/*
-	 * It is possible that the channel was paused and then the mode
-	 * was switched. Thus we are most likely changing modes not on
-	 * a message boundary. Thus we need to make a best guess as to
-	 * where to start playing.
-	 */
-	if (pchptr &&
-	    ((audio_info_t *)pchptr->ch_info.info)->play.pause &&
-	    (data = audio_sup_get_audio_data(pchptr)) != NULL) {
-		/* it's remotely possible that we happen to be at the end */
-		if (data->adata_pptr >= data->adata_peptr) {
-			ATRACE("am_set_compat_mode() end of data", statep);
-
-			/* don't let it be played again! */
-			audio_sup_free_audio_data(data);
-			return (AUDIO_SUCCESS);
-		} else if (data->adata_pptr == data->adata_proc) {
-			/* or that we are just about to use new data */
-			ATRACE("am_set_compat_mode() new message", statep);
-
-			/* put it back to use next */
-			audio_sup_putback_audio_data(pchptr, data);
-			return (AUDIO_SUCCESS);
-		}
-
-		/*
-		 * Make a guess as to where to point to. We make sure we are
-		 * on a 4 byte boundary. That way we don't have to worry
-		 * about being in the middle of sample.
-		 *
-		 * The equation:
-		 *	(offset of proc data from start)*(length of orig data)
-		 *	--------------------------------------------------------
-		 *		(length of proc data)
-		 */
-		ATRACE("am_set_compat_mode() orig", data->adata_orig);
-		ATRACE("am_set_compat_mode() optr", data->adata_optr);
-		ATRACE("am_set_compat_mode() oeptr", data->adata_oeptr);
-		ATRACE_32("am_set_compat_mode(): osize", data->adata_osize);
-		ATRACE("am_set_compat_mode() proc", data->adata_proc);
-		ATRACE("am_set_compat_mode() pptr", data->adata_pptr);
-		ATRACE("am_set_compat_mode() peptr", data->adata_peptr);
-		ATRACE_32("am_set_compat_mode(): psize", data->adata_psize);
-
-		tmp = (((char *)data->adata_pptr - (char *)data->adata_proc) *
-		    data->adata_osize) / data->adata_psize;
-
-		/*
-		 * tmp is an offset, which must be added to adata_orig to
-		 * get adata_optr. We mask off adata_optr so that regardless
-		 * of the format of the data we always are on a sample frame
-		 * boundary.
-		 */
-		data->adata_optr = (char *)data->adata_orig +
-		    (tmp & ~AM_MISC_MASK);
-		ATRACE("am_set_compat_mode() new optr", data->adata_optr);
-
-		/* put it back to use next */
-		audio_sup_putback_audio_data(pchptr, data);
-	}
-
-	ATRACE("am_set_compat_mode() done", 0);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_set_compat_mode() */
-
-/*
- * am_set_mixer_mode()
- *
- * Description:
- *	This routine is used to convert the mixer from COMPAT mode to MIXER
- *	mode. Any playing and recording channels should have been stopped
- *	before this routine is called.
- *
- *	When this routine is called there may be one playing and one recording
- *	channel.
- *
- *	Just like am_set_compat_mode(), psamples_f has already been added into
- *	the played sample count. So we don't need to do anything with it here.
- *
- *	NOTE: Only traditional Codecs will use this code.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Ptr to the channel changing the mode
- *	am_ad_info_t	*ad_infop	Ptr to the Audio Driver's config info
- *	am_apm_private_t **stpptr	Ptr to the mixer's private state data
- *	audio_apm_info_t *apm_infop	Ptr to the mixer's APM info structure
- *	audio_ch_t	*pchptr		Ptr to the play channel
- *	audio_ch_t	*rchptr		Ptr to the record channel
- *
- * Returns:
- *	AUDIO_SUCCESS		Mode change completed successfully.
- *	AUDIO_FAILURE		Mode change failed.
- */
-static int
-am_set_mixer_mode(audio_ch_t *chptr, am_ad_info_t *ad_infop,
-	am_apm_private_t *stpptr, audio_ch_t *pchptr, audio_ch_t *rchptr)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop;
-	audio_ch_t		nchptr;
-	am_ch_private_t		ch_private;
-	audio_info_t		*hw_info;
-	audio_info_t		new_info;
-	am_apm_persist_t	*persistp;
-
-	ATRACE("in am_set_mixer_mode()", statep);
-	ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_set_mixer_mode() audio_sup_get_apm_info() failed",
-		    statep);
-		return (AUDIO_FAILURE);
-	}
-	persistp = stpptr->am_pstate;
-	hw_info = apm_infop->apm_ad_state;
-
-	/*
-	 * see if we need to update the sample rate conv. routines. This should
-	 * be done before setting the apm_mode to AM_MIXER_MODE. Or, other code
-	 * path,e.g. am_flush, may find apm_mode == AM_MIXER_MODE and then find
-	 * the src module has not been update correctly and panic the system.
-	 * We update the src module before update the hardware so that if it
-	 * fails, the hardware state is still correct.
-	 */
-	if (pchptr) {
-		ATRACE("am_set_mixer_mode(), "
-		    "calling play src update", pchptr);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "am_set_mixer_mode calling play src update");
-		if (ad_infop->ad_play.ad_conv->ad_src_update(
-		    AM_SRC_CHPTR2HDL(pchptr),
-		    &((audio_info_t *)pchptr->ch_info.info)->
-		    play,
-		    &((audio_info_t *)apm_infop->apm_ad_state)->
-		    play,
-		    ((am_ad_info_t *)apm_infop->apm_ad_infop)->
-		    ad_play.ad_sr_info,
-		    AUDIO_PLAY) == AUDIO_FAILURE) {
-			ATRACE("am_set_mixer_mode() "
-			    "play src_update() failed", 0);
-			return (AUDIO_FAILURE);
-		}
-	}
-	if (rchptr) {
-		ATRACE("am_set_mixer_mode(), "
-		    "calling record src update", rchptr);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "am_set_mixer_mode calling record src update");
-		if (ad_infop->ad_record.ad_conv->ad_src_update(
-		    AM_SRC_CHPTR2HDL(rchptr),
-		    &((audio_info_t *)rchptr->ch_info.info)->
-		    record,
-		    &((audio_info_t *)apm_infop->apm_ad_state)->
-		    record,
-		    ((am_ad_info_t *)apm_infop->apm_ad_infop)->
-		    ad_record.ad_sr_info,
-		    AUDIO_RECORD) == AUDIO_FAILURE) {
-			ATRACE("am_set_mixer_mode() "
-			    "record src_update() failed", 0);
-			return (AUDIO_FAILURE);
-		}
-	}
-
-	/* copy the original channel structure to the temp, just in case */
-	bcopy(chptr, &nchptr, sizeof (nchptr));
-
-	/* we always reset the hardware, even if no play/rec channels */
-	AUDIO_INIT(&new_info, sizeof (new_info));
-	bzero(&ch_private, sizeof (ch_private));
-
-	if (pchptr) {
-		if (am_ck_sample_rate(&ad_infop->ad_play, AM_MIXER_MODE,
-		    hw_info->play.sample_rate) == AUDIO_FAILURE) {
-			ATRACE("am_set_mixer_mode() mixer can't play using "
-			    "unsupported sample rate",
-			    hw_info->play.sample_rate);
-			return (AUDIO_FAILURE);
-		}
-		new_info.play.samples =
-		    ((audio_info_t *)pchptr->ch_info.info)->play.samples;
-		ch_private.acp_writing = 1;
-	}
-	if (rchptr) {
-		if (am_ck_sample_rate(&ad_infop->ad_record, AM_MIXER_MODE,
-		    hw_info->record.sample_rate) == AUDIO_FAILURE) {
-			ATRACE("am_set_mixer_mode() mixer can't record using "
-			    "unsupported sample rate",
-			    hw_info->record.sample_rate);
-			return (AUDIO_FAILURE);
-		}
-		new_info.record.samples =
-		    ((audio_info_t *)rchptr->ch_info.info)->record.samples;
-		ch_private.acp_reading = 1;
-	}
-
-	if (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY) {
-		new_info.play.sample_rate = stpptr->am_save_psr;
-		new_info.play.gain = persistp->apm_mpgain;
-		new_info.play.balance = persistp->apm_mpbal;
-	}
-
-	if (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD) {
-		new_info.record.sample_rate = stpptr->am_save_rsr;
-		new_info.record.gain = persistp->apm_mrgain;
-		new_info.record.balance = persistp->apm_mrbal;
-	}
-
-	nchptr.ch_qptr = chptr->ch_qptr;
-	nchptr.ch_statep = chptr->ch_statep;
-	nchptr.ch_dir = AUDIO_BOTH;
-	nchptr.ch_info.dev_type = AUDIO;
-	nchptr.ch_apm_infop = apm_infop;
-	nchptr.ch_private = &ch_private;
-	nchptr.ch_info.info = &new_info;
-	/*
-	 * It's possible that when the bcopy above happens,
-	 * the ch_lock is held by someone else. It should be
-	 * cleared. Or, later when we try to hold it and find
-	 * it's held, deadlock may happen.
-	 */
-	mutex_init(&nchptr.ch_lock, NULL, MUTEX_DRIVER, NULL);
-
-	if (am_audio_set_info(&nchptr, &new_info, &new_info) == AUDIO_FAILURE) {
-		ATRACE("am_set_mixer_mode() am_audio_set_info() failed",
-		    &nchptr);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "set_mixer() "
-		    "couldn't reconfigure hardware");
-		mutex_destroy(&nchptr.ch_lock);
-		return (AUDIO_FAILURE);
-	}
-	mutex_destroy(&nchptr.ch_lock);
-
-
-	/*
-	 * We need to look like we've changed modes AFTER we try to set hw.
-	 * Otherwise am_audio_set_info() won't update the hardware. It'll
-	 * try to update the virtual channel.
-	 */
-	stpptr->am_pstate->apm_mode = AM_MIXER_MODE;
-	hw_info->sw_features_enabled |= AUDIO_SWFEATURE_MIXER;
-
-	/* clear the open flags in the hardware */
-	hw_info->play.open = 0;
-	hw_info->record.open = 0;
-
-	/* clear the hardware's waiting flags */
-	hw_info->play.waiting = 0;
-	hw_info->record.waiting = 0;
-
-	/* clear misc flags */
-	hw_info->play.eof = 0;
-
-	/*
-	 * Also update the hardware info for the number of channels,
-	 * precision, and encoding.
-	 */
-	hw_info->play.channels = stpptr->am_hw_pchs;
-	hw_info->record.channels = stpptr->am_hw_rchs;
-
-	hw_info->play.precision = stpptr->am_hw_pprec;
-	hw_info->record.precision = stpptr->am_hw_rprec;
-
-	hw_info->play.encoding = stpptr->am_hw_penc;
-	hw_info->record.encoding = stpptr->am_hw_renc;
-
-	ATRACE("am_set_mixer_mode() successful", statep);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_set_mixer_mode() */
-
-/*
- * am_wiocdata_mixerctl_chinfo()
- *
- * Description:
- *	We have the audio_channel_t data structure so we know how big the info
- *	structure is. We make sure the size of the info structure is correct,
- *	that there is a buffer, and that the channel number is reasonable.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_i_state_t	*state	Pointer to ioctl() state structure
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_wiocdata_mixerctl_chinfo(queue_t *q, mblk_t *mp, audio_i_state_t *state)
-{
-	STRUCT_HANDLE(audio_channel, audio_channel);
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	struct iocblk		*iocbp = (struct iocblk *)mp->b_rptr;
-	audio_info_t		*info;
-	int			ch_number;
-
-	ATRACE("in am_wiocdata_mixerctl_chinfo()", q);
-
-	/* get ready to do the model conversion */
-	STRUCT_SET_HANDLE(audio_channel, iocbp->ioc_flag,
-		(audio_channel_t *)mp->b_cont->b_rptr);
-
-	/*
-	 * Make sure the size is good, fortunately audio_info_t doesn't have
-	 * any pointers in it, so it's the same size, regardless of _ILP32
-	 * or _LP64.
-	 */
-	if (STRUCT_FGET(audio_channel, info_size) != sizeof (audio_info_t)) {
-		ATRACE_32("am_wiocdata_mixerctl_chinfo() bad size",
-		    STRUCT_FGET(audio_channel, info_size));
-		return (EINVAL);
-	}
-
-	/* make sure the app has a buffer to place the data into */
-	if ((info = STRUCT_FGETP(audio_channel, info)) == NULL) {
-		ATRACE("am_wiocdata_mixerctl_chinfo() no buffer",
-		    STRUCT_FGETP(audio_channel, info));
-		return (EINVAL);
-	}
-
-	/* get the channel number and make sure it's good */
-	ch_number = STRUCT_FGET(audio_channel, ch_number);
-	if (ch_number >= statep->as_max_chs || ch_number < 0) {
-		ATRACE_32("am_wiocdata_mixerctl_chinfo() bad ch number",
-		    STRUCT_FGET(audio_channel, ch_number));
-		return (EINVAL);
-	}
-
-	/* make sure this is a valid AUDIO/AUDIOCTL ch */
-	tchptr = &statep->as_channels[ch_number];
-	mutex_enter(&tchptr->ch_lock);
-	if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    tchptr->ch_info.pid == 0 ||
-	    (tchptr->ch_info.dev_type != AUDIO &&
-	    tchptr->ch_info.dev_type != AUDIOCTL)) {
-		mutex_exit(&tchptr->ch_lock);
-		ATRACE("am_wiocdata_mixerctl_chinfo() bad ch", tchptr);
-		return (EINVAL);
-	}
-	mutex_exit(&tchptr->ch_lock);
-
-	/*
-	 * We have a good audio_channel_t structure so we can do the next
-	 * step in the process, which is to ask for the audio_info structure.
-	 * But first we save the audio_channel structure for later use.
-	 * The STREAMS head will give us a new mp->b_cont message block
-	 * when it gives us the audio_info structure.
-	 */
-	state->ais_command = AM_COPY_IN_MIXCTL_SET_CHINFO2;
-	state->ais_address = (caddr_t)info;
-	state->ais_address2 = (caddr_t)mp->b_cont;
-
-	/* Set mp->b_cont = NULL so mcopyin() does not free the saved message */
-	mp->b_cont = NULL;
-
-	/* Setup for copyin */
-	ASSERT(state->ais_address != NULL);
-	mcopyin(mp, state, sizeof (audio_info_t), (caddr_t)state->ais_address);
-
-	/* send the copy in request */
-	qreply(q, mp);
-
-	ATRACE("am_wiocdata_mixerctl_chinfo() returning", q);
-
-	return (0);
-
-}	/* am_wiocdata_mixerctl_chinfo() */
-
-/*
- * am_wiocdata_mixerctl_get_chinfo()
- *
- * Description:
- *	The audio_info_t data structure has been copied out, so now we copy
- *	out the updated audio_channel_t structure.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_i_state_t	*state	Pointer to ioctl() state structure
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_wiocdata_mixerctl_get_chinfo(queue_t *q, mblk_t *mp, audio_i_state_t *state)
-{
-	mblk_t		*tmp;
-
-	ATRACE("in am_wiocdata_mixerctl_get_chinfo()", q);
-
-	state->ais_command = AM_COPY_OUT_MIXCTL_GET_CHINFO2;
-
-	tmp = (mblk_t *)state->ais_address2;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, tmp->b_wptr - tmp->b_rptr, state->ais_address, tmp);
-
-	state->ais_address2 = NULL;
-
-	qreply(q, mp);
-
-	ATRACE("am_wiocdata_mixerctl_get_chinfo() done", q);
-
-	return (0);
-
-}	/* am_wiocdata_mixerctl_get_chinfo() */
-
-/*
- * am_wiocdata_sr()
- *
- * Description:
- *	The next step in getting the sample rates. We've got the
- *	am_sample_rates_t structure so we can now get the number of sample
- *	rates to ask for, which we do. Thus we copy in the structure a
- *	second time, but this time it'll be larger.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	struct copyreq	*cqp	Pointer to copy request register
- *	audio_i_state_t	*state	Pointer to ioctl() state structure
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_wiocdata_sr(queue_t *q, mblk_t *mp, struct copyreq *cqp,
-	audio_i_state_t *state)
-{
-	am_sample_rates_t	*new;
-	size_t			size;
-
-	ATRACE("in am_wiocdata_sr()", q);
-
-	/*
-	 * We copy in just the am_sample_rates_t structure to get the
-	 * number of sample rates the samp_rates array can support.
-	 * Once we know this we make another call to get the whole
-	 * thing.
-	 */
-
-	/* get a pointer to the user supplied structure */
-	new = (am_sample_rates_t *)mp->b_cont->b_rptr;
-
-	/* make sure the number of array elements is sane */
-	if (new->num_samp_rates <= 0) {
-		ATRACE_32("am_wiocdata_sr() AM_COPY_IN_SAMP_RATES "
-		    "bad num_samp_rates", new->num_samp_rates);
-		return (EINVAL);
-	}
-
-	ATRACE("am_wiocdata_sr() new", new);
-	ATRACE_32("am_wiocdata_sr() num_samp_rates", new->num_samp_rates);
-	size = AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(new->num_samp_rates);
-
-	/*
-	 * Now that we know the number of array elements, we can ask
-	 * for the right number of bytes.
-	 *
-	 * Reuse the cq_private buffer, saving data for M_IOCDATA processing.
-	 */
-	state->ais_command = AM_COPY_IN_SAMP_RATES2;
-	ASSERT(cqp->cq_private == (mblk_t *)state);
-
-	/* Setup for copyin */
-	ASSERT(state->ais_address != NULL);
-	mcopyin(mp, state, size, (caddr_t)state->ais_address);
-
-	/* send the copy in request */
-	qreply(q, mp);
-
-	ATRACE("am_wiocdata_sr() returning success", 0);
-
-	return (0);
-
-}	/* am_wiocdata_sr() */
-
-/*
- * am_wioctl_copyin()
- *
- * Description:
- *	Common routine used to start the copy in process for many ioctl()s.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *	audio_i_state_t	*state	Pointer to ioctl() state structure
- *	int		cmd	IOCTL command
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_wioctl_copyin(queue_t *q, mblk_t *mp, audio_ch_t *chptr,
-	audio_i_state_t *state, int cmd)
-{
-	size_t		size;
-	int		new_cmd;
-
-	ATRACE("in am_wioctl_copyin()", q);
-
-	/* set copyin based on the ioctl() command */
-	switch (cmd) {
-	case AUDIO_DIAG_LOOPBACK: {
-		audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-		am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-
-		/* not all Audio Drivers and their hardware support loopbacks */
-		if (!(ad_infop->ad_diag_flags & AM_DIAG_INTERNAL_LOOP)) {
-			ATRACE_32("am_wioctl_diag_loopback() no loopbacks",
-			    ad_infop->ad_diag_flags);
-			return (ENOTTY);
-		}
-
-		new_cmd = AM_COPY_IN_DIAG_LOOPB;
-		size = sizeof (int);
-
-		break;
-	}
-
-	case AUDIO_SETINFO:
-		new_cmd = AM_COPY_IN_AUDIOINFO;
-		size = sizeof (audio_info_t);
-
-		break;
-	case AUDIO_MIXER_GET_SAMPLE_RATES:
-		new_cmd = AM_COPY_IN_SAMP_RATES;
-		size = sizeof (am_sample_rates_t);
-
-		break;
-	case AUDIO_MIXERCTL_SETINFO:
-		/* allowed only on AUDIOCTL channels */
-		if (chptr->ch_info.dev_type != AUDIOCTL) {
-			return (EINVAL);
-		}
-
-		new_cmd = AM_COPY_IN_MIXCTLINFO;
-		/*
-		 * We only need the dev_info part of am_control_t. Fortunately
-		 * this is at the front of the structure. So set the size to
-		 * copy in only the dev_info part. Otherwise we'll blow an
-		 * assert in am_mixerctl_setinfo_task().
-		 */
-		size = sizeof (audio_info_t);
-
-		break;
-	case AUDIO_MIXERCTL_GET_CHINFO:
-		/* allowed only on AUDIOCTL channels */
-		if (chptr->ch_info.dev_type != AUDIOCTL) {
-			return (EINVAL);
-		}
-
-		/* size can be different, depending on _ILP32 and _LP64 */
-		new_cmd = AM_COPY_IN_MIXCTL_GET_CHINFO;
-		size = SIZEOF_STRUCT(audio_channel,
-		    ((struct iocblk *)mp->b_rptr)->ioc_flag);
-
-		break;
-	case AUDIO_MIXERCTL_SET_CHINFO:
-		/* allowed only on AUDIOCTL channels */
-		if (chptr->ch_info.dev_type != AUDIOCTL) {
-			return (EINVAL);
-		}
-
-		/* size can be different, depending on _ILP32 and _LP64 */
-		new_cmd = AM_COPY_IN_MIXCTL_SET_CHINFO;
-		size = SIZEOF_STRUCT(audio_channel,
-		    ((struct iocblk *)mp->b_rptr)->ioc_flag);
-
-		break;
-	case AUDIO_MIXERCTL_SET_MODE:
-		/* allowed only on AUDIOCTL channels */
-		if (chptr->ch_info.dev_type != AUDIOCTL) {
-			return (EINVAL);
-		}
-
-		new_cmd = AM_COPY_IN_MIXCTL_MODE;
-		size = sizeof (int);
-
-		break;
-	default:
-		return (EIO);
-	}
-
-	/* set up the message */
-	state->ais_command = new_cmd;
-	state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-	/* Setup for copyin */
-	if (state->ais_address == NULL) {
-		return (EINVAL);
-	}
-	mcopyin(mp, state, size, state->ais_address);
-
-	/* send the copy in request */
-	qreply(q, mp);
-
-	ATRACE("am_wioctl_copyin() returning", chptr);
-
-	return (0);
-
-}	/* am_wioctl_copyin() */
-
-/*
- * am_wioctl_drain()
- *
- * Description:
- *	First make sure this is an AUDIO channel. Then see if there is
- *	any queued up audio. If there is then just return after setting
- *	a flag. Otherwise we've got an AUDIO_DRAIN before there's any
- *	played audio or after it has all played, thus we ACK.
- *
- *	Once all audio has been played am_audio_drained() is called and
- *	it generates the ACK.
- *
- *	This ioctl() doesn't use the taskq. Thus other channels won't be
- *	blocked by this ioctl().
- *
- *	Like all ioctl()s, AUDIO_DRAIN may be interrupted by a signal.
- *	The STREAMS head will let the ioctl() return to the app. We have
- *	no way of knowing this has happened, unless we use cv_wait_sig().
- *	This would block am_wput() so we aren't going to use it. However
- *	the fact we don't know about the signal is okay. The STREAMS head
- *	keeps a sequence number and as long as we save the original mp it
- *	can tell if this is an old ioctl() or not. If it is it just ignores
- *	the ACK. Further, am_close() will call am_audio_drained() which
- *	will send the ACK, which the STREAMS head will ignore if it wishes.
- *	Or we'll get another AUDIO_DRAIN and we still have the old mp, and
- *	thus we'll know the old AUDIO_DRAIN was interrupted. So we just
- *	free the mp and we're still okay.
- *
- *	XXX - I don't understand what this means if there's a mux pushed
- *	on top of the mixer. If this happens it is possible to have multiple
- *	AUDIO_DRAINs at once, but then you can also have multiple audio
- *	streams on one channel. So audio isn't going to work very well in
- *	that case anyway.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *	struct copyreq	*cqp	Pointer to copy request register
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-am_wioctl_drain(queue_t *q, mblk_t *mp, audio_ch_t *chptr,
-	struct copyreq *cqp)
-{
-	am_ch_private_t		*chpptr = (am_ch_private_t *)chptr->ch_private;
-	int			error = 0;
-
-	ATRACE("in am_wioctl_drain() chptr", chptr);
-
-	/* must be on a play audio channel */
-	mutex_enter(&chptr->ch_lock);
-	if (!chpptr->acp_writing || chptr->ch_info.dev_type != AUDIO) {
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_wioctl_drain() AUDIO_DRAIN bad type", chptr);
-		error = EINVAL;
-		goto done;
-	}
-
-	/*
-	 * See if we've got a new AUDIO_DRAIN, which means the last one
-	 * was interrupted.
-	 */
-	if (chpptr->acp_drain_mp) {
-		/* free it, it's lost and the STREAM head knows this */
-		freemsg(chpptr->acp_drain_mp);
-	}
-
-	/* see if we are empty, the easy case */
-	if (!(chpptr->acp_flags & AM_CHNL_MSG_ON_QUEUE) &&
-	    audio_sup_get_audio_data_cnt(chptr) == 0) {
-		chpptr->acp_drain_mp = NULL;
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_wioctl_drain() no messages", chptr);
-		goto done;
-	}
-
-	/* we need to wait for empty */
-	chpptr->acp_flags |= AM_CHNL_DRAIN;
-	chpptr->acp_drain_mp = mp;
-
-	ATRACE_32("am_wioctl_drain() messages, acp_flags", chpptr->acp_flags);
-	ATRACE("am_wioctl_drain() MP", mp);
-
-	mutex_exit(&chptr->ch_lock);
-
-	return;
-
-done:
-	ASSERT(cqp->cq_private == NULL);
-
-	if (error) {
-		miocnak(q, mp, 0, error);
-	} else {
-		miocack(q, mp, 0, 0);
-	}
-
-	ATRACE("am_wioctl_drain() drained", chptr);
-
-}	/* am_wioctl_drain() */
-
-/*
- * am_wioctl_getdev()
- *
- * Description:
- *	The first half of the AUDIO_GETDEV ioctl(). Ask to copy out
- *	the audio driver's device information structure.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the STREAMS queue
- *	mblk_t		*mp	Pointer to the message block
- *	audio_ch_t	*chptr	Pointer to this channel's state information
- *	audio_i_state_t	*state	Pointer to ioctl() state structure
- *
- * Returns:
- *	0			No error
- *	errno			Error number for the error
- */
-static int
-am_wioctl_getdev(queue_t *q, mblk_t *mp, audio_ch_t *chptr,
-	audio_i_state_t *state)
-{
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	audio_device_t		*devp;
-
-	ATRACE("in am_wioctl_getdev()", q);
-
-	/* set STREAMS for copy out of the audio_device structure */
-	state->ais_command = AM_COPY_OUT_GETDEV;
-	state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-	/* Setup for copyout */
-	if (state->ais_address == NULL) {
-		return (EINVAL);
-	}
-	mcopyout(mp, state, sizeof (*ad_infop->ad_dev_info), state->ais_address,
-	    NULL);
-
-	/* put the data in the buffer, but try to reuse it first */
-	if (audio_sup_mblk_alloc(mp, sizeof (*ad_infop->ad_dev_info)) ==
-	    AUDIO_FAILURE) {
-		return (ENOMEM);
-	}
-
-	/*
-	 * We don't bother to lock the state structure because this
-	 * is static data.
-	 */
-
-	devp = (audio_device_t *)mp->b_cont->b_rptr;
-
-	bcopy(ad_infop->ad_dev_info, devp,
-	    sizeof (*ad_infop->ad_dev_info));
-
-	/* send the copy out request */
-	qreply(q, mp);
-
-	ATRACE("am_wioctl_getdev() returning", chptr);
-
-	return (0);
-
-}	/* am_wioctl_getdev() */
-
-/*
- * Task queue callbacks.
- */
-
-/*
- * am_diag_loopback_task()
- *
- * Description:
- *	Called by the task queue to set the loopback mode in the audio
- *	driver.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_diag_loopback_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	int			error = 0;
-
-	ATRACE("in am_diag_lookback() arg", arg);
-
-	am_enter_rwlock();
-
-	if (*(int *)mp->b_cont->b_rptr) {
-		ATRACE_32("am_diag_loopback_task() enable",
-		    *(int *)mp->b_cont->b_rptr);
-
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_DIAG_MODE, NULL, 1, NULL,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			ATRACE("am_diag_loopback_task() "
-			    "AM_COPY_IN_DIAG_LOOPB enable failed", 0);
-			error = EIO;
-		}
-	} else {
-		ATRACE_32("am_diag_loopback_task() disable",
-		    *(int *)mp->b_cont->b_rptr);
-
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_DIAG_MODE, NULL, 0, NULL,
-		    AM_SERIALIZE) == AUDIO_FAILURE) {
-			ATRACE("am_diag_loopback_task() "
-			    "AM_COPY_IN_DIAG_LOOPB disable failed", 0);
-			error = EIO;
-		}
-	}
-
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE("am_diag_lookback() done", error);
-
-}	/* am_diag_loopback_task() */
-
-/*
- * am_get_chinfo_task()
- *
- * Description:
- *	Called by the task queue to get the channel's info. If the size of
- *	the info structure doesn't match then we return an EINVAL. This is
- *	better than trying to copy out to much data and cause a core dump
- *	in the application.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_get_chinfo_task(void *arg)
-{
-	STRUCT_HANDLE(audio_channel, audio_channel);
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	struct iocblk		*iocbp = (struct iocblk *)mp->b_rptr;
-	audio_info_t		*info;
-	mblk_t			*tmp;
-	int			ch_number;
-	int			error = 0;
-
-	ATRACE("in am_get_chinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* we have to check the mode when it's stable */
-	if (stpptr->am_pstate->apm_mode != AM_MIXER_MODE) {
-		ATRACE_32("am_get_chinfo_task() bad mode",
-		    chptr->ch_info.dev_type);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* get ready to do the model conversion */
-	STRUCT_SET_HANDLE(audio_channel, iocbp->ioc_flag,
-	    (audio_channel_t *)mp->b_cont->b_rptr);
-
-	/*
-	 * Make sure the size is good, fortunately audio_info_t doesn't have
-	 * any pointers in it, so it's the same size, regardless of _ILP32
-	 * or _LP64.
-	 */
-	if (STRUCT_FGET(audio_channel, info_size) != sizeof (audio_info_t)) {
-		ATRACE_32("am_get_chinfo_task() bad size",
-		    STRUCT_FGET(audio_channel, info_size));
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* make sure the app has a buffer to place the data into */
-	if ((info = STRUCT_FGETP(audio_channel, info)) == NULL) {
-		ATRACE("am_get_chinfo_task() no buffer",
-		    STRUCT_FGETP(audio_channel, info));
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* get the channel number and make sure it's good */
-	ch_number = STRUCT_FGET(audio_channel, ch_number);
-	if (ch_number >= statep->as_max_chs || ch_number < 0) {
-		ATRACE_32("am_get_chinfo_task() bad ch number",
-		    STRUCT_FGET(audio_channel, ch_number));
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* make sure this is a valid AUDIO/AUDIOCTL ch */
-	tchptr = &statep->as_channels[ch_number];
-	mutex_enter(&tchptr->ch_lock);
-	if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    tchptr->ch_info.pid == 0 ||
-	    (tchptr->ch_info.dev_type != AUDIO &&
-	    tchptr->ch_info.dev_type != AUDIOCTL)) {
-		ATRACE("am_get_chinfo_task() bad ch", tchptr);
-		error = EINVAL;
-		mutex_exit(&tchptr->ch_lock);
-		goto nack;
-	}
-	mutex_exit(&tchptr->ch_lock);
-
-	/* okay to update the channel's info */
-	STRUCT_FSET(audio_channel, pid, tchptr->ch_info.pid);
-	STRUCT_FSET(audio_channel, dev_type, tchptr->ch_info.dev_type);
-
-	/*
-	 * We have consistent data so now we can start the copy out,
-	 * beginning with the audio_info_t structure. That's because
-	 * we've got the _IPL32/_LP64 environment. Also, we can't use
-	 * the macros here because of the special address. So we save
-	 * the audio_channel_t data structure for later. And then get
-	 * a new mblk to put the audio_info structure into.
-	 */
-	if ((tmp = allocb(sizeof (audio_info_t), BPRI_HI)) == 0) {
-		error = ENOMEM;
-		goto nack;
-	}
-	state->ais_address2 = (caddr_t)mp->b_cont;
-
-	/*
-	 * Set mp->b_cont = NULL so the mcopyout() below does not free the
-	 * saved message in state->ais_address2 above when it sets
-	 * mp->b_cont = tmp
-	 */
-	mp->b_cont = NULL;
-
-	bcopy(tchptr->ch_info.info, tmp->b_wptr, sizeof (audio_info_t));
-	tmp->b_wptr = tmp->b_rptr + sizeof (audio_info_t);
-
-	state->ais_command = AM_COPY_OUT_MIXCTL_GET_CHINFO;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, sizeof (audio_info_t), (caddr_t)info, tmp);
-
-	ASSERT(mp->b_cont == tmp);
-
-	/* send the copy in request */
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_get_chinfo_task() return", chptr);
-
-	return;
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE("am_get_chinfo_task() returning nack", chptr);
-
-}	/* am_get_chinfo_task() */
-
-/*
- * am_get_mode_task()
- *
- * Description:
- *	This task gets the current mixer mode while the state is stable.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_get_mode_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	int 			error = 0;
-
-	ATRACE("in am_get_mode_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* set STREAMS for copy out of the mode */
-	state->ais_command = AM_COPY_OUT_MIXCTL_MODE;
-	state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-	/* Setup for copyout */
-	if (state->ais_address == NULL) {
-		error = EINVAL;
-		goto done;
-	}
-	mcopyout(mp, state, sizeof (int), state->ais_address, NULL);
-
-	/* put the data in the buffer, but try to reuse it first */
-	if (audio_sup_mblk_alloc(mp, sizeof (int)) == AUDIO_FAILURE) {
-		error = ENOMEM;
-		goto done;
-	} else {
-		*((int *)mp->b_cont->b_rptr) = stpptr->am_pstate->apm_mode;
-		mp->b_cont->b_wptr = mp->b_cont->b_rptr +
-		    sizeof (stpptr->am_pstate->apm_mode);
-		qreply(q, mp);
-	}
-
-done:
-	if (error) {
-		if (cqp->cq_private) {
-			kmem_free(cqp->cq_private, sizeof (audio_i_state_t));
-			cqp->cq_private = NULL;
-		}
-		miocnak(q, mp, 0, error);
-	}
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_get_mode_task() returning", chptr);
-
-}	/* am_get_mode_task() */
-
-/*
- * am_getinfo_task()
- *
- * Description:
- *	This is the task that gets serial access to the info data structure
- *	and copies out the audio_info data structure.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_getinfo_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_info_t		*info = chptr->ch_info.info;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_info_t		*info_out;
-	int			error = 0;
-
-	ATRACE("in am_getinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* set STREAMS for copy out of the audio_info structure */
-	state->ais_command = AM_COPY_OUT_AUDIOINFO;
-	state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-	/* Setup for copyout */
-	if (state->ais_address == NULL) {
-		error = EINVAL;
-		ATRACE("am_getinfo_task() with invalid address", chptr);
-		goto done;
-	}
-	mcopyout(mp, state, sizeof (*info_out), state->ais_address, NULL);
-
-	/* put the data in the buffer, but try to reuse it first */
-	if (audio_sup_mblk_alloc(mp, sizeof (*info_out)) == AUDIO_FAILURE) {
-		error = ENOMEM;
-		ATRACE("am_getinfo_task() failing to alloc mblk", chptr);
-		goto done;
-	}
-
-	info_out = (audio_info_t *)mp->b_cont->b_rptr;
-
-	bcopy(info, info_out, sizeof (*info_out));
-
-	/* update the played sample count */
-	mutex_enter(&chptr->ch_lock);
-	am_fix_info(chptr, info_out);
-	mutex_exit(&chptr->ch_lock);
-
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-done:
-	if (error) {
-		am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-	}
-	am_release_rwlock();
-
-	ATRACE("am_getinfo_task() returning", chptr);
-
-}	/* am_getinfo_task() */
-
-/*
- * am_mixerctl_getinfo_task()
- *
- * Description:
- *	This task gets serial access to the state to copy out the am_control
- *	data structure.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_mixerctl_getinfo_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_info_t		*info_out;
-	am_control_t		*ptr;
-	size_t			size;
-	int			i;
-	int			error = 0;
-	int			max_chs = statep->as_max_chs;
-
-	ATRACE("in am_mixerctl_getinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* we have to check the mode when it's stable, we also check the type */
-	if (stpptr->am_pstate->apm_mode != AM_MIXER_MODE ||
-	    chptr->ch_info.dev_type != AUDIOCTL) {
-		ATRACE_32("am_mixerctl_getinfo_task() bad mode/type",
-		    chptr->ch_info.dev_type);
-		error = EINVAL;
-		goto nack;
-	}
-
-	size = AUDIO_MIXER_CTL_STRUCT_SIZE(max_chs);
-
-	/* set STREAMS for copy out of the audio_info structure */
-	state->ais_command = AM_COPY_OUT_MIXCTLINFO;
-	state->ais_address = (caddr_t)(*(caddr_t *)mp->b_cont->b_rptr);
-
-	/* Setup for copyout */
-	if (state->ais_address == NULL) {
-		ATRACE("am_mixerctl_getinfo_task() bad address from user",
-		    NULL);
-		error = EINVAL;
-		goto nack;
-	}
-	mcopyout(mp, state, size, state->ais_address, NULL);
-
-	/* put the data in the buffer, but try to reuse it first */
-	if (audio_sup_mblk_alloc(mp, size) == AUDIO_FAILURE) {
-		ATRACE("am_mixerctl_getinfo_task() can't get memory", 0);
-		error = ENOMEM;
-		goto nack;
-	}
-
-	ptr = (am_control_t *)mp->b_cont->b_rptr;
-	info_out = &ptr->dev_info;
-
-	/*
-	 * We have to assemble this one by pieces. First we take
-	 * care of the hardware state, then extended hardware state.
-	 */
-
-	bcopy(apm_infop->apm_ad_state, info_out, sizeof (audio_info_t));
-
-	/* update the played sample count */
-	mutex_enter(&chptr->ch_lock);
-	am_fix_info(chptr, info_out);
-	mutex_exit(&chptr->ch_lock);
-
-	/* now get the channel information */
-	mutex_enter(&statep->as_lock);		/* freeze ch state */
-
-	for (i = 0, tchptr = &statep->as_channels[0]; i < max_chs;
-	    i++, tchptr++) {
-		mutex_enter(&tchptr->ch_lock);
-		if (tchptr->ch_info.pid) {
-			ptr->ch_open[i] = 1;
-		} else {
-			ptr->ch_open[i] = 0;
-		}
-		mutex_exit(&tchptr->ch_lock);
-	}
-
-	mutex_exit(&statep->as_lock);
-
-	/* send the copy out request */
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_mixerctl_getinfo_task() returning", chptr);
-
-	return;
-
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE("am_mixerctl_getinfo_task() returning nack", chptr);
-
-}	/* am_mixerctl_getinfo_task() */
-
-/*
- * am_mixerctl_setinfo_task()
- *
- * Description:
- *	This task gets serial access to the state to set it based on the
- *	am_control data structure. Can only set a few global things. The
- *	ch_open argument of the am_control data structure since this is
- *	read only.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_mixerctl_setinfo_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_info_t		ninfo;
-	audio_ch_t		ch;
-	audio_info_t		*tinfo;
-	am_control_t		*new;
-	int			error = 0;
-
-	ATRACE("in am_mixerctl_setinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* we have to check the mode when it's stable */
-	if (stpptr->am_pstate->apm_mode != AM_MIXER_MODE) {
-		ATRACE_32("am_mixerctl_setinfo_task() bad mode",
-		    chptr->ch_info.dev_type);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* get a pointer to the user supplied structure */
-	new = (am_control_t *)mp->b_cont->b_rptr;
-	tinfo = &new->dev_info;
-
-	/* we can only modify a few things so make sure that's all we touch */
-	AUDIO_INIT(&ninfo, sizeof (ninfo));
-	ninfo.play.gain = tinfo->play.gain;
-	ninfo.play.balance = tinfo->play.balance;
-	ninfo.play.port = tinfo->play.port;
-	ninfo.play.pause = tinfo->play.pause;
-	ninfo.record.gain = tinfo->record.gain;
-	ninfo.record.balance = tinfo->record.balance;
-	ninfo.record.port = tinfo->record.port;
-	ninfo.record.pause = tinfo->record.pause;
-	ninfo.monitor_gain = tinfo->monitor_gain;
-	ninfo.output_muted = tinfo->output_muted;
-
-	/* we always create a pseudo channel that points to the h/w */
-	bcopy(chptr, &ch, sizeof (*chptr));
-	ch.ch_info.info = hw_info;
-
-	/* too ugly to check here, so send to a utility routine */
-	ATRACE("am_mixerctl_setinfo_task() calling am_audio_set_info()", chptr);
-	if (am_audio_set_info(&ch, &ninfo, &ninfo) == AUDIO_FAILURE) {
-		ATRACE("am_mixerctl_setinfo_task() am_audio_set_info() failed",
-		    chptr);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* since there wasn't an error we succeeded, so return struct */
-	tinfo->play.gain = ninfo.play.gain;
-	tinfo->play.balance = ninfo.play.balance;
-	tinfo->play.port = ninfo.play.port;
-	tinfo->play.pause = ninfo.play.pause;
-	tinfo->record.gain = ninfo.record.gain;
-	tinfo->record.balance = ninfo.record.balance;
-	tinfo->record.port = ninfo.record.port;
-	tinfo->record.pause = ninfo.record.pause;
-	tinfo->monitor_gain = ninfo.monitor_gain;
-	tinfo->output_muted = ninfo.output_muted;
-
-	/*
-	 * Since there wasn't an error we were successful, now return
-	 * the updated structure.
-	 */
-	state->ais_command = AM_COPY_OUT_MIXCTLINFO;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, sizeof (*tinfo), state->ais_address, NULL);
-	ASSERT(mp->b_cont->b_wptr == (mp->b_cont->b_rptr + sizeof (*tinfo)));
-
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_mixerctl_setinfo_task() returning", chptr);
-
-	return;
-
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE("am_mixerctl_setinfo_task() returning error", chptr);
-
-}	/* am_mixerctl_setinfo_task() */
-
-/*
- * am_multiple_open_task()
- *
- * Description:
- *	The second part of the AUDIO_MIXER_MULTIPLE_OPEN ioctl(). We check
- *	the state here so we can rely on the state.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_multiple_open_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	int			error = 0;
-
-	ATRACE("in am_multiple_open_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/*
-	 * Don't allow this ioctl() if not in MIXER mode. We have to do this
-	 * check in the task because that's the only way we can rely on the
-	 * state.
-	 */
-	if (stpptr->am_pstate->apm_mode == AM_COMPAT_MODE) {
-		ATRACE_32("am_multiple_open_task() bad mode",
-		    stpptr->am_pstate->apm_mode);
-		error = EINVAL;
-		goto done;
-	}
-
-	/* just set the mode without checking what it is */
-	mutex_enter(&chptr->ch_lock);
-	chpptr->acp_flags |= AM_CHNL_MULTI_OPEN;
-	ATRACE_32("am_multiple_open_task() flags", chpptr->acp_flags);
-	mutex_exit(&chptr->ch_lock);
-
-	/* wake up any channels waiting on multiple open()s */
-	mutex_enter(&chptr->ch_statep->as_lock);
-	cv_broadcast(&chptr->ch_statep->as_cv);
-	mutex_exit(&chptr->ch_statep->as_lock);
-
-done:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE_32("am_multiple_open_task() done", error);
-
-}	/* am_multiple_open_task() */
-
-/*
- * am_sample_rate_task()
- *
- * Description:
- *	Now that we will know how many sample rates to return we can get
- *	them. We do this in a task because we need to know the mixer mode.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_sample_rate_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_sample_rates_t	*new;
-	am_ad_sample_rates_t	*src;
-	size_t			size;
-	int			error = 0;
-	int			i;
-	int			mode = stpptr->am_pstate->apm_mode;
-	int			num;
-
-	ATRACE("in am_sample_rate_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* get a pointer to the user supplied structure */
-	new = (am_sample_rates_t *)mp->b_cont->b_rptr;
-	/* new->flags is random data, so clear */
-	new->flags = 0;
-
-	/* make sure the number of array elements is sane */
-	if (new->num_samp_rates <= 0) {
-		ATRACE_32("am_sample_rate_task() AM_COPY_IN_SAMP_RATES2 "
-		    "bad num_samp_rates", new->num_samp_rates);
-		error = EINVAL;
-		goto nack;
-	}
-
-	size = AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(new->num_samp_rates);
-
-	ATRACE_32("am_sample_rate_task() AM_COPY_IN_SAMP_RATES2 type",
-	    new->type);
-	if (new->type == AUDIO_PLAY &&
-	    (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY)) {
-		if (mode == AM_MIXER_MODE) {
-			src = &ad_infop->ad_play.ad_mixer_srs;
-		} else {
-			ASSERT(mode == AM_COMPAT_MODE);
-			src = &ad_infop->ad_play.ad_compat_srs;
-		}
-	} else if (new->type == AUDIO_RECORD &&
-	    (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD)) {
-		if (mode == AM_MIXER_MODE) {
-			src = &ad_infop->ad_record.ad_mixer_srs;
-		} else {
-			ASSERT(mode == AM_COMPAT_MODE);
-			src = &ad_infop->ad_record.ad_compat_srs;
-		}
-	} else {
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* figure out how many sample rates we have */
-	for (num = 0; src->ad_srs[num] != 0; num++);
-
-	/* we don't copy more sample rates than we have */
-	if (num < new->num_samp_rates) {
-		new->num_samp_rates = num;
-	}
-
-	/* we reuse the buffer we got from user space */
-	for (i = 0; i < new->num_samp_rates; i++) {
-		/* get sample rate for array elements */
-		if (src->ad_srs[i] == 0) {
-			/* at the end of sample rates */
-			break;
-		}
-		new->samp_rates[i] = src->ad_srs[i];
-	}
-
-	/* let the app know there are more */
-	if (num > new->num_samp_rates) {
-		new->num_samp_rates = num;
-	}
-
-	/* type remains the same, but update others */
-	if (src->ad_limits & MIXER_SRS_FLAG_SR_LIMITS) {
-		new->flags = MIXER_SR_LIMITS;
-	}
-
-	/* ready to send the filled in structure back */
-	state->ais_command = AM_COPY_OUT_SAMP_RATES;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, size, state->ais_address, NULL);
-	ASSERT(mp->b_cont->b_wptr == (mp->b_cont->b_rptr + size));
-
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE_32("am_sample_rate_task() done", error);
-
-	return;
-
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE_32("am_sample_rate_task() returning error", error);
-
-}	/* am_sample_rate_task() */
-
-/*
- * am_set_chinfo_task()
- *
- * Description:
- *	Called by the task queue to set the channel's state. We've already
- *	verified the size and channel are okay, as well as there is a buffer.
- *	We have the audio_info structure, in mp->b_cont,  so new we can set
- *	the state.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_set_chinfo_task(void *arg)
-{
-	STRUCT_HANDLE(audio_channel, audio_channel);
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_info_t		*tinfo = (audio_info_t *)mp->b_cont->b_rptr;
-	audio_channel_t		*ch;
-	struct iocblk		*iocbp = (struct iocblk *)mp->b_rptr;
-	int			ch_number;
-	int			error = 0;
-
-	ATRACE("in am_set_chinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* we have to check the mode when it's stable */
-	if (stpptr->am_pstate->apm_mode != AM_MIXER_MODE) {
-		ATRACE_32("am_set_chinfo_task() bad mode",
-		    chptr->ch_info.dev_type);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* get ready to do the model conversion */
-	ch = (audio_channel_t *)((mblk_t *)state->ais_address2)->b_rptr;
-	STRUCT_SET_HANDLE(audio_channel, iocbp->ioc_flag, ch);
-
-	/*
-	 * We check things again because since the last check it could have
-	 * changed on us.
-	 */
-	if (STRUCT_FGET(audio_channel, info_size) != sizeof (audio_info_t)) {
-		ATRACE_32("am_set_chinfo_task() bad size",
-		    STRUCT_FGET(audio_channel, info_size));
-		error = EINVAL;
-		goto nack;
-	}
-	if (STRUCT_FGETP(audio_channel, info) == NULL) {
-		ATRACE("am_set_chinfo_task() no buffer",
-		    STRUCT_FGETP(audio_channel, info));
-		error = EINVAL;
-		goto nack;
-	}
-	ch_number = STRUCT_FGET(audio_channel, ch_number);
-	if (ch_number >= statep->as_max_chs || ch_number < 0) {
-		ATRACE_32("am_set_chinfo_task() bad ch number",
-		    STRUCT_FGET(audio_channel, ch_number));
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* make sure this is a valid AUDIO/AUDIOCTL ch */
-	tchptr = &statep->as_channels[ch_number];
-	mutex_enter(&tchptr->ch_lock);
-	if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-	    tchptr->ch_info.pid == 0 ||
-	    (tchptr->ch_info.dev_type != AUDIO &&
-	    tchptr->ch_info.dev_type != AUDIOCTL)) {
-		mutex_exit(&tchptr->ch_lock);
-		ATRACE("am_set_chinfo_task() bad ch", tchptr);
-		error = EINVAL;
-		goto nack;
-	}
-	mutex_exit(&tchptr->ch_lock);
-
-	/* because the channels are stable we update the audio_channel struct */
-	STRUCT_FSET(audio_channel, pid, tchptr->ch_info.pid);
-	STRUCT_FSET(audio_channel, dev_type, tchptr->ch_info.dev_type);
-
-	/* too ugly to check here, so send to a utility routine */
-	ATRACE("am_set_chinfo_task() calling am_audio_set_info()", chptr);
-	if (am_audio_set_info(tchptr, tinfo, tinfo) == AUDIO_FAILURE) {
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* update the played sample count */
-	mutex_enter(&chptr->ch_lock);
-	am_fix_info(chptr, tinfo);
-	mutex_exit(&chptr->ch_lock);
-
-	/*
-	 * Since there wasn't an error we were successful, now return
-	 * the updated structure.
-	 */
-	state->ais_command = AM_COPY_OUT_MIXCTL_GET_CHINFO;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, sizeof (*tinfo), state->ais_address, NULL);
-	ASSERT(mp->b_cont->b_wptr == (mp->b_cont->b_rptr + sizeof (*tinfo)));
-
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_set_chinfo_task() returning", chptr);
-
-	return;
-
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE("am_set_chinfo_task() returning error", chptr);
-
-}	/* am_set_chinfo_task() */
-
-/*
- * am_set_mode_task()
- *
- * Description:
- *	Switch modes. We have to shut down play and record first, then change
- *	modes, and finally restart. Since we are a low priority thread we
- *	can sleep!
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_set_mode_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_state_t		*statep = chptr->ch_statep;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_ad_info_t		*ad_infop = apm_infop->apm_ad_infop;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	audio_ch_t		*pchptr = NULL;
-	audio_ch_t		*rchptr = NULL;
-	audio_ch_t		*tchptr;
-	am_ch_private_t		*tchpptr;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	audio_info_t		*new_pinfo = NULL;
-	audio_info_t		*new_rinfo = NULL;
-	audio_info_t		old_hw_info;
-	audio_info_t		*tinfo;
-	int			doread;
-	int			dowrite;
-	int			error = 0;
-	int			i;
-	int			max_chs = statep->as_max_chs;
-	int			mode = stpptr->am_pstate->apm_mode;
-	int			new_mode;
-	int			ppid = 0;
-	int			rpid = 0;
-#ifdef DEBUG
-	int			pcount = 0;
-	int			rcount = 0;
-#endif
-
-	ATRACE("in am_set_mode_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/* get the new_mode and make sure it's good */
-	new_mode = *((int *)mp->b_cont->b_rptr);
-	ATRACE_32("am_set_mode_task() new mode", new_mode);
-	if (new_mode != AM_MIXER_MODE && new_mode != AM_COMPAT_MODE) {
-		ATRACE_32("am_set_mode_task() bad mode", new_mode);
-		error = EINVAL;
-		goto done;
-	}
-
-	/* make sure we aren't going into the same mode */
-	if (mode == new_mode) {
-		ATRACE_32("am_set_mode_task() same mode", new_mode);
-		goto done;
-	}
-
-	/* figure out the direction */
-	doread = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD;
-	dowrite = ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY;
-
-	/* we allocate this memory while it's easy to back out */
-	if (new_mode == AM_MIXER_MODE &&
-	    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			new_pinfo = kmem_alloc(sizeof (*new_pinfo), KM_SLEEP);
-			new_rinfo = kmem_alloc(sizeof (*new_rinfo), KM_SLEEP);
-	}
-
-	/* get the AUDIO apm_info pointer, we've got one for AUDIOCTL */
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		if (new_pinfo) {
-			kmem_free(new_pinfo, sizeof (*new_pinfo));
-		}
-		if (new_rinfo) {
-			kmem_free(new_rinfo, sizeof (*new_rinfo));
-		}
-		error = EIO;
-		goto done;
-	}
-
-	/*
-	 * Make sure we can go to COMPAT mode while we're locked. By
-	 * definition if we are going to MIXER mode we can't have more
-	 * than one in and one out channel allocated.
-	 */
-	mutex_enter(&statep->as_lock);
-	if (new_mode == AM_COMPAT_MODE && (stpptr->am_in_chs > 1 ||
-	    stpptr->am_out_chs > 1)) {
-		ATRACE("am_set_mode_task() AM_COPY_IN_MIXCTL_MODE busy", chptr);
-		ASSERT(new_pinfo == NULL);
-		ASSERT(new_rinfo == NULL);
-
-		error = EBUSY;
-		mutex_exit(&statep->as_lock);
-		goto done;
-	} else {
-		ASSERT(stpptr->am_in_chs <= 1);
-		ASSERT(stpptr->am_out_chs <= 1);
-	}
-	mutex_exit(&statep->as_lock);
-
-	/* once playing/recording has stopped we can clear this flag */
-	mutex_enter(&apm_infop->apm_lock);
-	stpptr->am_flags |= AM_PRIV_SW_MODES;
-	mutex_exit(&apm_infop->apm_lock);
-
-	/* how we do the switch is different based on the device */
-	if (ad_infop->ad_codec_type == AM_MS_CODEC) {
-		/* find the reading and writing channels */
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-
-			mutex_enter(&tchptr->ch_lock);
-
-			/* skip non-AUDIO and unallocated channels */
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr->ch_info.dev_type != AUDIO ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* is this channel playing?, recording? */
-			tchpptr = tchptr->ch_private;
-			ATRACE("am_set_mode_task() found ch flags",
-			    chpptr->acp_flags);
-			if (tchpptr->acp_writing) {
-				ATRACE_32("am_set_mode_task() MS found play ch",
-				    tchptr->ch_info.ch_number);
-				ASSERT(pchptr == NULL);
-				ASSERT(dowrite);
-				pchptr = tchptr;
-#ifdef DEBUG
-				pcount++;
-#endif
-			}
-			if (tchpptr->acp_reading) {
-				ATRACE_32(
-				    "am_set_mode_task() MS found record ch",
-				    tchptr->ch_info.ch_number);
-				ASSERT(rchptr == 0);
-				ASSERT(doread);
-				rchptr = tchptr;
-#ifdef DEBUG
-				rcount++;
-#endif
-			}
-			mutex_exit(&tchptr->ch_lock);
-
-			/* are we done finding active channels? */
-			if ((pchptr || !dowrite) && (rchptr || !doread)) {
-				break;
-			}
-		}
-		ASSERT(pcount <= 1);
-		ASSERT(rcount <= 1);
-
-		/* pause playing & recording, so the ISR isn't called */
-		if (dowrite) {
-			ATRACE("am_set_mode_task() pause play", chptr);
-			tinfo = pchptr->ch_info.info;
-			am_ad_pause_play(statep, stpptr, ad_infop,
-			    pchptr->ch_info.ch_number);
-			tinfo->play.active = 0;
-			hw_info->play.active = 0;
-			tinfo->play.pause = 1;
-		}
-		if (doread) {
-			ATRACE("am_set_mode_task() stop record", chptr);
-			tinfo = rchptr->ch_info.info;
-			am_ad_stop_record(statep, stpptr, ad_infop,
-			    rchptr->ch_info.ch_number);
-			tinfo->record.active = 0;
-			hw_info->record.active = 0;
-			tinfo->record.pause = 0;
-		}
-
-		/*
-		 * Multi-stream Codecs already use the virtual channel
-		 * configuration to set the hardware, so this is a trivial
-		 * change to make. Everything is halted so we don't need
-		 * to lock this.
-		 */
-		if (new_mode == AM_COMPAT_MODE) {
-			stpptr->am_pstate->apm_mode = AM_COMPAT_MODE;
-		} else {
-			ASSERT(new_mode == AM_MIXER_MODE);
-			stpptr->am_pstate->apm_mode = AM_MIXER_MODE;
-		}
-
-		ATRACE("am_set_mode_task() AM_MS_CODEC switch done", ad_infop);
-
-	} else {
-		ASSERT(ad_infop->ad_codec_type == AM_TRAD_CODEC);
-
-		/* wait for playing to end */
-		mutex_enter(&stpptr->am_mode_lock);
-
-		while (dowrite && hw_info->play.active) {
-			ATRACE_32("am_set_mode_task() wait to stop playing",
-			    hw_info->play.active);
-			if (cv_wait_sig(&stpptr->am_mode_cv,
-			    &stpptr->am_mode_lock) <= 0) {
-
-				ATRACE("am_set_mode_task() signal interrupt",
-				    hw_info->play.active);
-
-				mutex_exit(&stpptr->am_mode_lock);
-
-				/* we aren't switching modes any longer */
-				mutex_enter(&apm_infop->apm_lock);
-				stpptr->am_flags &= ~AM_PRIV_SW_MODES;
-				mutex_exit(&apm_infop->apm_lock);
-
-				/* we are bailing, so we need to restart */
-				am_restart(statep, hw_info);
-
-				if (new_mode == AM_MIXER_MODE) {
-					if (new_pinfo) {
-						kmem_free(new_pinfo,
-						    sizeof (*new_pinfo));
-					}
-					if (new_rinfo) {
-						kmem_free(new_rinfo,
-						    sizeof (*new_rinfo));
-					}
-				}
-
-				error = EINTR;
-				goto done;
-			}
-			ATRACE(
-			    "am_set_mode_task() signal returned normally", 0);
-		}
-		mutex_exit(&stpptr->am_mode_lock);
-
-		/*
-		 * Now we shutdown the record channel, if active.
-		 * We have to lock to make this call.
-		 */
-		if (doread && hw_info->record.active) {
-			am_ad_stop_record(statep, stpptr, ad_infop,
-			    chptr->ch_info.ch_number);
-			hw_info->record.active = 0;
-		}
-
-		/* wait as long as possible to save the old state for later */
-		bcopy(hw_info, &old_hw_info, sizeof (*hw_info));
-
-		/* find the play and record channels */
-		ASSERT(pcount == 0);
-		ASSERT(rcount == 0);
-		ASSERT(pchptr == 0);
-		ASSERT(rchptr == 0);
-		for (i = 0, tchptr = &statep->as_channels[0];
-			    i < max_chs; i++, tchptr++) {
-			/* skip non-AUDIO and unallocated channels */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr->ch_info.dev_type != AUDIO ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* is this channel playing?, recording? */
-			tchpptr = tchptr->ch_private;
-			ATRACE("am_set_mode_task() found ch flags",
-				    tchpptr->acp_flags);
-			if (dowrite && tchpptr->acp_writing) {
-				ATRACE_32("am_set_mode_task() T found play ch",
-				    tchptr->ch_info.ch_number);
-				/*
-				 * Disable the queue so that am_wsvc() won't
-				 * process any more data messages.
-				 */
-				noenable(tchptr->ch_qptr);
-
-				pchptr = tchptr;
-#ifdef DEBUG
-				pcount++;
-#endif
-			}
-			if (doread && tchpptr->acp_reading) {
-				ATRACE_32(
-				    "am_set_mode_task() T found record ch",
-				    tchptr->ch_info.ch_number);
-				rchptr = tchptr;
-#ifdef DEBUG
-				rcount++;
-#endif
-			}
-
-			mutex_exit(&tchptr->ch_lock);
-
-			/* are we done finding active channels? */
-			if ((pchptr || !dowrite) && (rchptr || !doread)) {
-				break;
-			}
-		}
-		ASSERT(pcount <= 1);
-		ASSERT(rcount <= 1);
-
-		/*
-		 * We stop playing because we have to force it to restart from
-		 * scratch, which means flushing the DMA engine. Otherwise
-		 * there's old data with the wrong format. This a problem only
-		 * when we are paused.
-		 */
-		if (pchptr &&
-		    ((audio_info_t *)pchptr->ch_info.info)->play.pause) {
-			am_ad_stop_play(statep, stpptr, ad_infop,
-			    AM_SET_CONFIG_BOARD);
-		}
-
-		if (new_mode == AM_MIXER_MODE) {
-			if (am_set_mixer_mode(chptr, ad_infop, stpptr,
-			    pchptr, rchptr) == AUDIO_FAILURE) {
-				/* we aren't switching modes any longer */
-				mutex_enter(&apm_infop->apm_lock);
-				stpptr->am_flags &= ~AM_PRIV_SW_MODES;
-				mutex_exit(&apm_infop->apm_lock);
-
-				/* we are bailing, so we need to restart */
-				am_restart(statep, hw_info);
-
-				if (new_pinfo) {
-					kmem_free(new_pinfo,
-					    sizeof (*new_pinfo));
-				}
-				if (new_rinfo) {
-					kmem_free(new_rinfo,
-					    sizeof (*new_rinfo));
-				}
-
-				error = EIO;
-				goto done;
-			}
-		} else {
-			if (am_set_compat_mode(chptr, ad_infop,
-			    pchptr, rchptr) == AUDIO_FAILURE) {
-				/* we aren't switching modes any longer */
-				mutex_enter(&apm_infop->apm_lock);
-				stpptr->am_flags &= ~AM_PRIV_SW_MODES;
-				mutex_exit(&apm_infop->apm_lock);
-
-				/* we are bailing, so we need to restart */
-				am_restart(statep, hw_info);
-
-				ASSERT(new_pinfo == NULL);
-				ASSERT(new_rinfo == NULL);
-
-				error = EIO;
-				goto done;
-			}
-		}
-	}
-	/* save the mode in persistent memory */
-	stpptr->am_pstate->apm_mode = new_mode;
-
-	/*
-	 * CAUTION: From here on out we cannot fail. We've changed modes
-	 *	and to fail would require setting it back. It is better
-	 *	to have a gap in audio then to go back at this point.
-	 */
-
-	/* we're in a new mode now, so fix the play/rec info pointers */
-	hw_info = &stpptr->am_hw_info;
-	if (new_mode == AM_MIXER_MODE) {
-		int	tpid;
-		/* rebuild info structures and get pids */
-		if (pchptr && pchptr == rchptr) {
-			/* this is play/record channel */
-			mutex_enter(&pchptr->ch_lock);
-			ppid = pchptr->ch_info.pid;
-			rpid = ppid;
-
-			tinfo = new_pinfo;
-			pchptr->ch_info.info = tinfo;
-			mutex_exit(&pchptr->ch_lock);
-
-			if (new_rinfo) {
-				kmem_free(new_rinfo, sizeof (*new_rinfo));
-			}
-			new_rinfo = NULL;
-
-			/* copy in the old play state */
-			bcopy(&old_hw_info.play, &tinfo->play,
-			    (2 * sizeof (old_hw_info.play)));
-
-			/* copy the current dev state */
-			tinfo->monitor_gain = hw_info->monitor_gain;
-			tinfo->output_muted = hw_info->output_muted;
-			tinfo->hw_features = hw_info->hw_features;
-			tinfo->sw_features = hw_info->sw_features;
-			tinfo->sw_features_enabled =
-			    hw_info->sw_features_enabled;
-			tinfo->ref_cnt = 1;
-		} else {
-			/* play or record channels */
-			ASSERT(pchptr != rchptr ||
-			    (pchptr == 0 && rchptr == 0));
-			if (pchptr) {
-				mutex_enter(&pchptr->ch_lock);
-				ppid = pchptr->ch_info.pid;
-
-				tinfo = new_pinfo;
-				AUDIO_INIT(tinfo, sizeof (*tinfo));
-				pchptr->ch_info.info = tinfo;
-
-				/* copy in the old play state */
-				bcopy(&old_hw_info.play, &tinfo->play,
-				    sizeof (tinfo->play));
-				/* copy the current dev state */
-				tinfo->monitor_gain = hw_info->monitor_gain;
-				tinfo->output_muted = hw_info->output_muted;
-				tinfo->hw_features = hw_info->hw_features;
-				tinfo->sw_features = hw_info->sw_features;
-				tinfo->sw_features_enabled =
-					hw_info->sw_features_enabled;
-				tinfo->ref_cnt = 1;
-				mutex_exit(&pchptr->ch_lock);
-			} else {
-				if (new_pinfo) {
-					kmem_free(new_pinfo,
-						sizeof (*new_pinfo));
-				}
-				new_pinfo = NULL;
-			}
-			if (rchptr) {
-				mutex_enter(&rchptr->ch_lock);
-				rpid = rchptr->ch_info.pid;
-
-				tinfo = new_rinfo;
-				AUDIO_INIT(tinfo, sizeof (*tinfo));
-				rchptr->ch_info.info = tinfo;
-
-				/* copy in the old record state */
-				bcopy(&old_hw_info.record, &tinfo->record,
-				    sizeof (old_hw_info.record));
-				/* copy the current dev state */
-				tinfo->monitor_gain = hw_info->monitor_gain;
-				tinfo->output_muted = hw_info->output_muted;
-				tinfo->hw_features = hw_info->hw_features;
-				tinfo->sw_features = hw_info->sw_features;
-				tinfo->sw_features_enabled =
-					hw_info->sw_features_enabled;
-				tinfo->ref_cnt = 1;
-				mutex_exit(&rchptr->ch_lock);
-			} else {
-				if (new_rinfo) {
-					kmem_free(new_rinfo,
-						sizeof (*new_rinfo));
-				}
-				new_rinfo = NULL;
-			}
-		}
-
-		/* start over with the reference count */
-		hw_info->ref_cnt = 1;
-
-		/* find AUDIOCTL channels to assoc. with AUDIO chs */
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-
-			/* skip if not AUDIOCTL or allocated chs */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr->ch_info.dev_type != AUDIOCTL ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			ASSERT(hw_info == tchptr->ch_info.info);
-
-			tpid = tchptr->ch_info.pid;
-
-			if (ppid && tpid == ppid) {
-				tchptr->ch_info.info = new_pinfo;
-				new_pinfo->ref_cnt++;
-			} else if (rpid && tpid == rpid) {
-				tchptr->ch_info.info = new_rinfo;
-				new_rinfo->ref_cnt++;
-			} else {
-				tchptr->ch_info.info = hw_info;
-				hw_info->ref_cnt++;
-			}
-			mutex_exit(&tchptr->ch_lock);
-		}
-
-		/* now we need to re-initialize the src structures */
-		if (pchptr) {
-			ATRACE("am_set_mode_task() calling am_fix_play_pause",
-				pchptr->ch_private);
-
-			am_fix_play_pause(pchptr);
-		}
-	} else {
-		/* start over with the reference count */
-		hw_info->ref_cnt = 1;
-
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-
-			/*
-			 * Skip if ! AUDIO, AUDIOCTL or allocated channels.
-			 *
-			 * It is possible that new audio will arrive in
-			 * am_wsvc(), thus we need to make sure we have this
-			 * lock and hold it if we have to manipulate the
-			 * channel's info structure.
-			 */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    (tchptr->ch_info.dev_type != AUDIO &&
-			    tchptr->ch_info.dev_type != AUDIOCTL) ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* see if we need to free the info structure */
-			tinfo = tchptr->ch_info.info;
-			if (tinfo != hw_info) {
-				/* not set to hw */
-				if (tinfo->ref_cnt == 1) {
-					/* not set to hw, and only ref so clr */
-					kmem_free(tinfo, sizeof (*tinfo));
-				} else {
-					/* someone else has a link to it */
-					tinfo->ref_cnt--;
-				}
-			}
-			/* now set to hardware */
-			tchptr->ch_info.info = hw_info;
-			hw_info->ref_cnt++;
-			mutex_exit(&tchptr->ch_lock);
-		}
-	}
-
-	/* we don't need this flag anymore, re-enabling ioctls */
-	mutex_enter(&apm_infop->apm_lock);
-	stpptr->am_flags &= ~AM_PRIV_SW_MODES;
-	mutex_exit(&apm_infop->apm_lock);
-
-	/* we're in the new mode, so restart I/O */
-	am_restart(statep, hw_info);
-
-	mutex_enter(&statep->as_lock);
-	/* if we have blocked processes they should unblock */
-	if (new_mode == AM_MIXER_MODE) {
-		cv_broadcast(&statep->as_cv);
-	}
-	mutex_exit(&statep->as_lock);
-
-	ATRACE_32("am_set_mode_task() done, new mode", new_mode);
-
-done:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	am_send_signal(statep, stpptr);
-
-	ATRACE_32("am_set_mode_task() returning ack", error);
-
-}	/* am_set_mode_task() */
-
-/*
- * am_setinfo_task()
- *
- * Description:
- *	The third part of the AUDIO_SETINFO ioctl(). This is the task that
- *	gets serial access to the info data structure and hardware.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_setinfo_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	audio_info_t		*tinfo = (audio_info_t *)mp->b_cont->b_rptr;
-	int			error = 0;
-
-	ATRACE("in am_setinfo_task() arg", arg);
-
-	am_enter_rwlock();
-
-	if (cqp->cq_cmd != AUDIO_SETINFO) {
-		ATRACE_32("am_wioctl() AUDIO_SETINFO bad command", cqp->cq_cmd);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/*
-	 * Don't allow this ioctl() when on an audioctl channel when multiple
-	 * audio channel open()s are allowed. We have to do this in the task
-	 * because that's the only way we can rely on the state.
-	 */
-	mutex_enter(&chptr->ch_lock);
-	if (chptr->ch_info.dev_type == AUDIOCTL &&
-	    (chpptr->acp_flags & AM_CHNL_MULTI_OPEN)) {
-		ATRACE_32("am_wioctl() AUDIO_SETINFO bad type",
-		    chptr->ch_info.dev_type);
-		mutex_exit(&chptr->ch_lock);
-		error = EINVAL;
-		goto nack;
-	}
-	mutex_exit(&chptr->ch_lock);
-
-	/* too ugly to check here, so send to a utility routine */
-	ATRACE("am_setinfo_task() calling am_audio_set_info()", chptr);
-	if (am_audio_set_info(chptr, tinfo, tinfo) == AUDIO_FAILURE) {
-		ATRACE_32("am_wioctl() AUDIO_SETINFO setinfo failed command",
-		    cqp->cq_cmd);
-		error = EINVAL;
-		goto nack;
-	}
-
-	/* update the played sample count */
-	mutex_enter(&chptr->ch_lock);
-	am_fix_info(chptr, tinfo);
-	mutex_exit(&chptr->ch_lock);
-
-	/*
-	 * Since there wasn't an error we were successful, now return
-	 * the updated structure.
-	 */
-	state->ais_command = AM_COPY_OUT_AUDIOINFO2;
-
-	/* Setup for copyout */
-	ASSERT(state->ais_address != NULL);
-	mcopyout(mp, state, sizeof (*tinfo), state->ais_address, NULL);
-	ASSERT(mp->b_cont->b_wptr == (mp->b_cont->b_rptr + sizeof (*tinfo)));
-
-	qreply(q, mp);
-
-	kmem_free(arg, sizeof (am_ioctl_args_t));
-
-	am_exit_task(chptr);
-
-	am_release_rwlock();
-
-	ATRACE("am_setinfo_task() returning", chptr);
-
-	return;
-
-nack:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE_32("am_setinfo_task() returning nack", error);
-
-}	/* am_setinfo_task() */
-
-/*
- * am_single_open_task()
- *
- * Description:
- *	The second part of the AUDIO_MIXER_SINGLE_OPEN ioctl(). We check
- *	the state here so we can rely on the state.
- *
- * Arguments:
- *	void		*arg	Argument structure
- *
- * Returns:
- *	void
- */
-static void
-am_single_open_task(void *arg)
-{
-	queue_t			*q = ((am_ioctl_args_t *)arg)->aia_q;
-	mblk_t			*mp = ((am_ioctl_args_t *)arg)->aia_mp;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	struct copyreq		*cqp = (struct copyreq *)mp->b_rptr;
-	audio_i_state_t		*state = (audio_i_state_t *)cqp->cq_private;
-	am_ch_private_t		*tchpptr;
-	pid_t			tpid;
-	int			error = 0;
-	int			i;
-	int			max_chs = statep->as_max_chs;
-	int			num_rd = 0;
-	int			num_wr = 0;
-
-	ATRACE("in am_single_open_task() arg", arg);
-
-	am_enter_rwlock();
-
-	/*
-	 * Don't allow this ioctl() if not in MIXER mode. We have to do this
-	 * check in the task because that's the only way we can rely on the
-	 * state.
-	 */
-	if (stpptr->am_pstate->apm_mode == AM_COMPAT_MODE) {
-		ATRACE_32("am_single_open_task() bad mode",
-		    stpptr->am_pstate->apm_mode);
-		error = EINVAL;
-		goto done;
-	}
-
-	/* see if there are multiple open()s already */
-	tpid = chptr->ch_info.pid;
-
-	/* we need to freeze channel allocation */
-	mutex_enter(&statep->as_lock);
-	for (i = 0, tchptr = &statep->as_channels[0]; i < max_chs;
-	    i++, tchptr++) {
-		/* ignore different processes */
-		mutex_enter(&tchptr->ch_lock);
-		if (tchptr->ch_info.pid != tpid) {
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-		tchpptr = tchptr->ch_private;
-		if (tchpptr->acp_reading) {
-			num_rd++;
-		}
-		if (tchpptr->acp_writing) {
-			num_wr++;
-		}
-		mutex_exit(&tchptr->ch_lock);
-	}
-	mutex_exit(&statep->as_lock);
-
-	/* we change back only if at most 1 read or write ch. open */
-	if (num_rd > 1 || num_wr > 1) {
-		/* too many channels open, so we have to fail */
-		error = EIO;
-	} else {
-		/* if we get here we know there is only 1 ch, ours */
-		mutex_enter(&chptr->ch_lock);
-		chpptr->acp_flags &= ~AM_CHNL_MULTI_OPEN;
-		mutex_exit(&chptr->ch_lock);
-
-		ATRACE("am_single_open_task() flags", chpptr->acp_flags);
-	}
-
-done:
-	am_mixer_task_acknack(state, chptr, q, mp, arg, error);
-
-	am_release_rwlock();
-
-	ATRACE_32("am_single_open_task() done", error);
-
-}	/* am_single_open_task() */
diff --git a/usr/src/uts/common/io/audio/sada/mixer/am_main.c b/usr/src/uts/common/io/audio/sada/mixer/am_main.c
deleted file mode 100644
index 856d21bef0..0000000000
--- a/usr/src/uts/common/io/audio/sada/mixer/am_main.c
+++ /dev/null
@@ -1,4273 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-/*
- * Mixer Audio Personality Module (mixer)
- *
- * This module is used by Audio Drivers that wish to have the audio(7I)
- * and mixer(7I) semantics for /dev/audio and /dev/audioctl. In general,
- * the mixer(7I) semantics are a superset of the audio(7I) semantics. Either
- * semantics may be enforced, depending on the mode of the mixer, AM_MIXER_MODE
- * or AM_COMPAT_MODE, respectively. The initial mode is set by the Audio
- * Driver and may be changed on the fly, even while playing or recording
- * audio. When mixing is enabled multiple play and record streams are allowed,
- * and new ioctls are also available. Some legacy applications may not behave
- * well in this mode, thus the availability of the compatibility mode.
- *
- * In addition to providing two sets of semantics, this module also supports
- * two types of audio Codecs, those that have a single play and record stream
- * (traditional Codecs) and those that allow multiple play and record streams
- * (multi-stream Codecs). Because multi-streaming Codecs must do sample rate
- * conversion in hardware in order to mix, the audio stream is not sample rate
- * converted. However, for traditional Codecs the  audio streams must be sample
- * rate converted.
- *
- * The mixer supports a number of data formats on all hardware devices. To do
- * this all incoming audio is converted to a canonical format, which is 16-bit
- * linear PCM. This audio is held in 32-bit integers, which allows it to be
- * mixed with other streams without losing data due to overflowing as would
- * happen if it was stored as a 16-bit short. It is also converted to mono or
- * stereo to match the hardware. When audio is played it is converted to the
- * format the hardware supports.
- *
- * Once the hardware format is set it is never changed again, except for the
- * sample rate.
- *
- * The following formats are supported:
- *	16-bit linear PCM, mono and stereo
- *	8-bit linear PCM, mono and stereo
- *	8-bit u-law, mono and stereo
- *	8-bit a-law, mono and stereo
- *
- * In order to present a uniform view of audio there are three different
- * views of the hardware.
- * 1.	The true configuration of the audio device. This view is not
- *	visible by applications. However it is used by the mixer to properly
- *	format audio for the device.
- * 2.	The COMPAT mode view. The is the view AUDIOCTL channels get when
- *	there aren't any AUDIO channels open in the application. This is
- *	called the master view. The reason this is not the true hardware view
- *	is because some hardware cannot support all formats. For example,
- *	many devices do not support u-law or a-law. The mixer can do this
- *	translation. However it would be confusing to show apps that the
- *	device was really set to linear PCM.
- * 2.	The MIXER mode view. This is the view AUDIO channels and AUDIOCTL
- *	channels in apps that have an AUDIO channel open see. It is mostly
- *	virtual. It always reflects how the channel was programmed. This
- *	becomes the COMPAT view when put into compatibility mode.
- *
- * Most ioctl()s are executed from a task queue. This gives them their own
- * thread and thus they can block without violating the STREAMS blocking rules.
- * The task queue is limited to just one thread. Thus these ioctl()s are
- * serialized and thus access to the hardware state structures and hardware
- * are protected.
- *
- *	NOTE: statep->as_max_chs is set when the audiosup module loads, so we
- *		don't need to protect it when we access it.
- *
- *	NOTE: All linear PCM is assumed to be signed. Therefore if the device
- *		only supports unsigned linear PCM we need to translate either
- *		before we send it to the device or after we take it from the
- *		device. This way we save each Audio Driver from having to do
- *		this.
- *
- *	NOTE: This module depends on the misc/audiosup module being loaded 1st.
- *
- * The audio mixer source code is broken up into three components:
- *	am_main.c:	module load and STREAMS entry points
- *	am_ad.c:	Audio Driver entry points
- *	am_ioctl.c:	ioctl() code
- *
- * These routines are called by the audio support module:
- *	am_open_audio()
- *	am_open_audioctl()
- *	am_close_audio()
- *	am_close_audioctl()
- *	am_restore_state()
- *	am_save_state()
- *	am_rput() - private
- *	am_rsvc() - private
- *	am_wput() - private
- *	am_wsvc() - private
- *
- * These routines are provided for use by the other mixer source code files:
- *	am_apply_gain_balance()
- *	am_convert_int_mono_stereo()
- *	am_convert_to_int()
- *	am_send_signal()
- *	am_update_conv_buffer()
- *	am_update_src_buffer()
- */
-
-#include <sys/modctl.h>
-#include <sys/debug.h>
-#include <sys/kmem.h>
-#include <sys/errno.h>
-#include <sys/stropts.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/file.h>
-#include <sys/fcntl.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_apm.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/impl/audio_mixer_impl.h>
-
-/*
- * Private primary routines provided by this file. These are provided to the
- * audio support module via am_open_audio() and am_open_audioctl().
- */
-static int am_rput(queue_t *, mblk_t *);
-static int am_rsvc(queue_t *);
-static int am_wput(queue_t *, mblk_t *);
-static int am_wsvc(queue_t *);
-
-/*
- * Local routine prototypes used only by this file.
- */
-static void am_flush(queue_t *, mblk_t *);
-static int am_p_process(audio_ch_t *, void *, size_t, int **, size_t *, int **,
-    size_t *);
-static void am_set_waiting(audio_state_t *, pid_t, int, boolean_t, boolean_t);
-
-/*
- * Module Linkage Structures
- */
-/* Linkage structure for loadable drivers */
-static struct modlmisc mixer_modlmisc = {
-	&mod_miscops,		/* drv_modops */
-	MIXER_MOD_NAME,		/* drv_linkinfo */
-};
-
-static struct modlinkage mixer_modlinkage =
-{
-	MODREV_1,		/* ml_rev */
-	(void*)&mixer_modlmisc,	/* ml_linkage */
-	NULL			/* NULL terminates the list */
-};
-
-/*
- *  Loadable Module Configuration Entry Points
- *
- *
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-	ATRACE("in mixer _init()", 0);
-
-	/* standard linkage call */
-	if ((error = mod_install(&mixer_modlinkage)) != 0) {
-		ATRACE_32("mixer _init() error 1", error);
-		return (error);
-	}
-
-	ATRACE("mixer _init() successful", 0);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description
- *	Module de-initialization, called when driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int	error;
-
-	ATRACE("in mixer _fini()", 0);
-
-	if ((error = mod_remove(&mixer_modlinkage)) != 0) {
-		ATRACE_32("mixer _fini() mod_remove failed", error);
-		return (error);
-	}
-
-	ATRACE_32("mixer _fini() successful", error);
-
-	return (error);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo	*modinfop	Pointer to an opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int		rc;
-
-	rc = mod_info(&mixer_modlinkage, modinfop);
-
-	ATRACE_32("mixer _info() returning", rc);
-
-	return (rc);
-
-}	/* _info() */
-
-/*
- * The public main routines for this file.
- */
-
-/*
- * am_open_audio()
- *
- * Description:
- *	AUDIO channel specific open() routine. There are lots of rules here,
- *	depending on audio vs. audioctl and backward compatibility vs. mixer
- *	mode. Thus mode switching must be frozen, which we do by freezing the
- *	taskq.
- *
- *	NOTE: In user context so it is okay for memory allocation to sleep.
- *
- *	NOTE: audio(7I) redefines the O_NONBLOCK/O_NDELAY open() flags to also
- *		mean don't block waiting for a play/record channel to exit.
- *
- * Arguments:
- *	queue_t		*q		Pointer to the read queue
- *	dev_t		*devp		Pointer to the device
- *	int		oflag		Open flags
- *	int		sflag		STREAMS flag
- *	cred_t		*credp		Ptr to the user's credential structure
- *
- * Returns:
- *	AUDIO_SUCCESS			Successfully opened the device
- *	errno				Error number for failed open()
- */
-/*ARGSUSED*/
-int
-am_open_audio(queue_t *q, dev_t *devp, int oflag, int sflag, cred_t *credp)
-{
-	audio_state_t		*statep;
-	audio_apm_info_t	*apm_infop;
-	am_ad_info_t		*ad_infop;
-	am_apm_private_t	*stpptr;
-	am_ch_private_t		*chpptr = NULL;
-	audio_ch_t		*chptr;
-	audio_ch_t		*tchptr;
-	audio_info_t		*default_info;
-	audio_info_t		*hw_info;
-	audio_info_t		*iptr;
-	pid_t			pid;
-	boolean_t		wantread = B_FALSE;
-	boolean_t		wantwrite = B_FALSE;
-	ulong_t			minor;
-	int			ch_flags = 0;
-	int			error;
-	int			i;
-	int			max_chs;
-	int			multi_open = 0;
-	int			mode;
-	int			rc;
-
-	ATRACE("in am_open_audio()", devp);
-
-	/* get the state structure */
-	if ((statep = audio_sup_devt_to_state(*devp)) == NULL) {
-		ATRACE_32(
-		    "am_open_audio() audio_sup_devt_to_state() failed", 0);
-		return (EIO);
-	}
-
-	/* this driver does only a conventional open(), i.e., no clone opens */
-	if (sflag) {
-		ATRACE("am_open_audio() clone open() failure", sflag);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "only conventional open()s are supported");
-		return (EIO);
-	}
-
-	/*
-	 * Determine if opening to FREAD and/or FWRITE. We also make sure
-	 * that at least one of FREAD or FWRITE is set.
-	 */
-	if (oflag & FREAD) {
-		wantread = B_TRUE;
-		ch_flags |= AUDIO_RECORD;
-		ATRACE_32("am_open_audio() allocate channel with read limits",
-		    ch_flags);
-	} else {
-		wantread = B_FALSE;
-	}
-	if (oflag & FWRITE) {
-		wantwrite = B_TRUE;
-		ch_flags |= AUDIO_PLAY;
-		ATRACE_32("am_open_audio() allocate channel with write limits",
-		    ch_flags);
-	} else {
-		wantwrite = B_FALSE;
-	}
-	if (wantread == B_FALSE && wantwrite == B_FALSE) {
-		ATRACE_32("am_open_audio(): must be RD, WR or RDWR", oflag);
-		return (EINVAL);
-	}
-
-	/* figure out if allocates should sleep or not */
-	ch_flags |= (oflag & (O_NONBLOCK|O_NDELAY)) ?
-	    AUDIO_NO_SLEEP : AUDIO_SLEEP;
-
-	/* get the PID for the process opening the channel */
-	pid = ddi_get_pid();
-
-	/* get pointers to various data structures */
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIO)) == NULL) {
-		ATRACE("am_open_audio() audio_sup_get_apm_info() failed", 0);
-		return (EIO);
-	}
-
-	mutex_enter(&apm_infop->apm_lock);
-	stpptr = apm_infop->apm_private;
-	hw_info = &stpptr->am_hw_info;
-	ad_infop = apm_infop->apm_ad_infop;
-	default_info = ad_infop->ad_defaults;
-	max_chs = statep->as_max_chs;
-
-	/* see if we've been offlined */
-	if (!(stpptr->am_flags & AM_PRIV_ON_LINE)) {
-		ATRACE_32("am_open_audio() offline #1", stpptr->am_flags);
-		mutex_exit(&apm_infop->apm_lock);
-		return (EIO);
-	}
-	mutex_exit(&apm_infop->apm_lock);
-
-	/*
-	 * The hardware may support only play or record and not the other.
-	 * If this is the case and the application asked for the direction
-	 * that isn't supported there's no way it can ever succeed. So we
-	 * fail right away.
-	 */
-	if ((wantwrite &&
-	    !(default_info->hw_features & AUDIO_HWFEATURE_PLAY)) ||
-	    (wantread &&
-	    !(default_info->hw_features & AUDIO_HWFEATURE_RECORD))) {
-		ATRACE("am_open_audio() trying to do illegal direction",
-		    default_info);
-		return (EINVAL);
-	}
-
-	/*
-	 * The hardware may be limited to simplex operation, i.e., it may
-	 * only play or record at any one time. We make sure we haven't
-	 * asked for something the hardware can't do before we continue.
-	 */
-	if ((default_info->hw_features & AUDIO_HWFEATURE_DUPLEX) == 0) {
-		ATRACE_32("am_open_audio() simplex", default_info->hw_features);
-		/* make sure we didn't open read/write */
-		if (wantread && wantwrite) {
-			/* we can never, ever succeed, so fail now */
-			ATRACE("am_open_audio() simplex failed, RD_WR", stpptr);
-			return (EBUSY);
-		}
-
-		/* we have to freeze the channels while we look at them */
-		mutex_enter(&statep->as_lock);
-
-		/* make sure we are asking for something we can have */
-		while ((stpptr->am_in_chs && wantwrite) ||
-		    (stpptr->am_out_chs && wantread)) {
-
-			ATRACE("am_open_audio() simplex blocked", stpptr);
-
-			/* is it okay to block and wait for the hw? */
-			if (ch_flags & AUDIO_NO_SLEEP) {
-				mutex_exit(&statep->as_lock);
-				return (EBUSY);
-			}
-
-			/*
-			 * Mark all AUDIO ch waiting flags. We may be
-			 * re-marking some, but there may be new chs since
-			 * the last loop through the channels. We also
-			 * mark both directions so that one direction knows
-			 * that the other is waiting.
-			 */
-			am_set_waiting(statep, AM_NO_PID, AM_SET_WAITING,
-			    AM_SET_PLAY, AM_SET_RECORD);
-
-			/* send a signal so other procs will wake up */
-			mutex_exit(&statep->as_lock);
-			am_send_signal(statep, stpptr);
-			mutex_enter(&statep->as_lock);
-
-			/* wait for a channel to be freed */
-			ATRACE("am_open_audio() simplex blocked", stpptr);
-			if (cv_wait_sig(&statep->as_cv,
-			    &statep->as_lock) <= 0) {
-
-				ATRACE("am_open_audio() simplex signal wakeup",
-				    statep);
-				/*
-				 * This channel may have had a signal, but
-				 * that doesn't mean any of the other channels
-				 * may proceed. So make sure every channel
-				 * gets another go. We clear the waiting flags
-				 * and then any others loop back and reset if
-				 * needed. That's why we do the cv_broadcast().
-				 */
-				am_set_waiting(statep, AM_NO_PID,
-				    AM_CLEAR_WAITING, AM_SET_PLAY,
-				    AM_SET_RECORD);
-				cv_broadcast(&statep->as_cv);
-				mutex_exit(&statep->as_lock);
-				return (EINTR);
-			}
-			/*
-			 * Normal wakeup, clear the waiting flags. If the
-			 * channels need to wait they'll set the flags. That's
-			 * why we wake all the channels up, so they'll go
-			 * through their loop.
-			 */
-			am_set_waiting(statep, AM_NO_PID, AM_CLEAR_WAITING,
-			    AM_SET_PLAY, AM_SET_RECORD);
-			cv_broadcast(&statep->as_cv);
-
-			ATRACE("am_open_audio() simplex normal wakeup", statep);
-		}
-
-		mutex_exit(&statep->as_lock);
-	}
-
-	ASSERT(!mutex_owned(&statep->as_lock));
-
-	/*
-	 * Before we go any further we allocate all the memory we'll need.
-	 * We do it now so that we can sleep while not holding any locks
-	 * or freezing the taskq, which would be bad if we have to wait
-	 * a long time.
-	 */
-	if ((chptr = audio_sup_alloc_ch(statep, &error, AUDIO, ch_flags)) ==
-	    NULL) {
-		ATRACE_32("am_open_audio() alloc returning", error);
-		return (error);
-	}
-	ASSERT(chptr->ch_info.pid == 0);
-
-	/*
-	 * Before we get the rest of the memory we need we make sure that we
-	 * can allocate a channel in the audio driver, if needed. This may
-	 * block for a while, which is why we do it now.
-	 */
-	if (am_ad_setup(statep, stpptr, ad_infop, chptr->ch_info.ch_number,
-	    ch_flags) == AUDIO_FAILURE) {
-			/*
-			 * This should always succeed because we have the
-			 * configuration information. So if we don't there's
-			 * something wrong and we fail.
-			 */
-			ATRACE("am_open_audio() ad_setup() failed", 0);
-			(void) audio_sup_free_ch(chptr);
-			return (EIO);
-	}
-	ATRACE("am_open_audio() ad_setup() succeeded", 0);
-
-	/*
-	 * CAUTION: From here on we have to call ad_teardown() to
-	 *	free the stream resources in the Audio Driver if
-	 *	there is an error.
-	 *
-	 * Now allocate the rest of memory.
-	 */
-	_NOTE(NOW_INVISIBLE_TO_OTHER_THREADS(*chpptr));
-	chpptr = kmem_zalloc(sizeof (*chpptr), KM_SLEEP);
-	chpptr->acp_psb_size = ad_infop->ad_play.ad_bsize;
-	chpptr->acp_play_samp_buf = kmem_zalloc(chpptr->acp_psb_size, KM_SLEEP);
-
-	/* get an audio structure to use */
-	iptr = kmem_alloc(sizeof (*iptr), KM_SLEEP);
-
-	/*
-	 * CAUTION: From here on we must free the channel and memory if we
-	 *	need to return on an error.
-	 *
-	 * Getting to here means there aren't any conflicts with the hardware,
-	 * so now the semantics of MIXER and COMPAT modes come to play. Because
-	 * these semantics depend on the mixer mode we have to block it from
-	 * changing. We do this by taking control of the taskq. Once we have
-	 * the taskq we know there's no way for the mode to change.
-	 */
-	ATRACE("am_open_audio() entering again:", chptr);
-
-again:	/* we loop back to here when cv_wait_sig returns due to a wakeup */
-	ATRACE("am_open_audio() again: loop", chptr);
-
-	ASSERT(!MUTEX_HELD(&statep->as_lock));
-
-	/* check again to see if we've been offlined */
-	mutex_enter(&apm_infop->apm_lock);
-	if (!(stpptr->am_flags & AM_PRIV_ON_LINE)) {
-		ATRACE_32("am_open_audio() offline #2", stpptr->am_flags);
-		mutex_exit(&apm_infop->apm_lock);
-		rc = EIO;
-		goto error;
-	}
-	mutex_exit(&apm_infop->apm_lock);
-
-	/*
-	 * CAUTION: We must keep the taskq blocked until we have everything
-	 *	done. Otherwise a mode switch will make the state inconsistent.
-	 *	Every time we jump to again: the taskq has been released so it
-	 *	is possible for ioctl()s to make progress. This also means we
-	 *	have to do a complete check every time since the hardware state
-	 *	could have changed.
-	 */
-	audio_sup_taskq_suspend(stpptr->am_taskq);
-
-	/* freeze the channels after freezing the taskq */
-	mutex_enter(&statep->as_lock);
-
-	mode = stpptr->am_pstate->apm_mode;	/* reget, it may have changed */
-
-	/*
-	 * If in MIXER mode then we look for multiple open()s with the same
-	 * PID. If no other AUDIO channels for the PID then we always succeed.
-	 * If in COMPAT mode we look for any open()s in the same direction.
-	 */
-	ASSERT(pid != 0);
-	if (mode == AM_MIXER_MODE) {
-		for (i = 0, multi_open = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-
-			/* skip non-audio channels and those not allocated */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    chptr == tchptr ||
-			    tchptr->ch_info.dev_type != AUDIO ||
-			    tchptr->ch_info.pid != pid) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/*
-			 * The same process can have separate read and write
-			 * open()s. But it can't have two reads or writes,
-			 * unless multiple open()s per process are allowed.
-			 */
-			if (((am_ch_private_t *)tchptr->ch_private)->
-			    acp_flags & AM_CHNL_MULTI_OPEN) {
-				mutex_exit(&tchptr->ch_lock);
-				ch_flags |= AM_CHNL_MULTI_OPEN;
-				/* don't need to look any further */
-				break;
-			} else if (
-			    (wantread && (tchptr->ch_dir & AUDIO_RECORD)) ||
-			    (wantwrite && (tchptr->ch_dir & AUDIO_PLAY))) {
-				mutex_exit(&tchptr->ch_lock);
-				/*
-				 * Multiple open()s not supported. So let the
-				 * taskq continue.
-				 */
-				audio_sup_taskq_resume(stpptr->am_taskq);
-
-				/* if O_NDELAY then we just return */
-				if (ch_flags & AUDIO_NO_SLEEP) {
-					mutex_exit(&statep->as_lock);
-					rc = EBUSY;
-					goto error;
-				}
-
-				/* set waiting flags and wait */
-				am_set_waiting(statep, pid, AM_SET_WAITING,
-				    wantwrite, wantread);
-
-				/* send a signal so other procs will wake up */
-				mutex_exit(&statep->as_lock);
-				am_send_signal(statep, stpptr);
-				mutex_enter(&statep->as_lock);
-
-				/*
-				 * Wait for channels to be freed so we can
-				 * try again.
-				 */
-				if (cv_wait_sig(&statep->as_cv,
-				    &statep->as_lock) <= 0) {
-					/*
-					 * Signal wakeup, clear waiting flags
-					 * for the PID. We wakeup all waiting
-					 * channels, thus they go through their
-					 * loops and remark waiting.
-					 */
-					am_set_waiting(statep, pid,
-					    AM_CLEAR_WAITING, wantwrite,
-					    wantread);
-					mutex_exit(&statep->as_lock);
-					rc = EINTR;
-					goto error;
-				}
-				/*
-				 * Normal wakeup, clear the waiting flags. If
-				 * the channels need to wait they'll set the
-				 * flags. That's why we wake all the channels
-				 * up, so they'll go through their loop.
-				 */
-				am_set_waiting(statep, pid, AM_CLEAR_WAITING,
-				    wantwrite, wantread);
-				cv_broadcast(&statep->as_cv);
-				mutex_exit(&statep->as_lock);
-				goto again;
-
-			}
-			mutex_exit(&tchptr->ch_lock);
-		}
-	} else {
-		ASSERT(mode == AM_COMPAT_MODE);
-
-		/* we can't have two reads or writes at one once */
-		mutex_enter(&chptr->ch_lock);
-		if ((wantread && stpptr->am_in_chs) ||
-		    (wantwrite && stpptr->am_out_chs)) {
-			mutex_exit(&chptr->ch_lock);
-			/*
-			 * Multiple open()s not supported. So let the
-			 * taskq continue.
-			 */
-			audio_sup_taskq_resume(stpptr->am_taskq);
-
-			/* if O_NDELAY then we just return */
-			if (ch_flags & AUDIO_NO_SLEEP) {
-				mutex_exit(&statep->as_lock);
-				rc = EBUSY;
-				goto error;
-			}
-
-			/* set waiting flags and wait */
-			am_set_waiting(statep, pid, AM_SET_WAITING, wantwrite,
-			    wantread);
-
-			/* send a signal so other procs will wake up */
-			mutex_exit(&statep->as_lock);
-			am_send_signal(statep, stpptr);
-			mutex_enter(&statep->as_lock);
-
-			/* wait for channels to be freed so we can try again */
-			if (cv_wait_sig(&statep->as_cv,
-			    &statep->as_lock) <= 0) {
-				/*
-				 * Signal wakeup, clear waiting flags for the
-				 * PID. We wakeup all waiting channels, thus
-				 * they go through their loops and remark
-				 * waiting.
-				 */
-				am_set_waiting(statep, pid, AM_CLEAR_WAITING,
-				    wantwrite, wantread);
-				mutex_exit(&statep->as_lock);
-				rc = EINTR;
-				goto error;
-			}
-			/*
-			 * Normal wakeup, clear the waiting flags. If the
-			 * channels need to wait they'll set the flags. That's
-			 * why we wake all the channels up, so they'll go
-			 * through their loop.
-			 */
-			am_set_waiting(statep, pid, AM_CLEAR_WAITING,
-			    wantwrite, wantread);
-			cv_broadcast(&statep->as_cv);
-			mutex_exit(&statep->as_lock);
-			goto again;
-
-		}
-		mutex_exit(&chptr->ch_lock);
-	}
-
-	ASSERT(MUTEX_HELD(&statep->as_lock));
-	ASSERT(audio_sup_taskq_suspended(stpptr->am_taskq) ==
-	    AUDIO_TASKQ_SUSPENDED);
-
-	/*
-	 * If we get here there are no conflicting open()s. However, if in
-	 * MIXER mode then we may be limited by the max number of channels,
-	 * max number of read channels, and the max number of write channels.
-	 * It wasn't easy to check above, so we do it now.
-	 */
-	if (mode == AM_MIXER_MODE &&
-	    (stpptr->am_channels >= max_chs ||
-	    (wantread && stpptr->am_in_chs >= stpptr->am_max_in_chs) ||
-	    (wantwrite && stpptr->am_out_chs >= stpptr->am_max_out_chs))) {
-		/* let the taskq continue working */
-		audio_sup_taskq_resume(stpptr->am_taskq);
-
-		/* if O_NDELAY then we just return */
-		if (ch_flags & AUDIO_NO_SLEEP) {
-			mutex_exit(&statep->as_lock);
-			rc = EBUSY;
-			goto error;
-		}
-
-		/* set waiting flags and wait */
-		am_set_waiting(statep, pid, AM_SET_WAITING, wantwrite,
-		    wantread);
-
-		/* send a signal so other procs will wake up */
-		mutex_exit(&statep->as_lock);
-		am_send_signal(statep, stpptr);
-		mutex_enter(&statep->as_lock);
-
-		if (cv_wait_sig(&statep->as_cv, &statep->as_lock) <= 0) {
-			/*
-			 * Signal wakeup, clear waiting flags for the PID.
-			 * We wakeup all waiting channels, thus they go
-			 * through their loops and remark waiting.
-			 */
-			am_set_waiting(statep, pid, AM_CLEAR_WAITING,
-			    wantwrite, wantread);
-			mutex_exit(&statep->as_lock);
-			rc = EINTR;
-			goto error;
-		}
-		/*
-		 * Normal wakeup, clear the waiting flags. If the channels
-		 * need to wait they'll set the flags. That's why we wake all
-		 * the channels up, so they'll go through their loop.
-		 */
-		am_set_waiting(statep, pid, AM_CLEAR_WAITING, wantwrite,
-		    wantread);
-		cv_broadcast(&statep->as_cv);
-		mutex_exit(&statep->as_lock);
-		goto again;
-	}
-
-	ASSERT(MUTEX_HELD(&statep->as_lock));
-	ASSERT(audio_sup_taskq_suspended(stpptr->am_taskq) ==
-	    AUDIO_TASKQ_SUSPENDED);
-
-	/*
-	 * We have a good channel, all open() semantics pass, so init the ch.
-	 *
-	 * CAUTION: pid isn't filled in until the very end. Otherwise
-	 *	other routines that look for AUDIO or AUDIOCTL channels may
-	 *	think that the channel is fully allocated and available for
-	 *	use.
-	 */
-	mutex_enter(&chptr->ch_lock);
-	ASSERT(chptr->ch_statep == statep);
-
-	chptr->ch_qptr =		q;
-	chptr->ch_wput =		am_wput;
-	chptr->ch_wsvc =		am_wsvc;
-	chptr->ch_rput =		am_rput;
-	chptr->ch_rsvc =		am_rsvc;
-	chptr->ch_dir =			ch_flags & AUDIO_BOTH;
-	chptr->ch_dev =			*devp;
-	chptr->ch_private =		chpptr;
-	chptr->ch_info.info_size =	sizeof (audio_info_t);
-	chptr->ch_dev_info =		ad_infop->ad_dev_info;
-
-	chpptr->acp_flags =		multi_open;
-	chpptr->acp_reading =		wantread;
-	chpptr->acp_writing =		wantwrite;
-
-	/* get the minor device for the new channel */
-	minor = audio_sup_ch_to_minor(statep, chptr->ch_info.ch_number);
-	ATRACE_32("am_open_audio() channel number", chptr->ch_info.ch_number);
-	ATRACE_32("am_open_audio() new minor number", minor);
-
-	/*
-	 * Setup the channel. We use the audio_info structure allocated above.
-	 * If we are in MIXER mode then we keep this structure and it becomes
-	 * the virtual state. If in COMPAT mode we free it after we set the
-	 * state.
-	 *
-	 * We init both play and record, that way it is always filled in.
-	 * Not all members are filled in, that happens in the ioctl() code.
-	 * Thus we have to init the structure so am_audio_set_info() will
-	 * ignore them.
-	 */
-	AUDIO_INIT(iptr, sizeof (*iptr));
-
-	iptr->record.sample_rate =	default_info->record.sample_rate;
-	iptr->record.channels =		AUDIO_CHANNELS_MONO;
-	iptr->record.precision =	AUDIO_PRECISION_8;
-	iptr->record.encoding =		AUDIO_ENCODING_ULAW;
-	iptr->record.buffer_size =	ad_infop->ad_record.ad_bsize;
-	iptr->record.samples =		0;
-	iptr->record.eof =		0;
-	iptr->record.pause =		0;
-	iptr->record.error =		0;
-	iptr->record.waiting =		0;
-	iptr->record.minordev =		minor;
-	iptr->record.open =		0;
-	iptr->record.active =		0;
-
-	iptr->play.sample_rate =	default_info->play.sample_rate;
-	iptr->play.channels =		AUDIO_CHANNELS_MONO;
-	iptr->play.precision =		AUDIO_PRECISION_8;
-	iptr->play.encoding =		AUDIO_ENCODING_ULAW;
-	iptr->play.buffer_size =	0;
-	iptr->play.samples =		0;
-	iptr->play.eof =		0;
-	iptr->play.pause =		0;
-	iptr->play.error =		0;
-	iptr->play.waiting =		0;
-	iptr->play.minordev =		minor;
-	iptr->play.open =		0;
-	iptr->play.active =		0;
-
-	if (mode == AM_MIXER_MODE) {	/* virtual channel */
-		iptr->record.gain =	stpptr->am_pstate->apm_rgain;
-		iptr->record.balance =	stpptr->am_pstate->apm_rbal;
-		iptr->play.gain =	stpptr->am_pstate->apm_pgain;
-		iptr->play.balance =	stpptr->am_pstate->apm_pbal;
-		chptr->ch_info.info = iptr;
-		iptr->output_muted = 0;
-		iptr->ref_cnt = 1;
-	} else {	/* AM_COMPAT_MODE, physical channel */
-		iptr->record.gain =	hw_info->record.gain;
-		iptr->record.balance =	hw_info->record.balance;
-		iptr->play.gain =	hw_info->play.gain;
-		iptr->play.balance =	hw_info->play.balance;
-		chptr->ch_info.info = hw_info;
-		iptr->output_muted = hw_info->output_muted;
-		iptr->ref_cnt = 0;		/* delete struct when done */
-	}
-	iptr->monitor_gain = hw_info->monitor_gain;
-
-	mutex_exit(&chptr->ch_lock);
-	ASSERT(MUTEX_HELD(&statep->as_lock));
-
-	/* before we setup the hardware we need to init src */
-	if (ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		if (wantread) {
-			ATRACE("am_open_audio() REC., calling src init", chptr);
-			mutex_enter(&chptr->ch_lock);
-			chpptr->acp_ch_rbuf_size =
-				ad_infop->ad_record.ad_conv->ad_src_init(
-					AM_SRC_CHPTR2HDL(chptr), AUDIO_RECORD);
-			mutex_exit(&chptr->ch_lock);
-		}
-		if (wantwrite) {
-			ATRACE("am_open_audio() PLAY, calling src init", chptr);
-			mutex_enter(&chptr->ch_lock);
-			chpptr->acp_ch_pbuf_size =
-				ad_infop->ad_play.ad_conv->ad_src_init(
-					AM_SRC_CHPTR2HDL(chptr), AUDIO_PLAY);
-			mutex_exit(&chptr->ch_lock);
-		}
-	}
-
-	/* set the open flags and increment the counts */
-	if (wantread) {
-		stpptr->am_in_chs++;
-	}
-	if (wantwrite) {
-		stpptr->am_out_chs++;
-	}
-	stpptr->am_channels++;
-
-	/* setting the hardware can take a long time, so let channels go */
-	mutex_exit(&statep->as_lock);
-
-	/* setup the hardware */
-	ASSERT(audio_sup_taskq_suspended(stpptr->am_taskq) ==
-	    AUDIO_TASKQ_SUSPENDED);
-	if (am_audio_set_info(chptr, iptr, NULL) == AUDIO_FAILURE) {
-		ATRACE("am_open_audio() hw set failed", chptr);
-
-		mutex_enter(&statep->as_lock);
-		if (wantread) {
-			stpptr->am_in_chs--;
-		}
-		if (wantwrite) {
-			stpptr->am_out_chs--;
-		}
-		stpptr->am_channels--;
-		mutex_exit(&statep->as_lock);
-
-		/* let the taskq continue */
-		audio_sup_taskq_resume(stpptr->am_taskq);
-
-		rc = EINVAL;
-
-		/* free up any sample rate conv. memory we allocated earlier */
-		if (wantread) {
-			ad_infop->ad_record.ad_conv->ad_src_exit(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_RECORD);
-		}
-		if (wantwrite) {
-			ad_infop->ad_play.ad_conv->ad_src_exit(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_PLAY);
-		}
-
-		goto error;
-	}
-
-	/*
-	 * From here on we can't fail. If in COMPAT mode we need to clean up
-	 * the hardware reference counts.
-	 */
-	if (mode == AM_COMPAT_MODE) {
-		kmem_free(iptr, sizeof (*iptr));
-		iptr = hw_info;
-		iptr->ref_cnt++;
-	}
-	if (wantread) {
-		iptr->record.open = 1;
-	}
-	if (wantwrite) {
-		iptr->play.open = 1;
-	}
-
-	/*
-	 * For mixer mode we see if there are any AUDIOCTL channels with
-	 * this process. If there are then we need to re-associate them
-	 * to this channel.
-	 *
-	 * NOTE: We must still have the taskq frozen.
-	 */
-	mutex_enter(&statep->as_lock);
-
-	if (mode == AM_MIXER_MODE) {
-		/* we have to keep the channels stable */
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-
-			/* skip myself, unallocated, and closing channels */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    chptr == tchptr ||
-			    (tchptr->ch_private &&
-			    ((am_ch_private_t *)tchptr->ch_private)->
-			    acp_flags & AM_CHNL_CLOSING)) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/*
-			 * Skip if different PIDs, including 0. pid is set
-			 * to 0 when am_close_audio*() is entered, so we aren't
-			 * associated with a closing channel.
-			 */
-			if (tchptr->ch_info.pid != pid) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* same PID, make sure it's AUDIOCTL */
-			if (tchptr->ch_info.dev_type != AUDIOCTL) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/*
-			 * Yes! It's possible that the AUDIOCTL ch is
-			 * already attached to an AUDIO ch. If so we
-			 * don't muck with it. From this point it is
-			 * indeterminate as to what happens with
-			 * AUDIOCTL channels. If it isn't we associate
-			 * it with this AUDIO channel.
-			 */
-			if (tchptr->ch_info.info == hw_info) {
-				hw_info->ref_cnt--;
-				ASSERT(hw_info->ref_cnt >= 1);
-
-				((am_ch_private_t *)
-				    tchptr->ch_private)->acp_flags |=
-				    AM_CHNL_CONTROL;
-				tchptr->ch_info.info = iptr;
-				iptr->ref_cnt++;
-			}
-
-			/*
-			 * we don't break because there can be more
-			 * than one AUDIOCTL for any one AUDIO channel.
-			 */
-			mutex_exit(&tchptr->ch_lock);
-		}
-
-		/*
-		 * Above we set Codec info, but we need to set the
-		 * rest of the info in the iptr structure, except for
-		 * hardware related members, which we fill in when
-		 * needed.
-		 */
-		iptr->play.avail_ports = hw_info->play.avail_ports;
-		iptr->play.mod_ports = hw_info->play.mod_ports;
-		iptr->record.avail_ports = hw_info->record.avail_ports;
-		iptr->record.mod_ports = hw_info->record.mod_ports;
-	}
-
-	ASSERT(iptr != NULL);
-	ASSERT(audio_sup_taskq_suspended(stpptr->am_taskq) ==
-	    AUDIO_TASKQ_SUSPENDED);
-
-	/* we start out open and empty */
-	chpptr->acp_flags |= (AM_CHNL_OPEN | AM_CHNL_EMPTY);
-
-	/* we've made it through all the checks, so it's safe to make the dev */
-	*devp = makedevice(getmajor(*devp), minor);
-	ATRACE("am_open_audio() made device", devp);
-
-	mutex_enter(&chptr->ch_lock);
-	chptr->ch_dev = *devp;
-	mutex_exit(&chptr->ch_lock);
-
-	ASSERT(chptr->ch_info.ch_number ==
-			audio_sup_minor_to_ch(statep, minor));
-
-	ATRACE("am_open_audio() qprocson()", chptr);
-
-	/*
-	 * WARNING: Do this after we can no longer fail or we will have a
-	 * potential memory leak. Also, do it before qprocson().
-	 */
-	audio_sup_set_qptr(q, *devp, chptr);
-
-	/* schedule the queue */
-	qprocson(q);
-
-	/*
-	 * Now we can set the pid. We don't need to lock the structure because
-	 * the worst thing that will happen is the channel will be skipped and
-	 * then picked up on the next sweep through the channels. Once this is
-	 * set the other mixer routines will see the channel.
-	 *
-	 * CAUTION: This must be after qprocson(), otherwise other threads will
-	 *	try to process STREAMS messages on a partially started stream.
-	 *	This will cause a panic. This shows up in am_send_signal().
-	 */
-	mutex_enter(&chptr->ch_lock);
-	chptr->ch_info.pid = pid;
-	mutex_exit(&chptr->ch_lock);
-
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*chpptr));
-
-	/* thaw the taskq AFTER setting the pid */
-	audio_sup_taskq_resume(stpptr->am_taskq);
-
-	mutex_exit(&statep->as_lock);
-
-	/* MUST be after inc. above - not locked any more, start recording */
-	if (wantread) {
-		mutex_enter(&chptr->ch_lock);
-		if (am_set_record_streams(chptr) == AUDIO_SUCCESS) {
-			chpptr->acp_flags |= AM_CHNL_RSTREAMS_SET;
-			ATRACE("am_open() record STREAMS water marks adjusted",
-			    0);
-#ifdef DEBUG
-		} else {
-			ATRACE("am_open() record STREAMS water marks not "
-			    "adjusted", 0);
-#endif
-		}
-		mutex_exit(&chptr->ch_lock);
-
-		/* start recording, regardless of mode or Codec type */
-		ATRACE("am_open_audio() starting record DMA engine", statep);
-		/* set before start for mode switch */
-		hw_info->record.active = 1;
-		iptr->record.active = 1;
-		iptr->record.pause = 0;
-		if (am_ad_start_record(statep, stpptr, ad_infop,
-		    chptr->ch_info.ch_number, AM_SERIALIZE) == AUDIO_FAILURE) {
-			iptr->record.active = 0;
-			hw_info->record.active = 0;
-			/* we don't change pause flag if failed to start */
-		}
-	}
-
-	ATRACE("am_open_audio() successful", statep);
-
-	am_send_signal(statep, stpptr);
-
-	return (AUDIO_SUCCESS);
-
-error:
-	kmem_free(chpptr->acp_play_samp_buf, chpptr->acp_psb_size);
-	kmem_free(chpptr, sizeof (*chpptr));
-	if (iptr && iptr != hw_info) {
-		kmem_free(iptr, sizeof (*iptr));
-	}
-	_NOTE(NOW_VISIBLE_TO_OTHER_THREADS(*chpptr));
-
-	mutex_enter(&chptr->ch_lock);
-	chptr->ch_private = NULL;
-	chptr->ch_info.info = NULL;
-	mutex_exit(&chptr->ch_lock);
-
-	(void) audio_sup_free_ch(chptr);
-
-	/* tell the Audio Driver to free up ch resources */
-	am_ad_teardown(statep, stpptr, ad_infop, chptr->ch_info.ch_number,
-	    (ch_flags & AUDIO_BOTH));
-	ATRACE("am_open_audio() ad_teardown() returned", 0);
-
-	ATRACE_32("am_open_audio() at \"error\"", rc);
-
-	return (rc);
-
-}	/* am_open_audio() */
-
-/*
- * am_open_audioctl()
- *
- * Description:
- *	AUDIOCTL channel specific open() routine. There are lots of rules here,
- *	but not as bad audio am_open_audio(). We don't worry about whether the
- *	channel is opened for read or writing as we don't read() or write()
- *	the device.
- *
- *	How AUDIOCTL channels are allocated:
- *	    COMPAT MODE
- *		always points to the hardware
- *	    MIXER MODE
- *		if no AUDIO channels open() for the process then points to
- *		the hardware
- *		if one AUDIO channel open() for the process then point to that
- *		channel, thus it is virtual
- *		if more than one AUDIO channel open() for the process then
- *		point to the first AUDIO channel, this includes multiple open()s
- *		as well as one for read and one for write, which isn't the same
- *		as a multiple open()
- *
- *	NOTE: In user context so it is okay for memory allocation to sleep.
- *
- * Arguments:
- *	queue_t		*q		Pointer to the read queue
- *	dev_t		*devp		Pointer to the device
- *	int		oflag		Open flags
- *	int		sflag		STREAMS flag
- *	cred_t		*credp		Ptr to the user's credential structure
- *
- * Returns:
- *	AUDIO_SUCCESS			Successfully opened the device
- *	errno				Error number for failed open()
- */
-/*ARGSUSED*/
-int
-am_open_audioctl(queue_t *q, dev_t *devp, int oflag, int sflag, cred_t *credp)
-{
-	audio_state_t		*statep;
-	audio_apm_info_t	*apm_infop;
-	audio_ch_t		*chptr;
-	audio_ch_t		*tchptr;
-	audio_info_t		*hw_info;
-	audio_info_t		*iptr;
-	am_ch_private_t		*chpptr = 0;
-	am_ad_info_t		*ad_infop;
-	am_apm_private_t	*stpptr;
-	pid_t			pid;
-	ulong_t			minor;
-	int			ch_flags;
-	int			error;
-	int			i;
-	int			max_chs;
-	int			mode;
-
-	ATRACE("in am_open_audioctl()", devp);
-
-	/* get the state structure */
-	if ((statep = audio_sup_devt_to_state(*devp)) == NULL) {
-		ATRACE_32(
-		    "am_open_audioctl() audio_sup_devt_to_state() failed", 0);
-		return (EIO);
-	}
-
-	/* this driver does only a conventional open(), i.e., no clone opens */
-	if (sflag) {
-		ATRACE("am_open_audioctl() clone open() failure", sflag);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE, "mixer:"
-		    "only conventional open()s are supported");
-		return (EIO);
-	}
-
-	/* figure out if allocates should sleep or not */
-	ch_flags = (oflag & (O_NONBLOCK|O_NDELAY)) ?
-	    AUDIO_NO_SLEEP : AUDIO_SLEEP;
-
-	/* get the PID for the process opening the channel */
-	pid = ddi_get_pid();
-
-	/* get pointers to various data structures */
-	if ((apm_infop = audio_sup_get_apm_info(statep, AUDIOCTL)) == NULL) {
-		ATRACE("am_open_audioctl() audio_sup_get_apm_info() failed", 0);
-		return (EIO);
-	}
-	stpptr = apm_infop->apm_private;
-	hw_info = &stpptr->am_hw_info;
-	ad_infop = apm_infop->apm_ad_infop;
-	max_chs = statep->as_max_chs;
-
-	/* see if we've been offlined */
-	mutex_enter(&apm_infop->apm_lock);
-	if (!(stpptr->am_flags & AM_PRIV_ON_LINE)) {
-		ATRACE_32("am_open_audio() offline", stpptr->am_flags);
-		mutex_exit(&apm_infop->apm_lock);
-		return (EIO);
-	}
-	mutex_exit(&apm_infop->apm_lock);
-
-	/*
-	 * Before we go any further we allocate the channel and all memory
-	 * that we'll need. That way we don't sleep holding any locks or
-	 * the taskq.
-	 */
-	if ((chptr = audio_sup_alloc_ch(statep, &error, AUDIOCTL, ch_flags)) ==
-	    NULL) {
-		ATRACE("am_open_audioctl() alloc returning", statep);
-		return (error);
-	}
-
-	ASSERT(chptr->ch_info.pid == 0);
-
-	chpptr = kmem_zalloc(sizeof (*chpptr), KM_SLEEP);
-
-
-	/*
-	 * CAUTION: We must keep the taskq blocked until we have everything
-	 *	done. Otherwise a mode switch will make the state inconsistent.
-	 */
-	audio_sup_taskq_suspend(stpptr->am_taskq);
-
-	/* freeze channel allocation */
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&chptr->ch_lock);
-
-	/* mode may have changed, so re-get */
-	mode = stpptr->am_pstate->apm_mode;
-
-	/*
-	 * We have a good channel, all open() semantics pass, so init the ch.
-	 *
-	 * CAUTION: pid isn't filled in until the very end. Otherwise
-	 *	other routines that look for AUDIO or AUDIOCTL channels may
-	 *	think that the channel is fully allocated and available for
-	 *	use.
-	 */
-	chptr->ch_qptr =		q;
-	ASSERT(chptr->ch_statep == statep);
-	chptr->ch_wput =		am_wput;
-	chptr->ch_wsvc =		am_wsvc;
-	chptr->ch_rput =		am_rput;
-	chptr->ch_rsvc =		am_rsvc;
-	chptr->ch_dir =			0;
-	chptr->ch_dev =			*devp;
-	chptr->ch_private =		chpptr;
-	chptr->ch_info.info_size =	sizeof (audio_info_t);
-	chptr->ch_dev_info =		ad_infop->ad_dev_info;
-
-	chpptr->acp_flags =		0;
-
-	/* get the minor device for the new channel */
-	minor = audio_sup_ch_to_minor(statep, chptr->ch_info.ch_number);
-	ATRACE_32("am_open_audioctl() channel number",
-	    chptr->ch_info.ch_number);
-	ATRACE_32("am_open_audioctl() new minor number", minor);
-
-	/*
-	 * Update the number of AUDIOCTL chs open for the mixer. We
-	 * don't bother updating the read or write counts because
-	 * they don't make sense for AUDIOCTL channels.
-	 */
-	stpptr->am_channels++;
-
-	mutex_exit(&chptr->ch_lock);
-
-	/* figure out the audio_info structure, and initialize */
-	if (mode == AM_MIXER_MODE) {
-		/* is this AUDIOCTL channel related to an AUDIO ch? */
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-			/* skip myself, unallocated & diff. PID channels */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr == chptr ||
-			    tchptr->ch_info.dev_type != AUDIO ||
-			    tchptr->ch_info.pid != pid) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* yes, so link the info structs */
-			ATRACE("am_open_audioctl() AUDIOCTL related", chptr);
-
-			/* we lock the channel, not the state */
-			iptr = tchptr->ch_info.info;
-			chptr->ch_info.info = iptr;
-			iptr->ref_cnt++;
-			chpptr->acp_flags |= AM_CHNL_CONTROL;
-			mutex_exit(&tchptr->ch_lock);
-			break;
-		}
-
-		if (i == max_chs) {	/* no, so link to HW */
-			ATRACE("am_open_audioctl() AUDIOCTL not related",
-			    chptr);
-
-			iptr = hw_info;
-			mutex_enter(&chptr->ch_lock);
-			chptr->ch_info.info = iptr;
-			mutex_exit(&chptr->ch_lock);
-			iptr->ref_cnt++;
-		}
-	} else {
-		/* in COMPAT mode there is only one state structure */
-		ATRACE("am_open_audioctl() AUDIOCTL: mode == AM_COMPAT_MODE",
-		    chptr);
-
-		iptr = hw_info;
-		mutex_enter(&chptr->ch_lock);
-		chptr->ch_info.info = iptr;
-		mutex_exit(&chptr->ch_lock);
-		iptr->ref_cnt++;
-	}
-
-	/* we have all the state info, so we can let the state go */
-	mutex_exit(&statep->as_lock);
-
-	ATRACE("am_open_audioctl() AUDIOCTL iptr", iptr);
-	ASSERT(iptr != NULL);
-
-	mutex_enter(&chptr->ch_lock);
-	/* we start out open and empty */
-	chpptr->acp_flags |= (AM_CHNL_OPEN | AM_CHNL_EMPTY);
-
-	ASSERT(iptr != NULL);
-
-	/* we've made it through all the checks, so it's safe to make the dev */
-	*devp = makedevice(getmajor(*devp), minor);
-	ATRACE("am_open_audioctl() made device", devp);
-
-	ASSERT(chptr->ch_info.ch_number ==
-			audio_sup_minor_to_ch(statep, minor));
-
-	ATRACE("am_open_audioctl() qprocson()", chptr);
-	mutex_exit(&chptr->ch_lock);
-
-	/*
-	 * WARNING: Do this after we can no longer fail or we will have a
-	 * potential memory leak. Also, do it before qprocson().
-	 */
-	audio_sup_set_qptr(q, *devp, chptr);
-
-	/* schedule the queue */
-	qprocson(q);
-
-	/*
-	 * Now we can set the pid. We don't need to lock the structure because
-	 * the worst thing that will happen is the channel will be skipped and
-	 * then picked up on the next sweep through the channels. Once this is
-	 * set the other mixer routines will see the channel.
-	 *
-	 * CAUTION: This must be after qprocson(), otherwise other threads will
-	 *	try to process STREAMS messages on a partially started stream.
-	 *	This will cause a panic. This shows up in am_send_signal().
-	 */
-	mutex_enter(&chptr->ch_lock);
-	chptr->ch_info.pid = pid;
-	mutex_exit(&chptr->ch_lock);
-
-	/* thaw the taskq AFTER setting the pid */
-	audio_sup_taskq_resume(stpptr->am_taskq);
-
-	ATRACE("am_open_audioctl() successful", statep);
-
-	am_send_signal(statep, stpptr);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_open_audioctl() */
-
-/*
- * am_close_audio()
- *
- * Description:
- *	Close a minor device, returning the minor number to the pool so that
- *	open(2) may use it again.
- *
- *	chpptr->acp_flags is used to coordinate draining the write queue.
- *	am_close_audio() sets flags to AM_CHNL_CLOSING. It then waits for the
- *	flags to have AM_CHNL_EMPTY set by am_get_audio() when all available
- *	data has been drained and played. If a signal interrupts the draining
- *	the queue is flushed and the AM_CHNL_OPEN flag is cleared.
- *	am_get_audio() then ignores this channel until it is open()ed again.
- *
- *	There are a number of rules that have to be followed when closing
- *	an audio channel. Some of them depend on the state of the mixer.
- *	Thus we must stop any mode switching. This is done by freezing the
- *	taskq.
- *
- *	NOTE: When ch_info.info is set and ref_cnt is changed statep->as_lock
- *		is used and the taskq MUST be frozen. Otherwise a mode switch
- *		will mess things up.
- *
- *	NOTE: We need to behave differently for a normal close() vs. the user
- *		app calling exit(). Unfortunately there isn't a DDI compliant
- *		method for doing this, so we take a look at the current thread
- *		and see if it's exiting or not. This is how the old diaudio
- *		module did it.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the read queue
- *	int		flag	File status flag
- *	cred_t		*credp	Pointer to the user's credential structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Successfully closed the device
- *	errno			Error number for failed close()
- */
-/*ARGSUSED*/
-int
-am_close_audio(queue_t *q, int flag,  cred_t *credp)
-{
-	audio_info_t		*info;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_ch_t		*tchptr;
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	audio_info_t		*alt_info = NULL;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	mblk_t			*rec_mp;
-	pid_t			save_pid;
-	int			codec_type = ad_infop->ad_codec_type;
-	int			dir = 0;
-	int			i;
-	int			max_chs;
-	int			mode;
-	int			tmp_stream;
-	int			was_reading;
-	int			was_writing;
-
-	ATRACE("in am_close_audio()", chptr);
-	ATRACE_32("am_close_audio() channel number", chptr->ch_info.ch_number);
-
-	ASSERT(q == chptr->ch_qptr);
-
-	/* mark the channel as in the process of being closed */
-	mutex_enter(&chptr->ch_lock);
-	was_reading = chpptr->acp_reading;
-	was_writing = chpptr->acp_writing;
-	chpptr->acp_flags |= AM_CHNL_CLOSING;
-	mutex_exit(&chptr->ch_lock);
-
-	/* set the direction for ad_teardown() */
-	if (was_reading) {
-		dir |= AUDIO_RECORD;
-	}
-	if (was_writing) {
-		dir |= AUDIO_PLAY;
-	}
-
-	/*
-	 * Wait for queue to drain, unless we were signaled in AUDIO_DRAIN
-	 * or the process is exiting (in which case we use the hack).
-	 */
-	ATRACE("am_close_audio() checking to see if need to wait", chptr);
-
-	if (was_writing) {
-		/* we now need the info, so protect it by freezing the taskq */
-		audio_sup_taskq_suspend(stpptr->am_taskq);
-
-		mutex_enter(&chptr->ch_lock);
-
-		info = chptr->ch_info.info;
-
-		if (info->play.active && !info->play.pause) {
-			/* release the taskq so it can continue */
-			audio_sup_taskq_resume(stpptr->am_taskq);
-			ATRACE("am_close_audio() need to wait", chptr);
-			while (!(chpptr->acp_flags & AM_CHNL_EMPTY) &&
-			    !(curthread->t_proc_flag & TP_LWPEXIT)) {
-				ATRACE_32("am_close_audio() not empty",
-				    chpptr->acp_flags);
-
-				/* wait for drain to complete */
-				if (cv_wait_sig(&chptr->ch_cv,
-				    &chptr->ch_lock) <= 0) {
-					ATRACE("am_close_audio() signal wakeup",
-					    chptr);
-
-					break;
-				}
-				ATRACE_32("am_close_audio() normal wakeup",
-				    chpptr->acp_flags);
-			}
-			ATRACE_32("am_close_audio() empty", chpptr->acp_flags);
-
-			/* clear the writing flag, for mode switching */
-			chpptr->acp_writing = 0;
-
-			mutex_exit(&chptr->ch_lock);
-		} else {
-			mutex_exit(&chptr->ch_lock);
-			/* release the taskq so it can continue */
-			audio_sup_taskq_resume(stpptr->am_taskq);
-		}
-	}
-
-	/*
-	 * It is possible an old AUDIO_DRAIN, which was interrupted, is
-	 * still outstanding. So just in case, we see of there's an mblk_t
-	 * hanging around. If so then we send it back. The STREAMS head will
-	 * ignore it if appropriate.
-	 *
-	 * The channel is marked as empty. Thus if the process is killed or
-	 * the cv_wait_sig() returns we force the AUDIO_DRAIN ioctl() to
-	 * return. This is okay since we will be flushing any queued up audio
-	 * and closing the STREAMS q.
-	 */
-	mutex_enter(&chptr->ch_lock);
-	chpptr->acp_flags |= AM_CHNL_EMPTY;
-	am_audio_drained(chptr);
-	mutex_exit(&chptr->ch_lock);
-
-	/* wait for queued tasks to clear */
-	audio_sup_taskq_wait(stpptr->am_taskq);
-
-	/* the mode & info struct mean something ONLY when the mode is frozen */
-	audio_sup_taskq_suspend(stpptr->am_taskq);
-
-	mode = stpptr->am_pstate->apm_mode;
-	info =	chptr->ch_info.info;
-
-	/*
-	 * Shutdown play and record. We shut down play first because it could
-	 * take a long time to shut down and we don't want to stop recording
-	 * too soon.
-	 */
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&chptr->ch_lock);
-
-	/* mark the channel closed so we can shut down the STREAMS queue */
-	chpptr->acp_flags &= ~AM_CHNL_OPEN;
-
-	/*
-	 * We shutdown the device if in COMPAT mode or for multi-stream
-	 * Codecs, or if this is the last playing/recording stream.
-	 */
-	if (was_writing) {
-		if (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC ||
-		    stpptr->am_out_chs == 1) {
-			ATRACE("am_close_audio() stopping play", statep);
-			mutex_exit(&statep->as_lock);
-			tmp_stream = chptr->ch_info.ch_number;
-			mutex_exit(&chptr->ch_lock);
-			am_ad_stop_play(statep, stpptr, ad_infop, tmp_stream);
-			mutex_enter(&statep->as_lock);
-			mutex_enter(&chptr->ch_lock);
-			info->play.active = 0;
-			hw_info->play.active = 0;
-			info->play.pause = 0;
-
-			/* make sure a mode switch wakes up */
-			mutex_enter(&stpptr->am_mode_lock);
-			cv_signal(&stpptr->am_mode_cv);
-			mutex_exit(&stpptr->am_mode_lock);
-		}
-
-		/* clear the writing flag, for mode switching */
-		chpptr->acp_writing = 0;
-	}
-	if (was_reading) {
-		if (mode == AM_COMPAT_MODE || codec_type == AM_MS_CODEC ||
-		    stpptr->am_in_chs == 1) {
-			ATRACE("am_close_audio() stopping record", statep);
-			mutex_exit(&statep->as_lock);
-			tmp_stream = chptr->ch_info.ch_number;
-			mutex_exit(&chptr->ch_lock);
-			am_ad_stop_record(statep, stpptr, ad_infop, tmp_stream);
-			mutex_enter(&statep->as_lock);
-			mutex_enter(&chptr->ch_lock);
-			info->record.active = 0;
-			hw_info->record.active = 0;
-			info->record.pause = 0;
-		}
-
-		/* send any recorded data that may still be hanging around */
-		if (chpptr->acp_rec_mp) {
-			info->record.samples += (chpptr->acp_rec_mp->b_wptr -
-			    chpptr->acp_rec_mp->b_rptr) /
-			    (info->record.channels *
-			    (info->record.precision >> AUDIO_PRECISION_SHIFT));
-
-			rec_mp = chpptr->acp_rec_mp;
-			chpptr->acp_rec_mp = NULL;
-
-			mutex_exit(&chptr->ch_lock);
-			mutex_exit(&statep->as_lock);
-			putnext(RD(q), rec_mp);
-			mutex_enter(&statep->as_lock);
-			mutex_enter(&chptr->ch_lock);
-
-			chpptr->acp_rec_mp = NULL;
-		}
-
-		/* clear the reading flag, for mode switching */
-		chpptr->acp_reading = 0;
-	}
-
-	/* save the gain and balance for the next open() */
-	if (was_writing) {
-		stpptr->am_pstate->apm_pgain = info->play.gain;
-		stpptr->am_pstate->apm_pbal = info->play.balance;
-	}
-	if (was_reading) {
-		stpptr->am_pstate->apm_rgain = info->record.gain;
-		stpptr->am_pstate->apm_rbal = info->record.balance;
-	}
-
-	/*
-	 * Clear the pid field - keeps this channel from being used while the
-	 * contents are being freed. But save the pid because we need it later.
-	 *
-	 * CAUTION: The taskq must be blocked before the PID is set to 0.
-	 *	Otherwise switching modes can happen in the middle of closing,
-	 *	which results in a bad reference count.
-	 */
-	save_pid = chptr->ch_info.pid;
-	chptr->ch_info.pid = 0;
-
-	mutex_exit(&chptr->ch_lock);
-
-	/* we are modifying mixer global data, so lock the mixer */
-	mutex_enter(&apm_infop->apm_lock);
-
-	ASSERT(stpptr->am_channels > 0);
-	ASSERT(stpptr->am_in_chs >= 0);
-	ASSERT(stpptr->am_out_chs >= 0);
-
-	stpptr->am_channels--;
-
-	if (was_reading) {
-		ASSERT(stpptr->am_in_chs > 0);
-		stpptr->am_in_chs--;
-		if (stpptr->am_in_chs == 0) {
-			/* turn off capture */
-			info->record.active = 0;
-		}
-
-		info->record.open = 0;
-		info->record.waiting = 0;
-		info->record.active = 0;
-		info->record.pause = 0;
-		info->record.samples = 0;
-		info->record.error = 0;
-	}
-
-	if (was_writing) {
-		ASSERT(stpptr->am_out_chs > 0);
-		stpptr->am_out_chs--;
-
-		info->play.open = 0;
-		info->play.waiting = 0;
-		info->play.active = 0;
-		info->play.pause = 0;
-		info->play.samples = 0;
-		info->play.eof = 0;
-		info->play.error = 0;
-	}
-
-	mutex_exit(&apm_infop->apm_lock);
-	mutex_exit(&statep->as_lock);
-
-	/*
-	 * If in MIXER mode the next step for closing the AUDIO channel is to
-	 * fix any AUDIOCTL channels that were pointing to this channel. Even
-	 * if multiple open()s aren't allowed we can still have one channel for
-	 * read and one for write. If there are multiple open()s then there
-	 * could be many channels. Thus we just look for the first AUDIO
-	 * channel with the same PID. If there isn't then we point to the
-	 * hardware.
-	 *
-	 * If in COMPAT mode then we are already pointing to hardware so
-	 * we skip this step.
-	 */
-	max_chs = statep->as_max_chs;
-	mutex_enter(&statep->as_lock);		/* freeze the channels */
-
-	if (mode == AM_MIXER_MODE) {
-		/* first find the first AUDIO channel */
-		ASSERT(alt_info == 0);
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-			/* skip the same and unallocated channels */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr == chptr ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* skip if not AUDIOCTL or different PIDs */
-			if (tchptr->ch_info.dev_type != AUDIO ||
-			    tchptr->ch_info.pid != save_pid) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			alt_info = tchptr->ch_info.info;
-
-			mutex_exit(&tchptr->ch_lock);
-
-			break;
-		}
-		if (alt_info == NULL) {
-			/* no match found, so set to hardware */
-			alt_info = &stpptr->am_hw_info;
-		}
-
-		/* next update the AUDIOCTL chs to point to the correct ch */
-		for (i = 0, tchptr = &statep->as_channels[0];
-		    i < max_chs; i++, tchptr++) {
-			/* skip the same and unallocated channels */
-			mutex_enter(&tchptr->ch_lock);
-			if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    tchptr == chptr ||
-			    tchptr->ch_info.pid == 0) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* skip if not AUDIOCTL or different PIDs */
-			if (tchptr->ch_info.dev_type != AUDIOCTL ||
-			    tchptr->ch_info.pid != save_pid) {
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			ATRACE("am_close_audio() setting AUDIOCTL info",
-			    alt_info);
-			ATRACE_32("am_close_audio() setting AUDIOCTL info, ch",
-			    i);
-
-			/*
-			 * This is the same PID, so fix info pointers. We check
-			 * to see if the AUDIOCTL channel is already pointing
-			 * to alt_info. If so then we don't have to do a thing.
-			 * But if it doesn't then we have to decrement the
-			 * count for where it points and then reset and
-			 * increment the new reference count.
-			 */
-			if (tchptr->ch_info.info == alt_info) {
-				/* the same, so continue */
-				mutex_exit(&tchptr->ch_lock);
-				continue;
-			}
-
-			/* different, so update counters and pointer */
-			((audio_info_t *)tchptr->ch_info.info)->ref_cnt--;
-			tchptr->ch_info.info = alt_info;
-			alt_info->ref_cnt++;
-			mutex_exit(&tchptr->ch_lock);
-		}
-	}
-
-	mutex_exit(&statep->as_lock);
-
-	/* make sure we aren't closing while someone is busy */
-	mutex_enter(&chptr->ch_lock);
-	while (chpptr->acp_busy_cnt) {
-		ATRACE_32("am_close_audio() in putnext(), calling cv_wait()",
-		    chpptr->acp_flags);
-
-		chpptr->acp_flags |= AM_CHNL_SIGNAL_NEEDED;
-
-		/* wait for the count to go to 0 */
-		cv_wait(&chptr->ch_cv, &chptr->ch_lock);
-		ATRACE("am_close_audio() putnext() cv wakeup", chptr);
-	}
-
-	/*
-	 * Mark that qprocsoff() has been called, even though technically it
-	 * hasn't. However, it will eliminate the race condition between
-	 * releasing the lock and the qprocsoff() below.
-	 */
-	chpptr->acp_flags |= AM_CHNL_QPROCSOFF;
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_close_audio() flushing q", chptr);
-	flushq(RD(q), FLUSHALL);
-
-	/* unschedule the queue */
-	ATRACE("am_close_audio() qprocsoff()", chptr);
-	qprocsoff(q);
-
-	/*
-	 * Remove the private data from the q, AFTER turning the q off.
-	 * If this is done before qprocsoff() then any STREAMS call
-	 * between these two would find bad qptr data and panic.
-	 */
-	audio_sup_free_qptr(q);
-
-	/* we have to reget this because it could have changed */
-	info =	chptr->ch_info.info;
-
-	/* take care of references to the audio state structure */
-	mutex_enter(&statep->as_lock);
-	if (info->ref_cnt <= 1) {
-		/*
-		 * Need to free the buffer. We don't need to lock because
-		 * only this thread can now be using this channel.
-		 */
-		ASSERT(info != apm_infop->apm_ad_state);
-		kmem_free(info, sizeof (audio_info_t));
-	} else {
-		info->ref_cnt--;
-	}
-	mutex_exit(&statep->as_lock);
-
-	/* release the taskq AFTER we don't care about the mode */
-	audio_sup_taskq_resume(stpptr->am_taskq);
-
-	/* free the sample rate conversion routine buffers */
-	if (codec_type == AM_TRAD_CODEC) {
-		if (was_writing) {
-			ad_infop->ad_play.ad_conv->ad_src_exit(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_PLAY);
-		}
-		if (was_reading) {
-			ad_infop->ad_record.ad_conv->ad_src_exit(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_RECORD);
-		}
-	}
-
-	/*
-	 * Tell the Audio Driver to free up any channel config info.
-	 * This may block, which is why we do it without any locks held
-	 * or the taskq frozen.
-	 */
-	am_ad_teardown(statep, stpptr, ad_infop, chptr->ch_info.ch_number, dir);
-	ATRACE("am_close_audio() ad_teardown() returned", 0);
-
-	mutex_enter(&chptr->ch_lock);
-
-	/* free all the buffers */
-	if (chpptr->acp_play_samp_buf) {
-		kmem_free(chpptr->acp_play_samp_buf, chpptr->acp_psb_size);
-		chpptr->acp_play_samp_buf = NULL;
-		chpptr->acp_psb_size = 0;
-	}
-	if (chpptr->acp_ch_psrc1) {
-		kmem_free(chpptr->acp_ch_psrc1, chpptr->acp_ch_psrc_siz);
-		chpptr->acp_ch_psrc1 = NULL;
-	}
-	if (chpptr->acp_ch_psrc2) {
-		kmem_free(chpptr->acp_ch_psrc2, chpptr->acp_ch_psrc_siz);
-		chpptr->acp_ch_psrc2 = NULL;
-	}
-	chpptr->acp_ch_psrc_siz = 0;
-	if (chpptr->acp_ch_pconv1) {
-		kmem_free(chpptr->acp_ch_pconv1, chpptr->acp_ch_pconv_siz);
-		chpptr->acp_ch_pconv1 = NULL;
-	}
-	if (chpptr->acp_ch_pconv2) {
-		kmem_free(chpptr->acp_ch_pconv2, chpptr->acp_ch_pconv_siz);
-		chpptr->acp_ch_pconv2 = NULL;
-	}
-	chpptr->acp_ch_pconv_siz = 0;
-	if (chpptr->acp_ch_rsrc1) {
-		kmem_free(chpptr->acp_ch_rsrc1, chpptr->acp_ch_rsrc_siz);
-		chpptr->acp_ch_rsrc1 = NULL;
-	}
-	if (chpptr->acp_ch_rsrc2) {
-		kmem_free(chpptr->acp_ch_rsrc2, chpptr->acp_ch_rsrc_siz);
-		chpptr->acp_ch_rsrc2 = NULL;
-	}
-	chpptr->acp_ch_rsrc_siz = 0;
-	if (chpptr->acp_ch_rconv1) {
-		kmem_free(chpptr->acp_ch_rconv1, chpptr->acp_ch_rconv_siz);
-		chpptr->acp_ch_rconv1 = NULL;
-	}
-	if (chpptr->acp_ch_rconv2) {
-		kmem_free(chpptr->acp_ch_rconv2, chpptr->acp_ch_rconv_siz);
-		chpptr->acp_ch_rconv2 = NULL;
-	}
-	chpptr->acp_ch_rconv_siz = 0;
-
-	ASSERT(chpptr->acp_rec_mp == NULL);
-
-	/* send the close signal */
-	mutex_exit(&chptr->ch_lock);
-	am_send_signal(statep, stpptr);
-	mutex_enter(&chptr->ch_lock);
-
-	kmem_free(chpptr, sizeof (*chpptr));
-	chptr->ch_private = NULL;
-
-	/* wait until the very end to flush */
-	ATRACE("am_close_audio() flushing messages", chptr);
-	audio_sup_flush_audio_data(chptr);
-
-	/* audio_sup_free_ch() requires the info ptr to be NULLed */
-	chptr->ch_info.info = NULL;
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_close_audio() calling audio_free_ch()", chptr);
-	if (audio_sup_free_ch(chptr) == AUDIO_FAILURE) {
-		/* not much we can do if this fails */
-		ATRACE("am_close_audio() audio_sup_free_ch() failed", chptr);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "close_audio() audio_sup_free_ch() error");
-	}
-
-	ATRACE("am_close_audio() successful", 0);
-	return (AUDIO_SUCCESS);
-
-}	/* am_close_audio() */
-
-/*
- * am_close_audioctl()
- *
- * Description:
- *	Close a minor device, returning the minor number to the pool so that
- *	open(2) may use it again.
- *
- *	NOTE: When ch_info.info is set and ref_cnt is changed statep->as_lock
- *		is used and the taskq MUST be frozen. Otherwise a mode switch
- *		will mess things up.
- *
- * Arguments:
- *	queue_t		*q	Pointer to the read queue
- *	int		flag	File status flag
- *	cred_t		*credp	Pointer to the user's credential structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Successfully closed the device
- *	errno			Error number for failed close()
- */
-/*ARGSUSED*/
-int
-am_close_audioctl(queue_t *q, int flag,  cred_t *credp)
-{
-	audio_info_t		*info;
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-
-	ATRACE("in am_close_audioctl()", chptr);
-	ATRACE_32("am_close_audioctl() channel number",
-	    chptr->ch_info.ch_number);
-
-	ASSERT(q == chptr->ch_qptr);
-
-	/* mark the channel as in the process of being closed */
-	mutex_enter(&chptr->ch_lock);
-	chpptr->acp_flags |= AM_CHNL_CLOSING;
-	mutex_exit(&chptr->ch_lock);
-
-	/* wait for queued tasks to clear */
-	audio_sup_taskq_wait(stpptr->am_taskq);
-
-	/* we now need everything stable, so freeze the taskq */
-	audio_sup_taskq_suspend(stpptr->am_taskq);
-
-	/* mark the channel as being closed */
-	mutex_enter(&chptr->ch_lock);
-	chpptr->acp_flags = 0;
-
-	/*
-	 * Clear the pid field - keeps this channel from being used while the
-	 * contents are being freed.
-	 *
-	 * CAUTION: The taskq must be blocked before the PID is set to 0.
-	 *	Otherwise switching modes can happen in the middle of closing,
-	 *	which results in a bad reference count.
-	 */
-	chptr->ch_info.pid = 0;
-
-	/* make sure we aren't closing while someone is busy */
-	while (chpptr->acp_busy_cnt) {
-		ATRACE_32("am_close_audioctl() in putnext(), calling cv_wait()",
-		    chpptr->acp_flags);
-
-		chpptr->acp_flags |= AM_CHNL_SIGNAL_NEEDED;
-
-		/* wait for the count to go to 0 */
-		cv_wait(&chptr->ch_cv, &chptr->ch_lock);
-		ATRACE("am_close_audioctl() putnext() cv wakeup", chptr);
-	}
-
-	/*
-	 * Mark that qprocsoff() has been called, even though technically it
-	 * hasn't. However, it will eliminate the race condition between
-	 * releasing the lock and the qprocsoff() below.
-	 */
-	chpptr->acp_flags |= AM_CHNL_QPROCSOFF;
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_close_audioctl() flushing q", chptr);
-	flushq(RD(q), FLUSHALL);
-
-	/* unschedule the queue */
-	ATRACE("am_close_audioctl() qprocsoff()", chptr);
-	qprocsoff(q);
-
-	/*
-	 * Remove the private data from the q, AFTER turning the q off.
-	 * If this is done before qprocsoff() then any STREAMS call
-	 * between these two would find bad qptr data and panic.
-	 */
-	audio_sup_free_qptr(q);
-
-	/* we are modifying mixer global data, so lock the mixer */
-	mutex_enter(&statep->as_lock);
-	mutex_enter(&apm_infop->apm_lock);
-
-	ASSERT(stpptr->am_channels > 0);
-	ASSERT(stpptr->am_in_chs >= 0);
-	ASSERT(stpptr->am_out_chs >= 0);
-
-	stpptr->am_channels--;
-
-	mutex_exit(&apm_infop->apm_lock);
-	mutex_exit(&statep->as_lock);
-
-	info = chptr->ch_info.info;
-
-	/*
-	 * Unlike AUDIO channels the reference count can never be 1. That's
-	 * because AUDIOCTL channels in COMPAT mode always increment the
-	 * count, as they do in MIXER mode when the process doesn't have an
-	 * AUDIO channel open as well. If in MIXER mode with an AUDIO channel
-	 * in the same process then the AUDIO channels reference count is
-	 * used, not the hardware.
-	 */
-	mutex_enter(&statep->as_lock);
-	info->ref_cnt--;
-	ASSERT(info->ref_cnt);
-	mutex_exit(&statep->as_lock);
-
-	/* release the taskq AFTER we don't care about the mode */
-	audio_sup_taskq_resume(stpptr->am_taskq);
-
-#ifdef DEBUG
-	mutex_enter(&chptr->ch_lock);
-	ASSERT(chpptr->acp_play_samp_buf == NULL);
-	ASSERT(chpptr->acp_ch_psrc1 == NULL);
-	ASSERT(chpptr->acp_ch_psrc2 == NULL);
-	ASSERT(chpptr->acp_ch_pconv1 == NULL);
-	ASSERT(chpptr->acp_ch_pconv2 == NULL);
-	ASSERT(chpptr->acp_ch_rsrc1 == NULL);
-	ASSERT(chpptr->acp_ch_rsrc2 == NULL);
-	ASSERT(chpptr->acp_ch_rconv1 == NULL);
-	ASSERT(chpptr->acp_ch_rconv2 == NULL);
-	ASSERT(chpptr->acp_rec_mp == NULL);
-	mutex_exit(&chptr->ch_lock);
-#endif
-
-	/* send the close signal */
-	am_send_signal(statep, stpptr);
-
-	mutex_enter(&chptr->ch_lock);
-
-	kmem_free(chpptr, sizeof (*chpptr));
-	chptr->ch_private = NULL;
-
-	/* audio_sup_free_ch() requires the info ptr to be NULLed */
-	chptr->ch_info.info = NULL;
-
-	mutex_exit(&chptr->ch_lock);
-
-	ATRACE("am_close_audioctl() calling audio_free_ch()", chptr);
-	if (audio_sup_free_ch(chptr) == AUDIO_FAILURE) {
-		/* not much we can do if this fails */
-		ATRACE("am_close_audioctl() audio_sup_free_ch() failed", chptr);
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-		    "close_audioctl() audio_sup_free_ch() error");
-	}
-
-	ATRACE("am_close_audioctl() successful", 0);
-	return (AUDIO_SUCCESS);
-
-}	/* am_close_audioctl() */
-
-/*
- * am_restore_state()
- *
- * Description:
- *	Restore the device's hardware state and restart playing and recording.
- *	If am_save_state() was called then we have the taskq frozen. Otherwise
- *	it isn't and we have to freeze it. That way hot unplug and hot plug
- *	events won't mess with open(), close(), or ioctl().
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instance's state
- *	audio_apm_info_t	*apm_infop	Ptr to the APM's state info
- *	int			dir		Direction to restore
- *
- * Returns:
- *	AUDIO_SUCCESS			State restored and restarted
- *	AUDIO_FAILURE			State not restored or restart failed
- */
-int
-am_restore_state(audio_state_t *statep, audio_apm_info_t *apm_infop, int dir)
-{
-	am_ad_info_t		*ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	audio_info_t		*hw_info;
-	audiohdl_t		handle = AUDIO_STATE2HDL(statep);
-	uint_t			pgain;		/* play gain to set h/w */
-	uint_t			pbalance;	/* play balance to set h/w */
-	uint_t			rgain;		/* record gain to set h/w */
-	uint_t			rbalance;	/* record balance to set h/w */
-	int			doread;		/* device supports record */
-	int			dowrite;	/* device supports play */
-	int			psr;
-	int			rsr;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE("in am_restore_state()", statep);
-
-	ad_infop = apm_infop->apm_ad_infop;
-	stpptr = apm_infop->apm_private;
-	hw_info = &stpptr->am_hw_info;
-
-	psr	=  hw_info->play.sample_rate;
-	pgain 	= hw_info->play.gain;
-	pbalance = hw_info->play.balance;
-	rsr 	= hw_info->record.sample_rate;
-	rgain 	= hw_info->record.gain;
-	rbalance = hw_info->record.balance;
-
-	/* figure out the direction, am_attach() already did the sanity ck */
-	dowrite =
-	    (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_PLAY) &&
-	    (dir & AUDIO_PLAY);
-	doread =
-	    (ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_RECORD) &&
-	    (dir & AUDIO_RECORD);
-	if (!doread && !dowrite) {
-		ATRACE("am_restore_state() nothing to do", dir);
-		mutex_enter(&stpptr->am_ad_lock);
-		stpptr->am_ad_in &= ~AM_APM_FREEZE;
-		mutex_exit(&stpptr->am_ad_lock);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* freeze the taskq */
-	audio_sup_taskq_suspend(stpptr->am_taskq);
-
-	mutex_enter(&stpptr->am_ad_lock);
-
-	/*
-	 * Set state saved in preparation for restore. We set this because it
-	 * is possible that am_save_state() was not called first, so this
-	 * simulates this call. Don't both with a check if it is set, this
-	 * wastes more time than just setting.
-	 */
-	stpptr->am_ad_in |= AM_APM_FREEZE;
-
-	mutex_exit(&stpptr->am_ad_lock);
-
-	ASSERT(doread || dowrite);
-
-	ATRACE("am_restore_state() dowrite", dowrite);
-	ATRACE("am_restore_state() doread", doread);
-
-	/*
-	 * CAUTION: Keep the calls the same as in am_attach(). There may be
-	 *	order dependencies and once the audio driver works we don't
-	 *	want to break it if we change the order.
-	 *
-	 *	Set the play format and gain, if present.
-	 */
-	if (dowrite) {
-		if (am_set_format(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-		    AUDIO_PLAY, psr, stpptr->am_hw_pchs, stpptr->am_hw_pprec,
-		    stpptr->am_hw_penc, AM_FORCE, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN, "am_restore_state() "
-			    "couldn't set play data format: %d %d %d %d",
-			    hw_info->play.sample_rate, stpptr->am_hw_pchs,
-			    stpptr->am_hw_pprec, stpptr->am_hw_penc);
-
-			goto error;
-		}
-
-		if (am_set_gain(statep, apm_infop, stpptr->am_hw_pchs,
-		    pgain, pbalance, AUDIO_PLAY, AM_SET_CONFIG_BOARD,
-		    AM_FORCE, AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN,
-			    "am_restore_state() couldn't set play gain");
-
-			goto error;
-		}
-	}
-
-	/* set the record format and gain, if present */
-	if (doread) {
-		if (am_set_format(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-		    AUDIO_RECORD, rsr, stpptr->am_hw_rchs, stpptr->am_hw_rprec,
-		    stpptr->am_hw_renc, AM_FORCE, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN,
-			    "am_restore_state() "
-			    "couldn't set record data format: %d %d %d %d",
-			    hw_info->record.sample_rate, stpptr->am_hw_rchs,
-			    stpptr->am_hw_rprec, stpptr->am_hw_renc);
-
-			goto error;
-		}
-
-		/* set the gains */
-		if (am_set_gain(statep, apm_infop, stpptr->am_hw_rchs,
-		    rgain, rbalance, AUDIO_RECORD, AM_SET_CONFIG_BOARD,
-		    AM_FORCE, AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN,
-			    "am_restore_state() couldn't set record gain");
-
-			goto error;
-		}
-	}
-
-	/* now set the ports, monitor gain, etc. */
-	if (dowrite) {
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_PORT, AUDIO_PLAY,
-		    hw_info->play.port, NULL, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN,
-			    "am_restore_state() couldn't set play port: 0x%x",
-			    ad_infop->ad_defaults->play.port);
-
-			goto error;
-		}
-	}
-
-	if (doread) {
-		if (am_ad_set_config(statep, stpptr, ad_infop,
-		    AM_SET_CONFIG_BOARD, AM_SET_PORT, AUDIO_RECORD,
-		    hw_info->record.port, NULL, AM_NO_SERIALIZE) ==
-		    AUDIO_FAILURE) {
-			audio_sup_log(handle, CE_WARN,
-			    "am_restore_state() couldn't set record port: 0x%x",
-			    ad_infop->ad_defaults->record.port);
-
-			goto error;
-		}
-	}
-
-	if ((ad_infop->ad_defaults->hw_features & AUDIO_HWFEATURE_IN2OUT) &&
-	    am_ad_set_config(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-	    AM_SET_MONITOR_GAIN, AUDIO_BOTH, hw_info->monitor_gain, NULL,
-	    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-		audio_sup_log(handle, CE_WARN,
-		    "am_restore_state() couldn't set monitor gain: 0x%x",
-		    ad_infop->ad_defaults->monitor_gain);
-
-		goto error;
-	}
-
-	if (dowrite && am_ad_set_config(statep, stpptr, ad_infop,
-	    AM_SET_CONFIG_BOARD, AM_OUTPUT_MUTE, AUDIO_PLAY,
-	    hw_info->output_muted, NULL, AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-		audio_sup_log(handle, CE_WARN,
-		    "am_restore_state() couldn't set output muted: 0x%x",
-		    ad_infop->ad_defaults->output_muted);
-
-		goto error;
-	}
-
-	if (doread && (ad_infop->ad_assist_flags & AM_ASSIST_MIC) &&
-	    am_ad_set_config(statep, stpptr, ad_infop, AM_SET_CONFIG_BOARD,
-	    AM_MIC_BOOST, AUDIO_RECORD,
-	    (ad_infop->ad_add_mode & AM_ADD_MODE_MIC_BOOST), NULL,
-	    AM_NO_SERIALIZE) == AUDIO_FAILURE) {
-		audio_sup_log(handle, CE_WARN,
-		    "am_restore_state() couldn't set mic boost: 0x%x",
-		    ad_infop->ad_add_mode);
-
-		goto error;
-	}
-
-	/*
-	 * Restart play and record, if there are any apps that have the
-	 * device open for the direction. If not then we don't waste time
-	 * restarting.
-	 *
-	 * It is legal for a play restart calls to fail. This can happen
-	 * when there's no audio to play. If it does fail due to no audio
-	 * then we are already stopped and the active flags should be cleared.
-	 * If play is restarted then the active flags should be set. We don't
-	 * check because there is a race condition and with this routine
-	 * and am_wsvc(). The flags will get set correctly very quickly, so
-	 * there's no reason to try to work around this race, which would not
-	 * be easy.
-	 *
-	 * The same is not true for record. If we start record for no recording
-	 * apps then the mixer will turn off the record later after the 1st
-	 * interrupt.
-	 */
-	if (ad_infop->ad_codec_type == AM_MS_CODEC) {
-		audio_ch_t	*chptr;
-		am_ch_private_t	*chpptr;
-		int		i;
-		int		max_chs = statep->as_max_chs;
-		int		restore_dir = 0;
-
-		restore_dir = doread | dowrite;
-		ATRACE_32("am_restore_state() restore_dir", restore_dir);
-
-		ATRACE("am_restore_state() restarting MS", 0);
-		for (i = 0,  chptr = &statep->as_channels[0]; i < max_chs;
-		    i++, chptr++, restore_dir = 0) {
-
-			/* lock the channel before we check it out */
-			mutex_enter(&chptr->ch_lock);
-
-			/* skip non-AUDIO and unallocated channels */
-			if (!(chptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-			    chptr->ch_info.dev_type != AUDIO ||
-			    chptr->ch_info.pid == 0) {
-				mutex_exit(&chptr->ch_lock);
-				continue;
-			}
-
-			/* make sure this channel is writing */
-			chpptr = chptr->ch_private;
-
-			/*
-			 * Figure which to restart for this channel. We do it
-			 * this way so we can free the mutex fore calling the
-			 * device's start routines.
-			 */
-			if (dowrite && chpptr->acp_writing) {
-				restore_dir |= AUDIO_PLAY;
-			}
-			if (doread && chpptr->acp_reading) {
-				restore_dir |= AUDIO_RECORD;
-			}
-
-			mutex_exit(&chptr->ch_lock);
-
-			if (restore_dir & AUDIO_PLAY) {
-				ATRACE("am_restore_state() MS restart play",
-				    chptr);
-				/*
-				 * Don't set active flags because they
-				 * should already be set.
-				 */
-				(void) am_ad_start_play(statep, stpptr,
-				    ad_infop, i, AM_NO_SERIALIZE);
-			}
-
-			if (restore_dir & AUDIO_RECORD) {
-				ATRACE("am_restore_state() MS restart record",
-				    chptr);
-				/*
-				 * Don't set active flags because they
-				 * should already be set.
-				 */
-				(void) am_ad_start_record(statep, stpptr,
-				    ad_infop, i, AM_NO_SERIALIZE);
-			}
-		}
-	} else {
-		mutex_enter(&statep->as_lock);
-		if (dowrite) {
-			/* do restart */
-			mutex_exit(&statep->as_lock);
-			ATRACE_32("am_restore_state() restarting play TRAD",
-			    stpptr->am_out_chs);
-
-			/*
-			 * We don't bother with the individual channel or
-			 * hardware active flags because they should already
-			 * be set.
-			 */
-			(void) am_ad_start_play(statep, stpptr, ad_infop,
-			    AM_SET_CONFIG_BOARD, AM_NO_SERIALIZE);
-			mutex_enter(&statep->as_lock);
-		}
-
-		if (doread) {
-			/* do restart */
-			mutex_exit(&statep->as_lock);
-			ATRACE_32("am_restore_state() restarting record TRAD",
-			    stpptr->am_in_chs);
-
-			/*
-			 * We don't bother with the individual channel or
-			 * hardware active flags because they should already
-			 * be set.
-			 */
-			(void) am_ad_start_record(statep, stpptr, ad_infop,
-			    AM_SET_CONFIG_BOARD, AM_NO_SERIALIZE);
-		} else {
-			mutex_exit(&statep->as_lock);
-		}
-	}
-
-	ATRACE("am_restore_state() done - success", statep);
-
-	rc = AUDIO_SUCCESS;
-
-error:
-	/* restore error, thaw that taskq and free flags */
-	audio_sup_taskq_resume(stpptr->am_taskq);
-
-	mutex_enter(&stpptr->am_ad_lock);
-
-	stpptr->am_ad_in &= ~AM_APM_FREEZE;
-
-	/* wake up a blocked driver access, just in case */
-	cv_signal(&stpptr->am_ad_cv);
-
-	mutex_exit(&stpptr->am_ad_lock);
-
-	ATRACE_32("am_restore_state() returning", rc);
-
-	return (rc);
-
-}	/* am_restore_state() */
-
-/*
- * am_save_state()
- *
- * Description:
- *	Flag that we are frozen. This stops calls into the audio driver,
- *	except for ioctl()s.
- *
- *	We don't care about which direction. Both act the same.
- *
- *	There is a very tiny window of opportunity between taskq_wait()
- *	and the mutex_enter() where in theory we could get another task added
- *	to the queue, would execute, and potentially cause problems.
- *
- * Arguments:
- *	audio_state_t		*statep		Ptr to the dev instance's state
- *	audio_apm_info_t	*apm_infop	Ptr to the APM's state info
- *	int			dir		Direction to save
- *
- * Returns:
- *	AUDIO_SUCCESS			State restored and restarted
- *	AUDIO_FAILURE			State not restored or restart failed
- */
-/*ARGSUSED*/
-int
-am_save_state(audio_state_t *statep, audio_apm_info_t *apm_infop, int dir)
-{
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-
-	ATRACE("in am_save_state()", statep);
-
-	audio_sup_taskq_wait(stpptr->am_taskq);
-
-	mutex_enter(&stpptr->am_ad_lock);
-
-	/* flag that we're frozen */
-	stpptr->am_ad_in |= AM_APM_FREEZE;
-
-	mutex_exit(&stpptr->am_ad_lock);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_save_state() */
-
-/*
- * The private main routines for this file.
- */
-
-/*
- * am_rput()
- *
- * Description:
- *	We have this here just for symmetry. There aren't any modules/drivers
- *	below this, so this should never be called. But just in case, we
- *	return.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *	mblk_t		*mp	Ptr to the msg block being passed to the queue
- *
- * Returns:
- *	0			Always returns 0
- */
-/*ARGSUSED*/
-static int
-am_rput(queue_t *q, mblk_t *mp)
-{
-	ATRACE("in am_rput()", q);
-
-	ATRACE("am_rput() returning 0", q);
-
-	freemsg(mp);
-
-	return (0);
-
-}	/* am_rput() */
-
-/*
- * am_rsvc()
- *
- * Description:
- *	We have this here just for symmetry. There aren't any modules/drivers
- *	below this, so this should never be called. But just in case, we
- *	return
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *
- * Returns:
- *	0			Always returns 0
- */
-/*ARGSUSED*/
-static int
-am_rsvc(queue_t *q)
-{
-	mblk_t		*mp;
-
-	ATRACE("in am_rsvc()", q);
-
-	/* we always have to drain the queue */
-	while ((mp = getq(q)) != NULL) {
-		freemsg(mp);
-	}
-
-	ATRACE("am_rsvc() returning 0", q);
-
-	return (0);
-
-}	/* am_rsvc() */
-
-/*
- * am_wput()
- *
- * Description:
- *	All messages to the mixer arrive here. We don't support very many
- *	messages.
- *		M_DATA		Passed on to the write svc() routine
- *		M_IOCTL		Calls am_wioctl() for further processing
- *		M_IOCDATA	Calls am_wiocdata() for further processing
- *		M_FLUSH		Flushes the input and/or output queues
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *	mblk_t		*mp	Ptr to the msg block being passed to the queue
- *
- * Returns:
- *	0			Always returns 0
- */
-static int
-am_wput(queue_t *q, mblk_t *mp)
-{
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_state_t		*statep = chptr->ch_statep;
-	int			error = EIO;
-
-	ATRACE("in am_wput()", q);
-
-	ASSERT(chptr);
-
-	/* figure out what kind of message we've got */
-	ATRACE_32("am_wput() type", mp->b_datap->db_type);
-	switch (mp->b_datap->db_type) {
-	case M_FLUSH:
-		ATRACE("am_wput() FLUSH", chptr);
-
-		ASSERT(WR(q) == q);
-
-		am_flush(q, mp);
-		return (0);
-
-	case M_IOCTL:
-		ATRACE("am_wput() IOCTL", chptr);
-		return (am_wioctl(q, mp, chptr));
-
-	case M_IOCDATA:
-		ATRACE("am_wput() IOCDATA", chptr);
-		return (am_wiocdata(q, mp, chptr));
-
-	case M_DATA:
-		ATRACE("am_wput() DATA", chptr);
-		/* make sure the write is on an AUDIO channel */
-		mutex_enter(&chptr->ch_lock);
-		if (chptr->ch_info.dev_type != AUDIO ||
-		    !((am_ch_private_t *)chptr->ch_private)->acp_writing) {
-
-			/* NOT an AUDIO channel, we don't allow write */
-			ATRACE_32("am_wput() not AUDIO",
-			    chptr->ch_info.dev_type);
-			mutex_exit(&chptr->ch_lock);
-			goto done;
-		}
-		mutex_exit(&chptr->ch_lock);
-		ATRACE("am_wput() putting msg on q", mp);
-
-		/*
-		 * First, concatenate the message. If in mixer mode
-		 * with a traditional Codec we do sample rate conversion
-		 * on the concatenated buffer before we save the data
-		 * for later use.
-		 */
-		if (pullupmsg(mp, -1)) {
-			ATRACE("am_wput() pullupmsg() successful", mp);
-			(void) putq(q, mp);	/* does qenable() */
-		} else {
-			ATRACE("am_wput() pullupmsg() failed", mp);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "wput() pullupmsg() failed, sound lost");
-			freemsg(mp);
-		}
-		ATRACE_32("am_wput() putting msg on q done", 0);
-
-		/* don't break because then we free a msg we've already freed */
-		ATRACE_32("am_wput() returning", 0);
-		return (0);
-
-	default:
-		ATRACE("am_wput() default", chptr);
-		break;
-	}
-
-	/* if we get here there was some kind of error, so send an M_ERROR */
-done:
-	ATRACE("am_wput() done:", chptr);
-
-	if (error) {
-		ATRACE_32("am_wput() error", error);
-		mp->b_datap->db_type = M_ERROR;
-		mp->b_rptr = mp->b_datap->db_base;
-		*(int *)mp->b_rptr = EIO;
-		mp->b_wptr = mp->b_rptr + sizeof (int *);
-		if (mp->b_cont) {
-			freemsg(mp->b_cont);
-			mp->b_cont = NULL;
-		}
-		qreply(q, mp);
-	} else {
-		if (mp) {
-			freemsg(mp);
-		}
-	}
-
-	ATRACE("am_wput() returning", chptr);
-
-	return (0);
-
-}	/* am_wput() */
-
-/*
- * am_wsvc()
- *
- * Description:
- *	Write service routine. By definition, this service routine grabs
- *	all messages from the queue before it returns.
- *
- *	The only message that we ever get is an M_DATA message, which is
- *	audio data. The audio data is converted to the canonical data format.
- *	If we need to sample rate convert then the data is converted.
- *
- *	We also make sure the play DMA engine is running.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *
- * Returns:
- *	0			Always returns 0
- */
-static int
-am_wsvc(queue_t *q)
-{
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	audio_state_t		*statep = chptr->ch_statep;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	audio_info_t		*hw_info = &stpptr->am_hw_info;
-	audio_info_t		*tinfo;
-	mblk_t			*mp;
-	int			EOF_count;
-	int			*orig_data;
-	int			*proc_data;
-	size_t			size;
-	size_t			orig_size;
-	size_t			proc_size;
-
-	ATRACE("in am_wsvc()", q);
-	ATRACE("am_wsvc() chptr", chptr);
-	ASSERT(RD(q) == chptr->ch_qptr);
-	ASSERT(!MUTEX_HELD(&chptr->ch_lock));
-
-	/* we always have to drain the queue */
-	while ((mp = getq(q)) != NULL) {
-		/* this is an AUDIO channel */
-		ATRACE("am_wsvc() processing data", mp);
-
-		/*
-		 * If this is an EOF marker, 0 size write(), we place an
-		 * empty audio data structure on the data list.
-		 */
-		size = mp->b_wptr - mp->b_rptr;
-		if (size == 0) {
-			ATRACE("am_wsvc() EOF message, putting on list", mp);
-			if (audio_sup_save_audio_data(chptr, NULL, 0, NULL,
-			    0) == AUDIO_FAILURE) {
-				audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-				    "am_wsvc() EOF marker lost");
-			}
-			freemsg(mp);
-#ifdef FLOW_CONTROL
-			goto flow;
-#else
-			continue;
-#endif
-		}
-
-		/* not an EOF message, so process it */
-		ATRACE("am_wsvc() calling am_p_process()", size);
-
-		/* don't let mode switch happen while we sample rate convert */
-		mutex_enter(&chptr->ch_lock);
-		mutex_enter(&stpptr->am_mode_lock);
-
-		/* if we are switching modes then return */
-		mutex_enter(&apm_infop->apm_lock);
-		if (stpptr->am_flags & AM_PRIV_SW_MODES) {
-			ATRACE_32("am_wsvc() switching modes",
-			    stpptr->am_flags);
-
-			/* put the message back on the queue */
-			(void) putbq(q, mp);
-
-			/* make sure the queue is off */
-			noenable(WR(q));
-
-			/* for AUDIO_DRAIN */
-			chpptr->acp_flags |= AM_CHNL_MSG_ON_QUEUE;
-
-			mutex_exit(&apm_infop->apm_lock);
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-
-			return (0);
-		}
-		mutex_exit(&apm_infop->apm_lock);
-
-		if (am_p_process(chptr, mp->b_rptr, size, &orig_data,
-		    &orig_size, &proc_data, &proc_size) == AUDIO_FAILURE) {
-			mutex_exit(&stpptr->am_mode_lock);
-			mutex_exit(&chptr->ch_lock);
-
-			ATRACE("am_wsvc() am_p_process() failed", chptr);
-
-			freemsg(mp);
-#ifdef FLOW_CONTROL
-			goto flow;
-#else
-			continue;
-#endif
-		}
-		mutex_exit(&stpptr->am_mode_lock);
-		mutex_exit(&chptr->ch_lock);
-
-		freemsg(mp);
-
-		/* save the audio data */
-		if (audio_sup_save_audio_data(chptr, orig_data, orig_size,
-		    proc_data, proc_size) == AUDIO_FAILURE) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "am_wsvc() data save failed, audio lost");
-#ifdef FLOW_CONTROL
-			goto flow;
-#else
-			continue;
-#endif
-		}
-
-		/*
-		 * Mark the channel as busy and not empty, but only if we
-		 * actually processed the message. If that failed then we
-		 * give up on this message and try again on the next one.
-		 */
-		mutex_enter(&chptr->ch_lock);
-		chpptr->acp_flags &= ~(AM_CHNL_EMPTY|AM_CHNL_ALMOST_EMPTY1|\
-		    AM_CHNL_ALMOST_EMPTY2);
-		mutex_exit(&chptr->ch_lock);
-
-#ifdef FLOW_CONTROL
-flow:
-		/* do we need to do flow control? */
-		mutex_enter(&chptr->ch_lock);
-		if (!(chpptr->acp_flags & AM_CHNL_PFLOW) &&
-		    (audio_sup_get_audio_data_size(chptr) >
-		    AM_MAX_QUEUED_MSGS_SIZE ||
-		    audio_sup_get_audio_data_cnt(chptr) >
-		    AM_MAX_QUEUED_MSGS_CNT)) {
-			/* yes, do flow control */
-			chpptr->acp_flags |= AM_CHNL_PFLOW;
-			mutex_exit(&chptr->ch_lock);
-
-			ATRACE("am_wsvc() flow control enabled, q off", q);
-
-			noenable(q);		/* keep putq() from enabling */
-
-			break;
-		}
-		mutex_exit(&chptr->ch_lock);
-#endif
-	}
-
-	/*
-	 * If !paused make sure the play engine is on.
-	 *
-	 * It is possible during a mode switch that the channel info
-	 * structure can change on us. So we use the channel lock to
-	 * make sure it remains stable.
-	 */
-	mutex_enter(&chptr->ch_lock);
-	tinfo = chptr->ch_info.info;
-
-	/* for AUDIO_DRAIN */
-	chpptr->acp_flags &= ~AM_CHNL_MSG_ON_QUEUE;
-	ATRACE("am_wsvc() starting play eng", statep);
-	if (!tinfo->play.pause) {
-		/*
-		 * Set the active bits before starting play so a switch mode
-		 * will sleep on the CV.
-		 */
-		tinfo = chptr->ch_info.info;
-		tinfo->play.active = 1;
-		hw_info->play.active = 1;
-		tinfo->play.pause = 0;
-		mutex_exit(&chptr->ch_lock);
-		if (am_ad_start_play(statep, stpptr, ad_infop,
-		    chptr->ch_info.ch_number, AM_SERIALIZE) == AUDIO_FAILURE) {
-			/* we don't change pause if failed to start */
-			mutex_enter(&chptr->ch_lock);
-			tinfo = chptr->ch_info.info;
-			tinfo->play.active = 0;
-			hw_info->play.active = 0;
-
-			/*
-			 * If we turn off the hardware then make sure any
-			 * queued up EOF messages go out. This is done in
-			 * am_get_samples(). Hopefully most of the EOFs will
-			 * be caught there. However, especially when in mixer
-			 * mode, it can be difficult to catch them all when
-			 * only EOFs are being written. Doing it again here
-			 * costs very little and we know nothing is lost.
-			 */
-			EOF_count = 0;
-			if (chpptr->acp_EOF[chpptr->acp_EOF_toggle]) {
-				EOF_count +=
-				    chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-				chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-			}
-			AUDIO_TOGGLE(chpptr->acp_EOF_toggle);
-			if (chpptr->acp_EOF[chpptr->acp_EOF_toggle]) {
-				EOF_count +=
-				    chpptr->acp_EOF[chpptr->acp_EOF_toggle];
-				chpptr->acp_EOF[chpptr->acp_EOF_toggle] = 0;
-			}
-			mutex_exit(&chptr->ch_lock);
-
-			for (; EOF_count; EOF_count--) {
-				tinfo->play.eof++;
-				am_send_signal(chptr->ch_statep, stpptr);
-			}
-
-			/*
-			 * It is possible that we tried to start playing
-			 * while in the middle of the audio driver calling
-			 * am_play_shutdown(). Thus the start would reload
-			 * the transfer engine but it would be shut down by
-			 * the ISR after am_play_shutdown() returns. The
-			 * appropriate flags in the audio driver will keep
-			 * the transfer engine from being shut back down.
-			 * However, if a mode switch happens after
-			 * am_play_shutdown() checks the mode switch flag
-			 * then when the ISR calls am_get_audio() the switch
-			 * mode flag will be set and no audio will ever be
-			 * transferred. Thus playing will be deadlocked with
-			 * the mode switch.
-			 *
-			 * The easiest way to expose this is to run showmetv
-			 * with 1894.mov and do insane mode switching. The
-			 * switch modes CV will never wake up, so we go ahead
-			 * and wake it up. The switch mode code will deal
-			 * with a bogus cv_signal().
-			 */
-			mutex_enter(&stpptr->am_mode_lock);
-			cv_signal(&stpptr->am_mode_cv);
-			mutex_exit(&stpptr->am_mode_lock);
-		}
-	} else {
-		mutex_exit(&chptr->ch_lock);
-	}
-	ATRACE("am_wsvc() start play eng ret", statep);
-
-	ATRACE("am_wsvc() returning", chptr);
-
-	return (0);
-
-}	/* am_wsvc() */
-
-/*
- * am_apply_gain_balance()
- *
- * Description:
- *	Apply gain and balance to the canonical audio data buffer.
- *
- * Arguments:
- *	int		*buf		Pointer to the canonical audio data
- *	int		samples		Number of samples
- *	int		channels	MONO or STEREO
- *	int		gain		Gain, 0 - 255
- *	int		balance		Balance, 0 - 64
- *
- * Returns:
- *	void
- */
-void
-am_apply_gain_balance(int *buf, int samples, int channels, int gain,
-    int balance)
-{
-	int		l_gain;
-	int		r_gain;
-
-	ATRACE("in am_apply_gain_balance()", buf);
-	ATRACE_32("am_apply_gain_balance() samples", samples);
-	ATRACE_32("am_apply_gain_balance() channels", channels);
-	ATRACE_32("am_apply_gain_balance() gain", gain);
-	ATRACE_32("am_apply_gain_balance() balance", balance);
-
-	if (channels == AUDIO_CHANNELS_MONO) {
-		l_gain = gain;
-		for (; samples; buf++, samples--) {
-			*buf = (*buf * l_gain) >> AM_MAX_GAIN_SHIFT;
-		}
-	} else {
-		ASSERT(channels == AUDIO_CHANNELS_STEREO);
-		ASSERT((samples % 1) == 0);
-
-		l_gain = r_gain = gain;
-
-		if (balance < AUDIO_MID_BALANCE) {
-			/* leave l gain alone and scale down r gain */
-			r_gain = (r_gain * balance) >> AM_TIMES_32_SHIFT;
-		} else if (balance > AUDIO_MID_BALANCE) {
-			/* leave r gain alone and scale down l gain */
-			l_gain = (l_gain * (64 - balance)) >> AM_TIMES_32_SHIFT;
-		}
-
-		for (; samples; buf += 2, samples -= 2) {
-			buf[0] = (buf[0] * l_gain) >> AM_MAX_GAIN_SHIFT;
-			buf[1] = (buf[1] * r_gain) >> AM_MAX_GAIN_SHIFT;
-		}
-	}
-
-	ATRACE("am_apply_gain_balance() done", buf);
-
-}	/* am_apply_gain_balance() */
-
-/*
- * am_convert_int_mono_stereo()
- *
- * Description:
- *	Convert a buffer between mono and stereo. Both the source and
- *	destination buffers are 32-bit integers. The number of samples
- *	is updated to match the new number of samples.
- *
- *	CAUTION: The calling routine must ensure that the dest is large
- *		enough for the data, or we'll panic.
- *
- * Arguments:
- *	int		*src		Input data buffer
- *	int		*dest		Output data buffer
- *	int		*samples	Ptr to he number of samples to convert
- *	int		src_chs		Input channels
- *	int		dest_chs	Output channels
- *
- * Returns:
- *	void
- */
-void
-am_convert_int_mono_stereo(int *src, int *dest, int *samples, int src_chs,
-    int dest_chs)
-{
-	size_t		size;
-	int		i;
-	int		val;
-
-	ATRACE("in am_convert_int_mono_stereo()", src);
-
-	if (src_chs == dest_chs) {
-		/*
-		 * The same size so no translation needed, just copy.
-		 * size = samples * sizeof (int)
-		 */
-		size = *samples << AM_TIMES_4_SHIFT;
-		bcopy(src, dest, size);
-
-	} else if (src_chs > dest_chs) {
-		/* convert from stereo to mono */
-		*samples >>= AM_TIMES_2_SHIFT;
-		for (i = *samples; i--; ) {
-			/* average the left and right channels */
-			val = *src++;
-			val += *src++;
-			*dest++ = val >> AM_HALF_ENERGY_SHIFT;
-		}
-	} else {
-		ASSERT(src_chs < dest_chs);
-		/* convert from mono to stereo */
-		for (i = *samples; i--; ) {
-			val = *src++;
-			*dest++ = val;
-			*dest++ = val;
-		}
-		*samples <<= AM_TIMES_2_SHIFT;
-	}
-
-	ATRACE("am_convert_int_mono_stereo() done", dest);
-
-}	/* am_convert_int_mono_stereo() */
-
-/*
- * am_convert_to_int()
- *
- * Description:
- *	Convert a buffer of various precisions and encodings into 16-bit
- *	linear PCM stored in a 32-bit int. If the input is unsigned PCM
- *	we convert it to signed PCM while converting it.
- *
- *	CAUTION: The calling routine must ensure that the outbuf is large
- *		enough for the data, or we'll panic.
- *
- * Arguments:
- *	void		*inbuf		Input data buffer
- *	int		*outbuf		Output data buffer
- *	int		samples		The number of samples to convert
- *	int		precision	The precision of the input buffer.
- *	int		encoding	The encoding of the input buffer.
- *	int		flags		Flags, including AM_PRIV_8/16_TRANS
- *
- * Returns:
- *	void
- */
-void
-am_convert_to_int(void *inbuf, int *outbuf, int samples, int precision,
-    int encoding, int flags)
-{
-	int		i;
-
-	ATRACE("in am_convert_to_int()", inbuf);
-
-	if (precision == AUDIO_PRECISION_16) {	/* do the easy case first */
-		int16_t		*src = (int16_t *)inbuf;
-
-		ASSERT(encoding == AUDIO_ENCODING_LINEAR);
-
-		ATRACE_32("am_convert_to_int() 16-Bit", samples);
-
-		if (flags & AM_PRIV_16_TRANS) {
-			for (i = samples; i--; ) {
-				*outbuf++ = (int)*src++ + INT16_MIN;
-			}
-		} else {
-			for (i = samples; i--; ) {
-				*outbuf++ = (int)*src++;
-			}
-		}
-	} else {		/* now the hard case, 8-bit */
-		int16_t		*aptr;
-		int8_t		*src = (int8_t *)inbuf;
-
-		ASSERT(precision == AUDIO_PRECISION_8);
-
-		if (encoding == AUDIO_ENCODING_ULAW) {
-			aptr = _8ulaw2linear16;
-			ATRACE("am_convert_to_int() 8-bit u-law", aptr);
-
-			/*
-			 * Copy the data into the buf. acp_ch_pptr1,
-			 * char -> int.
-			 */
-			for (i = samples; i--; ) {
-				/* the conv. array does the scaling */
-				*outbuf++ = (int)aptr[(unsigned char)*src++];
-			}
-		} else if (encoding == AUDIO_ENCODING_ALAW) {
-			aptr = _8alaw2linear16;
-			ATRACE("am_convert_to_int() 8-bit A-law", aptr);
-
-			/*
-			 * Copy the data into the buf. acp_ch_pptr1,
-			 * char -> int.
-			 */
-			for (i = samples; i--; ) {
-				/* the conv. array does the scaling */
-				*outbuf++ = (int)aptr[(unsigned char)*src++];
-			}
-		} else if (encoding == AUDIO_ENCODING_LINEAR8 ||
-		    (flags & AM_PRIV_8_TRANS)) {
-			/*
-			 * Copy the data into the buffer with a shift to
-			 * make signed.
-			 */
-			for (i = samples; i--; ) {
-				*outbuf++ = (((int)*src++ & AM_CHAR2INT_MASK) -
-				    INT8_MAX) << AM_256_SHIFT;
-			}
-		} else {
-			ASSERT(encoding == AUDIO_ENCODING_LINEAR);
-			/*
-			 * Copy the data into the buf. acp_ch_pptr1,
-			 * char -> int.
-			 */
-			for (i = samples; i--; ) {
-				*outbuf++ = (int)(*src++) << AM_256_SHIFT;
-			}
-		}
-	}
-
-	ATRACE("am_convert_to_int() done", outbuf);
-
-}	/* am_convert_to_int() */
-
-/*
- * am_reprocess()
- *
- * Description:
- *	Process the original data, which am_p_process() created, into sample
- *	rate converted audio.
- *
- *	Unlike am_p_process() we don't have the opportunity to minimize the
- *	sample rate conversion processing because we can't take advantage of
- *	converting between mono and stereo.
- *
- *	It is possible we need to convert original audio that has been
- *	partially played. So see if we may need to fix the processed
- *	data pointers.
- *
- *	CAUTION: This routine can be called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- *	CAUTION: It is not possible to update the sample rate converter in
- *		this routine because it may be called when switching modes
- *		and thus the configuration information may be in transition
- *		and not accurate. Thus the calling routines must ensure the
- *		the converter is ready.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	audio_data_t	*data		Original and new processed data struct
- *
- * Returns:
- *	AUDIO_SUCCESS		Data converted and saved
- *	AUDIO_FAILURE		Data conversion and save failed, audio lost
- */
-int
-am_reprocess(audio_ch_t *chptr, audio_data_t *data)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_ad_src_entry_t	*psrs = ad_infop->ad_play.ad_conv;
-	int			*tmp;
-	int			tmp_size;
-	uint_t			hw_channels = stpptr->am_hw_pchs;
-	int			mode = stpptr->am_pstate->apm_mode;
-	int			orig_samples;
-	int			src_samples;
-	int			*data_start;
-	int			*conv;
-	size_t			orig_size_plus_pb;
-	size_t			orig_size;
-
-	ATRACE("in am_reprocess()", data);
-
-	/* don't let mode switch happen while we sample rate convert */
-	mutex_enter(&stpptr->am_mode_lock);
-
-	/* make sure we need to process the data */
-	if (mode != AM_MIXER_MODE || ad_infop->ad_codec_type != AM_TRAD_CODEC) {
-		/* we don't SRC this data */
-		mutex_exit(&stpptr->am_mode_lock);
-		ATRACE_32("am_reprocess() don't SRC", mode);
-		return (AUDIO_FAILURE);
-	}
-
-	/* figure out the number of samples */
-	orig_samples = data->adata_osize >> AM_TIMES_4_SHIFT;
-	orig_size = orig_samples << AM_INT32_SHIFT;
-	orig_size_plus_pb = orig_size + chpptr->acp_ch_pbuf_size;
-	ATRACE_32("am_reprocess() BUF orig_samples", orig_samples);
-
-	/*
-	 * Make sure we've got good sample rate converter buffers. If we fail
-	 * the calling routine will throw away the audio. There aren't more
-	 * chances to process the audio.
-	 */
-	if (am_update_src_buffer(chptr, orig_samples, hw_channels,
-	    AUDIO_PLAY) == AUDIO_SUCCESS) {
-		ATRACE("am_reprocess() update_src_buffer() okay", 0);
-		src_samples = orig_samples;
-	} else {
-		mutex_exit(&stpptr->am_mode_lock);
-		ATRACE_32("am_reprocess() calling update_src_buffer() failed",
-		    mode);
-		return (AUDIO_FAILURE);
-	}
-	ATRACE_32("am_reprocess() src_samples", src_samples);
-
-	/* Only do extra work if necessary */
-	if (chpptr->acp_ch_pbuf_size > 0) {
-
-		/* Make sure we have good conversion buffers */
-		if (am_update_conv_buffer(chptr, orig_size_plus_pb,
-			AUDIO_PLAY) == AUDIO_FAILURE) {
-			mutex_exit(&stpptr->am_mode_lock);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-				"!process(1) couldn't allocate memory, "
-				"audio lost");
-			return (AUDIO_FAILURE);
-		}
-
-		/* Set conv pointing to pconv2 */
-		conv = chpptr->acp_ch_pconv2;
-
-		/* Set data_start pointing to where incoming data begins */
-		data_start = conv + (chpptr->acp_ch_pbuf_size >>
-			AM_TIMES_4_SHIFT);
-
-		/* We have to copy orig data to data_start */
-		bcopy(data->adata_orig, data_start, orig_size);
-	} else {
-		/* Set conv pointing to orig data */
-		conv = data->adata_orig;
-	}
-
-	tmp = psrs->ad_src_convert(AM_SRC_CHPTR2HDL(chptr), hw_channels,
-	    AUDIO_PLAY, conv, chpptr->acp_ch_psrc1,
-	    chpptr->acp_ch_psrc2, &src_samples);
-
-	ASSERT(src_samples <= (chpptr->acp_ch_psrc_siz >> AM_TIMES_4_SHIFT));
-
-	/* if successful then copy to proc */
-	if (tmp) {
-		/* sizeof (int) = 4 */
-		tmp_size = src_samples << AM_TIMES_4_SHIFT;
-		data->adata_proc = kmem_alloc(tmp_size, KM_NOSLEEP);
-		if (data->adata_proc) {
-			bcopy(tmp, data->adata_proc, tmp_size);
-			data->adata_psize = tmp_size;
-		} else {
-			mutex_exit(&stpptr->am_mode_lock);
-			ATRACE_32("am_reprocess() couldn't allocate new buffer",
-			    0);
-			return (AUDIO_FAILURE);
-		}
-		data->adata_psize = tmp_size;
-		data->adata_peptr = (char *)data->adata_proc + tmp_size;
-	} else {
-		mutex_exit(&stpptr->am_mode_lock);
-		ATRACE_32("am_reprocess() SRC failed", 0);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("am_reprocess() orig", data->adata_orig);
-	ATRACE("am_reprocess() optr", data->adata_optr);
-	ATRACE("am_reprocess() oeptr", data->adata_oeptr);
-	ATRACE_32("am_reprocess() osize", data->adata_osize);
-
-	/* see if we need to fix the processed data pointers */
-	if (data->adata_optr != data->adata_orig) {
-		long		tmp;
-
-		/*
-		 * Make a guess as to where to point to. We make sure we are
-		 * on a 4 byte boundary. That way we don't have to worry
-		 * about being in the middle of a sample.
-		 *
-		 * The equation:
-		 *	(offset of orig data from start)*(length of proc data)
-		 *	--------------------------------------------------------
-		 *		(length of orig data)
-		 */
-		tmp = ((char *)data->adata_optr - (char *)data->adata_orig) *
-		    data->adata_psize / data->adata_osize;
-
-		/*
-		 * tmp is an offset, which must be added to adata_proc to
-		 * get adata_pptr. We mask off adata_pptr so that regardless
-		 * of the format of the data we always are on a sample frame
-		 * boundary.
-		 */
-		data->adata_pptr = (char *)data->adata_proc +
-		    (tmp & ~AM_MISC_MASK);
-	} else {
-		/* on a boundary, so the pointers are easy */
-		data->adata_pptr = data->adata_proc;
-	}
-
-	ATRACE("am_reprocess() proc", data->adata_proc);
-	ATRACE("am_reprocess() pptr", data->adata_pptr);
-	ATRACE("am_reprocess() peptr", data->adata_peptr);
-	ATRACE_32("am_reprocess() psize", data->adata_psize);
-
-	/* we don't need to block mode switching now */
-	mutex_exit(&stpptr->am_mode_lock);
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_reprocess() */
-
-/*
- * am_send_signal()
- *
- * Description:
- *	This routine is used to send signals back to user land processes.
- *
- *	We always create a prototype signal message, but we use dupb() to
- *	actually send up the queue.
- *
- *	NOTE: We don't lock the tchptr because the state is frozen thus
- *		channels can't be allocated or freed. However they can
- *		have their state updated. The worst that can happen is
- *		we miss a channel to send a signal on, which isn't that
- *		bad. And this is only when open()ing or close()ing. Since
- *		those operations send a signal we're covered.
- *
- *	NOTE: This routine must be called with as_lock held.
- *
- * Arguments:
- *	audio_state_t		*statep	Pointer to the device instance's state
- *	am_apm_private_t	*stpptr	Pointer to APM private data
- *
- * Returns:
- *	void
- */
-void
-am_send_signal(audio_state_t *statep, am_apm_private_t *stpptr)
-{
-	audio_ch_t	*tchptr;
-	mblk_t		*mp = stpptr->am_sig_mp;
-	mblk_t		*dup_mp;
-	int		i;
-	int		max_chs;
-
-	ATRACE("in am_send_signal()", statep);
-
-	ASSERT(!MUTEX_HELD(&statep->as_lock));
-
-	/* get the number of chs for this instance */
-	max_chs = statep->as_max_chs;
-
-	ATRACE("am_send_signal() AM_SIGNAL_ALL_CTL", mp);
-
-	/* look for AUDIOCTL channels */
-	for (i = 0, tchptr = &statep->as_channels[0];
-	    i < max_chs; i++, tchptr++) {
-		/* skip unallocated, non-AUDIOCTL and closing channels */
-
-		mutex_enter(&tchptr->ch_lock);
-		if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    tchptr->ch_info.dev_type != AUDIOCTL ||
-		    tchptr->ch_info.pid == 0 ||
-		    (((am_ch_private_t *)tchptr->ch_private)->acp_flags &
-		    AM_CHNL_CLOSING)) {
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-		ATRACE("am_send_signal() tchptr", tchptr);
-		if ((dup_mp = dupb(mp)) == NULL) {
-			ATRACE("am_send_signal() AUDIOCTL "
-			    "couldn't allocate duplicate message", tchptr);
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "signal() couldn't allocate duplicate "
-			    "message to send signal, signal lost");
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-		ATRACE_32("am_send_signal() AM_SIGNAL_ALL_CTL putnext()",
-		    tchptr->ch_info.ch_number);
-		ASSERT((((am_ch_private_t *)tchptr->ch_private)->acp_flags &
-		    AM_CHNL_CLOSING) == 0);
-
-		am_safe_putnext(tchptr, dup_mp);
-
-		ATRACE("am_send_signal() "
-		    "AM_SIGNAL_ALL_CTL putnext() done", dup_mp);
-
-		mutex_exit(&tchptr->ch_lock);
-	}
-
-	ATRACE("am_send_signal() done", statep);
-
-}	/* am_send_signal() */
-
-/*
- * am_update_conv_buffer()
- *
- * Description:
- *	Make sure the conversion to 32-bit linear PCM buffers are large
- *	enough. If not then the ones are allocated. Both buffers are always
- *	set to the same size.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	size_t		size		The size we need
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, not both
- *
- * Returns:
- *	AUDIO_SUCCESS			Buffers okay
- *	AUDIO_FAILURE			Buffer allocation failed
- */
-int
-am_update_conv_buffer(audio_ch_t *chptr, size_t size, int dir)
-{
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	void			*tmp1;
-	void			*tmp2;
-
-	ATRACE("in am_update_conv_buffer()", chptr);
-	ATRACE("am_update_conv_buffer() size", size);
-
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	ASSERT(dir != AUDIO_BOTH);
-
-	if (dir == AUDIO_PLAY) {
-		if (chpptr->acp_ch_pconv_siz &&
-		    size <= chpptr->acp_ch_pconv_siz) {
-			/* buffers large enough */
-			ASSERT(chpptr->acp_ch_pconv1);
-			ASSERT(chpptr->acp_ch_pconv2);
-			return (AUDIO_SUCCESS);
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		if (chpptr->acp_ch_rconv_siz &&
-		    size <= chpptr->acp_ch_rconv_siz) {
-			/* buffers large enough */
-			ASSERT(chpptr->acp_ch_rconv1);
-			ASSERT(chpptr->acp_ch_rconv2);
-			return (AUDIO_SUCCESS);
-		}
-	}
-
-	if ((tmp1 = kmem_zalloc(size, KM_NOSLEEP)) == NULL) {
-		ATRACE("am_update_conv_buffer() kmem_zalloc(1) failed", NULL);
-		return (AUDIO_FAILURE);
-	}
-	if ((tmp2 = kmem_zalloc(size, KM_NOSLEEP)) == NULL) {
-		ATRACE("am_update_conv_buffer() kmem_zalloc(2) failed", NULL);
-		kmem_free(tmp1, size);
-		return (AUDIO_FAILURE);
-	}
-	ATRACE("am_update_conv_buffer() new buffers", size);
-
-	if (dir == AUDIO_PLAY) {
-		ASSERT(size >= chpptr->acp_ch_pbuf_size);
-		if (chpptr->acp_ch_pconv1) {
-			bcopy(chpptr->acp_ch_pconv1, tmp1,
-				chpptr->acp_ch_pbuf_size);
-			kmem_free(chpptr->acp_ch_pconv1,
-			    chpptr->acp_ch_pconv_siz);
-		}
-		if (chpptr->acp_ch_pconv2) {
-			bcopy(chpptr->acp_ch_pconv2, tmp2,
-				chpptr->acp_ch_pbuf_size);
-			kmem_free(chpptr->acp_ch_pconv2,
-			    chpptr->acp_ch_pconv_siz);
-		}
-		chpptr->acp_ch_pconv1 = tmp1;
-		chpptr->acp_ch_pconv2 = tmp2;
-		chpptr->acp_ch_pconv_siz = size;
-	} else {
-		ASSERT(size >= chpptr->acp_ch_rbuf_size);
-		if (chpptr->acp_ch_rconv1) {
-			bcopy(chpptr->acp_ch_rconv1, tmp1,
-				chpptr->acp_ch_rbuf_size);
-			kmem_free(chpptr->acp_ch_rconv1,
-			    chpptr->acp_ch_rconv_siz);
-		}
-		if (chpptr->acp_ch_rconv2) {
-			bcopy(chpptr->acp_ch_rconv2, tmp2,
-				chpptr->acp_ch_rbuf_size);
-			kmem_free(chpptr->acp_ch_rconv2,
-			    chpptr->acp_ch_rconv_siz);
-		}
-		chpptr->acp_ch_rconv1 = tmp1;
-		chpptr->acp_ch_rconv2 = tmp2;
-		chpptr->acp_ch_rconv_siz = size;
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_update_conv_buffer() */
-
-/*
- * am_update_src_buffer()
- *
- * Description:
- *	Make sure the sample rate conversion buffers are large enough. If
- *	not then the ones are allocated. Both buffers are always set to
- *	the same size.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	int		samples		The number of samples to convert
- *	uint_t		hw_channels	Number of hardware channels
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, not both
- *
- * Returns:
- *	AUDIO_SUCCESS			Buffers okay
- *	AUDIO_FAILURE			Buffer allocation failed, buffers 0ed
- */
-int
-am_update_src_buffer(audio_ch_t *chptr, int samples, uint_t hw_channels,
-    int dir)
-{
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_ad_src_entry_t	*srs;
-	audio_info_t		*info = chptr->ch_info.info;
-	size_t			size;
-	void			*tmp1;
-	void			*tmp2;
-
-	ATRACE("in am_update_src_buffer()", chptr);
-	ATRACE_32("am_update_src_buffer() samples", samples);
-
-	ASSERT(MUTEX_HELD(&stpptr->am_mode_lock));
-	ASSERT(MUTEX_HELD(&chptr->ch_lock));
-
-	ASSERT(dir != AUDIO_BOTH);
-
-	if (dir == AUDIO_PLAY) {
-		srs = ad_infop->ad_play.ad_conv;
-		size = srs->ad_src_size(AM_SRC_CHPTR2HDL(chptr),
-		    &info->play, AUDIO_PLAY, samples, hw_channels);
-		if (chpptr->acp_ch_psrc_siz &&
-			size <= chpptr->acp_ch_psrc_siz) {
-			/* buffers large enough */
-			ASSERT(chpptr->acp_ch_psrc1);
-			ASSERT(chpptr->acp_ch_psrc2);
-			return (AUDIO_SUCCESS);
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-		srs = ad_infop->ad_record.ad_conv;
-		size = srs->ad_src_size(AM_SRC_CHPTR2HDL(chptr),
-		    &info->record, AUDIO_RECORD, samples, hw_channels);
-		if (chpptr->acp_ch_rsrc_siz &&
-			size <= chpptr->acp_ch_rsrc_siz) {
-			/* buffers large enough */
-			ASSERT(chpptr->acp_ch_rsrc1);
-			ASSERT(chpptr->acp_ch_rsrc2);
-			return (AUDIO_SUCCESS);
-		}
-	}
-
-	if ((tmp1 = kmem_alloc(size, KM_NOSLEEP)) == NULL) {
-		ATRACE("am_update_src_buffer() kmem_alloc(1) failed", NULL);
-		return (AUDIO_FAILURE);
-	}
-	if ((tmp2 = kmem_alloc(size, KM_NOSLEEP)) == NULL) {
-		ATRACE("am_update_src_buffer() kmem_alloc(2) failed", NULL);
-		kmem_free(tmp1, size);
-		return (AUDIO_FAILURE);
-	}
-
-	if (dir == AUDIO_PLAY) {
-		if (chpptr->acp_ch_psrc1) {
-			kmem_free(chpptr->acp_ch_psrc1,
-			    chpptr->acp_ch_psrc_siz);
-		}
-		if (chpptr->acp_ch_psrc2) {
-			kmem_free(chpptr->acp_ch_psrc2,
-			    chpptr->acp_ch_psrc_siz);
-		}
-		chpptr->acp_ch_psrc1 = tmp1;
-		chpptr->acp_ch_psrc2 = tmp2;
-		chpptr->acp_ch_psrc_siz = size;
-	} else {
-		if (chpptr->acp_ch_rsrc1) {
-			kmem_free(chpptr->acp_ch_rsrc1,
-			    chpptr->acp_ch_rsrc_siz);
-		}
-		if (chpptr->acp_ch_rsrc2) {
-			kmem_free(chpptr->acp_ch_rsrc2,
-			    chpptr->acp_ch_rsrc_siz);
-		}
-		chpptr->acp_ch_rsrc1 = tmp1;
-		chpptr->acp_ch_rsrc2 = tmp2;
-		chpptr->acp_ch_rsrc_siz = size;
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_update_src_buffer() */
-
-/*
- * Private routines used only by this file.
- */
-
-/*
- * am_flush()
- *
- * Description:
- *	Flush the data stream. We handle both play and record. In order
- *	to flush we have to clear out any play buffers so we have to stop
- *	and then restart. This applies to both play and record.
- *
- * Arguments:
- *	queue_t		*q	Pointer to a queue
- *	mblk_t		*mp	Ptr to the msg block being passed to the queue
- *
- * Returns:
- *	void
- */
-static void
-am_flush(queue_t *q, mblk_t *mp)
-{
-	audio_ch_t		*chptr = (audio_ch_t *)
-				    audio_sup_get_qptr_data(q);
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_apm_info_t	*apm_infop = chptr->ch_apm_infop;
-	audio_info_t		*hw_info = apm_infop->apm_ad_state;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_apm_private_t	*stpptr = apm_infop->apm_private;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	audio_info_t		*tinfo;
-	mblk_t			*tmp;
-
-	ATRACE("in am_flush()", q);
-
-	ASSERT(chptr);
-
-	/* are we flushing the play side? */
-	if (*mp->b_rptr & FLUSHW) {
-		ATRACE("am_flush() flushing play side", 0);
-		flushq(q, FLUSHDATA);
-		*mp->b_rptr &= ~FLUSHW;
-
-		mutex_enter(&chptr->ch_lock);
-
-		/* flush accumulated data */
-		audio_sup_flush_audio_data(chptr);
-
-		/*
-		 * Flush the DMA engine and Codec, but only if this
-		 * channel points to the hardware and is an AUDIO
-		 * channel.
-		 */
-		tinfo = chptr->ch_info.info;
-		chpptr->acp_flags |= AM_CHNL_EMPTY;
-		if (chptr->ch_info.dev_type == AUDIO &&
-		    (chptr->ch_info.info == apm_infop->apm_ad_state ||
-		    ad_infop->ad_codec_type == AM_MS_CODEC)) {
-			/* before we can flush the DMA engine we must stop it */
-			mutex_exit(&chptr->ch_lock);
-			am_ad_stop_play(statep, stpptr, ad_infop,
-			    chptr->ch_info.ch_number);
-			mutex_enter(&chptr->ch_lock);
-
-			tinfo->play.active = 0;
-			hw_info->play.active = 0;
-			/* we don't change pause flag, it can be set */
-		}
-
-		/* update the played sample count */
-		if (stpptr->am_pstate->apm_mode == AM_MIXER_MODE &&
-		    ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-			tinfo->play.samples +=
-			    ad_infop->ad_play.ad_conv->ad_src_adjust(
-			    AM_SRC_CHPTR2HDL(chptr), AUDIO_PLAY,
-			    chpptr->acp_psamples_p);
-		} else {
-			tinfo->play.samples += chpptr->acp_psamples_p;
-		}
-		chpptr->acp_psamples_c = 0;
-		chpptr->acp_psamples_f = 0;
-		chpptr->acp_psamples_p = 0;
-
-		/* by definition we are empty */
-		am_audio_drained(chptr);
-		mutex_exit(&chptr->ch_lock);
-
-		ATRACE("am_flush() flushing play done", q);
-	}
-
-	/* now for the record side */
-	if (*mp->b_rptr & FLUSHR) {
-		ATRACE("am_flush() flushing record side", 0);
-
-		/*
-		 * Flush the DMA engine and Codec, but only if this channel
-		 * points to the hardware and is an AUDIO channel.
-		 */
-		if (chptr->ch_info.dev_type == AUDIO) {
-			/*
-			 * We only flush AUDIO channels, there's nothing on
-			 * AUDIOCTL channels to flush.
-			 */
-			mutex_enter(&chptr->ch_lock);
-			tinfo = chptr->ch_info.info;
-			if (tinfo->record.active &&
-			    (chptr->ch_info.info == apm_infop->apm_ad_state ||
-			    ad_infop->ad_codec_type == AM_MS_CODEC)) {
-				/*
-				 * Before we can flush the DMA engine we
-				 * must stop it.
-				 */
-				mutex_exit(&chptr->ch_lock);
-
-				am_ad_stop_record(statep, stpptr, ad_infop,
-				    chptr->ch_info.ch_number);
-
-				/*
-				 * Flush any partially captured data. This
-				 * needs to be done before we restart recording.
-				 */
-				mutex_enter(&chptr->ch_lock);
-				if (chpptr->acp_rec_mp) {
-					tmp = chpptr->acp_rec_mp;
-					chpptr->acp_rec_mp = NULL;
-					mutex_exit(&chptr->ch_lock);
-
-					freemsg(tmp);
-				} else {
-					mutex_exit(&chptr->ch_lock);
-				}
-
-				/* restart the record */
-				if (am_ad_start_record(statep,
-				    apm_infop->apm_private, ad_infop,
-				    chptr->ch_info.ch_number,
-				    AM_SERIALIZE) == AUDIO_FAILURE) {
-					audio_sup_log(AUDIO_STATE2HDL(statep),
-					    CE_WARN, "couldn't restart record "
-					    "after flush");
-					tinfo->record.active = 0;
-					hw_info->record.active = 0;
-				}
-			} else {
-				/*
-				 * Not currently recording but we still
-				 * need to flush any partial buffers.
-				 */
-				if (chpptr->acp_rec_mp) {
-					tmp = chpptr->acp_rec_mp;
-					chpptr->acp_rec_mp = NULL;
-					mutex_exit(&chptr->ch_lock);
-
-					freemsg(tmp);
-				} else {
-					mutex_exit(&chptr->ch_lock);
-				}
-			}
-		}
-
-		/* send the flush back up to the STREAMS head */
-		*mp->b_rptr &= ~FLUSHW;	/* clear the write */
-		qreply(q, mp);
-		mp = NULL;		/* stop freemsg() */
-
-		ATRACE("am_flush() flushing read done", q);
-	}
-
-	if (mp) {
-		freemsg(mp);
-	}
-
-	ATRACE("am_flush() returning", q);
-
-}	/* am_flush() */
-
-/*
- * am_p_process()
- *
- * Description:
- *	Process the message block into canonical form. If in MIXER mode
- *	then we also sample rate convert it to match the hardware. When
- *	done the original data and the converted data, if present, are
- *	saved in an audio_data_t structure. If sample rate conversion
- *	fails we don't worry about it. There is a second chance when
- *	the samples are mixed.
- *
- *	CAUTION: This routine is called from interrupt context, so memory
- *		allocation cannot sleep.
- *
- * Arguments:
- *	audio_ch_t	*chptr		Pointer to this channel's state info
- *	void		*buf		Ptr to the original msg block being
- *					processed
- *	size_t		size		The size of the data buffer in bytes
- *	int		**orig_data	Ptr to the original canonical audio ptr
- *	size_t		*orig_size	Ptr to the original audio size
- *	int		**proc_data	Ptr to the processed canonical audio ptr
- *	size_t		*proc_size	Ptr to the processed audio size
- *
- * Returns:
- *	AUDIO_SUCCESS		Data converted and saved
- *	AUDIO_FAILURE		Data conversion and save failed, audio lost
- */
-static int
-am_p_process(audio_ch_t *chptr, void *buf, size_t size, int **orig_data,
-    size_t *orig_size, int **proc_data, size_t *proc_size)
-{
-	audio_state_t		*statep = chptr->ch_statep;
-	audio_info_t		*info;
-	am_ch_private_t		*chpptr = chptr->ch_private;
-	am_apm_private_t	*stpptr = chptr->ch_apm_infop->apm_private;
-	am_ad_info_t		*ad_infop = chptr->ch_apm_infop->apm_ad_infop;
-	am_ad_src_entry_t	*psrs = ad_infop->ad_play.ad_conv;
-	int			*dest;
-	int			*src;
-	int			*tmp;
-	int			tmp_size;
-	uint_t			channels;
-	uint_t			encoding;
-	uint_t			precision;
-	uint_t			hw_channels;
-	int			i;
-	int			mode;
-	int			orig_samples;
-	int			src_samples;
-	int			val;
-	int			*conv;
-	int			*data_start;
-	size_t			orig_size_plus_pb;
-
-	ATRACE("in am_p_process()", buf);
-	ATRACE("am_p_process() size", size);
-
-	ASSERT(MUTEX_HELD(&stpptr->am_mode_lock));
-
-	/* set after we are frozen */
-	info = chptr->ch_info.info;
-	channels = info->play.channels;
-	encoding = info->play.encoding;
-	precision = info->play.precision;
-	hw_channels = stpptr->am_hw_pchs;
-	mode = stpptr->am_pstate->apm_mode;
-
-	/* figure out the number of samples */
-	orig_samples = size / (precision >> AUDIO_PRECISION_SHIFT);
-	*orig_size = orig_samples << AM_INT32_SHIFT;
-	orig_size_plus_pb = *orig_size + chpptr->acp_ch_pbuf_size;
-	ATRACE_32("am_p_process() BUF orig_size", *orig_size);
-	ATRACE_32("am_p_process() BUF orig_samples", orig_samples);
-
-	/*
-	 * It is possible that an odd sample write has happened. This can
-	 * happen when the system is very busy and partial STREAMS messages
-	 * are created. Or when an application just plain makes a mistake.
-	 * We chop off the partial sample and press on. This does mean it is
-	 * possible that the next buffer will be out of sync. If it was due to
-	 * the STREAMS message being chopped off then hopefully this just
-	 * finish off the messed up buffer and all will be back in sync. If it
-	 * was because the app is in error, then there's not much hope.
-	 *
-	 * This applies only for stereo, mono can be odd.
-	 */
-	if (channels == AUDIO_CHANNELS_STEREO) {
-		if ((orig_samples % AUDIO_CHANNELS_STEREO) != 0) {
-			ATRACE("am_p_process() stereo samples chopped",
-			    orig_samples);
-
-			/* fix */
-			orig_samples = orig_samples -
-			    (orig_samples % AUDIO_CHANNELS_STEREO);
-			*orig_size = orig_samples << AM_INT32_SHIFT;
-			orig_size_plus_pb = *orig_size +
-			    chpptr->acp_ch_pbuf_size;
-
-			ATRACE_32("am_p_process() new BUF orig_size",
-			    *orig_size);
-			ATRACE_32("am_p_process() new BUF orig_samples",
-			    orig_samples);
-		}
-		ATRACE("am_p_process() stereo samples okay", orig_samples);
-	}
-
-	/*
-	 * If we are going to need to do sample rate conversion then make sure
-	 * we've got good buffers.
-	 */
-	if (mode == AM_MIXER_MODE && ad_infop->ad_codec_type == AM_TRAD_CODEC) {
-		/*
-		 * Make sure we've got good sample rate converter buffers.
-		 * However, even if we fail we still proceed. We get a second
-		 * chance later on to do the sample rate conversion.
-		 */
-		if (am_update_src_buffer(chptr, orig_samples, hw_channels,
-		    AUDIO_PLAY) == AUDIO_SUCCESS) {
-			ATRACE("am_p_process() update_src_buffer() okay", 0);
-			src_samples = orig_samples;
-		} else {
-			/* mark the failure */
-			ATRACE_32(
-			    "am_p_process() calling update_src_buffer() failed",
-			    mode);
-			src_samples = 0;
-		}
-	} else {
-		/* setting to 0 also means don't do src */
-		src_samples = 0;
-	}
-	ATRACE_32("am_p_process() src_samples", src_samples);
-
-	/*
-	 * We convert to 16-bit linear in a 32-bit integer only if buf != NULL.
-	 *
-	 * For converting to canonical format or sample rate conversion
-	 * we need several buffers. We use acp_ch_pconv* for conversion and
-	 * acp_ch_pptr* for sample rate conversion. There are two buffers
-	 * for each, which gives us the space to toggle. We need different
-	 * buffers because we could be converting to 32-bit linear PCM the
-	 * same time we are sample rate converting.
-	 */
-	if (am_update_conv_buffer(chptr, orig_size_plus_pb,
-		AUDIO_PLAY) == AUDIO_FAILURE) {
-		audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			"!process(1) couldn't allocate memory, audio lost");
-		return (AUDIO_FAILURE);
-	}
-
-	/* prepare for failure */
-	*proc_data = NULL;
-	*proc_size = 0;
-
-	/*
-	 * Do the conversions. We try to minimize data copies as much as
-	 * possible. There are three possible situations, and how we deal
-	 * with the conversions. We make sure we do the sample rate conversion
-	 * on the minimum amount of audio data
-	 *
-	 * same (mono --> mono or stereo --> stereo)
-	 *	am_convert_to_int(buf)		--> orig data
-	 *	SRC(orig data)			--> proc data
-	 *
-	 * mono --> stereo
-	 *	am_convert_to_int(buf)		--> tmp data
-	 *	inline mono2stereo(tmp data)	--> orig data
-	 *	SRC(tmp data)			--> tmp data
-	 *	inline mono2stereo(tmp data)	--> proc data
-	 *
-	 * stereo --> mono
-	 *	am_convert_to_int(buf)		--> tmp data
-	 *	inline stereo2mono(tmp data)	--> orig data
-	 *	SRC(orig data)			--> proc data
-	 */
-	if (channels == hw_channels) {
-		/* either mono --> mono or stereo --> stereo */
-		ATRACE("am_p_process() channels == hw_channels", 0);
-
-		*orig_data = kmem_alloc(*orig_size, KM_NOSLEEP);
-		if (*orig_data == NULL) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!process(2) couldn't allocate"
-			    " memory, audio lost");
-			return (AUDIO_FAILURE);
-		}
-
-		/* convert to int and place in orig data */
-		am_convert_to_int(buf, *orig_data, orig_samples, precision,
-		    encoding, 0);
-
-		/* SRC the data if we need to */
-		if (src_samples) {
-
-			/* only do extra work if necessary */
-			if (chpptr->acp_ch_pbuf_size > 0) {
-
-				/* set conv pointing to pconv1 */
-				conv = chpptr->acp_ch_pconv1;
-
-				/* set data_start pointing to incoming data */
-				data_start = conv + (chpptr->acp_ch_pbuf_size
-					>> AM_TIMES_4_SHIFT);
-
-				/* we have to copy orig data to data_start */
-				bcopy(*orig_data, data_start, *orig_size);
-			} else {
-				/* set conv pointing to orig data */
-				conv = *orig_data;
-			}
-
-			tmp = psrs->ad_src_convert(AM_SRC_CHPTR2HDL(chptr),
-			    channels, AUDIO_PLAY, conv, chpptr->acp_ch_psrc1,
-			    chpptr->acp_ch_psrc2, &src_samples);
-
-			ASSERT(src_samples <= (chpptr->acp_ch_psrc_siz >>
-				AM_TIMES_4_SHIFT));
-
-			/* if successful then update info */
-			if (tmp) {
-				/* sizeof (int) = 4 */
-				tmp_size = src_samples << AM_TIMES_4_SHIFT;
-				*proc_data = kmem_alloc(tmp_size, KM_NOSLEEP);
-				if (*proc_data) {
-					bcopy(tmp, *proc_data, tmp_size);
-					*proc_size = tmp_size;
-				}
-			}
-		}
-	} else if (channels < hw_channels) {
-		/* mono --> stereo */
-		ATRACE_32("am_p_process() to stereo", channels);
-		ASSERT(channels == AUDIO_CHANNELS_MONO);
-		ASSERT(hw_channels == AUDIO_CHANNELS_STEREO);
-
-		/* set data_start pointing to incoming data */
-		data_start = (int *)chpptr->acp_ch_pconv1 +
-		    (chpptr->acp_ch_pbuf_size >> AM_TIMES_4_SHIFT);
-
-		/*
-		 * Convert to int and save. We need this mono data twice,
-		 * as the source for SRC and to be converted to stereo. If
-		 * we converted directly to stereo then we would have to
-		 * do SRC on stereo instead of mono.
-		 */
-		am_convert_to_int(buf, data_start, orig_samples, precision,
-		    encoding, 0);
-
-		/* double in size and allocate orig data to save */
-		*orig_size <<= AM_TIMES_2_SHIFT;
-		*orig_data = kmem_alloc(*orig_size, KM_NOSLEEP);
-		if (*orig_data == NULL) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!process(2) couldn't allocate"
-			    " memory, audio lost");
-			return (AUDIO_FAILURE);
-		}
-
-		/* convert to stereo */
-		src = data_start;
-		dest = *orig_data;
-		for (i = orig_samples; i--; ) {
-			/* dup to left and right channels */
-			val = *src++;
-			*dest++ = val;
-			*dest++ = val;
-		}
-
-		/* SRC the data and then convert to stereo */
-		if (src_samples) {
-			/* SRC the old mono data, not the new stereo data */
-			tmp = psrs->ad_src_convert(AM_SRC_CHPTR2HDL(chptr),
-			    AUDIO_CHANNELS_MONO, AUDIO_PLAY,
-			    chpptr->acp_ch_pconv1, chpptr->acp_ch_psrc1,
-			    chpptr->acp_ch_psrc2, &src_samples);
-
-			ASSERT(src_samples <= (chpptr->acp_ch_psrc_siz >>
-			    AM_TIMES_4_SHIFT));
-
-			/* if successful then convert to stereo and save */
-			if (tmp) {
-				/* sizeof (int) * 2 = 8 */
-				tmp_size = src_samples << AM_TIMES_8_SHIFT;
-				*proc_data = kmem_alloc(tmp_size, KM_NOSLEEP);
-				if (*proc_data) {
-					src = tmp;
-					dest = *proc_data;
-					for (i = src_samples; i--; ) {
-						/* dup to left & right chs */
-						val = *src++;
-						*dest++ = val;
-						*dest++ = val;
-					}
-					*proc_size = tmp_size;
-				}
-			}
-		}
-	} else {
-		/* stereo --> mono */
-		ATRACE_32("am_p_process() to mono", channels);
-		ASSERT(channels == AUDIO_CHANNELS_STEREO);
-		ASSERT(hw_channels == AUDIO_CHANNELS_MONO);
-
-		/* set data_start pointing to incoming data */
-		data_start = (int *)chpptr->acp_ch_pconv1 +
-		    (chpptr->acp_ch_pbuf_size >> AM_TIMES_4_SHIFT);
-
-		/* convert to int and then from stereo --> mono */
-		am_convert_to_int(buf, data_start, orig_samples, precision,
-		    encoding, 0);
-
-		/* divide in half the size and allocate orig data to save */
-		*orig_size >>= AM_TIMES_2_SHIFT;
-		*orig_data = kmem_alloc(*orig_size, KM_NOSLEEP);
-		if (*orig_data == NULL) {
-			audio_sup_log(AUDIO_STATE2HDL(statep), CE_NOTE,
-			    "!process(3) couldn't allocate"
-			    " memory, audio lost");
-			return (AUDIO_FAILURE);
-		}
-
-		/* convert to mono */
-		src = data_start;
-		dest = *orig_data;
-		for (i = (orig_samples >> AM_TIMES_2_SHIFT); i--; ) {
-			/* average the left and right channels */
-			val = *src++;
-			val += *src++;
-			*dest++ = val >> AM_HALF_ENERGY_SHIFT;
-		}
-
-		/* SRC the data, which is already mono */
-		if (src_samples) {
-			tmp = psrs->ad_src_convert(AM_SRC_CHPTR2HDL(chptr),
-			    AUDIO_CHANNELS_MONO, AUDIO_PLAY,
-			    chpptr->acp_ch_pconv1, chpptr->acp_ch_psrc1,
-			    chpptr->acp_ch_psrc2, &src_samples);
-
-			ASSERT(src_samples <= (chpptr->acp_ch_psrc_siz >>
-			    AM_TIMES_4_SHIFT));
-
-			/* if successful then update info */
-			if (tmp) {
-				/* sizeof (int) = 4 */
-				tmp_size = src_samples << AM_TIMES_4_SHIFT;
-				*proc_data = kmem_alloc(tmp_size, KM_NOSLEEP);
-				if (*proc_data) {
-					bcopy(tmp, *proc_data, tmp_size);
-					*proc_size = tmp_size;
-				}
-			}
-		}
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* am_p_process() */
-
-/*
- * am_set_waiting()
- *
- * Description:
- *	Go through all of the channels. If PID is set then we set the read
- *	and write waiting flags for that PID only. If PID is 0 then we set
- *	all of the AUDIO channels. We set the flag to the argument value.
- *	This lets this routine to be used to both set and clear the flag.
- *
- * Arguments:
- *	audio_state_t	*statep		Pointer to the device instance's state
- *	pid_t		pid		The PID to match
- *	int		value		Value to set the waiting flag
- *	boolean_t	wantwrite	If not 0 then set play.waiting
- *	boolean_t	wantread	If not 0 then set record.waiting
- *
- * Returns:
- *	void
- */
-static void
-am_set_waiting(audio_state_t *statep, pid_t pid, int value, boolean_t wantwrite,
-    boolean_t wantread)
-{
-	audio_ch_t		*tchptr;
-	audio_info_t		*info;
-	int			i;
-	int			max_chs = statep->as_max_chs;
-
-	ASSERT(MUTEX_HELD(&statep->as_lock));
-
-	for (i = 0, tchptr = &statep->as_channels[0]; i < max_chs;
-	    i++, tchptr++) {
-
-		/* skip non-audio channels */
-		mutex_enter(&tchptr->ch_lock);
-		if (!(tchptr->ch_flags & AUDIO_CHNL_ALLOCATED) ||
-		    tchptr->ch_info.dev_type != AUDIO ||
-		    tchptr->ch_info.pid == 0) {
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-		/* if pid then look for that PID, otherwise all others */
-		if (pid && tchptr->ch_info.pid != pid) {
-			mutex_exit(&tchptr->ch_lock);
-			continue;
-		}
-
-		/* pid == 0 or pid's match, so set flags */
-		info = tchptr->ch_info.info;
-		if (wantwrite) {
-			info->play.waiting = (uchar_t)value;
-		}
-		if (wantread) {
-			info->record.waiting = (uchar_t)value;
-		}
-		mutex_exit(&tchptr->ch_lock);
-	}
-
-	ASSERT(MUTEX_HELD(&statep->as_lock));
-
-}	/* am_set_waiting() */
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.c b/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.c
new file mode 100644
index 0000000000..66abd593d9
--- /dev/null
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.c
@@ -0,0 +1,1956 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#include <sys/note.h>
+#include <sys/varargs.h>
+#include <sys/stream.h>
+#include <sys/modctl.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/strsun.h>
+#include <sys/strsubr.h>
+#include <sys/taskq.h>
+
+#include <sys/audio.h>
+#include <sys/audio/audio_support.h>
+#include <sys/mixer.h>
+#include <sys/audio/audio_mixer.h>
+
+#include <sys/usb/usba/usbai_version.h>
+#include <sys/usb/usba.h>
+#include <sys/usb/clients/audio/usb_audio.h>
+#include <sys/usb/clients/audio/usb_mixer.h>
+#include <sys/usb/clients/audio/usb_ac/usb_ac.h>
+
+#include "audio_shim.h"
+
+extern void *usb_ac_statep;
+
+extern struct cb_ops audio_cb_ops;
+
+
+/*
+ * The code here is a "shim" connecting legacy SADA USB drivers with the
+ * Boomer audio framework, translating interfaces between the two.
+ * This is an interim measure until new Boomer-native USB drivers are in place.
+ */
+
+extern void audio_dump_bytes(const uint8_t *, int);
+extern void audio_dump_words(const uint16_t *, int);
+extern void audio_dump_dwords(const uint32_t *, int);
+
+
+#define	ASHIM_ENG_PLAY	0
+#define	ASHIM_ENG_REC	1
+
+#define	ASHIM_GET_ENG(statep, which)	(&(statep)->engines[(which)])
+#define	ASHIM_ENG_DIR(engp)	\
+	(((engp)->af_eflags & ENGINE_OUTPUT_CAP) ? AUDIO_PLAY : AUDIO_RECORD)
+
+static int ashim_eng_start(ashim_eng_t *);
+static void ashim_eng_stop(ashim_eng_t *);
+
+static int ashim_af_open(void *arg, int flag,
+    unsigned *fragszp, unsigned *nfragsp, caddr_t *bufp);
+static void ashim_af_close(void *arg);
+static uint64_t ashim_af_count(void *arg);
+static int ashim_af_start(void *arg);
+static void ashim_af_stop(void *arg);
+static int ashim_af_format(void *arg);
+static int ashim_af_channels(void *arg);
+static int ashim_af_rate(void *arg);
+static void ashim_af_sync(void *arg, unsigned nframes);
+static size_t ashim_af_qlen(void *arg);
+
+audio_engine_ops_t ashim_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	ashim_af_open,
+	ashim_af_close,
+	ashim_af_start,
+	ashim_af_stop,
+	ashim_af_count,
+	ashim_af_format,
+	ashim_af_channels,
+	ashim_af_rate,
+	ashim_af_sync,
+	ashim_af_qlen,
+};
+
+
+#define	ASHIM_BUFSCALE_MIN	1
+#define	ASHIM_BUFSCALE_MAX	2000
+#define	ASHIM_BUFSCALE_DEF	10
+
+/* engine buffer size in terms of fragments */
+int ashim_bufscale = ASHIM_BUFSCALE_DEF;
+
+/* use driver specified buffer size instead */
+int ashim_use_drvbuf = 0;
+
+/* format override */
+uint_t ashim_fmt_sr = 0;
+uint_t ashim_fmt_ch = 0;
+uint_t ashim_fmt_prec = 0;
+uint_t ashim_fmt_enc = 0; /* use SADA values e.g. AUDIO_ENCODING_LINEAR = 3 */
+
+/* open without starting engine */
+int ashim_eng_disable = 0;
+
+/* dump audio data */
+uint64_t ashim_dump_audio_start = 0;
+uint64_t ashim_dump_audio_len = 0;
+int ashim_dump_audio_bufsel = 0;	/* 0 = shim buf; 1 = drv/dma buf */
+
+/* dump i/o related counters */
+uint64_t ashim_dump_counters_start = 0;
+uint64_t ashim_dump_counters_len = 0;
+
+/* ignore errors when setting control values */
+int ashim_ctrl_ignore_errors = 0;
+
+/*
+ * *************************************************************************
+ * audio controls
+ *
+ * Note: we cannot determine if a SADA device truly supports play or record
+ * gain adjustment until we actually try the command.  Messages showing
+ * failed control updates will be printed at the DINFO ashim_debug level.
+ */
+
+#define	AUDIO_CTRL_STEREO_LEFT(v)	((uint8_t)((v) & 0xff))
+#define	AUDIO_CTRL_STEREO_RIGHT(v)	((uint8_t)(((v) >> 8) & 0xff))
+#define	AUDIO_CTRL_STEREO_VAL(l, r)	(((l) & 0xff) | (((r) & 0xff) << 8))
+
+/*
+ * framework gain range
+ */
+#define	AF_MAX_GAIN	100
+#define	AF_MIN_GAIN	0
+
+#define	AF_MAX_GAIN_ST	AUDIO_CTRL_STEREO_VAL(AF_MAX_GAIN, AF_MAX_GAIN)
+#define	AF_MIN_GAIN_ST	AUDIO_CTRL_STEREO_VAL(AF_MIN_GAIN, AF_MIN_GAIN)
+
+/*
+ * convert between framework and driver gain values
+ * must multiply (n) before dividing else D2F_GAIN result will be 0
+ */
+#define	F2D_GAIN(n)	(((n) * AUDIO_MAX_GAIN) / AF_MAX_GAIN)
+#define	D2F_GAIN(n)	(((n) * AF_MAX_GAIN) / AUDIO_MAX_GAIN)
+
+typedef struct ashim_ctrl_map ashim_ctrl_map_t;
+
+static uint64_t ashim_ctrl_defval(ashim_state_t *statep,
+    ashim_ctrl_map_t *mapp);
+static int ashim_ctrl_set_defaults(ashim_state_t *statep);
+static int ashim_ctrl_restore(ashim_state_t *statep);
+
+static int ashim_ctrl_rd(void *, uint64_t *);
+static int ashim_ctrl_wr_st(void *, uint64_t);
+static int ashim_ctrl_wr_mn(void *, uint64_t);
+static int ashim_ctrl_wr_bool(void *, uint64_t);
+static int ashim_ctrl_wr_ports(void *, uint64_t);
+
+static void ashim_rem_controls(ashim_state_t *statep);
+static int ashim_add_controls(ashim_state_t *statep);
+
+
+/*
+ * map framework port definitions to SADA ports
+ */
+
+typedef struct {
+	int		dport;
+	const char	*pname;
+} ashim_port_map_t;
+
+
+static ashim_port_map_t ashim_play_pmap[] = {
+	{AUDIO_SPEAKER,		AUDIO_PORT_SPEAKER},
+	{AUDIO_HEADPHONE,	AUDIO_PORT_HEADPHONES},
+	{AUDIO_LINE_OUT,	AUDIO_PORT_LINEOUT},
+	{AUDIO_SPDIF_OUT,	AUDIO_PORT_SPDIFOUT},
+	{AUDIO_AUX1_OUT,	AUDIO_PORT_AUX1OUT},
+	{AUDIO_AUX2_OUT,	AUDIO_PORT_AUX2OUT}
+};
+static int ashim_play_pmap_len =
+	sizeof (ashim_play_pmap) / sizeof (ashim_port_map_t);
+
+
+static ashim_port_map_t ashim_rec_pmap[] = {
+	{AUDIO_MICROPHONE,	AUDIO_PORT_MIC},
+	{AUDIO_LINE_IN,		AUDIO_PORT_LINEIN},
+	{AUDIO_CD,		AUDIO_PORT_CD},
+	{AUDIO_SPDIF_IN,	AUDIO_PORT_SPDIFIN},
+	{AUDIO_AUX1_IN,		AUDIO_PORT_AUX1IN},
+	{AUDIO_AUX2_IN,		AUDIO_PORT_AUX2IN},
+	{AUDIO_CODEC_LOOPB_IN,	AUDIO_PORT_STEREOMIX}
+};
+static int ashim_rec_pmap_len =
+	sizeof (ashim_rec_pmap) / sizeof (ashim_port_map_t);
+
+
+/*
+ * map frameowork controls to SADA controls
+ */
+struct ashim_ctrl_map {
+	int			dcmd;
+	int			dir;
+	audio_ctrl_desc_t	acd;
+	audio_ctrl_wr_t		af_wr;
+	audio_ctrl_rd_t		af_rd;
+};
+
+#define	DESC_ST(n, f)	{						\
+	.acd_name = (n),						\
+	.acd_type = AUDIO_CTRL_TYPE_STEREO,				\
+	.acd_flags = (f) |						\
+		AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE,	\
+	.acd_maxvalue = AF_MAX_GAIN,					\
+	.acd_minvalue = AF_MIN_GAIN,					\
+}
+#define	FUNC_ST		\
+	&ashim_ctrl_wr_st, &ashim_ctrl_rd
+
+
+#define	DESC_MN(n, f)	{						\
+	.acd_name = (n),						\
+	.acd_type = AUDIO_CTRL_TYPE_MONO,				\
+	.acd_flags = (f) |						\
+		AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE,	\
+	.acd_maxvalue = AF_MAX_GAIN,					\
+	.acd_minvalue = AF_MIN_GAIN,					\
+}
+#define	FUNC_MN		\
+	&ashim_ctrl_wr_mn, &ashim_ctrl_rd
+
+
+#define	DESC_BOOL(n, f)	{						\
+	.acd_name = (n),						\
+	.acd_type = AUDIO_CTRL_TYPE_BOOLEAN,				\
+	.acd_flags = (f) |						\
+		AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE,	\
+	.acd_maxvalue = 1,						\
+	.acd_minvalue = 0,						\
+}
+#define	FUNC_BOOL	\
+	&ashim_ctrl_wr_bool, &ashim_ctrl_rd
+
+
+#define	DESC_OUTS(n, f)	{						\
+	.acd_name = AUDIO_CTRL_ID_OUTPUTS,				\
+	.acd_type = AUDIO_CTRL_TYPE_ENUM,				\
+	.acd_flags = (f) |						\
+		AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE |	\
+		AUDIO_CTRL_FLAG_MULTI,					\
+	.acd_maxvalue = (n),						\
+	.acd_minvalue = (n),						\
+}
+
+#define	DESC_INS(n, f)	{						\
+	.acd_name = AUDIO_CTRL_ID_INPUTS,				\
+	.acd_type = AUDIO_CTRL_TYPE_ENUM,				\
+	.acd_flags = (f) |						\
+		AUDIO_CTRL_FLAG_READABLE | AUDIO_CTRL_FLAG_WRITEABLE,	\
+	.acd_maxvalue = (n),						\
+	.acd_minvalue = (n),						\
+}
+#define	FUNC_PORTS	\
+	&ashim_ctrl_wr_ports, &ashim_ctrl_rd
+
+
+/*
+ * sada <-> framework translation table
+ * a DESC_NULL() for audio_ctrl_desc_t is used to indicate a
+ * non-registered sada control that needs some initialization or be used
+ * internally by the shim
+ *
+ * Note: currently, usb_ac only notifies the framework of play volume changes
+ * from HID so the volume control the only one using AUDIO_CTRL_FLAG_POLL
+ */
+static ashim_ctrl_map_t ashim_ctrl_map[] = {
+	{AM_SET_GAIN,		AUDIO_PLAY,
+		DESC_ST(AUDIO_CTRL_ID_VOLUME, AUDIO_CTRL_FLAG_MAINVOL |
+			AUDIO_CTRL_FLAG_PLAY | AUDIO_CTRL_FLAG_POLL),
+		FUNC_ST},
+
+	{AM_SET_GAIN,		AUDIO_PLAY,
+		DESC_MN(AUDIO_CTRL_ID_VOLUME, AUDIO_CTRL_FLAG_MAINVOL |
+			AUDIO_CTRL_FLAG_PLAY | AUDIO_CTRL_FLAG_POLL),
+		FUNC_MN},
+
+	{AM_SET_GAIN,		AUDIO_RECORD,
+		DESC_ST(AUDIO_CTRL_ID_RECGAIN, AUDIO_CTRL_FLAG_RECVOL |
+			AUDIO_CTRL_FLAG_REC),
+		FUNC_ST},
+
+	{AM_SET_GAIN,		AUDIO_RECORD,
+		DESC_MN(AUDIO_CTRL_ID_RECGAIN, AUDIO_CTRL_FLAG_RECVOL |
+			AUDIO_CTRL_FLAG_REC),
+		FUNC_MN},
+
+	{AM_SET_MONITOR_GAIN,	AUDIO_RECORD,
+		DESC_MN(AUDIO_CTRL_ID_MONGAIN, AUDIO_CTRL_FLAG_MONVOL |
+			AUDIO_CTRL_FLAG_MONITOR),
+		FUNC_MN},
+
+	{AM_MIC_BOOST,		AUDIO_RECORD,
+		DESC_BOOL(AUDIO_CTRL_ID_MICBOOST, 0),
+		FUNC_BOOL},
+
+	{AM_SET_PORT,		AUDIO_PLAY,
+		DESC_OUTS(0, 0),
+		FUNC_PORTS},
+
+	{AM_SET_PORT,		AUDIO_RECORD,
+		DESC_INS(0, 0),
+		FUNC_PORTS},
+};
+static int ashim_ctrl_map_len =
+	sizeof (ashim_ctrl_map) / sizeof (ashim_ctrl_map_t);
+
+
+/*
+ * *************************************************************************
+ * shim support routines
+ */
+
+int ashim_debug = DBG_WARN;
+
+void
+vdprint(debug_level_t lvl, const char *fmt, va_list adx)
+{
+	if (ashim_debug < lvl)
+		return;
+
+	vcmn_err(CE_CONT, fmt, adx);
+}
+
+void
+dprint(debug_level_t lvl, const char *fmt, ...)
+{
+	va_list adx;
+
+	va_start(adx, fmt);
+	vdprint(lvl, fmt, adx);
+	va_end(adx);
+}
+
+void
+dwarn(const char *fmt, ...)
+{
+	va_list adx;
+
+	va_start(adx, fmt);
+	vdprint(DBG_WARN, fmt, adx);
+	va_end(adx);
+}
+
+void
+dinfo(const char *fmt, ...)
+{
+	va_list adx;
+
+	va_start(adx, fmt);
+	vdprint(DBG_INFO, fmt, adx);
+	va_end(adx);
+}
+
+void
+ddtl(const char *fmt, ...)
+{
+	va_list adx;
+
+	va_start(adx, fmt);
+	vdprint(DBG_DETAIL, fmt, adx);
+	va_end(adx);
+}
+
+
+/*
+ * get the maximum format specification the device supports
+ */
+static void
+ashim_max_fmt(ashim_state_t *statep, int dir, ashim_fmt_t *fmtp)
+{
+	am_ad_ch_cap_t *capp = &statep->ad_infop->ad_play;
+	am_ad_cap_comb_t *combp = statep->ad_infop->ad_play_comb;
+
+	uint_t *srs, *chs;
+	uint_t sr, ch, prec, enc, val;
+	int i;
+
+	if (dir == AUDIO_RECORD) {
+		capp = &statep->ad_infop->ad_record;
+		combp = statep->ad_infop->ad_rec_comb;
+	}
+	srs = capp->ad_mixer_srs.ad_srs;
+	chs = capp->ad_chs;
+
+	for (i = 0, sr = 0; srs[i]; i++) {
+		val = srs[i];
+		if (val > sr)
+			sr = val;
+	}
+
+	for (i = 0, ch = 0; chs[i]; i++) {
+		val = chs[i];
+		if (val > ch)
+			ch = val;
+	}
+
+	for (i = 0, prec = 0, enc = 0; combp[i].ad_prec; i++) {
+		val = combp[i].ad_prec;
+		if (val > prec)
+			prec = val;
+
+		val = combp[i].ad_enc;
+		if (val > enc)
+			enc = val;
+	}
+
+	fmtp->sr = ashim_fmt_sr ? ashim_fmt_sr : sr;
+	fmtp->ch = ashim_fmt_ch ? ashim_fmt_ch : ch;
+	fmtp->prec = ashim_fmt_prec ? ashim_fmt_prec : prec;
+	fmtp->enc = ashim_fmt_enc ? ashim_fmt_enc : enc;
+}
+
+
+/*
+ * calls the driver's setup routine if present
+ * For USB audio, this opens a pipe to the endpoint usb_as
+ * returns AUDIO_SUCCESS or AUDIO_FAILURE
+ */
+static int
+ashim_ad_setup(ashim_state_t *statep, int dir)
+{
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+
+	if (ad_entry->ad_setup == NULL)
+		return (AUDIO_SUCCESS);
+
+	return (ad_entry->ad_setup(AUDIO_SHIMST2HDL(statep), dir));
+}
+
+
+/*
+ * calls the driver's teardown routine if present.
+ * Note that the amount of teardowns must match the amount of setups used
+ * in each direction.
+ * returns AUDIO_SUCCESS or AUDIO_FAILURE
+ */
+static void
+ashim_ad_teardown(ashim_state_t *statep, int dir)
+{
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+
+	if (ad_entry->ad_teardown != NULL)
+		ad_entry->ad_teardown(AUDIO_SHIMST2HDL(statep), dir);
+}
+
+
+/*
+ * sets the audio format (sample rate, channels, precision, encoding)
+ * returns AUDIO_SUCCESS or AUDIO_FAILURE
+ */
+static int
+ashim_set_fmt(ashim_state_t *statep, int dir, ashim_fmt_t *fmtp)
+{
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+
+	return (ad_entry->ad_set_format(AUDIO_SHIMST2HDL(statep), dir,
+	    fmtp->sr, fmtp->ch, fmtp->prec, fmtp->enc));
+}
+
+
+static int
+ashim_af_fmt(ashim_fmt_t *fmtp)
+{
+	switch (fmtp->enc) {
+	case AUDIO_ENCODING_ULAW:
+		return (AUDIO_FORMAT_ULAW);
+	case AUDIO_ENCODING_ALAW:
+		return (AUDIO_FORMAT_ALAW);
+	case AUDIO_ENCODING_DVI:
+		return (AUDIO_FORMAT_NONE);
+	case AUDIO_ENCODING_LINEAR8:
+		return (AUDIO_FORMAT_U8);
+	case AUDIO_ENCODING_LINEAR:
+		break;
+	default:
+		return (AUDIO_FORMAT_NONE);
+	}
+
+	switch (fmtp->prec) {
+	case 8:
+		return (AUDIO_FORMAT_S8);
+	case 16:
+		return (AUDIO_FORMAT_S16_NE);
+	case 24:
+		return (AUDIO_FORMAT_S24_NE);
+	case 32:
+		return (AUDIO_FORMAT_S32_NE);
+	default:
+		break;
+	}
+	return (AUDIO_FORMAT_NONE);
+}
+
+
+static void
+ashim_rem_eng(ashim_state_t *statep, ashim_eng_t *engp)
+{
+	if (statep->af_devp == NULL || engp->af_engp == NULL)
+		return;
+
+	audio_dev_remove_engine(statep->af_devp, engp->af_engp);
+	audio_engine_free(engp->af_engp);
+	engp->af_engp = NULL;
+
+	if (statep->engcnt > 0)
+		statep->engcnt--;
+}
+
+
+static int
+ashim_add_eng(ashim_state_t *statep, int dir)
+{
+	audio_dev_t *af_devp = statep->af_devp;
+	ashim_eng_t *engp;
+	audio_engine_t *af_engp;
+	int rv = AUDIO_FAILURE;
+	int which;
+
+	if (dir == AUDIO_PLAY) {
+		which = ASHIM_ENG_PLAY;
+		engp = ASHIM_GET_ENG(statep, which);
+		engp->af_eflags = ENGINE_OUTPUT_CAP;
+		engp->prinfop = &statep->ad_infop->ad_defaults->play;
+		engp->name = "PLAY";
+	} else {
+		which = ASHIM_ENG_REC;
+		engp = ASHIM_GET_ENG(statep, which);
+		engp->af_eflags = ENGINE_INPUT_CAP;
+		engp->prinfop = &statep->ad_infop->ad_defaults->record;
+		engp->name = "RECORD";
+	}
+
+	mutex_init(&engp->lock, NULL, MUTEX_DRIVER, NULL);
+
+	engp->statep = statep;
+
+	ashim_max_fmt(statep, dir, &engp->fmt);
+	engp->af_fmt = ashim_af_fmt(&engp->fmt);
+
+	af_engp = audio_engine_alloc(&ashim_engine_ops, engp->af_eflags);
+	if (af_engp == NULL) {
+		audio_dev_warn(af_devp, "audio_engine_alloc failed");
+		goto OUT;
+	}
+
+	engp->af_engp = af_engp;
+	audio_engine_set_private(af_engp, engp);
+	audio_dev_add_engine(af_devp, af_engp);
+
+	engp->flags = ENG_ENABLED;
+
+	statep->engcnt++;
+	rv = AUDIO_SUCCESS;
+
+OUT:
+	if (rv != AUDIO_SUCCESS)
+		ashim_rem_eng(statep, engp);
+
+	return (rv);
+}
+
+
+static int
+ashim_set_config(ashim_state_t *statep, ashim_ctrl_t *ctrlp, int cmd, int dir,
+    int arg1, int arg2)
+{
+	int (*fn)(audiohdl_t, int, int, int, int) =
+	    statep->ad_infop->ad_entry->ad_set_config;
+
+	ddtl("%s - ashim_set_config: cmd 0x%x, dir %d, arg1 0x%x, arg2 0x%x\n",
+	    statep->dstr, cmd, dir, arg1, arg2);
+
+	if (fn(AUDIO_SHIMST2HDL(statep), cmd, dir, arg1, arg2) !=
+	    AUDIO_SUCCESS) {
+		dinfo("%s: failed update control %s "
+		    "with cmd 0x%x, dir %d, arg1 0x%x, arg2 0x%x",
+		    statep->dstr, ctrlp->acd.acd_name, cmd, dir, arg1, arg2);
+		return (EIO);
+	}
+
+	return (0);
+}
+
+
+static int
+ashim_ctrl_set_defaults(ashim_state_t *statep)
+{
+	ashim_ctrl_t *ctrlp;
+	int rv = AUDIO_SUCCESS;
+
+	for (ctrlp = statep->controls; ctrlp; ctrlp = ctrlp->nextp) {
+		if (audio_control_write(ctrlp->af_ctrlp, ctrlp->defval)) {
+			dinfo("%s: setting control %s to default value "
+			    "0x%llx failed\n",
+			    statep->dstr, ctrlp->acd.acd_name,
+			    (long long unsigned)ctrlp->defval);
+			rv = AUDIO_FAILURE;
+		}
+	}
+	return (rv);
+}
+
+
+static int
+ashim_ctrl_restore(ashim_state_t *statep)
+{
+	ashim_ctrl_t *ctrlp;
+	int rv = AUDIO_SUCCESS;
+
+	for (ctrlp = statep->controls; ctrlp; ctrlp = ctrlp->nextp) {
+		if (ctrlp->af_wr((void *)ctrlp, ctrlp->cval) != 0) {
+			dinfo("%s: restoring "
+			    "control %s to value 0x%llx failed",
+			    statep->dstr, ctrlp->acd.acd_name,
+			    (long long unsigned)ctrlp->cval);
+			rv = AUDIO_FAILURE;
+		}
+	}
+	return (rv);
+}
+
+
+static inline void
+ashim_dump_audio(ashim_eng_t *engp, void *buf, int sz)
+{
+	if (engp->io_count >= ashim_dump_audio_start &&
+	    engp->io_count - ashim_dump_audio_start < ashim_dump_audio_len) {
+		int samples = sz >> engp->smszshift;
+		int frames = samples >> engp->frsmshift;
+
+		cmn_err(CE_NOTE, "\n\n======= %s - %s: REQUEST #%llu, "
+		    "BUF I/O #%llu, SAMPLES %d (%d frames, %d bytes), "
+		    "BUF ADDR 0x%p =======\n",
+		    engp->statep->dstr, engp->name,
+		    (unsigned long long)engp->io_count,
+		    (unsigned long long)engp->bufio_count,
+		    samples, frames, sz, buf);
+
+		if (sz <= 0)
+			return;
+
+		switch (engp->fmt.prec) {
+		case 8:
+		case 24:
+			audio_dump_bytes((uint8_t *)buf, sz);
+			break;
+		case 16:
+			audio_dump_words((uint16_t *)buf, sz >> 1);
+			break;
+		case 32:
+			audio_dump_dwords((uint32_t *)buf, sz >> 2);
+			break;
+		}
+	}
+}
+
+
+static inline void
+ashim_dump_counters(ashim_eng_t *engp, unsigned frames, int bufcnt)
+{
+	if (engp->io_count >= ashim_dump_counters_start &&
+	    engp->io_count - ashim_dump_counters_start <
+	    ashim_dump_counters_len) {
+		int samples = frames << engp->frsmshift;
+		int sz = samples << engp->smszshift;
+
+		if (bufcnt >= 0) {
+			cmn_err(CE_CONT, "======= %s - %s: buf i/o %d of "
+			    "REQUEST #%llu, SAMPLES %d "
+			    "(%d frames, %d bytes)\n",
+			    engp->statep->dstr, engp->name, bufcnt,
+			    (unsigned long long)engp->io_count, samples,
+			    frames, sz);
+		} else {
+			cmn_err(CE_CONT, "======= %s - %s: REQUEST #%llu, "
+			    "SAMPLES %d (%d frames, %d bytes)\n",
+			    engp->statep->dstr, engp->name,
+			    (unsigned long long)engp->io_count,
+			    samples, frames, sz);
+		}
+	}
+}
+
+
+/*
+ * moves data between driver buffer and framework/shim buffer
+ */
+static void
+ashim_eng_bufio(ashim_eng_t *engp, void *buf, size_t sz)
+{
+	size_t cpsz = sz;
+	caddr_t *src, *dst, *dp;
+
+	if (engp->af_eflags & ENGINE_OUTPUT_CAP) {
+		src = &engp->bufpos;
+		dst = (caddr_t *)&buf;
+	} else {
+		src = (caddr_t *)&buf;
+		dst = &engp->bufpos;
+	}
+	dp = ashim_dump_audio_bufsel ? (caddr_t *)&buf : &engp->bufpos;
+
+	/*
+	 * Wrap.  If sz is exactly the remainder of the buffer
+	 * (bufpos + sz == bufendp) then the second cpsz should be 0 and so
+	 * the second memcpy() should have no effect, with bufpos updated
+	 * to the head of the buffer.
+	 */
+	if (engp->bufpos + sz >= engp->bufendp) {
+		cpsz = (size_t)engp->bufendp - (size_t)engp->bufpos;
+		(void) memcpy(*dst, *src, cpsz);
+
+		if (ashim_dump_audio_len)
+			ashim_dump_audio(engp, *dp, cpsz);
+
+		buf = (caddr_t)buf + cpsz;
+		engp->bufpos = engp->bufp;
+		cpsz = sz - cpsz;
+	}
+
+	if (cpsz) {
+		(void) memcpy(*dst, *src, cpsz);
+
+		if (ashim_dump_audio_len)
+			ashim_dump_audio(engp, *dp, cpsz);
+
+		engp->bufpos += cpsz;
+	}
+	engp->bufio_count++;
+}
+
+
+static void
+ashim_prtstats(ashim_eng_t *engp)
+{
+	ashim_state_t *statep = engp->statep;
+
+	dinfo("%s - %s: driver i/o: %llu, framework i/o: %llu, "
+	    "frames: %llu\n", statep->dstr, engp->name,
+	    (long long unsigned)engp->io_count,
+	    (long long unsigned)engp->bufio_count,
+	    (long long unsigned)engp->frames);
+}
+
+
+/*
+ * *************************************************************************
+ * audio control routines
+ */
+
+static uint64_t
+ashim_ctrl_defval(ashim_state_t *statep, ashim_ctrl_map_t *mapp)
+{
+	audio_prinfo_t *play_prp = &statep->ad_infop->ad_defaults->play;
+	audio_prinfo_t *rec_prp = &statep->ad_infop->ad_defaults->record;
+	const char *cname = mapp->acd.acd_name;
+	uint64_t cval = 0;
+	uint64_t gain;
+	ashim_fmt_t fmt;
+
+	if (strcmp(cname, AUDIO_CTRL_ID_VOLUME) == 0) {
+		ashim_max_fmt(statep, AUDIO_PLAY, &fmt);
+		if (fmt.ch == 0)
+			return (0);
+
+		gain = D2F_GAIN(play_prp->gain);
+		if (fmt.ch == 1)
+			cval = gain;
+		else
+			cval = AUDIO_CTRL_STEREO_VAL(gain, gain);
+
+	} else if (strcmp(cname, AUDIO_CTRL_ID_RECGAIN) == 0) {
+		ashim_max_fmt(statep, AUDIO_RECORD, &fmt);
+		if (fmt.ch == 0)
+			return (0);
+
+		gain = D2F_GAIN(rec_prp->gain);
+		if (fmt.ch == 1)
+			cval = gain;
+		else
+			cval = AUDIO_CTRL_STEREO_VAL(gain, gain);
+
+	} else if (strcmp(cname, AUDIO_CTRL_ID_MONGAIN) == 0) {
+		cval = D2F_GAIN(statep->ad_infop->ad_defaults->monitor_gain);
+
+	} else if (strcmp(cname, AUDIO_CTRL_ID_MICBOOST) == 0) {
+		cval = 0;
+
+	} else if (strcmp(cname, AUDIO_CTRL_ID_OUTPUTS) == 0) {
+		cval = play_prp->port;
+
+	} else if (strcmp(cname, AUDIO_CTRL_ID_INPUTS) == 0) {
+		cval = rec_prp->port;
+	}
+
+	return (cval);
+}
+
+
+static ashim_ctrl_t *
+ashim_find_ctrl_dcmd(ashim_state_t *statep, int dcmd, int dir)
+{
+	ashim_ctrl_t *ctrlp;
+
+	for (ctrlp = statep->controls; ctrlp != NULL; ctrlp = ctrlp->nextp) {
+		if (ctrlp->dcmd == dcmd && ctrlp->dir == dir)
+			break;
+	}
+	return (ctrlp);
+}
+
+
+/*
+ * control callback and related routines
+ */
+
+static int
+ashim_ctrl_gain_mutable(ashim_ctrl_t *ctrlp, int left, int right)
+{
+	ashim_state_t *statep = ctrlp->statep;
+	int gain;
+
+	if (left == 0 && right == 0)
+		gain = 0;
+	else if (left != 0)
+		gain = left;
+	else
+		gain = right;
+
+	if (ctrlp->dcmd == AM_SET_GAIN && ctrlp->dir == AUDIO_PLAY) {
+		/*
+		 * mute when gain = 0, and on the transition when gain != 0
+		 * but do not set unmute cmds during non-zero changes
+		 */
+		if (gain == 0) {
+			ctrlp->flags |= CTRL_MUTED;
+
+			if (ashim_set_config(statep, ctrlp, AM_OUTPUT_MUTE,
+			    AUDIO_PLAY, 1, 0) != AUDIO_SUCCESS)
+				return (EIO);
+
+		} else if (ctrlp->flags & CTRL_MUTED) {
+			ctrlp->flags &= ~CTRL_MUTED;
+
+			if (ashim_set_config(statep, ctrlp, AM_OUTPUT_MUTE,
+			    AUDIO_PLAY, 0, 0) != AUDIO_SUCCESS)
+				return (EIO);
+		}
+	}
+
+	return (0);
+}
+
+
+/*
+ * control read callback
+ */
+static int
+ashim_ctrl_rd(void *arg, uint64_t *cvalp)
+{
+	ashim_ctrl_t *ctrlp = arg;
+
+	mutex_enter(&ctrlp->lock);
+	*cvalp = ctrlp->cval;
+	mutex_exit(&ctrlp->lock);
+
+	return (0);
+}
+
+
+/*
+ * stereo level control callback
+ */
+static int
+ashim_ctrl_wr_st(void *arg, uint64_t cval)
+{
+	ashim_ctrl_t *ctrlp = arg;
+	ashim_state_t *statep = ctrlp->statep;
+	int rv = EIO;
+	int left, right;
+
+	ddtl("%s - control %s WRITE: 0x%llx\n", statep->dstr,
+	    ctrlp->acd.acd_name, (long long unsigned)cval);
+
+	left = AUDIO_CTRL_STEREO_LEFT(cval);
+	right = AUDIO_CTRL_STEREO_RIGHT(cval);
+
+	if (left < AF_MIN_GAIN || left > AF_MAX_GAIN ||
+	    right < AF_MIN_GAIN || right > AF_MAX_GAIN) {
+		dinfo("%s - control %s invalid value: 0x%llx\n", statep->dstr,
+		    ctrlp->acd.acd_name, (long long unsigned)cval);
+		return (EINVAL);
+	}
+
+	mutex_enter(&ctrlp->lock);
+	ctrlp->cval = cval;
+
+	left = F2D_GAIN(left);
+	right = F2D_GAIN(right);
+
+	if (ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir, left,
+	    0) != AUDIO_SUCCESS)
+		goto OUT;
+
+	if (ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir, right,
+	    1) != AUDIO_SUCCESS) {
+		/* restore previous left gain value */
+		(void) ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir,
+		    left, 0);
+		goto OUT;
+	}
+
+	rv = ashim_ctrl_gain_mutable(ctrlp, left, right);
+
+OUT:
+	mutex_exit(&ctrlp->lock);
+	return (ashim_ctrl_ignore_errors ? 0 : rv);
+}
+
+
+/*
+ * mono level control callback
+ */
+static int
+ashim_ctrl_wr_mn(void *arg, uint64_t cval)
+{
+	ashim_ctrl_t *ctrlp = arg;
+	ashim_state_t *statep = ctrlp->statep;
+	int rv = EIO;
+	int gain;
+
+	ddtl("%s - control %s WRITE: 0x%llx\n", statep->dstr,
+	    ctrlp->acd.acd_name, (long long unsigned)cval);
+
+	if (cval < (uint64_t)AF_MIN_GAIN || cval > (uint64_t)AF_MAX_GAIN) {
+		dinfo("%s - control %s invalid value: 0x%llx\n", statep->dstr,
+		    ctrlp->acd.acd_name, (long long unsigned)cval);
+		return (EINVAL);
+	}
+
+	mutex_enter(&ctrlp->lock);
+	ctrlp->cval = cval;
+
+	gain = (int)F2D_GAIN(cval);
+
+	if (ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir,
+	    gain, 0) != AUDIO_SUCCESS)
+		goto OUT;
+
+	rv = ashim_ctrl_gain_mutable(ctrlp, gain, 0);
+
+OUT:
+	mutex_exit(&ctrlp->lock);
+	return (ashim_ctrl_ignore_errors ? 0 : rv);
+}
+
+
+/*
+ * boolean control callback
+ */
+/*ARGSUSED*/
+static int
+ashim_ctrl_wr_bool(void *arg, uint64_t cval)
+{
+	ashim_ctrl_t *ctrlp = arg;
+	ashim_state_t *statep = ctrlp->statep;
+	int rv = EIO;
+
+	ddtl("%s - control %s WRITE: 0x%llx\n", statep->dstr,
+	    ctrlp->acd.acd_name, (long long unsigned)cval);
+
+	mutex_enter(&ctrlp->lock);
+	ctrlp->cval = cval;
+
+	if (ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir,
+	    (int)cval, 0) != AUDIO_SUCCESS)
+		goto OUT;
+
+	rv = 0;
+
+OUT:
+	mutex_exit(&ctrlp->lock);
+	return (ashim_ctrl_ignore_errors ? 0 : rv);
+}
+
+
+/*
+ * port selection control callback
+ */
+static int
+ashim_ctrl_wr_ports(void *arg, uint64_t cval)
+{
+	ashim_ctrl_t *ctrlp = arg;
+	ashim_state_t *statep = ctrlp->statep;
+	int rv = EIO;
+	int dports = cval & 0xff;
+
+	ddtl("%s - control %s WRITE: 0x%llx\n", statep->dstr,
+	    ctrlp->acd.acd_name, (long long unsigned)cval);
+
+	if ((cval & ~ctrlp->acd.acd_minvalue) !=
+	    (ctrlp->acd.acd_maxvalue & ~ctrlp->acd.acd_minvalue)) {
+		dinfo("%s - control %s invalid value: 0x%llx\n", statep->dstr,
+		    ctrlp->acd.acd_name, (long long unsigned)cval);
+		return (EINVAL);
+	}
+
+	mutex_enter(&ctrlp->lock);
+	ctrlp->cval = cval;
+
+	if (ashim_set_config(statep, ctrlp, ctrlp->dcmd, ctrlp->dir, dports,
+	    0) != AUDIO_SUCCESS)
+		goto OUT;
+
+	rv = 0;
+
+OUT:
+	mutex_exit(&ctrlp->lock);
+	return (ashim_ctrl_ignore_errors ? 0 : rv);
+}
+
+
+/*
+ * audio control registration related routines
+ */
+
+static ashim_ctrl_t *
+ashim_ctrl_alloc(void)
+{
+	return (kmem_zalloc(sizeof (ashim_ctrl_t), KM_SLEEP));
+}
+
+
+static void
+ashim_ctrl_free(ashim_ctrl_t *ctrlp)
+{
+	kmem_free(ctrlp, sizeof (ashim_ctrl_t));
+}
+
+
+static void
+ashim_ctrl_insert(ashim_state_t *statep, ashim_ctrl_t *ctrlp)
+{
+	ctrlp->nextp = statep->controls;
+	statep->controls = ctrlp;
+}
+
+
+/*
+ * returns the amount of modifiable ports
+ */
+static int
+ashim_ctrl_init_ports(ashim_state_t *statep, ashim_ctrl_t *ctrlp)
+{
+	audio_prinfo_t *prp;
+	ashim_port_map_t *pmap;
+	int pmaplen;
+	int i;
+	int count = 0;
+
+	if (ctrlp->dir == AUDIO_PLAY) {
+		prp = &statep->ad_infop->ad_defaults->play;
+		pmap = ashim_play_pmap;
+		pmaplen = ashim_play_pmap_len;
+	} else {
+		prp = &statep->ad_infop->ad_defaults->record;
+		pmap = ashim_rec_pmap;
+		pmaplen = ashim_rec_pmap_len;
+	}
+
+	/*
+	 * look at all SADA supported ports then set the corresponding
+	 * framework defined bits in the control description if the driver
+	 * informs us that it is present (avail_ports) and if it can be
+	 * toggled on/off (mod_ports)
+	 */
+	for (i = 0; i < pmaplen; i++) {
+		ctrlp->acd.acd_enum[i] = pmap[i].pname;
+
+		if (pmap[i].dport & prp->avail_ports) {
+			ctrlp->acd.acd_maxvalue |= pmap[i].dport;
+
+			dinfo("%s: available port: "
+			    "driver 0x%x, framework 0x%s\n",
+			    statep->dstr, pmap[i].dport, pmap[i].pname);
+		}
+		if (pmap[i].dport & prp->mod_ports) {
+			ctrlp->acd.acd_minvalue |= pmap[i].dport;
+
+			dinfo("%s: modifiable port: "
+			    "(driver 0x%x, framework %s)\n",
+			    statep->dstr, pmap[i].dport, pmap[i].pname);
+
+			count++;
+		}
+	}
+
+	return (count);
+}
+
+
+static void
+ashim_ctrl_fini(ashim_ctrl_t *ctrlp)
+{
+	mutex_destroy(&ctrlp->lock);
+}
+
+
+/*
+ * returns 0 if initialization is successful; failure to initialize is
+ * not fatal so caller should deallocate and continue on to the next control
+ */
+static int
+ashim_ctrl_init(ashim_state_t *statep, ashim_ctrl_t *ctrlp,
+    ashim_ctrl_map_t *mapp)
+{
+	ctrlp->dcmd = mapp->dcmd;
+	ctrlp->dir = mapp->dir;
+	ctrlp->acd = mapp->acd;
+	ctrlp->af_ctrlp = NULL;
+	ctrlp->defval = ashim_ctrl_defval(statep, mapp);
+	ctrlp->cval = ctrlp->defval;
+	ctrlp->af_wr = mapp->af_wr;
+	ctrlp->statep = statep;
+
+	if (mapp->acd.acd_type == AUDIO_CTRL_TYPE_ENUM &&
+	    ashim_ctrl_init_ports(statep, ctrlp) <= 1) {
+		dinfo("%s: no more than one modifiable port detected for "
+		    "control %s, enabling and skipping\n",
+		    statep->dstr, ctrlp->acd.acd_name);
+
+		(void) ctrlp->af_wr((void *)ctrlp, ctrlp->defval);
+		return (1);
+	}
+
+	mutex_init(&ctrlp->lock, NULL, MUTEX_DRIVER, NULL);
+	return (0);
+}
+
+
+/*
+ * heuristic to determine if a control is actually present by writing the
+ * default value and checking if the operation succeeds
+ */
+/*ARGSUSED*/
+static int
+ashim_ctrl_test(ashim_state_t *statep, ashim_ctrl_t *ctrlp)
+{
+	return (ctrlp->af_wr((void *)ctrlp, ctrlp->defval));
+}
+
+
+static void
+ashim_rem_controls(ashim_state_t *statep)
+{
+	ashim_ctrl_t *ctrlp = statep->controls;
+	ashim_ctrl_t *nextp;
+
+	while (ctrlp != NULL) {
+		if (ctrlp->af_ctrlp != NULL)
+			audio_dev_del_control(ctrlp->af_ctrlp);
+
+		nextp = ctrlp->nextp;
+		ashim_ctrl_fini(ctrlp);
+		ashim_ctrl_free(ctrlp);
+		ctrlp = nextp;
+	}
+
+	/* required */
+	statep->controls = NULL;
+}
+
+
+static int
+ashim_add_controls(ashim_state_t *statep)
+{
+	int rv = AUDIO_FAILURE;
+	int i;
+	ashim_ctrl_map_t *mapp;
+	ashim_ctrl_t *ctrlp;
+	audio_ctrl_t *af_ctrlp;
+	ashim_fmt_t playfmt = {0};
+	ashim_fmt_t recfmt = {0};
+	ashim_fmt_t *fmtp;
+	int ad_feat = statep->ad_infop->ad_defaults->hw_features;
+
+	if (ad_feat & AUDIO_HWFEATURE_PLAY)
+		ashim_max_fmt(statep, AUDIO_PLAY, &playfmt);
+
+	if (ad_feat & AUDIO_HWFEATURE_RECORD)
+		ashim_max_fmt(statep, AUDIO_RECORD, &recfmt);
+
+	for (i = 0; i < ashim_ctrl_map_len; i++) {
+		mapp = &ashim_ctrl_map[i];
+		fmtp = NULL;
+
+		if (mapp->dir == AUDIO_PLAY) {
+			if (!(ad_feat & AUDIO_HWFEATURE_PLAY))
+				continue;
+
+			fmtp = &playfmt;
+		}
+		if (mapp->dir == AUDIO_RECORD) {
+			if (!(ad_feat & AUDIO_HWFEATURE_RECORD))
+				continue;
+
+			fmtp = &recfmt;
+		}
+
+		if (mapp->dcmd == AM_SET_GAIN) {
+			if (fmtp->ch == 0)
+				continue;
+
+			if (mapp->acd.acd_type == AUDIO_CTRL_TYPE_MONO &&
+			    fmtp->ch != 1)
+				continue;
+
+			if (mapp->acd.acd_type == AUDIO_CTRL_TYPE_STEREO &&
+			    fmtp->ch != 2)
+				continue;
+		}
+
+		ctrlp = ashim_ctrl_alloc();
+
+		if (ashim_ctrl_init(statep, ctrlp, mapp)) {
+			ashim_ctrl_free(ctrlp);
+			continue;
+		}
+
+		if (ashim_ctrl_test(statep, ctrlp)) {
+			dinfo("%s: control %s tested invalid, ignoring\n",
+			    statep->dstr, ctrlp->acd.acd_name);
+
+			ashim_ctrl_fini(ctrlp);
+			ashim_ctrl_free(ctrlp);
+			continue;
+		}
+
+		af_ctrlp = audio_dev_add_control(statep->af_devp, &ctrlp->acd,
+		    mapp->af_rd, mapp->af_wr, (void *)ctrlp);
+
+		if (af_ctrlp == NULL) {
+			audio_dev_warn(statep->af_devp, "failed to add "
+			    "control %s", ctrlp->acd.acd_name);
+			ashim_ctrl_fini(ctrlp);
+			ashim_ctrl_free(ctrlp);
+			goto OUT;
+		}
+
+		dinfo("%s: added control %s, type: %d, "
+		    "flags: 0x%x, min: 0x%llx, max: 0x%llx, default: 0x%llx\n",
+		    statep->dstr, ctrlp->acd.acd_name, ctrlp->acd.acd_type,
+		    ctrlp->acd.acd_flags,
+		    ctrlp->acd.acd_minvalue, ctrlp->acd.acd_maxvalue,
+		    ctrlp->defval);
+
+		ctrlp->af_ctrlp = af_ctrlp;
+		ashim_ctrl_insert(statep, ctrlp);
+	}
+
+	rv = AUDIO_SUCCESS;
+
+OUT:
+	if (rv != AUDIO_SUCCESS)
+		ashim_rem_controls(statep);
+	return (rv);
+}
+
+
+/*
+ * **************************************************************************
+ * replacements for SADA framework interfaces
+ */
+
+void *
+audio_sup_get_private(audiohdl_t handle)
+{
+	return (AUDIO_HDL2SHIMST(handle)->private);
+}
+
+
+void
+audio_sup_set_private(audiohdl_t handle, void *private)
+{
+	AUDIO_HDL2SHIMST(handle)->private = private;
+}
+
+
+int
+audio_sup_unregister(audiohdl_t handle)
+{
+	ashim_state_t *statep;
+
+	if (handle == NULL)
+		return (AUDIO_SUCCESS);
+
+	statep = AUDIO_HDL2SHIMST(handle);
+
+	kmem_free(statep, sizeof (*statep));
+	return (AUDIO_SUCCESS);
+}
+
+
+audiohdl_t
+audio_sup_register(dev_info_t *dip)
+{
+	ashim_state_t *statep = NULL;
+	int inst;
+	const char *nm;
+
+	statep = kmem_zalloc(sizeof (*statep), KM_SLEEP);
+	statep->dip = dip;
+
+	inst = ddi_get_instance(dip);
+	nm = ddi_driver_name(dip);
+
+	(void) snprintf(statep->dstr, sizeof (statep->dstr),
+	    "%s#%d", nm, inst);
+
+	return (AUDIO_SHIMST2HDL(statep));
+}
+
+
+int
+am_unregister(audiohdl_t handle)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+
+	if (statep->af_devp == NULL)
+		return (AUDIO_SUCCESS);
+
+	if (statep->flags & AF_REGISTERED) {
+		if (audio_dev_unregister(statep->af_devp) != DDI_SUCCESS) {
+			dwarn("%s: am_unregister: audio_dev_unregister() "
+			    "failed\n", statep->dstr);
+
+			return (AUDIO_FAILURE);
+		}
+		statep->flags &= ~AF_REGISTERED;
+	}
+
+	return (AUDIO_SUCCESS);
+}
+
+
+/*ARGSUSED*/
+int
+am_detach(audiohdl_t handle, ddi_detach_cmd_t cmd)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	int i;
+
+	if (statep == NULL)
+		return (AUDIO_SUCCESS);
+
+	if (statep->af_devp == NULL)
+		return (AUDIO_SUCCESS);
+
+	if ((statep->flags & AF_REGISTERED) &&
+	    audio_dev_unregister(statep->af_devp) != DDI_SUCCESS) {
+		dwarn("%s: am_detach: audio_dev_unregister() failed\n",
+		    statep->dstr);
+		return (AUDIO_FAILURE);
+	}
+	statep->flags &= ~AF_REGISTERED;
+
+	for (i = 0; i < statep->engcnt; i++)
+		ashim_rem_eng(statep, &statep->engines[i]);
+
+	if (statep->controls != NULL)
+		ashim_rem_controls(statep);
+
+	audio_dev_free(statep->af_devp);
+	statep->af_devp = NULL;
+
+	return (AUDIO_SUCCESS);
+}
+
+
+int
+am_attach(audiohdl_t handle, ddi_attach_cmd_t cmd,
+    am_ad_info_t *ad_infop)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	audio_dev_t *af_devp;
+	int ad_feat = ad_infop->ad_defaults->hw_features;
+	int rv = AUDIO_FAILURE;
+
+	if (cmd != DDI_ATTACH)
+		return (AUDIO_FAILURE);
+
+	af_devp = audio_dev_alloc(statep->dip, 0);
+	audio_dev_set_description(af_devp, ad_infop->ad_dev_info->name);
+	audio_dev_set_version(af_devp, ad_infop->ad_dev_info->version);
+
+	statep->af_devp = af_devp;
+	statep->ad_infop = ad_infop;
+
+	/*
+	 * setup the engines
+	 */
+	statep->engcnt = 0;
+
+	/*
+	 * If the device supports both play and record we require duplex
+	 * functionality.  However, there are no known simplex SADA devices.
+	 * In this case we limit the device to play only.
+	 */
+	if ((ad_feat & AUDIO_HWFEATURE_PLAY) &&
+	    (ad_feat & AUDIO_HWFEATURE_RECORD) &&
+	    !(ad_feat & AUDIO_HWFEATURE_DUPLEX)) {
+		audio_dev_warn(af_devp, "missing duplex feature "
+		    "required for record");
+
+		ad_feat &= ~AUDIO_HWFEATURE_RECORD;
+		ad_infop->ad_defaults->hw_features = ad_feat;
+	}
+
+	if (ad_feat & AUDIO_HWFEATURE_PLAY) {
+		if (ashim_add_eng(statep, AUDIO_PLAY) != AUDIO_SUCCESS)
+			goto OUT;
+	}
+	if (ad_feat & AUDIO_HWFEATURE_RECORD) {
+		if (ashim_add_eng(statep, AUDIO_RECORD) != AUDIO_SUCCESS)
+			goto OUT;
+	}
+
+	if (ashim_add_controls(statep) != AUDIO_SUCCESS)
+		goto OUT;
+
+	if (ashim_ctrl_set_defaults(statep) != AUDIO_SUCCESS)
+		goto OUT;
+
+	if (audio_dev_register(af_devp) != DDI_SUCCESS) {
+		audio_dev_warn(af_devp, "audio_dev_register() failed");
+		goto OUT;
+	}
+	statep->flags |= AF_REGISTERED;
+
+	rv = AUDIO_SUCCESS;
+
+OUT:
+	if (rv != AUDIO_SUCCESS)
+		(void) am_detach(handle, DDI_DETACH);
+	return (rv);
+}
+
+
+int
+am_get_audio(audiohdl_t handle, void *buf, int samples)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	ashim_eng_t *engp = ASHIM_GET_ENG(statep, ASHIM_ENG_PLAY);
+	unsigned reqframes = samples >> engp->frsmshift;
+	unsigned frames;
+	unsigned i;
+	size_t sz;
+	int bufcnt = 0;
+	caddr_t bp = buf;
+
+	mutex_enter(&engp->lock);
+	if (ashim_dump_counters_len)
+		ashim_dump_counters(engp, reqframes, -1);
+
+	if (!(engp->flags & ENG_STARTED)) {
+		ddtl("%s - am_get_audio: stop in progress, ignoring\n",
+		    statep->dstr);
+		mutex_exit(&engp->lock);
+		return (0);
+	}
+	mutex_exit(&engp->lock);
+
+	/* break requests from the driver into fragment sized chunks */
+	for (i = 0; i < reqframes; i += frames) {
+		mutex_enter(&engp->lock);
+
+		frames = reqframes - i;
+		if (frames > engp->fragfr)
+			frames = engp->fragfr;
+
+		sz = (frames << engp->frsmshift) << engp->smszshift;
+
+		if (ashim_dump_counters_len) {
+			ashim_dump_counters(engp, frames, bufcnt);
+		}
+		bufcnt++;
+
+		/* must move data before updating framework */
+		ashim_eng_bufio(engp, bp, sz);
+		engp->frames += frames;
+		bp += sz;
+
+		mutex_exit(&engp->lock);
+		audio_engine_consume(engp->af_engp);
+	}
+
+	mutex_enter(&engp->lock);
+	engp->io_count++;
+	mutex_exit(&engp->lock);
+
+	return (samples);
+}
+
+
+void
+am_play_shutdown(audiohdl_t handle)
+{
+
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+
+	ad_entry->ad_stop_play(handle);
+
+	/*
+	 * XXX
+	 * used to notify framework that device's engine has stopped
+	 */
+}
+
+
+void
+am_send_audio(audiohdl_t handle, void *buf, int samples)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	ashim_eng_t *engp = ASHIM_GET_ENG(statep, ASHIM_ENG_REC);
+	unsigned reqframes = samples >> engp->frsmshift;
+	unsigned frames;
+	unsigned i;
+	size_t sz;
+	int bufcnt = 0;
+	caddr_t bp = buf;
+
+	mutex_enter(&engp->lock);
+	if (ashim_dump_counters_len)
+		ashim_dump_counters(engp, reqframes, -1);
+
+	if (!(engp->flags & ENG_STARTED)) {
+		ddtl("%s - am_send_audio: stop in progress, ignoring\n",
+		    statep->dstr);
+
+		mutex_exit(&engp->lock);
+		return;
+	}
+	mutex_exit(&engp->lock);
+
+	/* break requests from the driver into fragment sized chunks */
+	for (i = 0; i < reqframes; i += frames) {
+		mutex_enter(&engp->lock);
+
+		frames = reqframes - i;
+		if (frames > engp->fragfr)
+			frames = engp->fragfr;
+
+		sz = (frames << engp->frsmshift) << engp->smszshift;
+
+		if (ashim_dump_counters_len)
+			ashim_dump_counters(engp, frames, bufcnt);
+		bufcnt++;
+
+		/* must move data before updating framework */
+		ashim_eng_bufio(engp, bp, sz);
+		engp->frames += frames;
+		bp += sz;
+
+		mutex_exit(&engp->lock);
+		audio_engine_produce(engp->af_engp);
+	}
+
+	mutex_enter(&engp->lock);
+	engp->io_count++;
+	mutex_exit(&engp->lock);
+}
+
+void
+audio_sup_restore_state(audiohdl_t handle)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	ashim_eng_t *engp;
+	int i;
+	int start = 0;
+
+	(void) ashim_ctrl_restore(statep);
+
+	for (i = 0; i < statep->engcnt; i++) {
+		engp = &statep->engines[i];
+
+		mutex_enter(&engp->lock);
+		start = (engp->flags & ENG_STARTED);
+		mutex_exit(&engp->lock);
+
+		if (start)
+			(void) ashim_eng_start(engp);
+	}
+}
+
+
+/*
+ * **************************************************************************
+ * audio framework engine callbacks
+ */
+
+/*ARGSUSED*/
+static int
+ashim_af_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	ashim_eng_t *engp = (ashim_eng_t *)arg;
+	ashim_state_t *statep = engp->statep;
+	int rv = EIO;
+	int dir = ASHIM_ENG_DIR(engp);
+
+
+	if (usb_ac_open(statep->dip) != AUDIO_SUCCESS) {
+		audio_dev_warn(statep->af_devp, "usb_ac_open failed");
+		return (EIO);
+	}
+
+	mutex_enter(&engp->lock);
+
+	if (ashim_set_fmt(statep, dir, &engp->fmt) != AUDIO_SUCCESS) {
+		audio_dev_warn(statep->af_devp, "set format failed");
+		goto OUT;
+	}
+
+	engp->intrate = (engp->af_eflags & ENGINE_OUTPUT_CAP) ?
+	    statep->ad_infop->ad_play.ad_int_rate :
+	    statep->ad_infop->ad_record.ad_int_rate;
+
+	engp->sampsz = engp->fmt.prec / 8;
+	engp->framesz = engp->sampsz * engp->fmt.ch;
+
+	if (engp->fmt.ch > 2) {
+		audio_dev_warn(statep->af_devp, "unsupported ",
+		    "channel count: %u", engp->fmt.ch);
+		goto OUT;
+	}
+	if (engp->fmt.prec > 16) {
+		audio_dev_warn(statep->af_devp, "unsupported ",
+		    "precision: %u", engp->fmt.prec);
+		goto OUT;
+	}
+
+	engp->frsmshift = engp->fmt.ch / 2;
+	engp->smszshift = engp->sampsz / 2;
+
+	/*
+	 * In order to match the requested number of samples per interrupt
+	 * from SADA drivers when computing the fragment size,
+	 * we need to first truncate the floating point result from
+	 *	sample rate * channels / intr rate
+	 * then adjust up to an even number, before multiplying it
+	 * with the sample size
+	 */
+	engp->fragsz = engp->fmt.sr * engp->fmt.ch / engp->intrate;
+	if (engp->fragsz & 1)
+		engp->fragsz++;
+	engp->fragsz *= engp->sampsz;
+	engp->fragfr = engp->fragsz / engp->framesz;
+
+	if (ashim_use_drvbuf) {
+		engp->bufsz = ((engp)->af_eflags & ENGINE_OUTPUT_CAP) ?
+		    statep->ad_infop->ad_play.ad_bsize :
+		    statep->ad_infop->ad_record.ad_bsize;
+
+		engp->nfrags = engp->bufsz / engp->fragsz;
+
+		/* adjust buf size to frag boundary */
+		if (engp->nfrags * engp->fragsz < engp->bufsz)
+			engp->nfrags++;
+
+		engp->bufsz = engp->nfrags * engp->fragsz;
+	} else {
+		if (ashim_bufscale < ASHIM_BUFSCALE_MIN ||
+		    ashim_bufscale > ASHIM_BUFSCALE_MAX)
+			engp->nfrags = ASHIM_BUFSCALE_DEF;
+		else
+			engp->nfrags = ashim_bufscale;
+		engp->bufsz = engp->fragsz * engp->nfrags;
+	}
+
+	engp->bufp = kmem_zalloc(engp->bufsz, KM_SLEEP);
+	engp->bufpos = engp->bufp;
+	engp->bufendp = engp->bufp + engp->bufsz;
+	engp->frames = 0;
+	engp->io_count = 0;
+	engp->bufio_count = 0;
+
+	*fragfrp = engp->fragfr;
+	*nfragsp = engp->nfrags;
+	*bufp = engp->bufp;
+
+	if (ashim_ad_setup(statep, dir) != AUDIO_SUCCESS) {
+		audio_dev_warn(statep->af_devp, "device setup failed");
+		goto OUT;
+	}
+	statep->flags |= AD_SETUP;
+
+	rv = 0;
+
+	dinfo("%s - %s: "
+	    "frames per frag: %u, frags in buffer: %u, frag size: %u, "
+	    "intr rate: %u, buffer size: %u, buffer: 0x%p - 0x%p\n",
+	    statep->dstr, engp->name,
+	    *fragfrp, *nfragsp, engp->fragsz,
+	    engp->intrate, engp->bufsz, *bufp, engp->bufendp);
+
+OUT:
+	mutex_exit(&engp->lock);
+	if (rv != 0)
+		ashim_af_close(arg);
+
+	return (rv);
+}
+
+
+static void
+ashim_af_close(void *arg)
+{
+	ashim_eng_t *engp = (ashim_eng_t *)arg;
+	ashim_state_t *statep = engp->statep;
+
+	mutex_enter(&engp->lock);
+
+	if (statep->flags & AD_SETUP) {
+		ashim_ad_teardown(statep, ASHIM_ENG_DIR(engp));
+		statep->flags &= ~AD_SETUP;
+	}
+
+	if (engp->bufp != NULL) {
+		kmem_free(engp->bufp, engp->bufsz);
+		engp->bufp = NULL;
+		engp->bufpos = NULL;
+		engp->bufendp = NULL;
+	}
+
+	ashim_prtstats(engp);
+	mutex_exit(&engp->lock);
+
+	usb_ac_close(statep->dip);
+}
+
+
+static int
+ashim_eng_start(ashim_eng_t *engp)
+{
+	ashim_state_t *statep = engp->statep;
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+	int (*start)(audiohdl_t);
+	int rv = 0;
+
+	start = ((engp)->af_eflags & ENGINE_OUTPUT_CAP) ?
+	    ad_entry->ad_start_play : ad_entry->ad_start_record;
+
+	dinfo("%s: starting device %s engine\n", statep->dstr, engp->name);
+
+	if ((*start)(AUDIO_SHIMST2HDL(statep)) != AUDIO_SUCCESS) {
+		audio_dev_warn(statep->af_devp, "failed to start %s engine",
+		    engp->name);
+		rv = EIO;
+	}
+
+	return (rv);
+}
+
+
+static void
+ashim_eng_stop(ashim_eng_t *engp)
+{
+	ashim_state_t *statep = engp->statep;
+	am_ad_entry_t *ad_entry = statep->ad_infop->ad_entry;
+	void (*stop)(audiohdl_t);
+
+	stop = ((engp)->af_eflags & ENGINE_OUTPUT_CAP) ?
+	    ad_entry->ad_stop_play : ad_entry->ad_stop_record;
+
+	dinfo("%s: stopping device %s engine\n", statep->dstr, engp->name);
+
+	(*stop)(AUDIO_SHIMST2HDL(statep));
+}
+
+
+static int
+ashim_af_start(void *arg)
+{
+	ashim_eng_t *engp = (ashim_eng_t *)arg;
+	int rv = EIO;
+
+	if (ashim_eng_disable)
+		return (rv);
+
+	mutex_enter(&engp->lock);
+	engp->flags |= ENG_STARTED;
+	mutex_exit(&engp->lock);
+
+	rv = ashim_eng_start(engp);
+
+	return (rv);
+}
+
+
+static void
+ashim_af_stop(void *arg)
+{
+	ashim_eng_t *engp = (ashim_eng_t *)arg;
+
+	mutex_enter(&engp->lock);
+	engp->flags &= ~ENG_STARTED;
+	mutex_exit(&engp->lock);
+
+	ashim_eng_stop(engp);
+}
+
+
+static uint64_t
+ashim_af_count(void *arg)
+{
+	ashim_eng_t	*engp = arg;
+	uint64_t	val;
+
+	mutex_enter(&engp->lock);
+	val = engp->frames;
+	mutex_exit(&engp->lock);
+
+	return (val);
+}
+
+
+static int
+ashim_af_format(void *arg)
+{
+	ashim_eng_t *engp = arg;
+
+	return (engp->af_fmt);
+}
+
+static int
+ashim_af_channels(void *arg)
+{
+	ashim_eng_t *engp = arg;
+
+	return (engp->fmt.ch);
+}
+
+
+static int
+ashim_af_rate(void *arg)
+{
+	ashim_eng_t *engp = arg;
+
+	return (engp->fmt.sr);
+}
+
+
+/*ARGSUSED*/
+static void
+ashim_af_sync(void *arg, unsigned nframes)
+{
+	/*
+	 * drivers will call ddi_dma_sync() themselves after requesting data
+	 * on playback and before sending data on record through the shim
+	 */
+}
+
+
+static size_t
+ashim_af_qlen(void *arg)
+{
+	ashim_eng_t *engp = (ashim_eng_t *)arg;
+
+	return (engp->fragfr);
+}
+
+
+/*
+ * **************************************************************************
+ * interfaces used by USB audio
+ */
+
+/*ARGSUSED*/
+int
+am_hw_state_change(audiohdl_t handle, int cmd, int dir, int value,
+    int sleep)
+{
+	ashim_state_t *statep = AUDIO_HDL2SHIMST(handle);
+	ashim_ctrl_t *ctrlp;
+	uint64_t cval = 0;
+	int64_t left, right, delta = 0;
+	int dcmd = AM_SET_GAIN;
+
+	/* only known HWSC command used */
+	if (cmd != AM_HWSC_SET_GAIN_DELTA) {
+		audio_dev_warn(statep->af_devp, "invalid HW state change "
+		    "command recieved");
+		return (AUDIO_FAILURE);
+	}
+
+	ctrlp = ashim_find_ctrl_dcmd(statep, dcmd, dir);
+	if (ctrlp == NULL) {
+		audio_dev_warn(statep->af_devp, "driver control command %d "
+		    "not found for HW state change command %d", dcmd, cmd);
+		return (AUDIO_FAILURE);
+	}
+
+	mutex_enter(&ctrlp->lock);
+
+	delta = D2F_GAIN(value);
+	left = AUDIO_CTRL_STEREO_LEFT(ctrlp->cval) + delta;
+	right = AUDIO_CTRL_STEREO_RIGHT(ctrlp->cval) + delta;
+
+	if (left > AF_MAX_GAIN)
+		left = AF_MAX_GAIN;
+	if (right > AF_MAX_GAIN)
+		right = AF_MAX_GAIN;
+
+	if (left < AF_MIN_GAIN)
+		left = AF_MIN_GAIN;
+	if (right < AF_MIN_GAIN)
+		right = AF_MIN_GAIN;
+
+	cval = AUDIO_CTRL_STEREO_VAL(left, right);
+	mutex_exit(&ctrlp->lock);
+
+	if (audio_control_write(ctrlp->af_ctrlp, cval)) {
+		audio_dev_warn(statep->af_devp, "updating "
+		    "control %s to value 0x%llx by driver failed",
+		    ctrlp->acd.acd_name, (long long unsigned)cval);
+		return (AUDIO_FAILURE);
+	}
+	return (AUDIO_SUCCESS);
+}
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.h b/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.h
new file mode 100644
index 0000000000..3ec22cda37
--- /dev/null
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_ac/audio_shim.h
@@ -0,0 +1,161 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+#ifndef	_AUDIO_SHIM_H
+#define	_AUDIO_SHIM_H
+
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio.h>
+#include <sys/audio/audio_support.h>
+#include <sys/mixer.h>
+#include <sys/audio/audio_mixer.h>
+
+#include <sys/audio/audio_driver.h>
+
+#ifdef	__cplusplus
+extern "C" {
+#endif
+
+typedef enum {DBG_WARN = 1, DBG_INFO, DBG_DETAIL} debug_level_t;
+
+void vdprint(debug_level_t lvl, const char *fmt, va_list adx);
+void dprint(debug_level_t lvl, const char *fmt, ...);
+void dwarn(const char *fmt, ...);
+void dinfo(const char *fmt, ...);
+void ddtl(const char *fmt, ...);
+
+
+int am_unregister(audiohdl_t handle);
+
+#define	CTRL_MUTED	0x1		/* muted output gain */
+
+#define	TQ_NM_MAX	64
+#define	DSTR_MAX	64
+
+typedef struct ashim_config_arg ashim_config_arg_t;
+typedef struct ashim_ctrl ashim_ctrl_t;
+typedef struct ashim_fmt ashim_fmt_t;
+typedef struct ashim_eng ashim_eng_t;
+typedef struct ashim_state ashim_state_t;
+
+struct ashim_config_arg {
+	int		cmd;
+	int		dir;
+	int		arg1;
+	int		arg2;
+};
+
+struct ashim_ctrl {
+	int			dcmd;		/* SADA command */
+	int			dir;		/* play or record */
+	audio_ctrl_desc_t	acd;		/* description */
+	audio_ctrl_t		*af_ctrlp;	/* framework handle */
+	uint64_t		cval;		/* current control value */
+	uint64_t		defval;		/* default control value */
+	audio_ctrl_wr_t		af_wr;		/* write callback */
+
+	int			flags;		/* CTRL_XXX flags */
+	kmutex_t		lock;
+
+	ashim_state_t		*statep;
+	ashim_ctrl_t		*nextp;
+};
+
+struct ashim_fmt {
+	int		sr;	/* sample rate */
+	uint_t		ch;	/* channels */
+	uint_t		prec;	/* precision */
+	uint_t		enc;	/* encoding */
+};
+
+struct ashim_eng {
+	ashim_state_t	*statep;
+	audio_engine_t	*af_engp;
+	int		af_eflags;	/* ENGINE_* flags */
+	int		af_fmt;		/* AUDIO_FORMAT_* flags */
+	ashim_fmt_t	fmt;
+
+	unsigned	intrate;	/* interrupt rate */
+	unsigned	sampsz;		/* sample size */
+	unsigned	framesz;	/* frame size */
+	unsigned	fragsz;		/* fragment size */
+	unsigned	nfrags;		/* number of fragments in buffer */
+	unsigned	fragfr;		/* number of frames per fragment */
+	unsigned	frsmshift;	/* right shift: frames in sample cnt */
+	unsigned	smszshift;	/* left shift: sample cnt * sampsz */
+
+	caddr_t		bufp;		/* I/O buf; framework to/from drv */
+	unsigned	bufsz;		/* buffer size */
+	caddr_t		bufpos;		/* buffer position */
+	caddr_t		bufendp;	/* end of buffer */
+
+	audio_prinfo_t	*prinfop;	/* SADA ad_defaults play/record */
+
+	uint64_t	frames;		/* total frames processed since open */
+	uint64_t	io_count;	/* i/o requests from the driver */
+	uint64_t	bufio_count;	/* i/o requests to the framework */
+	char		*name;
+
+#define	ENG_STARTED	0x1
+#define	ENG_ENABLED	0x10
+	int		flags;
+
+	kmutex_t	lock;
+};
+
+struct ashim_state {
+	dev_info_t	*dip;
+	void		*private;	/* private audio driver data */
+	audio_dev_t	*af_devp;
+	am_ad_info_t	*ad_infop;
+
+#define	ASHIM_ENG_MAX	2
+	ashim_eng_t	engines[ASHIM_ENG_MAX];
+	int		engcnt;
+
+	ashim_ctrl_t	*controls;
+
+	char		*devnm;
+	char		dstr[DSTR_MAX];
+
+#define	AF_REGISTERED	0x1
+#define	AD_SETUP	0x10
+	int		flags;
+};
+
+/*
+ * Macros used to convert between audio handles and the shim state structure.
+ */
+#define	AUDIO_HDL2SHIMST(hdl)		((ashim_state_t *)(hdl))
+#define	AUDIO_SHIMST2HDL(statep)	((audiohdl_t)(statep))
+
+
+#ifdef	__cplusplus
+}
+#endif
+
+#endif /* _AUDIO_SHIM_H */
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.c b/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.c
index a26ba4b87b..57e9c3c0bd 100644
--- a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.c
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.c
@@ -23,13 +23,19 @@
  * Use is subject to license terms.
  */
 
-
 /*
- * AUDIO CONTROL Driver: usb_ac is a streams multiplexor that sits
- * on top of usb_as and hid and is responsible for
- * (1) providing the entry points to audio mixer framework, (2) passing
- * streams messages to and from usb_as and hid and (3) processing
- * control messages that it can handle.
+ * AUDIO CONTROL Driver:
+ * This driver is derived from the legacy SADA streams-based usb_ac driver
+ * and serves as an intermediate measure before the full conversion to the
+ * to the Boomer framework in a follow-on phase of the Boomer project, which
+ * will utilize more comprehensive USB audio features as well.  Multiplexor
+ * plumbing functionality that used to be in the usb_ac_dacf DACF module is
+ * now located here.
+ *
+ * usb_ac is a multiplexor that sits on top of usb_as and hid and is
+ * responsible for (1) providing the entry points to audio mixer framework,
+ * (2) passing control commands to and from usb_as and hid and (3) processing
+ * control messages from hid/usb_ah that it can handle.
  *
  * 1. Mixer entry points are: usb_ac_setup(), usb_ac_teardown(),
  *	usb_ac_set_config(), usb_ac_set_format(), usb_ac_start_play(),
@@ -43,10 +49,8 @@
  *	of sound control interface of the audio device in a synchronous
  *	manner.
  *
- * Serialization: usb_ac being a streams driver and having the requirement
- * of making non-blockings calls (USBA or streams or mixer) needs to drop
- * mutexes over such calls. But at the same time, a competing thread
- * can't be allowed to interfere with (1) pipe, (2) streams state.
+ * Serialization: A competing thread can't be allowed to interfere with
+ * (1) pipe, (2) streams state.
  * So we need some kind of serialization among the asynchronous
  * threads that can run in the driver. The serialization is mostly
  * needed to avoid races among open/close/events/power entry points
@@ -58,9 +62,6 @@
  * PM model in usb_ac: Raise power during attach. If a device is not at full
  * power, raise power in the entry points. After the command is over,
  * pm_idle_component() is called. The power is lowered in detach().
- *
- * locking: Warlock is not aware of the automatic locking mechanisms for
- * streams drivers.
  */
 #include <sys/usb/usba/usbai_version.h>
 #include <sys/usb/usba.h>
@@ -69,11 +70,12 @@
 #include <sys/ndi_impldefs.h>
 #include <sys/strsubr.h>
 #include <sys/strsun.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/sunldi.h>
 
 #include <sys/audio.h>
-#include <sys/audiovar.h>
 #include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
 #include <sys/mixer.h>
 #include <sys/audio/audio_mixer.h>
 
@@ -81,18 +83,22 @@
 #include <sys/usb/clients/audio/usb_mixer.h>
 #include <sys/usb/clients/audio/usb_ac/usb_ac.h>
 
+/* for getting the minor node info from hid */
+#include <sys/usb/clients/hid/hidminor.h>
+#include <sys/usb/clients/audio/usb_as/usb_as.h>
+
+#include "audio_shim.h"
+
 /* debug support */
 uint_t	usb_ac_errlevel 	= USB_LOG_L4;
 uint_t	usb_ac_errmask		= (uint_t)-1;
 uint_t	usb_ac_instance_debug	= (uint_t)-1;
 
-#ifdef DEBUG
 /*
- * tunable timeout for usb_as response, allow at least 10 secs for control
- * cmd to timeout
+ * wait period in seconds for the HID message processing thread
+ * used primarily to check when the stream has closed
  */
-int	usb_ac_wait_timeout = 10000000;
-#endif
+uint_t usb_ac_wait_hid = 1;
 
 /*
  * table for converting term types of input and output terminals
@@ -143,28 +149,19 @@ static struct {
 /*
  * Module linkage routines for the kernel
  */
-static int	usb_ac_getinfo(dev_info_t *, ddi_info_cmd_t, void *, void **);
 static int	usb_ac_attach(dev_info_t *, ddi_attach_cmd_t);
 static int	usb_ac_detach(dev_info_t *, ddi_detach_cmd_t);
 static int	usb_ac_power(dev_info_t *, int, int);
 
-/*
- * STREAMS module entry points
- */
-static int	usb_ac_open(queue_t *, dev_t *, int, int, cred_t *);
-static int	usb_ac_close(queue_t *, int, cred_t *);
-static int	usb_ac_uwput(queue_t *, mblk_t *);
-static int	usb_ac_lrput(queue_t *, mblk_t *);
-
 /* plumbing */
 static usb_ac_plumbed_t *usb_ac_get_plumb_info(usb_ac_state_t *, char *,
 				uchar_t);
-static usb_ac_plumbed_t *usb_ac_get_plumb_info_from_lrq(usb_ac_state_t *,
-				queue_t *);
 static uint_t	usb_ac_get_featureID(usb_ac_state_t *, uchar_t, uint_t,
 				uint_t);
-static void	usb_ac_plumb_ioctl(queue_t *, mblk_t *);
 
+/* module entry points */
+int		usb_ac_open(dev_info_t *);
+void		usb_ac_close(dev_info_t *);
 
 /* registration */
 static int	usb_ac_get_curr_n_channels(usb_ac_state_t *, int);
@@ -246,36 +243,52 @@ static int	usb_ac_set_mute(usb_ac_state_t *, uint_t, uint_t,
 static int	usb_ac_set_volume(usb_ac_state_t *, uint_t, short, int dir,
 				int);
 static int	usb_ac_get_maxmin_volume(usb_ac_state_t *, uint_t, int, int,
-				int);
-static void	usb_ac_free_mblk(mblk_t *);
-static mblk_t	*usb_ac_allocate_req_mblk(usb_ac_state_t *, int,
-				void *, uint_t);
+				int, short *);
 static int	usb_ac_send_as_cmd(usb_ac_state_t *, usb_ac_plumbed_t *,
 				int, void *);
-static int	usb_ac_send_format_cmd(audiohdl_t, int, int, int,
-				int, int, int);
-static int	usb_ac_do_setup(audiohdl_t, int, int);
-static void	usb_ac_do_teardown(audiohdl_t, int, int);
-static void	usb_ac_do_pause_play(audiohdl_t, int);
-static void	usb_ac_do_stop_record(audiohdl_t, int);
+static int	usb_ac_send_format_cmd(audiohdl_t, int, int, int, int, int);
+static int	usb_ac_do_setup(audiohdl_t, int);
+static void	usb_ac_do_teardown(audiohdl_t, int);
+static void	usb_ac_do_stop_play(audiohdl_t);
+static void	usb_ac_do_stop_record(audiohdl_t);
 
 /*  Mixer entry points */
-static int	usb_ac_setup(audiohdl_t, int, int);
-static void	usb_ac_teardown(audiohdl_t, int, int);
-static int	usb_ac_set_config(audiohdl_t, int, int, int, int, int);
-static int	usb_ac_set_format(audiohdl_t, int, int, int, int, int, int);
-static int	usb_ac_start_play(audiohdl_t, int);
-static void	usb_ac_pause_play(audiohdl_t, int);
-static void	usb_ac_stop_play(audiohdl_t, int);
-static int	usb_ac_start_record(audiohdl_t, int);
-static void	usb_ac_stop_record(audiohdl_t, int);
+static int	usb_ac_setup(audiohdl_t, int);
+static void	usb_ac_teardown(audiohdl_t, int);
+static int	usb_ac_set_config(audiohdl_t, int, int, int, int);
+static int	usb_ac_set_format(audiohdl_t, int, int, int, int, int);
+static int	usb_ac_start_play(audiohdl_t);
+static void	usb_ac_stop_play(audiohdl_t);
+static int	usb_ac_start_record(audiohdl_t);
+static void	usb_ac_stop_record(audiohdl_t);
 static int	usb_ac_restore_audio_state(usb_ac_state_t *, int);
 
 /*
- * External functions
+ * Mux
  */
-extern int	space_store(char *key, uintptr_t ptr);
-extern void	space_free(char *);
+static int	usb_ac_mux_walk_siblings(usb_ac_state_t *);
+static void	usb_ac_print_reg_data(usb_ac_state_t *,
+				usb_as_registration_t *);
+static int	usb_ac_get_reg_data(usb_ac_state_t *, ldi_handle_t, int);
+static int	usb_ac_setup_plumbed(usb_ac_state_t *, int, int, int);
+static int	usb_ac_mixer_registration(usb_ac_state_t *);
+static void	usb_ac_hold_siblings(usb_ac_state_t *);
+static int	usb_ac_online_siblings(usb_ac_state_t *);
+static void	usb_ac_rele_siblings(usb_ac_state_t *);
+static int	usb_ac_mux_plumbing(usb_ac_state_t *);
+static void	usb_ac_mux_plumbing_tq(void *);
+static int	usb_ac_mux_unplumbing(usb_ac_state_t *);
+static void	usb_ac_mux_unplumbing_tq(void *);
+static int	usb_ac_plumb(usb_ac_plumbed_t *);
+static void	usb_ac_unplumb(usb_ac_plumbed_t *);
+static void	usb_ac_reader(void *);
+static int	usb_ac_read_msg(usb_ac_plumbed_t *, mblk_t *);
+static int	usb_ac_do_plumbing(usb_ac_state_t *);
+static int	usb_ac_do_unplumbing(usb_ac_state_t *);
+
+/* just generic, USB Audio, 1.0 spec-compliant */
+static audio_device_t usb_dev_info =
+	{ {"USB Audio"}, {"1.0"}, {"external"} };
 
 
 /*
@@ -287,120 +300,30 @@ static am_ad_entry_t usb_ac_entry = {
 	usb_ac_set_config,	/* ad_set_config() */
 	usb_ac_set_format,	/* ad_set_format() */
 	usb_ac_start_play,	/* ad_start_play() */
-	usb_ac_pause_play,	/* ad_pause_play() */
 	usb_ac_stop_play,	/* ad_stop_play() */
 	usb_ac_start_record,	/* ad_start_record() */
 	usb_ac_stop_record,	/* ad_stop_record() */
-	NULL,			/* ad_ioctl() */
-	NULL			/* ad_iocdata() */
 };
 
 /* anchor for soft state structures */
-static void	*usb_ac_statep;
-
-/* for passing soft state etc. to usb_ac_dacf module */
-static usb_ac_state_space_t ssp;
-
-/* STREAMS driver id and limit value structure */
-static struct module_info usb_ac_modinfo = {
-	0xffff,				/* module ID number */
-	"usb_ac",			/* module name */
-	USB_AUDIO_MIN_PKTSZ,		/* minimum packet size */
-	USB_AUDIO_MAX_PKTSZ,		/* maximum packet size */
-	USB_AC_HIWATER,			/* high water mark */
-	USB_AC_LOWATER			/* low water mark */
-};
+void	*usb_ac_statep;
 
-/* STREAMS queue processing procedures structures */
-/* upper read queue */
-static struct qinit usb_ac_urqueue = {
-	NULL,			/* put procedure */
-	NULL,			/* service procedure */
-	usb_ac_open,		/* open procedure */
-	usb_ac_close,		/* close procedure */
-	NULL,			/* unused */
-	&usb_ac_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* upper write queue */
-static struct qinit usb_ac_uwqueue = {
-	usb_ac_uwput,		/* put procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&usb_ac_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* lower read queue */
-static struct qinit usb_ac_lrqueue = {
-	usb_ac_lrput,
-	NULL,
-	NULL,
-	NULL,
-	NULL,
-	&usb_ac_modinfo,	/* module parameters */
-	NULL
-};
-
-/* lower write queue */
-static struct qinit usb_ac_lwqueue = {
-	NULL,
-	NULL,
-	NULL,
-	NULL,
-	NULL,
-	&usb_ac_modinfo,	/* module parameters */
-	NULL
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab usb_ac_str_info = {
-	&usb_ac_urqueue,		/* upper read queue */
-	&usb_ac_uwqueue,		/* upper write queue */
-	&usb_ac_lrqueue,		/* lower read queue */
-	&usb_ac_lwqueue,		/* lower write queue */
-};
 
 /*
  * DDI Structures
- *
- * Entry points structure
  */
-static struct cb_ops usb_ac_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&usb_ac_str_info,	/* cb_str */
-	D_MP | D_MTPERQ,	/* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_arwite */
-};
 
 /* Device operations structure */
 static struct dev_ops usb_ac_dev_ops = {
 	DEVO_REV,		/* devo_rev */
 	0,			/* devo_refcnt */
-	usb_ac_getinfo,		/* devo_getinfo */
+	NULL,			/* devo_getinfo */
 	nulldev,		/* devo_identify - obsolete */
 	nulldev,		/* devo_probe - not needed */
 	usb_ac_attach,		/* devo_attach */
 	usb_ac_detach,		/* devo_detach */
 	nodev,			/* devo_reset */
-	&usb_ac_cb_ops,		/* devi_cb_ops */
+	NULL,			/* devi_cb_ops */
 	NULL,			/* devo_busb_ac_ops */
 	usb_ac_power,		/* devo_power */
 	ddi_quiesce_not_needed,	/* devo_quiesce */
@@ -420,12 +343,6 @@ static struct modlinkage usb_ac_modlinkage = {
 	NULL				/* NULL terminates the list */
 };
 
-/* warlock directives */
-_NOTE(SCHEME_PROTECTS_DATA("unique per call", iocblk))
-_NOTE(SCHEME_PROTECTS_DATA("unique per call", datab))
-_NOTE(SCHEME_PROTECTS_DATA("unique per call", msgb))
-_NOTE(SCHEME_PROTECTS_DATA("unique per call", queue))
-_NOTE(SCHEME_PROTECTS_DATA("stable data", usb_pipe_policy_t))
 
 /* standard entry points */
 int
@@ -439,19 +356,11 @@ _init(void)
 		return (rval);
 	}
 
+	audio_init_ops(&usb_ac_dev_ops, "usb_ac");
+
 	if ((rval = mod_install(&usb_ac_modlinkage)) != 0) {
 		ddi_soft_state_fini(&usb_ac_statep);
-	}
-
-	if (!rval) {
-		ssp.sp = usb_ac_statep;
-		ssp.restore_func = usb_ac_restore_audio_state;
-		ssp.get_featureID_func = usb_ac_get_featureID;
-		ssp.ac_entryp = &usb_ac_entry;
-		ssp.pm_busy_component = usb_ac_pm_busy_component;
-		ssp.pm_idle_component = usb_ac_pm_idle_component;
-
-		rval = space_store("usb_ac", (uintptr_t)&ssp);
+		audio_fini_ops(&usb_ac_dev_ops);
 	}
 
 	return (rval);
@@ -466,7 +375,7 @@ _fini(void)
 	if ((rval = mod_remove(&usb_ac_modlinkage)) == 0) {
 		/* Free the soft state internal structures */
 		ddi_soft_state_fini(&usb_ac_statep);
-		space_free("usb_ac");
+		audio_fini_ops(&usb_ac_dev_ops);
 	}
 
 	return (rval);
@@ -479,39 +388,6 @@ _info(struct modinfo *modinfop)
 	return (mod_info(&usb_ac_modlinkage, modinfop));
 }
 
-/*ARGSUSED*/
-static int
-usb_ac_getinfo(dev_info_t *dip, ddi_info_cmd_t infocmd,
-			void *arg, void **result)
-{
-	usb_ac_state_t	   *uacp = NULL;
-	int error = DDI_FAILURE;
-	int instance;
-
-	switch (infocmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((uacp = ddi_get_soft_state(usb_ac_statep,
-		    instance)) != NULL) {
-			*result = uacp->usb_ac_dip;
-			if (*result != NULL) {
-				error = DDI_SUCCESS;
-			}
-		} else {
-			*result = NULL;
-		}
-		break;
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void *)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-	default:
-		break;
-	}
-
-	return (error);
-}
 
 extern	uint_t		nproc;
 #define	INIT_PROCESS_CNT 3
@@ -520,11 +396,7 @@ static int
 usb_ac_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 {
 	usb_ac_state_t		*uacp = NULL;
-	audio_sup_reg_data_t	reg_data;
 	int			instance = ddi_get_instance(dip);
-	int			minor;
-	char			*key;
-	size_t			key_len, len;
 
 	switch (cmd) {
 		case DDI_ATTACH:
@@ -580,6 +452,9 @@ usb_ac_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 	uacp->usb_ac_instance = instance;
 	uacp->usb_ac_dip = dip;
 
+	(void) snprintf(uacp->dstr, sizeof (uacp->dstr), "%s#%d",
+	    ddi_driver_name(dip), instance);
+
 	if (usb_client_attach(dip, USBDRV_VERSION, 0) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
 		    "usb_client_attach failed");
@@ -608,46 +483,7 @@ usb_ac_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 
 	uacp->usb_ac_default_ph = uacp->usb_ac_dev_data->dev_default_ph;
 
-	/* register with audiosup */
-	reg_data.asrd_version	= AUDIOSUP_VERSION;
-
-	/*
-	 * we register with pathname, the mgf, product, and serial number
-	 * strings, vid.pid, and driver name which should be pretty unique
-	 */
-	key_len = 2 * MAXNAMELEN;
-	if (uacp->usb_ac_dev_data->dev_mfg) {
-		key_len += strlen(uacp->usb_ac_dev_data->dev_mfg);
-	}
-	if (uacp->usb_ac_dev_data->dev_product) {
-		key_len += strlen(uacp->usb_ac_dev_data->dev_product);
-	}
-	if (uacp->usb_ac_dev_data->dev_serial) {
-		key_len += strlen(uacp->usb_ac_dev_data->dev_serial);
-	}
-
-	key = kmem_alloc(key_len, KM_SLEEP);
-	(void) ddi_pathname(dip, key);
-
-	len = strlen(key);
-	(void) snprintf(&key[len], key_len - len, ",%s,%s,%s,%x.%x,%s",
-	    (uacp->usb_ac_dev_data->dev_mfg ?
-	    uacp->usb_ac_dev_data->dev_mfg : "-"),
-	    (uacp->usb_ac_dev_data->dev_product ?
-	    uacp->usb_ac_dev_data->dev_product : "-"),
-	    (uacp->usb_ac_dev_data->dev_serial ?
-	    uacp->usb_ac_dev_data->dev_serial : "-"),
-	    uacp->usb_ac_dev_data->dev_descr->idVendor,
-	    uacp->usb_ac_dev_data->dev_descr->idProduct,
-	    ddi_driver_name(dip));
-
-	reg_data.asrd_key = key;
-
-	USB_DPRINTF_L3(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "registering with key: %s", key);
-
-	uacp->usb_ac_audiohdl = audio_sup_register(dip, &reg_data);
-	kmem_free(key, key_len);
+	uacp->usb_ac_audiohdl = audio_sup_register(dip);
 
 	if (uacp->usb_ac_audiohdl == NULL) {
 		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
@@ -668,28 +504,9 @@ usb_ac_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 	/* we no longer need the descr tree */
 	usb_free_descr_tree(dip, uacp->usb_ac_dev_data);
 
-	/* read .conf file properties */
-	uacp->usb_ac_mixer_mode_enable = ddi_prop_get_int(DDI_DEV_T_ANY,
-	    dip, DDI_PROP_DONTPASS, "mixer-enabled", 1);
-
 	uacp->usb_ac_ser_acc = usb_init_serialization(dip,
 	    USB_INIT_SER_CHECK_SAME_THREAD);
 
-	/* create minor node */
-	minor = audio_sup_construct_minor(uacp->usb_ac_audiohdl, USER1);
-
-	USB_DPRINTF_L3(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "minor=%d", minor);
-
-	if ((ddi_create_minor_node(dip, "mux", S_IFCHR,
-	    minor, NULL, 0)) != DDI_SUCCESS) {
-		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-		    "usb_ac_attach: couldn't create minor node mux");
-
-		goto fail;
-	}
-	uacp->usb_ac_mux_minor = minor;
-
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	/* we are online */
@@ -715,17 +532,28 @@ usb_ac_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 		goto fail;
 	}
 
+	USB_DPRINTF_L4(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
+	    "usb_ac_attach: End");
+
 	/* report device */
 	ddi_report_dev(dip);
 
-	USB_DPRINTF_L4(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "usb_ac_attach: End");
+	if (usb_ac_do_plumbing(uacp) != USB_SUCCESS)
+		goto fail;
 
 	return (DDI_SUCCESS);
+
 fail:
 	if (uacp) {
 		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
 		    "attach failed");
+
+		/* wait for plumbing thread to finish */
+		if (uacp->tqp != NULL) {
+			ddi_taskq_wait(uacp->tqp);
+			ddi_taskq_destroy(uacp->tqp);
+			uacp->tqp = NULL;
+		}
 		(void) usb_ac_cleanup(dip, uacp);
 	}
 
@@ -738,19 +566,67 @@ usb_ac_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
 {
 	int instance = ddi_get_instance(dip);
 	usb_ac_state_t	*uacp;
-	int rval;
+	int rval = USB_FAILURE;
+	char *nm = "usb_ac_detach";
 
 	uacp = ddi_get_soft_state(usb_ac_statep, instance);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_detach:");
-
 	switch (cmd) {
 	case DDI_DETACH:
-		rval = usb_ac_cleanup(dip, uacp);
+		dinfo("%s: %s - detaching\n", uacp->dstr, nm);
+
+		/* wait for plumbing thread to finish */
+		if (uacp->tqp != NULL)
+			ddi_taskq_wait(uacp->tqp);
+
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		/* do not allow detach if still busy */
+		if (uacp->usb_ac_busy_count) {
+			dinfo("%s: %s - still busy: %d\n", uacp->dstr, nm,
+			    uacp->usb_ac_busy_count);
+
+			mutex_exit(&uacp->usb_ac_mutex);
+			return (USB_FAILURE);
+		}
+
+		/*
+		 * call am_unregister() to stop calls into our driver from
+		 * the framework; can fail if the framework is still busy
+		 */
+		if (uacp->usb_ac_audiohdl != NULL) {
+			mutex_exit(&uacp->usb_ac_mutex);
+
+			if (am_unregister(uacp->usb_ac_audiohdl) !=
+			    AUDIO_SUCCESS) {
+				dinfo("%s: %s - am_unregister failed, "
+				    "framework still busy\n",
+				    uacp->dstr, nm);
+
+				return (USB_FAILURE);
+			}
+			mutex_enter(&uacp->usb_ac_mutex);
+		}
+
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		/*
+		 * unplumb to stop activity from other modules, then
+		 * cleanup, which will also teardown audio framework state
+		 */
+		if (usb_ac_do_unplumbing(uacp) == USB_SUCCESS)
+			rval = usb_ac_cleanup(dip, uacp);
+
+		if (rval != USB_SUCCESS) {
+			USB_DPRINTF_L2(PRINT_MASK_ATTA,
+			    uacp->usb_ac_log_handle, "detach failed: %s%d",
+			    ddi_driver_name(dip), instance);
+		}
 
 		return ((rval == USB_SUCCESS) ? DDI_SUCCESS : DDI_FAILURE);
 	case DDI_SUSPEND:
+		dinfo("%s: %s - suspending\n", uacp->dstr, nm);
+
 		rval = usb_ac_cpr_suspend(dip);
 
 		return ((rval == USB_SUCCESS) ? DDI_SUCCESS : DDI_FAILURE);
@@ -776,8 +652,7 @@ usb_ac_cleanup(dev_info_t *dip, usb_ac_state_t *uacp)
 	mutex_enter(&uacp->usb_ac_mutex);
 	uacpm = uacp->usb_ac_pm;
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_cleanup: uacpm=0x%p", (void *)uacpm);
+	dinfo("%s: usb_ac_cleanup: uacpm=0x%p\n", uacp->dstr, (void *)uacpm);
 
 	ASSERT(uacp->usb_ac_busy_count == 0);
 
@@ -858,8 +733,6 @@ usb_ac_cleanup(dev_info_t *dip, usb_ac_state_t *uacp)
 
 	usb_fini_serialization(uacp->usb_ac_ser_acc);
 
-	ddi_remove_minor_node(dip, NULL);
-
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 	    "usb_ac_cleanup: Ending");
 
@@ -877,275 +750,79 @@ usb_ac_cleanup(dev_info_t *dip, usb_ac_state_t *uacp)
 }
 
 
-/*
- * usb_ac_open:
- *	Open entry point. Called on the plumbing minor node or
- *	audio or audioctl minor nodes which we pass to audio_sup_open()
- *	We do not raise power here and wait for the setup callback
- */
-/*ARGSUSED*/
-static int
-usb_ac_open(queue_t *q, dev_t *devp, int flag, int sflag, cred_t *credp)
-{
-	int minor = getminor(*devp);
-	int instance;
-	int rval;
-	usb_ac_state_t	*uacp;
-
-	instance = audio_sup_devt_to_instance(*devp);
-
-	uacp = ddi_get_soft_state(usb_ac_statep, instance);
-	if (uacp == NULL) {
-
-		return (ENXIO);
-	}
-
-	mutex_enter(&uacp->usb_ac_mutex);
-	uacp->usb_ac_busy_count++; /* This will prevent unplumbing */
-
-	USB_DPRINTF_L4(PRINT_MASK_OPEN, uacp->usb_ac_log_handle,
-	    "usb_ac_open: Begin q=0x%p, minor=0x%x instance=%d "
-	    "open cnt=%d", (void *)q, minor, instance, uacp->usb_ac_busy_count);
-
-	if (sflag) {
-		USB_DPRINTF_L2(PRINT_MASK_OPEN, uacp->usb_ac_log_handle,
-		    "usb_ac_open: clone open not supported");
-
-		uacp->usb_ac_busy_count--;
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		return (ENXIO);
-	}
-
-	if (minor == uacp->usb_ac_mux_minor) {
-
-		USB_DPRINTF_L4(PRINT_MASK_OPEN, uacp->usb_ac_log_handle,
-		    "usb_ac_open: opening mux");
-		/*
-		 * This is the plumbing open, initiated during attach/
-		 * connect_event_callback/cpr_resume/first user open.
-		 */
-		uacp->usb_ac_busy_count--;
-
-		/* Save the dev_t value of pluming q to use for lower q's */
-		uacp->usb_ac_dev = *devp;
-		audio_sup_set_qptr(q, *devp, (void *)uacp);
-
-		/* Initialize the queue pointers */
-		uacp->usb_ac_rq = q;
-		uacp->usb_ac_wq = WR(q);
-
-		/* release mutex while making streams framework call */
-		mutex_exit(&uacp->usb_ac_mutex);
-		qprocson(q);
-		mutex_enter(&uacp->usb_ac_mutex);
-
-	} else if (uacp->usb_ac_plumbing_state != USB_AC_STATE_PLUMBED) {
-		uacp->usb_ac_busy_count--;
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		return (EIO);
-	} else {
-		/* pass the open to audio_sup_open so SADA can do its work */
-		USB_DPRINTF_L4(PRINT_MASK_OPEN, uacp->usb_ac_log_handle,
-		    "usb_ac_open: calling audio_sup_open, q=0x%p, open_cnt=%d",
-		    (void *)q, uacp->usb_ac_busy_count);
-
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		/*
-		 * go to full power
-		 */
-		usb_ac_pm_busy_component(uacp);
-		(void) pm_raise_power(uacp->usb_ac_dip, 0, USB_DEV_OS_FULL_PWR);
-
-		rval = audio_sup_open(q, devp, flag, sflag, credp);
-
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		if (rval != 0) {
-			USB_DPRINTF_L4(PRINT_MASK_OPEN,
-			    uacp->usb_ac_log_handle,
-			    "audio_sup_open rval=%d", rval);
-
-			uacp->usb_ac_busy_count--;
-
-			mutex_exit(&uacp->usb_ac_mutex);
-
-			usb_ac_pm_idle_component(uacp);
-
-			return (rval);
-		}
-	}
-
-	USB_DPRINTF_L4(PRINT_MASK_OPEN, uacp->usb_ac_log_handle,
-	    "usb_ac_open: End q=0x%p, open cnt=%d",
-	    (void *)q, uacp->usb_ac_busy_count);
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	return (0);
-}
-
-
-/*
- * usb_ac_close :
- *	Close entry point
- */
-/*ARGSUSED*/
-static int
-usb_ac_close(queue_t *q, int flag, cred_t *credp)
+int
+usb_ac_open(dev_info_t *dip)
 {
-	dev_t dev = audio_sup_get_qptr_dev(q);
-	int minor = getminor(dev);
-	int instance = audio_sup_get_qptr_instance(q);
-	usb_ac_state_t *uacp = ddi_get_soft_state(usb_ac_statep, instance);
-	int rval;
+	int inst = ddi_get_instance(dip);
+	usb_ac_state_t *uacp = ddi_get_soft_state(usb_ac_statep, inst);
 
 	mutex_enter(&uacp->usb_ac_mutex);
 
-	USB_DPRINTF_L4(PRINT_MASK_CLOSE, uacp->usb_ac_log_handle,
-	    "usb_ac_close: Begin q=0x%p, opencount=%d",
-	    (void *)q, uacp->usb_ac_busy_count);
-
-	/* closing the mux? */
-	if (minor == uacp->usb_ac_mux_minor) {
-		USB_DPRINTF_L4(PRINT_MASK_CLOSE, uacp->usb_ac_log_handle,
-		    "usb_ac_close: closing mux plumbing stream");
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		/* Wait till all activity in the default pipe has drained */
-		usb_ac_serialize_access(uacp);
-		usb_ac_release_access(uacp);
-
-		audio_sup_free_qptr(q);
-		qprocsoff(q);
-
-		return (0);
-	}
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	rval = audio_sup_close(q, flag, credp);
-
-	if (rval != 0) {
-		USB_DPRINTF_L2(PRINT_MASK_CLOSE, uacp->usb_ac_log_handle,
-		    "audio_sup_close fails %d", rval);
-
-		return (rval);
-	}
-
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	/* normal streams closing */
-	ASSERT(uacp->usb_ac_plumbing_state >= USB_AC_STATE_PLUMBED);
-
-	uacp->usb_ac_busy_count --;
-
-	USB_DPRINTF_L4(PRINT_MASK_CLOSE, uacp->usb_ac_log_handle,
-	    "usb_ac_close: End rval=%d q=0x%p, opencount=%d",
-	    rval, (void *)q, uacp->usb_ac_busy_count);
+	uacp->usb_ac_busy_count++;
 
 	mutex_exit(&uacp->usb_ac_mutex);
 
-	usb_ac_pm_idle_component(uacp);
+	usb_ac_pm_busy_component(uacp);
+	(void) pm_raise_power(uacp->usb_ac_dip, 0, USB_DEV_OS_FULL_PWR);
 
 	return (0);
 }
 
 
-/*
- * usb_ac_uwput:
- *	write put entry point for the upper mux. Only PLUMB/UNPLUMB ioctls
- *	are processed here. All other ioctls are passed to audio_sup routines
- *	for further processing.
- */
-static int
-usb_ac_uwput(queue_t *q, mblk_t *mp)
+void
+usb_ac_close(dev_info_t *dip)
 {
-	int instance = audio_sup_get_qptr_instance(q);
-	usb_ac_state_t	*uacp = ddi_get_soft_state(usb_ac_statep, instance);
-	int error = DDI_SUCCESS;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_uwput: q=0x%p, mp=0x%p", (void *)q, (void *)mp);
-
-	ASSERT(mp != NULL);
-	ASSERT(mp->b_datap != NULL);
+	int inst = ddi_get_instance(dip);
+	usb_ac_state_t *uacp = ddi_get_soft_state(usb_ac_statep, inst);
 
 	mutex_enter(&uacp->usb_ac_mutex);
-	if (uacp->usb_ac_wq == q) {
-		ASSERT(mp->b_datap->db_type == M_IOCTL);
 
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		/* ioctl from plumbing thread (namely P_LINK) */
-		usb_ac_plumb_ioctl(q, mp);
+	if (uacp->usb_ac_busy_count > 0)
+		uacp->usb_ac_busy_count--;
 
-		return (error);
-	}
 	mutex_exit(&uacp->usb_ac_mutex);
 
-	/* Pass to audio_sup routine */
-	(void) audio_sup_wput(q, mp);
-
-	return (error);
+	usb_ac_pm_idle_component(uacp);
 }
 
 
 /*
- * usb_ac_lrput:
- *	read put entry point for the lower mux. Get the response from the
- *	lower module, signal usb_ac_send_as_cmd(), the thread that is waiting
- *	for a response to a message sent earlier anbd pass the response
- *	message	block.
+ * usb_ac_read_msg:
+ *	Handle asynchronous response from opened streams
  */
 static int
-usb_ac_lrput(queue_t *q, mblk_t *mp)
+usb_ac_read_msg(usb_ac_plumbed_t *plumb_infop, mblk_t *mp)
 {
-	int instance = audio_sup_get_qptr_instance(q);
-	usb_ac_state_t	*uacp;
+	usb_ac_state_t	*uacp = plumb_infop->acp_uacp;
 	int error = DDI_SUCCESS;
-	usb_ac_plumbed_t *plumb_infop;
-	usb_ac_streams_info_t *streams_infop = NULL;
 	int	val;
 	char	val1;
 	struct iocblk *iocp;
 
-	uacp = ddi_get_soft_state(usb_ac_statep, instance);
-
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_lrput: q=0x%p, mp=0x%p, instance=%d",
-	    (void *)q, (void *)mp, instance);
-	ASSERT(mp != NULL);
+	    "usb_ac_read_msg: mp=0x%p, instance=%d", (void *)mp,
+	    ddi_get_instance(uacp->usb_ac_dip));
 
-	mutex_enter(&uacp->usb_ac_mutex);
-	plumb_infop = usb_ac_get_plumb_info_from_lrq(uacp, q);
-	ASSERT(plumb_infop != NULL);
+	ASSERT(mp != NULL);
+	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
 
+	/*
+	 * typically an M_CTL is used between modules but in order to pass
+	 * through the streamhead, an M_PROTO type must be used instead
+	 */
 	switch (mp->b_datap->db_type) {
-	case M_CTL:
+	case M_PROTO:
 	case M_ERROR:
 		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "M_CTL/M_ERROR");
 
 		switch (plumb_infop->acp_driver) {
-		case USB_AS_PLUMBED:
-			USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "reply from usb_as, lrq=0x%p", (void *)q);
-			streams_infop = (usb_ac_streams_info_t *)
-			    plumb_infop->acp_data;
-			ASSERT(streams_infop != NULL);
-			streams_infop->acs_ac_to_as_req.acr_reply_mp = mp;
-			streams_infop->acs_ac_to_as_req.acr_wait_flag = 0;
-			cv_signal(&streams_infop->acs_ac_to_as_req.acr_cv);
-
-			break;
 		case USB_AH_PLUMBED:
 			USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "M_CTL from hid, lrq=0x%p", (void *)q);
+			    "message from hid, instance=%d",
+			    ddi_get_instance(plumb_infop->acp_dip));
 
-			iocp = (struct iocblk *)mp->b_rptr;
+			iocp = (struct iocblk *)(void *)mp->b_rptr;
 			ASSERT(mp->b_cont != NULL);
 
 			if (uacp->usb_ac_registered_with_mixer) {
@@ -1187,7 +864,7 @@ usb_ac_lrput(queue_t *q, mblk_t *mp)
 			break;
 		default:
 			USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "M_CTL from unknown module(%s)",
+			    "message from unknown module(%s)",
 			    ddi_driver_name(plumb_infop->acp_dip));
 			freemsg(mp);
 		}
@@ -1196,15 +873,11 @@ usb_ac_lrput(queue_t *q, mblk_t *mp)
 	default:
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "Unknown type=%d", mp->b_datap->db_type);
-		usb_ac_free_mblk(mp);
+		freemsg(mp);
 	}
-	mutex_exit(&uacp->usb_ac_mutex);
 
-	/*
-	 * Nobody is waiting; nothing to send up.
-	 */
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_lrput: done");
+	    "usb_ac_read_msg: done");
 
 	return (error);
 }
@@ -1408,121 +1081,6 @@ usb_ac_create_pm_components(dev_info_t *dip, usb_ac_state_t *uacp)
 
 
 /*
- * usb_ac_plumb_ioctl:
- *	IOCTL issued from plumbing thread (only P_LINK_LH/P_UNLINK for now
- *	caused by ldi_ioctl). Maybe we will need to use this function
- *	to issue other IOCTLS to children in future from plumbing thread
- */
-static void
-usb_ac_plumb_ioctl(queue_t *q, mblk_t *mp)
-{
-	int		instance = audio_sup_get_qptr_instance(q);
-	usb_ac_state_t	*uacp = ddi_get_soft_state(usb_ac_statep, instance);
-	struct iocblk	*iocp;
-	struct linkblk	*linkp;
-	int		n;
-	usb_ac_streams_info_t *streams_infop;
-
-	ASSERT(uacp != NULL);
-	ASSERT(mp != NULL);
-	ASSERT(mp->b_cont != NULL);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_plumb_ioctl, q=0x%p mp=0x%p instance=%d",
-	    (void *)q, (void *)mp, instance);
-
-	iocp = (struct iocblk *)mp->b_rptr;
-	mutex_enter(&uacp->usb_ac_mutex);
-	n = uacp->usb_ac_current_plumbed_index;
-
-	switch (iocp->ioc_cmd) {
-	case I_PLINK:
-		linkp = (struct linkblk *)mp->b_cont->b_rptr;
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "LINK ioctl, index=%d linkblk ptr=0x%p", n, (void *)linkp);
-
-		/*
-		 * We keep track of the module that is being
-		 * currently plumbed through usb_ac_current_plumbed_index
-		 * to the plumb structure array. We set the lwq field
-		 * of the plumb structure here.
-		 */
-		ASSERT(uacp->usb_ac_plumbed[n].acp_lwq == NULL);
-		uacp->usb_ac_plumbed[n].acp_lwq = linkp->l_qbot;
-		uacp->usb_ac_plumbed[n].acp_lrq = RD(linkp->l_qbot);
-
-		audio_sup_set_qptr(uacp->usb_ac_plumbed[n].acp_lrq,
-		    uacp->usb_ac_dev, (void *)uacp);
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "index=%d lwq=0x%p lrq=0x%p", n, (void *)linkp->l_qbot,
-		    (void *)RD(linkp->l_qbot));
-		break;
-	case I_UNLINK:
-	case I_PUNLINK:
-		linkp = (struct linkblk *)mp->b_cont->b_rptr;
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "UNLINK ioctl, linkblk ptr=0x%p", (void *)linkp);
-
-		audio_sup_free_qptr(RD(linkp->l_qbot));
-		uacp->usb_ac_dev = 0;
-
-		if (uacp->usb_ac_plumbed[n].acp_driver == USB_AS_PLUMBED) {
-
-			/*
-			 * we bzero the streams info and plumbed structure
-			 * since there is no guarantee that the next plumbing
-			 * will be identical
-			 */
-			streams_infop = (usb_ac_streams_info_t *)
-			    uacp->usb_ac_plumbed[n].acp_data;
-			cv_destroy(&(streams_infop->acs_ac_to_as_req.acr_cv));
-
-			/* bzero the relevant plumbing structure */
-			bzero(streams_infop, sizeof (usb_ac_streams_info_t));
-		}
-		bzero(&uacp->usb_ac_plumbed[n], sizeof (usb_ac_plumbed_t));
-
-		iocp->ioc_count = 0;
-		break;
-	default:
-		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "Unknown ioctl, cmd=%d", iocp->ioc_cmd);
-		iocp->ioc_error = EINVAL;
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		goto iocnak;
-	}
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	/*
-	 * Common exit path for calls that return a positive
-	 * acknowledgment with a return value of 0.
-	 */
-	iocp->ioc_rval = 0;
-	iocp->ioc_error = 0;
-	mp->b_datap->db_type = M_IOCACK;
-	qreply(q, mp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_plumb_ioctl: End (ACK)");
-
-	return;
-
-iocnak:
-
-	iocp->ioc_rval = 0;
-	mp->b_datap->db_type = M_IOCNAK;
-	qreply(q, mp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_plumb_ioctl: End: (NAK)");
-}
-
-
-/*
  * usb_ac_get_plumb_info:
  *	Get plumb_info pointer that matches module "name"
  *	If name = "usb_as", match the direction also (record or play)
@@ -1565,26 +1123,6 @@ usb_ac_get_plumb_info(usb_ac_state_t *uacp, char *name, uchar_t reg_play_type)
 
 
 /*
- * usb_ac_get_pinfo_from_lrq:
- *	Get plumb_info pointer that matches the lrq passed
- */
-static usb_ac_plumbed_t *
-usb_ac_get_plumb_info_from_lrq(usb_ac_state_t *uacp, queue_t *lrq)
-{
-	int	n;
-
-	for (n = 0; n < USB_AC_MAX_PLUMBED; n++) {
-		if (uacp->usb_ac_plumbed[n].acp_lrq == lrq) {
-
-			return (&uacp->usb_ac_plumbed[n]);
-		}
-	}
-
-	return (NULL);
-}
-
-
-/*
  * usb_ac_get_featureID:
  *	find out if there is at least one feature unit that supports
  *	the request controls.
@@ -2945,9 +2483,9 @@ usb_ac_disconnect_event_cb(dev_info_t *dip)
 {
 	usb_ac_state_t *uacp = (usb_ac_state_t *)ddi_get_soft_state(
 	    usb_ac_statep, ddi_get_instance(dip));
+	char *nm = "usb_ac_disconnect_event_cb";
 
-	USB_DPRINTF_L4(PRINT_MASK_EVENTS, uacp->usb_ac_log_handle,
-	    "usb_ac_disconnect_event_cb: dip=0x%p", (void *)dip);
+	dinfo("%s: %s - dip=%p\n", uacp->dstr, nm, (void *)dip);
 
 	usb_ac_serialize_access(uacp);
 
@@ -2962,10 +2500,8 @@ usb_ac_disconnect_event_cb(dev_info_t *dip)
 	}
 	mutex_exit(&uacp->usb_ac_mutex);
 
-	USB_DPRINTF_L3(PRINT_MASK_EVENTS, uacp->usb_ac_log_handle,
-	    "usb_ac_disconnect_event_cb: done");
-
 	usb_ac_release_access(uacp);
+	dinfo("%s: %s done\n", uacp->dstr, nm);
 
 	return (USB_SUCCESS);
 }
@@ -3008,9 +2544,9 @@ usb_ac_reconnect_event_cb(dev_info_t *dip)
 {
 	usb_ac_state_t *uacp = (usb_ac_state_t *)ddi_get_soft_state(
 	    usb_ac_statep, ddi_get_instance(dip));
+	char *nm = "usb_ac_reconnect_event_cb";
 
-	USB_DPRINTF_L4(PRINT_MASK_EVENTS, uacp->usb_ac_log_handle,
-	    "usb_ac_reconnect_event_cb: dip=0x%p", (void *)dip);
+	dinfo("%s: %s - dip=%p\n", uacp->dstr, nm, (void *)dip);
 
 	mutex_enter(&uacp->usb_ac_mutex);
 	mutex_exit(&uacp->usb_ac_mutex);
@@ -3032,9 +2568,11 @@ usb_ac_reconnect_event_cb(dev_info_t *dip)
 		USB_DPRINTF_L0(PRINT_MASK_EVENTS, uacp->usb_ac_log_handle,
 		    "busy device has been reconnected");
 	}
+
 	mutex_exit(&uacp->usb_ac_mutex);
 
 	usb_ac_release_access(uacp);
+	dinfo("%s: %s done\n", uacp->dstr, nm);
 
 	return (USB_SUCCESS);
 }
@@ -3149,8 +2687,7 @@ usb_ac_am_restore_state(void *arg)
 		 */
 		delay(USB_AC_RESTORE_DELAY);
 
-		(void) audio_sup_restore_state(uacp->usb_ac_audiohdl,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH);
+		audio_sup_restore_state(uacp->usb_ac_audiohdl);
 
 		mutex_enter(&uacp->usb_ac_mutex);
 		uacp->usb_ac_plumbing_state = USB_AC_STATE_PLUMBED;
@@ -3233,18 +2770,15 @@ usb_ac_restore_audio_state(usb_ac_state_t *uacp, int flag)
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static int
-usb_ac_setup(audiohdl_t ahdl, int stream, int flag)
+usb_ac_setup(audiohdl_t ahdl, int flag)
 {
 	int	rval = AUDIO_SUCCESS;
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_setup: Begin ahdl=0x%p, stream=%d, flag=%d",
-	    (void *)ahdl, stream, flag);
-
 	mutex_enter(&uacp->usb_ac_mutex);
+
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
 
@@ -3255,18 +2789,15 @@ usb_ac_setup(audiohdl_t ahdl, int stream, int flag)
 	usb_ac_serialize_access(uacp);
 
 	if (flag & AUDIO_PLAY) {
-		rval = usb_ac_do_setup(ahdl, stream, AUDIO_PLAY);
+		rval = usb_ac_do_setup(ahdl, AUDIO_PLAY);
 	}
 
 	if ((rval == USB_SUCCESS) && (flag & AUDIO_RECORD)) {
-		rval = usb_ac_do_setup(ahdl, stream, AUDIO_RECORD);
+		rval = usb_ac_do_setup(ahdl, AUDIO_RECORD);
 	}
 
 	usb_ac_release_access(uacp);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_setup: rval=%d", rval);
-
 	return ((rval == USB_SUCCESS) ? AUDIO_SUCCESS : AUDIO_FAILURE);
 }
 
@@ -3277,7 +2808,7 @@ usb_ac_setup(audiohdl_t ahdl, int stream, int flag)
  *	either from audio framework for usb_ac_set_format
  */
 static int
-usb_ac_do_setup(audiohdl_t ahdl, int stream, int flag)
+usb_ac_do_setup(audiohdl_t ahdl, int flag)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 	usb_ac_plumbed_t	*plumb_infop = NULL;
@@ -3286,10 +2817,6 @@ usb_ac_do_setup(audiohdl_t ahdl, int stream, int flag)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_setup: Begin ahdl=0x%p, stream=%d, flag=%d",
-	    (void *)ahdl, stream, flag);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	dir = (flag & AUDIO_PLAY) ? AUDIO_PLAY : AUDIO_RECORD;
@@ -3320,20 +2847,14 @@ usb_ac_do_setup(audiohdl_t ahdl, int stream, int flag)
 		    "usb_ac_do_setup: failure");
 
 		streams_infop->acs_setup_teardown_count--;
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		return (USB_FAILURE);
 	}
 
-	streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
-
 	mutex_exit(&uacp->usb_ac_mutex);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_setup: End");
-
 	return (USB_SUCCESS);
 }
 
@@ -3345,30 +2866,23 @@ usb_ac_do_setup(audiohdl_t ahdl, int stream, int flag)
  *	NOTE: allow teardown when disconnected
  */
 static void
-usb_ac_teardown(audiohdl_t ahdl, int stream, int flag)
+usb_ac_teardown(audiohdl_t ahdl, int flag)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_teardown: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	usb_ac_serialize_access(uacp);
 
 	if (flag & AUDIO_PLAY) {
-		usb_ac_do_teardown(ahdl, stream, AUDIO_PLAY);
+		usb_ac_do_teardown(ahdl, AUDIO_PLAY);
 	}
 
 	if (flag & AUDIO_RECORD) {
-		usb_ac_do_teardown(ahdl, stream, AUDIO_RECORD);
+		usb_ac_do_teardown(ahdl, AUDIO_RECORD);
 	}
 
 	usb_ac_release_access(uacp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_teardown: End");
 }
 
 
@@ -3377,7 +2891,7 @@ usb_ac_teardown(audiohdl_t ahdl, int stream, int flag)
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static void
-usb_ac_do_teardown(audiohdl_t ahdl, int stream, int flag)
+usb_ac_do_teardown(audiohdl_t ahdl, int flag)
 {
 	usb_ac_state_t		*uacp = audio_sup_get_private(ahdl);
 	usb_ac_plumbed_t	*plumb_infop = NULL;
@@ -3386,10 +2900,6 @@ usb_ac_do_teardown(audiohdl_t ahdl, int stream, int flag)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_teardown: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	dir = (flag & AUDIO_PLAY) ? AUDIO_PLAY : AUDIO_RECORD;
@@ -3426,15 +2936,12 @@ usb_ac_do_teardown(audiohdl_t ahdl, int stream, int flag)
 		    "usb_ac_do_teardown: failure");
 
 		streams_infop->acs_setup_teardown_count++;
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		return;
 	}
 
-	streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
-
 	mutex_exit(&uacp->usb_ac_mutex);
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
@@ -3450,8 +2957,7 @@ usb_ac_do_teardown(audiohdl_t ahdl, int stream, int flag)
  *	Check power is done here since we will access the default pipe
  */
 static int
-usb_ac_set_config(audiohdl_t ahdl, int stream, int command, int flag,
-	int arg1, int arg2)
+usb_ac_set_config(audiohdl_t ahdl, int command, int flag, int arg1, int arg2)
 {
 	usb_ac_state_t	*uacp = audio_sup_get_private(ahdl);
 	char		*what;
@@ -3463,12 +2969,8 @@ usb_ac_set_config(audiohdl_t ahdl, int stream, int command, int flag,
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_set_config: Begin ahdl=0x%p\n\t"
-	    "stream=%d, cmd=%d, flag=%d, arg1=%d, arg2=%d",
-	    (void *)ahdl, stream, command, flag, arg1, arg2);
-
 	mutex_enter(&uacp->usb_ac_mutex);
+
 	if (uacp->usb_ac_plumbing_state < USB_AC_STATE_PLUMBED) {
 		mutex_exit(&uacp->usb_ac_mutex);
 
@@ -3606,10 +3108,6 @@ usb_ac_set_config(audiohdl_t ahdl, int stream, int command, int flag,
 		what = "mic boost";
 		rval = AUDIO_SUCCESS;
 		break;
-	case AM_SET_GAIN_BAL:
-		what = "set gain bal";
-		rval = AUDIO_FAILURE;
-		break;
 	default:
 		what = "unknown";
 		rval = AUDIO_FAILURE;
@@ -3737,8 +3235,8 @@ usb_ac_set_gain(usb_ac_state_t *uacp, uint_t featureID,
 		return (USB_FAILURE);
 	}
 
-	if ((max = usb_ac_get_maxmin_volume(uacp, channel,
-	    USB_AUDIO_GET_MAX, dir, featureID)) == USB_FAILURE) {
+	if (usb_ac_get_maxmin_volume(uacp, channel,
+	    USB_AUDIO_GET_MAX, dir, featureID, &max) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_set_gain: getting max gain failed");
 
@@ -3748,8 +3246,8 @@ usb_ac_set_gain(usb_ac_state_t *uacp, uint_t featureID,
 	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 	    "usb_ac_set_gain: channel %d, max=%d", channel, max);
 
-	if ((min = usb_ac_get_maxmin_volume(uacp, channel,
-	    USB_AUDIO_GET_MIN, dir, featureID)) == USB_FAILURE) {
+	if (usb_ac_get_maxmin_volume(uacp, channel,
+	    USB_AUDIO_GET_MIN, dir, featureID, &min) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_set_gain: getting min gain failed");
 
@@ -3759,8 +3257,8 @@ usb_ac_set_gain(usb_ac_state_t *uacp, uint_t featureID,
 	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 	    "usb_ac_set_gain: channel=%d, min=%d", channel, min);
 
-	if ((current = usb_ac_get_maxmin_volume(uacp, channel,
-	    USB_AUDIO_GET_CUR, dir, featureID)) == USB_FAILURE) {
+	if (usb_ac_get_maxmin_volume(uacp, channel,
+	    USB_AUDIO_GET_CUR, dir, featureID, &current) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_set_gain: getting cur gain failed");
 
@@ -3798,8 +3296,8 @@ usb_ac_set_gain(usb_ac_state_t *uacp, uint_t featureID,
 	}
 
 	/* just curious, read it back, device may round up/down */
-	if ((current = usb_ac_get_maxmin_volume(uacp, channel,
-	    USB_AUDIO_GET_CUR, dir, featureID)) == USB_FAILURE) {
+	if (usb_ac_get_maxmin_volume(uacp, channel,
+	    USB_AUDIO_GET_CUR, dir, featureID, &current) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_set_gain: getting cur gain failed");
 	}
@@ -3827,7 +3325,7 @@ usb_ac_set_gain(usb_ac_state_t *uacp, uint_t featureID,
  *	usb_ac_send_as_cmd().
  */
 static int
-usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
+usb_ac_set_format(audiohdl_t ahdl, int flag,
 	int sample, int channels, int precision, int encoding)
 {
 	usb_ac_state_t		*uacp = audio_sup_get_private(ahdl);
@@ -3841,11 +3339,6 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_set_format: Begin ahdl=0x%p, stream=%d, flag=%d, "
-	    "sr=%d, chnls=%d, prec=%d, enc=%d", (void *)ahdl, stream, flag,
-	    sample, channels, precision, encoding);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
@@ -3879,7 +3372,7 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 
 		mutex_exit(&uacp->usb_ac_mutex);
 
-		rval = usb_ac_send_format_cmd(ahdl, stream, dir, sample,
+		rval = usb_ac_send_format_cmd(ahdl, dir, sample,
 		    channels, precision, encoding);
 
 		usb_ac_release_access(uacp);
@@ -3903,15 +3396,15 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 	mutex_exit(&uacp->usb_ac_mutex);
 
 	if (dir == AUDIO_PLAY) {
-		usb_ac_do_pause_play(ahdl, stream);
+		usb_ac_do_stop_play(ahdl);
 	} else if (dir == AUDIO_RECORD) {
-		usb_ac_do_stop_record(ahdl, stream);
+		usb_ac_do_stop_record(ahdl);
 	}
 
 	/* This blocks until the current isoc xfer is over */
-	usb_ac_do_teardown(ahdl, stream, dir);
+	usb_ac_do_teardown(ahdl, dir);
 
-	if (usb_ac_send_format_cmd(ahdl, stream, dir, sample,
+	if (usb_ac_send_format_cmd(ahdl, dir, sample,
 	    channels, precision, encoding) != USB_SUCCESS) {
 		/*
 		 * Setting new alternate has failed, try restoring
@@ -3920,13 +3413,13 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 		 * till bandwidth is available. Also we know that
 		 * there is a matching alternate, so that can't fail.
 		 */
-		if (usb_ac_send_format_cmd(ahdl, stream, dir,
+		if (usb_ac_send_format_cmd(ahdl, dir,
 		    old_format.fmt_sr, old_format.fmt_chns,
 		    old_format.fmt_precision, old_format.fmt_encoding) ==
 		    USB_FAILURE) {
 
 			/* We closed the pipe; reopen it */
-			(void) usb_ac_do_setup(ahdl, stream, dir);
+			(void) usb_ac_do_setup(ahdl, dir);
 
 			mutex_enter(&uacp->usb_ac_mutex);
 			streams_infop->acs_setup_teardown_count =
@@ -3940,7 +3433,7 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 	}
 
 	/* This should block until successful */
-	(void) usb_ac_do_setup(ahdl, stream, dir);
+	(void) usb_ac_do_setup(ahdl, dir);
 
 	mutex_enter(&uacp->usb_ac_mutex);
 	streams_infop->acs_setup_teardown_count = old_setup_teardown_count;
@@ -3948,9 +3441,6 @@ usb_ac_set_format(audiohdl_t ahdl, int stream, int flag,
 
 	usb_ac_release_access(uacp);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_set_format: End");
-
 	return (AUDIO_SUCCESS);
 }
 
@@ -4002,7 +3492,7 @@ usb_ac_get_curr_format(usb_ac_state_t *uacp, int dir)
  *	Send the set sample freq command down to usb_as.
  */
 static int
-usb_ac_send_format_cmd(audiohdl_t ahdl, int stream, int dir,
+usb_ac_send_format_cmd(audiohdl_t ahdl, int dir,
 	int sample, int channels, int precision, int encoding)
 {
 	usb_ac_state_t		*uacp = audio_sup_get_private(ahdl);
@@ -4012,11 +3502,6 @@ usb_ac_send_format_cmd(audiohdl_t ahdl, int stream, int dir,
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_send_format_cmd: Begin ahdl=0x%p, stream=%d, dir=%d, "
-	    "sr=%d, chnls=%d, prec=%d, enc=%d", (void *)ahdl, stream, dir,
-	    sample, channels, precision, encoding);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
@@ -4059,13 +3544,13 @@ usb_ac_send_format_cmd(audiohdl_t ahdl, int stream, int dir,
 		USB_DPRINTF_L2(PRINT_MASK_ALL,
 		    uacp->usb_ac_log_handle,
 		    "usb_ac_send_format_cmd: failed");
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		return (USB_FAILURE);
 	} else {
-		/* alternate number stored and reply mp freed */
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
+		/* store matching alternate number */
+		streams_infop->acs_ac_to_as_req.acr_curr_format.fmt_alt =
+		    format->fmt_alt;
 	}
 
 	/* Set the sample rate */
@@ -4074,19 +3559,13 @@ usb_ac_send_format_cmd(audiohdl_t ahdl, int stream, int dir,
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_send_format_cmd: setting format failed");
 
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		return (USB_FAILURE);
 	}
 
-	streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
-
 	mutex_exit(&uacp->usb_ac_mutex);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_send_format_cmd: End");
-
 	return (USB_SUCCESS);
 }
 
@@ -4097,7 +3576,7 @@ usb_ac_send_format_cmd(audiohdl_t ahdl, int stream, int dir,
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static int
-usb_ac_start_play(audiohdl_t ahdl, int stream)
+usb_ac_start_play(audiohdl_t ahdl)
 {
 	usb_ac_state_t		*uacp = audio_sup_get_private(ahdl);
 	usb_audio_formats_t	*cur_fmt;
@@ -4108,10 +3587,6 @@ usb_ac_start_play(audiohdl_t ahdl, int stream)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_start_play: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
@@ -4156,7 +3631,6 @@ usb_ac_start_play(audiohdl_t ahdl, int stream)
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_start_play: failure");
 
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		usb_ac_release_access(uacp);
@@ -4164,34 +3638,26 @@ usb_ac_start_play(audiohdl_t ahdl, int stream)
 		return (AUDIO_FAILURE);
 	}
 
-	streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 	mutex_exit(&uacp->usb_ac_mutex);
 
 	usb_ac_release_access(uacp);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_start_play: End");
-
 	return (AUDIO_SUCCESS);
 }
 
 
 /*
- * usb_ac_pause_play:
- *	Wrapper function for usb_ac_do_pause_play and gets
+ * usb_ac_stop_play:
+ *	Wrapper function for usb_ac_do_stop_play and gets
  *	called from mixer framework.
  */
 static void
-usb_ac_pause_play(audiohdl_t ahdl, int stream)
+usb_ac_stop_play(audiohdl_t ahdl)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_pause_play: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
@@ -4201,11 +3667,8 @@ usb_ac_pause_play(audiohdl_t ahdl, int stream)
 	mutex_exit(&uacp->usb_ac_mutex);
 
 	usb_ac_serialize_access(uacp);
-	usb_ac_do_pause_play(ahdl, stream);
+	usb_ac_do_stop_play(ahdl);
 	usb_ac_release_access(uacp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_pause_play: End");
 }
 
 /*
@@ -4214,7 +3677,7 @@ usb_ac_pause_play(audiohdl_t ahdl, int stream)
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static void
-usb_ac_do_pause_play(audiohdl_t ahdl, int stream)
+usb_ac_do_stop_play(audiohdl_t ahdl)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 	usb_ac_plumbed_t	*plumb_infop = NULL;
@@ -4222,10 +3685,6 @@ usb_ac_do_pause_play(audiohdl_t ahdl, int stream)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_pause_play: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	plumb_infop = usb_ac_get_plumb_info(uacp, "usb_as", AUDIO_PLAY);
@@ -4240,37 +3699,9 @@ usb_ac_do_pause_play(audiohdl_t ahdl, int stream)
 
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_do_pause_play: failure");
-
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 	}
 
 	mutex_exit(&uacp->usb_ac_mutex);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_pause_play: End");
-}
-
-
-/*
- * usb_ac_stop_play:
- *	Wrapper function for usb_ac_pause_play	and gets
- *	called from mixer framework.
- */
-static void
-usb_ac_stop_play(audiohdl_t ahdl, int stream)
-{
-	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
-
-	ASSERT(uacp != NULL);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_stop_play: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
-	usb_ac_pause_play(ahdl, stream);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_stop_play: End");
 }
 
 
@@ -4280,7 +3711,7 @@ usb_ac_stop_play(audiohdl_t ahdl, int stream)
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static int
-usb_ac_start_record(audiohdl_t ahdl, int stream)
+usb_ac_start_record(audiohdl_t ahdl)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 	usb_ac_plumbed_t	*plumb_infop = NULL;
@@ -4288,10 +3719,6 @@ usb_ac_start_record(audiohdl_t ahdl, int stream)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_start_record: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 	if (uacp->usb_ac_dev_state != USB_DEV_ONLINE) {
 		mutex_exit(&uacp->usb_ac_mutex);
@@ -4316,7 +3743,6 @@ usb_ac_start_record(audiohdl_t ahdl, int stream)
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_start_record: failure");
 
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 		mutex_exit(&uacp->usb_ac_mutex);
 
 		usb_ac_release_access(uacp);
@@ -4328,9 +3754,6 @@ usb_ac_start_record(audiohdl_t ahdl, int stream)
 
 	usb_ac_release_access(uacp);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_start_record: End");
-
 	return (AUDIO_SUCCESS);
 }
 
@@ -4341,22 +3764,15 @@ usb_ac_start_record(audiohdl_t ahdl, int stream)
  *	called form mixer framework.
  */
 static void
-usb_ac_stop_record(audiohdl_t ahdl, int stream)
+usb_ac_stop_record(audiohdl_t ahdl)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_stop_record: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	usb_ac_serialize_access(uacp);
-	usb_ac_do_stop_record(ahdl, stream);
+	usb_ac_do_stop_record(ahdl);
 	usb_ac_release_access(uacp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_stop_record: End");
 }
 
 
@@ -4366,7 +3782,7 @@ usb_ac_stop_record(audiohdl_t ahdl, int stream)
  *	Check power is done in usb_ac_send_as_cmd()
  */
 static void
-usb_ac_do_stop_record(audiohdl_t ahdl, int stream)
+usb_ac_do_stop_record(audiohdl_t ahdl)
 {
 	usb_ac_state_t *uacp = audio_sup_get_private(ahdl);
 	usb_ac_plumbed_t	*plumb_infop = NULL;
@@ -4374,10 +3790,6 @@ usb_ac_do_stop_record(audiohdl_t ahdl, int stream)
 
 	ASSERT(uacp != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_stop_record: Begin ahdl=0x%p, stream=%d",
-	    (void *)ahdl, stream);
-
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	plumb_infop = usb_ac_get_plumb_info(uacp, "usb_as", AUDIO_RECORD);
@@ -4392,14 +3804,9 @@ usb_ac_do_stop_record(audiohdl_t ahdl, int stream)
 
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 		    "usb_ac_do_stop_record: failure");
-
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
 	}
 
 	mutex_exit(&uacp->usb_ac_mutex);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_do_stop_record: End");
 }
 
 
@@ -4413,12 +3820,12 @@ usb_ac_do_stop_record(audiohdl_t ahdl, int stream)
  */
 static int
 usb_ac_get_maxmin_volume(usb_ac_state_t *uacp, uint_t channel, int cmd,
-    int dir, int feature_unitID)
+    int dir, int feature_unitID, short *max_or_minp)
 {
 	mblk_t		*data = NULL;
-	short		max_or_min;
 	usb_cr_t	cr;
 	usb_cb_flags_t	cb_flags;
+
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
 	    "usb_ac_get_maxmin_volume: channel=%d, cmd=%d dir=%d",
 	    channel, cmd, dir);
@@ -4454,14 +3861,14 @@ usb_ac_get_maxmin_volume(usb_ac_state_t *uacp, uint_t channel, int cmd,
 	mutex_enter(&uacp->usb_ac_mutex);
 	ASSERT(MBLKL(data) == 2);
 
-	max_or_min = (*(data->b_rptr+1) << 8) | *data->b_rptr;
+	*max_or_minp = (*(data->b_rptr+1) << 8) | *data->b_rptr;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_get_maxmin_volume: max_or_min=0x%x", max_or_min);
+	    "usb_ac_get_maxmin_volume: max_or_min=0x%x", *max_or_minp);
 
 	freemsg(data);
 
-	return (max_or_min);
+	return (USB_SUCCESS);
 }
 
 
@@ -4581,11 +3988,11 @@ static int
 usb_ac_send_as_cmd(usb_ac_state_t *uacp, usb_ac_plumbed_t *plumb_infop,
     int cmd, void *arg)
 {
-	mblk_t		*mp = NULL;
-	struct iocblk	*iocp;
-	queue_t 	*lwq = plumb_infop->acp_lwq;
 	usb_ac_streams_info_t *streams_infop;
-	int		error = USB_FAILURE;
+	int		rv;
+	int		rval;
+	ldi_handle_t	lh;
+	char		*nm = "usb_ac_send_as_cmd";
 
 	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
 	ASSERT(plumb_infop != NULL);
@@ -4593,276 +4000,1209 @@ usb_ac_send_as_cmd(usb_ac_state_t *uacp, usb_ac_plumbed_t *plumb_infop,
 	streams_infop = (usb_ac_streams_info_t *)plumb_infop->acp_data;
 	ASSERT(streams_infop != NULL);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_send_as_cmd: Begin lwq=0x%p, cmd=0x%x, arg=0x%p",
-	    (void *)lwq, cmd, arg);
+	dinfo("%s: %s - cmd=0x%x, arg=0x%p\n", uacp->dstr, nm, cmd, arg);
 
-	if (!canputnext(lwq)) {
-		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "usb_ac_send_as_cmd: canputnext failed");
+	lh = plumb_infop->acp_lh;
 
-		return (error);
+	rv = ldi_ioctl(lh, cmd, (intptr_t)arg, FKIOCTL, kcred, &rval);
+	if (rv != 0) {
+		dinfo("%s: %s - ldi_ioctl failed, error=%d\n", uacp->dstr, nm,
+		    rv);
+
+		return (USB_FAILURE);
 	}
 
-	/*
-	 * Allocate mblk for a particular command
-	 */
-	switch (cmd) {
-	case USB_AUDIO_SET_FORMAT:
-		mp = usb_ac_allocate_req_mblk(uacp, cmd, (void *)arg,
-		    sizeof (usb_audio_formats_t));
-		break;
-	case USB_AUDIO_TEARDOWN:
-	case USB_AUDIO_STOP_RECORD:
-	case USB_AUDIO_PAUSE_PLAY:
-	case USB_AUDIO_SETUP:
-		mp = usb_ac_allocate_req_mblk(uacp, cmd, NULL, 0);
-		break;
-	case USB_AUDIO_START_RECORD:
-		mp = usb_ac_allocate_req_mblk(uacp, cmd, (void *)arg,
-		    sizeof (audiohdl_t *));
-		break;
-	case USB_AUDIO_SET_SAMPLE_FREQ:
-		mp = usb_ac_allocate_req_mblk(uacp, cmd, (void *)arg,
-		    sizeof (int));
-		break;
-	case USB_AUDIO_START_PLAY:
-		mp = usb_ac_allocate_req_mblk(uacp, cmd, (void *)arg,
-		    sizeof (usb_audio_play_req_t));
-		break;
-	default:
+	return (USB_SUCCESS);
+}
+
+
+/*
+ * usb_ac_serialize/release_access:
+ */
+static void
+usb_ac_serialize_access(usb_ac_state_t	*uacp)
+{
+	(void) usb_serialize_access(uacp->usb_ac_ser_acc, USB_WAIT, 0);
+}
+
+static void
+usb_ac_release_access(usb_ac_state_t *uacp)
+{
+	usb_release_access(uacp->usb_ac_ser_acc);
+}
+
+
+static void
+usb_ac_pm_busy_component(usb_ac_state_t *usb_ac_statep)
+{
+	ASSERT(!mutex_owned(&usb_ac_statep->usb_ac_mutex));
+
+	if (usb_ac_statep->usb_ac_pm != NULL) {
+		mutex_enter(&usb_ac_statep->usb_ac_mutex);
+		usb_ac_statep->usb_ac_pm->acpm_pm_busy++;
+
+		USB_DPRINTF_L4(PRINT_MASK_PM,
+		    usb_ac_statep->usb_ac_log_handle,
+		    "usb_ac_pm_busy_component: %d",
+		    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+
+		mutex_exit(&usb_ac_statep->usb_ac_mutex);
+
+		if (pm_busy_component(usb_ac_statep->usb_ac_dip, 0) !=
+		    DDI_SUCCESS) {
+			mutex_enter(&usb_ac_statep->usb_ac_mutex);
+			usb_ac_statep->usb_ac_pm->acpm_pm_busy--;
+
+			USB_DPRINTF_L2(PRINT_MASK_PM,
+			    usb_ac_statep->usb_ac_log_handle,
+			    "usb_ac_pm_busy_component failed: %d",
+			    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+
+			mutex_exit(&usb_ac_statep->usb_ac_mutex);
+		}
+	}
+}
+
+
+static void
+usb_ac_pm_idle_component(usb_ac_state_t *usb_ac_statep)
+{
+	ASSERT(!mutex_owned(&usb_ac_statep->usb_ac_mutex));
+
+	if (usb_ac_statep->usb_ac_pm != NULL) {
+		if (pm_idle_component(usb_ac_statep->usb_ac_dip, 0) ==
+		    DDI_SUCCESS) {
+			mutex_enter(&usb_ac_statep->usb_ac_mutex);
+			ASSERT(usb_ac_statep->usb_ac_pm->acpm_pm_busy > 0);
+			usb_ac_statep->usb_ac_pm->acpm_pm_busy--;
+
+			USB_DPRINTF_L4(PRINT_MASK_PM,
+			    usb_ac_statep->usb_ac_log_handle,
+			    "usb_ac_pm_idle_component: %d",
+			    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+
+			mutex_exit(&usb_ac_statep->usb_ac_mutex);
+		}
+	}
+}
+
+
+/*
+ * handle read from plumbed drivers
+ */
+static void
+usb_ac_reader(void *argp)
+{
+	usb_ac_plumbed_t *acp = (usb_ac_plumbed_t *)argp;
+	usb_ac_state_t *uacp = acp->acp_uacp;
+	ldi_handle_t lh;
+	mblk_t *mp;
+	int rv;
+	timestruc_t tv = {0};
+
+	mutex_enter(&uacp->usb_ac_mutex);
+	lh = acp->acp_lh;
+	tv.tv_sec = usb_ac_wait_hid;
+
+	while (acp->acp_flags & ACP_ENABLED) {
+		mp = NULL;
+
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		rv = ldi_getmsg(lh, &mp, &tv);
+
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		if ((acp->acp_flags & ACP_ENABLED) && mp != NULL && rv == 0)
+			rv = usb_ac_read_msg(acp, mp);
+
+		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "%s%d read from %s%d, rv=%d",
+		    ddi_driver_name(uacp->usb_ac_dip),
+		    ddi_get_instance(uacp->usb_ac_dip),
+		    ddi_driver_name(acp->acp_dip),
+		    ddi_get_instance(acp->acp_dip), rv);
+	}
+	mutex_exit(&uacp->usb_ac_mutex);
+}
+
+
+/*
+ * setup threads to read from the other usb modules that may send unsolicited
+ * or asynchronous messages, which is only hid currently
+ */
+static int
+usb_ac_plumb(usb_ac_plumbed_t *acp)
+{
+	usb_ac_state_t	*uacp = acp->acp_uacp;
+	dev_info_t	*dip;
+	dev_info_t	*acp_dip;
+	int		acp_inst;
+	char		*acp_name;
+	char		tq_nm[128];
+	int		rv = USB_FAILURE;
+	char		*nm = "usb_ac_plumb";
+
+	mutex_enter(&uacp->usb_ac_mutex);
+
+	dip = uacp->usb_ac_dip;
+
+	acp_dip = acp->acp_dip;
+	acp_inst = ddi_get_instance(acp_dip);
+	acp_name = (char *)ddi_driver_name(acp_dip);
+
+	dinfo("%s: %s on %s%d\n", uacp->dstr, nm, acp_name, acp_inst);
+
+	if (strcmp(acp_name, "hid") != 0) {
+		rv = USB_SUCCESS;
+		goto OUT;
+	}
+
+	(void) snprintf(tq_nm, sizeof (tq_nm), "%s_%d_tq",
+	    ddi_driver_name(acp_dip), acp_inst);
+
+	acp->acp_tqp = ddi_taskq_create(dip, tq_nm, 1, TASKQ_DEFAULTPRI, 0);
+	if (acp->acp_tqp == NULL)
+		goto OUT;
+
+	if (ddi_taskq_dispatch(acp->acp_tqp, usb_ac_reader, (void *)acp,
+	    DDI_SLEEP) != DDI_SUCCESS)
+		goto OUT;
+
+	dinfo("%s: dispatched reader for %s%d\n", uacp->dstr,
+	    acp_name, acp_inst);
+
+	rv = USB_SUCCESS;
+
+OUT:
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	dinfo("%s: %s on %s%d done, rv=%d\n", uacp->dstr, nm,
+	    acp_name, acp_inst, rv);
+
+	return (rv);
+}
+
+
+static void
+usb_ac_mux_plumbing_tq(void *arg)
+{
+	usb_ac_state_t *uacp = (usb_ac_state_t *)arg;
+
+	if (usb_ac_mux_plumbing(uacp) != USB_SUCCESS)
+		dwarn("%s: usb_ac_mux_plumbing failed\n", uacp->dstr);
+}
+
+
+static int
+usb_ac_do_plumbing(usb_ac_state_t *uacp)
+{
+	dev_info_t *dip = uacp->usb_ac_dip;
+	int inst = ddi_get_instance(dip);
+	char tq_nm[128];
+	int rv = USB_FAILURE;
+
+	(void) snprintf(tq_nm, sizeof (tq_nm), "%s_%d_tq",
+	    ddi_driver_name(dip), inst);
+
+	uacp->tqp = ddi_taskq_create(dip, tq_nm, 1, TASKQ_DEFAULTPRI, 0);
+	if (uacp->tqp == NULL) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "usb_ac_send_as_cmd: unknown cmd=%d", cmd);
+		    "usb_ac_do_plumbing: ddi_taskq_create failed");
+		goto OUT;
+	}
 
-		return (error);
+	if (ddi_taskq_dispatch(uacp->tqp, usb_ac_mux_plumbing_tq, (void *)uacp,
+	    DDI_SLEEP) != DDI_SUCCESS) {
+		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "usb_ac_do_plumbing: ddi_taskq_dispatch failed");
+		goto OUT;
 	}
 
-	if (mp == NULL) {
+	rv = USB_SUCCESS;
+
+OUT:
+	return (rv);
+}
+
+
+
+static void
+usb_ac_mux_unplumbing_tq(void *arg)
+{
+	usb_ac_state_t *uacp = (usb_ac_state_t *)arg;
+
+	if (usb_ac_mux_unplumbing(uacp) != USB_SUCCESS)
+		dwarn("%s: usb_ac_mux_unplumbing failed\n", uacp->dstr);
+}
+
+
+static int
+usb_ac_do_unplumbing(usb_ac_state_t *uacp)
+{
+	int rv = USB_FAILURE;
+
+	if (uacp->tqp == NULL)
+		return (USB_SUCCESS);
+
+	if (ddi_taskq_dispatch(uacp->tqp, usb_ac_mux_unplumbing_tq,
+	    (void *)uacp, DDI_SLEEP) != DDI_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "usb_ac_send_as_cmd: can't get mblk to send cmd down");
+		    "usb_ac_do_unplumbing: ddi_taskq_dispatch failed");
+		goto OUT;
+	}
+
+	dinfo("%s: waiting for unplumb thread\n", uacp->dstr);
+
+	ddi_taskq_wait(uacp->tqp);
+	rv = USB_SUCCESS;
+
+	dinfo("%s: unplumb thread done\n", uacp->dstr);
 
-		return (error);
+OUT:
+	if (uacp->tqp != NULL) {
+		ddi_taskq_destroy(uacp->tqp);
+		uacp->tqp = NULL;
 	}
+	return (rv);
+}
 
-	/*
-	 * Set wait flag and send message down; we have made sure
-	 * before that canputnext succeeds. Note mp will be freed down
-	 */
-	streams_infop->acs_ac_to_as_req.acr_wait_flag = 1;
+
+/*
+ * teardown threads to the other usb modules
+ * and clear structures as part of unplumbing
+ */
+static void
+usb_ac_unplumb(usb_ac_plumbed_t *acp)
+{
+	usb_ac_streams_info_t *streams_infop;
+	usb_ac_state_t	*uacp = acp->acp_uacp;
+	dev_info_t	*acp_dip;
+	int		acp_inst;
+	char		*nm = "usb_ac_unplumb";
+	char		*acp_name;
+
+	acp_dip = acp->acp_dip;
+	acp_inst = ddi_get_instance(acp_dip);
+	acp_name = (char *)ddi_driver_name(acp_dip);
+
+	dinfo("%s: %s on %s%d\n", uacp->dstr, nm, acp_name, acp_inst);
+
+	if (acp->acp_tqp != NULL) {
+		dinfo("%s: %s - destroying taskq for %s%d\n", uacp->dstr, nm,
+		    acp_name, acp_inst);
+
+		ddi_taskq_destroy(acp->acp_tqp);
+	}
+
+	mutex_enter(&uacp->usb_ac_mutex);
+
+	if (acp->acp_driver == USB_AS_PLUMBED) {
+		/*
+		 * we bzero the streams info and plumbed structure
+		 * since there is no guarantee that the next plumbing
+		 * will be identical
+		 */
+		streams_infop = (usb_ac_streams_info_t *)acp->acp_data;
+
+		/* bzero the relevant plumbing structure */
+		bzero(streams_infop, sizeof (usb_ac_streams_info_t));
+	}
+	bzero(acp, sizeof (usb_ac_plumbed_t));
 
 	mutex_exit(&uacp->usb_ac_mutex);
-	putnext(lwq, mp);
+
+	dinfo("%s: %s on %s done\n", uacp->dstr, nm, acp_name, acp_inst);
+}
+
+
+/*ARGSUSED*/
+static int
+usb_ac_mux_plumbing(usb_ac_state_t *uacp)
+{
+	dev_info_t		*dip;
+	char			*nm = "usb_ac_mux_plumbing";
+
+	/* get the usb_ac dip */
+	dip = uacp->usb_ac_dip;
+
+	/* Access to the global variables is synchronized */
+	mutex_enter(&uacp->usb_ac_mutex);
+
+	dinfo("%s: %s state=%d\n", uacp->dstr, nm,
+	    uacp->usb_ac_plumbing_state);
+
+	if (uacp->usb_ac_plumbing_state >= USB_AC_STATE_PLUMBED) {
+		mutex_exit(&uacp->usb_ac_mutex);
+		dinfo("%s: audio streams driver already plumbed\n",
+		    uacp->dstr);
+
+		return (USB_SUCCESS);
+	}
+
+	/* usb_as and hid should be attached but double check */
+	if (usb_ac_online_siblings(uacp) != USB_SUCCESS) {
+		mutex_exit(&uacp->usb_ac_mutex);
+		dinfo("%s: no audio streams driver plumbed\n", uacp->dstr);
+
+		return (USB_FAILURE);
+	}
+
+	dinfo("%s: (%s) raising power\n", uacp->dstr, nm);
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	/* bring the device to full power */
+	usb_ac_pm_busy_component(uacp);
+	(void) pm_raise_power(dip, 0, USB_DEV_OS_FULL_PWR);
+
+	/* avoid dips disappearing while we are plumbing */
+	usb_ac_hold_siblings(uacp);
+
 	mutex_enter(&uacp->usb_ac_mutex);
 
 	/*
-	 * Wait for the response; reply will arrive through rput()
-	 * M_CTL and the cv_wait will be signaled there and wait flag
-	 * will be reset
+	 * walk all siblings and create the usb_ac<->usb_as and
+	 * usb_ac<->hid streams. return of 0 indicates no or
+	 * partial/failed plumbing
 	 */
-	while (streams_infop->acs_ac_to_as_req.acr_wait_flag) {
-#ifndef DEBUG
-		cv_wait(&streams_infop->acs_ac_to_as_req.acr_cv,
-		    &uacp->usb_ac_mutex);
-#else
-		clock_t tm = ddi_get_lbolt() +
-		    drv_usectohz(usb_ac_wait_timeout);
-		int rval;
-
-		rval = cv_timedwait(&streams_infop->acs_ac_to_as_req.acr_cv,
-		    &uacp->usb_ac_mutex, tm);
-
-		if (streams_infop->acs_ac_to_as_req.acr_wait_flag) {
-			if (rval == -1) {
-				USB_DPRINTF_L3(PRINT_MASK_ALL,
-				    uacp->usb_ac_log_handle,
-				    "usb_ac_send_as_cmd:"
-				    " timeout happen before cmd complete.");
-			} else {
-				USB_DPRINTF_L3(PRINT_MASK_ALL,
-				    uacp->usb_ac_log_handle,
-				    "usb_ac_send_as_cmd:"
-				    " not signaled by USB_AS_PLUMBED.");
+	if (usb_ac_mux_walk_siblings(uacp) == 0) {
+		/* pretend that we are plumbed so we can unplumb */
+		uacp->usb_ac_plumbing_state = USB_AC_STATE_PLUMBED;
+
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		(void) usb_ac_mux_unplumbing(uacp);
+
+		dinfo("%s: no audio streams driver plumbed\n", uacp->dstr);
+
+		usb_ac_rele_siblings(uacp);
+
+		usb_ac_pm_idle_component(uacp);
+
+		return (USB_FAILURE);
+	}
+	uacp->usb_ac_plumbing_state = USB_AC_STATE_PLUMBED;
+
+	/* restore state if we have already registered with the mixer */
+	if (uacp->usb_ac_registered_with_mixer) {
+		dinfo("%s: already registered with mixer, restoring state\n",
+		    uacp->dstr);
+
+		(void) usb_ac_restore_audio_state(uacp, USB_FLAGS_SLEEP);
+
+	} else if (usb_ac_mixer_registration(uacp) != USB_SUCCESS) {
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		dwarn("%s: mixer registration failed\n", uacp->dstr);
+
+		(void) usb_ac_mux_unplumbing(uacp);
+
+		usb_ac_rele_siblings(uacp);
+
+		usb_ac_pm_idle_component(uacp);
+
+		return (USB_FAILURE);
+	}
+
+	mutex_exit(&uacp->usb_ac_mutex);
+	usb_ac_rele_siblings(uacp);
+
+	usb_ac_pm_idle_component(uacp);
+
+	dinfo("%s: %s done\n", uacp->dstr, nm);
+
+	return (USB_SUCCESS);
+}
+
+
+static int
+usb_ac_mux_unplumbing(usb_ac_state_t *uacp)
+{
+	usb_ac_plumbed_t	*acp;
+	ldi_handle_t		lh;
+	dev_info_t		*acp_dip;
+	int			inst;
+	int			i;
+	dev_t			devt;
+	minor_t			minor;
+	int			maxlinked = 0;
+	char			*nm = "usb_ac_mux_unplumbing";
+
+	mutex_enter(&uacp->usb_ac_mutex);
+
+	dinfo("%s: %s state=%d\n", uacp->dstr, nm, uacp->usb_ac_plumbing_state);
+
+	if (uacp->usb_ac_plumbing_state == USB_AC_STATE_UNPLUMBED) {
+		dinfo("%s: %s - already unplumbed!\n", uacp->dstr, nm);
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		return (USB_SUCCESS);
+	}
+
+	/* usb_ac might not have anything plumbed yet */
+	if (uacp->usb_ac_current_plumbed_index == -1) {
+		dinfo("%s: %s - nothing plumbed!\n", uacp->dstr, nm);
+		uacp->usb_ac_plumbing_state = USB_AC_STATE_UNPLUMBED;
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		return (USB_SUCCESS);
+	}
+
+	/* do not allow detach if still busy */
+	if (uacp->usb_ac_busy_count) {
+		dinfo("%s: %s - mux still busy (%d)\n", uacp->dstr, nm,
+		    uacp->usb_ac_busy_count);
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		return (USB_FAILURE);
+	}
+
+	uacp->usb_ac_plumbing_state = USB_AC_STATE_UNPLUMBED;
+
+	/* close ac-as and ac-hid streams */
+	maxlinked = uacp->usb_ac_current_plumbed_index + 1;
+	dinfo("%s: %s - maxlinked %d\n", uacp->dstr, nm, maxlinked);
+
+	for (i = 0; i < maxlinked; i++) {
+		/*
+		 * we must save members of usb_ac_plumbed[] before calling
+		 * usb_ac_unplumb() because it clears the structure
+		 */
+		acp = &uacp->usb_ac_plumbed[i];
+		lh = acp->acp_lh;
+		acp_dip = acp->acp_dip;
+		devt = acp->acp_devt;
+
+		if (acp_dip == NULL) {
+			dinfo("%s: %s [%d] - skipping\n", uacp->dstr, nm, i);
+			continue;
+		}
+
+		minor = getminor(devt);
+		inst = ddi_get_instance(acp_dip);
+
+		uacp->usb_ac_current_plumbed_index = i;
+
+		dinfo("%s: %s [%d] - %s%d minor 0x%x\n", uacp->dstr, nm, i,
+		    ddi_driver_name(acp_dip), inst, minor);
+
+		if (lh != NULL) {
+			mutex_exit(&uacp->usb_ac_mutex);
+
+			acp->acp_flags &= ~ACP_ENABLED;
+
+			dinfo("%s: %s [%d] - closing\n", uacp->dstr, nm, i);
+
+			/* must close stream first before destroying thread */
+			(void) ldi_close(lh, FREAD|FWRITE, kcred);
+
+			usb_ac_unplumb(acp);
+
+			dinfo("%s: %s [%d] - unplumbed\n", uacp->dstr, nm, i);
+
+			mutex_enter(&uacp->usb_ac_mutex);
+		}
+	}
+
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	/* Wait till all activity in the default pipe has drained */
+	usb_ac_serialize_access(uacp);
+	usb_ac_release_access(uacp);
+
+	mutex_enter(&uacp->usb_ac_mutex);
+	uacp->usb_ac_current_plumbed_index = -1;
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	dinfo("%s: %s done\n", uacp->dstr, nm);
+
+	return (USB_SUCCESS);
+}
+
+
+/*
+ * walk all siblings and create the ac<->as and ac<->hid streams
+ */
+static int
+usb_ac_mux_walk_siblings(usb_ac_state_t *uacp)
+{
+	dev_info_t	*pdip;
+	dev_info_t	*child_dip;
+	major_t		drv_major;
+	minor_t		drv_minor;
+	int		drv_instance;
+	char		*drv_name;
+	dev_t		drv_devt;
+	ldi_handle_t	drv_lh;
+	ldi_ident_t	li;
+	int		error;
+	int		count = 0;
+	char		*nm = "usb_ac_mux_walk_siblings";
+
+	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
+
+	pdip = ddi_get_parent(uacp->usb_ac_dip);
+	child_dip = ddi_get_child(pdip);
+
+	dinfo("%s: %s parent=%s%d\n", uacp->dstr, nm,
+	    ddi_driver_name(pdip), ddi_get_instance(pdip));
+
+	while ((child_dip != NULL) && (count < USB_AC_MAX_PLUMBED)) {
+		drv_instance = ddi_get_instance(child_dip);
+		drv_name = (char *)ddi_driver_name(child_dip);
+
+		dinfo("%s: plumbing %s%d count=%d\n", uacp->dstr,
+		    drv_name, drv_instance, count);
+
+		/* ignore own dip */
+		if (child_dip == uacp->usb_ac_dip) {
+			child_dip = ddi_get_next_sibling(child_dip);
+			continue;
+		}
+		drv_instance = ddi_get_instance(child_dip);
+
+		/* ignore other dip other than usb_as and hid */
+		if (strcmp(ddi_driver_name(child_dip), "usb_as") == 0) {
+			uacp->usb_ac_plumbed[count].acp_driver = USB_AS_PLUMBED;
+			drv_minor = USB_AS_CONSTRUCT_MINOR(drv_instance);
+		} else if (strcmp(ddi_driver_name(child_dip), "hid") == 0) {
+			uacp->usb_ac_plumbed[count].acp_driver = USB_AH_PLUMBED;
+			drv_minor = HID_CONSTRUCT_EXTERNAL_MINOR(drv_instance);
+		} else {
+			drv_minor = drv_instance;
+			uacp->usb_ac_plumbed[count].acp_driver =
+			    UNKNOWN_PLUMBED;
+			child_dip = ddi_get_next_sibling(child_dip);
+
+			continue;
+		}
+
+		if (!i_ddi_devi_attached(child_dip)) {
+			child_dip = ddi_get_next_sibling(child_dip);
+
+			continue;
+		}
+
+		if (DEVI_IS_DEVICE_REMOVED(child_dip)) {
+			child_dip = ddi_get_next_sibling(child_dip);
+
+			continue;
+		}
+
+		drv_major = ddi_driver_major(child_dip);
+
+		uacp->usb_ac_current_plumbed_index = count;
+
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		drv_devt = makedevice(drv_major, drv_minor);
+
+		dinfo("%s: opening %s%d devt=(%d, 0x%x)\n", uacp->dstr,
+		    drv_name, drv_instance, drv_major, drv_minor);
+
+		error = ldi_ident_from_dip(uacp->usb_ac_dip, &li);
+		if (error == 0) {
+			uacp->usb_ac_plumbed[count].acp_flags |= ACP_ENABLED;
+
+			error = ldi_open_by_dev(&drv_devt, OTYP_CHR,
+			    FREAD|FWRITE, kcred, &drv_lh, li);
+			ldi_ident_release(li);
+		}
+
+		mutex_enter(&uacp->usb_ac_mutex);
+		if (error) {
+			dinfo("%s: open of devt=(%d, 0x%x) failed error=%d\n",
+			    uacp->dstr, drv_major, drv_minor, error);
+
+			return (0);
+		}
+
+		uacp->usb_ac_plumbed[count].acp_uacp = uacp;
+		uacp->usb_ac_plumbed[count].acp_devt = drv_devt;
+		uacp->usb_ac_plumbed[count].acp_lh = drv_lh;
+		uacp->usb_ac_plumbed[count].acp_dip = child_dip;
+		uacp->usb_ac_plumbed[count].acp_ifno =
+		    usb_get_if_number(child_dip);
+
+		if (uacp->usb_ac_plumbed[count].acp_driver == USB_AS_PLUMBED) {
+			/* get registration data */
+			if (usb_ac_get_reg_data(uacp, drv_lh, count) !=
+			    USB_SUCCESS) {
+
+				dinfo("%s: usb_ac_get_reg_data failed on "
+				    "%s%d\n", uacp->dstr,
+				    drv_name, drv_instance);
+
+				uacp->usb_ac_plumbed[count].acp_dip = NULL;
+
+				return (0);
+			}
+		} else if (uacp->usb_ac_plumbed[count].acp_driver ==
+		    USB_AH_PLUMBED) {
+			int rval;
+
+			dinfo("%s: pushing usb_ah on %s%d\n", uacp->dstr,
+			    drv_name, drv_instance);
+
+			mutex_exit(&uacp->usb_ac_mutex);
+
+			/* push usb_ah module on top of hid */
+			error = ldi_ioctl(drv_lh, I_PUSH, (intptr_t)"usb_ah",
+			    FKIOCTL, kcred, &rval);
+			mutex_enter(&uacp->usb_ac_mutex);
+
+			if (error) {
+				dinfo("%s: ldi_ioctl I_PUSH failed on "
+				    "%s%d, error=%d\n", uacp->dstr,
+				    drv_name, drv_instance, error);
+
+				uacp->usb_ac_plumbed[count].acp_dip = NULL;
+
+				/* skip plumbing the hid driver */
+				child_dip = ddi_get_next_sibling(child_dip);
+				continue;
 			}
+		} else {
+			/* should not be here */
+			dinfo("%s: %s - unknown module %s%d\n",
+			    uacp->dstr, nm, drv_name, drv_instance);
+			count--;
+
+			uacp->usb_ac_plumbed[count].acp_dip = NULL;
+
+			/* skip plumbing an unknown module */
+			child_dip = ddi_get_next_sibling(child_dip);
+			continue;
 		}
-#endif
+
+		mutex_exit(&uacp->usb_ac_mutex);
+		error = usb_ac_plumb(&uacp->usb_ac_plumbed[count]);
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		if (error != USB_SUCCESS) {
+			dinfo("%s: %s - usb_ac_plumb failed for %s%d\n",
+			    uacp->dstr, nm, drv_name, drv_instance);
+
+			return (0);
+		}
+
+		dinfo("%s: plumbed %s%d, minor 0x%x\n", uacp->dstr,
+		    drv_name, drv_instance, drv_minor);
+
+		child_dip = ddi_get_next_sibling(child_dip);
+		count++;
 	}
 
-	/* Wait is over, get the reply data */
-	mp = streams_infop->acs_ac_to_as_req.acr_reply_mp;
-	ASSERT(mp != NULL);
+	dinfo("%s: %s - %d drivers plumbed under usb_ac mux", uacp->dstr, nm,
+	    count);
 
-	iocp = (struct iocblk *)mp->b_rptr;
+	return (count);
+}
 
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_send_as_cmd: db_type=0x%x cmd=0x%x",
-	    mp->b_datap->db_type, iocp->ioc_cmd);
 
-	switch (mp->b_datap->db_type) {
-	case M_CTL:
-		switch (iocp->ioc_cmd) {
-		case USB_AUDIO_SET_FORMAT:
-			/*
-			 * This command sets mixer format data
-			 * and returns alternate setting that matches
-			 */
-			ASSERT(mp->b_cont != NULL);
-			ASSERT(MBLKL(mp->b_cont) == sizeof (int));
-			USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "alternate returned %d",
-			    *((int *)(mp->b_cont->b_rptr)));
-
-			streams_infop->acs_ac_to_as_req.acr_curr_format.
-			    fmt_alt = *((int *)(mp->b_cont->b_rptr));
-
-			/*FALLTHROUGH*/
-		case USB_AUDIO_SET_SAMPLE_FREQ:
-		case USB_AUDIO_SETUP:
-		case USB_AUDIO_START_PLAY:
-		case USB_AUDIO_PAUSE_PLAY:
-		case USB_AUDIO_START_RECORD:
-		case USB_AUDIO_STOP_RECORD:
-		case USB_AUDIO_TEARDOWN:
-			error = USB_SUCCESS;
-			break;
-		default:
+/*
+ * usb_ac_find_default_port:
+ */
+static int
+usb_ac_find_default_port(uint_t port)
+{
+	int i;
+
+	for (i = 0; i < 32; i++) {
+		if (port & (1 << i)) {
+
+			return (1 << i);
+		}
+	}
+
+	return (0);
+}
+
+
+/*
+ * Register with mixer only after first plumbing.
+ * Also do not register if earlier reg data
+ * couldn't be received from at least one
+ * streaming interface
+ */
+_NOTE(SCHEME_PROTECTS_DATA("private", am_ad_info))
+
+static int
+usb_ac_mixer_registration(usb_ac_state_t *uacp)
+{
+	am_ad_info_t	*info	= &uacp->usb_ac_am_ad_info;
+	audio_info_t	*dflts	= &uacp->usb_ac_am_ad_defaults;
+	usb_as_registration_t *asreg;
+	int		n, nplay, nrec;
+	char		*nm = "usb_ac_mixer_registration";
+
+	ASSERT(uacp != NULL);
+
+	dinfo("%s: %s infp=0x%p, dflts=0x%p, already registered=%d\n",
+	    uacp->dstr, nm, (void *)info, (void *)dflts,
+	    uacp->usb_ac_registered_with_mixer);
+
+	ASSERT(dflts != NULL);
+	ASSERT(info != NULL);
+
+	if (uacp->usb_ac_registered_with_mixer) {
+		return (USB_SUCCESS);
+	}
+
+	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n++) {
+		if (uacp->usb_ac_streams[n].acs_rcvd_reg_data) {
 			break;
 		}
-		break;
-	case M_ERROR:
-	default:
-		error = USB_FAILURE;
 	}
 
-	if (mp) {
-		usb_ac_free_mblk(mp);
-		streams_infop->acs_ac_to_as_req.acr_reply_mp = NULL;
+	/* Haven't found a streaming interface; fail mixer registration */
+	if (n > USB_AC_MAX_AS_PLUMBED) {
+		dinfo("%s: %s - no streaming interface found\n",
+		    uacp->dstr, nm);
+
+		return (USB_FAILURE);
 	}
 
-	return (error);
+	info->ad_defaults	= dflts;
+
+	dflts->monitor_gain	= 0;
+	dflts->output_muted	= B_FALSE;
+	dflts->hw_features	= 0;
+	dflts->sw_features	= AUDIO_SWFEATURE_MIXER;
+
+	/*
+	 * Fill out streaming interface specific stuff
+	 * Note that we handle only one playing and one recording
+	 * streaming interface at the most
+	 */
+	nplay = nrec = 0;
+	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n++) {
+		int ch, chs, default_gain, id;
+
+		if (uacp->usb_ac_streams[n].acs_rcvd_reg_data == 0) {
+			continue;
+		}
+
+		asreg = uacp->usb_ac_streams[n].acs_streams_reg;
+		if (asreg->reg_valid == 0) {
+			continue;
+		}
+
+		mutex_exit(&uacp->usb_ac_mutex);
+
+		dinfo("%s: %s [%d] - setting format on %s%d\n",
+		    uacp->dstr, nm, n,
+		    uacp->usb_ac_streams[n].acs_plumbed->acp_dip);
+
+		/* set first format so get_featureID can be succeed */
+		(void) usb_ac_set_format(uacp->usb_ac_audiohdl,
+		    asreg->reg_mode,
+		    asreg->reg_srs[0],
+		    asreg->reg_formats[0].fmt_chns,
+		    asreg->reg_formats[0].fmt_precision,
+		    asreg->reg_formats[0].fmt_encoding);
+
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		chs = asreg->reg_formats[0].fmt_chns;
+
+		/* check if any channel supports vol. control for this fmt */
+		for (ch = 0; ch <= chs; ch++) {
+			if ((id = usb_ac_get_featureID(uacp,
+			    asreg->reg_mode, ch,
+			    USB_AUDIO_VOLUME_CONTROL)) != -1) {
+				ddtl("%s: %s [%d] - dir=%d featureID=%d\n",
+				    uacp->dstr, nm, n, asreg->reg_mode, id);
+
+				break;
+			}
+		}
+		default_gain = (id == USB_AC_ID_NONE) ?
+		    AUDIO_MAX_GAIN : (AUDIO_MAX_GAIN/2);
+
+		ddtl("%s: %s [%d] - mode=%d chs=%d default_gain=%d id=%d\n",
+		    uacp->dstr, nm, n, asreg->reg_mode, chs, default_gain, id);
+
+		if (asreg->reg_mode == AUDIO_PLAY) {
+			nplay++;
+			ASSERT(nplay == 1);
+
+			dflts->play.sample_rate =
+			    asreg->reg_srs[0];
+			dflts->play.channels	=
+			    asreg->reg_formats[0].fmt_chns;
+			dflts->play.precision	=
+			    asreg->reg_formats[0].fmt_precision;
+			dflts->play.encoding	=
+			    asreg->reg_formats[0].fmt_encoding;
+			dflts->play.gain	= default_gain;
+			dflts->play.port	= usb_ac_find_default_port(
+			    uacp->usb_ac_output_ports);
+			dflts->play.avail_ports = uacp->usb_ac_output_ports;
+			dflts->play.mod_ports	= 0;
+						/* no support for mixer unit */
+			dflts->play.buffer_size = 8*1024;
+			dflts->hw_features	|= AUDIO_HWFEATURE_PLAY;
+
+			info->ad_play.ad_mixer_srs.ad_srs = asreg->reg_srs;
+
+			info->ad_play.ad_chs		= asreg->reg_channels;
+			info->ad_play.ad_int_rate	= 1000; /* every 1 ms */
+			info->ad_play.ad_bsize		= 8 * 1024;
+			info->ad_play_comb	= asreg->reg_combinations;
+		} else {
+			nrec++;
+			ASSERT(nrec == 1);
+
+			dflts->record.sample_rate =
+			    asreg->reg_srs[0];
+			dflts->record.channels	=
+			    asreg->reg_formats[0].fmt_chns;
+			dflts->record.precision =
+			    asreg->reg_formats[0].fmt_precision;
+			dflts->record.encoding	=
+			    asreg->reg_formats[0].fmt_encoding;
+			dflts->record.gain	= default_gain;
+			dflts->record.port	= usb_ac_find_default_port(
+			    uacp->usb_ac_input_ports);
+			dflts->record.avail_ports = uacp->usb_ac_input_ports;
+			dflts->record.mod_ports = uacp->usb_ac_input_ports;
+			dflts->record.buffer_size = 8*1024;
+			dflts->hw_features	|= AUDIO_HWFEATURE_RECORD;
+
+			info->ad_record.ad_mixer_srs.ad_srs = asreg->reg_srs;
+
+			info->ad_record.ad_chs		= asreg->reg_channels;
+			info->ad_record.ad_int_rate	= 1000; /* every 1 ms */
+			info->ad_record.ad_bsize	= 8 * 1024;
+			info->ad_num_mics		= 1;
+			info->ad_rec_comb	= asreg->reg_combinations;
+		}
+	}
+
+	if (nplay && nrec) {
+		dflts->hw_features	|= AUDIO_HWFEATURE_DUPLEX;
+	}
+
+	/* the rest */
+	info->ad_entry		= &usb_ac_entry;
+	info->ad_dev_info	= &usb_dev_info;
+
+	dinfo("%s: %s - calling am_attach\n", uacp->dstr, nm);
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	if (am_attach(uacp->usb_ac_audiohdl, DDI_ATTACH, info) ==
+	    AUDIO_FAILURE) {
+		dinfo("%s: %s - am_attach failed\n", uacp->dstr, nm);
+
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		return (USB_FAILURE);
+	}
+
+	mutex_enter(&uacp->usb_ac_mutex);
+
+	uacp->usb_ac_registered_with_mixer = 1;
+
+	dinfo("%s: %s - am_attach succeeded\n", uacp->dstr, nm);
+
+	return (USB_SUCCESS);
 }
 
 
 /*
- * usb_ac_allocate_req_mblk:
- *	Allocate a message block with the specified M_CTL cmd,
- *	The 2nd mblk contains the data for the command with a length len
+ * get registration data from usb_as driver unless we already
+ * have 2 registrations
  */
-static mblk_t *
-usb_ac_allocate_req_mblk(usb_ac_state_t *uacp, int cmd, void *buf, uint_t len)
+static int
+usb_ac_get_reg_data(usb_ac_state_t *uacp, ldi_handle_t drv_lh, int index)
 {
-	mblk_t	*mp, *mp2;
-	struct iocblk *mctlmsg;
+	int n, error, rval;
+	usb_as_registration_t *streams_reg;
+	char *nm = "usb_ac_get_reg_data";
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_allocate_req_mblk: cmd=0x%x, buf=0x%p, len=%d",
-	    cmd, buf, len);
+	dinfo("%s: %s index=%d, mixer registered=%d\n", uacp->dstr, nm, index,
+	    uacp->usb_ac_registered_with_mixer);
 
-	mp = allocb_wait(sizeof (struct iocblk), BPRI_HI, STR_NOSIG, NULL);
-	mp->b_datap->db_type = M_CTL;
-	mctlmsg = (struct iocblk *)mp->b_datap->db_base;
-	mctlmsg->ioc_cmd = cmd;
-	mctlmsg->ioc_count = len;
+	/* if already registered, just setup data structures again */
+	if (uacp->usb_ac_registered_with_mixer) {
 
-	mp->b_wptr = mp->b_wptr + sizeof (struct iocblk);
+		return (usb_ac_setup_plumbed(uacp, index, -1, -1));
+	}
 
-	if ((len == 0) || (buf == NULL)) {
+	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n ++) {
+		/*
+		 * We haven't received registration data
+		 * from n-th streaming interface in the array
+		 */
+		if (!uacp->usb_ac_streams[n].acs_rcvd_reg_data) {
+			break;
+		}
+	}
+
+	if (n >= USB_AC_MAX_AS_PLUMBED) {
+		dwarn("%s: More than 2 streaming interfaces (play "
+		    "and/or record) currently not supported\n", uacp->dstr);
 
-		return (mp);
+		return (USB_FAILURE);
 	}
 
-	mp2 = allocb_wait(len, BPRI_HI, STR_NOSIG, NULL);
-	mp->b_cont = mp2;
-	bcopy(buf, mp->b_cont->b_datap->db_base, len);
-	mp->b_cont->b_wptr = mp->b_cont->b_wptr + len;
+	/* take the stream reg struct with the same index */
+	streams_reg = &uacp->usb_ac_streams_reg[n];
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_allocate_req_mblk: mp=0x%p", (void *)mp);
+	dinfo("%s: regdata from usb_as: streams_reg=0x%p, n=%d\n", uacp->dstr,
+	    (void *)streams_reg, n);
+
+	mutex_exit(&uacp->usb_ac_mutex);
+
+	if ((error = ldi_ioctl(drv_lh, USB_AUDIO_MIXER_REGISTRATION,
+	    (intptr_t)streams_reg, FKIOCTL, kcred, &rval)) != 0) {
+		dinfo("%s: %s - ldi_ioctl failed for mixer registration, "
+		    "error=%d", uacp->dstr, nm, error);
+
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		return (USB_FAILURE);
+	} else {
+		mutex_enter(&uacp->usb_ac_mutex);
+
+		rval = usb_ac_setup_plumbed(uacp, index, n, n);
+
+		dinfo("%s: %s - usb_ac_streams[%d]: "
+		    "received_reg_data=%d type=%s", uacp->dstr, nm, index,
+		    uacp->usb_ac_streams[n].acs_rcvd_reg_data,
+		    ((streams_reg->reg_mode == AUDIO_PLAY) ?
+		    "play" : "record"));
 
-	return (mp);
+		usb_ac_print_reg_data(uacp, streams_reg);
+
+		return (rval);
+	}
 }
 
 
 /*
- * usb_ac_free_mblk:
- *	Free the message block
+ * setup plumbed and stream info structure, either initially or
+ * after replumbing
+ * On replumbing, str_idx and reg_idx are -1
  */
-static void
-usb_ac_free_mblk(mblk_t *mp)
+static int
+usb_ac_setup_plumbed(usb_ac_state_t *uacp, int plb_idx, int str_idx,
+    int reg_idx)
 {
-	if (mp->b_cont) {
-		freemsg(mp->b_cont);
-		mp->b_cont = NULL;
+	int i;
+	char *nm = "usb_ac_setup_plumbed";
+
+	dinfo("%s: %s plb_idx=%d str_idx=%d\n", uacp->dstr, nm,
+	    plb_idx, str_idx);
+
+	if (str_idx == -1) {
+		/* find a free streams info structure */
+		for (i = 0; i < USB_AC_MAX_AS_PLUMBED; i++) {
+			if (uacp->usb_ac_streams[i].acs_plumbed == NULL) {
+				break;
+			}
+		}
+		ASSERT(i < USB_AC_MAX_AS_PLUMBED);
+		str_idx = i;
 	}
-	freemsg(mp);
+
+	uacp->usb_ac_plumbed[plb_idx].acp_data =
+	    &uacp->usb_ac_streams[str_idx];
+	uacp->usb_ac_streams[str_idx].acs_plumbed =
+	    &uacp->usb_ac_plumbed[plb_idx];
+	uacp->usb_ac_streams[str_idx].acs_rcvd_reg_data = 1;
+
+	if (reg_idx == -1) {
+		/*
+		 * find the corresponding registration structure, match
+		 * on interface number and not on dip since dip may have
+		 * changed
+		 */
+		for (i = 0; i < USB_AC_MAX_AS_PLUMBED; i++) {
+			if (uacp->usb_ac_streams_reg[i].reg_ifno ==
+			    uacp->usb_ac_plumbed[plb_idx].acp_ifno) {
+				break;
+			}
+		}
+		if (i == USB_AC_MAX_AS_PLUMBED) {
+			dinfo("%s: %s - no corresponding registration "
+			    "structure\n", uacp->dstr, nm);
+
+			return (USB_FAILURE);
+		}
+		reg_idx = i;
+	}
+	uacp-> usb_ac_streams[str_idx].acs_streams_reg =
+	    &uacp->usb_ac_streams_reg[reg_idx];
+
+	dinfo("%s: %s done - plb_idx=%d str_idx=%d reg_idx=%d", uacp->dstr, nm,
+	    plb_idx, str_idx, reg_idx);
+
+	return (USB_SUCCESS);
 }
 
 
 /*
- * usb_ac_serialize/release_access:
+ * function to dump registration data
  */
 static void
-usb_ac_serialize_access(usb_ac_state_t	*uacp)
+usb_ac_print_reg_data(usb_ac_state_t *uacp,
+    usb_as_registration_t *reg)
 {
-	(void) usb_serialize_access(uacp->usb_ac_ser_acc, USB_WAIT, 0);
+	int n;
+
+	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+	    "usb_ac_print_reg_data: Begin valid=%d, play=%d, "
+	    "n_formats=%d, compat srs ptr=0x%p",
+	    reg->reg_valid, reg->reg_mode, reg->reg_n_formats,
+	    (void *)&reg->reg_srs);
+
+	for (n = 0; n < reg->reg_n_formats; n++) {
+		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "format%d: alt=%d chns=%d prec=%d enc=%d", n,
+		    reg->reg_formats[n].fmt_alt,
+		    reg->reg_formats[n].fmt_chns,
+		    reg->reg_formats[n].fmt_precision,
+		    reg->reg_formats[n].fmt_encoding);
+	}
+
+	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+	    "combinations: %d %d %d %d %d %d %d %d",
+	    reg->reg_combinations[0].ad_prec, reg->reg_combinations[0].ad_enc,
+	    reg->reg_combinations[1].ad_prec, reg->reg_combinations[1].ad_enc,
+	    reg->reg_combinations[2].ad_prec, reg->reg_combinations[2].ad_enc,
+	    reg->reg_combinations[3].ad_prec, reg->reg_combinations[3].ad_enc);
+
+
+	for (n = 0; n < USB_AS_N_FORMATS; n++) {
+		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "reg_formats[%d] ptr=0x%p", n,
+		    (void *)&reg->reg_formats[n]);
+	}
+
+	for (n = 0; n < USB_AS_N_CHANNELS; n++) {
+		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "reg_channels[%d]=%d", n, reg->reg_channels[n]);
+	}
+
+	for (n = 0; n < USB_AS_N_COMBINATIONS; n++) {
+		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+		    "reg_combinations[%d] ptr=0x%p", n,
+		    (void *)&reg->reg_combinations[n]);
+	}
+
+	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
+	    "usb_ac_print_reg_data: End");
 }
 
-static void
-usb_ac_release_access(usb_ac_state_t *uacp)
+
+static int
+usb_ac_online_siblings(usb_ac_state_t *uacp)
 {
-	usb_release_access(uacp->usb_ac_ser_acc);
+	dev_info_t	*pdip, *child_dip;
+	int		rval = USB_SUCCESS;
+	char		*nm = "usb_ac_online_siblings";
+
+	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
+
+	dinfo("%s: %s\n", uacp->dstr, nm);
+
+	pdip = ddi_get_parent(uacp->usb_ac_dip);
+
+	child_dip = ddi_get_child(pdip);
+	while (child_dip != NULL) {
+
+		dinfo("%s: onlining %s%d ref=%d\n", uacp->dstr,
+		    ddi_driver_name(child_dip),
+		    ddi_get_instance(child_dip),
+		    DEVI(child_dip)->devi_ref);
+
+		/* Online the child_dip of usb_as and hid,  if not already */
+		if ((strcmp(ddi_driver_name(child_dip), "usb_as") == 0) ||
+		    (strcmp(ddi_driver_name(child_dip), "hid") == 0)) {
+
+			mutex_exit(&uacp->usb_ac_mutex);
+			if (ndi_devi_online(child_dip, NDI_ONLINE_ATTACH) !=
+			    NDI_SUCCESS) {
+				dinfo("%s: failed to online device %s%d\n",
+				    uacp->dstr,
+				    ddi_driver_name(child_dip),
+				    ddi_get_instance(child_dip));
+
+				/* only onlining usb_as is fatal */
+				if (strcmp(ddi_driver_name(child_dip),
+				    "usb_as") == 0) {
+					mutex_enter(&uacp->usb_ac_mutex);
+					rval = USB_FAILURE;
+					break;
+				}
+			}
+			mutex_enter(&uacp->usb_ac_mutex);
+		}
+		child_dip = ddi_get_next_sibling(child_dip);
+	}
+
+	return (rval);
 }
 
 
+/*
+ * hold all audio children before or after plumbing
+ * online usb_as and hid, if not already
+ */
 static void
-usb_ac_pm_busy_component(usb_ac_state_t *usb_ac_statep)
+usb_ac_hold_siblings(usb_ac_state_t *uacp)
 {
-	ASSERT(!mutex_owned(&usb_ac_statep->usb_ac_mutex));
+	int		circ;
+	dev_info_t	*pdip, *child_dip;
+	char		*nm = "usb_ac_hold_siblings";
 
-	if (usb_ac_statep->usb_ac_pm != NULL) {
-		mutex_enter(&usb_ac_statep->usb_ac_mutex);
-		usb_ac_statep->usb_ac_pm->acpm_pm_busy++;
-
-		USB_DPRINTF_L4(PRINT_MASK_PM,
-		    usb_ac_statep->usb_ac_log_handle,
-		    "usb_ac_pm_busy_component: %d",
-		    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+	dinfo("%s: %s\n", uacp->dstr, nm);
 
-		mutex_exit(&usb_ac_statep->usb_ac_mutex);
+	/* hold all siblings and ourselves */
+	pdip = ddi_get_parent(uacp->usb_ac_dip);
 
-		if (pm_busy_component(usb_ac_statep->usb_ac_dip, 0) !=
-		    DDI_SUCCESS) {
-			mutex_enter(&usb_ac_statep->usb_ac_mutex);
-			usb_ac_statep->usb_ac_pm->acpm_pm_busy--;
+	/* hold the children */
+	ndi_devi_enter(pdip, &circ);
+	child_dip = ddi_get_child(pdip);
+	while (child_dip != NULL) {
+		ndi_hold_devi(child_dip);
 
-			USB_DPRINTF_L2(PRINT_MASK_PM,
-			    usb_ac_statep->usb_ac_log_handle,
-			    "usb_ac_pm_busy_component failed: %d",
-			    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+		dinfo("%s: held %s%d ref=%d\n", uacp->dstr,
+		    ddi_driver_name(child_dip), ddi_get_instance(child_dip),
+		    DEVI(child_dip)->devi_ref);
 
-			mutex_exit(&usb_ac_statep->usb_ac_mutex);
-		}
+		child_dip = ddi_get_next_sibling(child_dip);
 	}
+	ndi_devi_exit(pdip, circ);
 }
 
 
+/*
+ * release all audio children before or after plumbing
+ */
 static void
-usb_ac_pm_idle_component(usb_ac_state_t *usb_ac_statep)
+usb_ac_rele_siblings(usb_ac_state_t *uacp)
 {
-	ASSERT(!mutex_owned(&usb_ac_statep->usb_ac_mutex));
+	int		circ;
+	dev_info_t	*pdip, *child_dip;
+	char		*nm = "usb_ac_rele_siblings";
 
-	if (usb_ac_statep->usb_ac_pm != NULL) {
-		if (pm_idle_component(usb_ac_statep->usb_ac_dip, 0) ==
-		    DDI_SUCCESS) {
-			mutex_enter(&usb_ac_statep->usb_ac_mutex);
-			ASSERT(usb_ac_statep->usb_ac_pm->acpm_pm_busy > 0);
-			usb_ac_statep->usb_ac_pm->acpm_pm_busy--;
+	dinfo("%s: %s\n", uacp->dstr, nm);
 
-			USB_DPRINTF_L4(PRINT_MASK_PM,
-			    usb_ac_statep->usb_ac_log_handle,
-			    "usb_ac_pm_idle_component: %d",
-			    usb_ac_statep->usb_ac_pm->acpm_pm_busy);
+	/* release all siblings and ourselves */
+	pdip = ddi_get_parent(uacp->usb_ac_dip);
+	ndi_devi_enter(pdip, &circ);
+	child_dip = ddi_get_child(pdip);
+	while (child_dip != NULL) {
+		ndi_rele_devi(child_dip);
 
-			mutex_exit(&usb_ac_statep->usb_ac_mutex);
-		}
+		dinfo("%s: released %s%d ref=%d\n", uacp->dstr,
+		    ddi_driver_name(child_dip), ddi_get_instance(child_dip),
+		    DEVI(child_dip)->devi_ref);
+
+		child_dip = ddi_get_next_sibling(child_dip);
 	}
+	ndi_devi_exit(pdip, circ);
 }
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.conf b/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.conf
index 6318d089bd..cf1da6fa0d 100644
--- a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.conf
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac.conf
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 #
@@ -29,15 +29,6 @@
 #       WARNING: This is an UNSTABLE configuration file. Its contents
 #               may change at any time.
 #
-# mixer-mode determines the mode the mixer audio personality module
-# sets for this driver when it is loaded. The mixerctl(1) command may
-# be used to change modes on the fly.
-#
-#       mixer-enabled = 0;   <-- mixer disabled, old audio(7I) behavior
-#       mixer-enabled = 1;   <-- mixer enabled
-#
-
-mixer-enabled = 1;
 
 #
 # Uncomment reset-configuration to cause the usb_ac driver's state to
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac_dacf.c b/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac_dacf.c
deleted file mode 100644
index d23b379c7c..0000000000
--- a/usr/src/uts/common/io/usb/clients/audio/usb_ac/usb_ac_dacf.c
+++ /dev/null
@@ -1,1202 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-
-/*
- * This is a dacf module for usb audio plumbing/unplumbing based
- * upon the Extensions to Device Autoconfiguration project.  See
- * PSARC/1998/212 for more details.
- *
- * This module provides the dacf functions to be called after a
- * driver has attached and before it detaches.
- * Particularly, the usb_ac_mux_plumbing() is called after
- * usb_ac_attach() is called and usb_ac_mux_unplumbing() is called
- * before usb_ac_detach() is going to be called. This facilitates
- * the configuration of usb_as and hid under usb_ac and also
- * provides * usb_ac and usb_as hotplugging.
- *
- * This dacf module only supports usb_as and hid plumbing under
- * usb_ac.
- *
- * Note: This module shares some common data structures with usb_ac
- * module. So it would require dependency on usb_ac for the module
- * loading. But the side effect for usb_ac_dacf to depend on usb_ac
- * is a deadlock situation created when doing "modunload -i
- * <usb_ac_dacf_mod_id>" before "modunload -i <usb_ac_mod_id".
- * To avoid the dependency, space_store() and space_fetch() are
- * used to share the usb_ac data structure with usb_ac_dacf.
- */
-#include <sys/usb/usba/usbai_version.h>
-#include <sys/usb/usba.h>
-#include <sys/sunndi.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-
-#include <sys/usb/clients/audio/usb_audio.h>
-#include <sys/usb/clients/audio/usb_mixer.h>
-#include <sys/usb/clients/audio/usb_ac/usb_ac.h>
-
-#include <sys/stropts.h>
-#include <sys/dacf.h>
-
-/* for getting the minor node info from hid */
-#include <sys/usb/clients/hid/hidminor.h>
-#include <sys/usb/clients/audio/usb_as/usb_as.h>
-
-/*
- * Dacf entry points
- */
-static int	usb_ac_mux_plumbing(dacf_infohdl_t, dacf_arghdl_t, int);
-static int	usb_ac_mux_unplumbing(dacf_infohdl_t, dacf_arghdl_t, int);
-
-/*
- * External functions
- */
-extern	uint_t		nproc;
-#define	INIT_PROCESS_CNT 3
-
-extern uintptr_t space_fetch(char *);
-
-/*
- * Internal functions
- */
-static int	usb_ac_mux_walk_siblings(usb_ac_state_t *, ldi_handle_t);
-static void	usb_ac_print_reg_data(usb_ac_state_t *,
-				usb_as_registration_t *);
-static int	usb_ac_get_reg_data(usb_ac_state_t *, ldi_handle_t, int);
-static int	usb_ac_setup_plumbed(usb_ac_state_t *, int, int, int);
-static int	usb_ac_mixer_registration(usb_ac_state_t *,
-				usb_ac_state_space_t *);
-static void	usb_ac_hold_siblings(usb_ac_state_t *);
-static int	usb_ac_online_siblings(usb_ac_state_t *);
-static void	usb_ac_rele_siblings(usb_ac_state_t *);
-
-static am_ad_entry_t *usb_ac_entry;
-_NOTE(SCHEME_PROTECTS_DATA("stable data", usb_ac_entry))
-
-
-/* just generic, USB Audio, 1.0 spec-compliant */
-static audio_device_t usb_dev_info =
-	{ {"USB Audio"}, {"1.0"}, {"external"} };
-
-static dacf_op_t usb_ac_plumb_op[] = {
-	{ DACF_OPID_POSTATTACH,	usb_ac_mux_plumbing },
-	{ DACF_OPID_PREDETACH,	usb_ac_mux_unplumbing },
-	{ DACF_OPID_END,	NULL },
-};
-
-static dacf_opset_t opsets[] = {
-	{ "usb_audio_config", usb_ac_plumb_op },
-	{ NULL,		NULL }
-};
-
-struct dacfsw usb_audio_dacfsw = {
-	DACF_MODREV_1,
-	opsets,
-};
-
-struct modldacf usb_audio_dacf = {
-	&mod_dacfops,	/* Type of module */
-	"USB_AC_DACF",
-	&usb_audio_dacfsw
-};
-
-struct modlinkage usb_audio_modlinkage = {
-	MODREV_1, (void *)&usb_audio_dacf, NULL
-};
-
-
-int
-_init(void)
-{
-	return (mod_install(&usb_audio_modlinkage));
-}
-
-
-int
-_fini()
-{
-	return (mod_remove(&usb_audio_modlinkage));
-}
-
-
-int
-_info(struct modinfo *modinfop)
-{
-	return (mod_info(&usb_audio_modlinkage, modinfop));
-}
-
-
-/*
- * This routine is called at post attach time for the usb_ac module
- * by the DACF framework.
- */
-/*ARGSUSED*/
-static int
-usb_ac_mux_plumbing(dacf_infohdl_t info_hdl, dacf_arghdl_t arg_hdl, int flags)
-{
-	major_t			major;
-	minor_t			minor;
-	int			instance;
-	usb_ac_state_t		*uacp;
-	usb_ac_state_space_t	*ssp;
-	dev_info_t		*dip;
-	int			error;
-	ldi_handle_t		mux_lh;
-	dev_t			mux_devt;
-	ldi_ident_t		li;
-
-	/* get the usb_ac dip */
-	dip = dacf_devinfo_node(info_hdl);
-	instance = ddi_get_instance(dip);
-
-	/* Retrieve the soft state information */
-	if ((ssp = (usb_ac_state_space_t *)space_fetch("usb_ac")) ==
-	    NULL) {
-
-		return (DACF_FAILURE);
-	}
-
-	if ((uacp = (usb_ac_state_t *)ddi_get_soft_state(ssp->sp,
-	    instance)) == NULL) {
-
-		return (DACF_FAILURE);
-	}
-	ASSERT(dip == uacp->usb_ac_dip);
-
-	usb_ac_entry = ssp->ac_entryp;
-
-	/* Access to the global variables is synchronized */
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_mux_plumbing callback, state=%d",
-	    uacp->usb_ac_plumbing_state);
-
-	/*
-	 * by design, dacf will try to plumb again if an unplumb failed.
-	 * therefore, just return success
-	 */
-	if (uacp->usb_ac_plumbing_state >= USB_AC_STATE_PLUMBED) {
-		mutex_exit(&uacp->usb_ac_mutex);
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "audio streams driver already plumbed");
-
-		return (DACF_SUCCESS);
-	}
-
-	/* usb_as and hid should be attached but double check */
-	if (usb_ac_online_siblings(uacp) != USB_SUCCESS) {
-		mutex_exit(&uacp->usb_ac_mutex);
-		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "no audio streams driver plumbed");
-
-		return (DACF_FAILURE);
-	}
-
-	major = ddi_driver_major(dip);
-	ASSERT(major != (major_t)-1);
-
-	minor = dacf_minor_number(info_hdl);
-	ASSERT(minor == uacp->usb_ac_mux_minor);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "major 0x%x, minor %d  usb_ac_mux_plumbing: driver name "
-	    "%s\n", major, minor, (char *)dacf_driver_name(info_hdl));
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	/* bring the device to full power */
-	(ssp->pm_busy_component)(uacp);
-	(void) pm_raise_power(dip, 0, USB_DEV_OS_FULL_PWR);
-
-	/* avoid dips disappearing while we are plumbing */
-	usb_ac_hold_siblings(uacp);
-
-	/* opening usb_ac for plumbing underneath */
-	mux_devt = makedevice(major, minor);
-	error = ldi_ident_from_mod(&usb_audio_modlinkage, &li);
-	if (error == 0) {
-		error = ldi_open_by_dev(&mux_devt, OTYP_CHR,
-		    FREAD|FWRITE|FNOCTTY|FNONBLOCK, kcred, &mux_lh, li);
-		ldi_ident_release(li);
-	}
-	if (error) {
-		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "open failed, error=%d", error);
-		usb_ac_rele_siblings(uacp);
-
-		(ssp->pm_idle_component)(uacp);
-
-		return (DACF_FAILURE);
-	}
-
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	uacp->usb_ac_mux_lh = mux_lh;
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "mux_lh=0x%p", (void *)mux_lh);
-
-	/*
-	 * walk all siblings and create the usb_ac<->usb_as and
-	 * usb_ac<->hid streams. return of 0 indicates no or
-	 * partial/failed plumbing
-	 */
-	if (usb_ac_mux_walk_siblings(uacp, mux_lh) == 0) {
-		/* pretend that we are plumbed so we can unplumb */
-		uacp->usb_ac_plumbing_state = USB_AC_STATE_PLUMBED;
-
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		(void) usb_ac_mux_unplumbing(info_hdl, arg_hdl, flags);
-
-		USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "no audio streams driver plumbed");
-
-		usb_ac_rele_siblings(uacp);
-
-		(ssp->pm_idle_component)(uacp);
-
-		return (DACF_FAILURE);
-	}
-	uacp->usb_ac_plumbing_state = USB_AC_STATE_PLUMBED;
-
-	/* restore state if we have already registered with the mixer */
-	if (uacp->usb_ac_registered_with_mixer) {
-
-		(void) (ssp->restore_func)(uacp, USB_FLAGS_SLEEP);
-
-	} else if (usb_ac_mixer_registration(uacp, ssp) != USB_SUCCESS) {
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		(void) usb_ac_mux_unplumbing(info_hdl, arg_hdl, flags);
-
-		USB_DPRINTF_L1(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "mixer registration failed");
-
-		usb_ac_rele_siblings(uacp);
-
-		(ssp->pm_idle_component)(uacp);
-
-		return (DACF_FAILURE);
-	}
-
-	mutex_exit(&uacp->usb_ac_mutex);
-	usb_ac_rele_siblings(uacp);
-
-	(ssp->pm_idle_component)(uacp);
-
-	return (DACF_SUCCESS);
-}
-
-
-/*
- * This is the pre-detach routine invoked for usb_ac module by DACF framework
- */
-/*ARGSUSED*/
-static int
-usb_ac_mux_unplumbing(dacf_infohdl_t info_hdl, dacf_arghdl_t arg_hdl, int flags)
-{
-	usb_ac_state_t		*uacp;
-	usb_ac_state_space_t	*ssp;
-	dev_info_t		*dip;
-	ldi_handle_t		mux_lh;
-	dev_info_t		*child_dip;
-	int			i, error;
-	int			maxlinked = 0;
-
-	/* this is the usb_ac dip */
-	dip = dacf_devinfo_node(info_hdl);
-
-	/* Retrieve the soft state information */
-	if ((ssp = (usb_ac_state_space_t *)space_fetch("usb_ac")) ==
-	    NULL) {
-
-		return (DACF_FAILURE);
-	}
-
-	if ((uacp = (usb_ac_state_t *)ddi_get_soft_state(ssp->sp,
-	    ddi_get_instance(dip))) == NULL) {
-
-		return (DACF_FAILURE);
-	}
-
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_mux_unplumbing callback, state=%d",
-	    uacp->usb_ac_plumbing_state);
-
-	ASSERT(uacp->usb_ac_plumbing_state != USB_AC_STATE_UNPLUMBED);
-
-	if (uacp->usb_ac_plumbing_state == USB_AC_STATE_UNPLUMBED) {
-		USB_DPRINTF_L4(PRINT_MASK_ALL,
-		    uacp->usb_ac_log_handle,
-		    "already unplumbed!");
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		return (DACF_SUCCESS);
-	}
-
-	ASSERT(dip == uacp->usb_ac_dip);
-
-	mux_lh = uacp->usb_ac_mux_lh;
-
-	/* usb_ac might not have anything plumbed yet */
-	if (uacp->usb_ac_current_plumbed_index == -1) {
-		USB_DPRINTF_L4(PRINT_MASK_ALL,
-		    uacp->usb_ac_log_handle,
-		    "nothing plumbed!");
-		uacp->usb_ac_plumbing_state = USB_AC_STATE_UNPLUMBED;
-
-		goto close_mux;
-	}
-
-	/* do not allow detach if still busy */
-	if (uacp->usb_ac_busy_count) {
-		USB_DPRINTF_L2(PRINT_MASK_ALL,
-		    uacp->usb_ac_log_handle,
-		    "mux still open (%d %d)", uacp->usb_ac_busy_count,
-		    e_ddi_devi_holdcnt(dip));
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		return (DACF_FAILURE);
-	}
-
-	uacp->usb_ac_plumbing_state = USB_AC_STATE_UNPLUMBED;
-
-	/* wait till tasks have drained using save state */
-	mutex_exit(&uacp->usb_ac_mutex);
-	(void) audio_sup_save_state(uacp->usb_ac_audiohdl,
-	    AUDIO_ALL_DEVICES, AUDIO_BOTH);
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	mux_lh = uacp->usb_ac_mux_lh;
-	ASSERT(mux_lh != NULL);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_mux_unplumbing mux_lh 0x%p", (void *)mux_lh);
-
-	/* unlink and close ac-as and ac-hid streams */
-	maxlinked = uacp->usb_ac_current_plumbed_index + 1;
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_mux_unplumbing maxlinked %d", maxlinked);
-
-	for (i = 0; i < maxlinked; i++) {
-		int linkid = uacp->usb_ac_plumbed[i].acp_linkid;
-
-		child_dip = uacp->usb_ac_plumbed[i].acp_dip;
-		uacp->usb_ac_current_plumbed_index = i;
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "usb_ac_mux_unplumbing linkid %d ", linkid);
-
-		if (child_dip) {
-			int rval;
-
-			mutex_exit(&uacp->usb_ac_mutex);
-
-			error = ldi_ioctl(mux_lh, I_PUNLINK, (intptr_t)linkid,
-			    FKIOCTL, kcred, &rval);
-
-			USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "%s%d: unplink, error=%d",
-			    ddi_driver_name(child_dip),
-			    ddi_get_instance(child_dip), error);
-
-			mutex_enter(&uacp->usb_ac_mutex);
-		}
-
-	}
-
-close_mux:
-	if (mux_lh) {
-		mutex_exit(&uacp->usb_ac_mutex);
-		(void) ldi_close(mux_lh, FREAD|FWRITE, kcred);
-		mutex_enter(&uacp->usb_ac_mutex);
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "closed mux");
-		uacp->usb_ac_mux_lh = NULL;
-	}
-
-	uacp->usb_ac_current_plumbed_index = -1;
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_mux_unplumbing: done");
-
-	return (DACF_SUCCESS);
-}
-
-
-/*
- * walk all siblings and create the ac<->as and ac<->hid streams
- */
-static int
-usb_ac_mux_walk_siblings(usb_ac_state_t *uacp, ldi_handle_t mux_lh)
-{
-	dev_info_t	*pdip;
-	dev_info_t	*child_dip;
-	major_t		drv_major;
-	minor_t		drv_minor;
-	int		drv_instance;
-	dev_t		drv_devt;
-	ldi_handle_t	drv_lh;
-	ldi_ident_t	li;
-	int		linkid;
-	int		error;
-	int		count = 0;
-
-	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
-
-	mutex_exit(&uacp->usb_ac_mutex);
-	pdip = ddi_get_parent(uacp->usb_ac_dip);
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	child_dip = ddi_get_child(pdip);
-
-	while ((child_dip != NULL) && (count < USB_AC_MAX_PLUMBED)) {
-		drv_instance = ddi_get_instance(child_dip);
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "plumbing %s%d", ddi_driver_name(child_dip), drv_instance);
-
-		/* ignore own dip */
-		if (child_dip == uacp->usb_ac_dip) {
-			child_dip = ddi_get_next_sibling(child_dip);
-			continue;
-		}
-		drv_instance = ddi_get_instance(child_dip);
-
-		/* ignore other dip other than usb_as and hid */
-		if (strcmp(ddi_driver_name(child_dip), "usb_as") == 0) {
-			uacp->usb_ac_plumbed[count].acp_driver = USB_AS_PLUMBED;
-			drv_minor = USB_AS_CONSTRUCT_MINOR(drv_instance);
-		} else if (strcmp(ddi_driver_name(child_dip), "hid") == 0) {
-			uacp->usb_ac_plumbed[count].acp_driver = USB_AH_PLUMBED;
-			drv_minor = HID_CONSTRUCT_EXTERNAL_MINOR(drv_instance);
-		} else {
-			drv_minor = drv_instance;
-			uacp->usb_ac_plumbed[count].acp_driver =
-			    UNKNOWN_PLUMBED;
-			child_dip = ddi_get_next_sibling(child_dip);
-
-			continue;
-		}
-
-		if (!i_ddi_devi_attached(child_dip)) {
-			child_dip = ddi_get_next_sibling(child_dip);
-
-			continue;
-		}
-
-		if (DEVI_IS_DEVICE_REMOVED(child_dip)) {
-			child_dip = ddi_get_next_sibling(child_dip);
-
-			continue;
-		}
-
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "opening %s%d", ddi_driver_name(child_dip), drv_instance);
-
-		drv_major = ddi_driver_major(child_dip);
-
-		uacp->usb_ac_current_plumbed_index = count;
-
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		drv_devt = makedevice(drv_major, drv_minor);
-
-		error = ldi_ident_from_mod(&usb_audio_modlinkage, &li);
-		if (error == 0) {
-			error = ldi_open_by_dev(&drv_devt, OTYP_CHR,
-			    FREAD|FWRITE, kcred, &drv_lh, li);
-			ldi_ident_release(li);
-		}
-
-		mutex_enter(&uacp->usb_ac_mutex);
-		if (error) {
-			USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "ldi_open_by_dev failed on major = %d minor = %d, "
-			    "name = %s error=%d", drv_major, drv_instance,
-			    ddi_driver_name(child_dip), error);
-			mutex_exit(&uacp->usb_ac_mutex);
-			mutex_enter(&uacp->usb_ac_mutex);
-
-			return (0);
-		}
-
-		uacp->usb_ac_plumbed[count].acp_dip = child_dip;
-		uacp->usb_ac_plumbed[count].acp_ifno =
-		    usb_get_if_number(child_dip);
-
-		if (uacp->usb_ac_plumbed[count].acp_driver == USB_AS_PLUMBED) {
-			/* get registration data */
-			if (usb_ac_get_reg_data(uacp, drv_lh, count) !=
-			    USB_SUCCESS) {
-
-				USB_DPRINTF_L2(PRINT_MASK_ALL,
-				    uacp->usb_ac_log_handle,
-				    "get_reg_data failed on major = %d "
-				    "minor = %d, name = %s", drv_major,
-				    drv_instance,
-				    ddi_driver_name(child_dip));
-
-				mutex_exit(&uacp->usb_ac_mutex);
-				(void) ldi_close(drv_lh, FREAD|FWRITE, kcred);
-				mutex_enter(&uacp->usb_ac_mutex);
-				uacp->usb_ac_plumbed[count].acp_dip = NULL;
-
-				return (0);
-			}
-		} else if (uacp->usb_ac_plumbed[count].acp_driver ==
-		    USB_AH_PLUMBED) {
-			int rval;
-
-			mutex_exit(&uacp->usb_ac_mutex);
-
-			/* push usb_ah module on top of hid */
-			error = ldi_ioctl(drv_lh, I_PUSH, (intptr_t)"usb_ah",
-			    FKIOCTL, kcred, &rval);
-			mutex_enter(&uacp->usb_ac_mutex);
-
-			if (error) {
-				USB_DPRINTF_L2(PRINT_MASK_ALL,
-				    uacp->usb_ac_log_handle,
-				    "ldi_ioctl failed for usb_ah, "
-				    "major:%d minor:%d name:%s",
-				    drv_major, drv_instance,
-				    ddi_driver_name(child_dip));
-
-				mutex_exit(&uacp->usb_ac_mutex);
-				(void) ldi_close(drv_lh, FREAD|FWRITE, kcred);
-				mutex_enter(&uacp->usb_ac_mutex);
-				uacp->usb_ac_plumbed[count].acp_dip = NULL;
-
-				/* skip plumbing the hid driver */
-				child_dip = ddi_get_next_sibling(child_dip);
-				continue;
-			}
-		} else {
-			/* should not be here */
-			USB_DPRINTF_L2(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-			    "usb_ac_mux_plumbing: unknown module");
-			count --;
-
-			mutex_exit(&uacp->usb_ac_mutex);
-			(void) ldi_close(drv_lh, FREAD|FWRITE, kcred);
-			mutex_enter(&uacp->usb_ac_mutex);
-
-			uacp->usb_ac_plumbed[count].acp_dip = NULL;
-
-			/* skip plumbing an unknown module */
-			child_dip = ddi_get_next_sibling(child_dip);
-			continue;
-		}
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "linking %s%d", ddi_driver_name(child_dip), drv_instance);
-
-		/* usb_ac_plumb_ioctl sets lwq, this entry should be free */
-		ASSERT(uacp->usb_ac_plumbed[count].acp_lwq == NULL);
-
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		/* plumbing one at a time */
-		error = ldi_ioctl(mux_lh, I_PLINK, (intptr_t)drv_lh,
-		    FREAD|FWRITE|FNOCTTY|FKIOCTL, kcred, &linkid);
-
-		(void) ldi_close(drv_lh, FREAD|FWRITE, kcred);
-
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		if (error) {
-			USB_DPRINTF_L2(PRINT_MASK_ALL,
-			    uacp->usb_ac_log_handle,
-			    "plink failed for major:%d minor:%d"
-			    "name:%s", drv_major, drv_instance,
-			    ddi_driver_name(child_dip));
-
-			return (0);
-		}
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "link id %s%d 0x%x", ddi_driver_name(child_dip),
-		    ddi_get_instance(child_dip), linkid);
-
-		/* lwq should be set by usb_ac_plumb_ioctl by now */
-		ASSERT(uacp->usb_ac_plumbed[count].acp_lwq != NULL);
-
-		uacp->usb_ac_plumbed[count++].acp_linkid = linkid;
-
-		child_dip = ddi_get_next_sibling(child_dip);
-	}
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "%d drivers plumbed under usb_ac mux", count);
-
-	return (count);
-}
-
-
-/*
- * usb_ac_find_default_port:
- */
-static int
-usb_ac_find_default_port(uint_t port)
-{
-	int i;
-
-	for (i = 0; i < 32; i++) {
-		if (port & (1 << i)) {
-
-			return (1 << i);
-		}
-	}
-
-	return (0);
-}
-
-
-/*
- * Register with mixer only after first plumbing.
- * Also do not register if earlier reg data
- * couldn't be received from at least one
- * streaming interface
- */
-_NOTE(SCHEME_PROTECTS_DATA("private", am_ad_info))
-
-static int
-usb_ac_mixer_registration(usb_ac_state_t *uacp, usb_ac_state_space_t *ssp)
-{
-	am_ad_info_t	*info	= &uacp->usb_ac_am_ad_info;
-	audio_info_t	*dflts	= &uacp->usb_ac_am_ad_defaults;
-	usb_as_registration_t *asreg;
-	int		n, nplay, nrec;
-
-	ASSERT(uacp != NULL);
-
-	USB_DPRINTF_L4(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "usb_ac_mixer_registration: infp=0x%p, dflts=0x%p",
-	    (void *)info, (void *)dflts);
-
-	ASSERT(dflts != NULL);
-	ASSERT(info != NULL);
-
-	if (uacp->usb_ac_registered_with_mixer) {
-
-		return (USB_SUCCESS);
-	}
-
-	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n++) {
-		if (uacp->usb_ac_streams[n].acs_rcvd_reg_data) {
-			break;
-		}
-	}
-
-	/* Haven't found a streaming interface; fail mixer registration */
-	if (n > USB_AC_MAX_AS_PLUMBED) {
-		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-		    "no streaming interface");
-
-		return (USB_FAILURE);
-	}
-
-	USB_DPRINTF_L4(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "usb_ac_mixer_registration: mixer enabled =%d",
-	    uacp->usb_ac_mixer_mode_enable);
-
-	info->ad_int_vers	= AM_VERSION;
-	info->ad_mode		= (uacp->usb_ac_mixer_mode_enable ?
-	    AM_MIXER_MODE : AM_COMPAT_MODE);
-
-	info->ad_add_mode	= 0;
-	info->ad_codec_type	= AM_TRAD_CODEC;
-	info->ad_defaults	= dflts;
-
-	dflts->monitor_gain	= 0;
-	dflts->output_muted	= B_FALSE;
-	dflts->hw_features	= 0;
-	dflts->sw_features	= AUDIO_SWFEATURE_MIXER;
-
-	/*
-	 * Fill out streaming interface specific stuff
-	 * Note that we handle only one playing and one recording
-	 * streaming interface at the most
-	 */
-	nplay = nrec = 0;
-	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n++) {
-		int ch, chs, default_gain, id;
-
-		if (uacp->usb_ac_streams[n].acs_rcvd_reg_data == 0) {
-			continue;
-		}
-
-		asreg = uacp->usb_ac_streams[n].acs_streams_reg;
-		if (asreg->reg_valid == 0) {
-			continue;
-		}
-
-		mutex_exit(&uacp->usb_ac_mutex);
-
-		/* set first format so get_featureID can be succeed */
-		(void) ssp->ac_entryp->ad_set_format(uacp->usb_ac_audiohdl, 0,
-		    asreg->reg_mode,
-		    asreg->reg_compat_srs.ad_srs[0],
-		    asreg->reg_formats[0].fmt_chns,
-		    asreg->reg_formats[0].fmt_precision,
-		    asreg->reg_formats[0].fmt_encoding);
-
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		chs = asreg->reg_formats[0].fmt_chns;
-
-		/* check if any channel supports vol. control for this fmt */
-		for (ch = 0; ch <= chs; ch++) {
-			if ((id = ssp->get_featureID_func(uacp,
-			    asreg->reg_mode, ch,
-			    USB_AUDIO_VOLUME_CONTROL)) != -1) {
-				USB_DPRINTF_L3(PRINT_MASK_ATTA,
-				    uacp->usb_ac_log_handle,
-				    "dir=%d featureID=%d",
-				    asreg->reg_mode, id);
-
-				break;
-			}
-		}
-		default_gain = (id == USB_AC_ID_NONE) ?
-		    AUDIO_MAX_GAIN : (AUDIO_MAX_GAIN/2);
-
-		USB_DPRINTF_L3(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-		    "mode=%d chs=%d default_gain=%d id=%d",
-		    asreg->reg_mode, chs, default_gain, id);
-
-		if (asreg->reg_mode == AUDIO_PLAY) {
-			nplay++;
-			ASSERT(nplay == 1);
-
-			dflts->play.sample_rate =
-			    asreg->reg_compat_srs.ad_srs[0];
-			dflts->play.channels	=
-			    asreg->reg_formats[0].fmt_chns;
-			dflts->play.precision	=
-			    asreg->reg_formats[0].fmt_precision;
-			dflts->play.encoding	=
-			    asreg->reg_formats[0].fmt_encoding;
-			dflts->play.gain	= default_gain;
-			dflts->play.port	= usb_ac_find_default_port(
-			    uacp->usb_ac_output_ports);
-			dflts->play.avail_ports = uacp->usb_ac_output_ports;
-			dflts->play.mod_ports	= 0;
-						/* no support for mixer unit */
-			dflts->play.buffer_size = 8*1024;
-			dflts->play.balance	= AUDIO_MID_BALANCE;
-			dflts->hw_features	|= AUDIO_HWFEATURE_PLAY;
-
-			info->ad_play.ad_mixer_srs	= asreg->reg_mixer_srs;
-			info->ad_play.ad_compat_srs	= asreg->reg_compat_srs;
-			info->ad_play.ad_conv		= &am_src2;
-			info->ad_play.ad_sr_info	= NULL;
-			info->ad_play.ad_chs		= asreg->reg_channels;
-			info->ad_play.ad_int_rate	= 1000; /* every 1 ms */
-			info->ad_play.ad_max_chs	= 200;
-			info->ad_play.ad_bsize		= 8 * 1024;
-			info->ad_play_comb	= asreg->reg_combinations;
-		} else {
-			nrec++;
-			ASSERT(nrec == 1);
-
-			dflts->record.sample_rate =
-			    asreg->reg_compat_srs.ad_srs[0];
-			dflts->record.channels	=
-			    asreg->reg_formats[0].fmt_chns;
-			dflts->record.precision =
-			    asreg->reg_formats[0].fmt_precision;
-			dflts->record.encoding	=
-			    asreg->reg_formats[0].fmt_encoding;
-			dflts->record.gain	= default_gain;
-			dflts->record.port	= usb_ac_find_default_port(
-			    uacp->usb_ac_input_ports);
-			dflts->record.avail_ports = uacp->usb_ac_input_ports;
-			dflts->record.mod_ports = uacp->usb_ac_input_ports;
-			dflts->record.buffer_size = 8*1024;
-			dflts->record.balance	= AUDIO_MID_BALANCE;
-			dflts->hw_features	|= AUDIO_HWFEATURE_RECORD;
-
-			info->ad_record.ad_mixer_srs	= asreg->reg_mixer_srs;
-			info->ad_record.ad_compat_srs	= asreg->reg_compat_srs;
-			info->ad_record.ad_conv 	= &am_src2;
-			info->ad_record.ad_sr_info	= NULL;
-			info->ad_record.ad_chs		= asreg->reg_channels;
-			info->ad_record.ad_int_rate	= 1000; /* every 1 ms */
-			info->ad_record.ad_max_chs	= 200;
-			info->ad_record.ad_bsize	= 8 * 1024;
-			info->ad_num_mics		= 1;
-			info->ad_rec_comb	= asreg->reg_combinations;
-		}
-	}
-
-	if (nplay && nrec) {
-		/*
-		 * we pretend to always support AUDIO_HWFEATURE_IN2OUT
-		 * since there is no simple way to find out at this
-		 * point
-		 */
-		dflts->hw_features	|= AUDIO_HWFEATURE_DUPLEX |
-		    AUDIO_HWFEATURE_IN2OUT;
-	}
-
-	/* the rest */
-	info->ad_entry		= usb_ac_entry;
-	info->ad_dev_info	= &usb_dev_info;
-	info->ad_diag_flags	= 0;
-	info->ad_diff_flags = AM_DIFF_SR|AM_DIFF_CH|AM_DIFF_PREC|AM_DIFF_ENC;
-	info->ad_assist_flags	= 0;
-	info->ad_misc_flags	= AM_MISC_RP_EXCL;
-	info->ad_translate_flags = 0;
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	if (am_attach(uacp->usb_ac_audiohdl, DDI_ATTACH, info) ==
-	    AUDIO_FAILURE) {
-		USB_DPRINTF_L2(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-		    "am_attach: failed");
-
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		return (USB_FAILURE);
-	}
-
-	mutex_enter(&uacp->usb_ac_mutex);
-
-	uacp->usb_ac_registered_with_mixer = 1;
-
-	USB_DPRINTF_L4(PRINT_MASK_ATTA, uacp->usb_ac_log_handle,
-	    "am_attach succeeded");
-
-	return (USB_SUCCESS);
-}
-
-
-/*
- * get registration data from usb_as driver unless we already
- * have 2 registrations
- */
-static int
-usb_ac_get_reg_data(usb_ac_state_t *uacp, ldi_handle_t drv_lh, int index)
-{
-	int n, error, rval;
-	usb_as_registration_t *streams_reg;
-
-	/* if already registered, just setup data structures again */
-	if (uacp->usb_ac_registered_with_mixer) {
-
-		return (usb_ac_setup_plumbed(uacp, index, -1, -1));
-	}
-
-	for (n = 0; n < USB_AC_MAX_AS_PLUMBED; n ++) {
-		/*
-		 * We haven't received registration data
-		 * from n-th streaming interface in the array
-		 */
-		if (!uacp->usb_ac_streams[n].acs_rcvd_reg_data) {
-			break;
-		}
-	}
-
-	if (n >= USB_AC_MAX_AS_PLUMBED) {
-		USB_DPRINTF_L1(PRINT_MASK_ALL,
-		    uacp->usb_ac_log_handle,
-		    "More than 2 streaming interfaces (play "
-		    "and/or record) currently not supported");
-
-		return (USB_FAILURE);
-	}
-
-	/* take the stream reg struct with the same index */
-	streams_reg = &uacp->usb_ac_streams_reg[n];
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL,
-	    uacp->usb_ac_log_handle,
-	    "regdata from usb_as: streams_reg=0x%p, n=%d",
-	    (void *)streams_reg, n);
-
-	mutex_exit(&uacp->usb_ac_mutex);
-
-	if ((error = ldi_ioctl(drv_lh, USB_AUDIO_MIXER_REGISTRATION,
-	    (intptr_t)streams_reg, FKIOCTL, kcred, &rval)) != USB_SUCCESS) {
-		USB_DPRINTF_L2(PRINT_MASK_ALL,
-		    uacp->usb_ac_log_handle,
-		    "ldi_ioctl fails for mixer registration, error=%d", error);
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		return (USB_FAILURE);
-	} else {
-		mutex_enter(&uacp->usb_ac_mutex);
-
-		rval = usb_ac_setup_plumbed(uacp, index, n, n);
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "usb_ac_streams: index=%d, received_reg_data=%d type=%s",
-		    index, uacp->usb_ac_streams[n].acs_rcvd_reg_data,
-		    ((streams_reg->reg_mode == AUDIO_PLAY) ?
-		    "play" : "record"));
-
-		usb_ac_print_reg_data(uacp, streams_reg);
-
-		return (rval);
-	}
-}
-
-
-/*
- * setup plumbed and stream info structure, either initially or
- * after replumbing
- * On replumbing, str_idx and reg_idx are -1
- */
-static int
-usb_ac_setup_plumbed(usb_ac_state_t *uacp, int plb_idx, int str_idx,
-    int reg_idx)
-{
-	int i;
-
-	if (str_idx == -1) {
-		/* find a free streams info structure */
-		for (i = 0; i < USB_AC_MAX_AS_PLUMBED; i++) {
-			if (uacp->usb_ac_streams[i].acs_plumbed == NULL) {
-				break;
-			}
-		}
-		ASSERT(i < USB_AC_MAX_AS_PLUMBED);
-		str_idx = i;
-	}
-
-	uacp->usb_ac_plumbed[plb_idx].acp_data =
-	    &uacp->usb_ac_streams[str_idx];
-	uacp->usb_ac_streams[str_idx].acs_plumbed =
-	    &uacp->usb_ac_plumbed[plb_idx];
-	uacp->usb_ac_streams[str_idx].acs_rcvd_reg_data = 1;
-	cv_init(&(uacp->usb_ac_streams[str_idx].
-	    acs_ac_to_as_req.acr_cv), NULL, CV_DRIVER, NULL);
-
-	if (reg_idx == -1) {
-		/*
-		 * find the corresponding registration structure, match
-		 * on interface number and not on dip since dip may have
-		 * changed
-		 */
-		for (i = 0; i < USB_AC_MAX_AS_PLUMBED; i++) {
-			if (uacp->usb_ac_streams_reg[i].reg_ifno ==
-			    uacp->usb_ac_plumbed[plb_idx].acp_ifno) {
-				break;
-			}
-		}
-		if (i == USB_AC_MAX_AS_PLUMBED) {
-			USB_DPRINTF_L2(PRINT_MASK_ALL,
-			    uacp->usb_ac_log_handle,
-			    "no corresponding registration structure");
-
-			return (USB_FAILURE);
-		}
-		reg_idx = i;
-	}
-	uacp-> usb_ac_streams[str_idx].acs_streams_reg =
-	    &uacp->usb_ac_streams_reg[reg_idx];
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_setup_plumbed: plb_idx=%d str_idx=%d reg_idx=%d",
-	    plb_idx, str_idx, reg_idx);
-
-	return (USB_SUCCESS);
-}
-
-
-/*
- * function to dump registration data
- */
-static void
-usb_ac_print_reg_data(usb_ac_state_t *uacp,
-    usb_as_registration_t *reg)
-{
-	int n;
-
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_print_reg_data: Begin valid=%d, play=%d, "
-	    "n_formats=%d, mixer srs ptr=0x%p, compat srs ptr=0x%p",
-	    reg->reg_valid, reg->reg_mode, reg->reg_n_formats,
-	    (void *)&reg->reg_mixer_srs, (void *)&reg->reg_compat_srs);
-
-	for (n = 0; n < reg->reg_n_formats; n++) {
-		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "format%d: alt=%d chns=%d prec=%d enc=%d", n,
-		    reg->reg_formats[n].fmt_alt,
-		    reg->reg_formats[n].fmt_chns,
-		    reg->reg_formats[n].fmt_precision,
-		    reg->reg_formats[n].fmt_encoding);
-	}
-
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "combinations: %d %d %d %d %d %d %d %d",
-	    reg->reg_combinations[0].ad_prec, reg->reg_combinations[0].ad_enc,
-	    reg->reg_combinations[1].ad_prec, reg->reg_combinations[1].ad_enc,
-	    reg->reg_combinations[2].ad_prec, reg->reg_combinations[2].ad_enc,
-	    reg->reg_combinations[3].ad_prec, reg->reg_combinations[3].ad_enc);
-
-
-	for (n = 0; n < USB_AS_N_FORMATS; n++) {
-		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "reg_formats[%d] ptr=0x%p", n,
-		    (void *)&reg->reg_formats[n]);
-	}
-
-	for (n = 0; n < USB_AS_N_CHANNELS; n++) {
-		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "reg_channels[%d]=%d", n, reg->reg_channels[n]);
-	}
-
-	for (n = 0; n < USB_AS_N_COMBINATIONS; n++) {
-		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "reg_combinations[%d] ptr=0x%p", n,
-		    (void *)&reg->reg_combinations[n]);
-	}
-
-	USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-	    "usb_ac_print_reg_data: End");
-}
-
-
-static int
-usb_ac_online_siblings(usb_ac_state_t *uacp)
-{
-	dev_info_t	*pdip, *child_dip;
-	int		rval = USB_SUCCESS;
-
-	ASSERT(mutex_owned(&uacp->usb_ac_mutex));
-	USB_DPRINTF_L4(PRINT_MASK_PM, uacp->usb_ac_log_handle,
-	    "usb_ac_onlining siblings");
-
-	pdip = ddi_get_parent(uacp->usb_ac_dip);
-
-	child_dip = ddi_get_child(pdip);
-	while (child_dip != NULL) {
-		USB_DPRINTF_L3(PRINT_MASK_ALL, uacp->usb_ac_log_handle,
-		    "onlining %s%d ref=%d", ddi_driver_name(child_dip),
-		    ddi_get_instance(child_dip),
-		    DEVI(child_dip)->devi_ref);
-
-		/* Online the child_dip of usb_as and hid,  if not already */
-		if ((strcmp(ddi_driver_name(child_dip), "usb_as") == 0) ||
-		    (strcmp(ddi_driver_name(child_dip), "hid") == 0)) {
-			mutex_exit(&uacp->usb_ac_mutex);
-			if (ndi_devi_online(child_dip, NDI_ONLINE_ATTACH) !=
-			    NDI_SUCCESS) {
-				USB_DPRINTF_L2(PRINT_MASK_ALL,
-				    uacp->usb_ac_log_handle,
-				    "failure to online driver %s%d",
-				    ddi_driver_name(child_dip),
-				    ddi_get_instance(child_dip));
-
-				/* only onlining usb_as is fatal */
-				if (strcmp(ddi_driver_name(child_dip),
-				    "usb_as") == 0) {
-					mutex_enter(&uacp->usb_ac_mutex);
-					rval = USB_FAILURE;
-					break;
-				}
-			}
-			mutex_enter(&uacp->usb_ac_mutex);
-		}
-		child_dip = ddi_get_next_sibling(child_dip);
-	}
-
-	return (rval);
-}
-
-
-/*
- * hold all audio children before or after plumbing
- * online usb_as and hid, if not already
- */
-static void
-usb_ac_hold_siblings(usb_ac_state_t *uacp)
-{
-	int		circ;
-	dev_info_t	*pdip, *child_dip;
-
-	USB_DPRINTF_L4(PRINT_MASK_PM, uacp->usb_ac_log_handle,
-	    "usb_ac_hold_siblings:");
-
-	/* hold all siblings and ourselves */
-	pdip = ddi_get_parent(uacp->usb_ac_dip);
-
-	/* hold the children */
-	ndi_devi_enter(pdip, &circ);
-	child_dip = ddi_get_child(pdip);
-	while (child_dip != NULL) {
-
-		ndi_hold_devi(child_dip);
-
-		USB_DPRINTF_L3(PRINT_MASK_PM, uacp->usb_ac_log_handle,
-		    "usb_ac_hold_siblings: %s%d ref=%d",
-		    ddi_driver_name(child_dip), ddi_get_instance(child_dip),
-		    DEVI(child_dip)->devi_ref);
-
-		child_dip = ddi_get_next_sibling(child_dip);
-	}
-	ndi_devi_exit(pdip, circ);
-}
-
-
-/*
- * release all audio children before or after plumbing
- */
-static void
-usb_ac_rele_siblings(usb_ac_state_t *uacp)
-{
-	int		circ;
-	dev_info_t	*pdip, *child_dip;
-
-	USB_DPRINTF_L4(PRINT_MASK_PM, uacp->usb_ac_log_handle,
-	    "usb_ac_rele_siblings:");
-
-	/* release all siblings and ourselves */
-	pdip = ddi_get_parent(uacp->usb_ac_dip);
-	ndi_devi_enter(pdip, &circ);
-	child_dip = ddi_get_child(pdip);
-	while (child_dip != NULL) {
-		ndi_rele_devi(child_dip);
-		USB_DPRINTF_L3(PRINT_MASK_PM, uacp->usb_ac_log_handle,
-		    "usb_ac_rele_siblings: %s%d ref=%d",
-		    ddi_driver_name(child_dip), ddi_get_instance(child_dip),
-		    DEVI(child_dip)->devi_ref);
-		child_dip = ddi_get_next_sibling(child_dip);
-	}
-	ndi_devi_exit(pdip, circ);
-}
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_ah/usb_ah.c b/usr/src/uts/common/io/usb/clients/audio/usb_ah/usb_ah.c
index 796408aebc..e0358f5ecf 100644
--- a/usr/src/uts/common/io/usb/clients/audio/usb_ah/usb_ah.c
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_ah/usb_ah.c
@@ -19,28 +19,28 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
 
 /*
  * USB audio hid streams module - processes hid data
- * from HID driver and converts to a format that SADA
+ * from HID driver and converts to a format that usb_ac
  * understands. The stack looks like this :
  *	hid --> usb_ah --> usb_ac --> audio framework
  * usb_ac just acts as a passthrough layer for the converted data.
  *
- * During open, usb_ah gets the parser handle from hid and gets
- * the hardware information passed as report descriptor. Then
- * it finds out the relevant usages and stores the bitmap and other
- * information in internal data structure. When a button is pressed
- * to. say, increase/decrease the volume, a report is generated and
- * hid sends that data up through the streams. usb_ah, upon getting
- * this information and with the prior knowledge about the bitmap
- * for each button, calculates the value and sends up to SADA
- * through usb_ac. SADA in turn sends a command down to speaker to
- * increase the volume of the speaker that is managed by usb_ac.
+ * During open, usb_ah gets the parser handle from hid and gets the
+ * hardware information passed as report descriptor. Then it finds out
+ * the relevant usages and stores the bitmap and other information in
+ * internal data structure. When a button is pressed to. say,
+ * increase/decrease the volume, a report is generated and hid sends
+ * that data up through the streams. usb_ah, upon getting this
+ * information and with the prior knowledge about the bitmap for each
+ * button, calculates the value and sends up to usb_ac.  usb_ac in
+ * turn sends a command down to speaker to increase the volume of the
+ * speaker that is managed by usb_ac.
  */
 #include <sys/usb/usba.h>
 #include <sys/usb/clients/hid/hid.h>
@@ -49,12 +49,9 @@
 #include <sys/strsun.h>
 
 #include <sys/audio.h>
-#include <sys/audiovar.h>
 #include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
 #include <sys/mixer.h>
 #include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src1.h>
 
 #include <sys/usb/clients/audio/usb_audio.h>
 #include <sys/usb/clients/audio/usb_mixer.h>
@@ -76,12 +73,12 @@ static void	usb_ah_repeat_send(usb_ah_state_t *, usb_ah_button_descr_t *,
 static void	usb_ah_cancel_timeout(usb_ah_state_t *);
 static void	usb_ah_check_usage_send_data(usb_ah_state_t *, mblk_t *);
 static int	usb_ah_get_cooked_rd(usb_ah_state_t *);
+static mblk_t	*usb_ah_mk_mctl(struct iocblk, void *, size_t);
 
 /* stream qinit functions defined here */
 static int	usb_ah_open(queue_t *, dev_t *, int, int, cred_t *);
 static int	usb_ah_close(queue_t *, int, cred_t *);
-static void	usb_ah_wput(queue_t *, mblk_t *);
-static void	usb_ah_rput(queue_t *, mblk_t *);
+static int	usb_ah_rput(queue_t *, mblk_t *);
 
 /*
  * Global Variables
@@ -137,21 +134,21 @@ static struct module_info usb_ah_minfo = {
 
 /* read side for key data and ioctl replies */
 static struct qinit usb_ah_rinit = {
-	(int (*)())usb_ah_rput,
-	(int (*)())NULL,		/* service not used */
+	usb_ah_rput,
+	NULL,		/* service not used */
 	usb_ah_open,
 	usb_ah_close,
-	(int (*)())NULL,
+	NULL,
 	&usb_ah_minfo
 	};
 
-/* write side for ioctls */
+/* write side -- just pass everything down */
 static struct qinit usb_ah_winit = {
-	(int (*)())usb_ah_wput,
-	(int (*)())NULL,
+	(int (*)(queue_t *, mblk_t *))putnext,
+	NULL,
 	usb_ah_open,
 	usb_ah_close,
-	(int (*)())NULL,
+	NULL,
 	&usb_ah_minfo
 	};
 
@@ -218,17 +215,8 @@ usb_ah_open(queue_t *q, dev_t *devp, int oflag, int sflag, cred_t *crp)
 		return (0); /* already opened */
 	}
 
-	switch (sflag) {
-	case MODOPEN:
-
-		break;
-	case CLONEOPEN:
-		USB_DPRINTF_L3(PRINT_MASK_OPEN, usb_ah_log_handle,
-		    "usb_ah_open: Clone open not supported");
-
-		/* FALLTHRU */
-	default:
-
+	if (sflag != MODOPEN) {
+		/* Only module open supported */
 		return (EINVAL);
 	}
 
@@ -323,9 +311,6 @@ usb_ah_open(queue_t *q, dev_t *devp, int oflag, int sflag, cred_t *crp)
 
 	usb_ahd->usb_ah_flags |= USB_AH_OPEN;
 
-	USB_DPRINTF_L3(PRINT_MASK_OPEN, usb_ah_log_handle,
-	    "usb_ah_open exiting");
-
 	return (0);
 }
 
@@ -362,81 +347,33 @@ usb_ah_close(register queue_t *q, int flag, cred_t *crp)
 	mutex_destroy(&usb_ahd->usb_ah_mutex);
 	kmem_free(usb_ahd, sizeof (usb_ah_state_t));
 
-	USB_DPRINTF_L4(PRINT_MASK_CLOSE, usb_ah_log_handle,
-	    "usb_ah_close exiting");
-
 	return (0);
 }
 
 
 /*
- * usb_ah_wput :
- *	usb_ah	module output queue put procedure: handles M_IOCTL
- *	messages.
- */
-static void
-usb_ah_wput(register queue_t *q, register mblk_t *mp)
-{
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, usb_ah_log_handle,
-	    "usb_ah_wput entering");
-
-	switch (mp->b_datap->db_type) {
-	case M_FLUSH:
-		if (*mp->b_rptr & FLUSHW) {
-			flushq(q, FLUSHDATA);
-		}
-		if (*mp->b_rptr & FLUSHR) {
-			flushq(RD(q), FLUSHDATA);
-		}
-
-		break;
-	default:
-		break;
-	}
-
-	putnext(q, mp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, usb_ah_log_handle,
-	    "usb_ah_wput exiting:3");
-}
-
-
-/*
  * usb_ah_rput :
  *	Put procedure for input from driver end of stream (read queue).
  */
-static void
+static int
 usb_ah_rput(register queue_t *q, register mblk_t *mp)
 {
 	usb_ah_state_t		*usb_ahd;
 
 	usb_ahd = (usb_ah_state_t *)q->q_ptr;
 
-	USB_DPRINTF_L3(PRINT_MASK_ALL, usb_ah_log_handle,
-	    "usb_ah_rput: Begin, usb_ah state=0x%p, q=0x%p, mp=0x%p",
-	    (void *)usb_ahd, (void *)q, (void *)mp);
-
 	if (usb_ahd == 0) {
 		freemsg(mp);	/* nobody's listening */
 
-		return;
+		return (0);
 	}
 
 	switch (mp->b_datap->db_type) {
-	case M_FLUSH:
-		if (*mp->b_rptr & FLUSHW)
-			flushq(WR(q), FLUSHDATA);
-		if (*mp->b_rptr & FLUSHR)
-			flushq(q, FLUSHDATA);
-		freemsg(mp);
 
-		return;
 	case M_DATA:
 		if (!(usb_ahd->usb_ah_flags & USB_AH_OPEN)) {
 			freemsg(mp);	/* not ready to listen */
 
-			return;
 		} else if (MBLKL(mp) == usb_ahd->usb_ah_packet_size) {
 
 			/*
@@ -463,20 +400,23 @@ usb_ah_rput(register queue_t *q, register mblk_t *mp)
 		}
 
 		break;
+
 	case M_CTL:
 		usb_ah_mctl_receive(q, mp);
+		break;
 
-		return;
+	case M_FLUSH:
 	case M_IOCACK:
 	case M_IOCNAK:
 		putnext(q, mp);
+		break;
 
-		return;
 	default:
 		putnext(q, mp);
-
-		return;
+		break;
 	}
+
+	return (0);
 }
 
 
@@ -546,7 +486,7 @@ usb_ah_repeat_send(usb_ah_state_t *usb_ahd, usb_ah_button_descr_t *bd,
 {
 	mblk_t	*dup_mp;
 
-	bd->mblk = usba_mk_mctl(mctlmsg, buf, len);
+	bd->mblk = usb_ah_mk_mctl(mctlmsg, buf, len);
 
 	if (bd->mblk != NULL) {
 		dup_mp = usb_ah_cp_mblk(bd->mblk);
@@ -579,8 +519,6 @@ usb_ah_timeout(void *addr)
 	usb_ahd = (usb_ah_state_t *)bd->uahp;
 
 	mutex_enter(&usb_ahd->usb_ah_mutex);
-	USB_DPRINTF_L4(PRINT_MASK_ALL, usb_ah_log_handle,
-	    "usb_ah_timeout: tid=0x%p", usb_ahd->usb_ah_tid);
 
 	/*
 	 * If a release event still hasn't reached, tid will be non-zero
@@ -612,9 +550,6 @@ usb_ah_cancel_timeout(usb_ah_state_t *usb_ahd)
 {
 	queue_t	*rq = usb_ahd->usb_ah_readq;
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, usb_ah_log_handle,
-	    "usb_ah_cancel_timeout: tid=0x%p", usb_ahd->usb_ah_tid);
-
 	if (usb_ahd->usb_ah_tid) {
 		(void) quntimeout(rq, usb_ahd->usb_ah_tid);
 		usb_ahd->usb_ah_tid = 0;
@@ -816,7 +751,7 @@ usb_ah_check_usage_send_data(usb_ah_state_t *usb_ahd, mblk_t *mp)
 					/* Relative volume */
 					mctlmsg.ioc_cmd = USB_AUDIO_VOL_CHANGE;
 					mctlmsg.ioc_count = sizeof (uint_t);
-					mctl_ptr = usba_mk_mctl(mctlmsg,
+					mctl_ptr = usb_ah_mk_mctl(mctlmsg,
 					    &val, mctlmsg.ioc_count);
 					if (mctl_ptr != NULL) {
 						mutex_exit(&usb_ahd->
@@ -865,7 +800,7 @@ usb_ah_check_usage_send_data(usb_ah_state_t *usb_ahd, mblk_t *mp)
 			if (val) {
 				mctlmsg.ioc_cmd = USB_AUDIO_MUTE;
 				mctlmsg.ioc_count = sizeof (uint_t);
-				mctl_ptr = usba_mk_mctl(mctlmsg,
+				mctl_ptr = usb_ah_mk_mctl(mctlmsg,
 				    &val, mctlmsg.ioc_count);
 				if (mctl_ptr != NULL) {
 					mutex_exit(&usb_ahd->usb_ah_mutex);
@@ -887,3 +822,21 @@ usb_ah_check_usage_send_data(usb_ah_state_t *usb_ahd, mblk_t *mp)
 	mutex_exit(&usb_ahd->usb_ah_mutex);
 	freemsg(mp);
 }
+
+
+/*
+ * since usb_ac now uses LDI to access HID streams, we must change the msg
+ * type from M_CTL to M_PROTO since the streamhead will not pass M_CTLs up
+ */
+static mblk_t *
+usb_ah_mk_mctl(struct iocblk mctlmsg, void *buf, size_t len)
+{
+	mblk_t *mp;
+
+	mp = usba_mk_mctl(mctlmsg, buf, len);
+	if (mp == NULL)
+		return (NULL);
+
+	mp->b_datap->db_type = M_PROTO;
+	return (mp);
+}
diff --git a/usr/src/uts/common/io/usb/clients/audio/usb_as/usb_as.c b/usr/src/uts/common/io/usb/clients/audio/usb_as/usb_as.c
index 75ae231c8a..aad709cb37 100644
--- a/usr/src/uts/common/io/usb/clients/audio/usb_as/usb_as.c
+++ b/usr/src/uts/common/io/usb/clients/audio/usb_as/usb_as.c
@@ -23,27 +23,25 @@
  * Use is subject to license terms.
  */
 
-
 /*
- * Audio Streams Driver: This driver is responsible for
- * (1) Processing audio data messages during play and record and
- * management of isoc pipe, (2) Selecting correct alternate that matches
- * a set of parameters and management of control pipe. This streams driver
- * is pushed under usb_ac and interacts with usb_ac using streams messages.
- * When a streams message has been received from usb_ac, it is immediately
- * put on WQ. The write side service routine loops thru all the queued
- * messages, processes them and sends up a reply. If the processing involves
- * an async USBA command, the reqly is sent up after completion of the
- * command.
+ * Audio Streams Interface Driver:
+ * This driver is derived from the legacy SADA streams-based usb_as driver
+ * and serves as an intermediate measure before the full conversion to the
+ * to the Boomer framework in a follow-on phase of the Boomer project, which
+ * will utilize more comprehensive USB audio features as well.
+ *
+ * usb_as is responsible for (1) Processing audio data messages during
+ * play and record and management of isoc pipe, (2) Selecting correct
+ * alternate that matches a set of parameters and management of control pipe.
+ * This driver is opened by usb_ac and interacts with usb_ac synchronously
+ * using ioctls. If the processing involves an async USBA command, the ioctl
+ * returns after completion of the command.
  *
- * Note: (1) All streams messages from usb_ac are M_CTL messages.
- * (2) When there is a play/record, usb_as calls mixer routines directly for
- * data (play) or sends data to mixer (record).
+ * Note: When there is a play/record, usb_as calls framework routines
+ * directly for data (play) or sends data to mixer (record).
  *
- * Serialization: usb_as being a streams driver and having the requirement
- * making non-blockings calls (USBA or streams or mixer) needs to drop
- * mutexes over such calls.  But at the same time, a competing thread
- * can't be allowed to interfere with (1) pipe, (2) streams state.
+ * Serialization: A competing thread can't be allowed to interfere with
+ * (1) pipe, (2) streams state.
  * So we need some kind of serialization among the asynchronous
  * threads that can run in the driver. The serialization is mostly
  * needed to avoid races among open/close/events/power entry points
@@ -54,34 +52,23 @@
  *
  * PM Model in usb_as: Raise power during attach and lower power in detach.
  * If device is not fully powered, synchronous raise power in wsrv entry points.
- *
- * locking: Warlock is not aware of the automatic locking mechanisms for
- * streams drivers. This driver is single threaded per queue instance.
- *
- * TODO:
- *	- mdb dcmds
- *	- dump
- *	- kstat
  */
 #include <sys/usb/usba/usbai_version.h>
 #include <sys/usb/usba.h>
-#include <sys/stropts.h>
-#include <sys/strsun.h>
-#include <sys/strsubr.h>
-#include <sys/strsun.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
 
 #include <sys/audio.h>
-#include <sys/audiovar.h>
 #include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
 #include <sys/mixer.h>
 #include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
 
 #include <sys/usb/clients/audio/usb_audio.h>
 #include <sys/usb/clients/audio/usb_mixer.h>
 #include <sys/usb/clients/audio/usb_as/usb_as.h>
 
+#include "../usb_ac/audio_shim.h"
+
 /* debug support */
 uint_t	usb_as_errlevel	= USB_LOG_L4;
 uint_t	usb_as_errmask	= (uint_t)-1;
@@ -95,12 +82,9 @@ static int	usb_as_detach(dev_info_t *, ddi_detach_cmd_t);
 static int	usb_as_power(dev_info_t *, int, int);
 static int	usb_as_getinfo(dev_info_t *, ddi_info_cmd_t, void *, void **);
 
-/*
- * STREAMS module entry points
- */
-static int usb_as_open();
-static int usb_as_close();
-static int usb_as_wsrv();
+static int usb_as_open(dev_t *, int, int, cred_t *);
+static int usb_as_close(dev_t, int, int, cred_t *);
+
 
 /* support functions */
 static void	usb_as_cleanup(dev_info_t *, usb_as_state_t *);
@@ -118,11 +102,7 @@ static int	usb_as_reconnect_event_cb(dev_info_t *);
 static int	usb_as_cpr_suspend(dev_info_t *);
 static void	usb_as_cpr_resume(dev_info_t *);
 
-static int	usb_as_ioctl(queue_t *, mblk_t *);
-static int	usb_as_mctl_rcv(queue_t *, mblk_t *);
-
-static void	usb_as_default_xfer_cb(usb_pipe_handle_t, usb_ctrl_req_t *);
-static void	usb_as_default_xfer_exc_cb(usb_pipe_handle_t, usb_ctrl_req_t *);
+static int	usb_as_ioctl(dev_t, int, intptr_t, int, cred_t *, int *);
 
 static int	usb_as_pwrlvl0(usb_as_state_t *);
 static int	usb_as_pwrlvl1(usb_as_state_t *);
@@ -132,80 +112,32 @@ static void	usb_as_pm_busy_component(usb_as_state_t *);
 static void	usb_as_pm_idle_component(usb_as_state_t *);
 
 static void	usb_as_restore_device_state(dev_info_t *, usb_as_state_t *);
-static int	usb_as_setup(usb_as_state_t *, mblk_t *);
-static void	usb_as_teardown(usb_as_state_t *, mblk_t *);
-static int	usb_as_start_play(usb_as_state_t *, mblk_t *);
+static int	usb_as_setup(usb_as_state_t *);
+static void	usb_as_teardown(usb_as_state_t *);
+static int	usb_as_start_play(usb_as_state_t *, usb_audio_play_req_t *);
 static void	usb_as_continue_play(usb_as_state_t *);
-static void	usb_as_pause_play(usb_as_state_t *, mblk_t *);
+static void	usb_as_pause_play(usb_as_state_t *);
 
-static void	usb_as_qreply_error(usb_as_state_t *, queue_t *, mblk_t *);
-static void	usb_as_send_merr_up(usb_as_state_t *, mblk_t *);
-static void	usb_as_send_mctl_up(usb_as_state_t *, mblk_t *);
-static int	usb_as_set_format(usb_as_state_t *, mblk_t *);
-static int	usb_as_set_sample_freq(usb_as_state_t *, mblk_t *);
+static int	usb_as_set_format(usb_as_state_t *, usb_audio_formats_t *);
+static int	usb_as_set_sample_freq(usb_as_state_t *, int);
 static int	usb_as_send_ctrl_cmd(usb_as_state_t *, uchar_t, uchar_t,
 			ushort_t, ushort_t, ushort_t, mblk_t *, boolean_t);
 
-static void	usb_as_isoc_close_cb(usb_pipe_handle_t ph,
-				usb_opaque_t arg, int, usb_cb_flags_t);
-static int	usb_as_start_record(usb_as_state_t *, mblk_t *);
-static int	usb_as_stop_record(usb_as_state_t *, mblk_t *);
+static int	usb_as_start_record(usb_as_state_t *, audiohdl_t);
+static int	usb_as_stop_record(usb_as_state_t *);
 static void	usb_as_play_cb(usb_pipe_handle_t, usb_isoc_req_t *);
 static void	usb_as_record_cb(usb_pipe_handle_t, usb_isoc_req_t *);
 static void	usb_as_play_exc_cb(usb_pipe_handle_t, usb_isoc_req_t  *);
 static void	usb_as_record_exc_cb(usb_pipe_handle_t, usb_isoc_req_t	*);
 static int	usb_as_get_pktsize(usb_as_state_t *, usb_audio_formats_t *,
-				usb_frame_number_t);
-static void	usb_as_handle_shutdown(usb_as_state_t *, mblk_t *);
-static int	usb_as_play_isoc_data(usb_as_state_t *, mblk_t *);
+			usb_frame_number_t);
+static void	usb_as_handle_shutdown(usb_as_state_t *);
+static int	usb_as_play_isoc_data(usb_as_state_t *,
+			usb_audio_play_req_t *);
 
 /* anchor for soft state structures */
 static void	*usb_as_statep;
 
-/*
- * STREAMS Structures
- */
-
-/* STREAMS driver id and limit value structure */
-static struct module_info usb_as_modinfo = {
-	0xffff,				/* module ID number */
-	"usb_as",			/* module name */
-	USB_AUDIO_MIN_PKTSZ,		/* minimum packet size */
-	USB_AUDIO_MAX_PKTSZ,		/* maximum packet size */
-	USB_AS_HIWATER,			/* high water mark */
-	USB_AS_LOWATER			/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit usb_as_rqueue = {
-	NULL,			/* put procedure */
-	NULL,			/* service procedure */
-	usb_as_open,		/* open procedure */
-	usb_as_close,		/* close procedure */
-	NULL,			/* unused */
-	&usb_as_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit usb_as_wqueue = {
-	putq,		/* put procedure */
-	usb_as_wsrv,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&usb_as_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab usb_as_str_info = {
-	&usb_as_rqueue,	/* read queue */
-	&usb_as_wqueue,	/* write queue */
-	NULL,		/* mux lower read queue */
-	NULL,		/* mux lower write queue */
-};
 
 /*
  * DDI Structures
@@ -213,21 +145,21 @@ static struct streamtab usb_as_str_info = {
 
 /* Entry points structure */
 static struct cb_ops usb_as_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
+	usb_as_open,		/* cb_open */
+	usb_as_close,		/* cb_close */
 	nodev,			/* cb_strategy */
 	nodev,			/* cb_print */
 	nodev,			/* cb_dump */
 	nodev,			/* cb_read */
 	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
+	usb_as_ioctl,		/* cb_ioctl */
 	nodev,			/* cb_devmap */
 	nodev,			/* cb_mmap */
 	nodev,			/* cb_segmap */
 	nochpoll,		/* cb_chpoll */
 	ddi_prop_op,		/* cb_prop_op */
-	&usb_as_str_info,	/* cb_str */
-	D_MP | D_MTPERQ,	/* cb_flag */
+	NULL,			/* cb_str */
+	D_MP | D_64BIT,		/* cb_flag */
 	CB_REV,			/* cb_rev */
 	nodev,			/* cb_aread */
 	nodev,			/* cb_arwite */
@@ -263,14 +195,6 @@ static struct modlinkage usb_as_modlinkage = {
 	NULL				/* NULL terminates the list */
 };
 
-/* warlock directives */
-_NOTE(SCHEME_PROTECTS_DATA("unshared", iocblk))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", datab))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", msgb))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", queue))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", usb_pipe_policy_t))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", usb_isoc_pkt_descr))
-_NOTE(SCHEME_PROTECTS_DATA("unshared", usb_isoc_req))
 
 static usb_event_t usb_as_events = {
 	usb_as_disconnect_event_cb,
@@ -289,10 +213,6 @@ static uint_t usb_as_default_srs[] = {
 	32000,	33075, 37800, 44100, 48000, 0
 };
 
-static uint_t usb_as_mixer_srs[] = {
-	8000,	48000,	0
-};
-
 
 int
 _init(void)
@@ -413,6 +333,9 @@ usb_as_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
 	uasp->usb_as_instance = instance;
 	uasp->usb_as_dip = dip;
 
+	(void) snprintf(uasp->dstr, sizeof (uasp->dstr), "%s#%d",
+	    ddi_driver_name(dip), instance);
+
 	if (usb_client_attach(dip, USBDRV_VERSION, 0) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ATTA, uasp->usb_as_log_handle,
 		    "usb_client_attach failed");
@@ -605,42 +528,40 @@ usb_as_cleanup(dev_info_t *dip, usb_as_state_t *uasp)
  */
 /*ARGSUSED*/
 static int
-usb_as_open(queue_t *q, dev_t *devp, int flag, int sflag, cred_t *credp)
+usb_as_open(dev_t *devp, int flag, int otyp, cred_t *credp)
 {
-	usb_as_state_t	*uasp =
-	    ddi_get_soft_state(usb_as_statep,
-	    USB_AS_MINOR_TO_INSTANCE(getminor(*devp)));
+	int		inst = USB_AS_MINOR_TO_INSTANCE(getminor(*devp));
+	usb_as_state_t	*uasp = ddi_get_soft_state(usb_as_statep, inst);
+	char		*nm = "usb_as_open";
+
 	if (uasp == NULL) {
 
 		return (ENXIO);
 	}
 
 	/* Do mux plumbing stuff */
-	USB_DPRINTF_L4(PRINT_MASK_OPEN, uasp->usb_as_log_handle,
-	    "usb_as_open: Begin q=0x%p", (void *)q);
+	dinfo("%s: %s\n", uasp->dstr, nm);
+
+	mutex_enter(&uasp->usb_as_mutex);
 
-	if (sflag) {
-		USB_DPRINTF_L2(PRINT_MASK_OPEN, uasp->usb_as_log_handle,
-		    "usb_as_open: clone open not supported");
+	if (uasp->usb_as_flag == USB_AS_OPEN || credp != kcred) {
+		dwarn("%s: multiple opens or opens from userspace not "
+		    "supported\n", uasp->dstr);
+		mutex_exit(&uasp->usb_as_mutex);
 
 		return (ENXIO);
 	}
 
-	mutex_enter(&uasp->usb_as_mutex);
-
 	/* fail open on a disconnected device */
 	if (uasp->usb_as_dev_state == USB_DEV_DISCONNECTED) {
+		dinfo("%s: %s disconnected\n", uasp->dstr, nm);
 		mutex_exit(&uasp->usb_as_mutex);
 
 		return (ENODEV);
 	}
 
-	/* Initialize the queue pointers */
-	q->q_ptr = uasp;
-	WR(q)->q_ptr = uasp;
-	uasp->usb_as_rq = q;
-	uasp->usb_as_wq = WR(q);
-	uasp->usb_as_streams_flag = USB_AS_STREAMS_OPEN;
+	/* Initialize state */
+	uasp->usb_as_flag = USB_AS_OPEN;
 	mutex_exit(&uasp->usb_as_mutex);
 
 	/*
@@ -649,10 +570,7 @@ usb_as_open(queue_t *q, dev_t *devp, int flag, int sflag, cred_t *credp)
 	usb_as_pm_busy_component(uasp);
 	(void) pm_raise_power(uasp->usb_as_dip, 0, USB_DEV_OS_FULL_PWR);
 
-	qprocson(q);
-
-	USB_DPRINTF_L4(PRINT_MASK_OPEN, uasp->usb_as_log_handle,
-	    "usb_as_open: End q=0x%p", (void *)q);
+	dinfo("%s: %s done\n", uasp->dstr, nm);
 
 	return (0);
 }
@@ -664,15 +582,16 @@ usb_as_open(queue_t *q, dev_t *devp, int flag, int sflag, cred_t *credp)
  */
 /*ARGSUSED*/
 static int
-usb_as_close(queue_t *q, int flag, cred_t *credp)
+usb_as_close(dev_t dev, int flag, int otyp, cred_t *credp)
 {
-	usb_as_state_t	*uasp = (usb_as_state_t *)q->q_ptr;
+	int		inst = USB_AS_MINOR_TO_INSTANCE(getminor(dev));
+	usb_as_state_t	*uasp = ddi_get_soft_state(usb_as_statep, inst);
 
 	USB_DPRINTF_L4(PRINT_MASK_CLOSE, uasp->usb_as_log_handle,
-	    "usb_as_close: q=0x%p", (void *)q);
+	    "usb_as_close: inst=%d", inst);
 
 	mutex_enter(&uasp->usb_as_mutex);
-	uasp->usb_as_streams_flag = USB_AS_STREAMS_DISMANTLING;
+	uasp->usb_as_flag = USB_AS_DISMANTLING;
 	mutex_exit(&uasp->usb_as_mutex);
 
 	/*
@@ -685,256 +604,88 @@ usb_as_close(queue_t *q, int flag, cred_t *credp)
 
 	usb_release_access(uasp->usb_as_ser_acc);
 
-	qprocsoff(q);
-
 	/* we can now power down */
 	usb_as_pm_idle_component(uasp);
 
-	return (0);
-}
-
-
-static void
-usb_as_qreply_error(usb_as_state_t *uasp, queue_t *q, mblk_t *mp)
-{
 	mutex_enter(&uasp->usb_as_mutex);
-	uasp->usb_as_def_mblk = NULL;
+	uasp->usb_as_flag = 0;
 	mutex_exit(&uasp->usb_as_mutex);
 
-	if (!canputnext(RD(q))) {
-		freemsg(mp);
-	} else {
-		/*
-		 * Pass an error message up.
-		 */
-		mp->b_datap->db_type = M_ERROR;
-		if (mp->b_cont) {
-			freemsg(mp->b_cont);
-			mp->b_cont = NULL;
-		}
-		mp->b_rptr = mp->b_datap->db_base;
-		mp->b_wptr = mp->b_rptr + sizeof (char);
-		*mp->b_rptr = EINVAL;
-		qreply(q, mp);
-	}
-	USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_qreply_error: sending M_ERROR up q=0x%p,mp=0x%p",
-	    (void *)q, (void *)mp);
-}
-
-
-/*
- * usb_as_wsrv
- *	write service routine, processes all the queued mblks.
- *	returns DDI_SUCCESS or DDI_FAILURE
- */
-static int
-usb_as_wsrv(queue_t *q)
-{
-	int		error;
-	usb_as_state_t	*uasp = q->q_ptr;
-	mblk_t		*mp = NULL;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_wsrv: Begin q=0x%p", (void *)q);
-
-	/* process all message blocks on the queue */
-	while ((mp = getq(q)) != NULL) {
-		ASSERT(mp->b_datap != NULL);
-
-		switch (mp->b_datap->db_type) {
-		case M_FLUSH:
-			/*
-			 * Canonical flush handling :
-			 *	mp will be freed by usb_ac since it passes
-			 *	the same mp
-			 */
-			if (*mp->b_rptr & FLUSHW) {
-				flushq(q, FLUSHDATA);
-			}
-			/* read queue not used so just send up */
-			if (*mp->b_rptr & FLUSHR) {
-				*mp->b_rptr &= ~FLUSHW;
-				qreply(q, mp);
-			} else {
-				freemsg(mp);
-			}
-
-			break;
-		case M_IOCTL:
-			/* only ioctl is mixer registration data */
-			error = usb_as_ioctl(q, mp);
-
-			break;
-		case M_CTL:
-			/* process the message */
-			mutex_enter(&uasp->usb_as_mutex);
-			ASSERT(uasp->usb_as_def_mblk == NULL);
-			uasp->usb_as_def_mblk = mp;
-			mutex_exit(&uasp->usb_as_mutex);
-
-			error = usb_as_mctl_rcv(q, mp);
-			if (error != USB_SUCCESS) {
-				usb_as_qreply_error(uasp, q, mp);
-			}
-
-			break;
-		default:
-			usb_as_qreply_error(uasp, q, mp);
-
-			break;
-		}
-	}
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_wsrv: End q=0x%p", (void *)q);
-
-	return (DDI_SUCCESS);
+	return (0);
 }
 
 
 /*
- * usb_as_ioctl:
- *	usb_as handles only USB_AUDIO_MIXER_REGISTRATION ioctl
- *	NACK all other ioctl requests
- *	Returns USB_SUCCESS or USB_FAILURE
+ *
  */
+/*ARGSUSED*/
 static int
-usb_as_ioctl(queue_t *q, mblk_t *mp)
+usb_as_ioctl(dev_t dev, int cmd, intptr_t arg, int mode, cred_t *credp,
+    int *rvalp)
 {
-	int		error = USB_FAILURE;
-	usb_as_state_t	*uasp = q->q_ptr;
-	register struct iocblk *iocp;
-
-	iocp = (struct iocblk *)mp->b_rptr;
+	int		inst = USB_AS_MINOR_TO_INSTANCE(getminor(dev));
+	usb_as_state_t	*uasp = ddi_get_soft_state(usb_as_statep, inst);
+	int		rv = USB_SUCCESS;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_ioctl: Begin q=0x%p, mp=0x%p", (void *)q, (void *)mp);
+	    "usb_as_ioctl: Begin inst=%d, cmd=0x%x, arg=0x%p",
+	    inst, cmd, (void *)arg);
 
-	if (mp->b_cont == NULL) {
+	if (!(mode & FKIOCTL)) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-		    "usb_as_ioctl: no data block, q=0x%p, mp=0x%p",
-		    (void *)q, (void *)mp);
-	} else {
-		switch (iocp->ioc_cmd) {
-		case USB_AUDIO_MIXER_REGISTRATION:
-			USB_DPRINTF_L4(PRINT_MASK_ALL,
-			    uasp->usb_as_log_handle,
-			    "usb_as_ioctl(mixer reg): q=0x%p, "
-			    "mp=0x%p, b_cont_rptr=0x%p, b_cont_wptr=0x%p",
-			    (void *)q, (void *)mp, (void *)mp->b_cont->b_rptr,
-			    (void *)mp->b_cont->b_wptr);
-
-			mutex_enter(&uasp->usb_as_mutex);
-
-			/*
-			 * Copy the usb_as_reg structure to the structure
-			 * that usb_ac passed. Note that this is a structure
-			 * assignment and not a pointer assignment!
-			 */
-			*((usb_as_registration_t *)(*((
-			    usb_as_registration_t **)mp->
-			    b_cont->b_rptr))) = uasp->usb_as_reg;
-
-			mp->b_cont->b_wptr = mp->b_cont->b_rptr +
-			    sizeof (usb_as_registration_t *);
-
-			mutex_exit(&uasp->usb_as_mutex);
-			error = USB_SUCCESS;
-			break;
-		default:
-			USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-			    "usb_as_ioctl: unknown IOCTL, cmd=%d",
-			    iocp->ioc_cmd);
-			break;
-		}
-	}
-
-	iocp->ioc_rval = 0;
-	if (error == USB_FAILURE) {
-		iocp->ioc_error = ENOTTY;
-		mp->b_datap->db_type = M_IOCNAK;
-	} else {
-		iocp->ioc_error = 0;
-		mp->b_datap->db_type = M_IOCACK;
+		    "usb_as_ioctl: inst=%d, user space not supported", inst);
+		return (ENXIO);
 	}
 
-	/*
-	 * Send the response up
-	 */
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_ioctl: error=%d, q=0x%p, mp=0x%p", error,
-	    (void *)q, (void *)mp);
-
-	qreply(q, mp);
-
-	return (error);
-}
-
-
-/*
- * usb_as_mctl_rcv:
- *	Handle M_CTL requests from usb_ac.
- *	Returns USB_SUCCESS/FAILURE
- */
-static int
-usb_as_mctl_rcv(queue_t *q, mblk_t *mp)
-{
-	int		error = USB_FAILURE;
-	usb_as_state_t	*uasp = q->q_ptr;
-	struct iocblk	*iocp;
+	mutex_enter(&uasp->usb_as_mutex);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_mctl_rcv: Begin q=0x%p mp=0x%p",
-	    (void *)q, (void *)mp);
+	switch (cmd) {
+	case USB_AUDIO_MIXER_REGISTRATION:
+		USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
+		    "usb_as_ioctl(mixer reg): inst=%d", inst);
 
-	ASSERT(mp != NULL);
+		/*
+		 * Copy the usb_as_reg structure to the structure
+		 * that usb_ac passed. Note that this is a structure
+		 * assignment and not a pointer assignment!
+		 */
+		*(usb_as_registration_t *)arg = uasp->usb_as_reg;
 
-	/*
-	 * Uopn success, each function sends up a reply either immediately,
-	 * or on callback. On failure, reply is send up in the wsrv.
-	 */
-	iocp = (struct iocblk *)mp->b_rptr;
-	mutex_enter(&uasp->usb_as_mutex);
-	switch (iocp->ioc_cmd) {
-		case USB_AUDIO_SET_FORMAT:
-			error = usb_as_set_format(uasp, mp);
-			break;
-		case USB_AUDIO_SET_SAMPLE_FREQ:
-			error = usb_as_set_sample_freq(uasp, mp);
-			break;
-		case USB_AUDIO_SETUP:
-			error = usb_as_setup(uasp, mp);
-			break;
-		case USB_AUDIO_TEARDOWN:
-			usb_as_teardown(uasp, mp);
-			error = USB_SUCCESS;
-			break;
-		case USB_AUDIO_START_PLAY:
-			error = usb_as_start_play(uasp, mp);
-			break;
-		case USB_AUDIO_STOP_PLAY:
-		case USB_AUDIO_PAUSE_PLAY:
-			usb_as_pause_play(uasp, mp);
-			error = USB_SUCCESS;
-			break;
-		case USB_AUDIO_START_RECORD:
-			error = usb_as_start_record(uasp, mp);
-			break;
-		case USB_AUDIO_STOP_RECORD:
-			error = usb_as_stop_record(uasp, mp);
-			break;
-		default:
-			break;
+		break;
+	case USB_AUDIO_SET_FORMAT:
+		rv = usb_as_set_format(uasp, (usb_audio_formats_t *)arg);
+		break;
+	case USB_AUDIO_SET_SAMPLE_FREQ:
+		rv = usb_as_set_sample_freq(uasp, *(int *)arg);
+		break;
+	case USB_AUDIO_SETUP:
+		rv = usb_as_setup(uasp);
+		break;
+	case USB_AUDIO_TEARDOWN:
+		usb_as_teardown(uasp);
+		break;
+	case USB_AUDIO_START_PLAY:
+		rv = usb_as_start_play(uasp, (usb_audio_play_req_t *)arg);
+		break;
+	case USB_AUDIO_STOP_PLAY:
+	case USB_AUDIO_PAUSE_PLAY:
+		usb_as_pause_play(uasp);
+		break;
+	case USB_AUDIO_START_RECORD:
+		rv = usb_as_start_record(uasp, *(audiohdl_t *)arg);
+		break;
+	case USB_AUDIO_STOP_RECORD:
+		rv = usb_as_stop_record(uasp);
+		break;
+	default:
+		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
+		    "usb_as_ioctl: unknown IOCTL, cmd=%d", cmd);
+		break;
 	}
 
 	mutex_exit(&uasp->usb_as_mutex);
 
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_mctl_rcv: End q=0x%p mp=0x%p error=%d",
-	    (void *)q, (void *)mp, error);
-
-	return (error);
+	return (rv == USB_SUCCESS ? 0 : ENXIO);
 }
 
 
@@ -948,35 +699,25 @@ usb_as_mctl_rcv(queue_t *q, mblk_t *mp)
  *	Therefore, we ignore errors and carry on regardless
  */
 static int
-usb_as_set_sample_freq(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_set_sample_freq(usb_as_state_t *uasp, int freq)
 {
-	int	freq, alt, ep;
+	int	alt, ep;
 	mblk_t	*data;
 	int	rval = USB_FAILURE;
 	boolean_t ignore_errors;
 
-	ASSERT(mp != NULL);
-	ASSERT(mp->b_cont != NULL);
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	alt = uasp->usb_as_alternate;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_set_sample_freq: mp=0x%p cont_sr=%d", (void *)mp,
-	    uasp->usb_as_alts[alt].alt_continuous_sr);
+	    "usb_as_set_sample_freq: inst=%d cont_sr=%d freq=%d",
+	    ddi_get_instance(uasp->usb_as_dip),
+	    uasp->usb_as_alts[alt].alt_continuous_sr, freq);
 
 	ignore_errors = B_TRUE;
 
 	ep = uasp->usb_as_alts[alt].alt_ep->bEndpointAddress;
-	freq = *((int *)mp->b_cont->b_rptr);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_set_sample_freq: freq = %d", freq);
-
-	if (mp->b_cont) {
-		freemsg(mp->b_cont);
-		mp->b_cont = NULL;
-	}
 
 	data = allocb(4, BPRI_HI);
 	if (data) {
@@ -998,11 +739,10 @@ usb_as_set_sample_freq(usb_as_state_t *uasp, mblk_t *mp)
 		    ignore_errors)) != USB_SUCCESS) {
 			USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 			    "usb_as_set_sample_freq: set sample freq failed");
-
-			freemsg(data);
 		}
 		mutex_enter(&uasp->usb_as_mutex);
 	}
+	freemsg(data);
 
 	return (rval);
 }
@@ -1011,25 +751,23 @@ usb_as_set_sample_freq(usb_as_state_t *uasp, mblk_t *mp)
 /*
  * usb_as_set_format:
  *	Matches channel, encoding and precision and find out
- *	the right alternate. Sets alternate interface.
+ *	the right alternate. Sets alternate interface and returns it.
  */
 static int
-usb_as_set_format(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_set_format(usb_as_state_t *uasp, usb_audio_formats_t *format)
 {
 	int		n;
 	usb_as_registration_t *reg;
-	usb_audio_formats_t *format;
 	int		alt, rval;
 	uint_t		interface;
 
-	ASSERT(mp != NULL);
-	ASSERT(mp->b_cont != NULL);
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	if (uasp->usb_as_request_count) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-		    "usb_as_set_format: failing mp=0x%p, rq_cnt=%d",
-		    (void *)mp, uasp->usb_as_request_count);
+		    "usb_as_set_format: failing inst=%d, rq_cnt=%d",
+		    ddi_get_instance(uasp->usb_as_dip),
+		    uasp->usb_as_request_count);
 
 		return (USB_FAILURE);
 	}
@@ -1038,13 +776,12 @@ usb_as_set_format(usb_as_state_t *uasp, mblk_t *mp)
 
 	reg = &uasp->usb_as_reg;
 	interface = uasp->usb_as_ifno;
-	format = (usb_audio_formats_t *)mp->b_cont->b_rptr;
 
-	bcopy(format, &uasp->usb_as_curr_format, sizeof (usb_audio_formats_t));
+	uasp->usb_as_curr_format = *format;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_set_format: mp=0x%p, reg=0x%p, format=0x%p",
-	    (void *)mp, (void *)reg, (void *)format);
+	    "usb_as_set_format: inst=%d, reg=0x%p, format=0x%p",
+	    ddi_get_instance(uasp->usb_as_dip), (void *)reg, (void *)format);
 
 	for (n = 0; n < reg->reg_n_formats; n++) {
 		if ((format->fmt_chns == reg->reg_formats[n].fmt_chns) &&
@@ -1060,7 +797,7 @@ usb_as_set_format(usb_as_state_t *uasp, mblk_t *mp)
 		}
 	}
 
-	if (n > reg->reg_n_formats) {
+	if (n >= reg->reg_n_formats) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 		    "usb_as_set_format: Didn't find a matching alt");
 
@@ -1075,19 +812,23 @@ usb_as_set_format(usb_as_state_t *uasp, mblk_t *mp)
 
 	mutex_exit(&uasp->usb_as_mutex);
 
-	if ((rval = usb_as_send_ctrl_cmd(uasp,
+	rval = usb_as_send_ctrl_cmd(uasp,
 					/* bmRequestType */
 	    USB_DEV_REQ_HOST_TO_DEV | USB_DEV_REQ_RCPT_IF,
 	    USB_REQ_SET_IF,		/* bRequest */
 	    alt,			/* wValue */
 	    interface,			/* wIndex */
 	    0,				/* wLength */
-	    NULL, B_FALSE)) != USB_SUCCESS) {
+	    NULL, B_FALSE);
+
+	mutex_enter(&uasp->usb_as_mutex);
+
+	if (rval != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 		    "usb_as_set_format: set_alternate failed");
-
+	} else {
+		format->fmt_alt = (uchar_t)alt;
 	}
-	mutex_enter(&uasp->usb_as_mutex);
 
 	return (rval);
 }
@@ -1099,17 +840,17 @@ usb_as_set_format(usb_as_state_t *uasp, mblk_t *mp)
  *	is available.
  */
 static int
-usb_as_setup(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_setup(usb_as_state_t *uasp)
 {
 	int alt = uasp->usb_as_alternate;
 	usb_ep_descr_t *ep = (usb_ep_descr_t *)uasp->usb_as_alts[alt].alt_ep;
 	int rval;
 
-	ASSERT(mp != NULL);
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_setup: Begin usb_as_setup, mp=0x%p", (void *)mp);
+	    "usb_as_setup: Begin usb_as_setup, inst=%d",
+	    ddi_get_instance(uasp->usb_as_dip));
 
 	ASSERT(uasp->usb_as_request_count == 0);
 
@@ -1124,7 +865,7 @@ usb_as_setup(usb_as_state_t *uasp, mblk_t *mp)
 
 	/* open isoc pipe, may fail if there is no bandwidth  */
 	rval = usb_pipe_open(uasp->usb_as_dip, ep, &uasp->usb_as_isoc_pp,
-	    0, &uasp->usb_as_isoc_ph);
+	    USB_FLAGS_SLEEP, &uasp->usb_as_isoc_ph);
 
 	if (rval != USB_SUCCESS) {
 		switch (rval) {
@@ -1151,11 +892,9 @@ usb_as_setup(usb_as_state_t *uasp, mblk_t *mp)
 
 	(void) usb_pipe_set_private(uasp->usb_as_isoc_ph, (usb_opaque_t)uasp);
 
-	/* return reply up */
 	mutex_enter(&uasp->usb_as_mutex);
 	uasp->usb_as_audio_state = USB_AS_IDLE;
 	uasp->usb_as_setup_cnt++;
-	usb_as_send_mctl_up(uasp, NULL);
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_setup: End");
@@ -1169,10 +908,12 @@ usb_as_setup(usb_as_state_t *uasp, mblk_t *mp)
  *
  */
 static void
-usb_as_teardown(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_teardown(usb_as_state_t *uasp)
 {
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_teardown: Begin mp=0x%p", (void *)mp);
+	    "usb_as_teardown: Begin inst=%d",
+	    ddi_get_instance(uasp->usb_as_dip));
+
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	uasp->usb_as_audio_state = USB_AS_IDLE;
@@ -1185,10 +926,9 @@ usb_as_teardown(usb_as_state_t *uasp, mblk_t *mp)
 		mutex_exit(&uasp->usb_as_mutex);
 
 		/* reply mp will be sent up in isoc close callback */
-		usb_pipe_close(uasp->usb_as_dip, uasp->usb_as_isoc_ph, 0,
-		    usb_as_isoc_close_cb, (usb_opaque_t)uasp);
+		usb_pipe_close(uasp->usb_as_dip, uasp->usb_as_isoc_ph,
+		    USB_FLAGS_SLEEP, NULL, (usb_opaque_t)NULL);
 
-		/* wait for callback to send up a reply */
 		mutex_enter(&uasp->usb_as_mutex);
 		uasp->usb_as_isoc_ph = NULL;
 
@@ -1198,8 +938,6 @@ usb_as_teardown(usb_as_state_t *uasp, mblk_t *mp)
 	} else {
 		USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 		    "usb_as_teardown: Pipe already closed");
-
-		usb_as_send_mctl_up(uasp, NULL);
 	}
 
 	ASSERT(uasp->usb_as_setup_cnt == 0);
@@ -1210,32 +948,24 @@ usb_as_teardown(usb_as_state_t *uasp, mblk_t *mp)
 
 
 /*
- * usb_as_start_play:
- *	this function is called from usb_as_mctl_rcv
+ * usb_as_start_play
  */
 static int
-usb_as_start_play(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_start_play(usb_as_state_t *uasp, usb_audio_play_req_t *play_req)
 {
-	usb_audio_play_req_t *play_req;
-	int		samples;
 	int		n_requests;
 	int		rval = USB_FAILURE;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_start_play: Begin mp=0x%p, req_cnt=%d",
-	    (void *)mp, uasp->usb_as_request_count);
+	    "usb_as_start_play: Begin inst=%d, req_cnt=%d",
+	    ddi_get_instance(uasp->usb_as_dip), uasp->usb_as_request_count);
 
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
-	ASSERT(mp && mp->b_cont);
-
-	play_req = (usb_audio_play_req_t *)mp->b_cont->b_rptr;
 	uasp->usb_as_request_samples = play_req->up_samples;
 	uasp->usb_as_ahdl = play_req->up_handle;
 	uasp->usb_as_audio_state = USB_AS_ACTIVE;
 
-	samples = uasp->usb_as_request_samples;
-
 	if ((uasp->usb_as_request_count >= USB_AS_MAX_REQUEST_COUNT) ||
 	    (uasp->usb_as_audio_state == USB_AS_IDLE) ||
 	    (uasp->usb_as_audio_state == USB_AS_PLAY_PAUSED)) {
@@ -1246,27 +976,18 @@ usb_as_start_play(usb_as_state_t *uasp, mblk_t *mp)
 	} else {
 		USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 		    "usb_as_start_play: samples=%d requestcount=%d ",
-		    samples, uasp->usb_as_request_count);
+		    uasp->usb_as_request_samples, uasp->usb_as_request_count);
 
 		/* queue up as many requests as allowed */
 		for (n_requests = uasp->usb_as_request_count;
 		    n_requests < USB_AS_MAX_REQUEST_COUNT; n_requests++) {
-			if ((rval = usb_as_play_isoc_data(uasp, mp)) !=
+			if ((rval = usb_as_play_isoc_data(uasp, play_req)) !=
 			    USB_SUCCESS) {
 				break;
 			}
 		}
 	}
 
-	/*
-	 * send mctl up for success. For failure, usb_as_wsrv
-	 * will send an merr up.
-	 */
-	if (rval == USB_SUCCESS) {
-		usb_as_send_mctl_up(uasp, NULL);
-	}
-
-
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_start_play: End");
 
@@ -1281,7 +1002,6 @@ usb_as_start_play(usb_as_state_t *uasp, mblk_t *mp)
 static void
 usb_as_continue_play(usb_as_state_t *uasp)
 {
-	int		samples;
 	int		n_requests;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
@@ -1291,13 +1011,11 @@ usb_as_continue_play(usb_as_state_t *uasp)
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	if (uasp->usb_as_dev_state == USB_DEV_DISCONNECTED) {
-		usb_as_handle_shutdown(uasp, NULL);
+		usb_as_handle_shutdown(uasp);
 
 		return;
 	}
 
-	samples = uasp->usb_as_request_samples;
-
 	if ((uasp->usb_as_request_count >= USB_AS_MAX_REQUEST_COUNT) ||
 	    (uasp->usb_as_audio_state == USB_AS_IDLE) ||
 	    (uasp->usb_as_audio_state == USB_AS_PLAY_PAUSED)) {
@@ -1307,7 +1025,7 @@ usb_as_continue_play(usb_as_state_t *uasp)
 	} else {
 		USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 		    "usb_as_continue_play: samples=%d requestcount=%d ",
-		    samples, uasp->usb_as_request_count);
+		    uasp->usb_as_request_samples, uasp->usb_as_request_count);
 
 		/* queue up as many requests as allowed */
 		for (n_requests = uasp->usb_as_request_count;
@@ -1326,16 +1044,13 @@ usb_as_continue_play(usb_as_state_t *uasp)
 
 
 static void
-usb_as_handle_shutdown(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_handle_shutdown(usb_as_state_t *uasp)
 {
 	audiohdl_t	ahdl;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_handl_shutdown, mp=0x%p", (void *)mp);
-
-	if (mp != NULL) {
-		usb_as_send_mctl_up(uasp, NULL);
-	}
+	    "usb_as_handle_shutdown, inst=%d",
+	    ddi_get_instance(uasp->usb_as_dip));
 
 	USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_handle_shutdown: am_play_shutdown");
@@ -1345,13 +1060,13 @@ usb_as_handle_shutdown(usb_as_state_t *uasp, mblk_t *mp)
 	ahdl = uasp->usb_as_ahdl;
 
 	mutex_exit(&uasp->usb_as_mutex);
-	am_play_shutdown(ahdl, AUDIO_NO_CHANNEL);
+	am_play_shutdown(ahdl);
 	mutex_enter(&uasp->usb_as_mutex);
 }
 
 
 static int
-usb_as_play_isoc_data(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_play_isoc_data(usb_as_state_t *uasp, usb_audio_play_req_t *play_req)
 {
 	int		rval = USB_FAILURE;
 
@@ -1389,8 +1104,8 @@ usb_as_play_isoc_data(usb_as_state_t *uasp, mblk_t *mp)
 	}
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_play_isoc_data: Begin bufsize=0x%lx, mp=0x%p", bufsize,
-	    (void *)mp);
+	    "usb_as_play_isoc_data: Begin bufsize=0x%lx, inst=%d", bufsize,
+	    ddi_get_instance(uasp->usb_as_dip));
 
 	mutex_exit(&uasp->usb_as_mutex);
 
@@ -1402,14 +1117,22 @@ usb_as_play_isoc_data(usb_as_state_t *uasp, mblk_t *mp)
 		goto done;
 	}
 
-	if ((count = am_get_audio(ahdl, (void *)data->b_wptr,
-	    AUDIO_NO_CHANNEL, bufsize / precision)) == 0) {
+	/*
+	 * restriction of Boomer: cannot call am_get_audio() in the context
+	 * of start so we play a fragment of silence at first
+	 */
+	if (play_req != NULL) {
+		bzero(data->b_wptr, bufsize);
+		count = bufsize / precision;
+
+	} else if ((count = am_get_audio(ahdl, (void *)data->b_wptr,
+	    bufsize / precision)) == 0) {
 		mutex_enter(&uasp->usb_as_mutex);
 		if (uasp->usb_as_request_count == 0) {
-			usb_as_handle_shutdown(uasp, NULL);
+			usb_as_handle_shutdown(uasp);
 
 			/* Don't return failure for 0 bytes of data sent */
-			if (mp) {
+			if (play_req) {
 				/*
 				 * Since we set rval to SUCCESS
 				 * we treat it as a special case
@@ -1536,13 +1259,13 @@ done:
 }
 
 
-/*ARGSUSED*/
 static void
-usb_as_pause_play(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_pause_play(usb_as_state_t *uasp)
 {
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
+
+	/* this will stop the isoc request in the play callback */
 	uasp->usb_as_audio_state = USB_AS_PLAY_PAUSED;
-	usb_as_send_mctl_up(uasp, NULL);
 }
 
 
@@ -1628,7 +1351,7 @@ usb_as_play_exc_cb(usb_pipe_handle_t ph, usb_isoc_req_t *isoc_req)
 	mutex_enter(&uasp->usb_as_mutex);
 	uasp->usb_as_rcv_debug_count++;
 	uasp->usb_as_request_count--;
-	usb_as_handle_shutdown(uasp, NULL);
+	usb_as_handle_shutdown(uasp);
 
 	USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_play_exc_cb: SEND CNT=%d, RCV COUNT=%d",
@@ -1646,7 +1369,7 @@ usb_as_play_exc_cb(usb_pipe_handle_t ph, usb_isoc_req_t *isoc_req)
  * usb_as_start_record
  */
 static int
-usb_as_start_record(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_start_record(usb_as_state_t *uasp, audiohdl_t ahdl)
 {
 	int		rval = USB_FAILURE;
 	usb_isoc_req_t *isoc_req;
@@ -1654,10 +1377,9 @@ usb_as_start_record(usb_as_state_t *uasp, mblk_t *mp)
 	ushort_t	n_pkt = 1, pkt;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_start_record: mp=0x%p", (void *)mp);
+	    "usb_as_start_record: inst=%d",
+	    ddi_get_instance(uasp->usb_as_dip));
 
-	ASSERT(mp != NULL);
-	ASSERT(mp->b_cont != NULL);
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	/*
@@ -1668,7 +1390,7 @@ usb_as_start_record(usb_as_state_t *uasp, mblk_t *mp)
 
 	if (uasp->usb_as_audio_state == USB_AS_IDLE) {
 
-		uasp->usb_as_ahdl = *((audiohdl_t *)mp->b_cont->b_rptr);
+		uasp->usb_as_ahdl = ahdl;
 		uasp->usb_as_audio_state = USB_AS_ACTIVE;
 		mutex_exit(&uasp->usb_as_mutex);
 
@@ -1712,8 +1434,6 @@ usb_as_start_record(usb_as_state_t *uasp, mblk_t *mp)
 			usb_free_isoc_req(isoc_req);
 			isoc_req = NULL;
 		}
-	} else {
-		usb_as_send_mctl_up(uasp, NULL);
 	}
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
@@ -1723,20 +1443,16 @@ usb_as_start_record(usb_as_state_t *uasp, mblk_t *mp)
 }
 
 
-/*ARGSUSED*/
 static int
-usb_as_stop_record(usb_as_state_t *uasp, mblk_t *mp)
+usb_as_stop_record(usb_as_state_t *uasp)
 {
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_stop_record: ");
 	ASSERT(mutex_owned(&uasp->usb_as_mutex));
 
 	/* if we are disconnected, the pipe will be closed anyways */
-	if (uasp->usb_as_dev_state == USB_DEV_DISCONNECTED) {
-		usb_as_send_mctl_up(uasp, NULL);
-
+	if (uasp->usb_as_dev_state == USB_DEV_DISCONNECTED)
 		return (USB_SUCCESS);
-	}
 
 	switch (uasp->usb_as_audio_state) {
 	case USB_AS_ACTIVE:
@@ -1750,17 +1466,14 @@ usb_as_stop_record(usb_as_state_t *uasp, mblk_t *mp)
 		    USB_FLAGS_SLEEP);
 		mutex_enter(&uasp->usb_as_mutex);
 
-		usb_as_send_mctl_up(uasp, NULL);
-
 		break;
 	case USB_AS_STOP_POLLING_STARTED:
 		/* A stop polling in progress, wait for completion and reply */
 		break;
 	default:
-		usb_as_send_mctl_up(uasp, NULL);
+		break;
 	}
 
-
 	return (USB_SUCCESS);
 }
 
@@ -1895,8 +1608,7 @@ usb_as_record_cb(usb_pipe_handle_t ph, usb_isoc_req_t *isoc_req)
 
 			am_send_audio(ahdl,
 			    isoc_req->isoc_data->b_rptr + offset,
-			    AUDIO_NO_CHANNEL, isoc_req->
-			    isoc_pkt_descr[i].isoc_pkt_actual_length /
+			    isoc_req->isoc_pkt_descr[i].isoc_pkt_actual_length /
 			    precision);
 
 			mutex_enter(&uasp->usb_as_mutex);
@@ -2000,250 +1712,36 @@ usb_as_send_ctrl_cmd(usb_as_state_t *uasp,
 	ushort_t wValue, ushort_t wIndex, ushort_t wLength,
 	mblk_t	*data, boolean_t ignore_errors)
 {
-	usb_ctrl_req_t *reqp;
-
+	usb_ctrl_setup_t setup;
+	usb_cr_t cr;
+	usb_cb_flags_t cf;
 
 	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
 	    "usb_as_send_ctrl_cmd: Begin bmRequestType=%d,\n\t"
 	    "bRequest=%d, wValue=%d, wIndex=%d, wLength=%d, data=0x%p",
 	    bmRequestType, bRequest, wValue, wIndex, wLength, (void *)data);
 
-	if ((reqp = usb_alloc_ctrl_req(uasp->usb_as_dip, 0, 0)) == NULL) {
-
-		mutex_enter(&uasp->usb_as_mutex);
-		uasp->usb_as_xfer_cr = USB_AS_SEND_MERR;
-		mutex_exit(&uasp->usb_as_mutex);
-
-		return (USB_FAILURE);
-	}
-
-	reqp->ctrl_bmRequestType	= bmRequestType;
-	reqp->ctrl_bRequest		= bRequest;
-	reqp->ctrl_wValue		= wValue;
-	reqp->ctrl_wIndex		= wIndex;
-	reqp->ctrl_wLength		= wLength;
-	reqp->ctrl_data			= data;
-	reqp->ctrl_attributes		= 0;
-	reqp->ctrl_client_private	= (usb_opaque_t)uasp;
-	reqp->ctrl_cb			= usb_as_default_xfer_cb;
-	reqp->ctrl_exc_cb		= ignore_errors ?
-	    usb_as_default_xfer_cb : usb_as_default_xfer_exc_cb;
-
-	/* Send async command down */
-	if (usb_pipe_ctrl_xfer(uasp->usb_as_default_ph, reqp, 0) !=
-	    USB_SUCCESS) {
+	setup.bmRequestType	= bmRequestType & ~USB_DEV_REQ_DEV_TO_HOST;
+	setup.bRequest		= bRequest;
+	setup.wValue		= wValue;
+	setup.wIndex		= wIndex;
+	setup.wLength		= wLength;
+	setup.attrs		= 0;
 
+	if (usb_pipe_ctrl_xfer_wait(uasp->usb_as_default_ph, &setup, &data,
+	    &cr, &cf, 0) != USB_SUCCESS) {
 		USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-		    "usb_as_send_ctrl_cmd: usba xfer failed (req=%d)",
-		    bRequest);
+		    "usb_as_send_ctrl_cmd: usba xfer failed (req=%d), "
+		    "completion reason: 0x%x, completion flags: 0x%x",
+		    bRequest, cr, cf);
 
-		mutex_enter(&uasp->usb_as_mutex);
-		uasp->usb_as_xfer_cr = USB_AS_SEND_MERR;
-		mutex_exit(&uasp->usb_as_mutex);
-		usb_free_ctrl_req(reqp);
-
-		return (USB_FAILURE);
+		return (ignore_errors ? USB_SUCCESS: USB_FAILURE);
 	}
 
 	return (USB_SUCCESS);
 }
 
 
-static void
-usb_as_send_merr_up(usb_as_state_t *uasp, mblk_t *mp)
-{
-	queue_t *q;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_send_merr_up: data=0x%p", (void *)mp);
-
-	ASSERT(mutex_owned(&uasp->usb_as_mutex));
-	q = uasp->usb_as_rq;
-
-	mp->b_datap->db_type = M_ERROR;
-
-	if (mp->b_cont) {
-		freemsg(mp->b_cont);
-		mp->b_cont = NULL;
-	}
-
-	mp->b_rptr = mp->b_datap->db_base;
-	mp->b_wptr = mp->b_rptr + sizeof (char);
-	*mp->b_rptr = EINVAL;
-
-	mutex_exit(&uasp->usb_as_mutex);
-	if (!canputnext(RD(q))) {
-		freemsg(mp);
-		mp = NULL;
-	} else {
-		putnext(RD(q), mp);
-	}
-	mutex_enter(&uasp->usb_as_mutex);
-}
-
-
-static void
-usb_as_send_mctl_up(usb_as_state_t *uasp, mblk_t *data)
-{
-	mblk_t		*tmp, *mp;
-	queue_t		*q;
-	struct iocblk	*iocp;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_send_mctl_up: data=0x%p", (void *)data);
-	ASSERT(mutex_owned(&uasp->usb_as_mutex));
-
-	q = uasp->usb_as_rq;
-	mp = uasp->usb_as_def_mblk;
-	ASSERT(mp != NULL);
-
-	/* Free the b_cont of the original mblk_t, if any */
-	if (mp->b_cont) {
-		freemsg(mp->b_cont);
-		mp->b_cont = NULL;
-	}
-
-	/*
-	 * If we have response to send up, attach it at the b_cont
-	 * of the mctl message. Otherwise just send the mctl message
-	 * up and the module above will decode the command
-	 */
-	iocp = (struct iocblk *)mp->b_rptr;
-	iocp->ioc_error = 0;
-
-	switch (iocp->ioc_cmd) {
-	case USB_AUDIO_SET_FORMAT:
-		freemsg(data);
-
-		/*
-		 * we cannot easily recover if we can't get an mblk
-		 * so we have to sleep here
-		 */
-		tmp = allocb_wait(sizeof (int), BPRI_HI,
-		    STR_NOSIG, NULL);
-		iocp->ioc_count = sizeof (int);
-		*(int *)tmp->b_wptr = uasp->usb_as_alternate;
-		tmp->b_wptr += sizeof (int);
-		mp->b_cont = tmp;
-
-		USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-		    "usb_as_send_mctl_up: set_format returning,alt=%d",
-		    uasp->usb_as_alternate);
-
-		break;
-	default:
-		if (data != NULL) {
-			/*
-			 * Use the original mp to send the message up
-			 * This should already have the right ioc_cmd in.
-			 */
-			iocp->ioc_count = MBLKL(data);
-			mp->b_cont = data;
-		} else {
-			iocp->ioc_count = 0;
-		}
-		break;
-	}
-	uasp->usb_as_def_mblk = NULL;
-	mutex_exit(&uasp->usb_as_mutex);
-	if (!canputnext(q)) {
-		freemsg(mp);
-		mp = NULL;
-	} else {
-		putnext(q, mp);
-	}
-	mutex_enter(&uasp->usb_as_mutex);
-}
-
-
-/*
- * usb_as_default_xfer_cb:
- *	Callback routine for the async control xfer. Reply mctl here.
- */
-/*ARGSUSED*/
-static void
-usb_as_default_xfer_cb(usb_pipe_handle_t def, usb_ctrl_req_t *reqp)
-{
-	usb_as_state_t	*uasp = (usb_as_state_t *)reqp->ctrl_client_private;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_default_xfer_cb: ph=0x%p, reqp=0x%p",
-	    (void *)def, (void *)reqp);
-
-	ASSERT((reqp->ctrl_cb_flags & USB_CB_INTR_CONTEXT) == 0);
-
-	mutex_enter(&uasp->usb_as_mutex);
-	uasp->usb_as_xfer_cr = USB_AS_SEND_MCTL;
-	usb_as_send_mctl_up(uasp, NULL);
-	mutex_exit(&uasp->usb_as_mutex);
-
-	usb_free_ctrl_req(reqp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_default_xfer_cb: End");
-}
-
-
-/*
- * usb_as_isoc_close_cb()
- *	called from teardown usb_pipe_close
- */
-static void
-usb_as_isoc_close_cb(usb_pipe_handle_t ph, usb_opaque_t arg,
-	int rval, usb_cb_flags_t cb_flags)
-{
-	usb_as_state_t	*uasp = (usb_as_state_t  *)arg;
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_isoc_close_cb: ph=0x%p arg=0x%p cb_flags=0x%x",
-	    (void *)ph, (void *)arg, cb_flags);
-
-	/* pipe close cannot fail */
-	ASSERT(rval == USB_SUCCESS);
-
-	mutex_enter(&uasp->usb_as_mutex);
-	usb_as_send_mctl_up(uasp, NULL);
-	mutex_exit(&uasp->usb_as_mutex);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_isoc_close_cb: End");
-}
-
-
-/*
- * usb_as_default_exc_cb:
- *	Exception callback for the default pipe. Autoclearing took care
- *	of the recovery
- */
-/*ARGSUSED*/
-static void
-usb_as_default_xfer_exc_cb(usb_pipe_handle_t def, usb_ctrl_req_t *reqp)
-{
-	usb_as_state_t	*uasp = (usb_as_state_t *)reqp->ctrl_client_private;
-	mblk_t		*mp;
-
-	USB_DPRINTF_L2(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_default_xfer_exc_cb: ph=0x%p, reqp=0x%p",
-	    (void *)def, (void *)reqp);
-
-	ASSERT((reqp->ctrl_cb_flags & USB_CB_INTR_CONTEXT) == 0);
-
-	mutex_enter(&uasp->usb_as_mutex);
-	uasp->usb_as_xfer_cr = USB_AS_SEND_MERR;
-	mp = uasp->usb_as_def_mblk;
-	uasp->usb_as_def_mblk = NULL;
-
-	usb_as_send_merr_up(uasp, mp);
-
-	mutex_exit(&uasp->usb_as_mutex);
-
-	usb_free_ctrl_req(reqp);
-
-	USB_DPRINTF_L4(PRINT_MASK_ALL, uasp->usb_as_log_handle,
-	    "usb_as_default_xfer_exc_cb: End");
-}
-
-
 /*
  * Power management
  */
@@ -2801,30 +2299,9 @@ usb_as_prepare_registration_data(usb_as_state_t   *uasp)
 		}
 	}
 
-	/*
-	 * we prefer that a valid format supports all our
-	 * default sample rates. If not we delete sample rates
-	 * to get a set that is supported by all formats.
-	 *
-	 * Continuous sample rate will be checked in set_format
-	 * command for a particular alternate. This is interface
-	 * specific registration data and not per alternate.
-	 */
-	reg->reg_mixer_srs.ad_srs = reg->reg_mixer_srs_list;
-	reg->reg_mixer_srs.ad_limits = MIXER_SRS_FLAG_SR_LIMITS;
-
-	/* copy over sample rate table	but zero it first */
-	bzero(reg->reg_mixer_srs_list, sizeof (reg->reg_mixer_srs_list));
-	bcopy(usb_as_mixer_srs, reg->reg_mixer_srs_list,
-	    sizeof (usb_as_mixer_srs));
-
-	reg->reg_compat_srs.ad_srs = reg->reg_compat_srs_list;
-	reg->reg_compat_srs.ad_limits = MIXER_SRS_FLAG_SR_NOT_LIMITS;
-
 	/* copy over sample rate table	but zero it first */
-	bzero(reg->reg_compat_srs_list, sizeof (reg->reg_compat_srs_list));
-	bcopy(usb_as_default_srs, reg->reg_compat_srs_list,
-	    sizeof (usb_as_default_srs));
+	bzero(reg->reg_srs, sizeof (reg->reg_srs));
+	bcopy(usb_as_default_srs, reg->reg_srs, sizeof (usb_as_default_srs));
 
 	channels[1] = channels[2] = 0;
 
@@ -2841,8 +2318,8 @@ usb_as_prepare_registration_data(usb_as_state_t   *uasp)
 		if (uasp->usb_as_alts[alt].alt_valid &&
 		    (n < USB_AS_N_FORMATS) &&
 		    (usb_as_valid_format(uasp, alt,
-		    reg->reg_compat_srs_list,
-		    (sizeof (reg->reg_compat_srs_list)/
+		    reg->reg_srs,
+		    (sizeof (reg->reg_srs)/
 		    sizeof (uint_t)) - 1)) == USB_SUCCESS) {
 			reg->reg_formats[n].fmt_termlink =
 			    uasp->usb_as_alts[alt].alt_general->
diff --git a/usr/src/uts/common/io/usb/inc.flg b/usr/src/uts/common/io/usb/inc.flg
deleted file mode 100644
index f130c48245..0000000000
--- a/usr/src/uts/common/io/usb/inc.flg
+++ /dev/null
@@ -1,191 +0,0 @@
-#!/bin/sh
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-
-#
-# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-
-
-# This file brings down all that is needed to build just 
-# USB drivers.
-#
-
-find_files "s.*.x"	\
-	usr/src/uts/common/rpc \
-	usr/src/uts/common/rpcsvc \
-	usr/src/uts/common/gssapi
-
-find_files "s.*.h"	\
-	usr/src/uts/common/vm \
-	usr/src/uts/common/sys \
-	usr/src/uts/common/rpc \
-	usr/src/uts/common/gssapi \
-	usr/src/uts/common/netinet \
-	usr/src/uts/common/net \
-	usr/src/uts/intel/sys \
-	usr/src/uts/sparc/sys \
-	usr/src/uts/sparc/v7/sys \
-	usr/src/uts/sparc/v9/sys \
-	usr/src/uts/sun/sys
-
-echo_file usr/src/uts/common/io/cmlb.c
-
-find_files "s.*"	\
-	usr/src/uts/common/io/warlock \
-	usr/src/uts/common/io/scsi \
-	usr/src/uts/common/io/audio \
-	usr/src/uts/sparc/hid	\
-	usr/src/uts/sparc/hidparser	\
-	usr/src/uts/sparc/usba	\
-	usr/src/uts/sparc/usba10 \
-	usr/src/uts/sparc/uhci	\
-	usr/src/uts/sparc/ohci	\
-	usr/src/uts/sparc/ehci	\
-	usr/src/uts/sparc/hubd	\
-	usr/src/uts/sparc/usb_mid \
-	usr/src/uts/sparc/usb_ia \
-	usr/src/uts/sparc/usbkbm \
-	usr/src/uts/sparc/usbms \
-	usr/src/uts/sparc/scsa2usb \
-	usr/src/uts/sparc/scsi \
-	usr/src/uts/sparc/sd \
-	usr/src/uts/sparc/st \
-	usr/src/uts/sparc/cmlb \
-	usr/src/uts/sparc/usbprn \
-	usr/src/uts/sparc/usb_ac \
-	usr/src/uts/sparc/usb_ac_dacf \
-	usr/src/uts/sparc/usb_as \
-	usr/src/uts/sparc/usb_ah \
-	usr/src/uts/sparc/audiosup \
-	usr/src/uts/sparc/mixer \
-	usr/src/uts/sparc/amsrc2 \
-	usr/src/uts/sparc/ugen \
-	usr/src/uts/sparc/usbftdi \
-	usr/src/uts/sparc/usbser \
-	usr/src/uts/sparc/usbsksp \
-	usr/src/uts/sparc/usbsprl \
-	usr/src/uts/sparc/usbsacm \
-	usr/src/uts/sparc/usbvc \
-	usr/src/uts/sparc/wusb_ca \
-	usr/src/uts/sparc/hwa1480_fw \
-	usr/src/uts/sparc/hwahc \
-	usr/src/uts/sparc/hwarc \
-	usr/src/uts/sparc/wusb_df \
-	usr/src/uts/sparc/uwba \
-	usr/src/uts/sparc/usbskel \
-	usr/src/uts/sparc/warlock \
-	usr/src/uts/sparc/asm \
-	usr/src/uts/intel/hid	\
-	usr/src/uts/intel/hidparser	\
-	usr/src/uts/intel/usba	\
-	usr/src/uts/intel/usba10 \
-	usr/src/uts/intel/uhci	\
-	usr/src/uts/intel/ohci	\
-	usr/src/uts/intel/ehci	\
-	usr/src/uts/intel/hubd	\
-	usr/src/uts/intel/usb_mid \
-	usr/src/uts/intel/usb_ia \
-	usr/src/uts/intel/usbkbm \
-	usr/src/uts/intel/usbms	\
-	usr/src/uts/intel/scsa2usb \
-	usr/src/uts/intel/scsi \
-	usr/src/uts/intel/sd \
-	usr/src/uts/intel/st \
-	usr/src/uts/intel/cmlb \
-	usr/src/uts/intel/usbprn \
-	usr/src/uts/intel/usb_ac \
-	usr/src/uts/intel/usb_ac_dacf \
-	usr/src/uts/intel/usb_as \
-	usr/src/uts/intel/usb_ah \
-	usr/src/uts/intel/audiosup \
-	usr/src/uts/intel/mixer \
-	usr/src/uts/intel/amsrc2 \
-	usr/src/uts/intel/ugen \
-	usr/src/uts/intel/usbftdi \
-	usr/src/uts/intel/usbser \
-	usr/src/uts/intel/usbsksp \
-	usr/src/uts/intel/usbsprl \
-	usr/src/uts/intel/usbsacm \
-	usr/src/uts/intel/usbvc \
-	usr/src/uts/intel/wusb_ca \
-	usr/src/uts/intel/hwa1480_fw \
-	usr/src/uts/intel/hwahc \
-	usr/src/uts/intel/hwarc \
-	usr/src/uts/intel/wusb_df \
-	usr/src/uts/intel/uwba \
-	usr/src/uts/intel/usbskel \
-	usr/src/uts/intel/warlock \
-	usr/src/uts/intel/ia32 \
-	usr/src/uts/intel/amd64 \
-	usr/src/uts/intel/asm 
-
-
-echo_file usr/src/uts/sparc/os/driver_classes
-echo_file usr/src/uts/sparc/os/driver_aliases
-echo_file usr/src/uts/sparc/os/name_to_major
-echo_file usr/src/uts/intel/os/driver_classes
-echo_file usr/src/uts/intel/os/driver_aliases
-echo_file usr/src/uts/intel/os/name_to_major
-
-echo_file  usr/src/Makefile.master
-echo_file  usr/src/Makefile.master.64
-echo_file  usr/src/req.flg
-echo_file  usr/src/Makefile.psm
-echo_file  usr/src/Makefile.psm.targ
-echo_file  usr/src/uts/Makefile
-echo_file  usr/src/uts/Makefile.targ
-echo_file  usr/src/uts/Makefile.uts
-echo_file  usr/src/uts/sun/Makefile.files
-echo_file  usr/src/uts/sun/Makefile.rules
-echo_file  usr/src/uts/common/Makefile.files
-echo_file  usr/src/uts/common/Makefile.rules
-echo_file  usr/src/uts/common/sys/Makefile.syshdrs
-echo_file  usr/src/uts/sparc/Makefile
-echo_file  usr/src/uts/sparc/Makefile.files
-echo_file  usr/src/uts/sparc/Makefile.rules
-echo_file  usr/src/uts/sparc/Makefile.sparc
-echo_file  usr/src/uts/sparc/Makefile.sparc.shared
-echo_file  usr/src/uts/sparc/Makefile.targ
-echo_file  usr/src/uts/sparc/Makefile.targ.shared
-echo_file  usr/src/uts/sparc/req.flg
-echo_file  usr/src/uts/sparc/ml/sparc.il
-echo_file  usr/src/uts/sparc/v9/Makefile.files
-echo_file  usr/src/uts/sparc/v9/Makefile.rules
-echo_file  usr/src/uts/sun4u/Makefile.rules
-echo_file  usr/src/uts/sun4/Makefile.rules
-echo_file  usr/src/uts/intel/Makefile
-echo_file  usr/src/uts/intel/Makefile.files
-echo_file  usr/src/uts/intel/Makefile.rules
-echo_file  usr/src/uts/intel/Makefile.intel
-echo_file  usr/src/uts/intel/Makefile.intel.shared
-echo_file  usr/src/uts/intel/Makefile.targ
-echo_file  usr/src/uts/intel/Makefile.targ.shared
-echo_file  usr/src/uts/sparc/v9/Makefile.files
-echo_file  usr/src/uts/sparc/v9/Makefile.rules
-echo_file  usr/src/uts/sun4u/Makefile.rules
-echo_file  usr/src/uts/sun4/Makefile.rules
-echo_file  usr/src/uts/common/sys/Makefile
-echo_file  usr/src/uts/common/os/privs.awk
-echo_file  usr/src/uts/common/os/priv_defs
-echo_file  usr/src/uts/common/rpc/Makefile
-echo_file  usr/src/uts/common/rpcsvc/Makefile
-echo_file  usr/src/uts/common/gssapi/Makefile
diff --git a/usr/src/uts/common/io/warlock/usb_ac.wlcmd b/usr/src/uts/common/io/warlock/usb_ac.wlcmd
index 37bf53167a..6a68d90059 100644
--- a/usr/src/uts/common/io/warlock/usb_ac.wlcmd
+++ b/usr/src/uts/common/io/warlock/usb_ac.wlcmd
@@ -19,10 +19,9 @@
 # CDDL HEADER END
 #
 
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 one usb_ac_state
 
@@ -46,9 +45,6 @@ root usb_ac_stop_play
 root usb_ac_stop_record
 root usb_ac_teardown
 
-root usb_ac_dacf.c:usb_ac_mux_plumbing
-root usb_ac_dacf.c:usb_ac_mux_unplumbing
-
 add  bus_ops::bus_add_eventcall targets warlock_dummy
 add  bus_ops::bus_get_eventcookie targets warlock_dummy
 add  bus_ops::bus_post_event targets warlock_dummy
diff --git a/usr/src/uts/common/io/warlock/usb_ac_with_sada.wlcmd b/usr/src/uts/common/io/warlock/usb_ac_with_sada.wlcmd
index 67cc818a42..9442430d7f 100644
--- a/usr/src/uts/common/io/warlock/usb_ac_with_sada.wlcmd
+++ b/usr/src/uts/common/io/warlock/usb_ac_with_sada.wlcmd
@@ -19,10 +19,9 @@
 # CDDL HEADER END
 #
 
-# Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 # warlock command file
 
@@ -86,8 +85,6 @@ root	am_single_open_task
 root	am_hw_task
 
 ### specify the usb_ac root functions
-root 	usb_ac_dacf.c:usb_ac_mux_plumbing
-root 	usb_ac_dacf.c:usb_ac_mux_unplumbing
 root 	usb_ac_open
 root 	usb_ac_close
 root 	usb_ac_disconnect_event_cb
diff --git a/usr/src/uts/common/io/warlock/usb_ac_with_usba.wlcmd b/usr/src/uts/common/io/warlock/usb_ac_with_usba.wlcmd
index e5441a38e2..aba4357455 100644
--- a/usr/src/uts/common/io/warlock/usb_ac_with_usba.wlcmd
+++ b/usr/src/uts/common/io/warlock/usb_ac_with_usba.wlcmd
@@ -1,5 +1,5 @@
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # CDDL HEADER START
@@ -21,7 +21,6 @@
 #
 # CDDL HEADER END
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 one ohci_state
 one ehci_state
@@ -147,8 +146,6 @@ root	ohci_intr
 root	ehci_intr
 
 ### specify the usb_ac root functions
-root usb_ac_dacf.c:usb_ac_mux_plumbing
-root usb_ac_dacf.c:usb_ac_mux_unplumbing
 root usb_ac_open
 root usb_ac_close
 root usb_ac_disconnect_event_cb
diff --git a/usr/src/uts/common/sys/Makefile b/usr/src/uts/common/sys/Makefile
index 75a8d27fc0..4f0cb2dd1c 100644
--- a/usr/src/uts/common/sys/Makefile
+++ b/usr/src/uts/common/sys/Makefile
@@ -43,9 +43,6 @@ i386_HDRS=			\
 	agp/agptarget_io.h	\
 	agpgart.h		\
 	asy.h			\
-	audio/audio810.h	\
-	audio/audiohd.h	\
-	audio/audioixp.h	\
 	fd_debug.h		\
 	fdc.h			\
 	fdmedia.h		\
@@ -55,8 +52,6 @@ i386_HDRS=			\
 	ucode.h
 
 sparc_HDRS= 			\
-	audio/audiots.h		\
-	audio/am_src2.h		\
 	mouse.h			\
 	scsi/targets/ssddef.h	\
 	$(MDESCHDRS)
@@ -85,7 +80,6 @@ CHKHDRS=			\
 	attr.h			\
 	audio.h			\
 	audioio.h		\
-	audiovar.h		\
 	autoconf.h		\
 	auxv.h			\
 	auxv_386.h		\
@@ -505,6 +499,7 @@ CHKHDRS=			\
 	socketvar.h		\
 	sockio.h		\
 	sodirect.h		\
+	soundcard.h		\
 	squeue.h		\
 	squeue_impl.h		\
 	srn.h			\
@@ -653,12 +648,9 @@ HDRS=				\
 
 AUDIOHDRS=			\
 	ac97.h			\
-	am_src1.h		\
-	audio_apm.h		\
-	audio_mixer.h		\
-	audio_src.h		\
-	audio_support.h		\
-	audio_trace.h		\
+	audio_common.h		\
+	audio_driver.h		\
+	audio_oss.h		\
 	g711.h
 
 AVHDRS=				\
diff --git a/usr/src/uts/common/sys/audio/ac97.h b/usr/src/uts/common/sys/audio/ac97.h
index 2a2f7afb33..ebeae69d2e 100644
--- a/usr/src/uts/common/sys/audio/ac97.h
+++ b/usr/src/uts/common/sys/audio/ac97.h
@@ -20,20 +20,15 @@
  */
 
 /*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-/*
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
 #ifndef	_SYS_AC97_H
 #define	_SYS_AC97_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
+#include <sys/types.h>
+#include <sys/audio/audio_common.h>
 
 #ifdef	__cplusplus
 extern "C" {
@@ -42,89 +37,12 @@ extern "C" {
 #ifdef _KERNEL
 
 /*
- * This header file describes the AC-97 V2.1 Codec register set. See the
+ * This header file describes the AC-97 Codec register set. See the
  * spec for a detailed description of each register.
  */
 
-struct ac97_v21 {
-	uint16_t	ac97_reset;			/* 00h */
-	uint16_t	ac97_master_volume;		/* 02h */
-	uint16_t	ac97_headphone_volume;		/* 04h, optional */
-	uint16_t	ac97_master_mono_volume;	/* 06h, optional */
-	uint16_t	ac97_master_tone;		/* 08h, optional */
-	uint16_t	ac97_pc_beep_volume;		/* 0ah, optional */
-	uint16_t	ac97_phone_volume;		/* 0ch, optional */
-	uint16_t	ac97_mic_volume;		/* 0eh */
-	uint16_t	ac97_line_in_volume;		/* 10h */
-	uint16_t	ac97_cd_volume;			/* 12h */
-	uint16_t	ac97_video_volume;		/* 14h, optional */
-	uint16_t	ac97_aux_volume;		/* 16h, optional */
-	uint16_t	ac97_PCM_out_volume;		/* 18h */
-	uint16_t	ac97_record_select;		/* 1ah */
-	uint16_t	ac97_record_gain;		/* 1ch */
-	uint16_t	ac97_record_gain_mic;		/* 1eh, optional */
-	uint16_t	ac97_general_purpose;		/* 20h, optional */
-	uint16_t	ac97_threeD_control;		/* 22h, optional */
-	uint16_t	ac97_reserved_1;		/* 24h */
-	uint16_t	ac97_pwrdwn_ctrl_stat;		/* 26h */
-
-	/* extended audio registers */
-
-	uint16_t	ac97_ext_audio_id;		/* 28h, optional */
-	uint16_t	ac97_ext_audio_stat_ctrl;	/* 2ah, optional */
-	uint16_t	ac97_ext_front_DAC_rate;	/* 2ch, optional */
-	uint16_t	ac97_ext_surround_DAC_rate;	/* 2eh, optional */
-	uint16_t	ac97_ext_LFE_DAC_rate;		/* 30h, optional */
-	uint16_t	ac97_ext_LR_ADC_rate;		/* 32h, optional */
-	uint16_t	ac97_ext_mic_ADC_rate;		/* 34h, optional */
-	uint16_t	ac97_ext_C_LFE_volume;		/* 36h, optional */
-	uint16_t	ac97_ext_LR_surround_volume;	/* 38h, optional */
-	uint16_t	ac97_ext_reserved_1;		/* 3ah, optional */
-
-	/* extended modem registers */
-
-	uint16_t	ac97_ext_modem_id;		/* 3ch, optional */
-	uint16_t	ac97_ext_modem_stat_ctrl;	/* 3eh, optional */
-	uint16_t	ac97_ext_line1_DAC_ADC_rate;	/* 40h, optional */
-	uint16_t	ac97_ext_line2_DAC_ADC_rate;	/* 42h, optional */
-	uint16_t	ac97_ext_hndst_DAC_ADC_rate;	/* 44h, optional */
-	uint16_t	ac97_ext_line1_DAC_ADC_level;	/* 46h, optional */
-	uint16_t	ac97_ext_line2_DAC_ADC_level;	/* 48h, optional */
-	uint16_t	ac97_ext_hndst_DAC_ADC_level;	/* 4ah, optional */
-	uint16_t	ac97_ext_GPIO_pin_config;	/* 4ch, optional */
-	uint16_t	ac97_ext_GPIO_pin_polarity;	/* 4eh, optional */
-	uint16_t	ac97_ext_GPIO_pin_sticky;	/* 50h, optional */
-	uint16_t	ac97_ext_GPIO_pin_wakeup;	/* 52h, optional */
-	uint16_t	ac97_ext_GPIO_pin_status;	/* 54h, optional */
-	uint16_t	ac97_ext_modem_AFE_stat_ctrl;	/* 56h, optional */
-	uint16_t	ac97_ext_reserved_2;		/* 58h, optional */
-
-	/* reserved for vendor usage */
-	uint16_t	ac97_vendor_01;			/* 5ah, optional */
-	uint16_t	ac97_vendor_02;			/* 5ch, optional */
-	uint16_t	ac97_vendor_03;			/* 5eh, optional */
-	uint16_t	ac97_vendor_04;			/* 60h, optional */
-	uint16_t	ac97_vendor_05;			/* 62h, optional */
-	uint16_t	ac97_vendor_06;			/* 64h, optional */
-	uint16_t	ac97_vendor_07;			/* 66h, optional */
-	uint16_t	ac97_vendor_08;			/* 68h, optional */
-	uint16_t	ac97_vendor_09;			/* 6ah, optional */
-	uint16_t	ac97_vendor_10;			/* 6ch, optional */
-	uint16_t	ac97_vendor_11;			/* 6eh, optional */
-	uint16_t	ac97_vendor_12;			/* 70h, optional */
-	uint16_t	ac97_vendor_13;			/* 72h, optional */
-	uint16_t	ac97_vendor_14;			/* 74h, optional */
-	uint16_t	ac97_vendor_15;			/* 76h, optional */
-	uint16_t	ac97_vendor_16;			/* 78h, optional */
-	uint16_t	ac97_vendor_17;			/* 7ah, optional */
-
-	uint16_t	ac97_vendor_id1;		/* 7ch */
-	uint16_t	ac97_vendor_id2;		/* 7eh */
-};
-typedef struct ac97_v21 ac97_v21_t;
-
 /*
- * Defines for the V2.1 registers.
+ * Defines for the registers.
  */
 
 /* Reset Register					Index 00h */
@@ -162,7 +80,7 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	HPVR_MUTE					0x8000
 
 /* Mono Master Volume Register				Index 06h - Optional */
-#define	AC97_MONO_MASTER_VOLUME_REGSITER		0x06
+#define	AC97_MONO_MASTER_VOLUME_REGISTER		0x06
 #define	MMVR_MASK					0x001f
 #define	MMVR_0dB_ATTEN					0x0000
 #define	MMVR_OPTIONAL_MASK				0x003f
@@ -239,7 +157,7 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	AUXVR_LEFT_MAX_ATTEN				0x1f00
 #define	AUXVR_MUTE					0x8000
 
-/* PCM Out Input Volume Register			Index 18h */
+/* PCM Out Volume Register				Index 18h */
 #define	AC97_PCM_OUT_VOLUME_REGISTER			0x18
 #define	PCMOVR_RIGHT_GAIN_MASK				0x001f
 #define	PCMOVR_RIGHT_0dB_GAIN				0x0010
@@ -306,6 +224,14 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	TDCR_CENTER_MASK				0x0f00
 #define	TDCR_NULL					0x0000
 
+/* Audio Interrupt and Paging Mechanism			Index 24h - r2.3 */
+#define	AC97_INTERRUPT_PAGING_REGISTER			0x24
+#define	IPR_IS						0x8000
+#define	IPR_CAUSE_MASK					0x6000
+#define	IPR_SC						0x1000
+#define	IPR_IE						0x0800
+#define	IPR_PG_MASK					0x000f
+
 /* Powerdown Control Status Register			Index 26h */
 #define	AC97_POWERDOWN_CTRL_STAT_REGISTER		0x26
 #define	PCSR_ADC					0x0001
@@ -327,11 +253,17 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	AC97_EXTENDED_AUDIO_REGISTER			0x28
 #define	EAR_VRA						0x0001
 #define	EAR_DRA						0x0002
+#define	EAR_SPDIF					0x0004
 #define	EAR_VRM						0x0008
+#define	EAR_DSA_MASK					0x0030
 #define	EAR_CDAC					0x0040
 #define	EAR_SDAC					0x0080
 #define	EAR_LDAC					0x0100
 #define	EAR_AMAP					0x0200
+#define	EAR_REV_MASK					0x0c00
+#define	EAR_REV_21					0x0000
+#define	EAR_REV_22					0x0400
+#define	EAR_REV_23					0x0800
 #define	EAR_PRIMARY_CODEC				0x0000
 #define	EAR_SECONDARY_01_CODEC				0x4000
 #define	EAR_SECONDARY_10_CODEC				0x8000
@@ -341,15 +273,23 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER		0x2a
 #define	EASCR_VRA					0x0001
 #define	EASCR_DRA					0x0002
+#define	EASCR_SPDIF					0x0004
 #define	EASCR_VRM					0x0008
+#define	EASCR_SPSA_MASK					0x0030
+#define	EASCR_SPSA_3_4					0x0000
+#define	EASCR_SPSA_7_8					0x0010
+#define	EASCR_SPSA_6_9					0x0020
+#define	EASCR_SPSA_10_11				0x0030
 #define	EASCR_CDAC					0x0040
 #define	EASCR_SDAC					0x0080
 #define	EASCR_LDAC					0x0100
 #define	EASCR_MADC					0x0200
+#define	EASCR_SPCV					0x0400
 #define	EASCR_PRI					0x0800
 #define	EASCR_PRJ					0x1000
 #define	EASCR_PRK					0x2000
 #define	EASCR_PRL					0x4000
+#define	EASCR_VCFG					0x8000
 
 /* Extended Front DAC Rate Register			2ch - Optional */
 #define	AC97_EXTENDED_FRONT_DAC_RATE_REGISTER		0x2c
@@ -371,7 +311,7 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	AC97_EXTENDED_C_LFE_VOLUME_REGISTER		0x36
 #define	EXLFEVR_CENTER_MASK				0x001f
 #define	EXLFEVR_CENTER_OPTIONAL_MASK			0x003f
-#define	EXLFEVR_CENTER_MTUE				0x0080
+#define	EXLFEVR_CENTER_MUTE				0x0080
 #define	EXLFEVR_LFE_MASK				0x1f00
 #define	EXLFEVR_LFE_OPTIONAL_MASK			0x3f00
 #define	EXLFEVR_LFE_MUTE				0x8000
@@ -385,6 +325,25 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	EXLFEVR_LEFT_OPTIONAL_MASK			0x3f00
 #define	EXLFEVR_LEFT_MUTE				0x8000
 
+/* S/PDIF Control Register				3ah - Optional */
+#define	AC97_SPDIF_CONTROL_REGISTER			0x3a
+#define	SPCR_PRO					0x0001
+#define	SPCR_AUDIO					0x0002
+#define	SPCR_COPY					0x0004
+#define	SPCR_PRE					0x0008
+#define	SPCR_CC_MASK					0x07f0
+#define	SPCR_L						0x0800
+#define	SPCR_SPSR_MASK					0x3000
+#define	SPCR_SPSR_44100					0x0000
+#define	SPCR_SPSR_48000					0x2000
+#define	SPCR_SPSR_32000					0x3000
+#define	SPCR_DRS					0x4000
+#define	SPCR_V						0x8000
+
+/*
+ * Modem only registers from 3ch - 58h.
+ */
+
 /* Extended Modem ID Register				3ch - Optional */
 #define	AC97_EXTENDED_MODEM_ID_REGISTER			0x3c
 #define	EMIDR_LINE1					0x0001
@@ -478,7 +437,11 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	EMAFESCR_CID1					0x4000
 #define	EMAFESCR_CID2					0x8000
 
-/* Vender Reserved Registers				5ah - 7ah - Optional */
+/* Vendor Reserved Registers				5ah - 7ah - Optional */
+/*
+ * Note that 60h - 6eh is also defined as the extended codec page area in
+ * AC'97 r2.3.
+ */
 #define	AC97_VENDOR_REGISTER_01				0x5a
 #define	AC97_VENDOR_REGISTER_02				0x5c
 #define	AC97_VENDOR_REGISTER_03				0x5e
@@ -497,6 +460,18 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	AC97_VENDOR_REGISTER_16				0x78
 #define	AC97_VENDOR_REGISTER_17				0x7a
 
+/*
+ * Page 01 Extended Codec Registers
+ */
+#define	AC97_PAGE01_CODEC_CLASS_REV_REGISTER		0x60
+#define	AC97_PAGE01_PCI_SVID_REGISTER			0x62
+#define	AC97_PAGE01_PCI_SID_REGISTER			0x64
+#define	AC97_PAGE01_FUNCTION_SELECT_REGISTER		0x66
+#define	AC97_PAGE01_FUNCTION_INFORMATION_REGISTER	0x68
+#define	AC97_PAGE01_SENSE_DETAILS_REGISTER		0x6a
+#define	AC97_PAGE01_DAC_SLOT_MAPPING_REGISTER		0x6c
+#define	AC97_PAGE01_ADC_SLOT_MAPPING_REGISTER		0x6e
+
 /* Vendor ID1 Register					7ch */
 #define	AC97_VENDOR_ID1_REGISTER			0x7c
 #define	VID1R_CHAR2_MASK				0x00ff
@@ -507,6 +482,133 @@ typedef struct ac97_v21 ac97_v21_t;
 #define	VID2R_REVISION_MASK				0x00ff
 #define	VID2R_CHAR3_MASK				0xff00
 
+/*
+ * Property names used by AC97.  We should probably have a better way
+ * of dealing with some of these.  (LINEIN_FUNC and MIC_FUNC should really
+ * be saved/restored with other global settings.)
+ */
+#define	AC97_PROP_AMPLIFIER	"ac97-amplifier"
+#define	AC97_PROP_SPEAKER	"ac97-speaker"
+#define	AC97_PROP_MICBOOST	"ac97-micboost"
+#define	AC97_PROP_NO_HEADPHONE	"ac97-no-headphone"
+#define	AC97_PROP_NO_AUXOUT	"ac97-no-auxout"
+#define	AC97_PROP_NO_CDROM	"ac97-no-cdrom"
+#define	AC97_PROP_NO_VIDEO	"ac97-no-video"
+#define	AC97_PROP_NO_AUXIN	"ac97-no-auxin"
+#define	AC97_PROP_NO_MIC	"ac97-no-mic"
+#define	AC97_PROP_NO_LINEIN	"ac97-no-linein"
+#define	AC97_PROP_LINEIN_FUNC	"ac97-linein-function"	/* 1=linein, 2=surr */
+#define	AC97_PROP_MIC_FUNC	"ac97-mic-function"	/* 1=mic, 2=cen/lfe */
+#define	AC97_PROP_DOWNMIX	"ac97-downmix"
+#define	AC97_PROP_SPREAD	"ac97-spread"
+
+/*
+ * Known Codec vendors.
+ */
+#define	AC97_VENDOR_ADS			0x41445300	/* Analog Devices */
+#define	AC97_VENDOR_AKM			0x414b4d00	/* Asahi Kasei */
+#define	AC97_VENDOR_ALC			0x414c4300	/* Realtek */
+#define	AC97_VENDOR_ALG			0x414c4700	/* Realtek */
+#define	AC97_VENDOR_CMI			0x43d44900	/* Cmedia */
+#define	AC97_VENDOR_CRY			0x43525900	/* Cirrus Logic */
+#define	AC97_VENDOR_CXT			0x43585400	/* Conexant */
+#define	AC97_VENDOR_EMC			0x454d4300	/* eMicro */
+#define	AC97_VENDOR_EV			0x000f8300	/* Ectiva */
+#define	AC97_VENDOR_ESS			0x45838300	/* ESS */
+#define	AC97_VENDOR_ICE			0x49434500	/* ICEnsemble */
+#define	AC97_VENDOR_ST			0x83847600	/* SigmaTel */
+#define	AC97_VENDOR_TRA			0x54524100	/* TriTech */
+#define	AC97_VENDOR_VIA			0x56494100	/* VIA */
+#define	AC97_VENDOR_WML			0x574d4c00	/* Wolfson */
+#define	AC97_VENDOR_YMH			0x594d4800	/* Yamaha */
+
+/*
+ * Known Codec IDs.
+ */
+#define	AC97_CODEC_AD1819B		0x41445303
+#define	AC97_CODEC_AD1881		0x41445340
+#define	AC97_CODEC_AD1881A		0x41445348
+#define	AC97_CODEC_AD1885		0x41445360
+#define	AC97_CODEC_AD1886		0x41445361
+#define	AC97_CODEC_AD1887		0x41445362
+#define	AC97_CODEC_AD1888		0x41445368
+#define	AC97_CODEC_AD1980		0x41445370
+#define	AC97_CODEC_AD1981A		0x41445371
+#define	AC97_CODEC_AD1981		0x41445372
+#define	AC97_CODEC_AD1981B		0x41445374
+#define	AC97_CODEC_AD1985		0x41445375
+#define	AC97_CODEC_AK4540		0x414b4d00
+#define	AC97_CODEC_ALC100		0x414c4326
+#define	AC97_CODEC_ALC200P		0x414c4710
+#define	AC97_CODEC_ALC202		0x414c4740
+#define	AC97_CODEC_ALC203		0x414c4770
+#define	AC97_CODEC_ALC250		0x414c4750
+#define	AC97_CODEC_ALC250_2		0x414c4752
+#define	AC97_CODEC_ALC650		0x414c4720
+#define	AC97_CODEC_ALC655		0x414c4760
+#define	AC97_CODEC_ALC658		0x414c4780
+#define	AC97_CODEC_ALC850		0x414c4790
+#define	AC97_CODEC_CMI9738		0x434d4941
+#define	AC97_CODEC_CMI9739		0x434d4961
+#define	AC97_CODEC_CMI9780		0x434d4969
+#define	AC97_CODEC_CMI9761		0x434d4978
+#define	AC97_CODEC_CMI9761_2		0x434d4982
+#define	AC97_CODEC_CMI9761_3		0x434d4983
+#define	AC97_CODEC_CS4202		0x43525970
+#define	AC97_CODEC_CS4205		0x43525950
+#define	AC97_CODEC_CS4294		0x43525920
+#define	AC97_CODEC_CS4297		0x43525900
+#define	AC97_CODEC_CS4297A		0x43525910
+#define	AC97_CODEC_CS4299		0x43525930
+#define	AC97_CODEC_CX20468		0x43585428
+#define	AC97_CODEC_CX20468_2		0x43585429
+#define	AC97_CODEC_CX20468_21		0x43585430
+#define	AC97_CODEC_EM28028		0x454d4328
+#define	AC97_CODEC_ES1921		0x45838308
+#define	AC97_CODEC_EV1938		0x000f8384
+#define	AC97_CODEC_ICE1232		0x49434511
+#define	AC97_CODEC_STAC9700		0x83847600
+#define	AC97_CODEC_STAC9701		0x83847601
+#define	AC97_CODEC_STAC9701_2		0xc250c250
+#define	AC97_CODEC_STAC9704		0x83847604
+#define	AC97_CODEC_STAC9705		0x83847605
+#define	AC97_CODEC_STAC9708		0x83847608
+#define	AC97_CODEC_STAC9721		0x83847609
+#define	AC97_CODEC_STAC9744		0x83847644
+#define	AC97_CODEC_STAC9750		0x83847650
+#define	AC97_CODEC_STAC9752		0x83847652
+#define	AC97_CODEC_STAC9756		0x83847656
+#define	AC97_CODEC_STAC9758		0x83847658
+#define	AC97_CODEC_STAC9766		0x83847666
+#define	AC97_CODEC_TR28023		0x54524103
+#define	AC97_CODEC_TR28023_2		0x54524123
+#define	AC97_CODEC_TR28028		0x54524108
+#define	AC97_CODEC_TR28028_2		0x54524128
+#define	AC97_CODEC_VT1612A		0x56494161
+#define	AC97_CODEC_VT1617A		0x56494170
+#define	AC97_CODEC_VT1616		0x49434551
+#define	AC97_CODEC_VT1616A		0x49434552
+#define	AC97_CODEC_VT1618		0x56494182
+#define	AC97_CODEC_WM9701A		0x574d4c00
+#define	AC97_CODEC_WM9703		0x574d4c03
+#define	AC97_CODEC_WM9704		0x574d4c04
+#define	AC97_CODEC_YMF743		0x594d4800
+#define	AC97_CODEC_YMF753		0x594d4803
+
+/*
+ * Functions for drivers to interact with the common ac97 module.
+ */
+typedef struct ac97 ac97_t;
+typedef void (*ac97_wr_t)(void *, uint8_t, uint16_t);
+typedef uint16_t (*ac97_rd_t)(void *, uint8_t);
+
+ac97_t *ac97_alloc(dev_info_t *, ac97_rd_t, ac97_wr_t, void *);
+void ac97_free(ac97_t *);
+int ac97_init(ac97_t *, audio_dev_t *);
+void ac97_suspend(ac97_t *);
+void ac97_resume(ac97_t *);
+void ac97_reset(ac97_t *);
+
 #endif	/* _KERNEL */
 
 #ifdef	__cplusplus
diff --git a/usr/src/uts/common/sys/audio/am_src1.h b/usr/src/uts/common/sys/audio/am_src1.h
deleted file mode 100644
index 9e5fcd41f9..0000000000
--- a/usr/src/uts/common/sys/audio/am_src1.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright (c) 2000-2001 by Sun Microsystems, Inc.
- * All rights reserved.
- *
- * This header file defines the public interfaces for the default
- * audio mixer sample rate converter.
- *
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
-#ifndef	_SYS_AM_SRC1_H
-#define	_SYS_AM_SRC1_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-
-#define	AM_SRC1_MOD_NAME	"Audio Sample Rate Conv. #1"
-
-#ifdef _KERNEL
-
-/*
- * am_ad_src1_info_t	- mixer mode Audio Driver sample rate conversion #1 info
- */
-struct am_ad_src1_info {
-	uint_t		ad_from_sr;	/* going from this sample rate */
-	uint_t		ad_to_sr;	/* to this sample rate */
-	uint_t		ad_nconv;	/* number of conversions */
-	uint_t		ad_u0;		/* up conversion parameter #1 */
-	uint_t		ad_u1;		/* up conversion parameter #2 */
-	uint_t		ad_u2;		/* up conversion parameter #3 */
-	uint_t		ad_u3;		/* up conversion parameter #4 */
-	uint_t		ad_d0;		/* conversion parameter #1 */
-	uint_t		ad_d1;		/* conversion parameter #2 */
-	uint_t		ad_d2;		/* conversion parameter #3 */
-	uint_t		ad_d3;		/* conversion parameter #4 */
-	uint_t		ad_k;		/* filter shift */
-};
-typedef struct am_ad_src1_info am_ad_src1_info_t;
-
-#define	AM_SRC1_FILTER		0x40000000	/* filter if ORed with ad_p?? */
-
-extern am_ad_src_entry_t am_src1;	/* sample rate conversion routine #1 */
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AM_SRC1_H */
diff --git a/usr/src/uts/common/sys/audio/am_src2.h b/usr/src/uts/common/sys/audio/am_src2.h
deleted file mode 100644
index 7c11d904c9..0000000000
--- a/usr/src/uts/common/sys/audio/am_src2.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the public interfaces for the alternative
- * audio mixer sample rate converter.
- *
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
-#ifndef	_SYS_AM_SRC2_H
-#define	_SYS_AM_SRC2_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#define	AM_SRC2_MOD_NAME	"Audio Sample Rate Conv. #2"
-
-#ifdef _KERNEL
-
-extern am_ad_src_entry_t am_src2;	/* sample rate conversion routine #2 */
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AM_SRC2_H */
diff --git a/usr/src/uts/common/sys/audio/audio810.h b/usr/src/uts/common/sys/audio/audio810.h
deleted file mode 100644
index 9af6f9c382..0000000000
--- a/usr/src/uts/common/sys/audio/audio810.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIO810_H_
-#define	_SYS_AUDIO810_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/*
- * Header file for the audio810 device driver
- */
-
-/*
- * Values returned by the AUDIO_GETDEV ioctl()
- */
-#define	I810_DEV_NAME			"SUNW,audio810"
-#define	I810_DEV_CONFIG			"onboard1"
-#define	I810_DEV_VERSION		"a"
-
-/*
- * Driver supported configuration information
- */
-#define	I810_NAME			"audio810"
-#define	I810_MOD_NAME			"mixer audio driver"
-
-#define	I810_SAMPR5510			(5510)
-#define	I810_SAMPR6620			(6620)
-#define	I810_SAMPR8000			(8000)
-#define	I810_SAMPR9600			(9600)
-#define	I810_SAMPR11025			(11025)
-#define	I810_SAMPR16000			(16000)
-#define	I810_SAMPR18900			(18900)
-#define	I810_SAMPR22050			(22050)
-#define	I810_SAMPR27420			(27420)
-#define	I810_SAMPR32000			(32000)
-#define	I810_SAMPR33075			(33075)
-#define	I810_SAMPR37800			(37800)
-#define	I810_SAMPR44100			(44100)
-#define	I810_SAMPR48000			(48000)
-
-#define	I810_DEFAULT_SR			I810_SAMPR8000
-#define	I810_DEFAULT_CH			AUDIO_CHANNELS_MONO
-#define	I810_DEFAULT_PREC		AUDIO_PRECISION_8
-#define	I810_DEFAULT_ENC		AUDIO_ENCODING_ULAW
-#define	I810_DEFAULT_PGAIN		(AUDIO_MAX_GAIN * 3 / 4)
-#define	I810_DEFAULT_RGAIN		(127)
-#define	I810_DEFAULT_MONITOR_GAIN	(0)
-#define	I810_DEFAULT_BAL		AUDIO_MID_BALANCE
-#define	I810_INTS			(175)	/* default interrupt rate */
-#define	I810_MIN_INTS			(24)	/* minimum interrupt rate */
-#define	I810_MAX_INTS			(5000)	/* maximum interrupt rate */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif /* _SYS_AUDIO810_H_ */
diff --git a/usr/src/uts/common/sys/audio/audio_apm.h b/usr/src/uts/common/sys/audio/audio_apm.h
deleted file mode 100644
index bf79a12b40..0000000000
--- a/usr/src/uts/common/sys/audio/audio_apm.h
+++ /dev/null
@@ -1,331 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 1999-2003 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the audio support module's public interfaces
- * that may be used by audio personality modules. Audio drivers must NOT
- * include this header file.
- *
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
-#ifndef	_SYS_AUDIO_APM_H
-#define	_SYS_AUDIO_APM_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#ifdef _KERNEL
-
-/*
- * Miscellaneous defines.
- */
-#define	AUDIO_NUM_DEVS		10	/* /dev/audio, /dev/audioctl, etc. */
-#define	AUDIO_MINOR_PER_INST	128	/* max # of channels to each instance */
-#define	AUDIO_MIN_CLONE_CHS	1	/* minimum number of clone channels */
-#define	AUDIO_CLONE_CHANLIM	(AUDIO_MINOR_PER_INST - AUDIO_NUM_DEVS)
-					/* max # of clones to each instance */
-
-#define	AUDIO_MINOR_TO_INST(d)	(getminor(d) >> 7)
-
-#define	AUDIO_TASKQ_SUSPENDED	0
-#define	AUDIO_TASKQ_RUNNING	(~AUDIO_TASKQ_SUSPENDED)
-
-/*
- * audio_data_t		- struct used to store original and processed audio data
- */
-struct audio_data {
-	void			*adata_orig;	/* the original data */
-	void			*adata_optr;	/* marker into the orig. data */
-	void			*adata_oeptr;	/* end of the original data */
-	size_t			adata_osize;	/* size of the original data */
-	void			*adata_proc;	/* the processed data */
-	void			*adata_pptr;	/* marker into the proc. data */
-	void			*adata_peptr;	/* ptr to end of proc. data */
-	size_t			adata_psize;	/* size of the processed data */
-	struct audio_data	*adata_next;	/* pointer to the next struct */
-};
-typedef struct audio_data audio_data_t;
-
-/*
- * The members of the audio_data structure, except adata_next, are protected
- * by scheme because audio_sup_get_audio_data() removes one of these structures
- * from the list and only that thread has access to the data structure.
- */
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_orig))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_optr))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_oeptr))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_osize))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_proc))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_pptr))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_peptr))
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_data::adata_psize))
-
-/*
- * audio_ch_t		- per channel state and operation data
- */
-struct audio_ch {
-	queue_t			*ch_qptr;	/* channel queue pointer */
-	struct audio_state	*ch_statep;	/* channel instance state ptr */
-	kmutex_t		ch_lock;	/* channel lock */
-	kcondvar_t		ch_cv;		/* available for use by ch */
-	struct audio_apm_info	*ch_apm_infop;	/* pointer to ch APM info */
-	int			(*ch_wput)(queue_t *, mblk_t *);
-						/* APM's write put rtn */
-	int			(*ch_wsvc)(queue_t *);
-						/* APM's write svc rtn */
-	int			(*ch_rput)(queue_t *, mblk_t *);
-						/* APM's read put routine */
-	int			(*ch_rsvc)(queue_t *);
-						/* APM's read svc routine */
-	int			ch_dir;		/* I/O direction */
-	uint_t			ch_flags;	/* channel state flags */
-	dev_t			ch_dev;		/* channel device number */
-	void			*ch_private;	/* channel private data */
-	audio_channel_t		ch_info;	/* channel state info */
-	audio_device_t		*ch_dev_info;	/* Audio Driver device info */
-	kmutex_t		ch_adata_lock;	/* audio data list lock */
-	audio_data_t		*ch_adata;	/* audio data structure list */
-	audio_data_t		*ch_adata_end;	/* end of audio data list */
-	int			ch_adata_cnt;	/* # of queued data structs */
-};
-typedef struct audio_ch audio_ch_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_lock, audio_ch))
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_adata_lock, audio_ch::ch_adata))
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_adata_lock, audio_ch::ch_adata_cnt))
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_adata_lock, audio_ch::ch_adata_end))
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_adata_lock, audio_data::adata_next))
-
-/*
- * Further analysis is needed for these structure members. We are
- * deferring this analysis until later.
- */
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_qptr))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_statep))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_apm_infop))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_wput))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_wsvc))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_rput))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_rsvc))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_dir))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_private))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_info))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_dev_info))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_ch::ch_dev))
-
-/* audio_ch.ch_flags defines */
-#define	AUDIO_CHNL_ALLOCATED	0x0001u		/* the channel is allocated */
-#define	AUDIO_CHNL_ACTIVE	0x0002u		/* the channel is active */
-
-/*
- * audio_state_t	- per instance state and operation data
- */
-struct audio_state {
-	kmutex_t		as_lock;	/* instance state lock */
-	kcondvar_t		as_cv;		/* cv for blocked ch alloc */
-	int			as_max_chs;	/* max # of open channels */
-	int			as_minors_per_inst; /* #minors per instance */
-	int			as_audio_reserved; /* #audio devices */
-	dev_info_t		*as_dip;	/* known at attach time */
-	int			as_dev_instance; /* Audio Driver dev inst. # */
-	major_t			as_major;	/* Audio Driver major number */
-	uint_t			as_ch_inuse;	/* # of channels in use */
-	struct audio_apm_info	*as_apm_info_list;	/* APM info list */
-	void			*as_private;	/* private audio driver data */
-	void			*as_persistp;	/* persistent data */
-	audio_ch_t		as_channels[AUDIO_CLONE_CHANLIM]; /* channels */
-};
-typedef struct audio_state audio_state_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_state::as_lock, audio_state))
-
-/* these audio_state structure members are read only once set */
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_max_chs))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_dip))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_dev_instance))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_max_chs))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_audio_reserved))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_state::as_minors_per_inst))
-
-/*
- * Further analysis is needed for this structure member. We are
- * deferring this analysis until later.
- */
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_state::as_private))
-
-/*
- * audio_apm_info_t	- audio personality module state information
- */
-struct audio_apm_info {
-	kmutex_t		apm_lock;	/* APM structure state lock */
-	int			(*apm_open)(queue_t *, dev_t *,
-						int, int, cred_t *);
-						/* APM open() routine */
-	int			(*apm_close)(queue_t *, int, cred_t *);
-						/* APM close() routine */
-	int			(*apm_restore_state)(audio_state_t *,
-				    struct audio_apm_info *, int);
-						/* APM state retsore routine */
-	int			(*apm_save_state)(audio_state_t *,
-				    struct audio_apm_info *, int);
-						/* APM state save routine */
-	audio_device_t		*apm_info;	/* audio_device_t structure */
-	audio_device_type_e	apm_type;	/* the device type */
-	void			*apm_private;	/* private APM data */
-	void			*apm_ad_infop;	/* device capabilities */
-	void			*apm_ad_state;	/* state of the device */
-	struct audio_apm_info	*apm_next;	/* pointer to the next struct */
-};
-typedef struct audio_apm_info audio_apm_info_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_apm_info::apm_lock, audio_apm_info))
-/* these audio_apm_info structure members are read only once set */
-_NOTE(DATA_READABLE_WITHOUT_LOCK(
-    audio_apm_info::apm_private		audio_apm_info::apm_type
-    audio_apm_info::apm_close		audio_apm_info::apm_open
-    audio_apm_info::apm_restore_state	audio_apm_info::apm_save_state
-    audio_apm_info::apm_info		audio_apm_info::apm_ad_infop
-    audio_apm_info::apm_ad_state))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(audio_apm_info::apm_next))
-_NOTE(MUTEX_PROTECTS_DATA(audio_state::as_lock, audio_apm_info::apm_next))
-
-/*
- * audio_apm_reg	- This structure holds all of the data needed to
- *			  register an Audio Personality Module for use.
- */
-struct audio_apm_reg {
-	int		aar_version;		/* structure version */
-	int		(*aar_apm_open)
-			    (queue_t *q, dev_t *devp, int flag, int sflags,
-			    cred_t *credp);
-						/* APM open() routine */
-	int		(*aar_apm_close)(queue_t *q, int flag, cred_t *credp);
-						/* APM close() routine */
-	int		(*aar_apm_save_state)(audio_state_t *statep,
-			    audio_apm_info_t *, int);
-						/* APM save routine */
-	int		(*aar_apm_restore_state)(audio_state_t *statep,
-			    audio_apm_info_t *, int dir);
-						/* APM restore routine */
-	void		*aar_private;		/* APM private data */
-	void		*aar_info;		/* APM info structure */
-	void		*aar_state;		/* APM state structure */
-	audio_device_t	*aar_dev_info;		/* APM device info pointer */
-};
-typedef struct audio_apm_reg audio_apm_reg_t;
-#define	AM_AAR_VERSION		AM_AAR_VERS1
-#define	AM_AAR_VERS1		1		/* supported register version */
-
-/*
- * Macros used to convert between audio handles and the state structure.
- */
-#define	AUDIO_HDL2STATE(hdl)		((audio_state_t *)(hdl))
-#define	AUDIO_STATE2HDL(statep)		((audiohdl_t)(statep))
-
-/*
- * Audio Support Module Channel Routines
- */
-audio_ch_t *audio_sup_alloc_ch(audio_state_t *statep, int *error,
-    audio_device_type_e type, int oflag);
-int audio_sup_free_ch(audio_ch_t *chptr);
-
-/*
- * Audio Support Module State Routines
- */
-audio_state_t *audio_sup_devt_to_state(dev_t dev);
-audio_state_t *audio_sup_devinfo_to_state(dev_info_t *dip);
-
-/*
- * Audio Support Module Persistent Memory Routines
- */
-void *audio_sup_get_persist_state(audio_state_t *state,
-    audio_device_type_e dev_type);
-int audio_sup_free_persist_state(audio_state_t *state,
-    audio_device_type_e dev_type);
-int audio_sup_set_persist_state(audio_state_t *state,
-    audio_device_type_e dev_type, void *state_data, size_t state_size);
-
-/*
- * Audio Support Module Minor Routines
- */
-int audio_sup_ch_to_minor(audio_state_t *statep, int channel);
-int audio_sup_minor_to_ch(audio_state_t *statep, minor_t minor);
-int audio_sup_type_to_minor(audio_device_type_e type);
-
-/*
- * Audio Support Module Audio Data Routines
- */
-void audio_sup_flush_audio_data(audio_ch_t *chptr);
-void audio_sup_free_audio_data(audio_data_t *adata);
-audio_data_t *audio_sup_get_audio_data(audio_ch_t *chptr);
-int audio_sup_get_audio_data_cnt(audio_ch_t *chptr);
-int audio_sup_get_audio_data_size(audio_ch_t *chptr);
-void audio_sup_putback_audio_data(audio_ch_t *chptr, audio_data_t *adata);
-int audio_sup_save_audio_data(audio_ch_t *chptr, void *adata_orig,
-    size_t adata_osize, void *adata_proc, size_t adata_psize);
-
-/*
- * Audio Support Module Registration Routines
- */
-audio_apm_info_t *audio_sup_register_apm(audio_state_t *statep,
-    audio_device_type_e type, audio_apm_reg_t *reg_info);
-int audio_sup_unregister_apm(audio_state_t *statep, audio_device_type_e type);
-
-/*
- * Audio Support Module Task Queue Routines
- */
-audio_taskq_t audio_sup_taskq_create(const char *q_name);
-void audio_sup_taskq_destroy(audio_taskq_t tq_handle);
-int audio_sup_taskq_dispatch(audio_taskq_t tq_handle,
-    void (*task_function)(void *arg), void *arg, int sleep);
-void audio_sup_taskq_resume(audio_taskq_t tq_handle);
-void audio_sup_taskq_suspend(audio_taskq_t tq_handle);
-int audio_sup_taskq_suspended(audio_taskq_t tq_handle);
-void audio_sup_taskq_wait(audio_taskq_t tq_handle);
-
-/*
- * Audio Support Module Miscellaneous Routines
- */
-audio_device_type_e audio_sup_devt_to_ch_type(audio_state_t *statep,
-    dev_t dev);
-int audio_sup_get_channel_number(queue_t *q);
-audio_apm_info_t *audio_sup_get_apm_info(audio_state_t *statep,
-    audio_device_type_e);
-void *audio_sup_get_info(queue_t *q);
-int audio_sup_mblk_alloc(mblk_t *mp, size_t size);
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_APM_H */
diff --git a/usr/src/uts/common/sys/audio/audio_common.h b/usr/src/uts/common/sys/audio/audio_common.h
new file mode 100644
index 0000000000..296880b6f7
--- /dev/null
+++ b/usr/src/uts/common/sys/audio/audio_common.h
@@ -0,0 +1,304 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_SYS_AUDIO_AUDIO_COMMON_H
+#define	_SYS_AUDIO_AUDIO_COMMON_H
+
+#include <sys/mkdev.h>
+#include <sys/types.h>
+
+#ifdef	__cplusplus
+extern "C" {
+#endif
+
+#ifdef	_KERNEL
+
+/* Shared data structures */
+typedef struct audio_parms audio_parms_t;
+typedef struct audio_buffer audio_buffer_t;
+typedef struct audio_stream audio_stream_t;
+typedef struct audio_engine audio_engine_t;
+typedef struct audio_client audio_client_t;
+typedef struct audio_dev audio_dev_t;
+typedef struct audio_mixer_ops audio_mixer_ops_t;
+typedef struct audio_engine_ops audio_engine_ops_t;
+typedef struct audio_ctrl audio_ctrl_t;
+typedef struct audio_ctrl_desc audio_ctrl_desc_t;
+
+struct audio_ctrl_desc {
+	const char		*acd_name;		/* Controls Mnemonic */
+	uint32_t		acd_type;		/* Entry type */
+	uint64_t		acd_flags;		/* Characteristics */
+	/*
+	 * Minimum and Maximum values for this control.  The value
+	 * must be between these values inclusive.  For
+	 * AUDIO_CTRL_TYPE_ENUM, the maxvalue is a bitmask of
+	 * supported controls.
+	 */
+	uint64_t		acd_maxvalue;		/* max value control */
+	uint64_t		acd_minvalue;		/* min value control */
+	/*
+	 * Array of pointers to names for each enum position. This
+	 * should be null for all but AUDIO_CTRL_TYPE_ENUM.
+	 */
+	const char		*acd_enum[64];
+};
+
+/*
+ * Audio data formats.  Note that these are represented int a bit
+ * field, to allow for multiple values to be represented in the same
+ * integer (in certain portions of the API.)
+ */
+#define	AUDIO_FORMAT_NONE		0x00000000U
+#define	AUDIO_FORMAT_ULAW		0x00000001U
+#define	AUDIO_FORMAT_ALAW		0x00000002U
+#define	AUDIO_FORMAT_S8			0x00000004U
+#define	AUDIO_FORMAT_U8			0x00000008U
+#define	AUDIO_FORMAT_S16_LE		0x00000010U
+#define	AUDIO_FORMAT_S16_BE		0x00000020U
+#define	AUDIO_FORMAT_U16_LE		0x00000040U
+#define	AUDIO_FORMAT_U16_BE		0x00000080U
+#define	AUDIO_FORMAT_S24_LE		0x00000100U
+#define	AUDIO_FORMAT_S24_BE		0x00000200U
+#define	AUDIO_FORMAT_S32_LE		0x00000400U
+#define	AUDIO_FORMAT_S32_BE		0x00000800U
+#define	AUDIO_FORMAT_S24_PACKED		0x00001000U
+#define	AUDIO_FORMAT_AC3		0x00010000U
+#define	AUDIO_FORMAT_OPAQUE_MASK	0xffff0000U
+#define	AUDIO_FORMAT_CONVERTIBLE	0x0000ffffU
+/*
+ * We only support signed 16, 24, and 32 bit format conversions in the
+ * engines, for simplicity.  (We haven't run into any engines that
+ * require other formats.)
+ */
+#define	AUDIO_FORMAT_PCM		0x00000f30
+
+/*
+ * Some big endian/little endian handling macros (native endian and opposite
+ * endian formats). The usage of these macros is described in the OSS
+ * Programmer's Manual.
+ */
+
+#if defined(_BIG_ENDIAN)
+
+#define	AUDIO_FORMAT_S16_NE	AUDIO_FORMAT_S16_BE
+#define	AUDIO_FORMAT_U16_NE	AUDIO_FORMAT_U16_BE
+#define	AUDIO_FORMAT_S32_NE	AUDIO_FORMAT_S32_BE
+#define	AUDIO_FORMAT_S24_NE	AUDIO_FORMAT_S24_BE
+#define	AUDIO_FORMAT_S16_OE	AUDIO_FORMAT_S16_LE
+#define	AUDIO_FORMAT_U16_OE	AUDIO_FORMAT_U16_LE
+#define	AUDIO_FORMAT_S32_OE	AUDIO_FORMAT_S32_LE
+#define	AUDIO_FORMAT_S24_OE	AUDIO_FORMAT_S24_LE
+
+#elif defined(_LITTLE_ENDIAN)
+#define	AUDIO_FORMAT_S16_NE	AUDIO_FORMAT_S16_LE
+#define	AUDIO_FORMAT_U16_NE	AUDIO_FORMAT_U16_LE
+#define	AUDIO_FORMAT_S32_NE	AUDIO_FORMAT_S32_LE
+#define	AUDIO_FORMAT_S24_NE	AUDIO_FORMAT_S24_LE
+#define	AUDIO_FORMAT_S16_OE	AUDIO_FORMAT_S16_BE
+#define	AUDIO_FORMAT_U16_OE	AUDIO_FORMAT_U16_BE
+#define	AUDIO_FORMAT_S32_OE	AUDIO_FORMAT_S32_BE
+#define	AUDIO_FORMAT_S24_OE	AUDIO_FORMAT_S24_BE
+
+#else
+#error "Machine endianness undefined"
+#endif
+
+/*
+ * These are parameterized around the maximum minor number available
+ * for use in the filesystem.  Unfortunately, we have to use 32-bit limits,
+ * because we could have 32-bit userland apps (we usually will, in fact).
+ */
+#define	AUDIO_MN_CLONE_NBITS	(NBITSMINOR32 - 1)
+#define	AUDIO_MN_CLONE_MASK	(1U << (AUDIO_MN_CLONE_NBITS - 1))
+#define	AUDIO_MN_TYPE_NBITS	(4)
+#define	AUDIO_MN_TYPE_SHIFT	(0)
+#define	AUDIO_MN_TYPE_MASK	((1U << AUDIO_MN_TYPE_NBITS) - 1)
+#define	AUDIO_MN_INST_NBITS	((NBITSMINOR32 - 1) - AUDIO_MN_TYPE_NBITS)
+#define	AUDIO_MN_INST_MASK	((1U << AUDIO_MN_INST_NBITS) - 1)
+#define	AUDIO_MN_INST_SHIFT	(AUDIO_MN_TYPE_NBITS)
+#define	AUDIO_MKMN(inst, typ)	\
+	(((inst) << AUDIO_MN_INST_SHIFT) | ((typ) << AUDIO_MN_TYPE_SHIFT))
+
+#define	AUDIO_MINOR_MIXER	(0)
+#define	AUDIO_MINOR_DSP		(1)
+/* 2 is reserved for now */
+#define	AUDIO_MINOR_DEVAUDIO	(3)
+#define	AUDIO_MINOR_DEVAUDIOCTL	(4)
+#define	AUDIO_MINOR_SNDSTAT	(AUDIO_MN_TYPE_MASK)
+
+/* reserved minors for driver specific use */
+#define	AUDIO_MINOR_DRV1	(AUDIO_MINOR_SNDSTAT - 1)
+#define	AUDIO_MINOR_DRV2	(AUDIO_MINOR_SNDSTAT - 2)
+
+
+/* Various controls */
+#define	AUDIO_CTRL_ID_VOLUME	"volume"
+#define	AUDIO_CTRL_ID_LINEOUT	"line-out"
+#define	AUDIO_CTRL_ID_FRONT	"front"
+#define	AUDIO_CTRL_ID_REAR	"rear"
+#define	AUDIO_CTRL_ID_HEADPHONE	"headphones"
+#define	AUDIO_CTRL_ID_CENTER	"center"
+#define	AUDIO_CTRL_ID_LFE	"lfe"
+#define	AUDIO_CTRL_ID_SURROUND	"surround"
+#define	AUDIO_CTRL_ID_SPEAKER	"speaker"
+#define	AUDIO_CTRL_ID_AUX1OUT	"aux1-out"
+#define	AUDIO_CTRL_ID_AUX2OUT	"aux2-out"
+#define	AUDIO_CTRL_ID_BASS	"bass"
+#define	AUDIO_CTRL_ID_TREBLE	"treble"
+#define	AUDIO_CTRL_ID_3DDEPTH	"3d-depth"
+#define	AUDIO_CTRL_ID_3DCENT	"3d-center"
+#define	AUDIO_CTRL_ID_3DENHANCE	"3d-enhance"
+#define	AUDIO_CTRL_ID_PHONE	"phone"
+#define	AUDIO_CTRL_ID_MIC	"mic"
+#define	AUDIO_CTRL_ID_LINEIN	"line-in"
+#define	AUDIO_CTRL_ID_CD	"cd"
+#define	AUDIO_CTRL_ID_VIDEO	"video"
+#define	AUDIO_CTRL_ID_AUX1IN	"aux1-in"
+#define	AUDIO_CTRL_ID_PCMIN	"pcm"
+#define	AUDIO_CTRL_ID_RECGAIN	"record-gain"
+#define	AUDIO_CTRL_ID_AUX2IN	"aux2-in"
+#define	AUDIO_CTRL_ID_MICBOOST	"micboost"
+#define	AUDIO_CTRL_ID_LOOPBACK	"loopback"
+#define	AUDIO_CTRL_ID_LOUDNESS	"loudness"
+#define	AUDIO_CTRL_ID_OUTPUTS	"outputs"
+#define	AUDIO_CTRL_ID_INPUTS	"inputs"
+#define	AUDIO_CTRL_ID_RECSRC	"record-source"
+#define	AUDIO_CTRL_ID_MONSRC	"monitor-source"
+#define	AUDIO_CTRL_ID_DIAG	"diag"
+#define	AUDIO_CTRL_ID_BEEP	"beep"
+#define	AUDIO_CTRL_ID_MONGAIN	"monitor-gain"
+#define	AUDIO_CTRL_ID_STEREOSIM	"stereo-simulate"	/* AC'97 feature */
+#define	AUDIO_CTRL_ID_MICGAIN	"mic-gain"		/* mono mic gain */
+#define	AUDIO_CTRL_ID_SPKSRC	"speaker-source"	/* AC'97 feature */
+#define	AUDIO_CTRL_ID_MICSRC	"mic-source"		/* AC'97 feature */
+#define	AUDIO_CTRL_ID_JACK1	"jack1"			/* jack repurposing */
+#define	AUDIO_CTRL_ID_JACK2	"jack2"
+#define	AUDIO_CTRL_ID_JACK3	"jack3"
+#define	AUDIO_CTRL_ID_JACK4	"jack4"
+#define	AUDIO_CTRL_ID_JACK5	"jack5"
+#define	AUDIO_CTRL_ID_JACK6	"jack6"
+#define	AUDIO_CTRL_ID_JACK7	"jack7"
+#define	AUDIO_CTRL_ID_DOWNMIX	"downmix"
+#define	AUDIO_CTRL_ID_SPREAD	"spread"
+
+/*
+ * Names for ports.
+ */
+#define	AUDIO_PORT_MIC			"mic"
+#define	AUDIO_PORT_CD			"cd"
+#define	AUDIO_PORT_VIDEO		"video"
+#define	AUDIO_PORT_AUX1OUT		"aux1-out"
+#define	AUDIO_PORT_AUX2OUT		"aux2-out"
+#define	AUDIO_PORT_LINEOUT		"line-out"
+#define	AUDIO_PORT_STEREOMIX		"stereo-mix"
+#define	AUDIO_PORT_MONOMIX		"mono-mix"
+#define	AUDIO_PORT_PHONE		"phone"
+#define	AUDIO_PORT_REAR			"rear"
+#define	AUDIO_PORT_CENTER		"center"
+#define	AUDIO_PORT_SURROUND		"surround"
+#define	AUDIO_PORT_LFE			"lfe"
+#define	AUDIO_PORT_SPEAKER		"speaker"
+#define	AUDIO_PORT_LINEIN		"line-in"
+#define	AUDIO_PORT_AUX1IN		"aux1-in"
+#define	AUDIO_PORT_AUX2IN		"aux2-in"
+#define	AUDIO_PORT_HEADPHONES		"headphones"
+#define	AUDIO_PORT_SPDIFIN		"spdif-in"
+#define	AUDIO_PORT_SPDIFOUT		"spdif-out"
+#define	AUDIO_PORT_CENLFE		"center/lfe"	/* combined jack use */
+#define	AUDIO_PORT_MIC1			"mic1"
+#define	AUDIO_PORT_MIC2			"mic2"
+#define	AUDIO_PORT_DIGOUT		"digital-out"
+#define	AUDIO_PORT_DIGIN		"digital-in"
+#define	AUDIO_PORT_HDMI			"hdmi"
+#define	AUDIO_PORT_MODEM		"modem"
+#define	AUDIO_PORT_HANDSET		"handset"
+#define	AUDIO_PORT_OTHER		"other"
+#define	AUDIO_PORT_STEREO		"stereo"	/* e.g. mic array */
+#define	AUDIO_PORT_NONE			"none"
+
+/*
+ * A few common values that sometimes we see.
+ */
+#define	AUDIO_VALUE_ON			"on"
+#define	AUDIO_VALUE_OFF			"off"
+#define	AUDIO_VALUE_VERYLOW		"very-low"
+#define	AUDIO_VALUE_LOW			"low"
+#define	AUDIO_VALUE_MEDIUM		"medium"
+#define	AUDIO_VALUE_HIGH		"high"
+#define	AUDIO_VALUE_VERYHIGH		"very-high"
+
+/*
+ * Posible return values for walk callback function
+ */
+#define	AUDIO_WALK_CONTINUE	1	/* continue walk */
+#define	AUDIO_WALK_STOP		2	/* stop the walk */
+#define	AUDIO_WALK_RESTART	3	/* restart the walk from beginning */
+
+/*
+ * Control types
+ */
+#define	AUDIO_CTRL_TYPE_BOOLEAN		1	/* ON/OFF control */
+#define	AUDIO_CTRL_TYPE_ENUM		2	/* Enumerated list */
+#define	AUDIO_CTRL_TYPE_STEREO		3	/* stereo level control */
+#define	AUDIO_CTRL_TYPE_MONO		4	/* mono level control */
+#define	AUDIO_CTRL_TYPE_METER		5	/* VU meter */
+
+/*
+ * Control characteristics flags
+ */
+#define	AUDIO_CTRL_FLAG_READABLE	0x00000001	/* Control readable */
+#define	AUDIO_CTRL_FLAG_WRITEABLE	0x00000002	/* Control writable */
+#define	AUDIO_CTRL_FLAG_RW		0x00000003	/* Read/writeable */
+#define	AUDIO_CTRL_FLAG_VUPEAK		0x00000004	/* peak meter */
+#define	AUDIO_CTRL_FLAG_CENTIBEL	0x00000008	/* Centibel (0.1 dB) */
+#define	AUDIO_CTRL_FLAG_DECIBEL		0x00000010	/* Step size of 1 dB */
+#define	AUDIO_CTRL_FLAG_POLL		0x00000020	/* May change itself */
+#define	AUDIO_CTRL_FLAG_MAINVOL		0x00000100	/* Main volume ctrl */
+#define	AUDIO_CTRL_FLAG_PCMVOL		0x00000200	/* PCM output volume */
+#define	AUDIO_CTRL_FLAG_RECVOL		0x00000400	/* PCM record volume */
+#define	AUDIO_CTRL_FLAG_MONVOL		0x00000800	/* Monitor volume */
+#define	AUDIO_CTRL_FLAG_PLAY		0x00001000	/* Playback control */
+#define	AUDIO_CTRL_FLAG_REC		0x00002000	/* Record control */
+#define	AUDIO_CTRL_FLAG_3D		0x00004000	/* 3D effect control */
+#define	AUDIO_CTRL_FLAG_TONE		0x00008000	/* Tone control */
+#define	AUDIO_CTRL_FLAG_MONITOR		0x00010000	/* Monitor control */
+#define	AUDIO_CTRL_FLAG_DIGITAL		0x00020000	/* Digital control */
+
+/*
+ * AUDIO_CTRL_TYPE_ENUM might allow more than a single value to be
+ * selected.  (Value is a bitmask.)
+ */
+#define	AUDIO_CTRL_FLAG_MULTI		0x00000040
+
+#endif	/* _KERNEL */
+
+#ifdef	__cplusplus
+}
+#endif
+
+#endif	/* _SYS_AUDIO_AUDIO_COMMON_H */
diff --git a/usr/src/uts/common/sys/audio/audio_driver.h b/usr/src/uts/common/sys/audio/audio_driver.h
new file mode 100644
index 0000000000..256698c1fe
--- /dev/null
+++ b/usr/src/uts/common/sys/audio/audio_driver.h
@@ -0,0 +1,241 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_SYS_AUDIO_AUDIO_DRIVER_H
+#define	_SYS_AUDIO_AUDIO_DRIVER_H
+
+#include <sys/types.h>
+#include <sys/list.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/audio/audio_common.h>
+
+
+#ifdef	__cplusplus
+extern "C" {
+#endif
+
+#ifdef	_KERNEL
+
+struct audio_engine_ops {
+	int	audio_engine_version;
+#define	AUDIO_ENGINE_VERSION	0
+
+	/*
+	 * Initialize engine, including buffer allocation.  Arguments
+	 * that are pointers are hints.  On return, they are updated with
+	 * the actual values configured by the driver.
+	 */
+	int	(*audio_engine_open)(void *, int flags,
+		    unsigned *fragfr, unsigned *nfrags, caddr_t *buf);
+	void	(*audio_engine_close)(void *);
+
+	/*
+	 * Start and stop are used to actually get the hardware running
+	 * or stop the hardware.  Until this is kicked off, the engine
+	 * will not actually transfer data.  These are not destructive to
+	 * ring positions, etc.  (Think of it like pause/play).
+	 */
+	int	(*audio_engine_start)(void *);
+	void	(*audio_engine_stop)(void *);
+
+	/*
+	 * Obtain the engine offset.  Offsets start at zero at engine_open,
+	 * and keep counting upwards.  Count is returned in frames.
+	 */
+	uint64_t	(*audio_engine_count)(void *);
+
+	/*
+	 * The following entry points return the currently configured
+	 * status of the engine.  It is assumed that the engine's
+	 * configuration is relatively fixed, and does not change
+	 * while open, or in response to open.
+	 *
+	 * However, in the future we might like to allow for the
+	 * device to change the settings while it is not open, which
+	 * could allow for mixerctl to change the configured channels,
+	 * for example.  In order to synchronize this properly, we'll
+	 * need the engine to perform a notification/request.  That
+	 * will be added later.
+	 *
+	 * AC3: We will have to figure out how to support dynamically
+	 * selecting different sampling frequencies for AC3, since
+	 * it needs to be able to support 32, 44.1, and 48 kHz.
+	 * Perhaps special flags used during open() would do the trick.
+	 */
+	int	(*audio_engine_format)(void *);
+	int	(*audio_engine_channels)(void *);
+	int	(*audio_engine_rate)(void *);
+
+	/*
+	 * DMA cache synchronization.  The framework does this on
+	 * behalf of the driver for both input and output.  The driver
+	 * is responsible for tracking the direction (based on the
+	 * flags passed to ae_open()), and dealing with any partial
+	 * synchronization if any is needed.
+	 */
+	void	(*audio_engine_sync)(void *, unsigned);
+
+	/*
+	 * The framework may like to know how deep the device queues data.
+	 * This can be used to provide a more accurate latency calculation.
+	 */
+	size_t	(*audio_engine_qlen)(void *);
+
+	/*
+	 * If the driver doesn't use simple interleaving, then we need to
+	 * know more about the offsets of channels within the buffer.
+	 * We obtain both the starting offset within the buffer, and the
+	 * increment for each new sample.  As usual, these are given in
+	 * samples.  If this entry point is NULL, the framework assumes
+	 * that simple interlevaing is used instead.
+	 */
+	void	(*audio_engine_chinfo)(void *, int chan, unsigned *offset,
+	    unsigned *incr);
+};
+
+void audio_init_ops(struct dev_ops *, const char *);
+void audio_fini_ops(struct dev_ops *);
+
+audio_dev_t *audio_dev_alloc(dev_info_t *, int);
+void audio_dev_free(audio_dev_t *);
+
+void audio_dev_set_description(audio_dev_t *, const char *);
+void audio_dev_set_version(audio_dev_t *, const char *);
+void audio_dev_add_info(audio_dev_t *, const char *);
+
+audio_engine_t *audio_engine_alloc(audio_engine_ops_t *, unsigned);
+void audio_engine_set_private(audio_engine_t *, void *);
+void *audio_engine_get_private(audio_engine_t *);
+void audio_engine_free(audio_engine_t *);
+
+void audio_dev_add_engine(audio_dev_t *, audio_engine_t *);
+void audio_dev_remove_engine(audio_dev_t *, audio_engine_t *);
+int audio_dev_register(audio_dev_t *);
+int audio_dev_unregister(audio_dev_t *);
+void audio_dev_warn(audio_dev_t *, const char *, ...);
+/* DEBUG ONLY */
+void audio_dump_bytes(const uint8_t *w, int dcount);
+void audio_dump_words(const uint16_t *w, int dcount);
+void audio_dump_dwords(const uint32_t *w, int dcount);
+
+/*
+ * Drivers call these.
+ */
+void audio_engine_consume(audio_engine_t *);
+void audio_engine_produce(audio_engine_t *);
+void audio_engine_reset(audio_engine_t *);
+
+/* Engine flags */
+#define	ENGINE_OUTPUT_CAP	(1U << 2)
+#define	ENGINE_INPUT_CAP	(1U << 3)
+#define	ENGINE_CAPS		(ENGINE_OUTPUT_CAP | ENGINE_INPUT_CAP)
+#define	ENGINE_DRIVER_FLAGS	(0xffff)	/* flags usable by driver */
+
+#define	ENGINE_OUTPUT		(1U << 16)	/* fields not for driver use */
+#define	ENGINE_INPUT		(1U << 17)
+#define	ENGINE_OPEN		(1U << 18)
+#define	ENGINE_RUNNING		(1U << 19)
+#define	ENGINE_EXCLUSIVE	(1U << 20)	/* exclusive use, e.g. AC3 */
+#define	ENGINE_NDELAY		(1U << 21)	/* non-blocking open */
+#define	ENGINE_WAKE		(1U << 22)	/* wakeup tq running */
+
+/*
+ * entry points used by legacy SADA drivers
+ */
+int audio_legacy_open(queue_t *, dev_t *, int, int, cred_t *);
+int audio_legacy_close(queue_t *, int, cred_t *);
+int audio_legacy_wput(queue_t *, mblk_t *);
+int audio_legacy_wsrv(queue_t *);
+
+
+
+/*
+ * Audio device controls
+ */
+
+/*
+ * Control read or write driver function type.
+ *
+ * Returns zero on success, errno on failure.
+ */
+typedef int (*audio_ctrl_wr_t)(void *, uint64_t);
+typedef int (*audio_ctrl_rd_t)(void *, uint64_t *);
+
+
+/*
+ * This will allocate and register a control for my audio device.
+ *
+ * On success this will return a control structure else NULL.
+ */
+audio_ctrl_t *audio_dev_add_control(audio_dev_t *,
+    audio_ctrl_desc_t *, audio_ctrl_rd_t, audio_ctrl_wr_t, void *);
+
+/*
+ * Add a synthetic PCM volume control.  This should only be used by
+ * devices which have no physical PCM volume controls.  The control
+ * implements a simple attenuator on the PCM data; unlike AC'97 there
+ * is no "gain", so using this instead of a hardware control may
+ * result in loss range.  The control is implemented using
+ * AUDIO_CTRL_ID_VOLUME.
+ */
+int audio_dev_add_soft_volume(audio_dev_t *);
+
+/*
+ * This will remove a control from an audio device.
+ */
+void audio_dev_del_control(audio_ctrl_t *);
+
+/*
+ * This will tell the framework that controls have changed
+ * and it should update its values.
+ */
+void audio_dev_update_controls(audio_dev_t *);
+
+/*
+ * This is used to read the current value of a control.
+ * Note, this will cause a callback into the driver to get the value.
+ *
+ * On return zero is returned on success else errno is returned.
+ */
+int audio_control_read(audio_ctrl_t *, uint64_t *);
+
+/*
+ * This is used to write a value to a control.
+ * Note, this will cause a callback into the driver to write the value.
+ *
+ * On return zero is returned on success else errno is returned.
+ */
+int audio_control_write(audio_ctrl_t *, uint64_t);
+
+#endif	/* _KERNEL */
+
+#ifdef	__cplusplus
+}
+#endif
+
+#endif	/* _SYS_AUDIO_AUDIO_DRIVER_H */
diff --git a/usr/src/uts/common/sys/audio/audio_mixer.h b/usr/src/uts/common/sys/audio/audio_mixer.h
index 4e6571dba8..4f84acb44f 100644
--- a/usr/src/uts/common/sys/audio/audio_mixer.h
+++ b/usr/src/uts/common/sys/audio/audio_mixer.h
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,8 +19,8 @@
  * CDDL HEADER END
  */
 /*
- * Copyright (c) 1999-2001 by Sun Microsystems, Inc.
- * All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
  *
  * This header file defines the public interfaces for the audio mixer
  * audio personality module.
@@ -34,8 +33,6 @@
 #ifndef	_SYS_AUDIO_MIXER_H
 #define	_SYS_AUDIO_MIXER_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 #ifdef	__cplusplus
 extern "C" {
 #endif
@@ -45,17 +42,16 @@ extern "C" {
 /*
  * Miscellaneous defines
  */
-#define	AM_MAX_QUEUED_MSGS_SIZE		(49000*2*4)	/* ~1 secs of audio */
-#define	AM_MIN_QUEUED_MSGS_SIZE		(24500*2*4)	/* ~0.5 secs of audio */
+struct am_ad_sample_rates {
+	uint_t		*ad_srs;	/* NULL term. list of sample rates */
+};
+typedef struct am_ad_sample_rates am_ad_sample_rates_t;
 
 /*
  * am_ad_ch_cap_t	- Audio Driver play/record capabilities
  */
 struct am_ad_ch_cap {
 	am_ad_sample_rates_t	ad_mixer_srs;	/* mixer mode sample rates */
-	am_ad_sample_rates_t	ad_compat_srs;	/* compat mode sample rates */
-	am_ad_src_entry_t	*ad_conv;	/* sample rate conv. routines */
-	void			*ad_sr_info;	/* sample rate conv. info */
 	uint_t			*ad_chs;	/* list of channel types */
 	int			ad_int_rate;	/* interrupt rate */
 	int			ad_max_chs;	/* max channels */
@@ -76,50 +72,31 @@ typedef struct am_ad_cap_comb am_ad_cap_comb_t;
  * am_ad_entry_t	- Audio Driver ops vector definition
  */
 struct am_ad_entry {
-	int		(*ad_setup)(audiohdl_t handle, int stream, int flags);
-	void		(*ad_teardown)(audiohdl_t handle, int stream, int dir);
-	int		(*ad_set_config)(audiohdl_t handle, int stream,
+	int		(*ad_setup)(audiohdl_t handle, int flags);
+	void		(*ad_teardown)(audiohdl_t handle, int dir);
+	int		(*ad_set_config)(audiohdl_t handle,
 				int command, int dir, int arg1, int arg2);
-	int		(*ad_set_format)(audiohdl_t handle, int stream,
+	int		(*ad_set_format)(audiohdl_t handle,
 				int dir, int sample_rate, int channels,
 				int precision, int encoding);
-	int		(*ad_start_play)(audiohdl_t handle, int stream);
-	void		(*ad_pause_play)(audiohdl_t handle, int stream);
-	void		(*ad_stop_play)(audiohdl_t handle, int stream);
-	int		(*ad_start_record)(audiohdl_t handle, int stream);
-	void		(*ad_stop_record)(audiohdl_t handle, int stream);
-	int		(*ad_ioctl)(audiohdl_t handle, int channel,
-				queue_t *q, mblk_t *mp, int *error);
-	int		(*ad_iocdata)(audiohdl_t handle, int channel,
-				queue_t *q, mblk_t *mp, int *error);
+	int		(*ad_start_play)(audiohdl_t handle);
+	void		(*ad_stop_play)(audiohdl_t handle);
+	int		(*ad_start_record)(audiohdl_t handle);
+	void		(*ad_stop_record)(audiohdl_t handle);
 };
 typedef struct am_ad_entry am_ad_entry_t;
 
-/* ad_set_config() and ad_set_format() stream # */
-#define	AM_SET_CONFIG_BOARD	(-1)	/* for the whole board */
-
 /* ad_set_config() commands */
 #define	AM_SET_GAIN		0x01	/* set input/ouput channel gain */
-#define	AM_SET_GAIN_BAL		0x02	/* set input/ouput channel gain */
 #define	AM_SET_PORT		0x03	/* set input/output port */
 #define	AM_SET_MONITOR_GAIN	0x04	/* set monitor gain */
 #define	AM_OUTPUT_MUTE		0x05	/* mute output */
-#define	AM_MONO_MIC		0x06	/* set which mono microphone */
 #define	AM_MIC_BOOST		0x07	/* enable/disable mic preamp */
-#define	AM_BASS_BOOST		0x08	/* boost output bass */
-#define	AM_MID_BOOST		0x09	/* boost output mid range */
-#define	AM_TREBLE_BOOST		0x0a	/* boost output treble */
-#define	AM_LOUDNESS		0x0b	/* enable/disable output loudness */
-#define	AM_SET_DIAG_MODE	0x0c	/* set diagnostics mode */
 
 /*
  * am_ad_info_t	- Audio Driver configuration information structure
  */
 struct am_ad_info {
-	int		ad_int_vers;	/* Audio Driver interface version */
-	int		ad_mode;	/* MIXER or COMPAT mode */
-	uint_t		ad_add_mode;	/* additional mode information */
-	int		ad_codec_type;	/* Codec type */
 	audio_info_t	*ad_defaults;	/* Audio Driver audio_info_t struct */
 	am_ad_ch_cap_t	ad_play;	/* play capabilities */
 	am_ad_ch_cap_t	ad_record;	/* record capabilities */
@@ -127,66 +104,13 @@ struct am_ad_info {
 	am_ad_cap_comb_t *ad_rec_comb;	/* list of rec cap. combinations */
 	am_ad_entry_t	*ad_entry;	/* Audio Driver entry points */
 	audio_device_t	*ad_dev_info;	/* device information */
-	uint_t		ad_diag_flags;	/* flags that specify diagnostics sup */
-	uint_t		ad_diff_flags;	/* format difference flags */
-	uint_t		ad_assist_flags; /* audio stream assist flags */
-	uint_t		ad_misc_flags;	/* misc. flags */
-	uint_t		ad_translate_flags; /* translate flags */
 	int		ad_num_mics;	/* # of mic inputs */
-	uint_t		_xxx[4];	/* reserved for future use */
 };
 typedef struct am_ad_info am_ad_info_t;
 
 /* the taskq lock must be held in order for ad_mode to be valid */
 _NOTE(SCHEME_PROTECTS_DATA("method", am_ad_info::ad_mode))
 
-/* am_ad_info.ad_int_vers defines */
-#define	AM_VERSION	AM_VERS2
-#define	AM_VERS2	2		/* Supported interface version */
-
-/* am_ad_info.ad_add_mode defines */
-#define	AM_ADD_MODE_DIAG_MODE	0x00000001u	/* dev supports diagnostics */
-#define	AM_ADD_MODE_MIC_BOOST	0x00000002u	/* mic boost enabled */
-
-/* am_ad_info.ad_codec_type defines */
-#define	AM_TRAD_CODEC		0x00000001u	/* traditional Codec */
-#define	AM_MS_CODEC		0x00000002u	/* multi-stream Codec */
-
-/* am_ad_info.ad_diag_flags defines */
-#define	AM_DIAG_INTERNAL_LOOP	0x00000001u	/* dev has internal loopbacks */
-
-/* am_ad_info.ad_diff_flags defines */
-#define	AM_DIFF_SR		0x00000001u	/* p/r sample rate may differ */
-#define	AM_DIFF_CH		0x00000002u	/* p/r channels may differ */
-#define	AM_DIFF_PREC		0x00000004u	/* p/r precision may differ */
-#define	AM_DIFF_ENC		0x00000008u	/* p/r encoding may differ */
-
-/* am_ad_info.ad_assist_flags defines */
-#define	AM_ASSIST_BASE		0x00000001u	/* device has base boost */
-#define	AM_ASSIST_MID		0x00000002u	/* device has mid range boost */
-#define	AM_ASSIST_TREBLE	0x00000004u	/* device has treble boost */
-#define	AM_ASSIST_LOUDNESS	0x00000008u	/* device has loudness boost */
-#define	AM_ASSIST_MIC		0x00000010u	/* mic has preamp boost */
-
-/* am_ad_info.ad_misc_flags defines */
-#define	AM_MISC_PP_EXCL		0x00000001u	/* play ports are exclusive */
-#define	AM_MISC_RP_EXCL		0x00000002u	/* record ports are exclusive */
-#define	AM_MISC_MONO_MIC	0x00000004u	/* mono mic */
-#define	AM_MISC_MONO_DUP	0x00000008u	/* mono is duped to all chs */
-
-/* am_ad_info.ad_translate_flags */
-#define	AM_MISC_8_P_TRANSLATE	0x00000001u	/* trans. signed to unsigned */
-#define	AM_MISC_16_P_TRANSLATE	0x00000002u	/* trans. signed to unsigned */
-#define	AM_MISC_8_R_TRANSLATE	0x00010000u	/* trans. unsigned to signed */
-#define	AM_MISC_16_R_TRANSLATE	0x00020000u	/* trans. unsigned to signed */
-
-/*
- * Support for custom audio driver ioctl()s.
- */
-#define	AM_WIOCDATA			0	/* returned by Audio Driver */
-#define	AM_ACK				1	/* with private ioctl() & */
-#define	AM_NACK				2	/* iocdata() routines */
-
 /*
  * Audio Mixer Driver Entry Point Routines
  */
@@ -196,13 +120,13 @@ int am_detach(audiohdl_t handle, ddi_detach_cmd_t cmd);
 /*
  * Audio Mixer Driver Device Dependent Driver Play Routines
  */
-int am_get_audio(audiohdl_t handle, void *buf, int channel, int samples);
-void am_play_shutdown(audiohdl_t handle, int channel);
+int am_get_audio(audiohdl_t handle, void *buf, int samples);
+void am_play_shutdown(audiohdl_t handle);
 
 /*
  * Audio Mixer Driver Device Dependent Driver Record Routines
  */
-void am_send_audio(audiohdl_t handle, void *buf, int channel, int samples);
+void am_send_audio(audiohdl_t handle, void *buf, int samples);
 
 /*
  * Audio Mixer Driver Device Dependent Driver Miscellaneous Routines
diff --git a/usr/src/uts/common/sys/audio/audio_oss.h b/usr/src/uts/common/sys/audio/audio_oss.h
new file mode 100644
index 0000000000..3cb3b9548a
--- /dev/null
+++ b/usr/src/uts/common/sys/audio/audio_oss.h
@@ -0,0 +1,908 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright (C) 4Front Technologies 1996-2008.
+ *
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+#ifndef	_SYS_AUDIO_OSS_H
+#define	_SYS_AUDIO_OSS_H
+
+#include <sys/types.h>
+#include <sys/time.h>
+
+/*
+ * These are the ioctl calls for all Solaris /dev/dsp and /dev/mixer audio
+ * devices.
+ *
+ * Note that the contents of this file include definitions which exist
+ * primarily for compatibility.  Many of the defines here are not
+ * actually implemented, but exist solely to facilitate compilation of
+ * programs from other operating systems.  Other definitions here may
+ * not be fully supported or may otherwise be obsolete. There are many
+ * things in this file which should not be used on SunOS.
+ *
+ * Please read the documentation to determine which portions of the
+ * API are fully supported and recommended for use in new
+ * applications.
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Buffer status queries.
+ * SNDCTL_DSP_GETOSPACE and SNDCTL_DSP_GETISPACE
+ */
+typedef struct audio_buf_info {
+	int fragments;		/* # of available fragments */
+	int fragstotal;		/* Total # of fragments allocated */
+	int fragsize;		/* Size of a fragment in bytes */
+	int bytes;		/* Available space in bytes */
+	/* Note! 'bytes' could be more than fragments*fragsize */
+} audio_buf_info;
+
+/*
+ * Sync groups for audio devices.
+ * SNDCTL_DSP_SYNCGROUP and SNDCTL_DSP_SYNCSTART
+ */
+typedef struct oss_syncgroup {
+	int id;
+	int mode;
+	int filler[16];
+} oss_syncgroup;
+
+/*
+ * SNDCTL_DSP_GETERROR
+ */
+typedef struct audio_errinfo {
+	int play_underruns;
+	int rec_overruns;
+	unsigned int play_ptradjust;
+	unsigned int rec_ptradjust;
+	int play_errorcount;
+	int rec_errorcount;
+	int play_lasterror;
+	int rec_lasterror;
+	int play_errorparm;
+	int rec_errorparm;
+	int filler[16];
+} audio_errinfo;
+
+/*
+ * SNDCTL_DSP_GETIPTR and SNDCTL_DSP_GETOPTR
+ */
+typedef struct count_info {
+	unsigned int bytes;	/* Total # of bytes processed */
+	int blocks;		/* # of fragment transitions since last time */
+	int ptr;		/* Current DMA pointer value */
+} count_info;
+
+/*
+ * SNDCTL_DSP_CURENT_IPTR and SNDCTL_DSP_CURRENT_OPTR
+ */
+typedef struct {
+	long long samples;	/* Total # of samples */
+	int fifo_samples;	/* Samples in device FIFO */
+	int filler[32];		/* For future use */
+} oss_count_t;
+
+/*
+ * SNDCTL_DSP_GET_RECSRC_NAMES and SNDCTL_DSP_GET_PLAYTGT_NAMES
+ */
+#define	OSS_ENUM_MAXVALUE	255
+typedef struct oss_mixer_enuminfo {
+	int dev;
+	int ctrl;
+	int nvalues;
+	int version;
+	short strindex[OSS_ENUM_MAXVALUE];
+	char strings[3000];
+} oss_mixer_enuminfo;
+
+/*
+ * Digital interface (S/PDIF) control interface
+ * SNDCTL_DSP_READCTL and SNDCTL_DSP_WRITECTL
+ */
+typedef struct oss_digital_control {
+	unsigned int caps;
+#define	DIG_CBITIN_NONE		0x00000000
+#define	DIG_CBITIN_LIMITED	0x00000001
+#define	DIG_CBITIN_DATA		0x00000002
+#define	DIG_CBITIN_BYTE0	0x00000004
+#define	DIG_CBITIN_FULL		0x00000008
+#define	DIG_CBITIN_MASK		0x0000000f
+#define	DIG_CBITOUT_NONE	0x00000000
+#define	DIG_CBITOUT_LIMITED	0x00000010
+#define	DIG_CBITOUT_BYTE0	0x00000020
+#define	DIG_CBITOUT_FULL	0x00000040
+#define	DIG_CBITOUT_DATA	0x00000080
+#define	DIG_CBITOUT_MASK	0x000000f0
+#define	DIG_UBITIN		0x00000100
+#define	DIG_UBITOUT		0x00000200
+#define	DIG_VBITOUT		0x00000400
+#define	DIG_OUTRATE		0x00000800
+#define	DIG_INRATE		0x00001000
+#define	DIG_INBITS		0x00002000
+#define	DIG_OUTBITS		0x00004000
+#define	DIG_EXACT		0x00010000
+#define	DIG_PRO			0x00020000
+#define	DIG_CONSUMER		0x00040000
+#define	DIG_PASSTHROUGH		0x00080000
+#define	DIG_OUTSEL		0x00100000
+
+	unsigned int valid;
+#define	VAL_CBITIN		0x00000001
+#define	VAL_UBITIN		0x00000002
+#define	VAL_CBITOUT		0x00000004
+#define	VAL_UBITOUT		0x00000008
+#define	VAL_ISTATUS		0x00000010
+#define	VAL_IRATE		0x00000020
+#define	VAL_ORATE		0x00000040
+#define	VAL_INBITS		0x00000080
+#define	VAL_OUTBITS		0x00000100
+#define	VAL_REQUEST		0x00000200
+#define	VAL_OUTSEL		0x00000400
+
+#define	VAL_OUTMASK (VAL_CBITOUT|VAL_UBITOUT|VAL_ORATE|VAL_OUTBITS|VAL_OUTSEL)
+
+	unsigned int request;
+	unsigned int param;
+#define	SPD_RQ_PASSTHROUGH	1
+
+	unsigned char cbitin[24];
+	unsigned char ubitin[24];
+	unsigned char cbitout[24];
+	unsigned char ubitout[24];
+
+	unsigned int outsel;
+#define	OUTSEL_DIGITAL		1
+#define	OUTSEL_ANALOG		2
+#define	OUTSEL_BOTH		(OUTSEL_DIGITAL|OUTSEL_ANALOG)
+
+	int in_data;		/* Audio/data if autodetectable by receiver */
+#define	IND_UNKNOWN		0
+#define	IND_AUDIO		1
+#define	IND_DATA		2
+
+	int in_locked;		/* Receiver locked */
+#define	LOCK_NOT_INDICATED	0
+#define	LOCK_UNLOCKED		1
+#define	LOCK_LOCKED		2
+
+	int in_quality;		/* Input signal quality */
+#define	IN_QUAL_NOT_INDICATED	0
+#define	IN_QUAL_POOR		1
+#define	IN_QUAL_GOOD		2
+
+	int in_vbit;
+	int out_vbit;		/* V bits */
+#define	VBIT_NOT_INDICATED	0
+#define	VBIT_OFF		1
+#define	VBIT_ON			2
+
+	unsigned int in_errors;	/* Various input error conditions */
+#define	INERR_CRC		0x0001
+#define	INERR_QCODE_CRC		0x0002
+#define	INERR_PARITY		0x0004
+#define	INERR_BIPHASE		0x0008
+
+	int srate_in;
+	int srate_out;
+	int bits_in;
+	int bits_out;
+
+	int filler[32];
+} oss_digital_control;
+
+/*
+ * The "new" mixer API.
+ *
+ * This improved mixer API makes it possible to access every possible feature
+ * of every possible device. However you should read the mixer programming
+ * section of the OSS API Developer's Manual. There is no chance that you
+ * could use this interface correctly just by examining this header.
+ */
+#define	OSS_VERSION		0x040003
+#define	SOUND_VERSION		OSS_VERSION
+
+typedef struct oss_sysinfo {
+	char product[32];	/* E.g. SunOS Audio */
+	char version[32];	/* E.g. 4.0a */
+	int versionnum;		/* See OSS_GETVERSION */
+	char options[128];	/* NOT SUPPORTED */
+
+	int numaudios;		/* # of audio/dsp devices */
+	int openedaudio[8];	/* Mask of audio devices are busy */
+
+	int numsynths;		/* NOT SUPPORTED, always 0 */
+	int nummidis;		/* NOT SUPPORTED, always 0 */
+	int numtimers;		/* NOT SUPPORTED, always 0 */
+	int nummixers;		/* # of mixer devices */
+
+	int openedmidi[8];	/* Mask of midi devices are busy */
+	int numcards;		/* Number of sound cards in the system */
+	int numaudioengines;	/* Number of audio engines in the system */
+	char license[16];	/* E.g. "GPL" or "CDDL" */
+	char revision_info[256];	/* For internal use */
+	int filler[172];	/* For future expansion */
+} oss_sysinfo;
+
+typedef struct oss_mixext {
+	int dev;		/* Mixer device number */
+	int ctrl;		/* Extension number */
+	int type;		/* Entry type */
+#define	MIXT_DEVROOT		0	/* Device root entry */
+#define	MIXT_GROUP		1	/* Controller group */
+#define	MIXT_ONOFF		2	/* OFF (0) or ON (1) */
+#define	MIXT_ENUM		3	/* Enumerated (0 to maxvalue) */
+#define	MIXT_MONOSLIDER		4	/* Mono slider (0 to 255) */
+#define	MIXT_STEREOSLIDER	5	/* Stereo slider (dual 0 to 255) */
+#define	MIXT_MESSAGE		6	/* (Readable) textual message */
+#define	MIXT_MONOVU		7	/* VU meter value (mono) */
+#define	MIXT_STEREOVU		8	/* VU meter value (stereo) */
+#define	MIXT_MONOPEAK		9	/* VU meter peak value (mono) */
+#define	MIXT_STEREOPEAK		10	/* VU meter peak value (stereo) */
+#define	MIXT_RADIOGROUP		11	/* Radio button group */
+#define	MIXT_MARKER		12	/* Separator between entries */
+#define	MIXT_VALUE		13	/* Decimal value entry */
+#define	MIXT_HEXVALUE		14	/* Hexadecimal value entry */
+#define	MIXT_MONODB		15	/* OBSOLETE */
+#define	MIXT_STEREODB		16	/* OBSOLETE */
+#define	MIXT_SLIDER		17	/* Slider (mono, 31 bit int range) */
+#define	MIXT_3D			18
+#define	MIXT_MONOSLIDER16	19	/* Mono slider (0-32767) */
+#define	MIXT_STEREOSLIDER16	20	/* Stereo slider (dual 0-32767) */
+#define	MIXT_MUTE		21	/* Mute=1, unmute=0 */
+
+	/* Possible value range (minvalue to maxvalue) */
+	/* Note that maxvalue may also be smaller than minvalue */
+	int maxvalue;
+	int minvalue;
+
+	int flags;
+#define	MIXF_READABLE	0x00000001	/* Has readable value */
+#define	MIXF_WRITEABLE	0x00000002	/* Has writeable value */
+#define	MIXF_POLL	0x00000004	/* May change itself */
+#define	MIXF_HZ		0x00000008	/* Hertz scale */
+#define	MIXF_STRING	0x00000010	/* Use dynamic extensions for value */
+#define	MIXF_DYNAMIC	0x00000010	/* Supports dynamic extensions */
+#define	MIXF_OKFAIL	0x00000020	/* Interpret value as 1=OK, 0=FAIL */
+#define	MIXF_FLAT	0x00000040	/* NOT SUPPORTED */
+#define	MIXF_LEGACY	0x00000080	/* NOT SUPPORTED */
+#define	MIXF_CENTIBEL	0x00000100	/* Centibel (0.1 dB) step size */
+#define	MIXF_DECIBEL	0x00000200	/* Step size of 1 dB */
+#define	MIXF_MAINVOL	0x00000400	/* Main volume control */
+#define	MIXF_PCMVOL	0x00000800	/* PCM output volume control */
+#define	MIXF_RECVOL	0x00001000	/* PCM recording volume control */
+#define	MIXF_MONVOL	0x00002000	/* Input->output monitor volume */
+#define	MIXF_WIDE	0x00004000	/* NOT SUPPORTED */
+#define	MIXF_DESCR	0x00008000	/* NOT SUPPORTED */
+#define	MIXF_DISABLE	0x00010000	/* Control has been disabled */
+
+	char id[16];			/* Mnemonic ID (internal use) */
+	int parent;			/* Entry# of parent (-1 if root) */
+
+	int dummy;			/* NOT SUPPORTED */
+
+	int timestamp;
+
+	char data[64];			/* Misc data (entry type dependent) */
+	unsigned char enum_present[32];	/* Mask of allowed enum values */
+	int control_no;			/* NOT SUPPORTED, always -1 */
+
+	unsigned int desc;		/* Scope flags, etc */
+#define	MIXEXT_SCOPE_MASK		0x0000003f
+#define	MIXEXT_SCOPE_OTHER		0x00000000
+#define	MIXEXT_SCOPE_INPUT		0x00000001
+#define	MIXEXT_SCOPE_OUTPUT		0x00000002
+#define	MIXEXT_SCOPE_MONITOR		0x00000003
+#define	MIXEXT_SCOPE_RECSWITCH		0x00000004
+
+	char extname[32];
+	int update_counter;
+#ifdef	_KERNEL
+	int filler[6];
+	int enumbit;
+#else
+	int filler[7];
+#endif
+} oss_mixext;
+
+typedef struct oss_mixext_root {
+	char id[16];
+	char name[48];
+} oss_mixext_root;
+
+typedef struct oss_mixer_value {
+	int dev;
+	int ctrl;
+	int value;
+	int flags;		/* Reserved for future use. Initialize to 0 */
+	int timestamp;		/* Must be set to oss_mixext.timestamp */
+	int filler[8];		/* Reserved for future use. Initialize to 0 */
+} oss_mixer_value;
+
+#define	OSS_LONGNAME_SIZE	64
+#define	OSS_LABEL_SIZE		16
+#define	OSS_DEVNODE_SIZE	32
+typedef	char	oss_longname_t[OSS_LONGNAME_SIZE];
+typedef	char	oss_label_t[OSS_LABEL_SIZE];
+typedef	char	oss_devnode_t[OSS_DEVNODE_SIZE];
+
+
+typedef struct oss_audioinfo {
+	int dev;		/* Audio device number */
+	char name[64];
+	int busy;		/* 0, OPEN_READ, OPEN_WRITE, OPEN_READWRITE */
+	int pid;		/* Process ID, not used in SunOS */
+	int caps;		/* PCM_CAP_INPUT, PCM_CAP_OUTPUT */
+	int iformats;		/* Supported input formats */
+	int oformats;		/* Supported output formats */
+	int magic;		/* Internal use only */
+	char cmd[64];		/* Command using the device (if known) */
+	int card_number;
+	int port_number;
+	int mixer_dev;
+	int legacy_device;	/* Obsolete field. Replaced by devnode */
+	int enabled;		/* 1=enabled, 0=device not ready */
+	int flags;		/* internal use only - no practical meaning */
+	int min_rate;		/* Minimum sample rate */
+	int max_rate;		/* Maximum sample rate */
+	int min_channels;	/* Minimum number of channels */
+	int max_channels;	/* Maximum number of channels */
+	int binding;		/* DSP_BIND_FRONT, etc. 0 means undefined */
+	int rate_source;
+	char handle[32];
+#define	OSS_MAX_SAMPLE_RATES	20	/* Cannot be changed  */
+	unsigned int nrates;	/* Array of supported sample rates */
+	unsigned int rates[OSS_MAX_SAMPLE_RATES];
+	oss_longname_t song_name;	/* Song name (if given) */
+	oss_label_t label;	/* Device label (if given) */
+	int latency;		/* In usecs, -1=unknown */
+	oss_devnode_t devnode;	/* Device special file name (absolute path) */
+	int next_play_engine;
+	int next_rec_engine;
+	int filler[184];
+} oss_audioinfo;
+
+typedef struct oss_mixerinfo {
+	int dev;
+	char id[16];
+	char name[32];
+	int modify_counter;
+	int card_number;
+	int port_number;
+	char handle[32];
+	int magic;		/* Reserved */
+	int enabled;		/* Reserved */
+	int caps;
+#define	MIXER_CAP_VIRTUAL	0x00000001
+#define	MIXER_CAP_LAYOUT_B	0x00000002	/* For internal use only */
+#define	MIXER_CAP_NARROW	0x00000004	/* Conserve horiz space */
+	int flags;		/* Reserved */
+	int nrext;
+	/*
+	 * The priority field can be used to select the default
+	 * (motherboard) mixer device. The mixer with the highest
+	 * priority is the most preferred one. -2 or less means that
+	 * this device cannot be used as the default mixer.
+	 */
+	int priority;
+	oss_devnode_t devnode;  /* Device special file name (absolute path) */
+	int legacy_device;
+	int filler[245];	/* Reserved */
+} oss_mixerinfo;
+
+typedef struct oss_card_info {
+	int card;
+	char shortname[16];
+	char longname[128];
+	int flags;
+	char hw_info[400];
+	int intr_count;
+	int ack_count;
+	int filler[154];
+} oss_card_info;
+
+typedef struct mixer_info {	/* OBSOLETE */
+	char id[16];
+	char name[32];
+	int modify_counter;
+	int card_number;
+	int port_number;
+	char handle[32];
+} mixer_info;
+
+#define	MAX_PEAK_CHANNELS	128
+typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS];
+
+/* For use with SNDCTL_DSP_GET_CHNORDER */
+#define	CHID_UNDEF		0
+#define	CHID_L			1
+#define	CHID_R			2
+#define	CHID_C			3
+#define	CHID_LFE		4
+#define	CHID_LS			5
+#define	CHID_RS			6
+#define	CHID_LR			7
+#define	CHID_RR			8
+#define	CHNORDER_UNDEF		0x0000000000000000ULL
+#define	CHNORDER_NORMAL		0x0000000087654321ULL
+
+
+#define	OSSIOCPARM_MASK	0x1fff		/* parameters must be < 8192 bytes */
+#define	OSSIOC_VOID	0x00000000	/* no parameters */
+#define	OSSIOC_OUT	0x20000000	/* copy out parameters */
+#define	OSSIOC_IN	0x40000000	/* copy in parameters */
+#define	OSSIOC_INOUT	(OSSIOC_IN|OSSIOC_OUT)
+#define	OSSIOC_SZ(t)	((sizeof (t) & OSSIOCPARM_MASK) << 16)
+#define	OSSIOC_GETSZ(x)	(((x) >> 16) & OSSIOCPARM_MASK)
+
+#define	__OSSIO(x, y)		((int)(OSSIOC_VOID|(x<<8)|y))
+#define	__OSSIOR(x, y, t)	((int)(OSSIOC_OUT|OSSIOC_SZ(t)|(x<<8)|y))
+#define	__OSSIOW(x, y, t)	((int)(OSSIOC_IN|OSSIOC_SZ(t)|(x<<8)|y))
+#define	__OSSIOWR(x, y, t)	((int)(OSSIOC_INOUT|OSSIOC_SZ(t)|(x<<8)|y))
+
+#define	SNDCTL_SYSINFO		__OSSIOR('X', 1, oss_sysinfo)
+#define	OSS_SYSINFO		SNDCTL_SYSINFO  /* Old name */
+
+#define	SNDCTL_MIX_NRMIX	__OSSIOR('X', 2, int)
+#define	SNDCTL_MIX_NREXT	__OSSIOWR('X', 3, int)
+#define	SNDCTL_MIX_EXTINFO	__OSSIOWR('X', 4, oss_mixext)
+#define	SNDCTL_MIX_READ		__OSSIOWR('X', 5, oss_mixer_value)
+#define	SNDCTL_MIX_WRITE	__OSSIOWR('X', 6, oss_mixer_value)
+
+#define	SNDCTL_AUDIOINFO	__OSSIOWR('X', 7, oss_audioinfo)
+#define	SNDCTL_MIX_ENUMINFO	__OSSIOWR('X', 8, oss_mixer_enuminfo)
+#define	SNDCTL_MIDIINFO		__OSSIO('X', 9)
+#define	SNDCTL_MIXERINFO	__OSSIOWR('X', 10, oss_mixerinfo)
+#define	SNDCTL_CARDINFO		__OSSIOWR('X', 11, oss_card_info)
+#define	SNDCTL_ENGINEINFO	__OSSIOWR('X', 12, oss_audioinfo)
+#define	SNDCTL_AUDIOINFO_EX	__OSSIOWR('X', 13, oss_audioinfo)
+#define	SNDCTL_MIX_DESCRIPTION	__OSSIOWR('X', 14, oss_mixer_enuminfo)
+
+/* ioctl codes 'X', 200-255 are reserved for internal use */
+
+/*
+ * Few more "globally" available ioctl calls.
+ */
+#define	SNDCTL_SETSONG		__OSSIOW('Y', 2, oss_longname_t)
+#define	SNDCTL_GETSONG		__OSSIOR('Y', 2, oss_longname_t)
+#define	SNDCTL_SETNAME		__OSSIOW('Y', 3, oss_longname_t)
+#define	SNDCTL_SETLABEL		__OSSIOW('Y', 4, oss_label_t)
+#define	SNDCTL_GETLABEL		__OSSIOR('Y', 4, oss_label_t)
+
+/*
+ * IOCTL commands for /dev/dsp
+ */
+#define	SNDCTL_DSP_HALT		__OSSIO('P', 0)
+#define	SNDCTL_DSP_RESET	SNDCTL_DSP_HALT /* Old name */
+#define	SNDCTL_DSP_SYNC		__OSSIO('P', 1)
+#define	SNDCTL_DSP_SPEED	__OSSIOWR('P', 2, int)
+
+#define	SNDCTL_DSP_STEREO	__OSSIOWR('P', 3, int)	/* OBSOLETE */
+
+#define	SNDCTL_DSP_GETBLKSIZE	__OSSIOWR('P', 4, int)
+#define	SNDCTL_DSP_SAMPLESIZE	SNDCTL_DSP_SETFMT
+#define	SNDCTL_DSP_CHANNELS	__OSSIOWR('P', 6, int)
+#define	SNDCTL_DSP_POST		__OSSIO('P', 8)
+#define	SNDCTL_DSP_SUBDIVIDE	__OSSIOWR('P', 9, int)
+#define	SNDCTL_DSP_SETFRAGMENT	__OSSIOWR('P', 10, int)
+
+#define	SNDCTL_DSP_GETFMTS	__OSSIOR('P', 11, int)	/* Returns a mask */
+#define	SNDCTL_DSP_SETFMT	__OSSIOWR('P', 5, int)	/* Selects ONE fmt */
+
+#define	SNDCTL_DSP_GETOSPACE	__OSSIOR('P', 12, audio_buf_info)
+#define	SNDCTL_DSP_GETISPACE	__OSSIOR('P', 13, audio_buf_info)
+#define	SNDCTL_DSP_GETCAPS	__OSSIOR('P', 15, int)
+
+#define	SNDCTL_DSP_GETTRIGGER	__OSSIOR('P', 16, int)
+#define	SNDCTL_DSP_SETTRIGGER	__OSSIOW('P', 16, int)
+
+#define	SNDCTL_DSP_GETIPTR	__OSSIOR('P', 17, count_info)
+#define	SNDCTL_DSP_GETOPTR	__OSSIOR('P', 18, count_info)
+
+#define	SNDCTL_DSP_SETSYNCRO	__OSSIO('P', 21)
+#define	SNDCTL_DSP_SETDUPLEX	__OSSIO('P', 22)
+
+#define	SNDCTL_DSP_PROFILE	__OSSIOW('P', 23, int)   /* OBSOLETE */
+#define	APF_NORMAL	0	/* Normal applications */
+#define	APF_NETWORK	1	/* Underruns caused by "external" delay */
+#define	APF_CPUINTENS	2	/* Underruns caused by "overheating" the CPU */
+
+
+#define	SNDCTL_DSP_GETODELAY	__OSSIOR('P', 23, int)
+
+#define	SNDCTL_DSP_GETPLAYVOL	__OSSIOR('P', 24, int)
+#define	SNDCTL_DSP_SETPLAYVOL	__OSSIOWR('P', 24, int)
+#define	SNDCTL_DSP_GETERROR	__OSSIOR('P', 25, audio_errinfo)
+
+#define	SNDCTL_DSP_READCTL	__OSSIOWR('P', 26, oss_digital_control)
+#define	SNDCTL_DSP_WRITECTL	__OSSIOWR('P', 27, oss_digital_control)
+
+#define	SNDCTL_DSP_SYNCGROUP	__OSSIOWR('P', 28, oss_syncgroup)
+#define	SNDCTL_DSP_SYNCSTART	__OSSIOW('P', 29, int)
+
+#define	SNDCTL_DSP_COOKEDMODE	__OSSIOW('P', 30, int)
+
+#define	SNDCTL_DSP_SILENCE	__OSSIO('P', 31)
+#define	SNDCTL_DSP_SKIP		__OSSIO('P', 32)
+
+#define	SNDCTL_DSP_HALT_INPUT	__OSSIO('P', 33)
+#define	SNDCTL_DSP_RESET_INPUT	SNDCTL_DSP_HALT_INPUT   /* Old name */
+#define	SNDCTL_DSP_HALT_OUTPUT	__OSSIO('P', 34)
+#define	SNDCTL_DSP_RESET_OUTPUT	SNDCTL_DSP_HALT_OUTPUT  /* Old name */
+
+#define	SNDCTL_DSP_LOW_WATER	__OSSIOW('P', 34, int)
+
+#define	SNDCTL_DSP_CURRENT_IPTR	__OSSIOR('P', 35, oss_count_t)
+#define	SNDCTL_DSP_CURRENT_OPTR	__OSSIOR('P', 36, oss_count_t)
+
+#define	SNDCTL_DSP_GET_RECSRC_NAMES	__OSSIOR('P', 37, oss_mixer_enuminfo)
+#define	SNDCTL_DSP_GET_RECSRC	__OSSIOR('P', 38, int)
+#define	SNDCTL_DSP_SET_RECSRC	__OSSIOWR('P', 38, int)
+
+#define	SNDCTL_DSP_GET_PLAYTGT_NAMES	__OSSIOR('P', 39, oss_mixer_enuminfo)
+#define	SNDCTL_DSP_GET_PLAYTGT	__OSSIOR('P', 40, int)
+#define	SNDCTL_DSP_SET_PLAYTGT	__OSSIOWR('P', 40, int)
+#define	SNDCTL_DSP_GETRECVOL	__OSSIOR('P', 41, int)
+#define	SNDCTL_DSP_SETRECVOL	__OSSIOWR('P', 41, int)
+
+#define	SNDCTL_DSP_GET_CHNORDER	__OSSIOR('P', 42, unsigned long long)
+#define	SNDCTL_DSP_SET_CHNORDER	__OSSIOWR('P', 42, unsigned long long)
+
+#define	SNDCTL_DSP_GETIPEAKS	__OSSIOR('P', 43, oss_peaks_t)
+#define	SNDCTL_DSP_GETOPEAKS	__OSSIOR('P', 44, oss_peaks_t)
+
+#define	SNDCTL_DSP_POLICY	__OSSIOW('P', 45, int)    /* See the manual */
+
+#define	SNDCTL_DSP_GETCHANNELMASK	__OSSIOWR('P', 64, int)
+#define	SNDCTL_DSP_BIND_CHANNEL	__OSSIOWR('P', 65, int)
+
+/*
+ * These definitions are here for the benefit of compiling application
+ * code.  Most of these are NOT implemented in the Solaris code,
+ * however.  This is the older 3.x OSS API, and only the master input and
+ * output levels are actually supported.
+ */
+#define	SOUND_MIXER_NRDEVICES	28
+#define	SOUND_MIXER_VOLUME	0
+#define	SOUND_MIXER_BASS	1
+#define	SOUND_MIXER_TREBLE	2
+#define	SOUND_MIXER_SYNTH	3
+#define	SOUND_MIXER_PCM		4
+#define	SOUND_MIXER_SPEAKER	5
+#define	SOUND_MIXER_LINE	6
+#define	SOUND_MIXER_MIC		7
+#define	SOUND_MIXER_CD		8
+#define	SOUND_MIXER_IMIX	9	/*  Recording monitor  */
+#define	SOUND_MIXER_ALTPCM	10
+#define	SOUND_MIXER_RECLEV	11	/* Recording level */
+#define	SOUND_MIXER_IGAIN	12	/* Input gain */
+#define	SOUND_MIXER_OGAIN	13	/* Output gain */
+#define	SOUND_MIXER_LINE1	14	/* Input source 1  (aux1) */
+#define	SOUND_MIXER_LINE2	15	/* Input source 2  (aux2) */
+#define	SOUND_MIXER_LINE3	16	/* Input source 3  (line) */
+#define	SOUND_MIXER_DIGITAL1	17	/* Digital I/O 1 */
+#define	SOUND_MIXER_DIGITAL2	18	/* Digital I/O 2 */
+#define	SOUND_MIXER_DIGITAL3	19	/* Digital I/O 3 */
+#define	SOUND_MIXER_PHONE	20	/* Phone */
+#define	SOUND_MIXER_MONO	21	/* Mono Output */
+#define	SOUND_MIXER_VIDEO	22	/* Video/TV (audio) in */
+#define	SOUND_MIXER_RADIO	23	/* Radio in */
+#define	SOUND_MIXER_DEPTH	24	/* Surround depth */
+#define	SOUND_MIXER_REARVOL	25	/* Rear/Surround speaker vol */
+#define	SOUND_MIXER_CENTERVOL	26	/* Center/LFE speaker vol */
+#define	SOUND_MIXER_SIDEVOL	27	/* Side-Surround (8speaker) vol */
+#define	SOUND_MIXER_SURRVOL	SOUND_MIXER_SIDEVOL
+#define	SOUND_ONOFF_MIN		28
+#define	SOUND_ONOFF_MAX		30
+#define	SOUND_MIXER_NONE	31
+
+#define	SOUND_MIXER_RECSRC	0xff	/* Recording sources */
+#define	SOUND_MIXER_DEVMASK	0xfe	/* Supported devices */
+#define	SOUND_MIXER_RECMASK	0xfd	/* Recording sources */
+#define	SOUND_MIXER_CAPS	0xfc	/* Mixer capabilities (do not use) */
+#define	SOUND_MIXER_STEREODEVS	0xfb	/* Mixer channels supporting stereo */
+#define	SOUND_MIXER_OUTSRC	0xfa
+#define	SOUND_MIXER_OUTMASK	0xf9
+
+#define	SOUND_MIXER_ENHANCE	SOUND_MIXER_NONE
+#define	SOUND_MIXER_MUTE	SOUND_MIXER_NONE
+#define	SOUND_MIXER_LOUD	SOUND_MIXER_NONE
+
+#define	SOUND_MASK_VOLUME	(1 << SOUND_MIXER_VOLUME)
+#define	SOUND_MASK_BASS		(1 << SOUND_MIXER_BASS)
+#define	SOUND_MASK_TREBLE	(1 << SOUND_MIXER_TREBLE)
+#define	SOUND_MASK_SYNTH	(1 << SOUND_MIXER_SYNTH)
+#define	SOUND_MASK_PCM		(1 << SOUND_MIXER_PCM)
+#define	SOUND_MASK_SPEAKER	(1 << SOUND_MIXER_SPEAKER)
+#define	SOUND_MASK_LINE		(1 << SOUND_MIXER_LINE)
+#define	SOUND_MASK_MIC		(1 << SOUND_MIXER_MIC)
+#define	SOUND_MASK_CD		(1 << SOUND_MIXER_CD)
+#define	SOUND_MASK_IMIX		(1 << SOUND_MIXER_IMIX)
+#define	SOUND_MASK_ALTPCM	(1 << SOUND_MIXER_ALTPCM)
+#define	SOUND_MASK_RECLEV	(1 << SOUND_MIXER_RECLEV)
+#define	SOUND_MASK_IGAIN	(1 << SOUND_MIXER_IGAIN)
+#define	SOUND_MASK_OGAIN	(1 << SOUND_MIXER_OGAIN)
+#define	SOUND_MASK_LINE1	(1 << SOUND_MIXER_LINE1)
+#define	SOUND_MASK_LINE2	(1 << SOUND_MIXER_LINE2)
+#define	SOUND_MASK_LINE3	(1 << SOUND_MIXER_LINE3)
+#define	SOUND_MASK_DIGITAL1	(1 << SOUND_MIXER_DIGITAL1)
+#define	SOUND_MASK_DIGITAL2	(1 << SOUND_MIXER_DIGITAL2)
+#define	SOUND_MASK_DIGITAL3	(1 << SOUND_MIXER_DIGITAL3)
+#define	SOUND_MASK_MONO		(1 << SOUND_MIXER_MONO)
+#define	SOUND_MASK_PHONE	(1 << SOUND_MIXER_PHONE)
+#define	SOUND_MASK_RADIO	(1 << SOUND_MIXER_RADIO)
+#define	SOUND_MASK_VIDEO	(1 << SOUND_MIXER_VIDEO)
+#define	SOUND_MASK_DEPTH	(1 << SOUND_MIXER_DEPTH)
+#define	SOUND_MASK_REARVOL	(1 << SOUND_MIXER_REARVOL)
+#define	SOUND_MASK_CENTERVOL	(1 << SOUND_MIXER_CENTERVOL)
+#define	SOUND_MASK_SIDEVOL	(1 << SOUND_MIXER_SIDEVOL)
+#define	SOUND_MASK_SURRVOL	SOUND_MASK_SIDEVOL
+#define	SOUND_MASK_MUTE		(1 << SOUND_MIXER_MUTE)
+#define	SOUND_MASK_ENHANCE	(1 << SOUND_MIXER_ENHANCE)
+#define	SOUND_MASK_LOUD		(1 << SOUND_MIXER_LOUD)
+
+/*
+ * Again, DO NOT USE the following two macros.  They are here for SOURCE
+ * COMPATIBILITY ONLY.
+ */
+#define	SOUND_DEVICE_LABELS	{					   \
+	"Vol  ", "Bass ", "Treble", "Synth", "Pcm  ", "Speaker ", "Line ", \
+	"Mic  ", "CD   ", "Mix  ", "Pcm2 ", "Rec  ", "IGain", "OGain",     \
+	"Aux1", "Aux2", "Aux3", "Digital1", "Digital2", "Digital3",	   \
+	"Phone", "Mono", "Video", "Radio", "Depth",		           \
+	"Rear", "Center", "Side" }
+
+#define	SOUND_DEVICE_NAMES { 						\
+	"vol", "bass", "treble", "synth", "pcm", "speaker", "line",	\
+	"mic", "cd", "mix", "pcm2", "rec", "igain", "ogain",		\
+	"aux1", "aux2", "aux3", "dig1", "dig2", "dig3",			\
+	"phone", "mono", "video", "radio", "depth",			\
+	"rear", "center", "side" }
+
+#define	MIXER_READ(dev)			__OSSIOR('M', dev, int)
+#define	MIXER_WRITE(dev)		__OSSIOWR('M', dev, int)
+#define	SOUND_MIXER_INFO		__OSSIOR('M', 101, mixer_info)
+#define	OSS_GETVERSION			__OSSIOR('M', 118, int)
+
+/*
+ * These macros are useful for some applications.  They are implemented
+ * as soft values for the application, and do not affect real hardware.
+ */
+#define	SOUND_MIXER_READ_VOLUME		MIXER_READ(SOUND_MIXER_VOLUME)
+#define	SOUND_MIXER_READ_OGAIN		MIXER_READ(SOUND_MIXER_OGAIN)
+#define	SOUND_MIXER_READ_PCM		MIXER_READ(SOUND_MIXER_PCM)
+#define	SOUND_MIXER_READ_IGAIN		MIXER_READ(SOUND_MIXER_IGAIN)
+#define	SOUND_MIXER_READ_RECLEV		MIXER_READ(SOUND_MIXER_RECLEV)
+#define	SOUND_MIXER_READ_RECSRC		MIXER_READ(SOUND_MIXER_RECSRC)
+#define	SOUND_MIXER_READ_DEVMASK	MIXER_READ(SOUND_MIXER_DEVMASK)
+#define	SOUND_MIXER_READ_RECMASK	MIXER_READ(SOUND_MIXER_RECMASK)
+#define	SOUND_MIXER_READ_CAPS		MIXER_READ(SOUND_MIXER_CAPS)
+#define	SOUND_MIXER_READ_STEREODEVS	MIXER_READ(SOUND_MIXER_STEREODEVS)
+#define	SOUND_MIXER_READ_RECGAIN	__OSSIOR('M', 119, int)
+#define	SOUND_MIXER_READ_MONGAIN	__OSSIOR('M', 120, int)
+
+#define	SOUND_MIXER_WRITE_VOLUME	MIXER_WRITE(SOUND_MIXER_VOLUME)
+#define	SOUND_MIXER_WRITE_OGAIN		MIXER_WRITE(SOUND_MIXER_OGAIN)
+#define	SOUND_MIXER_WRITE_PCM		MIXER_WRITE(SOUND_MIXER_PCM)
+#define	SOUND_MIXER_WRITE_IGAIN		MIXER_WRITE(SOUND_MIXER_IGAIN)
+#define	SOUND_MIXER_WRITE_RECLEV	MIXER_WRITE(SOUND_MIXER_RECLEV)
+#define	SOUND_MIXER_WRITE_RECSRC	MIXER_WRITE(SOUND_MIXER_RECSRC)
+#define	SOUND_MIXER_WRITE_RECGAIN	__OSSIOWR('M', 119, int)
+#define	SOUND_MIXER_WRITE_MONGAIN	__OSSIOWR('M', 120, int)
+
+/*
+ * These macros are here for source compatibility.  They intentionally don't
+ * map to any real hardware.  NOT SUPPORTED!
+ */
+#define	SOUND_MIXER_READ_BASS		MIXER_READ(SOUND_MIXER_BASS)
+#define	SOUND_MIXER_READ_TREBLE		MIXER_READ(SOUND_MIXER_TREBLE)
+#define	SOUND_MIXER_READ_SYNTH		MIXER_READ(SOUND_MIXER_SYNTH)
+#define	SOUND_MIXER_READ_SPEAKER	MIXER_READ(SOUND_MIXER_SPEAKER)
+#define	SOUND_MIXER_READ_LINE		MIXER_READ(SOUND_MIXER_LINE)
+#define	SOUND_MIXER_READ_MIC		MIXER_READ(SOUND_MIXER_MIC)
+#define	SOUND_MIXER_READ_CD		MIXER_READ(SOUND_MIXER_CD)
+#define	SOUND_MIXER_READ_IMIX		MIXER_READ(SOUND_MIXER_IMIX)
+#define	SOUND_MIXER_READ_ALTPCM		MIXER_READ(SOUND_MIXER_ALTPCM)
+#define	SOUND_MIXER_READ_LINE1		MIXER_READ(SOUND_MIXER_LINE1)
+#define	SOUND_MIXER_READ_LINE2		MIXER_READ(SOUND_MIXER_LINE2)
+#define	SOUND_MIXER_READ_LINE3		MIXER_READ(SOUND_MIXER_LINE3)
+
+#define	SOUND_MIXER_WRITE_BASS		MIXER_WRITE(SOUND_MIXER_BASS)
+#define	SOUND_MIXER_WRITE_TREBLE	MIXER_WRITE(SOUND_MIXER_TREBLE)
+#define	SOUND_MIXER_WRITE_SYNTH		MIXER_WRITE(SOUND_MIXER_SYNTH)
+#define	SOUND_MIXER_WRITE_SPEAKER	MIXER_WRITE(SOUND_MIXER_SPEAKER)
+#define	SOUND_MIXER_WRITE_LINE		MIXER_WRITE(SOUND_MIXER_LINE)
+#define	SOUND_MIXER_WRITE_MIC		MIXER_WRITE(SOUND_MIXER_MIC)
+#define	SOUND_MIXER_WRITE_CD		MIXER_WRITE(SOUND_MIXER_CD)
+#define	SOUND_MIXER_WRITE_IMIX		MIXER_WRITE(SOUND_MIXER_IMIX)
+#define	SOUND_MIXER_WRITE_ALTPCM	MIXER_WRITE(SOUND_MIXER_ALTPCM)
+#define	SOUND_MIXER_WRITE_LINE1		MIXER_WRITE(SOUND_MIXER_LINE1)
+#define	SOUND_MIXER_WRITE_LINE2		MIXER_WRITE(SOUND_MIXER_LINE2)
+#define	SOUND_MIXER_WRITE_LINE3		MIXER_WRITE(SOUND_MIXER_LINE3)
+
+/*
+ * Audio encoding types (Note! U8=8 and S16_LE=16 for compatibility)
+ */
+#define	AFMT_QUERY	0x00000000	/* Return current fmt */
+#define	AFMT_MU_LAW	0x00000001
+#define	AFMT_A_LAW	0x00000002
+#define	AFMT_IMA_ADPCM	0x00000004
+#define	AFMT_U8		0x00000008
+#define	AFMT_S16_LE	0x00000010
+#define	AFMT_S16_BE	0x00000020
+#define	AFMT_S8		0x00000040
+#define	AFMT_U16_LE	0x00000080
+#define	AFMT_U16_BE	0x00000100
+#define	AFMT_MPEG	0x00000200	/* NOT SUPPORTED: MPEG (2) audio */
+#define	AFMT_AC3	0x00000400	/* NOT SUPPORTED: AC3 compressed */
+#define	AFMT_VORBIS	0x00000800	/* NOT SUPPORTED: Ogg Vorbis */
+#define	AFMT_S32_LE	0x00001000
+#define	AFMT_S32_BE	0x00002000
+#define	AFMT_FLOAT	0x00004000	/* NOT SUPPORTED: IEEE double float */
+#define	AFMT_S24_LE	0x00008000	/* LSB aligned in 32 bit word */
+#define	AFMT_S24_BE	0x00010000	/* LSB aligned in 32 bit word */
+#define	AFMT_SPDIF_RAW	0x00020000	/* NOT SUPPORTED: Raw S/PDIF frames */
+#define	AFMT_S24_PACKED	0x00040000	/* 24 bit packed little endian */
+/*
+ * Some big endian/little endian handling macros (native endian and
+ * opposite endian formats).
+ */
+#if defined(_BIG_ENDIAN)
+#define	AFMT_S16_NE	AFMT_S16_BE
+#define	AFMT_U16_NE	AFMT_U16_BE
+#define	AFMT_S32_NE	AFMT_S32_BE
+#define	AFMT_S24_NE	AFMT_S24_BE
+#define	AFMT_S16_OE	AFMT_S16_LE
+#define	AFMT_S32_OE	AFMT_S32_LE
+#define	AFMT_S24_OE	AFMT_S24_LE
+#else
+#define	AFMT_S16_NE	AFMT_S16_LE
+#define	AFMT_U16_NE	AFMT_U16_LE
+#define	AFMT_S32_NE	AFMT_S32_LE
+#define	AFMT_S24_NE	AFMT_S24_LE
+#define	AFMT_S16_OE	AFMT_S16_BE
+#define	AFMT_S32_OE	AFMT_S32_BE
+#define	AFMT_S24_OE	AFMT_S24_BE
+#endif
+
+/*
+ * SNDCTL_DSP_GETCAPS bits
+ */
+#define	PCM_CAP_REVISION	0x000000ff	/* Revision level (0 to 255) */
+#define	PCM_CAP_DUPLEX		0x00000100	/* Full duplex rec/play */
+#define	PCM_CAP_REALTIME	0x00000200	/* NOT SUPPORTED */
+#define	PCM_CAP_BATCH		0x00000400	/* NOT SUPPORTED */
+#define	PCM_CAP_COPROC		0x00000800	/* NOT SUPPORTED */
+#define	PCM_CAP_TRIGGER		0x00001000	/* Supports SETTRIGGER */
+#define	PCM_CAP_MMAP		0x00002000	/* Supports mmap() */
+#define	PCM_CAP_MULTI		0x00004000	/* Supports multiple open */
+#define	PCM_CAP_BIND		0x00008000	/* Supports channel binding */
+#define	PCM_CAP_INPUT		0x00010000	/* Supports recording */
+#define	PCM_CAP_OUTPUT		0x00020000	/* Supports playback */
+#define	PCM_CAP_VIRTUAL		0x00040000	/* Virtual device */
+#define	PCM_CAP_ANALOGOUT	0x00100000	/* NOT SUPPORTED */
+#define	PCM_CAP_ANALOGIN	0x00200000	/* NOT SUPPORTED */
+#define	PCM_CAP_DIGITALOUT	0x00400000	/* NOT SUPPORTED */
+#define	PCM_CAP_DIGITALIN	0x00800000	/* NOT SUPPORTED */
+#define	PCM_CAP_ADMASK		0x00f00000	/* NOT SUPPORTED */
+#define	PCM_CAP_SHADOW		0x01000000	/* "Shadow" device */
+#define	PCM_CAP_CH_MASK		0x06000000	/* See DSP_CH_MASK below */
+#define	PCM_CAP_HIDDEN		0x08000000	/* NOT SUPPORTED */
+#define	PCM_CAP_FREERATE	0x10000000
+#define	PCM_CAP_MODEM		0x20000000	/* NOT SUPPORTED */
+#define	PCM_CAP_DEFAULT		0x40000000	/* "Default" device */
+
+/*
+ * Preferred channel usage. These bits can be used to give
+ * recommendations to the application. Used by few drivers.  For
+ * example if ((caps & DSP_CH_MASK) == DSP_CH_MONO) means that the
+ * device works best in mono mode. However it doesn't necessarily mean
+ * that the device cannot be used in stereo. These bits should only be
+ * used by special applications such as multi track hard disk
+ * recorders to find out the initial setup. However the user should be
+ * able to override this selection.
+ *
+ * To find out which modes are actually supported the application
+ * should try to select them using SNDCTL_DSP_CHANNELS.
+ */
+#define	DSP_CH_MASK		0x06000000	/* Mask */
+#define	DSP_CH_ANY		0x00000000	/* No preferred mode */
+#define	DSP_CH_MONO		0x02000000
+#define	DSP_CH_STEREO		0x04000000
+#define	DSP_CH_MULTI		0x06000000	/* More than two channels */
+
+
+/*
+ * The PCM_CAP_* capability names used to be known as DSP_CAP_*, so
+ * it's necessary to define the older names too.
+ */
+#define	DSP_CAP_ADMASK		PCM_CAP_ADMASK
+#define	DSP_CAP_ANALOGIN	PCM_CAP_ANALOGIN
+#define	DSP_CAP_ANALOGOUT	PCM_CAP_ANALOGOUT
+#define	DSP_CAP_BATCH		PCM_CAP_BATCH
+#define	DSP_CAP_BIND		PCM_CAP_BIND
+#define	DSP_CAP_COPROC		PCM_CAP_COPROC
+#define	DSP_CAP_DEFAULT		PCM_CAP_DEFAULT
+#define	DSP_CAP_DIGITALIN	PCM_CAP_DIGITALIN
+#define	DSP_CAP_DIGITALOUT	PCM_CAP_DIGITALOUT
+#define	DSP_CAP_DUPLEX		PCM_CAP_DUPLEX
+#define	DSP_CAP_FREERATE	PCM_CAP_FREERATE
+#define	DSP_CAP_HIDDEN		PCM_CAP_HIDDEN
+#define	DSP_CAP_INPUT		PCM_CAP_INPUT
+#define	DSP_CAP_MMAP		PCM_CAP_MMAP
+#define	DSP_CAP_MODEM		PCM_CAP_MODEM
+#define	DSP_CAP_MULTI		PCM_CAP_MULTI
+#define	DSP_CAP_OUTPUT		PCM_CAP_OUTPUT
+#define	DSP_CAP_REALTIME	PCM_CAP_REALTIME
+#define	DSP_CAP_REVISION	PCM_CAP_REVISION
+#define	DSP_CAP_SHADOW		PCM_CAP_SHADOW
+#define	DSP_CAP_TRIGGER		PCM_CAP_TRIGGER
+#define	DSP_CAP_VIRTUAL		PCM_CAP_VIRTUAL
+
+/*
+ * SNDCTL_DSP_GETTRIGGER and SNDCTL_DSP_SETTRIGGER
+ */
+#define	PCM_ENABLE_INPUT	0x00000001
+#define	PCM_ENABLE_OUTPUT	0x00000002
+
+/*
+ * SNDCTL_DSP_BIND_CHANNEL
+ */
+#define	DSP_BIND_QUERY		0x00000000
+#define	DSP_BIND_FRONT		0x00000001
+#define	DSP_BIND_SURR		0x00000002
+#define	DSP_BIND_CENTER_LFE	0x00000004
+#define	DSP_BIND_HANDSET	0x00000008
+#define	DSP_BIND_MIC		0x00000010
+#define	DSP_BIND_MODEM1		0x00000020
+#define	DSP_BIND_MODEM2		0x00000040
+#define	DSP_BIND_I2S		0x00000080
+#define	DSP_BIND_SPDIF		0x00000100
+#define	DSP_BIND_REAR		0x00000200
+
+/*
+ * SOUND_MIXER_READ_CAPS
+ */
+#define	SOUND_CAP_EXCL_INPUT	0x00000001
+#define	SOUND_CAP_NOLEGACY	0x00000004
+#define	SOUND_CAP_NORECSRC	0x00000008
+
+/*
+ * The following ioctl is for internal use only -- it is used to
+ * coordinate /dev/sndstat numbering with file names in /dev/sound.
+ * Applications must not use it.  (This is duplicated in sys/audioio.h
+ * as well.)
+ */
+#define	SNDCTL_SUN_SEND_NUMBER	__OSSIOW('X', 200, int)
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* _SYS_AUDIO_OSS_H */
diff --git a/usr/src/uts/common/sys/audio/audio_src.h b/usr/src/uts/common/sys/audio/audio_src.h
deleted file mode 100644
index 55ee5071dc..0000000000
--- a/usr/src/uts/common/sys/audio/audio_src.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 1999-2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the public interfaces for the audio mixer
- * sample rate conversion routines. Only the audio mixer and sample rate
- * conversion routines may include this header file.
- *
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
-#ifndef	_SYS_AUDIO_SRC_H
-#define	_SYS_AUDIO_SRC_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#ifdef _KERNEL
-
-#define	AM_SRC_VERSION		AM_SRC_VERSION2
-#define	AM_SRC_VERSION2		2
-
-/*
- * The handle used by the mixer and the sample rate conversion routine.
- */
-typedef struct src_handle *srchdl_t;
-
-/*
- * am_ad_src_entry_t	- Audio Driver sample rate conversion routines
- */
-struct am_ad_src_entry {
-	int		ad_version;
-	size_t		(*ad_src_init)(srchdl_t handle, int dir);
-	void		(*ad_src_exit)(srchdl_t handle, int dir);
-	int		(*ad_src_update)(srchdl_t handle,
-			    audio_prinfo_t *ch_prinfo,
-			    audio_prinfo_t *hw_prinfo,
-			    void *src_info, int dir);
-	int		(*ad_src_adjust)(srchdl_t handle, int dir,
-			    int samples);
-	int		*(*ad_src_convert)(srchdl_t handle, int channels,
-			    int dir, int *src, int *ptr1, int *ptr2,
-			    int *samples);
-	size_t		(*ad_src_size)(srchdl_t handle, audio_prinfo_t *prinfo,
-			    int dir, int samples, int hw_channels);
-};
-typedef struct am_ad_src_entry am_ad_src_entry_t;
-
-/*
- * am_ad_sample_rates_t		- supported sample rates
- */
-struct am_ad_sample_rates {
-	int		ad_limits;	/* 0 if sample rates not limits */
-	uint_t		*ad_srs;	/* NULL term. list of sample rates */
-};
-typedef struct am_ad_sample_rates am_ad_sample_rates_t;
-
-/* am_ad_ample_rates.ad_limits */
-#define	MIXER_SRS_FLAG_SR_NOT_LIMITS	0x00000000u
-						/* samp rates not limits */
-#define	MIXER_SRS_FLAG_SR_LIMITS	MIXER_SR_LIMITS
-						/* samp rates set limits */
-
-/*
- * Audio Mixer Audio Driver Miscellaneous Routines
- */
-void *am_get_src_data(srchdl_t handle, int dir);
-void am_set_src_data(srchdl_t handle, int dir, void *data);
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_SRC_H */
diff --git a/usr/src/uts/common/sys/audio/audio_support.h b/usr/src/uts/common/sys/audio/audio_support.h
index 46e7ecdf78..d6e5504156 100644
--- a/usr/src/uts/common/sys/audio/audio_support.h
+++ b/usr/src/uts/common/sys/audio/audio_support.h
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,12 +19,11 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  *
- * This header file defines the public interfaces for the audio support
- * module. These definitions are available for use by all audio modules
- * and drivers.
+ * This header file defines a legacy interface for audio drivers.
+ * It should not be used in any code.
  *
  * CAUTION: This header file has not gone through a formal review process.
  *	Thus its commitment level is very low and may change or be removed
@@ -35,8 +33,6 @@
 #ifndef	_SYS_AUDIO_SUPPORT_H
 #define	_SYS_AUDIO_SUPPORT_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 #ifdef	__cplusplus
 extern "C" {
 #endif
@@ -50,121 +46,23 @@ extern "C" {
 #define	AUDIO_FAILURE			(-1)
 
 /*
- * Miscellaneous defines.
- */
-#define	AUDIO_PRECISION_SHIFT		3
-#define	AUDIO_NO_CHANNEL		(-1)
-#define	AUDIO_TOGGLE(X)			(X) ^= 1
-
-/*
  * Opaque handle used by all audio drivers to reference the audio
  * support module and all audio personality modules.
  */
 typedef struct audio_handle *audiohdl_t;
 
 /*
- * Opaque handle used by all audio personality modules to reference the
- * audio task queue.
- */
-typedef struct audio_taskq *audio_taskq_t;
-
-/*
- * audio_i_state	- This structure is used to hold state information
- *			  between M_IOCTL and M_IOCDATA messages from the
- *			  STREAMS head.
- */
-struct audio_i_state {
-	long	ais_command;	/* the M_IOCDATA command to execute next */
-	caddr_t	ais_address;	/* address to M_COPYOUT/M_COPYIN data from/to */
-	caddr_t	ais_address2;	/* address to M_COPYOUT/M_COPYIN data from/to */
-};
-typedef struct audio_i_state audio_i_state_t;
-
-/*
- * audio_sup_reg_data	- This structure is used to provide registration
- *			  data from the audio driver to the audiosup module.
- */
-struct audio_sup_reg_data {
-	int		asrd_version;	/* version of this data structure */
-	char		*asrd_key;	/* unique string used to ID drvr inst */
-};
-typedef struct audio_sup_reg_data audio_sup_reg_data_t;
-
-_NOTE(SCHEME_PROTECTS_DATA("private data", audio_sup_reg_data::asrd_key))
-
-#define	AUDIOSUP_VERSION	AUDIOSUP_VERSION1
-#define	AUDIOSUP_VERSION1	1
-
-/*
  * Audio Support Module Entry Point Routines
  */
-audiohdl_t audio_sup_register(dev_info_t *dip, audio_sup_reg_data_t *data);
+audiohdl_t audio_sup_register(dev_info_t *dip);
 int audio_sup_unregister(audiohdl_t handle);
-int audio_sup_open(queue_t *q, dev_t *devp, int flag, int sflags,
-    cred_t *credp);
-int audio_sup_close(queue_t *q, int flag, cred_t *credp);
-int audio_sup_getinfo(dev_info_t *dip, ddi_info_cmd_t infocmd,
-    void *arg, void **result);
-int audio_sup_restore_state(audiohdl_t handle, audio_device_type_e device,
-    int dir);
-int audio_sup_save_state(audiohdl_t handle, audio_device_type_e device,
-    int dir);
-
-#define	AUDIO_ALL_DEVICES	UNDEFINED
-
-int audio_sup_rput(queue_t *q, mblk_t *mp);
-int audio_sup_rsvc(queue_t *q);
-int audio_sup_wput(queue_t *q, mblk_t *mp);
-int audio_sup_wsvc(queue_t *q);
-
-/*
- * Audio Support Module Entry Point Routines
- *	CAUTION: These routines will be removed from the next
- *		release of Solaris. Migrate to audio_sup_register()
- *		and audio_sup_unregister().
- */
-audiohdl_t audio_sup_attach(dev_info_t *dip, ddi_attach_cmd_t cmd);
-int audio_sup_detach(audiohdl_t handle, ddi_detach_cmd_t cmd);
-
-/*
- * Audio Support Module STREAMS Private Data Routines
- */
-void audio_sup_set_qptr(queue_t *q, dev_t dev, void *data);
-void audio_sup_free_qptr(queue_t *q);
-dev_t audio_sup_get_qptr_dev(queue_t *q);
-void *audio_sup_get_qptr_data(queue_t *q);
-int audio_sup_get_qptr_instance(queue_t *q);
-
-/*
- * Audio Support Module Minor Routines
- */
-int audio_sup_get_max_chs(audiohdl_t handle);
-int audio_sup_get_minors_per_inst(audiohdl_t handle);
-int audio_sup_construct_minor(audiohdl_t handle, audio_device_type_e type);
-int audio_sup_devt_to_instance(dev_t devt);
+void audio_sup_restore_state(audiohdl_t handle);
 
 /*
  * Audio Support Module Miscellaneous Routines
  */
-dev_info_t *audio_sup_get_dip(audiohdl_t handle);
 void *audio_sup_get_private(audiohdl_t handle);
 void audio_sup_set_private(audiohdl_t handle, void *private);
-/*PRINTFLIKE3*/
-extern void audio_sup_log(audiohdl_t handle, uint_t level, char *fmt, ...)
-    __KPRINTFLIKE(3);
-int audio_sup_update_persist_key(dev_info_t *dip, char *new_key, int sleep);
-
-
-#ifdef _SYSCALL32
-/* ILP32 view of the audio_channel structure */
-struct audio_channel32 {
-	pid32_t			pid;		/* process ID */
-	uint32_t		ch_number;	/* channel number */
-	int32_t			dev_type;	/* device type */
-	size32_t		info_size;	/* size of info structure */
-	caddr32_t		info;		/* pointer to info structure */
-};
-#endif
 
 #endif	/* _KERNEL */
 
diff --git a/usr/src/uts/common/sys/audio/audio_trace.h b/usr/src/uts/common/sys/audio/audio_trace.h
deleted file mode 100644
index 61eeb45ed0..0000000000
--- a/usr/src/uts/common/sys/audio/audio_trace.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-/*
- * This header file defines the public interfaces for audio tracing.
- *
- * CAUTION: This header file has not gone through a formal review process.
- *	Thus its commitment level is very low and may change or be removed
- *	at any time.
- */
-
-#ifndef	_SYS_AUDIO_TRACE_H
-#define	_SYS_AUDIO_TRACE_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#ifdef _KERNEL
-
-/*
- * Trace declarations and defines.
- */
-
-struct audio_trace_buf {
-	uint_t		atb_seq;	/* trace sequence number */
-	char		*atb_comment;	/* trace comment string */
-	uintptr_t	atb_data;	/* data to go with string */
-};
-typedef struct audio_trace_buf audio_trace_buf_t;
-
-#define	AUDIO_TRACE_BUFFER_SIZE		1024
-
-#ifdef DEBUG
-
-#ifndef __lock_lint
-extern audio_trace_buf_t audio_trace_buffer[AUDIO_TRACE_BUFFER_SIZE];
-extern kmutex_t audio_tb_lock;	/* global trace buffer lock */
-extern size_t audio_tb_siz;
-extern int audio_tb_pos;
-extern uint_t audio_tb_seq;
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_tb_lock, audio_tb_pos))
-_NOTE(MUTEX_PROTECTS_DATA(audio_tb_lock, audio_tb_seq))
-_NOTE(MUTEX_PROTECTS_DATA(audio_tb_lock, audio_trace_buffer))
-
-#define	ATRACE(M, D) {							\
-	mutex_enter(&audio_tb_lock);					\
-	audio_trace_buffer[audio_tb_pos].atb_seq = audio_tb_seq++;	\
-	audio_trace_buffer[audio_tb_pos].atb_comment = (M);		\
-	audio_trace_buffer[audio_tb_pos++].atb_data = (uintptr_t)(D);	\
-	if (audio_tb_pos >= audio_tb_siz)				\
-		audio_tb_pos = 0;					\
-	mutex_exit(&audio_tb_lock);					\
-	}
-
-#define	ATRACE_64(M, D)		ATRACE(M, D)
-
-#define	ATRACE_32(M, D) {						\
-	mutex_enter(&audio_tb_lock);					\
-	audio_trace_buffer[audio_tb_pos].atb_seq = audio_tb_seq++;	\
-	audio_trace_buffer[audio_tb_pos].atb_comment = (M);		\
-	audio_trace_buffer[audio_tb_pos++].atb_data = 			\
-	    (uintptr_t)(uint32_t)(D);					\
-	if (audio_tb_pos >= audio_tb_siz)				\
-		audio_tb_pos = 0;					\
-	mutex_exit(&audio_tb_lock);					\
-	}
-
-#define	ATRACE_16(M, D) {						\
-	mutex_enter(&audio_tb_lock);					\
-	audio_trace_buffer[audio_tb_pos].atb_seq = audio_tb_seq++;	\
-	audio_trace_buffer[audio_tb_pos].atb_comment = (M);		\
-	audio_trace_buffer[audio_tb_pos++].atb_data =			\
-	    (uintptr_t)(uint16_t)(D);					\
-	if (audio_tb_pos >= audio_tb_siz)				\
-		audio_tb_pos = 0;					\
-	mutex_exit(&audio_tb_lock);					\
-	}
-
-#define	ATRACE_8(M, D) {						\
-	mutex_enter(&audio_tb_lock);					\
-	audio_trace_buffer[audio_tb_pos].atb_seq = audio_tb_seq++;	\
-	audio_trace_buffer[audio_tb_pos].atb_comment = (M);		\
-	audio_trace_buffer[audio_tb_pos++].atb_data =			\
-	    (uintptr_t)(uint8_t)(D);					\
-	if (audio_tb_pos >= audio_tb_siz)				\
-		audio_tb_pos = 0;					\
-	mutex_exit(&audio_tb_lock);					\
-	}
-#else
-
-#define	ATRACE(M, D)
-#define	ATRACE_32(M, D)
-#define	ATRACE_16(M, D)
-#define	ATRACE_8(M, D)
-
-#endif
-
-#else	/* DEBUG */
-
-#define	ATRACE(M, D)
-#define	ATRACE_32(M, D)
-#define	ATRACE_16(M, D)
-#define	ATRACE_8(M, D)
-
-#endif	/* DEBUG */
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_TRACE_H */
diff --git a/usr/src/uts/common/sys/audio/audiohd.h b/usr/src/uts/common/sys/audio/audiohd.h
deleted file mode 100755
index 81f660c614..0000000000
--- a/usr/src/uts/common/sys/audio/audiohd.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIOHD_H_
-#define	_SYS_AUDIOHD_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#include <sys/types.h>
-#include <sys/thread.h>
-#include <sys/synch.h>
-#include <sys/kstat.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/audio.h>
-#include <sys/audio/audio_support.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-
-
-#define	AUDIOHD_SAMPR5510		5510
-#define	AUDIOHD_SAMPR6620		6620
-#define	AUDIOHD_SAMPR8000		8000
-#define	AUDIOHD_SAMPR9600		9600
-#define	AUDIOHD_SAMPR11025		11025
-#define	AUDIOHD_SAMPR16000		16000
-#define	AUDIOHD_SAMPR18900		18900
-#define	AUDIOHD_SAMPR22050		22050
-#define	AUDIOHD_SAMPR27420		27420
-#define	AUDIOHD_SAMPR32000		32000
-#define	AUDIOHD_SAMPR33075		33075
-#define	AUDIOHD_SAMPR37800		37800
-#define	AUDIOHD_SAMPR44100		44100
-#define	AUDIOHD_SAMPR48000		48000
-
-#define	AUDIOHD_SAMPLER_MAX	AUDIOHD_SAMPR48000
-#define	AUDIOHD_MIN_INTS	32
-#define	AUDIOHD_MAX_INTS	1500
-#define	AUDIOHD_INTS	50
-#define	AUDIOHD_MAX_PRECISION	AUDIO_PRECISION_16
-#define	AUDIOHD_MAX_CHANNELS	AUDIO_CHANNELS_STEREO
-#define	AUDIOHD_MAX_OUT_CHANNELS	32
-#define	AUDIOHD_MAX_IN_CHANNELS		AUDIOHD_MAX_OUT_CHANNELS
-
-#define	AUDIOHD_BSIZE		8192
-#define	AUDIOHD_DEFAULT_SR	8000
-#define	AUDIOHD_DEFAULT_CH	AUDIO_CHANNELS_MONO
-#define	AUDIOHD_DEFAULT_PREC	AUDIO_PRECISION_8
-#define	AUDIOHD_DEFAULT_ENC		AUDIO_ENCODING_ULAW
-#define	AUDIOHD_DEFAULT_PGAIN	(AUDIO_MAX_GAIN * 3 / 4)
-#define	AUDIOHD_DEFAULT_RGAIN	127
-#define	AUDIOHD_DEFAULT_BAL			AUDIO_MID_BALANCE
-#define	AUDIOHD_DEFAULT_MONITOR_GAIN		0
-
-#define	AUDIOHD_DEV_NAME	"SUNW,audiohd"
-#define	AUDIOHD_DEV_CONFIG	"onboard1"
-#define	AUDIOHD_DEV_VERSION	"a"
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIOHD_H_ */
diff --git a/usr/src/uts/common/sys/audio/audioixp.h b/usr/src/uts/common/sys/audio/audioixp.h
deleted file mode 100644
index 3d0d6ed17a..0000000000
--- a/usr/src/uts/common/sys/audio/audioixp.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIOIXP_H_
-#define	_SYS_AUDIOIXP_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/*
- * Header file for the audioixp device driver
- */
-
-/*
- * Values returned by the AUDIO_GETDEV ioctl()
- */
-#define	IXP_DEV_NAME		"SUNW,audioixp"
-#define	IXP_DEV_CONFIG		"onboard1"
-#define	IXP_DEV_VERSION		"a"
-
-/*
- * Driver supported configuration information
- */
-#define	IXP_NAME		"audioixp"
-#define	IXP_MOD_NAME		"mixer audio driver"
-
-#define	IXP_SAMPR8000		(8000)
-#define	IXP_SAMPR11025		(11025)
-#define	IXP_SAMPR16000		(16000)
-#define	IXP_SAMPR22050		(22050)
-#define	IXP_SAMPR24000		(24000)
-#define	IXP_SAMPR32000		(32000)
-#define	IXP_SAMPR44100		(44100)
-#define	IXP_SAMPR48000		(48000)
-
-#define	IXP_DEFAULT_SR		IXP_SAMPR8000
-#define	IXP_DEFAULT_CH		AUDIO_CHANNELS_MONO
-#define	IXP_DEFAULT_PREC	AUDIO_PRECISION_8
-#define	IXP_DEFAULT_ENC		AUDIO_ENCODING_ULAW
-#define	IXP_DEFAULT_PGAIN	(AUDIO_MAX_GAIN * 3 / 4)
-#define	IXP_DEFAULT_RGAIN		(127)
-#define	IXP_DEFAULT_MONITOR_GAIN	(0)
-#define	IXP_DEFAULT_BAL		AUDIO_MID_BALANCE
-#define	IXP_INTS		(175)	/* default interrupt rate */
-#define	IXP_MIN_INTS		(24)	/* minimum interrupt rate */
-#define	IXP_MAX_INTS		(500)	/* maximum interrupt rate */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif /* _SYS_AUDIOIXP_H_ */
diff --git a/usr/src/uts/common/sys/audio/audiots.h b/usr/src/uts/common/sys/audio/audiots.h
deleted file mode 100644
index 686ba89062..0000000000
--- a/usr/src/uts/common/sys/audio/audiots.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright (c) 2000-2001 by Sun Microsystems, Inc.
- * All rights reserved.
- */
-
-#ifndef	_SYS_AUDIOTS_H
-#define	_SYS_AUDIOTS_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/*
- * Header file for the audiots device driver.
- */
-
-/*
- * Values returned by the AUDIO_GETDEV ioctl()
- */
-#define	TS_DEV_NAME		"SUNW,audiots"
-#define	TS_DEV_CONFIG		"onboard1"
-#define	TS_DEV_VERSION_A	"a"		/* grover, sngl strming play */
-#define	TS_DEV_VERSION_B	"b"		/* birdsnest, sngl strm play */
-
-/*
- * Driver supported configuration information
- */
-#define	TS_NAME			"audiots"
-#define	TS_MOD_NAME		"mixer audio driver"
-
-#define	TS_SAMPR4000		(4000)
-#define	TS_SAMPR5510		(5510)
-#define	TS_SAMPR6620		(6620)
-#define	TS_SAMPR8000		(8000)
-#define	TS_SAMPR9600		(9600)
-#define	TS_SAMPR11025		(11025)
-#define	TS_SAMPR16000		(16000)
-#define	TS_SAMPR18900		(18900)
-#define	TS_SAMPR22050		(22050)
-#define	TS_SAMPR27420		(27420)
-#define	TS_SAMPR32000		(32000)
-#define	TS_SAMPR33075		(33075)
-#define	TS_SAMPR37800		(37800)
-#define	TS_SAMPR44100		(44100)
-#define	TS_SAMPR48000		(48000)
-
-#define	TS_DEFAULT_SR		TS_SAMPR8000
-#define	TS_DEFAULT_CH		AUDIO_CHANNELS_MONO
-#define	TS_DEFAULT_PREC		AUDIO_PRECISION_8
-#define	TS_DEFAULT_ENC		AUDIO_ENCODING_ULAW
-#define	TS_DEFAULT_PGAIN	(AUDIO_MAX_GAIN * 3 / 4)
-#define	TS_DEFAULT_RGAIN	(127)
-#define	TS_DEFAULT_MONITOR_GAIN	(0)
-#define	TS_DEFAULT_BAL		AUDIO_MID_BALANCE	/* MUST be mid */
-#define	TS_INTS			(175)		/* default interrupt rate */
-#define	TS_MIN_INTS		(24)		/* minimum interrupt rate */
-#define	TS_MAX_INTS		(2000)		/* maximum interrupt rate */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIOTS_H */
diff --git a/usr/src/uts/common/sys/audio/impl/am_src1_impl.h b/usr/src/uts/common/sys/audio/impl/am_src1_impl.h
deleted file mode 100644
index ca5e04cbf1..0000000000
--- a/usr/src/uts/common/sys/audio/impl/am_src1_impl.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright (c) 2000-2001 by Sun Microsystems, Inc.
- * All rights reserved.
- *
- * This header file defines the internal interfaces for the default audio
- * mixer sample rate converted. It is NOT to be distributed with Solaris.
- */
-
-#ifndef	_SYS_AM_SRC1_IMPL_H
-#define	_SYS_AM_SRC1_IMPL_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#include <sys/note.h>
-#include <sys/types.h>
-#include <sys/audio/am_src1.h>
-
-/*
- * am_src1_data_t	- sample rate conversion algorithm #1 data
- */
-#define	AM_SRC1_MUPSIZE1	4		/* max up/down conversions */
-struct am_src1_data {
-	kmutex_t	src1_lock;		/* protects this structure */
-	uint_t		src1_inFs;		/* input sample rate */
-	uint_t		src1_outFs;		/* device sample rate */
-	int		src1_k;			/* filter parameter */
-	int		src1_up[AM_SRC1_MUPSIZE1];	/* up sample steps */
-	int		src1_down[AM_SRC1_MUPSIZE1];	/* down sample steps */
-	int		src1_ustart1[AM_SRC1_MUPSIZE1];
-						/* up smpl saved samps, L */
-	int		src1_ustart2[AM_SRC1_MUPSIZE1];
-						/* up smpl saved samps, R */
-	int		src1_dstart[AM_SRC1_MUPSIZE1];
-						/* dwn sample saved samples */
-	int		src1_out1[AM_SRC1_MUPSIZE1];
-						/* down sample saved samples */
-	int		src1_out2[AM_SRC1_MUPSIZE1];
-						/* down sample saved samples */
-	int		(*src1_up0)(struct am_src1_data *, int, int,
-				int *, int *, int);
-	int		(*src1_up1)(struct am_src1_data *, int, int,
-				int *, int *, int);
-	int		(*src1_up2)(struct am_src1_data *, int, int,
-				int *, int *, int);
-	int		(*src1_up3)(struct am_src1_data *, int, int,
-				int *, int *, int);
-	int		src1_up_factor;		/* total up conversions */
-	int		src1_down_factor;	/* total down conversions */
-	int		src1_count;		/* number of up conversions */
-};
-typedef struct am_src1_data am_src1_data_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(am_src1_data::src1_lock, am_src1_data))
-
-/* fudge factor to make sure we've got enough memory */
-#define	AM_SRC1_BUFFER		(8 * sizeof (int))
-
-/* misc. defines */
-#define	AM_SRC1_SHIFT1		1
-#define	AM_SRC1_SHIFT2		2
-#define	AM_SRC1_SHIFT3		3
-#define	AM_SRC1_SHIFT8		8
-#define	AM_INT32_SHIFT		2
-#define	SRC_MAX(a, b)		((a) > (b) ? (a) : (b))
-#define	SRC_MAX4(w, x, y, z)	SRC_MAX(SRC_MAX(w, x), SRC_MAX(y, z))
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AM_SRC1_IMPL_H */
diff --git a/usr/src/uts/common/sys/audio/impl/am_src2_impl.h b/usr/src/uts/common/sys/audio/impl/am_src2_impl.h
deleted file mode 100644
index 5b2ce698ba..0000000000
--- a/usr/src/uts/common/sys/audio/impl/am_src2_impl.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the internal interfaces for the alternative audio
- * mixer sample rate converter. It is NOT to be distributed with Solaris.
- */
-
-#ifndef	_SYS_AM_SRC2_IMPL_H
-#define	_SYS_AM_SRC2_IMPL_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#include <sys/note.h>
-#include <sys/types.h>
-#include <sys/audio/am_src2.h>
-
-/* misc. defines */
-#define	AM_SRC2_NZCS		(10)		/* number of zero crossings */
-#define	AM_SRC2_CPZC		(256)		/* coeffs per zero crossing */
-#define	AM_SRC2_CPZC_SHIFT	(8)		/* 1 << CPZC_SHIFT = CPZC */
-#define	AM_SRC2_CPZC_MASK	(0xFF)		/* for modulo arithmetic */
-#define	AM_SRC2_STEREO_MASK	(0x1)		/* for modulo arithmetic */
-#define	AM_SRC2_COFF_SHIFT	(16)		/* for scaling output */
-#define	AM_SRC2_COFFS		((2 * AM_SRC2_NZCS) + 1) /* coefficients */
-#define	AM_SRC2_SHIFT1		(1)
-#define	AM_SRC2_SHIFT2		(2)
-
-/* We define the location of the first, middle and last coefficients */
-#define	AM_SRC2_START		(AM_SRC2_NZCS)
-#define	AM_SRC2_MIDDLE		((AM_SRC2_NZCS * AM_SRC2_CPZC) + AM_SRC2_CPZC)
-#define	AM_SRC2_END		((2 * AM_SRC2_NZCS * AM_SRC2_CPZC) + \
-				AM_SRC2_CPZC)
-
-/*
- * We define the maximum amount of prebuffer room required.
- * If playing, it is assumed we are increasing the sample rate
- * in which case the amount of room required is given by
- * 2 * max_channels * nzcs.
- */
-#define	AM_SRC2_PBUFFER		(2 * 2 * AM_SRC2_NZCS)
-
-/*
- * When down sampling the amount of prebuffer room required
- * depends on the conversion factor. Here we assume that this
- * factor will never be greater than 12 (i.e. 48kHz -> 4kHz)
- * and we allocate enough memory to cope with that. A little
- * wasteful but worth it for the simplicity achieved. Here
- * the room required is 2 * max_channels * max_factor * nzcs.
- */
-#define	AM_SRC2_RBUFFER		(2 * 2 * 12 * AM_SRC2_NZCS)
-
-/*
- * am_src2_data_t	- sample rate conversion algorithm #2 data
- */
-struct am_src2_data {
-	kmutex_t	src2_lock;	/* protects this structure */
-	int		src2_inFs;	/* input sample rate */
-	int		src2_outFs;	/* device sample rate */
-	int		(*src2_resample)(struct am_src2_data *, int *, int *,
-				int);	/* resampling function */
-	int		src2_pbsize;	/* room in front of samples */
-	int		src2_delta_c;	/* distance to nearest input sample */
-	int		src2_delta_n;	/* normalised delta_c */
-	int		src2_i_index;	/* start point in input */
-	int		src2_csteps;	/* step size through input */
-	int		src2_cmod;	/* fraction of csteps left over */
-	int		src2_cover;	/* fraction of csteps counter */
-	int		src2_tsteps;	/* step size through filter */
-	int		*src2_table;	/* filter coefficients */
-	int		**src2_tables;  /* individual tables */
-};
-
-typedef struct am_src2_data am_src2_data_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(am_src2_data::src2_lock, am_src2_data))
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AM_SRC2_IMPL_H */
diff --git a/usr/src/uts/common/sys/audio/impl/am_src2_table.h b/usr/src/uts/common/sys/audio/impl/am_src2_table.h
deleted file mode 100644
index 294d756df3..0000000000
--- a/usr/src/uts/common/sys/audio/impl/am_src2_table.h
+++ /dev/null
@@ -1,1493 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file contains sample rate conversion tables for amsrc2.
- */
-
-#ifndef	_SYS_AM_SRC2_TABLE_H
-#define	_SYS_AM_SRC2_TABLE_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/* The table used in down sampling */
-int32_t _amsrc2table[5377] = {
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0, 0,
-0, -1, -1, -2, -2, -3, -3, -4,
--4, -5, -6, -6, -7, -7, -8, -9,
--9, -10, -10, -11, -12, -12, -13, -14,
--14, -15, -16, -16, -17, -18, -19, -19,
--20, -21, -21, -22, -23, -24, -24, -25,
--26, -26, -27, -28, -29, -29, -30, -31,
--32, -33, -33, -34, -35, -36, -36, -37,
--38, -39, -40, -40, -41, -42, -43, -43,
--44, -45, -46, -47, -47, -48, -49, -50,
--51, -51, -52, -53, -54, -54, -55, -56,
--57, -58, -58, -59, -60, -61, -61, -62,
--63, -64, -64, -65, -66, -67, -67, -68,
--69, -69, -70, -71, -71, -72, -73, -73,
--74, -75, -75, -76, -77, -77, -78, -78,
--79, -80, -80, -81, -81, -82, -82, -83,
--83, -84, -84, -85, -85, -86, -86, -87,
--87, -88, -88, -88, -89, -89, -89, -90,
--90, -90, -91, -91, -91, -91, -92, -92,
--92, -92, -93, -93, -93, -93, -93, -93,
--93, -93, -93, -93, -94, -94, -94, -93,
--93, -93, -93, -93, -93, -93, -93, -93,
--92, -92, -92, -92, -92, -91, -91, -91,
--90, -90, -90, -89, -89, -88, -88, -87,
--87, -86, -86, -85, -85, -84, -84, -83,
--82, -82, -81, -80, -80, -79, -78, -77,
--76, -76, -75, -74, -73, -72, -71, -70,
--69, -68, -67, -66, -65, -64, -63, -62,
--61, -60, -58, -57, -56, -55, -54, -52,
--51, -50, -48, -47, -46, -44, -43, -41,
--40, -38, -37, -35, -34, -32, -31, -29,
--28, -26, -24, -23, -21, -19, -18, -16,
--14, -13, -11, -9, -7, -6, -4, -2,
-0, 2, 4, 6, 8, 9, 11, 13,
-15, 17, 19, 21, 23, 25, 27, 29,
-31, 33, 36, 38, 40, 42, 44, 46,
-48, 50, 52, 55, 57, 59, 61, 63,
-65, 68, 70, 72, 74, 76, 79, 81,
-83, 85, 87, 90, 92, 94, 96, 99,
-101, 103, 105, 107, 110, 112, 114, 116,
-119, 121, 123, 125, 127, 129, 132, 134,
-136, 138, 140, 142, 145, 147, 149, 151,
-153, 155, 157, 159, 161, 163, 165, 167,
-169, 171, 173, 175, 177, 179, 181, 183,
-185, 187, 188, 190, 192, 194, 195, 197,
-199, 201, 202, 204, 206, 207, 209, 210,
-212, 213, 215, 216, 218, 219, 220, 222,
-223, 224, 226, 227, 228, 229, 230, 231,
-233, 234, 235, 236, 237, 237, 238, 239,
-240, 241, 242, 242, 243, 244, 244, 245,
-245, 246, 246, 247, 247, 247, 248, 248,
-248, 248, 249, 249, 249, 249, 249, 249,
-249, 248, 248, 248, 248, 248, 247, 247,
-246, 246, 245, 245, 244, 244, 243, 242,
-241, 240, 240, 239, 238, 237, 236, 234,
-233, 232, 231, 230, 228, 227, 225, 224,
-222, 221, 219, 218, 216, 214, 212, 211,
-209, 207, 205, 203, 201, 199, 196, 194,
-192, 190, 187, 185, 182, 180, 177, 175,
-172, 170, 167, 164, 161, 159, 156, 153,
-150, 147, 144, 141, 138, 134, 131, 128,
-125, 121, 118, 115, 111, 108, 104, 100,
-97, 93, 90, 86, 82, 78, 74, 71,
-67, 63, 59, 55, 51, 47, 43, 38,
-34, 30, 26, 22, 17, 13, 9, 4,
-0, -4, -9, -13, -18, -22, -27, -31,
--36, -41, -45, -50, -54, -59, -64, -69,
--73, -78, -83, -88, -92, -97, -102, -107,
--112, -116, -121, -126, -131, -136, -141, -146,
--151, -155, -160, -165, -170, -175, -180, -185,
--190, -195, -199, -204, -209, -214, -219, -224,
--229, -234, -238, -243, -248, -253, -258, -262,
--267, -272, -277, -281, -286, -291, -295, -300,
--305, -309, -314, -318, -323, -327, -332, -336,
--340, -345, -349, -353, -358, -362, -366, -370,
--374, -378, -382, -386, -390, -394, -398, -402,
--406, -410, -413, -417, -421, -424, -428, -431,
--435, -438, -441, -445, -448, -451, -454, -457,
--460, -463, -466, -469, -471, -474, -477, -479,
--482, -484, -487, -489, -491, -493, -495, -497,
--499, -501, -503, -505, -507, -508, -510, -511,
--513, -514, -515, -516, -518, -519, -519, -520,
--521, -522, -522, -523, -523, -524, -524, -524,
--525, -525, -525, -524, -524, -524, -524, -523,
--523, -522, -521, -520, -520, -519, -518, -516,
--515, -514, -512, -511, -509, -508, -506, -504,
--502, -500, -498, -496, -493, -491, -488, -486,
--483, -480, -478, -475, -472, -468, -465, -462,
--458, -455, -451, -448, -444, -440, -436, -432,
--428, -424, -419, -415, -411, -406, -401, -397,
--392, -387, -382, -377, -372, -366, -361, -356,
--350, -344, -339, -333, -327, -321, -315, -309,
--303, -297, -290, -284, -278, -271, -264, -258,
--251, -244, -237, -230, -223, -216, -209, -201,
--194, -187, -179, -172, -164, -156, -149, -141,
--133, -125, -117, -109, -101, -93, -85, -76,
--68, -60, -51, -43, -34, -26, -17, -9,
-0, 9, 17, 26, 35, 44, 53, 62,
-71, 80, 89, 98, 107, 116, 125, 135,
-144, 153, 162, 172, 181, 190, 199, 209,
-218, 228, 237, 246, 256, 265, 274, 284,
-293, 303, 312, 321, 331, 340, 349, 359,
-368, 377, 387, 396, 405, 414, 424, 433,
-442, 451, 460, 469, 479, 488, 497, 506,
-514, 523, 532, 541, 550, 558, 567, 576,
-584, 593, 601, 610, 618, 626, 634, 643,
-651, 659, 667, 675, 682, 690, 698, 706,
-713, 721, 728, 735, 742, 750, 757, 764,
-770, 777, 784, 791, 797, 803, 810, 816,
-822, 828, 834, 840, 846, 851, 857, 862,
-867, 872, 877, 882, 887, 892, 896, 901,
-905, 909, 914, 917, 921, 925, 929, 932,
-935, 939, 942, 944, 947, 950, 952, 955,
-957, 959, 961, 963, 964, 966, 967, 969,
-970, 971, 971, 972, 972, 973, 973, 973,
-973, 973, 972, 972, 971, 970, 969, 968,
-966, 965, 963, 961, 959, 957, 955, 952,
-950, 947, 944, 941, 937, 934, 930, 927,
-923, 919, 914, 910, 905, 900, 896, 890,
-885, 880, 874, 869, 863, 857, 850, 844,
-838, 831, 824, 817, 810, 803, 795, 787,
-780, 772, 764, 755, 747, 738, 730, 721,
-712, 702, 693, 684, 674, 664, 654, 644,
-634, 624, 613, 603, 592, 581, 570, 559,
-547, 536, 524, 513, 501, 489, 477, 464,
-452, 440, 427, 414, 401, 388, 375, 362,
-349, 335, 322, 308, 294, 281, 267, 253,
-238, 224, 210, 195, 181, 166, 152, 137,
-122, 107, 92, 77, 62, 46, 31, 15,
-0, -16, -31, -47, -63, -78, -94, -110,
--126, -142, -158, -174, -190, -206, -223, -239,
--255, -271, -288, -304, -320, -337, -353, -370,
--386, -402, -419, -435, -452, -468, -485, -501,
--517, -534, -550, -567, -583, -599, -616, -632,
--648, -664, -680, -697, -713, -729, -745, -761,
--777, -792, -808, -824, -840, -855, -871, -886,
--901, -917, -932, -947, -962, -977, -992, -1007,
--1022, -1036, -1051, -1065, -1079, -1093, -1107, -1121,
--1135, -1149, -1162, -1176, -1189, -1202, -1215, -1228,
--1241, -1254, -1266, -1279, -1291, -1303, -1315, -1327,
--1338, -1350, -1361, -1372, -1383, -1394, -1404, -1415,
--1425, -1435, -1445, -1454, -1464, -1473, -1482, -1491,
--1500, -1508, -1517, -1525, -1533, -1540, -1548, -1555,
--1562, -1569, -1576, -1582, -1588, -1594, -1600, -1605,
--1611, -1616, -1621, -1625, -1630, -1634, -1638, -1641,
--1645, -1648, -1651, -1653, -1656, -1658, -1660, -1662,
--1663, -1664, -1665, -1666, -1666, -1666, -1666, -1666,
--1665, -1665, -1663, -1662, -1660, -1658, -1656, -1654,
--1651, -1648, -1645, -1641, -1637, -1633, -1629, -1624,
--1619, -1614, -1609, -1603, -1597, -1591, -1584, -1577,
--1570, -1563, -1555, -1547, -1539, -1531, -1522, -1513,
--1504, -1494, -1485, -1475, -1464, -1454, -1443, -1432,
--1420, -1409, -1397, -1384, -1372, -1359, -1346, -1333,
--1320, -1306, -1292, -1278, -1263, -1248, -1233, -1218,
--1203, -1187, -1171, -1154, -1138, -1121, -1104, -1087,
--1070, -1052, -1034, -1016, -997, -979, -960, -941,
--922, -902, -883, -863, -842, -822, -802, -781,
--760, -739, -717, -696, -674, -652, -630, -608,
--585, -563, -540, -517, -494, -470, -447, -423,
--400, -376, -351, -327, -303, -278, -254, -229,
--204, -179, -154, -128, -103, -77, -52, -26,
-0, 26, 52, 78, 104, 131, 157, 184,
-210, 237, 263, 290, 317, 344, 371, 398,
-425, 452, 479, 506, 533, 560, 587, 615,
-642, 669, 696, 723, 751, 778, 805, 832,
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--3535, -3505, -3475, -3444, -3413, -3381, -3349, -3317,
--3284, -3251, -3217, -3183, -3149, -3114, -3079, -3044,
--3008, -2972, -2936, -2899, -2862, -2825, -2788, -2750,
--2712, -2673, -2634, -2595, -2556, -2517, -2477, -2437,
--2397, -2357, -2316, -2275, -2234, -2193, -2152, -2110,
--2068, -2026, -1984, -1942, -1900, -1857, -1815, -1772,
--1729, -1686, -1643, -1600, -1557, -1514, -1470, -1427,
--1383, -1340, -1296, -1252, -1209, -1165, -1121, -1078,
--1034, -990, -946, -903, -859, -815, -772, -728,
--685, -641, -598, -554, -511, -468, -425, -382,
--339, -296, -253, -211, -168, -126, -84, -42,
-0, 42, 83, 125, 166, 207, 248, 289,
-329, 369, 409, 449, 489, 528, 568, 607,
-645, 684, 722, 760, 798, 835, 873, 910,
-946, 983, 1019, 1055, 1090, 1125, 1160, 1195,
-1229, 1263, 1297, 1330, 1363, 1396, 1429, 1461,
-1492, 1524, 1555, 1585, 1616, 1645, 1675, 1704,
-1733, 1761, 1790, 1817, 1845, 1872, 1898, 1924,
-1950, 1975, 2000, 2025, 2049, 2073, 2096, 2119,
-2142, 2164, 2186, 2207, 2228, 2248, 2268, 2288,
-2307, 2326, 2344, 2362, 2379, 2397, 2413, 2429,
-2445, 2460, 2475, 2490, 2504, 2517, 2530, 2543,
-2555, 2567, 2579, 2589, 2600, 2610, 2620, 2629,
-2638, 2646, 2654, 2661, 2668, 2675, 2681, 2686,
-2692, 2697, 2701, 2705, 2708, 2711, 2714, 2716,
-2718, 2719, 2720, 2721, 2721, 2721, 2720, 2719,
-2717, 2715, 2713, 2710, 2707, 2703, 2699, 2695,
-2690, 2685, 2679, 2673, 2667, 2660, 2653, 2645,
-2637, 2629, 2621, 2611, 2602, 2592, 2582, 2572,
-2561, 2550, 2538, 2526, 2514, 2501, 2488, 2475,
-2461, 2447, 2433, 2419, 2404, 2388, 2373, 2357,
-2341, 2325, 2308, 2291, 2273, 2256, 2238, 2220,
-2201, 2183, 2164, 2144, 2125, 2105, 2085, 2065,
-2044, 2024, 2003, 1982, 1960, 1938, 1917, 1895,
-1872, 1850, 1827, 1804, 1781, 1758, 1734, 1711,
-1687, 1663, 1639, 1615, 1590, 1565, 1541, 1516,
-1491, 1466, 1440, 1415, 1389, 1364, 1338, 1312,
-1286, 1260, 1234, 1207, 1181, 1154, 1128, 1101,
-1075, 1048, 1021, 994, 967, 940, 913, 886,
-859, 832, 805, 778, 751, 723, 696, 669,
-642, 615, 587, 560, 533, 506, 479, 452,
-425, 398, 371, 344, 317, 290, 263, 237,
-210, 184, 157, 131, 104, 78, 52, 26,
-0, -26, -52, -77, -103, -128, -154, -179,
--204, -229, -254, -278, -303, -327, -351, -376,
--400, -423, -447, -470, -494, -517, -540, -563,
--585, -608, -630, -652, -674, -696, -717, -739,
--760, -781, -802, -822, -842, -863, -883, -902,
--922, -941, -960, -979, -997, -1016, -1034, -1052,
--1070, -1087, -1104, -1121, -1138, -1154, -1171, -1187,
--1203, -1218, -1233, -1248, -1263, -1278, -1292, -1306,
--1320, -1333, -1346, -1359, -1372, -1384, -1397, -1409,
--1420, -1432, -1443, -1454, -1464, -1475, -1485, -1494,
--1504, -1513, -1522, -1531, -1539, -1547, -1555, -1563,
--1570, -1577, -1584, -1591, -1597, -1603, -1609, -1614,
--1619, -1624, -1629, -1633, -1637, -1641, -1645, -1648,
--1651, -1654, -1656, -1658, -1660, -1662, -1663, -1665,
--1665, -1666, -1666, -1666, -1666, -1666, -1665, -1664,
--1663, -1662, -1660, -1658, -1656, -1653, -1651, -1648,
--1645, -1641, -1638, -1634, -1630, -1625, -1621, -1616,
--1611, -1605, -1600, -1594, -1588, -1582, -1576, -1569,
--1562, -1555, -1548, -1540, -1533, -1525, -1517, -1508,
--1500, -1491, -1482, -1473, -1464, -1454, -1445, -1435,
--1425, -1415, -1404, -1394, -1383, -1372, -1361, -1350,
--1338, -1327, -1315, -1303, -1291, -1279, -1266, -1254,
--1241, -1228, -1215, -1202, -1189, -1176, -1162, -1149,
--1135, -1121, -1107, -1093, -1079, -1065, -1051, -1036,
--1022, -1007, -992, -977, -962, -947, -932, -917,
--901, -886, -871, -855, -840, -824, -808, -792,
--777, -761, -745, -729, -713, -697, -680, -664,
--648, -632, -616, -599, -583, -567, -550, -534,
--517, -501, -485, -468, -452, -435, -419, -402,
--386, -370, -353, -337, -320, -304, -288, -271,
--255, -239, -223, -206, -190, -174, -158, -142,
--126, -110, -94, -78, -63, -47, -31, -16,
-0, 15, 31, 46, 62, 77, 92, 107,
-122, 137, 152, 166, 181, 195, 210, 224,
-238, 253, 267, 281, 294, 308, 322, 335,
-349, 362, 375, 388, 401, 414, 427, 440,
-452, 464, 477, 489, 501, 513, 524, 536,
-547, 559, 570, 581, 592, 603, 613, 624,
-634, 644, 654, 664, 674, 684, 693, 702,
-712, 721, 730, 738, 747, 755, 764, 772,
-780, 787, 795, 803, 810, 817, 824, 831,
-838, 844, 850, 857, 863, 869, 874, 880,
-885, 890, 896, 900, 905, 910, 914, 919,
-923, 927, 930, 934, 937, 941, 944, 947,
-950, 952, 955, 957, 959, 961, 963, 965,
-966, 968, 969, 970, 971, 972, 972, 973,
-973, 973, 973, 973, 972, 972, 971, 971,
-970, 969, 967, 966, 964, 963, 961, 959,
-957, 955, 952, 950, 947, 944, 942, 939,
-935, 932, 929, 925, 921, 917, 914, 909,
-905, 901, 896, 892, 887, 882, 877, 872,
-867, 862, 857, 851, 846, 840, 834, 828,
-822, 816, 810, 803, 797, 791, 784, 777,
-770, 764, 757, 750, 742, 735, 728, 721,
-713, 706, 698, 690, 682, 675, 667, 659,
-651, 643, 634, 626, 618, 610, 601, 593,
-584, 576, 567, 558, 550, 541, 532, 523,
-514, 506, 497, 488, 479, 469, 460, 451,
-442, 433, 424, 414, 405, 396, 387, 377,
-368, 359, 349, 340, 331, 321, 312, 303,
-293, 284, 274, 265, 256, 246, 237, 228,
-218, 209, 199, 190, 181, 172, 162, 153,
-144, 135, 125, 116, 107, 98, 89, 80,
-71, 62, 53, 44, 35, 26, 17, 9,
-0, -9, -17, -26, -34, -43, -51, -60,
--68, -76, -85, -93, -101, -109, -117, -125,
--133, -141, -149, -156, -164, -172, -179, -187,
--194, -201, -209, -216, -223, -230, -237, -244,
--251, -258, -264, -271, -278, -284, -290, -297,
--303, -309, -315, -321, -327, -333, -339, -344,
--350, -356, -361, -366, -372, -377, -382, -387,
--392, -397, -401, -406, -411, -415, -419, -424,
--428, -432, -436, -440, -444, -448, -451, -455,
--458, -462, -465, -468, -472, -475, -478, -480,
--483, -486, -488, -491, -493, -496, -498, -500,
--502, -504, -506, -508, -509, -511, -512, -514,
--515, -516, -518, -519, -520, -520, -521, -522,
--523, -523, -524, -524, -524, -524, -525, -525,
--525, -524, -524, -524, -523, -523, -522, -522,
--521, -520, -519, -519, -518, -516, -515, -514,
--513, -511, -510, -508, -507, -505, -503, -501,
--499, -497, -495, -493, -491, -489, -487, -484,
--482, -479, -477, -474, -471, -469, -466, -463,
--460, -457, -454, -451, -448, -445, -441, -438,
--435, -431, -428, -424, -421, -417, -413, -410,
--406, -402, -398, -394, -390, -386, -382, -378,
--374, -370, -366, -362, -358, -353, -349, -345,
--340, -336, -332, -327, -323, -318, -314, -309,
--305, -300, -295, -291, -286, -281, -277, -272,
--267, -262, -258, -253, -248, -243, -238, -234,
--229, -224, -219, -214, -209, -204, -199, -195,
--190, -185, -180, -175, -170, -165, -160, -155,
--151, -146, -141, -136, -131, -126, -121, -116,
--112, -107, -102, -97, -92, -88, -83, -78,
--73, -69, -64, -59, -54, -50, -45, -41,
--36, -31, -27, -22, -18, -13, -9, -4,
-0, 4, 9, 13, 17, 22, 26, 30,
-34, 38, 43, 47, 51, 55, 59, 63,
-67, 71, 74, 78, 82, 86, 90, 93,
-97, 100, 104, 108, 111, 115, 118, 121,
-125, 128, 131, 134, 138, 141, 144, 147,
-150, 153, 156, 159, 161, 164, 167, 170,
-172, 175, 177, 180, 182, 185, 187, 190,
-192, 194, 196, 199, 201, 203, 205, 207,
-209, 211, 212, 214, 216, 218, 219, 221,
-222, 224, 225, 227, 228, 230, 231, 232,
-233, 234, 236, 237, 238, 239, 240, 240,
-241, 242, 243, 244, 244, 245, 245, 246,
-246, 247, 247, 248, 248, 248, 248, 248,
-249, 249, 249, 249, 249, 249, 249, 248,
-248, 248, 248, 247, 247, 247, 246, 246,
-245, 245, 244, 244, 243, 242, 242, 241,
-240, 239, 238, 237, 237, 236, 235, 234,
-233, 231, 230, 229, 228, 227, 226, 224,
-223, 222, 220, 219, 218, 216, 215, 213,
-212, 210, 209, 207, 206, 204, 202, 201,
-199, 197, 195, 194, 192, 190, 188, 187,
-185, 183, 181, 179, 177, 175, 173, 171,
-169, 167, 165, 163, 161, 159, 157, 155,
-153, 151, 149, 147, 145, 142, 140, 138,
-136, 134, 132, 129, 127, 125, 123, 121,
-119, 116, 114, 112, 110, 107, 105, 103,
-101, 99, 96, 94, 92, 90, 87, 85,
-83, 81, 79, 76, 74, 72, 70, 68,
-65, 63, 61, 59, 57, 55, 52, 50,
-48, 46, 44, 42, 40, 38, 36, 33,
-31, 29, 27, 25, 23, 21, 19, 17,
-15, 13, 11, 9, 8, 6, 4, 2,
-0, -2, -4, -6, -7, -9, -11, -13,
--14, -16, -18, -19, -21, -23, -24, -26,
--28, -29, -31, -32, -34, -35, -37, -38,
--40, -41, -43, -44, -46, -47, -48, -50,
--51, -52, -54, -55, -56, -57, -58, -60,
--61, -62, -63, -64, -65, -66, -67, -68,
--69, -70, -71, -72, -73, -74, -75, -76,
--76, -77, -78, -79, -80, -80, -81, -82,
--82, -83, -84, -84, -85, -85, -86, -86,
--87, -87, -88, -88, -89, -89, -90, -90,
--90, -91, -91, -91, -92, -92, -92, -92,
--92, -93, -93, -93, -93, -93, -93, -93,
--93, -93, -94, -94, -94, -93, -93, -93,
--93, -93, -93, -93, -93, -93, -93, -92,
--92, -92, -92, -91, -91, -91, -91, -90,
--90, -90, -89, -89, -89, -88, -88, -88,
--87, -87, -86, -86, -85, -85, -84, -84,
--83, -83, -82, -82, -81, -81, -80, -80,
--79, -78, -78, -77, -77, -76, -75, -75,
--74, -73, -73, -72, -71, -71, -70, -69,
--69, -68, -67, -67, -66, -65, -64, -64,
--63, -62, -61, -61, -60, -59, -58, -58,
--57, -56, -55, -54, -54, -53, -52, -51,
--51, -50, -49, -48, -47, -47, -46, -45,
--44, -43, -43, -42, -41, -40, -40, -39,
--38, -37, -36, -36, -35, -34, -33, -33,
--32, -31, -30, -29, -29, -28, -27, -26,
--26, -25, -24, -24, -23, -22, -21, -21,
--20, -19, -19, -18, -17, -16, -16, -15,
--14, -14, -13, -12, -12, -11, -10, -10,
--9, -9, -8, -7, -7, -6, -6, -5,
--4, -4, -3, -3, -2, -2, -1, -1, 0,
-};
-
-/* The table used in up sampling */
-int32_t _amsrc2tables[5397] = {
-0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 0, 65536, 0, 0,
-0, 0, 0, 0, 0, 0, 0,
-0, -1, 2, -4, 9, -16, 26,
--42, 68, -116, 251, 65535, -248, 115,
--67, 42, -26, 15, -9, 4, -2,
-0, -1, 4, -9, 17, -31, 52,
--84, 135, -232, 503, 65530, -495, 230,
--134, 83, -52, 31, -17, 9, -4,
-0, -2, 6, -13, 26, -47, 78,
--126, 203, -349, 758, 65523, -739, 344,
--201, 125, -77, 46, -26, 13, -6,
-0, -2, 8, -18, 35, -63, 104,
--168, 272, -466, 1014, 65511, -982, 457,
--268, 166, -103, 62, -34, 17, -7,
-0, -3, 9, -22, 44, -78, 131,
--211, 340, -584, 1273, 65497, -1222, 570,
--334, 207, -128, 77, -43, 22, -9,
-0, -3, 11, -27, 53, -94, 157,
--253, 409, -702, 1533, 65479, -1460, 682,
--400, 248, -154, 92, -51, 26, -11,
-0, -4, 13, -31, 62, -110, 184,
--296, 478, -821, 1796, 65458, -1696, 794,
--465, 289, -179, 107, -60, 30, -13,
-0, -4, 15, -36, 71, -126, 210,
--339, 547, -940, 2061, 65433, -1929, 905,
--530, 329, -204, 122, -68, 34, -14,
-0, -5, 17, -41, 80, -142, 237,
--382, 616, -1059, 2327, 65405, -2161, 1015,
--595, 369, -229, 137, -76, 38, -16,
-0, -6, 19, -45, 89, -158, 263,
--425, 686, -1179, 2595, 65374, -2391, 1125,
--660, 409, -254, 152, -85, 43, -18,
-0, -6, 21, -50, 98, -174, 290,
--468, 755, -1299, 2866, 65340, -2618, 1234,
--724, 449, -278, 166, -93, 47, -19,
-0, -7, 23, -54, 107, -190, 317,
--511, 825, -1420, 3138, 65302, -2843, 1342,
--788, 489, -303, 181, -101, 51, -21,
-0, -7, 25, -59, 116, -206, 344,
--554, 895, -1541, 3412, 65261, -3066, 1449,
--851, 528, -327, 195, -109, 55, -23,
-0, -8, 27, -64, 125, -223, 371,
--598, 965, -1663, 3688, 65216, -3287, 1556,
--915, 568, -351, 210, -117, 59, -24,
-0, -9, 29, -69, 135, -239, 398,
--641, 1035, -1784, 3965, 65168, -3505, 1662,
--977, 607, -376, 224, -125, 63, -26,
-0, -9, 31, -73, 144, -255, 425,
--685, 1106, -1907, 4245, 65117, -3721, 1768,
--1040, 645, -400, 238, -133, 67, -28,
-0, -10, 33, -78, 153, -271, 452,
--728, 1176, -2029, 4526, 65063, -3935, 1872,
--1102, 684, -423, 253, -141, 71, -29,
-0, -10, 36, -83, 162, -288, 479,
--772, 1247, -2152, 4809, 65005, -4147, 1976,
--1163, 722, -447, 267, -149, 74, -31,
-0, -11, 38, -88, 172, -304, 506,
--815, 1317, -2274, 5094, 64944, -4357, 2079,
--1224, 760, -470, 281, -156, 78, -32,
-0, -12, 40, -92, 181, -320, 533,
--859, 1388, -2398, 5380, 64880, -4564, 2182,
--1285, 798, -494, 294, -164, 82, -34,
-0, -12, 42, -97, 190, -337, 560,
--903, 1459, -2521, 5669, 64812, -4769, 2283,
--1345, 835, -517, 308, -172, 86, -35,
-0, -13, 44, -102, 199, -353, 587,
--946, 1530, -2645, 5959, 64741, -4971, 2384,
--1405, 873, -540, 322, -179, 90, -37,
-0, -14, 46, -107, 209, -370, 615,
--990, 1600, -2769, 6250, 64667, -5172, 2483,
--1465, 910, -563, 335, -187, 93, -38,
-0, -14, 48, -112, 218, -386, 642,
--1034, 1671, -2893, 6544, 64589, -5370, 2582,
--1524, 946, -585, 349, -194, 97, -40,
-0, -15, 50, -116, 228, -402, 669,
--1078, 1742, -3017, 6839, 64509, -5566, 2681,
--1582, 983, -608, 362, -201, 100, -41,
-0, -16, 52, -121, 237, -419, 696,
--1121, 1813, -3141, 7135, 64425, -5759, 2778,
--1640, 1019, -630, 375, -209, 104, -43,
-0, -16, 55, -126, 246, -435, 723,
--1165, 1884, -3265, 7434, 64338, -5950, 2874,
--1698, 1055, -652, 388, -216, 108, -44,
-0, -17, 57, -131, 256, -452, 751,
--1209, 1955, -3390, 7733, 64247, -6139, 2970,
--1755, 1090, -674, 401, -223, 111, -46,
-0, -18, 59, -136, 265, -468, 778,
--1252, 2026, -3515, 8035, 64154, -6325, 3064,
--1811, 1125, -696, 414, -230, 115, -47,
-0, -19, 61, -141, 274, -485, 805,
--1296, 2097, -3639, 8338, 64057, -6509, 3158,
--1867, 1160, -717, 427, -237, 118, -48,
-0, -19, 63, -146, 284, -501, 832,
--1340, 2167, -3764, 8642, 63957, -6691, 3251,
--1923, 1195, -739, 440, -244, 121, -50,
-0, -20, 65, -151, 293, -517, 859,
--1383, 2238, -3889, 8948, 63854, -6871, 3343,
--1978, 1229, -760, 452, -251, 125, -51,
-0, -21, 68, -155, 303, -534, 886,
--1427, 2309, -4014, 9255, 63747, -7048, 3434,
--2033, 1263, -781, 464, -258, 128, -52,
-0, -21, 70, -160, 312, -550, 913,
--1470, 2379, -4139, 9564, 63638, -7222, 3524,
--2087, 1297, -802, 477, -264, 131, -54,
-0, -22, 72, -165, 321, -567, 940,
--1514, 2450, -4263, 9875, 63525, -7394, 3613,
--2140, 1330, -822, 489, -271, 134, -55,
-0, -23, 74, -170, 331, -583, 967,
--1557, 2520, -4388, 10186, 63409, -7564, 3701,
--2193, 1363, -842, 501, -278, 138, -56,
-0, -24, 76, -175, 340, -599, 994,
--1600, 2591, -4513, 10499, 63290, -7732, 3789,
--2246, 1396, -863, 513, -284, 141, -57,
-0, -24, 79, -180, 349, -616, 1021,
--1643, 2661, -4638, 10814, 63167, -7897, 3875,
--2298, 1429, -883, 524, -290, 144, -58,
-0, -25, 81, -185, 359, -632, 1048,
--1686, 2731, -4762, 11130, 63042, -8060, 3960,
--2349, 1461, -902, 536, -297, 147, -60,
-0, -26, 83, -190, 368, -648, 1075,
--1729, 2801, -4887, 11447, 62914, -8220, 4044,
--2400, 1492, -922, 547, -303, 150, -61,
-0, -26, 85, -195, 377, -664, 1101,
--1772, 2870, -5011, 11765, 62782, -8378, 4128,
--2450, 1524, -941, 559, -309, 153, -62,
-0, -27, 87, -199, 387, -680, 1128,
--1815, 2940, -5135, 12085, 62647, -8533, 4210,
--2500, 1555, -960, 570, -315, 156, -63,
-0, -28, 90, -204, 396, -697, 1154,
--1857, 3009, -5259, 12406, 62510, -8686, 4291,
--2549, 1585, -979, 581, -321, 159, -64,
-0, -29, 92, -209, 405, -713, 1181,
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-0, -94, 248, -520, 959, -1637, 2668,
--4283, 7087, -13445, 50111, 31943, -11422, 6308,
--3859, 2404, -1464, 846, -448, 206, -71,
-0, -94, 248, -519, 957, -1633, 2661,
--4271, 7070, -13427, 50395, 31593, -11340, 6267,
--3835, 2388, -1454, 840, -445, 204, -71,
-0, -94, 247, -518, 955, -1629, 2654,
--4260, 7052, -13406, 50677, 31244, -11256, 6225,
--3810, 2373, -1445, 834, -441, 202, -70,
-0, -93, 247, -516, 952, -1624, 2646,
--4247, 7033, -13383, 50957, 30894, -11171, 6183,
--3785, 2357, -1435, 828, -438, 201, -69,
-0, -93, 246, -515, 950, -1619, 2638,
--4234, 7013, -13358, 51235, 30544, -11085, 6139,
--3759, 2341, -1425, 822, -435, 199, -69,
-0, -93, 246, -514, 947, -1614, 2629,
--4220, 6992, -13330, 51511, 30194, -10998, 6095,
--3733, 2325, -1415, 816, -431, 197, -68,
-0, -93, 245, -512, 944, -1609, 2620,
--4205, 6969, -13301, 51785, 29844, -10909, 6051,
--3706, 2308, -1404, 810, -428, 195, -67,
-0, -93, 245, -511, 941, -1603, 2610,
--4190, 6946, -13269, 52057, 29493, -10820, 6005,
--3679, 2291, -1394, 803, -424, 194, -67,
-0, -93, 244, -509, 937, -1597, 2600,
--4174, 6921, -13235, 52326, 29143, -10729, 5959,
--3651, 2273, -1383, 797, -421, 192, -66,
-0, -93, 244, -508, 934, -1591, 2589,
--4157, 6895, -13198, 52594, 28792, -10637, 5912,
--3623, 2256, -1372, 791, -417, 190, -65,
-0, -93, 243, -506, 930, -1584, 2579,
--4139, 6868, -13160, 52859, 28441, -10544, 5864,
--3594, 2238, -1361, 784, -413, 188, -64,
-0, -93, 242, -504, 927, -1577, 2567,
--4121, 6840, -13119, 53123, 28090, -10450, 5815,
--3565, 2220, -1350, 777, -410, 187, -64,
-0, -92, 241, -502, 923, -1570, 2555,
--4102, 6810, -13076, 53384, 27740, -10354, 5766,
--3535, 2201, -1338, 770, -406, 185, -63,
-0, -92, 240, -500, 919, -1563, 2543,
--4083, 6779, -13031, 53643, 27389, -10258, 5717,
--3505, 2183, -1327, 764, -402, 183, -62,
-0, -92, 240, -498, 914, -1555, 2530,
--4063, 6748, -12983, 53899, 27038, -10161, 5666,
--3475, 2164, -1315, 757, -398, 181, -61,
-0, -92, 239, -496, 910, -1547, 2517,
--4042, 6715, -12933, 54154, 26688, -10062, 5615,
--3444, 2144, -1303, 750, -394, 179, -61,
-0, -92, 238, -493, 905, -1539, 2504,
--4020, 6680, -12881, 54406, 26337, -9963, 5563,
--3413, 2125, -1291, 742, -390, 177, -60,
-0, -91, 237, -491, 900, -1531, 2490,
--3997, 6645, -12826, 54656, 25987, -9862, 5511,
--3381, 2105, -1279, 735, -386, 175, -59,
-0, -91, 236, -488, 896, -1522, 2475,
--3974, 6609, -12769, 54903, 25636, -9761, 5457,
--3349, 2085, -1266, 728, -382, 173, -58,
-0, -91, 234, -486, 890, -1513, 2460,
--3951, 6571, -12710, 55148, 25286, -9658, 5404,
--3317, 2065, -1254, 721, -378, 171, -58,
-0, -90, 233, -483, 885, -1504, 2445,
--3926, 6532, -12648, 55391, 24937, -9555, 5349,
--3284, 2044, -1241, 713, -374, 169, -57,
-0, -90, 232, -480, 880, -1494, 2429,
--3901, 6492, -12584, 55631, 24587, -9451, 5294,
--3251, 2024, -1228, 706, -370, 167, -56,
-0, -90, 231, -478, 874, -1485, 2413,
--3875, 6451, -12518, 55869, 24238, -9346, 5239,
--3217, 2003, -1215, 698, -366, 165, -55,
-0, -89, 230, -475, 869, -1475, 2397,
--3848, 6408, -12449, 56104, 23889, -9240, 5183,
--3183, 1982, -1202, 690, -362, 163, -54,
-0, -89, 228, -472, 863, -1464, 2379,
--3821, 6365, -12378, 56337, 23540, -9133, 5126,
--3149, 1960, -1189, 682, -358, 161, -54,
-0, -88, 227, -468, 857, -1454, 2362,
--3793, 6320, -12305, 56568, 23192, -9025, 5069,
--3114, 1938, -1176, 675, -353, 159, -53,
-0, -88, 225, -465, 850, -1443, 2344,
--3765, 6274, -12229, 56796, 22844, -8917, 5011,
--3079, 1917, -1162, 667, -349, 157, -52,
-0, -87, 224, -462, 844, -1432, 2326,
--3735, 6227, -12151, 57021, 22497, -8808, 4953,
--3044, 1895, -1149, 659, -345, 155, -51,
-0, -87, 222, -458, 838, -1420, 2307,
--3705, 6179, -12071, 57244, 22150, -8697, 4894,
--3008, 1872, -1135, 651, -340, 153, -51,
-0, -86, 221, -455, 831, -1409, 2288,
--3675, 6130, -11988, 57464, 21803, -8587, 4835,
--2972, 1850, -1121, 643, -336, 151, -50,
-0, -86, 219, -451, 824, -1397, 2268,
--3643, 6079, -11902, 57682, 21457, -8475, 4775,
--2936, 1827, -1107, 634, -332, 149, -49,
-0, -85, 218, -448, 817, -1384, 2248,
--3611, 6028, -11815, 57897, 21112, -8363, 4714,
--2899, 1804, -1093, 626, -327, 147, -48,
-0, -85, 216, -444, 810, -1372, 2228,
--3578, 5975, -11725, 58109, 20767, -8250, 4654,
--2862, 1781, -1079, 618, -323, 145, -47,
-0, -84, 214, -440, 803, -1359, 2207,
--3545, 5921, -11632, 58319, 20423, -8136, 4592,
--2825, 1758, -1065, 610, -318, 142, -47,
-0, -84, 212, -436, 795, -1346, 2186,
--3511, 5866, -11537, 58526, 20079, -8022, 4531,
--2788, 1734, -1051, 601, -314, 140, -46,
-0, -83, 211, -432, 787, -1333, 2164,
--3476, 5809, -11440, 58730, 19736, -7907, 4468,
--2750, 1711, -1036, 593, -309, 138, -45,
-0, -82, 209, -428, 780, -1320, 2142,
--3441, 5752, -11340, 58932, 19394, -7792, 4406,
--2712, 1687, -1022, 584, -305, 136, -44,
-0, -82, 207, -424, 772, -1306, 2119,
--3405, 5693, -11238, 59131, 19052, -7676, 4343,
--2673, 1663, -1007, 576, -300, 134, -43,
-0, -81, 205, -419, 764, -1292, 2096,
--3368, 5634, -11134, 59327, 18711, -7559, 4279,
--2634, 1639, -992, 567, -295, 132, -43,
-0, -80, 203, -415, 755, -1278, 2073,
--3330, 5573, -11027, 59521, 18371, -7442, 4215,
--2595, 1615, -977, 558, -291, 129, -42,
-0, -80, 201, -411, 747, -1263, 2049,
--3292, 5511, -10917, 59711, 18031, -7324, 4151,
--2556, 1590, -962, 550, -286, 127, -41,
-0, -79, 199, -406, 738, -1248, 2025,
--3254, 5448, -10806, 59899, 17693, -7206, 4086,
--2517, 1565, -947, 541, -281, 125, -40,
-0, -78, 196, -401, 730, -1233, 2000,
--3214, 5384, -10691, 60084, 17355, -7087, 4021,
--2477, 1541, -932, 532, -277, 123, -40,
-0, -77, 194, -397, 721, -1218, 1975,
--3174, 5318, -10575, 60266, 17018, -6968, 3956,
--2437, 1516, -917, 523, -272, 121, -39,
-0, -76, 192, -392, 712, -1203, 1950,
--3134, 5252, -10455, 60446, 16682, -6848, 3890,
--2397, 1491, -901, 514, -267, 119, -38,
-0, -76, 190, -387, 702, -1187, 1924,
--3092, 5185, -10334, 60622, 16347, -6728, 3824,
--2357, 1466, -886, 506, -262, 116, -37,
-0, -75, 187, -382, 693, -1171, 1898,
--3051, 5116, -10210, 60796, 16013, -6607, 3758,
--2316, 1440, -871, 497, -258, 114, -36,
-0, -74, 185, -377, 684, -1154, 1872,
--3008, 5046, -10084, 60966, 15680, -6486, 3691,
--2275, 1415, -855, 488, -253, 112, -36,
-0, -73, 182, -372, 674, -1138, 1845,
--2965, 4976, -9955, 61134, 15348, -6365, 3624,
--2234, 1389, -840, 479, -248, 110, -35,
-0, -72, 180, -366, 664, -1121, 1817,
--2921, 4904, -9824, 61299, 15016, -6243, 3557,
--2193, 1364, -824, 469, -243, 107, -34,
-0, -71, 177, -361, 654, -1104, 1790,
--2877, 4831, -9690, 61461, 14686, -6121, 3489,
--2152, 1338, -808, 460, -238, 105, -33,
-0, -70, 175, -356, 644, -1087, 1761,
--2832, 4757, -9554, 61620, 14357, -5999, 3421,
--2110, 1312, -792, 451, -234, 103, -33,
-0, -69, 172, -350, 634, -1070, 1733,
--2786, 4682, -9415, 61775, 14029, -5876, 3353,
--2068, 1286, -777, 442, -229, 101, -32,
-0, -68, 170, -344, 624, -1052, 1704,
--2740, 4606, -9274, 61928, 13702, -5753, 3285,
--2026, 1260, -761, 433, -224, 99, -31,
-0, -67, 167, -339, 613, -1034, 1675,
--2693, 4529, -9131, 62078, 13376, -5630, 3216,
--1984, 1234, -745, 424, -219, 96, -30,
-0, -66, 164, -333, 603, -1016, 1645,
--2646, 4451, -8985, 62225, 13052, -5507, 3148,
--1942, 1207, -729, 414, -214, 94, -29,
-0, -65, 161, -327, 592, -997, 1616,
--2598, 4371, -8837, 62369, 12728, -5383, 3079,
--1900, 1181, -713, 405, -209, 92, -29,
-0, -64, 159, -321, 581, -979, 1585,
--2549, 4291, -8686, 62510, 12406, -5259, 3009,
--1857, 1154, -697, 396, -204, 90, -28,
-0, -63, 156, -315, 570, -960, 1555,
--2500, 4210, -8533, 62647, 12085, -5135, 2940,
--1815, 1128, -680, 387, -199, 87, -27,
-0, -62, 153, -309, 559, -941, 1524,
--2450, 4128, -8378, 62782, 11765, -5011, 2870,
--1772, 1101, -664, 377, -195, 85, -26,
-0, -61, 150, -303, 547, -922, 1492,
--2400, 4044, -8220, 62914, 11447, -4887, 2801,
--1729, 1075, -648, 368, -190, 83, -26,
-0, -60, 147, -297, 536, -902, 1461,
--2349, 3960, -8060, 63042, 11130, -4762, 2731,
--1686, 1048, -632, 359, -185, 81, -25,
-0, -58, 144, -290, 524, -883, 1429,
--2298, 3875, -7897, 63167, 10814, -4638, 2661,
--1643, 1021, -616, 349, -180, 79, -24,
-0, -57, 141, -284, 513, -863, 1396,
--2246, 3789, -7732, 63290, 10499, -4513, 2591,
--1600, 994, -599, 340, -175, 76, -24,
-0, -56, 138, -278, 501, -842, 1363,
--2193, 3701, -7564, 63409, 10186, -4388, 2520,
--1557, 967, -583, 331, -170, 74, -23,
-0, -55, 134, -271, 489, -822, 1330,
--2140, 3613, -7394, 63525, 9875, -4263, 2450,
--1514, 940, -567, 321, -165, 72, -22,
-0, -54, 131, -264, 477, -802, 1297,
--2087, 3524, -7222, 63638, 9564, -4139, 2379,
--1470, 913, -550, 312, -160, 70, -21,
-0, -52, 128, -258, 464, -781, 1263,
--2033, 3434, -7048, 63747, 9255, -4014, 2309,
--1427, 886, -534, 303, -155, 68, -21,
-0, -51, 125, -251, 452, -760, 1229,
--1978, 3343, -6871, 63854, 8948, -3889, 2238,
--1383, 859, -517, 293, -151, 65, -20,
-0, -50, 121, -244, 440, -739, 1195,
--1923, 3251, -6691, 63957, 8642, -3764, 2167,
--1340, 832, -501, 284, -146, 63, -19,
-0, -48, 118, -237, 427, -717, 1160,
--1867, 3158, -6509, 64057, 8338, -3639, 2097,
--1296, 805, -485, 274, -141, 61, -19,
-0, -47, 115, -230, 414, -696, 1125,
--1811, 3064, -6325, 64154, 8035, -3515, 2026,
--1252, 778, -468, 265, -136, 59, -18,
-0, -46, 111, -223, 401, -674, 1090,
--1755, 2970, -6139, 64247, 7733, -3390, 1955,
--1209, 751, -452, 256, -131, 57, -17,
-0, -44, 108, -216, 388, -652, 1055,
--1698, 2874, -5950, 64338, 7434, -3265, 1884,
--1165, 723, -435, 246, -126, 55, -16,
-0, -43, 104, -209, 375, -630, 1019,
--1640, 2778, -5759, 64425, 7135, -3141, 1813,
--1121, 696, -419, 237, -121, 52, -16,
-0, -41, 100, -201, 362, -608, 983,
--1582, 2681, -5566, 64509, 6839, -3017, 1742,
--1078, 669, -402, 228, -116, 50, -15,
-0, -40, 97, -194, 349, -585, 946,
--1524, 2582, -5370, 64589, 6544, -2893, 1671,
--1034, 642, -386, 218, -112, 48, -14,
-0, -38, 93, -187, 335, -563, 910,
--1465, 2483, -5172, 64667, 6250, -2769, 1600,
--990, 615, -370, 209, -107, 46, -14,
-0, -37, 90, -179, 322, -540, 873,
--1405, 2384, -4971, 64741, 5959, -2645, 1530,
--946, 587, -353, 199, -102, 44, -13,
-0, -35, 86, -172, 308, -517, 835,
--1345, 2283, -4769, 64812, 5669, -2521, 1459,
--903, 560, -337, 190, -97, 42, -12,
-0, -34, 82, -164, 294, -494, 798,
--1285, 2182, -4564, 64880, 5380, -2398, 1388,
--859, 533, -320, 181, -92, 40, -12,
-0, -32, 78, -156, 281, -470, 760,
--1224, 2079, -4357, 64944, 5094, -2274, 1317,
--815, 506, -304, 172, -88, 38, -11,
-0, -31, 74, -149, 267, -447, 722,
--1163, 1976, -4147, 65005, 4809, -2152, 1247,
--772, 479, -288, 162, -83, 36, -10,
-0, -29, 71, -141, 253, -423, 684,
--1102, 1872, -3935, 65063, 4526, -2029, 1176,
--728, 452, -271, 153, -78, 33, -10,
-0, -28, 67, -133, 238, -400, 645,
--1040, 1768, -3721, 65117, 4245, -1907, 1106,
--685, 425, -255, 144, -73, 31, -9,
-0, -26, 63, -125, 224, -376, 607,
--977, 1662, -3505, 65168, 3965, -1784, 1035,
--641, 398, -239, 135, -69, 29, -9,
-0, -24, 59, -117, 210, -351, 568,
--915, 1556, -3287, 65216, 3688, -1663, 965,
--598, 371, -223, 125, -64, 27, -8,
-0, -23, 55, -109, 195, -327, 528,
--851, 1449, -3066, 65261, 3412, -1541, 895,
--554, 344, -206, 116, -59, 25, -7,
-0, -21, 51, -101, 181, -303, 489,
--788, 1342, -2843, 65302, 3138, -1420, 825,
--511, 317, -190, 107, -54, 23, -7,
-0, -19, 47, -93, 166, -278, 449,
--724, 1234, -2618, 65340, 2866, -1299, 755,
--468, 290, -174, 98, -50, 21, -6,
-0, -18, 43, -85, 152, -254, 409,
--660, 1125, -2391, 65374, 2595, -1179, 686,
--425, 263, -158, 89, -45, 19, -6,
-0, -16, 38, -76, 137, -229, 369,
--595, 1015, -2161, 65405, 2327, -1059, 616,
--382, 237, -142, 80, -41, 17, -5,
-0, -14, 34, -68, 122, -204, 329,
--530, 905, -1929, 65433, 2061, -940, 547,
--339, 210, -126, 71, -36, 15, -4,
-0, -13, 30, -60, 107, -179, 289,
--465, 794, -1696, 65458, 1796, -821, 478,
--296, 184, -110, 62, -31, 13, -4,
-0, -11, 26, -51, 92, -154, 248,
--400, 682, -1460, 65479, 1533, -702, 409,
--253, 157, -94, 53, -27, 11, -3,
-0, -9, 22, -43, 77, -128, 207,
--334, 570, -1222, 65497, 1273, -584, 340,
--211, 131, -78, 44, -22, 9, -3,
-0, -7, 17, -34, 62, -103, 166,
--268, 457, -982, 65511, 1014, -466, 272,
--168, 104, -63, 35, -18, 8, -2,
-0, -6, 13, -26, 46, -77, 125,
--201, 344, -739, 65523, 758, -349, 203,
--126, 78, -47, 26, -13, 6, -2,
-0, -4, 9, -17, 31, -52, 83,
--134, 230, -495, 65530, 503, -232, 135,
--84, 52, -31, 17, -9, 4, -1,
-0, -2, 4, -9, 15, -26, 42,
--67, 115, -248, 65535, 251, -116, 68,
--42, 26, -16, 9, -4, 2, -1,
-0, 0, 0, 0, 0, 0, 0,
-0, 0, 0, 65536, 0, 0, 0,
-0, 0, 0, 0, 0, 0, 0,
-};
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AM_SRC2_TABLE_H */
diff --git a/usr/src/uts/common/sys/audio/impl/audio810_impl.h b/usr/src/uts/common/sys/audio/impl/audio810_impl.h
deleted file mode 100644
index 05baf89564..0000000000
--- a/usr/src/uts/common/sys/audio/impl/audio810_impl.h
+++ /dev/null
@@ -1,350 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIO810_IMPL_H_
-#define	_SYS_AUDIO810_IMPL_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#ifdef _KERNEL
-
-
-/*
- * Misc. defines
- */
-#define	I810_IDNUM			(0x6175)
-#define	I810_MINPACKET			(0)
-#define	I810_MAXPACKET			(1*1024)
-#define	I810_HIWATER			(64*1024)
-#define	I810_LOWATER			(32*1024)
-
-#define	I810_DMA_PCM_IN			(1)
-#define	I810_DMA_PCM_OUT		(2)
-
-#define	I810_LAST_AC_REG		(0x3A)
-
-#define	I810_INIT_NO_RESTORE		(1)
-#define	I810_INIT_RESTORE		(0)
-#define	I810_CODEC_REG(r)		((r) >> 1)
-
-#define	I845_AM_REGS_SIZE		(0x200)
-#define	I845_BM_REGS_SIZE		(0x100)
-#define	I810_AM_REGS_SIZE		(0x100)
-#define	I810_BM_REGS_SIZE		(0x40)
-#define	I810_BD_NUMS			(32)
-#define	I810_BD_SIZE	\
-	(I810_BD_NUMS * sizeof (i810_bd_entry_t))
-
-#define	I810_BSIZE			(8*1024)
-
-#define	I810_NOT_SUSPENDED		(0)
-#define	I810_SUSPENDED			(~I810_NOT_SUSPENDED)
-
-#define	I810_MAX_CHANNELS		(200)		/* force max # chs */
-#define	I810_MAX_HW_CHANNELS		(32)
-#define	I810_MAX_IN_CHANNELS		(1)
-#define	I810_MAX_OUT_CHANNELS	\
-	(I810_MAX_HW_CHANNELS - I810_MAX_IN_CHANNELS)
-#define	I810_INPUT_STREAM		(31)
-#define	I810_PORT_UNMUTE		(0xffffffff)
-
-#define	I810_KIOP(X)	((kstat_intr_t *)(X->i810_ksp->ks_data))
-
-#define	I810_MOD_SIZE			(16)
-#define	I810_PLAY_BUF_SZ		(1024)
-#define	I810_RECORD_BUF_SZ		(1024)
-#define	I810_BUF_MIN			(512)
-#define	I810_BUF_MAX			(8192)
-
-/* The size of each entry of "reg" property is 5 integers */
-#define	I810_INTS_PER_REG_PROP		5
-
-/* The index to the size of address space for each entry of "reg" property */
-#define	I810_REG_PROP_ADDR_LEN_IDX	4
-
-/* offset from the base of specified DMA engine */
-#define	I810_OFFSET_BD_BASE		(0x00)
-#define	I810_OFFSET_CIV			(0x04)
-#define	I810_OFFSET_LVI			(0x05)
-#define	I810_OFFSET_SR			(0x06)
-#define	I810_OFFSET_PICB		(0x08)
-#define	I810_OFFSET_PIV			(0x0A)
-#define	I810_OFFSET_CR			(0x0B)
-
-/* DMA engine offset from base */
-#define	I810_BASE_PCM_IN		(0x00)
-#define	I810_BASE_PCM_OUT		(0x10)
-#define	I810_BASE_MIC			(0x20)
-
-/* PCM in bus master registers */
-#define	I810_PCM_IN_BD_BASE	(I810_OFFSET_BD_BASE)
-#define	I810_PCM_IN_CIV		(I810_OFFSET_CIV)
-#define	I810_PCM_IN_LVI		(I810_OFFSET_LVI)
-#define	I810_PCM_IN_SR		(I810_OFFSET_SR)
-#define	I810_PCM_IN_PICB	(I810_OFFSET_PICB)
-#define	I810_PCM_IN_PIV		(I810_OFFSET_PIV)
-#define	I810_PCM_IN_CR		(I810_OFFSET_CR)
-
-#define	I810_PCM_OUT_BD_BASE	(I810_OFFSET_BD_BASE + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_CIV	(I810_OFFSET_CIV + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_LVI	(I810_OFFSET_LVI + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_SR		(I810_OFFSET_SR + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_PICB	(I810_OFFSET_PICB + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_PIV	(I810_OFFSET_PIV + I810_BASE_PCM_OUT)
-#define	I810_PCM_OUT_CR		(I810_OFFSET_CR + I810_BASE_PCM_OUT)
-
-
-#define	I810_MIC_BD_BASE	(I810_OFFSET_BD_BASE + I810_BASE_MIC)
-#define	I810_MIC_CIV		(I810_OFFSET_CIV + I810_BASE_MIC)
-#define	I810_MIC_LVI		(I810_OFFSET_LVI +I810_BASE_MIC)
-#define	I810_MIC_SR		(I810_OFFSET_SR + I810_BASE_MIC)
-#define	I810_MIC_PICB		(I810_OFFSET_PICB + I810_BASE_MIC)
-#define	I810_MIC_PIV		(I810_OFFSET_PIV + I810_BASE_MIC)
-#define	I810_MIC_CR		(I810_OFFSET_CR + I810_BASE_MIC)
-
-#define	I810_REG_GCR			0x2C
-#define	I810_REG_GSR			0x30
-#define	I810_REG_CASR			0x34
-
-/* bits of bus master status register */
-#define	I810_BM_SR_DCH			0x01
-#define	I810_BM_SR_CELV			0x02
-#define	I810_BM_SR_LVBCI		0x04
-#define	I810_BM_SR_BCIS			0x08
-#define	I810_BM_SR_FIFOE		0x10
-
-/* bits of bus master control register */
-#define	I810_BM_CR_RUN			0x01
-#define	I810_BM_CR_RST			0x02
-#define	I810_BM_CR_LVBIE		0x04
-#define	I810_BM_CR_FEIE			0x08
-#define	I810_BM_CR_IOCE			0x10
-
-#define	I810_BM_CR_PAUSE		0x00
-
-/*
- * Global Control Register
- */
-#define	I810_GCR_GPIE			0x00000001
-#define	I810_GCR_COLD_RST		0x00000002
-#define	I810_GCR_WARM_RST		0x00000004
-#define	I810_GCR_ACLINK_OFF		0x00000008
-#define	I810_GCR_PRI_INTR_ENABLE	0x00000010
-#define	I810_GCR_SEC_INTR_ENABLE	0x00000020
-
-/* For ICH2 or more, bit21:20 is the PCM 4/6-channel enable bits */
-#define	I810_GCR_2_CHANNELS		(0 << 20)
-#define	I810_GCR_4_CHANNELS		(1 << 20)
-#define	I810_GCR_6_CHANNELS		(2 << 20)
-#define	I810_GCR_CHANNELS_MASK		(3 << 20)
-
-/*
- * Global Status Register
- */
-#define	I810_GSR_TRI_READY		0x10000000	/* for ICH4/5 */
-#define	I810_GSR_READ_COMPL		0x00008000
-#define	I810_GSR_INTR_SEC_RESUME	0x00000800
-#define	I810_GSR_INTR_PRI_RESUME	0x00000400
-#define	I810_GSR_SEC_READY		0x00000200
-#define	I810_GSR_PRI_READY		0x00000100
-#define	I810_GSR_INTR_MIC		0x00000080
-#define	I810_GSR_INTR_POUT		0x00000040
-#define	I810_GSR_INTR_PIN		0x00000020
-#define	I810_GSR_INTR_MO		0x00000004
-#define	I810_GSR_INTR_MI		0x00000002
-#define	I810_GSR_INTR_GSI		0x00000001
-#define	I810_GSR_USE_INTR		0x00000060	/* PCM-IN ,PCM-OUT */
-
-/*
- * Macro for AD1980 codec
- */
-#define	AD1980_VID1		0x4144
-#define	AD1980_VID2		0x5370
-#define	AD1985_VID2		0x5375
-#define	CODEC_AD_REG_MISC	0x76	/* offset of ad1980 misc control reg */
-#define	AD1980_MISC_LOSEL	0x0020	/* Line-out amplifier output selector */
-#define	AD1980_MISC_HPSEL	0x0400	/* HP-out amplifier output selector */
-#define	AD1980_SURR_MUTE	0x8080	/* Mute for surround volume register */
-/*
- * chunk buffer
- */
-struct i810_bdlist_chunk {
-	caddr_t			data_buf;	/* virtual address of buffer */
-	uint32_t		addr_phy;	/* physical address of buffer */
-	ddi_dma_handle_t	dma_handle;	/* dma handle */
-	ddi_acc_handle_t	acc_handle;	/* access handle */
-	size_t			real_len;	/* real len */
-};
-typedef struct i810_bdlist_chunk	i810_bdlist_chunk_t;
-
-/*
- * sample buffer
- */
-struct i810_sample_buf {
-	boolean_t	io_started;	/* start/stop state for play/record */
-	int		avail;		/* the number of available chunk(s) */
-	uint8_t		tail;		/* For CPU, 1st available BD entry */
-	uint8_t		head;		/* For CPU, 1st BD entry to reclaim */
-	i810_bdlist_chunk_t	chunk[2];	/* 2 chunks for each buffers */
-};
-typedef struct i810_sample_buf	i810_sample_buf_t;
-
-
-/*
- * buffer descripter list entry, sees datasheet
- */
-struct i810_bd_entry {
-	uint32_t	buf_base;	/* the address of the buffer */
-	uint16_t	buf_len;	/* the number of samples */
-	uint16_t	reserved:14;
-	uint8_t		cmd_bup:1;
-	uint8_t		cmd_ioc:1;
-};
-typedef struct	i810_bd_entry	i810_bd_entry_t;
-
-/*
- * audio810_state_t	-per instance state and operation data
- */
-struct audio810_state {
-	kmutex_t		inst_lock;	/* state protection lock */
-	ddi_iblock_cookie_t	intr_iblock;
-	dev_info_t		*dip;	/* used by audio810_getinfo() */
-	audiohdl_t		audio_handle;	/* audio handle */
-	am_ad_info_t		ad_info;	/* audio device info state */
-	uint16_t		codec_shadow[64]; /* shadow of AC97 registers */
-
-	boolean_t		var_sr;		/* variable sample rate ? */
-	boolean_t		swap_out;	/* swap line-out and sur-out */
-	ddi_acc_handle_t	pci_conf_handle; /* pci configuration space */
-	ddi_acc_handle_t	am_regs_handle;	/* for audio mixer register */
-	ddi_acc_handle_t	bm_regs_handle;	/* for bus master register */
-	caddr_t			am_regs_base;	/* base of audio mixer regs */
-	caddr_t			bm_regs_base;	/* base of bus master regs */
-
-	ddi_dma_handle_t	bdl_dma_handle; /* for buffer descriptor list */
-	ddi_acc_handle_t	bdl_acc_handle;	/* access handle of bdlist */
-	void			*bdl_virtual;	/* virtual address of BDL */
-	size_t			bdl_size;	/* real len of BDL */
-	i810_bd_entry_t		*bdl_virt_pin;  /* vaddr of PCM in BDL */
-	i810_bd_entry_t		*bdl_virt_pout; /* vaddr of PCM out BDL */
-	uint32_t		bdl_phys_pin;	/* phys addr of PCM in BDL */
-	uint32_t		bdl_phys_pout;	/* phys addr of PCM in BDL */
-
-	i810_sample_buf_t	play_buf;	/* buffer for playback */
-	i810_sample_buf_t	record_buf;	/* buffer for record */
-	int			play_buf_size;	/* the size of play buffer */
-	int			record_buf_size;	/* size of in buffer */
-
-	audio_info_t		i810_defaults;	/* default state for dev */
-	audio_device_t		i810_dev_info;	/* audio device info state */
-	uint16_t		vol_bits_mask;	/* bits used to ctrl volume */
-
-	kstat_t			*i810_ksp;	/* kernel statistics */
-	uint32_t		flags;		/* state flags */
-
-	uint_t			i810_psample_rate;	/* play sample rate */
-	uint_t			i810_pchannels;		/* play channels */
-	uint_t			i810_pprecision;	/* play precision */
-	uint_t			i810_csample_rate;	/* record sample rate */
-	uint_t			i810_cchannels;		/* record channels */
-	uint_t			i810_cprecision;	/* record precision */
-	uint_t			i810_output_port;	/* current out port */
-	uint_t			i810_input_port;	/* current input port */
-	uint_t			i810_monitor_gain;	/* monitor gain */
-	int			i810_csamples;	/* pcm-in samples/interrupt */
-	int			i810_psamples;	/* pcm-out samples/intr */
-
-	uint32_t		i810_res_flags;		/* resource flags */
-	int			i810_suspended;	/* suspend/resume state */
-	int			i810_busy_cnt;	/* device busy count */
-	kcondvar_t		i810_cv;	/* suspend/resume cond. var */
-};
-typedef struct audio810_state	 audio810_state_t;
-
-/* audio810_state_t.flags defines */
-#define	I810_DMA_PLAY_STARTED	0x00000001	/* play DMA eng. initialized */
-#define	I810_DMA_PLAY_PAUSED	0x00000002	/* play DMA engine paused */
-#define	I810_DMA_PLAY_EMPTY	0x00000004	/* play DMA engine empty */
-#define	I810_DMA_RECD_STARTED	0x00000010	/* record DMA engine started */
-
-
-
-/* bits of audio810_state_t.i810_res_flags */
-#define	I810_RS_PCI_REGS		0x0001
-#define	I810_RS_AM_REGS			0x0002
-#define	I810_RS_BM_REGS			0x0004
-#define	I810_RS_DMA_BDL_HANDLE		0x0008
-#define	I810_RS_DMA_BDL_MEM		0x0010
-#define	I810_RS_DMA_BDL_BIND		0x0020
-
-
-/*
- * Useful bit twiddlers
- */
-#define	I810_BM_GET8(reg)	\
-	ddi_get8(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)))
-
-#define	I810_BM_GET16(reg)	\
-	ddi_get16(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)))
-
-#define	I810_BM_GET32(reg)	\
-	ddi_get32(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)))
-
-#define	I810_BM_PUT8(reg, val)	\
-	ddi_put8(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)), (val))
-
-#define	I810_BM_PUT16(reg, val)	\
-	ddi_put16(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)), (val))
-
-#define	I810_BM_PUT32(reg, val)	\
-	ddi_put32(statep->bm_regs_handle, \
-	(void *)((char *)statep->bm_regs_base + (reg)), (val))
-
-#define	I810_AM_GET16(reg)	\
-	ddi_get16(statep->am_regs_handle, \
-	(void *)((char *)statep->am_regs_base + (reg)))
-
-#define	I810_AM_PUT16(reg, val)	\
-	ddi_put16(statep->am_regs_handle, \
-	(void *)((char *)statep->am_regs_base + (reg)), (val))
-
-#endif	/* _KERNEL */
-
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO810_IMPL_H_ */
diff --git a/usr/src/uts/common/sys/audio/impl/audio_mixer_impl.h b/usr/src/uts/common/sys/audio/impl/audio_mixer_impl.h
deleted file mode 100644
index 0dec342e63..0000000000
--- a/usr/src/uts/common/sys/audio/impl/audio_mixer_impl.h
+++ /dev/null
@@ -1,425 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the internal interfaces for the audio mixer
- * audio personality module. It is NOT to be distributed with Solaris or
- * included in any audio drivers.
- */
-
-#ifndef	_SYS_AUDIO_MIXER_IMPL_H
-#define	_SYS_AUDIO_MIXER_IMPL_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#define	AM_MAX_GAIN_SHIFT		8	/* 1 more then 255, but close */
-#define	AM_BALANCE_SHIFT		5
-
-#define	AM_MIN_CHS			1
-
-/*
- * Flow control is used to keep too many buffers from being allocated.
- * However, sometimes we come across apps that have a problem with flow
- * control. Therefore we can comment out and turn off flow control temporarily
- * so we can debug the app and come up with a work around.
- */
-#define	FLOW_CONTROL
-
-/*
- * Miscellaneous defines
- */
-#define	AM_INT16_SHIFT			1
-#define	AM_INT32_SHIFT			2
-#define	AM_256_SHIFT			8
-#define	AM_8_SHIFT			3
-#define	AM_HALF_ENERGY_SHIFT		1
-#define	AM_TIMES_2_SHIFT		1
-#define	AM_TIMES_4_SHIFT		2
-#define	AM_TIMES_8_SHIFT		3
-#define	AM_TIMES_16_SHIFT		4
-#define	AM_TIMES_32_SHIFT		5
-
-#define	AM_MIN_MIX_BUFSIZE		48000
-#define	AM_DEFAULT_MIX_BUFSIZE		(AM_MIN_MIX_BUFSIZE * 4 * 2)
-#define	AM_MAX_QUEUED_MSGS_CNT		500
-#define	AM_MIN_QUEUED_MSGS_CNT		200
-
-#define	AM_DEFAULT_MIXER_GAIN		(AUDIO_MAX_GAIN*3/4)
-
-#define	AM_MAX_QUEUED_MSGS		4
-
-#define	AM_MAX_SAMPLES			(1024*1024)
-
-#define	AM_SINGLE_OPEN			0
-#define	AM_MULTIPLE_OPEN		1
-
-#define	AM_NOT_MUTED			0
-#define	AM_MUTED			(~AM_NOT_MUTED)
-
-#define	AM_NO_FORCE			0
-#define	AM_FORCE			(~AM_NO_FORCE)
-
-#define	AM_NO_SERIALIZE			0
-#define	AM_SERIALIZE			(~AM_NO_SERIALIZE)
-
-#define	AM_MISC_MASK			0x00000003
-#define	AM_CHAR2INT_MASK		0x000000ff
-
-#define	AM_NO_PID			0
-#define	AM_SET_WAITING			1
-#define	AM_CLEAR_WAITING		0
-#define	AM_SET_RECORD			1
-#define	AM_NOT_RECORD			0
-#define	AM_SET_PLAY			1
-#define	AM_NOT_PLAY			0
-
-#define	AM_EXTRA_SAMPLES		16
-
-/* Private mixer return codes */
-#define	AM_CHNL_CLOSED		(-2)	/* Channel closed while processing */
-
-/*
- * The sample rate conversion handle is implemented as the channel pointer.
- * But the sample rate converter doesn't have to know about this at all.
- */
-#define	AM_SRC_CHPTR2HDL(chptr)		((srchdl_t)(chptr))
-#define	AM_SRC_HDL2CHPTR(hdl)		((audio_ch_t *)(hdl))
-
-/* audio mixer ioctl/iocdata commands */
-#define	AM_COPY_OUT_AUDIOINFO	(MIOC|1)	/* AUDIO_GETINFO */
-#define	AM_COPY_OUT_AUDIOINFO2	(MIOC|2)	/* AUDIO_SETINFO */
-#define	AM_COPY_IN_AUDIOINFO	(MIOC|3)	/* AUDIO_SETINFO */
-#define	AM_COPY_IN_DIAG_LOOPB	(MIOC|4)	/* AUDIO_DIAG_LOOPBACK */
-#define	AM_COPY_OUT_GETDEV	(MIOC|5)	/* AUDIO_GETDEV */
-#define	AM_COPY_OUT_SAMP_RATES	(MIOC|6)	/* AUDIO_MIXER_GET_SAMPLE... */
-#define	AM_COPY_IN_SAMP_RATES	(MIOC|7)	/* AUDIO_MIXER_GET_SAMPLE... */
-#define	AM_COPY_IN_SAMP_RATES2	(MIOC|8)	/* AUDIO_MIXER_GET_SAMPLE... */
-#define	AM_COPY_OUT_MIXCTLINFO	(MIOC|9)	/* AUDIO_MIXERCTL_GETINFO */
-#define	AM_COPY_IN_MIXCTLINFO	(MIOC|10)	/* AUDIO_MIXERCTL_SETINFO */
-#define	AM_COPY_OUT_MIXCTL_CHINFO (MIOC|11)	/* AUDIO_MIXERCTL_GET_CHINFO */
-#define	AM_COPY_OUT_MIXCTL_CHINFO2 (MIOC|12)	/* AUDIO_MIXERCTL_GET_CHINFO */
-#define	AM_COPY_IN_MIXCTL_GET_CHINFO (MIOC|13)	/* AUDIO_MIXERCTL_SET_CHINFO */
-#define	AM_COPY_OUT_MIXCTL_GET_CHINFO (MIOC|14)	/* AUDIO_MIXERCTL_GET_CHINFO */
-#define	AM_COPY_OUT_MIXCTL_GET_CHINFO2 (MIOC|15) /* AUDIO_MIXERCTL_GET_CHINFO */
-#define	AM_COPY_IN_MIXCTL_SET_CHINFO (MIOC|16)	/* AUDIO_MIXERCTL_SET_CHINFO */
-#define	AM_COPY_IN_MIXCTL_SET_CHINFO2 (MIOC|17)	/* AUDIO_MIXERCTL_SET_CHINFO */
-#define	AM_COPY_OUT_MIXCTL_MODE	(MIOC|18)	/* AUDIO_MIXERCTL_GET_MODE */
-#define	AM_COPY_IN_MIXCTL_MODE	(MIOC|19)	/* AUDIO_MIXERCTL_SET_MODE */
-
-/*
- * am_ch_private_t	- audio mixer channel private data
- */
-struct am_ch_private {
-	uint_t			acp_flags;	/* channel flags */
-	boolean_t		acp_reading;	/* true for RD channel */
-	boolean_t		acp_writing;	/* true for WR channel */
-	int			acp_EOF[2];	/* # of EOF signals to send */
-	int			acp_EOF_toggle;	/* toggle for EOF signals */
-	int			acp_psamples_f;	/* sample frame count mp_orig */
-	int			acp_psamples_c;	/* samples in buf to play */
-	int			acp_psamples_p;	/* samples in played buf */
-	int			acp_busy_cnt;	/* # of calls outstanding */
-	mblk_t			*acp_drain_mp;	/* saved mblk_t for DRAIN */
-	mblk_t			*acp_rec_mp;	/* record message block */
-	int			acp_rec_remaining; /* # bytes left in mp buf */
-	int			*acp_play_samp_buf; /* play sample buf space */
-	size_t			acp_psb_size;	/* size of play_samp_buf */
-	void			*acp_play_src_data; /* play src data */
-	void			*acp_rec_src_data; /* rec. src data */
-	void			*acp_ch_psrc1;	/* play src buffer #1 */
-	void			*acp_ch_psrc2;	/* play src buffer #2 */
-	void			*acp_ch_pconv1;	/* play conversion buffer #1 */
-	void			*acp_ch_pconv2;	/* play conversion buffer #1 */
-	size_t			acp_ch_psrc_siz; /* play src buffer size */
-	size_t			acp_ch_pconv_siz; /* play converter buf. size */
-	size_t			acp_ch_pbuf_size; /* play prebuffer size */
-	void			*acp_ch_rsrc1;	/* record src buffer #1 */
-	void			*acp_ch_rsrc2;	/* record src buffer #2 */
-	void			*acp_ch_rconv1;	/* record conv. buffer #1 */
-	void			*acp_ch_rconv2;	/* record conv. buffer #1 */
-	size_t			acp_ch_rsrc_siz; /* record src buffer size */
-	size_t			acp_ch_rconv_siz; /* record converter buf siz */
-	size_t			acp_ch_rbuf_size; /* record prebuffer size */
-};
-typedef struct am_ch_private am_ch_private_t;
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_ch::ch_lock, am_ch_private))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_ch_private::acp_play_src_data))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_ch_private::acp_rec_src_data))
-
-/* am_ch_private.acp_flags defines */
-#define	AM_CHNL_OPEN		0x00001u /* channel open if set */
-#define	AM_CHNL_MULTI_OPEN	0x00002u /* PID may open multiple streams */
-#define	AM_CHNL_DRAIN		0x00004u /* want drain semantics if set */
-#define	AM_CHNL_DRAIN_NEXT_INT	0x00008u /* signal drain on next intr, step 1 */
-#define	AM_CHNL_CLOSING		0x00010u /* the channel is being closed */
-#define	AM_CHNL_ALMOST_EMPTY1	0x00020u /* 0 data for ch but data in DMA buf */
-#define	AM_CHNL_ALMOST_EMPTY2	0x00040u /* 0 data for ch but data in DMA buf */
-#define	AM_CHNL_EMPTY		0x00080u /* the channel doesn't have any data */
-#define	AM_CHNL_CONTROL		0x00100u /* AUDIOCTL in same proc as AUDIO */
-#define	AM_CHNL_CLIMIT		0x00200u /* used 2 limit allocb() failed msgs */
-#define	AM_CHNL_PFLOW		0x00400u /* play side has been flow cntrlled */
-#define	AM_CHNL_RSTREAMS_SET	0x00800u /* rec side's STREAMS H2O marks set */
-#define	AM_CHNL_MSG_ON_QUEUE	0x01000u /* data message on STREAMS Q */
-#define	AM_CHNL_IOCTL_TASK	0x08000u /* ioctl() task scheduled */
-#define	AM_CHNL_QPROCSOFF	0x10000u /* qprocsoff() has been called */
-#define	AM_CHNL_SIGNAL_NEEDED	0x20000u /* someone needs to send a signal */
-
-/*
- * am_apm_persist_t	- audio mixer persistent private state data
- */
-struct am_apm_persist {
-	int		apm_mode;	/* mixer mode */
-	int		apm_mpgain;	/* saved master play gain, mixer mode */
-	int		apm_mpbal;	/* saved master play bal, mixer mode */
-	int		apm_mrgain;	/* saved master rec. gain, mixer mode */
-	int		apm_mrbal;	/* saved master rec. bal, mixer mode */
-	uint_t		apm_pgain;	/* saved play gain */
-	uint_t		apm_pbal;	/* saved play balance */
-	uint_t		apm_pport;	/* saved play port */
-	uint_t		apm_pmute;	/* saved output muted */
-	uint_t		apm_rgain;	/* saved record gain */
-	uint_t		apm_rbal;	/* saved record balance */
-	uint_t		apm_rport;	/* saved record port */
-	uint_t		apm_mgain;	/* saved monitor gain */
-};
-typedef struct am_apm_persist am_apm_persist_t;
-
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mode))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mpgain))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mpbal))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mrgain))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mrbal))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_pgain))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_pbal))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_pport))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_pmute))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_rgain))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_rbal))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_rport))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_persist::apm_mgain))
-
-/*
- * am_apm_private_t	- audio mixer state private data
- */
-struct am_apm_private {
-	kmutex_t		am_mode_lock;	/* lock for mode switch cv */
-	kcondvar_t		am_mode_cv;	/* used to switch modes only */
-	kmutex_t		am_ad_lock;	/* lock for calling driver */
-	kcondvar_t		am_ad_cv;	/* serializes calls into drvr */
-	int			am_ad_in;	/* set when calling driver */
-	audio_info_t		am_hw_info;	/* pseudo hardware state */
-	mblk_t			*am_sig_mp;	/* signal STREAMS message */
-	audio_taskq_t		am_taskq;	/* h/w task queue */
-	void			*am_args;	/* qtimeout() args */
-	int			*am_mix_buf;	/* buffer to mix audio in */
-	size_t			am_mix_size;	/* the size of the buffer */
-	int			*am_send_buf;	/* buffer to send audio from */
-	size_t			am_send_size;	/* the size of the buffer */
-	uint_t			am_flags;	/* flags for the audio mixer */
-	uint_t			am_pflags;	/* play flags for the mixer */
-	uint_t			am_rflags;	/* record flags for the mixer */
-	int			am_channels;	/* current channels */
-	int			am_in_chs;	/* current record channels */
-	int			am_out_chs;	/* current play channels */
-	int			am_max_in_chs;	/* R/O, max input channels */
-	int			am_max_out_chs;	/* R/O, max output channels */
-	uint_t			am_hw_pchs;	/* the real h/w play channels */
-	uint_t			am_hw_pprec;	/* the real h/w play prec. */
-	uint_t			am_hw_penc;	/* the real h/w play encoding */
-	uint_t			am_hw_rchs;	/* the real h/w rec. channels */
-	uint_t			am_hw_rprec;	/* the real h/w rec. prec. */
-	uint_t			am_hw_renc;	/* the real h/w rec. encoding */
-	uint_t			am_save_psr;	/* saved play sample rate */
-	uint_t			am_save_rsr;	/* saved record sample rate */
-	uint_t			am_save_hw_rgain; /* saved h/w record gain */
-	am_apm_persist_t	*am_pstate;	/* persistent state */
-};
-typedef struct am_apm_private am_apm_private_t;
-
-/* defines for am_apm_private.am_ad_in */
-#define	AM_APM_IN_DRIVER	0x00000001	/* mixer is calling driver */
-#define	AM_APM_FREEZE		0x00000002	/* mixer can't call driver */
-#define	AM_APM_FORCE		0x00000004	/* force driver calls */
-
-_NOTE(MUTEX_PROTECTS_DATA(audio_apm_info::apm_lock, am_apm_private::am_flags))
-_NOTE(MUTEX_PROTECTS_DATA(audio_apm_info::apm_lock,
-    am_apm_private::am_send_size))
-_NOTE(MUTEX_PROTECTS_DATA(audio_state::as_lock, am_apm_private::am_channels))
-_NOTE(MUTEX_PROTECTS_DATA(audio_state::as_lock, am_apm_private::am_in_chs))
-_NOTE(MUTEX_PROTECTS_DATA(audio_state::as_lock, am_apm_private::am_out_chs))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_private::am_send_buf))
-
-/*
- * These are modified only when the taskq lock is held, which ensures a
- * single thread.
- */
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_private::am_hw_info))
-_NOTE(SCHEME_PROTECTS_DATA("method", am_apm_private::am_save_hw_rgain))
-
-/*
- * Further analysis is needed on the structure members of am_apm_private.
- */
-_NOTE(MUTEX_PROTECTS_DATA(am_apm_private::am_ad_lock, am_apm_private::am_ad_in))
-
-_NOTE(READ_ONLY_DATA(am_apm_private::am_hw_pchs
-	am_apm_private::am_hw_penc
-	am_apm_private::am_hw_pprec
-	am_apm_private::am_hw_rchs
-	am_apm_private::am_hw_renc
-	am_apm_private::am_hw_rprec))
-
-/* am_apm_private.am_flags defines */
-#define	AM_PRIV_ON_LINE		0x00000001u	/* device on line */
-#define	AM_PRIV_SW_MODES	0x00000002u	/* switch between M & C mode */
-
-/* am_apm_private.am_pflags and .am_rflags defines */
-#define	AM_PRIV_CH_MONO		0x00000001u	/* mono supported */
-#define	AM_PRIV_CH_STEREO	0x00000002u	/* stereo supported */
-#define	AM_PRIV_16_PCM		0x00000004u	/* 16-bit PCM supported */
-#define	AM_PRIV_8_PCM		0x00000008u	/* 8-bit PCM supported */
-#define	AM_PRIV_8_ULAW		0x00000010u	/* 8-bit u-Law supported */
-#define	AM_PRIV_8_ALAW		0x00000020u	/* 8-bit A-Law supported */
-#define	AM_PRIV_16_TRANS	0x00000040u	/* do 16-bit unsigned trans */
-#define	AM_PRIV_8_TRANS		0x00000080u	/* do 8-bit unsigned trans */
-
-#define	AM_PRIV_TASKQ_NAME	"audio_mixer_taskq"	/* taskq name */
-
-/*
- * The following data structures are used by taskq_dispatch() to queue up
- * ioctl() requests.
- */
-struct am_ioctl_args {
-	queue_t		*aia_q;			/* STREAMS queue */
-	mblk_t		*aia_mp;		/* ioctl() message block */
-};
-typedef struct am_ioctl_args am_ioctl_args_t;
-
-struct am_state_ch_args {
-	audio_state_t	*asca_statep;		/* device state structure */
-	audio_apm_info_t *asca_apm_infop;	/* mixer infop */
-	int		asca_cmd;		/* hw state change command */
-	int		asca_dir;		/* direction for the update */
-	int		asca_value;		/* value for cmd to use */
-};
-typedef struct am_state_ch_args am_state_ch_args_t;
-
-/*
- * Used only to execute a task off the taskq, so it is not shared.
- * am_sample_rates_t is defined in mixer.h.
- */
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", am_state_ch_args))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", am_sample_rates))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", am_ioctl_args))
-
-/* steams stuff */
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", copyreq))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", copyresp))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", datab))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", iocblk))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", msgb))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", queue))
-
-/* other unshared/stable or no lock needed stuff */
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", audio_channel))
-_NOTE(SCHEME_PROTECTS_DATA("Unshared data", audio_i_state))
-_NOTE(SCHEME_PROTECTS_DATA("method", audio_info))
-
-
-/*
- * The following routines are provided by the different source code
- * modules that make up the audio mixer.
- *
- * From am_main.c
- */
-int am_open_audio(queue_t *, dev_t *, int, int, cred_t *);
-int am_open_audioctl(queue_t *, dev_t *, int, int, cred_t *);
-int am_close_audio(queue_t *, int, cred_t *);
-int am_close_audioctl(queue_t *, int, cred_t *);
-int am_restore_state(audio_state_t *, audio_apm_info_t *, int);
-int am_save_state(audio_state_t *, audio_apm_info_t *, int);
-void am_apply_gain_balance(int *, int, int, int, int);
-void am_convert_int_mono_stereo(int *, int *, int *, int, int);
-void am_convert_to_int(void *, int *, int, int, int, int);
-int am_reprocess(audio_ch_t *, audio_data_t *data);
-void am_send_signal(audio_state_t *, am_apm_private_t *);
-int am_update_conv_buffer(audio_ch_t *, size_t, int);
-int am_update_src_buffer(audio_ch_t *, int, uint_t, int);
-
-/*
- * From am_ad.c
- */
-void am_ad_pause_play(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int);
-int am_ad_set_config(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int,
-    int, int, int, int, int);
-int am_ad_set_format(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int,
-    int, int, int, int, int, int);
-int am_ad_setup(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int, int);
-int am_ad_start_play(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int,
-    int);
-int am_ad_start_record(audio_state_t *, am_apm_private_t *, am_ad_info_t *,
-    int, int);
-void am_ad_stop_play(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int);
-void am_ad_stop_record(audio_state_t *, am_apm_private_t *, am_ad_info_t *,
-    int);
-void am_ad_teardown(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int,
-    int);
-int am_ck_channels(am_ad_ch_cap_t *, uint_t, boolean_t);
-int am_ck_combinations(am_ad_cap_comb_t *, int, int, boolean_t);
-int am_ck_sample_rate(am_ad_ch_cap_t *, int, int);
-void am_safe_putnext(audio_ch_t *, mblk_t *);
-int am_test_canputnext(audio_ch_t *);
-int am_set_record_streams(audio_ch_t *);
-
-/*
- * From am_ioctl.c
- */
-int am_wiocdata(queue_t *, mblk_t *, audio_ch_t *);
-int am_wioctl(queue_t *, mblk_t *, audio_ch_t *);
-void am_audio_drained(audio_ch_t *);
-int am_audio_set_info(audio_ch_t *, audio_info_t *, audio_info_t *);
-int am_set_format(audio_state_t *, am_apm_private_t *, am_ad_info_t *, int,
-    int, int, int, int, int, int, int);
-int am_set_gain(audio_state_t *, audio_apm_info_t *, uint_t, uint_t,
-    uint_t, int, int, int, int);
-
-/*
- * From g711.h
- */
-extern int16_t _8alaw2linear16[256];
-extern int8_t _8alaw2linear8[256];
-extern int16_t _8ulaw2linear16[256];
-extern int8_t _8ulaw2linear8[256];
-extern uint8_t _13linear2alaw8[0x2000];
-extern uint8_t _14linear2ulaw8[0x4000];
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_MIXER_IMPL_H */
diff --git a/usr/src/uts/common/sys/audio/impl/audio_support_impl.h b/usr/src/uts/common/sys/audio/impl/audio_support_impl.h
deleted file mode 100644
index 8b77323ef6..0000000000
--- a/usr/src/uts/common/sys/audio/impl/audio_support_impl.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 1999-2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- *
- * This header file defines the internal interfaces for the audio support
- * module. It is NOT to be distributed with Solaris.
- */
-
-#ifndef	_SYS_AUDIO_SUPPORT_IMPL_H
-#define	_SYS_AUDIO_SUPPORT_IMPL_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#define	AUDIO_MINOR_AUDIO		0	/* /dev/audio */
-#define	AUDIO_MINOR_AUDIOCTL		1	/* /dev/audioctl */
-#define	AUDIO_MINOR_WAVE_TABLE		2	/* reserved for future */
-#define	AUDIO_MINOR_MIDI_PORT		3	/* reserved for future */
-#define	AUDIO_MINOR_TIME		4	/* reserved for future */
-#define	AUDIO_MINOR_USER1		7	/* reserved for future */
-#define	AUDIO_MINOR_USER2		8	/* reserved for future */
-#define	AUDIO_MINOR_USER3		9	/* reserved for future */
-
-#define	AUDIO_MAX(a, b)			((a) > (b) ? (a) : (b))
-#define	AUDIO_MAX4(w, x, y, z)		AUDIO_MAX(AUDIO_MAX(w, x), \
-						AUDIO_MAX(y, z))
-
-/* audio support ioctl/iocdata commands */
-#define	AUDIO_COPY_OUT_CH_NUMBER	(AIOC|1)	/* AUDIO_GET_CH_NUM */
-#define	AUDIO_COPY_OUT_CH_TYPE		(AIOC|2)	/* AUDIO_GET_CH_TYPE */
-#define	AUDIO_COPY_OUT_NUM_CHS		(AIOC|3)	/* AUDIO_GET_NUM_CHS */
-#define	AUDIO_COPY_OUT_AD_DEV		(AIOC|4)	/* AUDIO_GET_AD_DEV */
-#define	AUDIO_COPY_OUT_APM_DEV		(AIOC|5)	/* AUDIO_GET_APM_DEV */
-#define	AUDIO_COPY_OUT_AS_DEV		(AIOC|6)	/* AUDIO_GET_AS_DEV */
-
-/*
- * Macros used to convert between the opaque audio taskq handles and the
- * true taskq handles. Also standard taskq defines.
- */
-#define	AUDIO_AUDIOTQHDL2TQHDL(atqhdl)		((taskq_t *)(atqhdl))
-#define	AUDIO_TQHDL2AUDIOTQHDL(tqhdl)		((audio_taskq_t)(tqhdl))
-
-#define	AUDIO_SUP_TASKQ_NTHREADS		(1)
-#define	AUDIO_SUP_TASKQ_MINALLOC		4	/* min taskq structs */
-#define	AUDIO_SUP_TASKQ_MAXALLOC		100	/* max taskq structs */
-
-/*
- * audio_inst_list_t	- structure that describes the audio device instance
- */
-struct audio_inst_info {
-	struct audio_inst_info  *ail_next;	/* linking in driver list */
-	audio_state_t		ail_state;	/* state struct for this inst */
-};
-typedef struct audio_inst_info audio_inst_info_t;
-
-/*
- * audio_qptr_t		- structure used to store private data in the STREAM
- */
-struct audio_qptr {
-	dev_t			aq_dev;		/* device name */
-	void			*aq_data;	/* STREAM private data */
-};
-typedef struct audio_qptr audio_qptr_t;
-
-/* audiosup module key for persistent memory, one per system */
-#define	AUDIO_SUP_KEY			"AUDIO:audiosup master anchor key"
-
-/* audiosup module instance key for persistent memory, one per dev instance */
-#define	AUDIO_KEY_CLASS			"AUDIO:"
-
-/*
- * audio_apm_persist_t	- structure for instance APM persistent memory storage
- */
-struct audio_apm_persist {
-	audio_device_type_e	ap_apm_type;	/* APM type, one per instance */
-	void			*ap_data;	/* per APM data */
-	size_t			ap_size;	/* size of the data */
-	struct audio_apm_persist *ap_next;	/* next item */
-};
-typedef struct audio_apm_persist audio_apm_persist_t;
-
-/*
- * audio_inst_persist_t	- instance persistent data
- */
-struct audio_inst_persist {
-	char			*amp_key;	/* persistent data key */
-	major_t			amp_major;	/* device major number */
-	int			amp_instance;	/* instance number */
-	audio_apm_persist_t	*amp_apmp;	/* ptr to APM persistent data */
-	struct audio_inst_persist *amp_next;	/* next structure */
-};
-typedef struct audio_inst_persist audio_inst_persist_t;
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_SUPPORT_IMPL_H */
diff --git a/usr/src/uts/common/sys/audio/impl/audioixp_impl.h b/usr/src/uts/common/sys/audio/impl/audioixp_impl.h
deleted file mode 100644
index 4ce55d8e16..0000000000
--- a/usr/src/uts/common/sys/audio/impl/audioixp_impl.h
+++ /dev/null
@@ -1,459 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIOIXP_IMPL_H_
-#define	_SYS_AUDIOIXP_IMPL_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-#ifdef _KERNEL
-
-#define	IXP_CONFIG_REGS		(0)	/* PCI configure register */
-#define	IXP_IO_AM_REGS		(1)	/* PCI base register 0x10 */
-
-#define	IXP_IDNUM			(0x6175)
-#define	IXP_MINPACKET			(0)
-#define	IXP_MAXPACKET			(1*1024)
-#define	IXP_HIWATER			(64*1024)
-#define	IXP_LOWATER			(32*1024)
-
-#define	IXP_DMA_PCM_IN		(1)
-#define	IXP_DMA_PCM_OUT		(2)
-
-#define	IXP_KIOP(X)	((kstat_intr_t *)(X->ixp_ksp->ks_data))
-
-/*
- * PCI configuration registers and bits
- */
-#define	IXP_PCI_REG_VID				(0x00)
-#define	IXP_PCI_VID				(0x1002)
-
-#define	IXP_PCI_REG_DID				(0x02)
-#define	IXP_PCI_DID				(0x4370)
-
-#define	IXP_PCI_REG_CMD				(0x04)
-#define	IXP_PCI_CMD_MEM_ACC_EN			(0x0002)
-#define	IXP_PCI_CMD_MASTER_EN			(0x0004)
-#define	IXP_PCI_CMD_MEM_WR_INVAL_EN		(0x0010)
-#define	IXP_PCI_CMD_PARITY_ERR_RESP		(0x0040)
-#define	IXP_PCI_CMD_SERR_EN			(0x0100)
-#define	IXP_PCI_CMD_INTA_EN			(0x0400)
-
-#define	IXP_PCI_REG_STS				(0x06)
-#define	IXP_PCI_STS_INTA			(0x0008)
-#define	IXP_PCI_STS_CAP_LIST			(0x0010)
-#define	IXP_PCI_STS_66M_CAP			(0x0020)
-#define	IXP_PCI_STS_FAST_B2B_CAP		(0x0080)
-#define	IXP_PCI_STS_MASTER_PARITY_ERROR		(0x0100)
-#define	IXP_PCI_STS_RECEIVED_TARGET_ABORT	(0x1000)
-#define	IXP_PCI_STS_RECEIVED_MASTER_ABORT	(0x2000)
-#define	IXP_PCI_STS_SERR			(0x4000)
-#define	IXP_PCI_STS_PARITY_ERR			(0x8000)
-
-#define	IXP_PCI_REG_REV_ID_CLS_CODE		(0x08)
-#define	IXP_PCI_REV_ID_400			(0x00)
-#define	IXP_PCI_REV_ID_450			(0x80)
-#define	IXP_PCI_CLS_CODE			(0x040100)
-
-#define	IXP_PCI_REG_CACHE_LINE_SIZE		(0x0c)
-#define	IXP_PCI_REG_LATENCY_TIMER		(0x0d)
-#define	IXP_PCI_REG_HEADER_TYPE			(0x0e)
-#define	IXP_PCI_REG_BUILTIN_SELF_TEST		(0x0f)
-#define	IXP_PCI_REG_BA0				(0x10)
-#define	IXP_PCI_REG_BA1				(0x14)
-#define	IXP_PCI_REG_BA2				(0x18)
-#define	IXP_PCI_REG_BA3				(0x1c)
-#define	IXP_PCI_REG_BA4				(0x20)
-#define	IXP_PCI_REG_BA5				(0x24)
-#define	IXP_PCI_REG_CARDBUS_CIS_POINTER		(0x28)
-#define	IXP_PCI_REG_SUB_ID			(0x2c)
-#define	IXP_PCI_REG_EX_ROM_BA			(0x30)
-#define	IXP_PCI_REG_CAP_P			(0x34)
-#define	IXP_PCI_REG_INT_LINE			(0x3c)
-#define	IXP_PCI_REG_INT_PIN			(0x3d)
-#define	IXP_PCI_REG_MIN_GRANT			(0x3e)
-#define	IXP_PCI_REG_MAX_LATENCY			(0x3f)
-#define	IXP_PCI_REG_MSI_CAP_REG_SET_ID		(0x40)
-#define	IXP_PCI_REG_MSI_MSG_CTRL		(0x42)
-#define	IXP_PCI_REG_MSI_MSG_ADDR		(0x44)
-#define	IXP_PCI_REG_MSI_MSG_DATA		(0x48)
-#define	IXP_PCI_REG_MSI_PROGRAM_WEIGHT		(0x4c)
-#define	IXP_PCI_REG_UNMASK_LATENCY_TIMER_EXPIRATION	(0x50)
-
-/*
- * Audio controller registers and bits
- */
-#define	IXP_AUDIO_INT				(0x00)
-#define	IXP_AUDIO_INT_IN_DMA_OVERFLOW		(1U<<0)
-#define	IXP_AUDIO_INT_IN_DMA			(1U<<1)
-#define	IXP_AUDIO_INT_OUT_DMA_UNDERFLOW		(1U<<2)
-#define	IXP_AUDIO_INT_OUT_DMA			(1U<<3)
-#define	IXP_AUDIO_INT_CODEC0_NOT_READY		(1U<<10)
-#define	IXP_AUDIO_INT_CODEC1_NOT_READY		(1U<<11)
-#define	IXP_AUDIO_INT_CODEC2_NOT_READY		(1U<<12)
-#define	IXP_AUDIO_INT_NEW_FRAME			(1U<<13)
-
-#define	IXP_AUDIO_INT_EN			(0x04)
-#define	IXP_AUDIO_INT_EN_IN_DMA_OVERFLOW	(1U<<0)
-#define	IXP_AUDIO_INT_EN_STATUS			(1U<<1)
-#define	IXP_AUDIO_INT_EN_OUT_DMA_UNDERFLOW	(1U<<2)
-#define	IXP_AUDIO_INT_EN_CODEC0_NOT_READY	(1U<<10)
-#define	IXP_AUDIO_INT_EN_CODEC1_NOT_READY	(1U<<11)
-#define	IXP_AUDIO_INT_EN_CODEC2_NOT_READY	(1U<<12)
-#define	IXP_AUDIO_INT_EN_NEW_FRAME		(1U<<13)
-
-#define	IXP_AUDIO_CMD				(0x08)
-#define	IXP_AUDIO_CMD_POWER_DOWN		(1U<<0)
-#define	IXP_AUDIO_CMD_EN_IN			(1U<<1)
-#define	IXP_AUDIO_CMD_EN_OUT			(1U<<2)
-#define	IXP_AUDIO_CMD_EN_IN_DMA			(1U<<8)
-#define	IXP_AUDIO_CMD_EN_OUT_DMA		(1U<<9)
-#define	IXP_AUDIO_CMD_INTER_IN			(1U<<21)
-#define	IXP_AUDIO_CMD_INTER_OUT			(1U<<22)
-#define	IXP_AUDIO_CMD_BURST_EN			(1U<<25)
-#define	IXP_AUDIO_CMD_AC_ACTIVE			(1U<<28)
-#define	IXP_AUDIO_CMD_AC_SOFT_RESET		(1U<<29)
-#define	IXP_AUDIO_CMD_AC_SYNC			(1U<<30)
-#define	IXP_AUDIO_CMD_AC_RESET			(1U<<31)
-
-#define	IXP_AUDIO_OUT_PHY_ADDR_DATA		(0x0c)
-#define	IXP_AUDIO_OUT_PHY_PRIMARY_CODEC	(0u)
-#define	IXP_AUDIO_OUT_PHY_SECOND_CODEC	(1u)
-#define	IXP_AUDIO_OUT_PHY_THIRD_CODEC	(2u)
-#define	IXP_AUDIO_OUT_PHY_READ		(1u<<2)
-#define	IXP_AUDIO_OUT_PHY_WRITE		(0u)
-#define	IXP_AUDIO_OUT_PHY_EN			(1u<<8)
-#define	IXP_AUDIO_OUT_PHY_ADDR_SHIFT		(9)
-#define	IXP_AUDIO_OUT_PHY_ADDR_MASK		(0x7fu<<9)
-#define	IXP_AUDIO_OUT_PHY_DATA_SHIFT		(16)
-#define	IXP_AUDIO_OUT_PHY_DATA_MASK		(0xffffu<<16)
-
-#define	IXP_AUDIO_IN_PHY_ADDR_DATA		(0x10)
-#define	IXP_AUDIO_IN_PHY_READY			(1u<<8)
-#define	IXP_AUDIO_IN_PHY_ADDR_SHIFT		(9)
-#define	IXP_AUDIO_IN_PHY_ADDR_MASK		(0x7fu<<9)
-#define	IXP_AUDIO_IN_PHY_DATA_SHIFT		(16)
-#define	IXP_AUDIO_IN_PHY_DATA_MASK		(0xffffu<<16)
-
-#define	IXP_AUDIO_SLOTREQ			(0x14)
-#define	IXP_AUDIO_COUNTER			(0x18)
-#define	IXP_AUDIO_IN_FIFO_THREASHOLD		(0x1c)
-#define	IXP_AUDIO_IN_DMA_LINK_P			(0x20)
-#define	IXP_AUDIO_IN_DMA_LINK_P_EN		(1u<<0)
-
-#define	IXP_AUDIO_IN_DMA_DT_START		(0x24)
-#define	IXP_AUDIO_IN_DMA_DT_NEXT		(0x28)
-#define	IXP_AUDIO_IN_DMA_DT_CUR			(0x2c)
-#define	IXP_AUDIO_IN_DT_SIZE_FIFO_INFO		(0x30)
-
-#define	IXP_AUDIO_OUT_DMA_SLOT_EN_THRESHOLD	(0x34)
-#define	IXP_AUDIO_OUT_DMA_SLOT_3		(1U<<0)
-#define	IXP_AUDIO_OUT_DMA_SLOT_4		(1U<<1)
-#define	IXP_AUDIO_OUT_DMA_SLOT_5		(1U<<2)
-#define	IXP_AUDIO_OUT_DMA_SLOT_6		(1U<<3)
-#define	IXP_AUDIO_OUT_DMA_SLOT_7		(1U<<4)
-#define	IXP_AUDIO_OUT_DMA_SLOT_8		(1U<<5)
-#define	IXP_AUDIO_OUT_DMA_SLOT_9		(1U<<6)
-#define	IXP_AUDIO_OUT_DMA_SLOT_10		(1U<<7)
-#define	IXP_AUDIO_OUT_DMA_SLOT_11		(1U<<8)
-#define	IXP_AUDIO_OUT_DMA_SLOT_12		(1U<<9)
-#define	IXP_AUDIO_OUT_DMA_THRESHOLD_MASK	(0x7fU<<11)
-#define	IXP_AUDIO_OUT_DMA_THRESHOLD_SHIFT	(11)
-
-#define	IXP_AUDIO_OUT_DMA_LINK_P		(0x38)
-#define	IXP_AUDIO_OUT_DMA_LINK_P_EN		(1U<<0)
-
-#define	IXP_AUDIO_OUT_DMA_DT_START		(0x3c)
-#define	IXP_AUDIO_OUT_DMA_DT_NEXT		(0x40)
-#define	IXP_AUDIO_OUT_DMA_DT_CUR		(0x44)
-#define	IXP_AUDIO_OUT_DT_SIZE_USED_FREE		(0x48)
-#define	IXP_AUDIO_SPDIF_CMD			(0x4c)
-#define	IXP_AUDIO_SPDIF_LINK_P			(0x50)
-#define	IXP_AUDIO_SPDIF_DT_START		(0x54)
-#define	IXP_AUDIO_SPDIF_DT_NEXT			(0x58)
-#define	IXP_AUDIO_SPDIF_DT_CUR			(0x5c)
-#define	IXP_AUDIO_SPDIF_DT_SIZE_FIFO_INFO	(0x60)
-#define	IXP_AUDIO_MODEM_MIRROR			(0x7c)
-#define	IXP_AUDIO_AUDIO_MIRROR			(0x80)
-#define	IXP_AUDIO_6CH_RECORDER_EN		(0x84)
-#define	IXP_AUDIO_FIFO_FLUSH		(0x88)
-#define	IXP_AUDIO_FIFO_FLUSH_OUT		(1u<<0)
-#define	IXP_AUDIO_FIFO_FLUSH_IN			(1u<<1)
-
-#define	IXP_AUDIO_OUT_FIFO_INFO		(0x8c)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG1	(0x90)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG2	(0x94)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG3	(0x98)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG4	(0x9c)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG5	(0xa0)
-#define	IXP_AUDIO_SPDIF_STATUS_BITS_REG6	(0xa4)
-#define	IXP_AUDIO_PHY_SEMA			(0xa8)
-
-#define	IXP_INIT_NO_RESTORE		(1)
-#define	IXP_INIT_RESTORE		(0)
-#define	IXP_CODEC_REG(r)		((r) >> 1)
-
-/*
- * AC97 status and link control registers are located
- * in PCI configuration space.
- */
-#define	IXP_REG_GSR				0x40
-#define	IXP_REG_GCR				0x41
-
-/* AC link interface status register */
-#define	IXP_GSR_PRI_READY			0x01
-#define	IXP_GSR_SEC_READY			0x04
-#define	IXP_GSR_TRI_READY			0x10
-#define	IXP_GSR_FOUR_READY			0x20
-
-#define	IXP_LAST_AC_REG				(0x3a)
-
-/* AC link interface control register */
-#define	IXP_GCR_ENAC97				0x80
-#define	IXP_GCR_RST				0x40
-#define	IXP_GCR_RSYNCHI				0x20
-#define	IXP_GCR_SDO				0x10
-#define	IXP_GCR_VSR				0x08
-#define	IXP_GCR_3D_AUDIO_CHANNEL		0x04
-
-/*
- * Macro for AD1980 codec
- */
-#define	AD1980_VID1		0x4144
-#define	AD1980_VID2		0x5370
-#define	AD1985_VID2		0x5375
-#define	CODEC_AD_REG_MISC	0x76	/* offset of ad1980 misc control reg */
-#define	AD1980_MISC_LOSEL	0x0020	/* Line-out amplifier output selector */
-#define	AD1980_MISC_HPSEL	0x0400	/* HP-out amplifier output selector */
-#define	AD1980_SURR_MUTE	0x8080	/* Mute for surround volume register */
-
-/*
- * Macro for ALC202 codec
- */
-#define	ALC202_VID1		0x414c
-#define	ALC202_VID2		0x4740
-
-/* audioixp_state_t.flags defines */
-#define	IXP_DMA_PLAY_STARTED	0x00000001
-#define	IXP_DMA_PLAY_PAUSED	0x00000002
-#define	IXP_DMA_PLAY_EMPTY	0x00000004
-#define	IXP_DMA_RECD_STARTED	0x00000010
-
-#define	IXP_BD_NUMS			(2)
-
-/* we always have 2 chunks */
-#define	IXP_CHUNK_MASK			(0x1l)
-
-#define	IXP_BSIZE			(8*1024)
-
-#define	IXP_MAX_CHANNELS		(200)		/* force max # chs */
-#define	IXP_MAX_HW_CHANNELS		(32)
-#define	IXP_MAX_IN_CHANNELS		(1)
-#define	IXP_MAX_OUT_CHANNELS	\
-	(IXP_MAX_HW_CHANNELS - IXP_MAX_IN_CHANNELS)
-#define	IXP_IN_STREAM		(31)
-#define	IXP_PORT_UNMUTE		(0xffffffff)
-
-#define	IXP_MOD_SIZE			(32)
-#define	IXP_PLAY_BUF_SZ			(1024)
-#define	IXP_RECORD_BUF_SZ		(1024)
-#define	IXP_BUF_MIN			(512)
-#define	IXP_BUF_MAX			(8192)
-
-/*
- * chunk buffer
- */
-struct audioixp_bdlist_chunk {
-	caddr_t			data_buf;	/* virtual address of buffer */
-	uint32_t		addr_phy;	/* physical address of buffer */
-	ddi_dma_handle_t	dma_handle;	/* dma handle */
-	ddi_acc_handle_t	acc_handle;	/* access handle */
-	size_t			real_len;	/* real len */
-};
-typedef struct audioixp_bdlist_chunk	audioixp_bdlist_chunk_t;
-
-/*
- * sample buffer
- */
-struct audioixp_sample_buf {
-	boolean_t	io_started;	/* start/stop state for play/record */
-	uint8_t		avail;		/* the number of available chunk(s) */
-	uint8_t		next;		/* next bd entry to process */
-	audioixp_bdlist_chunk_t
-			chunk[2];	/* 2 chunks for each buffers */
-	uint32_t	last_hw_pointer;
-};
-typedef struct audioixp_sample_buf	audioixp_sample_buf_t;
-
-
-/*
- * buffer descriptor list entry, see datasheet
- */
-struct audioixp_bd_entry {
-	uint32_t	buf_base;	/* the address of the buffer */
-	uint16_t	status;		/* status of the buffer */
-	uint16_t	buf_len;	/* size of the buffer in DWORD */
-	uint32_t	next;		/* physical addr of next bd_entry */
-};
-typedef struct	audioixp_bd_entry	audioixp_bd_entry_t;
-
-
-/*
- * we allocate all buffer descriptors lists in continuous dma memory,
- * so just define the struct
- */
-struct audioixp_bd_list {
-	audioixp_bd_entry_t		pcm_in[IXP_BD_NUMS];
-	audioixp_bd_entry_t		pcm_out[IXP_BD_NUMS];
-};
-typedef struct audioixp_bd_list	audioixp_bd_list_t;
-
-
-/*
- * audioixp_state_t	-per instance state and operation data
- */
-struct audioixp_state {
-	kmutex_t		inst_lock;	/* state protection lock */
-	ddi_iblock_cookie_t	intr_iblock;
-	dev_info_t		*dip;	/* used by audioixp_getinfo() */
-	audiohdl_t		audio_handle;	/* audio handle */
-	am_ad_info_t		ad_info;	/* audio device info state */
-	uint16_t		codec_shadow[64];
-						/* shadow of AC97 registers */
-	boolean_t		var_sr;		/* variable sample rate ? */
-	boolean_t		swap_out;	/* swap line-out and sur-out */
-	ddi_acc_handle_t	pci_conf_handle; /* pci configuration space */
-	ddi_acc_handle_t	am_regs_handle;	/* for audio mixer register */
-	caddr_t			am_regs_base;	/* base of audio mixer regs */
-
-	ddi_dma_handle_t	bdl_dma_handle; /* for buffer descriptor list */
-	ddi_acc_handle_t	bdl_acc_handle;	/* access handle of bdlist */
-	audioixp_bd_list_t	*bdl_virtual;	/* virtual address of BDL */
-	audioixp_bd_list_t	*bdl_phys;	/* Physical address of BDL */
-	size_t			bdl_size;	/* real len of BDL */
-
-	audioixp_sample_buf_t	play_buf; /* buffer for playback */
-	audioixp_sample_buf_t	record_buf; /* buffer for record */
-	int			play_buf_size;	/* the size of play buffer */
-	int			record_buf_size; /* size of in buffer */
-
-	audio_info_t		ixp_defaults; /* default state for dev */
-	audio_device_t		ixp_dev_info; /* audio device info state */
-	uint16_t		vol_bits_mask;	/* bits used to ctrl volume */
-
-	kstat_t			*ixp_ksp;	/* kernel statistics */
-	uint32_t		flags;		/* state flags */
-
-	uint_t			ixp_psample_rate;	/* play sample rate */
-	uint_t			ixp_pchannels;	/* play channels */
-	uint_t			ixp_pprecision;	/* play precision */
-	uint_t			ixp_csample_rate;	/* record sample rate */
-	uint_t			ixp_cchannels;	/* record channels */
-	uint_t			ixp_cprecision;	/* record precision */
-	uint_t			ixp_output_port;	/* current out port */
-	uint_t			ixp_input_port;	/* current input port */
-	uint_t			ixp_monitor_gain;	/* monitor gain */
-	int			ixp_csamples; /* pcm-in samples/intr */
-	int			ixp_psamples; /* pcm-out samples/intr */
-
-	uint32_t		ixp_res_flags;	/* resource flags */
-	uint32_t		ixp_codec_not_ready_bits; /* for codec detect */
-};
-typedef struct audioixp_state	 audioixp_state_t;
-
-/* bits of audioixp_state_t.IXP_res_flags */
-#define	IXP_RS_PCI_REGS		(1u<<0)
-#define	IXP_RS_AM_REGS		(1u<<1)
-#define	IXP_RS_DMA_BDL_HANDLE	(1u<<2)
-#define	IXP_RS_DMA_BDL_MEM	(1u<<3)
-#define	IXP_RS_DMA_BDL_BIND	(1u<<4)
-
-/*
- * Useful bit twiddlers
- */
-#define	IXP_BM_GET8(reg)	\
-	pci_config_get8(statep->pci_conf_handle, reg)
-
-#define	IXP_BM_GET16(reg)	\
-	pci_config_get16(statep->pci_conf_handle, reg)
-
-#define	IXP_BM_GET32(reg)	\
-	pci_config_get32(statep->pci_conf_handle, reg)
-
-#define	IXP_BM_PUT8(reg, val)	\
-	pci_config_put8(statep->pci_conf_handle, reg, val)
-
-#define	IXP_BM_PUT16(reg, val)	\
-	pci_config_put16(statep->pci_conf_handle, reg, val)
-
-#define	IXP_BM_PUT32(reg, val)	\
-	pci_config_put32(statep->pci_conf_handle, reg, val)
-
-#define	IXP_AM_GET32(reg)	\
-	ddi_get32(statep->am_regs_handle, \
-	(void *)((char *)statep->am_regs_base + (reg)))
-
-#define	IXP_AM_PUT32(reg, val)	\
-	ddi_put32(statep->am_regs_handle, \
-	(void *)((char *)statep->am_regs_base + (reg)), (val))
-
-#define	IXP_AM_UPDATE8(reg, mask, value) \
-	{ 				\
-		int8_t	tmp;		\
-		tmp = IXP_AM_GET8((reg));	\
-		tmp &= ~(mask);		\
-		tmp |= (value);		\
-		IXP_PUT8((reg), (tmp));	\
-	}
-
-#define	IXP_AM_UPDATE16(reg, mask, value) \
-	{ 				\
-		int16_t	tmp;		\
-		tmp = IXP_AM_GET16((reg));	\
-		tmp &= ~(mask);		\
-		tmp |= (value);		\
-		IXP_PUT16((reg), (tmp));	\
-	}
-
-#define	IXP_AM_UPDATE32(reg, mask, value) \
-	{ 				\
-		int32_t	tmp;		\
-		tmp = IXP_AM_GET32((reg));	\
-		tmp &= ~(mask);		\
-		tmp |= (value);		\
-		IXP_AM_PUT32((reg), (tmp));	\
-	}
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIOIXP_IMPL_H_ */
diff --git a/usr/src/uts/common/sys/audioio.h b/usr/src/uts/common/sys/audioio.h
index 2814eb7040..d8b4e25e68 100644
--- a/usr/src/uts/common/sys/audioio.h
+++ b/usr/src/uts/common/sys/audioio.h
@@ -19,15 +19,13 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2006 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
 #ifndef	_SYS_AUDIOIO_H
 #define	_SYS_AUDIOIO_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 #include <sys/types.h>
 #include <sys/types32.h>
 #include <sys/time.h>
@@ -266,22 +264,6 @@ typedef struct audio_device audio_device_t;
  */
 #define	AUDIO_DIAG_LOOPBACK	_IOW('A', 101, int)
 
-
-/*
- * Structure sent up as a M_PROTO message on trace streams
- */
-struct audtrace_hdr {
-	uint_t seq;		/* Sequence number (per-aud_stream) */
-	int type;		/* device-dependent */
-#if defined(_LP64) || defined(_I32LPx)
-	struct timeval32 timestamp;
-#else
-	struct timeval timestamp;
-#endif
-	char _f[8];		/* filler */
-};
-typedef struct audtrace_hdr audtrace_hdr_t;
-
 #ifdef __cplusplus
 }
 #endif
diff --git a/usr/src/uts/common/sys/audiovar.h b/usr/src/uts/common/sys/audiovar.h
deleted file mode 100644
index 83aff40454..0000000000
--- a/usr/src/uts/common/sys/audiovar.h
+++ /dev/null
@@ -1,472 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright (c) 1991-1992, 1997-2000 by Sun Microsystems, Inc.
- * All rights reserved.
- */
-
-#ifndef	_SYS_AUDIOVAR_H
-#define	_SYS_AUDIOVAR_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * The audio driver is divided into generic (device-independent) and
- * device-specific modules.  The generic routines handle most STREAMS
- * protocol issues, communicating with the device-specific code via
- * function callouts and a chained control block structure.
- *
- * Separate control block lists are maintained for reading (record) and
- * writing (play).  These control blocks simulate a chained-DMA
- * structure, in that each block controls the transfer of data between
- * the device and a single contiguous memory segment.
- *
- * The command block contains buffer start and stop addresses, a link
- * address to the next block in the chain, a 'done' flag, a 'skip' flag
- * (indicating that this command block contains no data), and a pointer
- * to the STREAMS data block structure which is private to the generic
- * driver.
- *
- * The device-specific audio driver code is expected to honor the 'skip'
- * flag and set the 'done' flag when it has completed processing the
- * command block (i.e., the data transfer, if any, is complete).  For
- * record command blocks, it is also expected to add to the 'data'
- * pointer the number of bytes successfully read from the device.
- *
- * The device-specific driver module must initialize the static STREAMS
- * control structures and must provide an identify routine (sbus-only),
- * an attach routine, and an open routine.  The open routine verifies the
- * device unit number and calls the generic open routine with the address
- * of the audio_state structure for that unit.
- *
- * The generic audio driver makes calls to the device-specific code
- * through an ops-vector table.  The following routines must be provided:
- *
- * The 'close' routine is called after either the play or record stream
- * is closed.  It may perform device-specific cleanup and initialization.
- *
- * void dev_close(as)
- * 	aud_stream_t		*as;	// Pointer to audio device state
- *
- *
- * The 'ioctl' routine is called from the STREAMS write put procedure
- * when a non-generic ioctl is encountered (AUDIO_SETINFO, AUDIO_GETINFO,
- * and AUDIO_DRAIN are the generic ioctls).  Any required data mblk_t is
- * allocated; its address is given by mp->b_cont (if this is a read/write
- * ioctl, the user-supplied buffer at mp->b_cont is reused).  If data is
- * successfully returned, the iocp->ioc_count field should be set with
- * the number of bytes returned.  If an error occurs, the 'ioctl' routine
- * should set iocp->ioc_error to the appropriate error code.  Otherwise,
- * the returned value should be AUDRETURN_CHANGE if a device state change
- * occurred (in which case a signal is sent to the control device, if
- * any) and AUDRETURN_NOCHANGE if no signal should be sent. If the ioctl
- * can not complete right away, it should return AUDRETURN_DELAYED
- * indicating that it will ack the ioctl at a later time.
- *
- * aud_return_t dev_ioctl(as, mp, iocp)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- * 	mblk_t		*mp;		// ioctl STREAMS message block
- * 	struct iocblk	*iocp;		// M_IOCTL message data
- *
- *
- * The 'start' routine is called to start i/o.  Ordinarily, it is called
- * from the device-specific 'queuecmd' routine, but it is also called
- * when paused output is resumed.
- *
- * void dev_start(as)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- *
- *
- * The 'stop' routine is called to stop i/o.  It is called when i/o is
- * paused, flushed, or the device is closed.  Note that currently queued
- * command blocks should not be flushed by this routine, since i/o may be
- * resumed from the current point.
- *
- * void dev_stop(as)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- *
- *
- * The 'setflag' routine is called to get a single device-specific flag.
- * The flag argument is either AUD_ACTIVE (return the active flag) or
- * AUD_ERRORRESET (zero the error flag, returning its previous value).
- * (The val argument is currently ignored.)
- *
- * void dev_setflag(as, flag, val)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- * 	enum aud_opflag	flag;		// AUD_ACTIVE || AUD_ERRORESET
- *
- *
- * The 'setinfo' routine is called to get or set device-specific fields
- * in the audio state structure.  If mp is NULL, the sample counters and
- * active flags should be set in v.  If mp is not NULL, then
- * mp->b_cont->data points to the audio_info_t structure supplied in an
- * AUDIO_SETINFO ioctl (ip).  All device-specific fields (gains, ports,
- * sample counts) in both v and the device itself should be updated, as
- * long as the corresponding field in ip is not set to AUD_INIT_VALUE.
- * When the sample counters are set, the value returned in v should be
- * the previous value. If the setinfo can not complete right away, it
- * should return AUDRETURN_DELAYED indicating that it will ack the ioctl
- * at a later time. If an error occurs on setinfo, the iocp->ioc_error
- * should be set as in dev_ioctl
- *
- * aud_return_t dev_setinfo(as, mp, iocp)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- * 	mblk_t		*mp;		// user info structure or NULL
- * 	struct iocblk	*iocp;		// M_IOCTL message data
- *
- *
- * The 'queuecmd' routine is called whenever a new command block is
- * linked into the chained command list.  The device-specific code must
- * ensure that the command is enqueued to the device and that i/o, if not
- * currently active, is started.
- *
- * void dev_queuecmd(as, cmdp)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- * 	struct aud_cmd	*cmdp;		// new command block to queue
- *
- *
- * The 'flushcmd' routine is called whenever the chained command list is
- * flushed.  It is only called after i/o has been stopped (via the 'stop'
- * routine) and after the command list in the audio state structure has
- * been cleared.  The device-specific code should flush the device's
- * queued command list.
- *
- * void dev_flushcmd(as)
- * 	aud_stream_t	*as;		// Pointer to audio device state
- */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-/*
- * Various generic audio driver constants
- */
-#define	AUD_INITVALUE	(~0)
-#define	Modify(X)	((uint_t)(X) != AUD_INITVALUE)
-#define	Modifys(X)	((X) != (ushort_t)AUD_INITVALUE)
-#define	Modifyc(X)	((X) != (uchar_t)AUD_INITVALUE)
-
-
-/*
- * Define the virtual chained-DMA control structure
- */
-typedef struct aud_cmd aud_cmd_t;
-struct aud_cmd {
-	/*
-	 * Data pointers
-	 */
-	uchar_t *data;		/* address of next transfer */
-	uchar_t *enddata;	/* address+1 of last transfer */
-
-	/*
-	 * Linked list management
-	 */
-	aud_cmd_t *next;	/* pointer to next or NULL */
-	aud_cmd_t *lastfragment; /* last fragment in packet */
-
-	/*
-	 * Flags
-	 */
-	uint_t :0;		/* Force word alignment */
-	uchar_t skip;		/* TRUE => no xfers on buffer */
-	uchar_t done;		/* TRUE => buffer processed */
-
-	uchar_t iotype;		/* copy of mblk's db_type */
-	boolean_t processed;	/* TRUE if processed cmd at head of list */
-
-	audtrace_hdr_t tracehdr; /* trace info */
-
-	/*
-	 * Device-independent private, opaque storage
-	 */
-	void *dihandle;
-};
-
-
-/*
- * Define the list-head for queued control structures
- */
-typedef struct aud_cmdlist aud_cmdlist_t;
-struct aud_cmdlist {
-	aud_cmd_t *head;	/* First queued command block */
-	aud_cmd_t *tail;	/* Last queued command block */
-	aud_cmd_t *free;	/* Head of free list */
-};
-
-
-/*
- * Define possible return values from the setinfo and ioctl calls
- */
-typedef enum {
-	AUDRETURN_CHANGE,
-	AUDRETURN_NOCHANGE,
-	AUDRETURN_DELAYED
-} aud_return_t;
-
-
-/*
- * Define legal values for the 'flag' argument to the 'setflag' callout
- */
-enum aud_opflag {
-	AUD_ACTIVE,		/* active flag */
-	AUD_ERRORRESET		/* error flag (reset after read) */
-};
-
-
-/*
- * The audio stream type determines the legal operations for a stream in the
- * generic portion of an audio driver.
- */
-typedef enum {
-	AUDTYPE_NONE = 00,	/* Not a legal device */
-	AUDTYPE_DATA = 01,	/* Data, IOCTL, etc., but not signals */
-	AUDTYPE_CONTROL = 02,	/* IOCTL, etc., but not M_DATA */
-	AUDTYPE_BOTH = 03	/* Anything is ok, signals delivered */
-} aud_streamtype_t;
-
-#define	ISPLAYSTREAM(as)	(ISDATASTREAM(as) && (as->openflag & FWRITE))
-#define	ISRECORDSTREAM(as)	(ISDATASTREAM(as) && (as->openflag & FREAD))
-#define	ISDATASTREAM(as)	(((as->type) & (AUDTYPE_DATA)) != 0)
-#define	ISCONTROLSTREAM(as)	(((as->type) & (AUDTYPE_CONTROL)) != 0)
-
-
-typedef enum {
-	AUDMODE_NONE = 00,	/* Not a legal mode */
-	AUDMODE_AUDIO = 01,	/* Transparent audio mode */
-	AUDMODE_HDLC = 02	/* HDLC datacomm mode */
-} aud_modetype_t;
-
-
-/*
- * This structure describes the state of the audio device and queues
- */
-typedef struct aud_state aud_state_t;
-struct aud_state {
-	/*
-	 * Back-pointer to the device-dependent audio state
-	 */
-	void *ddstate;
-
-	/*
-	 * Device-independent audio state
-	 */
-	uint_t monitor_gain;	/* input to output mix: 0 - 255 */
-	boolean_t output_muted;	/* true if output is muted */
-	uint_t hw_features;	/* hardware features this driver supports */
-	uint_t sw_features;	/* software features this driver supports */
-	uint_t sw_features_enabled;	/* supported SW features enabled */
-
-	/*
-	 * Audio ops vector
-	 */
-	struct aud_ops *ops;
-};
-
-/*
- * STREAMS routines pass the address of a 'struct audstream' when calling
- * put and service procedures.  This structure points to the STREAMS
- * queues and back to the containing 'struct aud_state'.
- */
-typedef struct aud_stream aud_stream_t;
-struct aud_stream {
-	aud_state_t *distate;	/* pointer to driver data */
-	aud_streamtype_t type;	/* defines legal operations */
-	aud_modetype_t mode;	/* Audio or HDLC data */
-	boolean_t signals_okay;	/* Can send sigs up this aud_stream */
-
-	/*
-	 * Sideways pointers to related aud_stream_t structures
-	 */
-	aud_stream_t *control_as; /* control stream */
-	aud_stream_t *output_as; /* play stream */
-	aud_stream_t *input_as;	/* record stream */
-
-	/*
-	 * Software state
-	 */
-	aud_cmdlist_t cmdlist;	/* command chains */
-	audio_prinfo_t info;	/* info for this stream side */
-	int openflag;		/* open flag & (FREAD|FWRITE) */
-	boolean_t draining;	/* TRUE if output draining */
-	int maxfrag_size;	/* max aud_cmd_t fragment size */
-	struct {
-		int action;		/* IOCTL action */
-		mblk_t *mp;		/* Pending ioctl */
-		ulong_t priv;		/* private state */
-		uint_t ioctl_id;	/* from ioc_id */
-		cred_t *credp;		/* from ioc_cr */
-		int reason;		/* HW implementation dep. reason */
-		boolean_t (*handler)(aud_stream_t *, mblk_t *, int,
-		    boolean_t);
-	} dioctl;		/* Delayed ioctls */
-	uint_t sequence;	/* packet sequence number */
-
-	/*
-	 * STREAMS information
-	 */
-	queue_t *readq;		/* STREAMS read queue */
-	queue_t *writeq;	/* STREAMS write queue */
-	queue_t *traceq;	/* STREAMS trace queue */
-
-	/*
-	 * OS-Dependent information
-	 *
-	 * NB - For now we lock on a per-unit basis, so this points to
-	 * the mutex of the unit it belongs to.  Other arrangements can
-	 * be made later
-	 *
-	 * The condition variable in a output stream is used to wait for
-	 * output to drain.
-	 *
-	 * The condition variable in a control stream is used to wait on
-	 * open if the device is in use.
-	 */
-	kmutex_t *lock;		/* low-level lock */
-	kcondvar_t cv;		/* generic condition variable */
-};
-
-#define	LOCK_AS(as)	mutex_enter((as)->lock)
-#define	UNLOCK_AS(as)	mutex_exit((as)->lock)
-#define	ASSERT_ASLOCKED(as) ASSERT(MUTEX_HELD((as)->lock))
-
-#define	AUDIOCACTION_INIT	(0) /* no ioctl in progress */
-#define	AUDIOCACTION_WAIT	(1) /* copyout response not received */
-#define	AUDIOCACTION_WAITING	(2) /* read to ack/nak */
-
-
-/*
- * Argument for audio_sensig
- */
-typedef enum {
-	AUDIO_SENDSIG_NONE = 0,	/* Default */
-	AUDIO_SENDSIG_EXPLICIT,	/* Send signal up this aud_stream only */
-	AUDIO_SENDSIG_ALL	/* Send signal up all related aud_streams */
-} audio_sendsig_t;
-
-
-/*
- * Define the ops-vector table for device-specific callouts
- *
- * close	close routine
- * ioctl	ioctl routine
- * start	routine to start I/O on a stream
- * stop		routine to stop I/O on a stream
- * setflag	routine to get or set a flag value
- * setinfo	routine to get or set the audio state structure
- * queuecmd	routine to queue a command on the HW command list
- * flushcmd	routine to flush the HW's command list
- */
-typedef struct aud_ops	aud_ops_t;
-struct aud_ops {
-#ifdef __STDC__
-	void (*close)(aud_stream_t *);
-	aud_return_t (*ioctl)(aud_stream_t *, queue_t *, mblk_t *);
-	aud_return_t (*mproto)(aud_stream_t *, mblk_t *);
-	void (*start)(aud_stream_t *);
-	void (*stop)(aud_stream_t *);
-	uint_t (*setflag)(aud_stream_t *, enum aud_opflag, uint_t);
-	aud_return_t (*setinfo)(aud_stream_t *, mblk_t *, int *);
-	void (*queuecmd)(aud_stream_t *, aud_cmd_t *);
-	void (*flushcmd)(aud_stream_t *);
-#else /* __STDC__ */
-	void (*close)();
-	aud_return_t (*ioctl)();
-	aud_return_t (*mproto)();
-	void (*start)();
-	void (*stop)();
-	uint_t (*setflag)();
-	aud_return_t (*setinfo)();
-	void (*queuecmd)();
-	void (*flushcmd)();
-#endif /* __STDC__ */
-};
-
-
-/*
- * Define pseudo-routine names for the device-specific callouts
- */
-#define	AUD_CLOSE(A)		(*(A)->distate->ops->close)(A)
-#define	AUD_IOCTL(A, Q, M)	(*(A)->distate->ops->ioctl)(A, Q, M)
-#define	AUD_MPROTO(A, M)	(*(A)->distate->ops->mproto)(A, M)
-#define	AUD_START(A)		(*(A)->distate->ops->start)(A)
-#define	AUD_STOP(A)		(*(A)->distate->ops->stop)(A)
-#define	AUD_SETFLAG(A, F, X)	(*(A)->distate->ops->setflag)(A, F, X)
-#define	AUD_GETFLAG(A, F)	(*(A)->distate->ops->setflag)(A, F, \
-    AUD_INITVALUE)
-#define	AUD_SETINFO(A, M, E)	(*(A)->distate->ops->setinfo)(A, M, E)
-#define	AUD_GETINFO(A)		(*(A)->distate->ops->setinfo)(A, NULL, NULL)
-#define	AUD_QUEUECMD(A, C)	(*(A)->distate->ops->queuecmd)(A, C)
-#define	AUD_FLUSHCMD(A)		(*(A)->distate->ops->flushcmd)(A)
-
-
-/*
- * Device Independent Audio driver function prototypes
- */
-#ifdef __STDC__
-extern int	audio_open(aud_stream_t *, queue_t *, dev_t *, int, int);
-extern int	audio_close(queue_t *, int, cred_t *);
-extern int	audio_wput(queue_t *, mblk_t *);
-extern int	audio_wsrv(queue_t *);
-extern int	audio_rput(queue_t *, mblk_t *);
-extern int	audio_rsrv(queue_t *);
-extern void	audio_gc_output(aud_stream_t *);
-extern void	audio_process_output(aud_stream_t *);
-extern void	audio_process_input(aud_stream_t *);
-extern void	audio_sendsig(aud_stream_t *, audio_sendsig_t);
-extern void	audio_flush_cmdlist(aud_stream_t *);
-extern void	audio_ack(queue_t *, mblk_t *, int);
-extern void	audio_copyout(queue_t *, mblk_t *, caddr_t, uint_t);
-extern void	audio_pause_play(aud_stream_t *);
-extern void	audio_pause_record(aud_stream_t *);
-extern void	audio_resume_play(aud_stream_t *);
-extern void	audio_resume_record(aud_stream_t *);
-extern void	audio_trace(aud_stream_t *, aud_cmd_t *);
-extern void	audio_trace_hdr(aud_stream_t *, audtrace_hdr_t *);
-#else /* __STDC__ */
-extern int	audio_open();
-extern int	audio_close();
-extern int	audio_wput();
-extern int	audio_wsrv();
-extern int	audio_rput();
-extern int	audio_rsrv();
-extern void	audio_gc_output();
-extern void	audio_process_output();
-extern void	audio_process_input();
-extern void	audio_sendsig();
-extern void	audio_flush_cmdlist();
-extern void	audio_ack();
-extern void	audio_copyout();
-extern void	audio_pause_play();
-extern void	audio_pause_record();
-extern void	audio_resume_play();
-extern void	audio_resume_record();
-extern void	audio_trace()
-extern void	audio_trace_hdr();
-#endif /* __STDC__ */
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif /* _SYS_AUDIOVAR_H */
diff --git a/usr/src/uts/common/sys/mixer.h b/usr/src/uts/common/sys/mixer.h
index 8491fc73a6..9abcd6121d 100644
--- a/usr/src/uts/common/sys/mixer.h
+++ b/usr/src/uts/common/sys/mixer.h
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,26 +19,19 @@
  * CDDL HEADER END
  */
 /*
- * Copyright (c) 1999-2000 by Sun Microsystems, Inc.
- * All rights reserved
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
  */
 
 #ifndef	_SYS_MIXER_H
 #define	_SYS_MIXER_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
 #ifdef	__cplusplus
 extern "C" {
 #endif
 
 #include <sys/audio.h>
 
-#define	MIXER_NAME		"audio mixer"	/* STREAMS module name */
-#define	MIXER_VERSION		"Rev 1"		/* 1st version of audio mixer */
-#define	MIXER_CONFIGURATION	"Config A"	/* 1st configuration */
-#define	MIXER_MOD_NAME		"Audio Mixer"	/* STREAMS modldrv name */
-
 #define	AM_MIXER_MODE			0
 #define	AM_COMPAT_MODE			1
 
diff --git a/usr/src/uts/common/sys/soundcard.h b/usr/src/uts/common/sys/soundcard.h
new file mode 100644
index 0000000000..5167b1720c
--- /dev/null
+++ b/usr/src/uts/common/sys/soundcard.h
@@ -0,0 +1,31 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+#ifndef	_SYS_SOUNDCARD_H
+#define	_SYS_SOUNDCARD_H
+
+#include <sys/audio/audio_oss.h>
+
+#endif /* _SYS_SOUNDCARD_H */
diff --git a/usr/src/uts/common/sys/usb/clients/audio/usb_ac/usb_ac.h b/usr/src/uts/common/sys/usb/clients/audio/usb_ac/usb_ac.h
index 5ed299cccb..55e3d72d06 100644
--- a/usr/src/uts/common/sys/usb/clients/audio/usb_ac/usb_ac.h
+++ b/usr/src/uts/common/sys/usb/clients/audio/usb_ac/usb_ac.h
@@ -19,7 +19,7 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
@@ -35,9 +35,8 @@ extern "C" {
 #include <sys/sunldi.h>
 #include <sys/usb/usba/usbai_private.h>
 
-/* driver specific macros */
-#define	USB_AC_HIWATER		(AM_MAX_QUEUED_MSGS_SIZE)
-#define	USB_AC_LOWATER		(32*1024)
+int usb_ac_open(dev_info_t *);
+void usb_ac_close(dev_info_t *);
 
 
 /* structure for each unit described by descriptors */
@@ -57,12 +56,17 @@ typedef struct usb_ac_unit_list {
  * plumbing data; info per plumbed module
  */
 typedef struct usb_ac_plumbed {
+	struct usb_ac_state *acp_uacp;	/* usb_ac state pointer */
 	dev_info_t	*acp_dip;	/* devinfo pointer */
 	uint_t		acp_ifno;	/* interface number */
-	int		acp_linkid;	/* link ID for plumbing */
 	int		acp_driver;	/* Plumbed driver, see value below */
-	queue_t		*acp_lrq;	/* lower read queue */
-	queue_t		*acp_lwq;	/* lower write queue */
+
+	ldi_handle_t	acp_lh;		/* ldi handle of plumbed driver */
+	dev_t		acp_devt;	/* devt of plumbed driver */
+	ddi_taskq_t	*acp_tqp;	/* taskq for I/O to plumbed driver */
+	int		acp_flags;
+#define	ACP_ENABLED	1
+
 	void		*acp_data;	/* ptr to streams or hid data */
 } usb_ac_plumbed_t;
 
@@ -72,9 +76,6 @@ typedef struct usb_ac_plumbed {
  * only one active at a time.
  */
 typedef struct usb_ac_to_as_req {
-	int		acr_wait_flag;	/* an mblk sent wait on this flag */
-	kcondvar_t	acr_cv;		/* an mblk sent; wait on this cv */
-	mblk_t		*acr_reply_mp;	/* response to current request */
 	usb_audio_formats_t acr_curr_format; /* format data from mixer */
 	int		acr_curr_dir;
 } usb_ac_to_as_req_t;
@@ -164,11 +165,6 @@ typedef struct usb_ac_state {
 	/* pipe handle */
 	usb_pipe_handle_t	usb_ac_default_ph;
 
-	/* streams management */
-	queue_t			*usb_ac_rq;		/* read q ptr */
-	queue_t			*usb_ac_wq;		/* write q ptr */
-	dev_t			usb_ac_dev;	/* dev_t of plumbing open */
-
 	/* serial access */
 	usb_serialization_t	usb_ac_ser_acc;
 
@@ -176,13 +172,10 @@ typedef struct usb_ac_state {
 	usb_ac_power_t		*usb_ac_pm; /* power capabilities */
 
 	/* mixer registration data */
-	uint_t			usb_ac_mixer_mode_enable;
 	uint_t			usb_ac_registered_with_mixer;
 
 	/* plumbing management */
-	int			usb_ac_mux_minor;
 	uint_t			usb_ac_plumbing_state;
-	ldi_handle_t		usb_ac_mux_lh;
 	ushort_t		usb_ac_busy_count;
 	usb_ac_plumbed_t	usb_ac_plumbed[USB_AC_MAX_PLUMBED];
 
@@ -197,25 +190,11 @@ typedef struct usb_ac_state {
 	 * copy registration data
 	 */
 	usb_as_registration_t	usb_ac_streams_reg[USB_AC_MAX_AS_PLUMBED];
-} usb_ac_state_t;
 
-typedef struct usb_ac_state_space {
-	void			*sp;	/* soft state for the instance */
-				/* ptr to usb_ac_restore_audio_state */
-	int			(*restore_func)
-					(usb_ac_state_t *, int);
-				/* ptr to usb_ac_get_featureID */
-	uint_t			(* get_featureID_func)
-					(usb_ac_state_t *, uchar_t,
-					uint_t, uint_t);
-				/* ptr to the usb_ac entry points */
-	am_ad_entry_t		*ac_entryp;
-				/* ptr to pm_busy/idle calls */
-	void			(*pm_busy_component)
-					(usb_ac_state_t *);
-	void			(*pm_idle_component)
-					(usb_ac_state_t *);
-} usb_ac_state_space_t;
+	ddi_taskq_t		*tqp;
+
+	char			dstr[64];
+} usb_ac_state_t;
 
 /* warlock directives, stable data */
 _NOTE(MUTEX_PROTECTS_DATA(usb_ac_state_t::usb_ac_mutex, usb_ac_state_t))
diff --git a/usr/src/uts/common/sys/usb/clients/audio/usb_as/usb_as.h b/usr/src/uts/common/sys/usb/clients/audio/usb_as/usb_as.h
index 7592a78663..e88168b391 100644
--- a/usr/src/uts/common/sys/usb/clients/audio/usb_as/usb_as.h
+++ b/usr/src/uts/common/sys/usb/clients/audio/usb_as/usb_as.h
@@ -19,7 +19,7 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
@@ -86,9 +86,7 @@ typedef struct usb_as_state {
 	uint_t			usb_as_dev_state;
 	uint_t			usb_as_ifno;
 	kmutex_t		usb_as_mutex;
-	queue_t			*usb_as_rq;		/* read q ptr */
-	queue_t			*usb_as_wq;		/* write q ptr */
-	uint_t			usb_as_streams_flag;	/* streams status */
+	uint_t			usb_as_flag;		/* status */
 
 	/* mblk containing the current control command */
 	mblk_t			*usb_as_def_mblk;
@@ -133,6 +131,7 @@ typedef struct usb_as_state {
 	/* debug support */
 	uint_t			usb_as_send_debug_count;
 	uint_t			usb_as_rcv_debug_count;
+	char			dstr[64];
 } usb_as_state_t;
 
 /* warlock directives, stable data */
@@ -141,8 +140,6 @@ _NOTE(MUTEX_PROTECTS_DATA(usb_as_state_t::usb_as_mutex, usb_as_power_t))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_dip))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_pm))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_instance))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_rq))
-_NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_wq))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_default_ph))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_isoc_ph))
 _NOTE(DATA_READABLE_WITHOUT_LOCK(usb_as_state_t::usb_as_log_handle))
@@ -202,9 +199,9 @@ typedef struct usb_as_req {
 } usb_as_req_t;
 
 
-/* Streams status */
-#define	USB_AS_STREAMS_OPEN		1
-#define	USB_AS_STREAMS_DISMANTLING	2
+/* status */
+#define	USB_AS_OPEN		1
+#define	USB_AS_DISMANTLING	2
 
 #define	USB_AS_BUFFER_SIZE		256	/* descriptor buffer size */
 
diff --git a/usr/src/uts/common/sys/usb/clients/audio/usb_mixer.h b/usr/src/uts/common/sys/usb/clients/audio/usb_mixer.h
index c6e2220944..87a0d2953e 100644
--- a/usr/src/uts/common/sys/usb/clients/audio/usb_mixer.h
+++ b/usr/src/uts/common/sys/usb/clients/audio/usb_mixer.h
@@ -19,7 +19,7 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
@@ -62,10 +62,7 @@ typedef struct usb_as_registration {
 	uchar_t			reg_mode;	/* play or record */
 	uchar_t			reg_n_formats;
 	int			reg_ifno;
-	am_ad_sample_rates_t	reg_mixer_srs;
-	am_ad_sample_rates_t	reg_compat_srs;
-	uint_t			reg_mixer_srs_list[USB_AS_N_SRS];
-	uint_t			reg_compat_srs_list[USB_AS_N_SRS];
+	uint_t			reg_srs[USB_AS_N_SRS];
 	usb_audio_formats_t	reg_formats[USB_AS_N_FORMATS];
 	uint_t			reg_channels[USB_AS_N_CHANNELS];
 	am_ad_cap_comb_t	reg_combinations[USB_AS_N_COMBINATIONS];
diff --git a/usr/src/uts/intel/Makefile.intel.shared b/usr/src/uts/intel/Makefile.intel.shared
index c45b46cefa..f597720339 100644
--- a/usr/src/uts/intel/Makefile.intel.shared
+++ b/usr/src/uts/intel/Makefile.intel.shared
@@ -195,9 +195,14 @@ DRV_KMODS	+= asy
 DRV_KMODS	+= ata
 DRV_KMODS	+= ath
 DRV_KMODS	+= atu
+DRV_KMODS	+= audio
 DRV_KMODS	+= audio810
+DRV_KMODS	+= audioens
 DRV_KMODS	+= audiohd
 DRV_KMODS	+= audioixp
+DRV_KMODS	+= audiopci
+DRV_KMODS	+= audiovia823x
+DRV_KMODS	+= austr
 DRV_KMODS	+= bl 
 DRV_KMODS	+= bge
 DRV_KMODS	+= bofi
@@ -351,8 +356,6 @@ DRV_KMODS	+= ii rdc rdcsrv rdcstub
 #
 $(CLOSED_BUILD)DRV_KMODS	+= dca
 
-$(CLOSED_BUILD)CLOSED_DRV_KMODS		+= audioens
-$(CLOSED_BUILD)CLOSED_DRV_KMODS		+= audiovia823x
 $(CLOSED_BUILD)CLOSED_DRV_KMODS		+= bmc
 $(CLOSED_BUILD)CLOSED_DRV_KMODS		+= elxl
 $(CLOSED_BUILD)CLOSED_DRV_KMODS		+= glm
@@ -546,10 +549,9 @@ SYS_KMODS	+= acctctl
 #	MISC_KMODS_32 are built only 32-bit
 #	MISC_KMODS_64 are built only 64-bit
 #
+MISC_KMODS	+= ac97
 MISC_KMODS	+= acpica
 MISC_KMODS	+= agpmaster
-MISC_KMODS	+= amsrc2
-MISC_KMODS	+= audiosup
 MISC_KMODS	+= bignum
 MISC_KMODS	+= blk2scsa
 MISC_KMODS	+= bootdev
@@ -582,7 +584,6 @@ MISC_KMODS	+= kmech_dummy
 MISC_KMODS	+= kmech_krb5
 MISC_KMODS	+= ksocket
 MISC_KMODS	+= mac
-MISC_KMODS	+= mixer
 MISC_KMODS	+= net80211
 MISC_KMODS	+= nfs_dlboot
 MISC_KMODS	+= nfssrv
@@ -621,7 +622,6 @@ MISC_KMODS	+= qlc_fw_8100
 MISC_KMODS	+= hwa1480_fw
 MISC_KMODS	+= uwba
 
-$(CLOSED_BUILD)CLOSED_MISC_KMODS	+= amsrc1
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= klmmod klmops
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= scsi_vhci_f_asym_lsi
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= scsi_vhci_f_asym_emc
@@ -688,7 +688,6 @@ $(CLOSED_BUILD)CLOSED_XMODS =		\
 #
 #	'Dacf' Modules (/kernel/dacf):
 #
-DACF_KMODS	+= usb_ac_dacf
 
 #
 #	Performance Counter BackEnd modules (/usr/kernel/pcbe)
diff --git a/usr/src/uts/intel/amsrc2/Makefile b/usr/src/uts/intel/ac97/Makefile
index 7de69fa670..1161de1ade 100644
--- a/usr/src/uts/intel/amsrc2/Makefile
+++ b/usr/src/uts/intel/ac97/Makefile
@@ -18,17 +18,14 @@
 #
 # CDDL HEADER END
 #
-
-# uts/intel/amsrc2/Makefile
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# uts/intel/ac97/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
-
 #
-#	This makefile drives the production of the audio mixer
-#	sample rate conversion routine #1 (amsrc2).
+#	This makefile drives the production of the ac97 support module.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 #
 #	Path to the base of the uts directory tree (usually /usr/src/uts).
@@ -38,12 +35,13 @@ UTSBASE = ../..
 #
 #	Define the module and object file sets.
 #
-MODULE		= amsrc2
-OBJECTS		= $(AMSRC2_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AMSRC2_OBJS:%.o=$(LINTS_DIR)/%.ln)
+MODULE		= ac97
+OBJECTS		= $(AC97_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AC97_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT	= $(AMSRC2_OBJS:%.o=%.ll)
+WARLOCK_OUT	= $(AC97_OBJS:%.o=%.ll)
 WARLOCK_OK	= $(MODULE).ok
+WLCMD_DIR	= $(UTSBASE)/common/io/warlock
 
 #
 #	Include common rules.
@@ -57,10 +55,7 @@ ALL_TARGET	= $(BINARY)
 LINT_TARGET	= $(MODULE).lint
 INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
 
-#
-#	Depends on misc/amsrc2, misc/audiosup, and misc/mixer
-#
-LDFLAGS		+= -dy -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS		+= -dy -Ndrv/audio
 
 #
 #	Default build targets.
@@ -89,22 +84,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-WARLOCK		= warlock
-WLCC		= wlcc
-TOUCH		= touch
-TEST		= test
-
-#
-#	Warlock rules
-#
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT)
-	$(TOUCH) $(WARLOCK_OK) 
-
-%.ll:	$(UTSBASE)/common/io/audio/sada/amsrc2/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
diff --git a/usr/src/uts/intel/audio/Makefile b/usr/src/uts/intel/audio/Makefile
new file mode 100644
index 0000000000..e9c4ca1bc3
--- /dev/null
+++ b/usr/src/uts/intel/audio/Makefile
@@ -0,0 +1,80 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/intel/audio/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audio support module.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audio
+OBJECTS		= $(AUDIO_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIO_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/impl
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/intel/Makefile.intel
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/intel/Makefile.targ
diff --git a/usr/src/uts/intel/audio810/Makefile b/usr/src/uts/intel/audio810/Makefile
index d1714978f0..a3d3df4247 100644
--- a/usr/src/uts/intel/audio810/Makefile
+++ b/usr/src/uts/intel/audio810/Makefile
@@ -19,14 +19,12 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/audio810/Makefile
 #
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of Metropolis audio driver
 #	(audio810) kernel module.
 #
@@ -42,34 +40,17 @@ MODULE		= audio810
 OBJECTS		= $(AUDIO810_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIO810_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audio810
-WARLOCK_OUT	= $(AUDIO810_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audio810
 
 #
 #	Include common rules.
 #
 include $(UTSBASE)/intel/Makefile.intel
 
-# 
-#	Overrides, lint pass one enforcement 
-# 
-CFLAGS		+= $(CCVERBOSE)
-DEBUG_FLGS	=
-$(NOT_RELEASE_BUILD)DEBUG_DEFS	+= $(DEBUG_FLGS)
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
 #
-#	Depends on misc/audiosup
+#	Depends on drv/audio and misc/ac97
 #
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
 
 #
 #	Define targets
@@ -88,10 +69,8 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS)
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -105,47 +84,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/intel/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/intel/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_standalone \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audio810_with_sada.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audio810_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audio810/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audio810.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audio810.wlcmd $(WARLOCK_OUT) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../../intel/warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../../intel/audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../../intel/amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../../intel/mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/intel/audioens/Makefile b/usr/src/uts/intel/audioens/Makefile
new file mode 100644
index 0000000000..2a03ba5668
--- /dev/null
+++ b/usr/src/uts/intel/audioens/Makefile
@@ -0,0 +1,82 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/intel/audioens/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audioens driver.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audioens
+OBJECTS		= $(AUDIOENS_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIOENS_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audioens
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/intel/Makefile.intel
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/intel/Makefile.targ
diff --git a/usr/src/uts/intel/audiohd/Makefile b/usr/src/uts/intel/audiohd/Makefile
index 4659db6c74..a0487ca3f7 100644
--- a/usr/src/uts/intel/audiohd/Makefile
+++ b/usr/src/uts/intel/audiohd/Makefile
@@ -19,14 +19,11 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/audiohd/Makefile
 #
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of high definition audio
 #	driver (audiohd) kernel module.
 #
@@ -42,10 +39,7 @@ MODULE		= audiohd
 OBJECTS		= $(AUDIOHD_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIOHD_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audiohd
-WARLOCK_OUT 	= $(AUDIOHD_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audiohd
 
 #
 #	Include common rules.
@@ -67,10 +61,7 @@ $(NOT_RELEASE_BUILD)DEBUG_DEFS	+= $(DEBUG_FLGS)
 LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
 LINTTAGS	+= -erroff=E_ASSIGN_NARROW_CONV
 
-#
-#	Depends on misc/audiosup
-#
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS		+= -dy -Ndrv/audio
 
 #
 #	Define targets
@@ -92,7 +83,6 @@ clean:		$(CLEAN_DEPS)
 		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -106,47 +96,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/intel/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/intel/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files  \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audiohd_with_sada.wlcmd warlock_standalone
-	$(WARLOCK) -c $(WLCMD_DIR)/audiohd_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audiohd/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiohd.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiohd.wlcmd $(WARLOCK_OUT) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../../intel/warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../../intel/audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../../intel/amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../../intel/mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/intel/audioixp/Makefile b/usr/src/uts/intel/audioixp/Makefile
index 414ee60224..ead1d79da8 100644
--- a/usr/src/uts/intel/audioixp/Makefile
+++ b/usr/src/uts/intel/audioixp/Makefile
@@ -19,14 +19,12 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/audioixp/Makefile
 #
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of high definition audio
 #	driver (audioixp) kernel module.
 #
@@ -42,10 +40,7 @@ MODULE		= audioixp
 OBJECTS		= $(AUDIOIXP_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIOIXP_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audioixp
-WARLOCK_OUT 	= $(AUDIOIXP_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audioixp
 
 #
 #	Include common rules.
@@ -62,7 +57,7 @@ $(NOT_RELEASE_BUILD)DEBUG_DEFS	+= $(DEBUG_FLGS)
 #
 #	Depends on misc/audiosup
 #
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
 
 #
 #	Define targets
@@ -72,13 +67,6 @@ LINT_TARGET	= $(MODULE).lint
 INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
 
 #
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
-#
 #	Default build targets.
 #
 .KEEP_STATE:
@@ -88,10 +76,8 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS)
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -105,50 +91,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-# NOTE: there will be warnings about q_lock which is the simulated
-# rwlock of the taskq framework
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/intel/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/intel/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_standalone \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audioixp_with_sada.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audioixp_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audioixp/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audioixp.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audioixp.wlcmd $(WARLOCK_OUT) \
-	-l ../../intel/warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../../intel/warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../../intel/audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../../intel/amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../../intel/mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/intel/audiopci/Makefile b/usr/src/uts/intel/audiopci/Makefile
new file mode 100644
index 0000000000..ca07ef76cd
--- /dev/null
+++ b/usr/src/uts/intel/audiopci/Makefile
@@ -0,0 +1,82 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/intel/audiopci/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audiopci driver.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audiopci
+OBJECTS		= $(AUDIOPCI_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIOPCI_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audiopci
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/intel/Makefile.intel
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+LDFLAGS		+= -dy -Ndrv/audio
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/intel/Makefile.targ
diff --git a/usr/src/uts/intel/usb_ac_dacf/Makefile b/usr/src/uts/intel/audiovia823x/Makefile
index 0c119ec8a1..4f6dbb58be 100644
--- a/usr/src/uts/intel/usb_ac_dacf/Makefile
+++ b/usr/src/uts/intel/audiovia823x/Makefile
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,29 +19,28 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# uts/intel/usb_ac_dacf/Makefile
+# uts/intel/audiovia823x/Makefile
 #
-#	This makefile drives the production of the usb_ac driver
-#	kernel module. Intel architecture dependent
 #
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
+#	This makefile drives the production of the Via 823x audio driver
+#	(audiovia823x) kernel module.
+#
 #
 #	Path to the base of the uts directory tree (usually /usr/src/uts).
 #
-UTSBASE	= ../..
+UTSBASE = ../..
 
 #
 #	Define the module and object file sets.
 #
-MODULE		= usb_ac_dacf
-OBJECTS		= $(USB_AC_DACF_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(USB_AC_DACF_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_DACF_DIR)/$(MODULE)
+MODULE		= audiovia823x
+OBJECTS		= $(AUDIOVIA823X_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIOVIA823X_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audiovia823x
 
 #
 #	Include common rules.
@@ -50,23 +48,18 @@ ROOTMODULE	= $(ROOT_DACF_DIR)/$(MODULE)
 include $(UTSBASE)/intel/Makefile.intel
 
 #
-#	Define targets
+#	Depends on drv/audio and misc/ac97
 #
-ALL_TARGET	= $(BINARY)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
 
 #
-#	Override defaults to build a unique, local modstubs.o.
+#	Define targets
 #
-MODSTUBS_DIR	 = $(OBJS_DIR)
-CLEANFILES	+= $(MODSTUBS_O)
+ALL_TARGET	= $(BINARY) $(SRC_CONFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
 
 #
-# depends on misc/usba
-#
-LDFLAGS += -dy -Nmisc/usba -Nmisc/amsrc2 -Nmisc/audiosup -Nmisc/mixer
-#
 #	Default build targets.
 #
 .KEEP_STATE:
@@ -91,4 +84,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/intel/Makefile.targ
-
diff --git a/usr/src/uts/intel/austr/Makefile b/usr/src/uts/intel/austr/Makefile
new file mode 100644
index 0000000000..e839641126
--- /dev/null
+++ b/usr/src/uts/intel/austr/Makefile
@@ -0,0 +1,85 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/intel/austr/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audio streams module.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= austr
+OBJECTS		= $(AUSTR_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUSTR_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/impl
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/intel/Makefile.intel
+
+#
+#	Depends on drv/audio
+#
+LDFLAGS		+= -dy -Ndrv/audio
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/intel/Makefile.targ
diff --git a/usr/src/uts/intel/mixer/Makefile b/usr/src/uts/intel/mixer/Makefile
deleted file mode 100644
index cf6b5094a5..0000000000
--- a/usr/src/uts/intel/mixer/Makefile
+++ /dev/null
@@ -1,154 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# uts/intel/mixer/Makefile
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-#	This makefile drives the production of the audio mixer (mixer)
-#	Audio Personality Module.
-#
-#ident	"%Z%%M%	%I%	%E% SMI"
-
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= mixer
-OBJECTS		= $(MIXER_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(MIXER_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT	= $(MIXER_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR	= $(UTSBASE)/common/io/warlock
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/intel/Makefile.intel
-
-#
-#	Define targets
-#
-ALL_TARGET	= $(BINARY)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
-
-#
-#	Depends on misc/audiosup
-#
-LDFLAGS		+= -dy -Nmisc/audiosup
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-LINTTAGS	+= -erroff=E_PTRDIFF_OVERFLOW
-LINTTAGS	+= -erroff=E_ASSIGN_NARROW_CONV
-LINTTAGS	+= -erroff=E_SUSPICIOUS_COMPARISON
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		$(DEF_DEPS)
-
-all:		$(ALL_DEPS)
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-lint:		$(LINT_DEPS)
-
-modlintlib:	$(MODLINTLIB_DEPS)
-
-clean.lint:	$(CLEAN_LINT_DEPS)
-
-install:	$(INSTALL_DEPS)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/intel/audiosup/%.ll)
-AMSRC1_FILES = $(AMSRC1_OBJS:%.o=$(CLOSED)/uts/intel/amsrc1/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
-
-WARLOCK_TARGETS = $(WARLOCK_OUT) warlock_ddi.files warlock_audiosup.files \
-	warlock_standalone $(WLCMD_DIR)/mixer_with_audiosup.wlcmd
-
-$(CLOSED_BUILD)WARLOCK_CLOSED = warlock_closed
-
-warlock: warlock_open $(WARLOCK_CLOSED)
-	$(TOUCH) $(WARLOCK_OK)
-
-warlock_open: $(WARLOCK_TARGETS) warlock_amsrc2.files
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer_with_audiosup.wlcmd $(WARLOCK_OUT) \
-	$(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_closed: $(WARLOCK_TARGETS) warlock_amsrc1.files
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer_with_audiosup.wlcmd \
-	$(WARLOCK_OUT) $(AUDIO_SUP_FILES) $(AMSRC1_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-
-%.ll: $(UTSBASE)/common/io/audio/sada/mixer/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone:  $(WARLOCK_OUT) warlock_ddi.files \
-	$(WLCMD_DIR)/mixer.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc1.files:
-	@cd $(CLOSED)/uts/intel/amsrc1; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/intel/os/dacf.conf b/usr/src/uts/intel/os/dacf.conf
index 9ccb1b8282..0475a5c840 100644
--- a/usr/src/uts/intel/os/dacf.conf
+++ b/usr/src/uts/intel/os/dacf.conf
@@ -19,14 +19,13 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 #
 # NOTICE: This file contains important KERNEL STATE.  Do not edit this file.
 # Its format and contents are subject to change in future releases of Solaris.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
 
 #
 # keyboard and mouse configuration rules
@@ -61,13 +60,6 @@ driver-minorname="mouse8042:internal_mouse" consconfig_dacf:ms_config post-attac
 driver-minorname="mouse8042:internal_mouse" consconfig_dacf:ms_config pre-detach - pushmod="vuid3ps2"
 
 #
-# Configure and/or unconfigure usb audio devices.
-# usb_ac as a mux for hid and usb_as devices.
-#
-driver-minorname="usb_ac:mux" usb_ac_dacf:usb_audio_config post-attach -
-driver-minorname="usb_ac:mux" usb_ac_dacf:usb_audio_config pre-detach -
-
-#
 # Configure and/or unconfigure DDI_NT_NET devices.
 #
 minor-nodetype="ddi_network" net_dacf:net_config post-attach -
diff --git a/usr/src/uts/intel/usb_ac/Makefile b/usr/src/uts/intel/usb_ac/Makefile
index cfeb9d8525..29ce6e73fb 100644
--- a/usr/src/uts/intel/usb_ac/Makefile
+++ b/usr/src/uts/intel/usb_ac/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/usb_ac/Makefile
@@ -42,7 +42,7 @@ OBJECTS		= $(USB_AC_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(USB_AC_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
 CONF_SRCDIR	= $(UTSBASE)/common/io/usb/clients/audio/usb_ac
-WARLOCK_OUT	= $(USB_AC_OBJS:%.o=%.ll) usb_ac_dacf.ll
+WARLOCK_OUT	= $(USB_AC_OBJS:%.o=%.ll)
 WARLOCK_OK	= $(MODULE).ok
 WLCMD_DIR	= $(UTSBASE)/common/io/warlock
 
@@ -67,14 +67,7 @@ CLEANFILES	+= $(MODSTUBS_O)
 #
 # depends on misc/usba
 #
-LDFLAGS += -dy -Nmisc/usba -Nmisc/audiosup -Nmisc/mixer
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
+LDFLAGS += -dy -Nmisc/usba -Ndrv/audio
 
 #
 #	Default build targets.
@@ -119,9 +112,6 @@ USBA_FILES	= $(USBA_OBJS:%.o=../usba/%.ll)
 UHCI_FILES	= $(UHCI_OBJS:%.o=../uhci/%.ll)
 OHCI_FILES	= $(OHCI_OBJS:%.o=../ohci/%.ll)
 EHCI_FILES	= $(EHCI_OBJS:%.o=../ehci/%.ll)
-MIXER_FILES	= $(MIXER_OBJS:%.o=../mixer/%.ll)
-AUDIO_SUP_FILES	= $(AUDIO_SUP_OBJS:%.o=../audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
 
 
 #
@@ -138,7 +128,6 @@ warlock_with_sada: $(WLCMD_DIR)/usb_ac_with_sada.wlcmd \
 	$(WARLOCK_OUT) \
 	warlock_audiosup.files  warlock_mixer.files warlock_amsrc2.files 
 	$(WARLOCK) -c $(WLCMD_DIR)/usb_ac_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
 	-l ../warlock/ddi_dki_impl.ll
 
 
diff --git a/usr/src/uts/intel/usb_ah/Makefile b/usr/src/uts/intel/usb_ah/Makefile
index 534aa9e7e9..eaa8cb675d 100644
--- a/usr/src/uts/intel/usb_ah/Makefile
+++ b/usr/src/uts/intel/usb_ah/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/usb_ah/Makefile
@@ -66,7 +66,7 @@ CLEANFILES	+= $(MODSTUBS_O)
 #
 # depends on misc/usba
 #
-LDFLAGS += -dy -Nmisc/usba -Nhidparser -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS += -dy -Nmisc/usba -Nhidparser -Ndrv/usb_ac
 
 #
 # For now, disable these lint checks; maintainers should endeavor
diff --git a/usr/src/uts/intel/usb_as/Makefile b/usr/src/uts/intel/usb_as/Makefile
index f4a62920ee..9d82ee38a0 100644
--- a/usr/src/uts/intel/usb_as/Makefile
+++ b/usr/src/uts/intel/usb_as/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/usb_as/Makefile
@@ -63,11 +63,7 @@ INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
 MODSTUBS_DIR	 = $(OBJS_DIR)
 CLEANFILES	+= $(MODSTUBS_O)
 
-#
-# do not remove depends on amsrc2. If removed, amsrc2 gets unloaded when
-# usb_ac_dacf gets unloaded
-#
-LDFLAGS		+= -dy -Nmisc/usba -Nmisc/audiosup -Nmisc/mixer -Nmisc/amsrc2
+LDFLAGS		+= -dy -Nmisc/usba -Ndrv/usb_ac
 
 #
 # For now, disable these lint checks; maintainers should endeavor
@@ -120,22 +116,18 @@ USBA_FILES	= $(USBA_OBJS:%.o=../usba/%.ll)
 UHCI_FILES	= $(UHCI_OBJS:%.o=../uhci/%.ll)
 OHCI_FILES	= $(OHCI_OBJS:%.o=../ohci/%.ll)
 EHCI_FILES	= $(EHCI_OBJS:%.o=../ehci/%.ll)
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/intel/mixer/%.ll)
 AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/intel/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/intel/amsrc2/%.ll)
 
 #	Warlock targets
 #
 warlock: $(WARLOCK_OK)
 
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files  \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
+$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_audiosup.files \
 	warlock_standalone warlock_with_usba  warlock_with_sada
 	$(TOUCH) $(WARLOCK_OK)
 
 warlock_with_sada: $(WLCMD_DIR)/usb_as_with_sada.wlcmd \
-	$(WARLOCK_OUT) \
-	warlock_audiosup.files warlock_amsrc2.files warlock_mixer.files
+	$(WARLOCK_OUT) warlock_audiosup.files 
 	$(WARLOCK) -c $(WLCMD_DIR)/usb_as_with_sada.wlcmd $(WARLOCK_OUT) \
 	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
 	-l ../warlock/ddi_dki_impl.ll
@@ -171,9 +163,3 @@ warlock_ddi.files:
 
 warlock_audiosup.files:
 	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/intel/warlock/Makefile b/usr/src/uts/intel/warlock/Makefile
index fbefb9b95d..2b877db841 100644
--- a/usr/src/uts/intel/warlock/Makefile
+++ b/usr/src/uts/intel/warlock/Makefile
@@ -20,7 +20,7 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/intel/warlock/Makefile
@@ -51,7 +51,7 @@ include $(UTSBASE)/intel/Makefile.intel
 #
 #	lock_lint rules
 #
-all:	warlock warlock.1394 warlock.audio warlock.ecpp warlock.scsi \
+all:	warlock warlock.1394 warlock.ecpp warlock.scsi \
 	warlock.usb warlock.ib warlock.sata warlock.sdcard warlock.wc
 
 warlock: $(MODULE).ok 
@@ -101,21 +101,6 @@ warlock.1394:
 warlock.ecpp:
 	@cd ../ecpp; $(MAKE) clean; $(MAKE) warlock
 
-warlock.audio:
-	@cd ../audiosup; $(MAKE) clean; $(MAKE) warlock
-	@cd ../mixer; $(MAKE) clean; $(MAKE) warlock
-	@cd ../amsrc2; $(MAKE) clean; $(MAKE) warlock
-	@cd ../../intel/audio810; $(MAKE) clean; $(MAKE) warlock
-	@cd ../../intel/audiohd; $(MAKE) clean; $(MAKE) warlock
-	@cd ../../intel/audioixp; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_ac; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_as; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_ah; $(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/intel/amsrc1; $(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/intel/audiovia823x; \
-		$(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/intel/audioens; $(MAKE) clean; $(MAKE) warlock
-
 warlock.ib:
 	@cd ../ibmf; $(MAKE) clean; $(MAKE) warlock
 	@cd ../ib; $(MAKE) clean; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/Makefile.sparc.shared b/usr/src/uts/sparc/Makefile.sparc.shared
index a452f8ef67..4829b38c36 100644
--- a/usr/src/uts/sparc/Makefile.sparc.shared
+++ b/usr/src/uts/sparc/Makefile.sparc.shared
@@ -261,7 +261,7 @@ $(CLOSED_BUILD)CLOSED_DRV_KMODS	+= ixgb
 #
 #	Machine Specific Driver Modules (/kernel/drv):
 #
-DRV_KMODS	+= audio1575 audio810 audiohd audiocs audiots
+DRV_KMODS	+= audio audio1575 audioens audiocs audiots audiopci austr
 DRV_KMODS	+= bge bpp dmfe eri esp fas hme qfe
 DRV_KMODS	+= openeepr options sd ses st
 DRV_KMODS	+= ssd
@@ -306,8 +306,6 @@ DRV_KMODS	+= ii rdc rdcsrv rdcstub
 DRV_KMODS	+= iscsi
 DRV_KMODS	+= emlxs
 
-$(CLOSED_BUILD)CLOSED_DRV_KMODS	+= audioens
-$(CLOSED_BUILD)CLOSED_DRV_KMODS	+= audiovia823x
 $(CLOSED_BUILD)CLOSED_DRV_KMODS	+= ifp
 $(CLOSED_BUILD)CLOSED_DRV_KMODS	+= uata
 $(CLOSED_BUILD)CLOSED_DRV_KMODS	+= usbser_edge
@@ -378,7 +376,7 @@ SYS_KMODS	+= doorfs pset acctctl portfs
 #
 #	'User' Modules (/kernel/misc):
 #
-MISC_KMODS	+= amsrc2 audiosup mixer
+MISC_KMODS	+= ac97
 MISC_KMODS	+= bignum
 MISC_KMODS	+= consconfig gld ipc nfs_dlboot nfssrv scsi
 MISC_KMODS	+= strplumb swapgeneric tlimod
@@ -420,7 +418,6 @@ MISC_KMODS	+= sda
 MISC_KMODS	+= spuni
 MISC_KMODS	+= hwa1480_fw uwba
 
-$(CLOSED_BUILD)CLOSED_MISC_KMODS	+= amsrc1
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= klmmod klmops
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= scsi_vhci_f_asym_lsi
 $(CLOSED_BUILD)CLOSED_MISC_KMODS	+= scsi_vhci_f_asym_emc
@@ -483,7 +480,7 @@ $(CLOSED_BUILD)CLOSED_XMODS =		\
 #
 #	'Dacf' Modules (/kernel/dacf):
 #
-DACF_KMODS	+= usb_ac_dacf net_dacf
+DACF_KMODS	+= net_dacf
 
 #
 #	MAC-Type Plugin Modules (/kernel/mac)
diff --git a/usr/src/uts/intel/audiosup/Makefile b/usr/src/uts/sparc/ac97/Makefile
index 2285741889..84b7c69ea7 100644
--- a/usr/src/uts/intel/audiosup/Makefile
+++ b/usr/src/uts/sparc/ac97/Makefile
@@ -19,15 +19,13 @@
 # CDDL HEADER END
 #
 #
-# uts/intel/audiosup/Makefile
+# uts/sparc/ac97/Makefile
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#	This makefile drives the production of the audio support (audiosup)
-#	module.
+#	This makefile drives the production of the ac97 support module.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 #
 #	Path to the base of the uts directory tree (usually /usr/src/uts).
@@ -37,18 +35,18 @@ UTSBASE = ../..
 #
 #	Define the module and object file sets.
 #
-MODULE		= audiosup
-OBJECTS		= $(AUDIO_SUP_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AUDIO_SUP_OBJS:%.o=$(LINTS_DIR)/%.ln)
+MODULE		= ac97
+OBJECTS		= $(AC97_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AC97_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT	= $(AUDIO_SUP_OBJS:%.o=%.ll)
+WARLOCK_OUT	= $(AC97_OBJS:%.o=%.ll)
 WARLOCK_OK	= $(MODULE).ok
 WLCMD_DIR	= $(UTSBASE)/common/io/warlock
 
 #
 #	Include common rules.
 #
-include $(UTSBASE)/intel/Makefile.intel
+include $(UTSBASE)/sparc/Makefile.sparc
 
 #
 #	Define targets
@@ -57,12 +55,7 @@ ALL_TARGET	= $(BINARY)
 LINT_TARGET	= $(MODULE).lint
 INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
 
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
+LDFLAGS		+= -dy -Ndrv/audio
 
 #
 #	Default build targets.
@@ -90,28 +83,4 @@ install:	$(INSTALL_DEPS)
 #
 #	Include common targets.
 #
-include $(UTSBASE)/intel/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiosup.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiosup.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/framework/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
+include $(UTSBASE)/sparc/Makefile.targ
diff --git a/usr/src/uts/sparc/amsrc2/main.c b/usr/src/uts/sparc/amsrc2/main.c
deleted file mode 100644
index 613e960581..0000000000
--- a/usr/src/uts/sparc/amsrc2/main.c
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2002 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * main.c - A pseudo-kernel to use when analyzing am_src2.c with warlock.
- *
- * The main idea here is to represent all of the ways that the kernel can
- * call into the amsrc2, so that warlock has the correct view of the call
- * graph.
- */
-
-#include <sys/modctl.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/impl/am_src2_impl.h>
-
-am_ad_src_entry_t *src = &am_src2;
-
-main() {
-
-	_init();
-	_fini();
-	_info(NULL);
-
-	/*
-	 * ad_src_init() is called by am_open() before audio may begin playing
-	 * or recording. So this can be viewed as single threaded.
-	 */
-	src->ad_src_init(NULL, 0);
-
-	_NOTE(COMPETING_THREADS_NOW)
-
-	src->ad_src_adjust(NULL, 0, 0);
-	src->ad_src_convert(NULL, 0, 0, NULL, NULL, NULL, NULL);
-	src->ad_src_size(NULL, NULL, 0, 0, 0);
-	src->ad_src_update(NULL, NULL, NULL, NULL, 0);
-
-	_NOTE(NO_COMPETING_THREADS_NOW)
-
-	/*
-	 * ad_src_exit() is called by am_close() only when play and record have
-	 * stopped, thus it also is single threaded.
-	 */
-	src->ad_src_exit(NULL, 0);
-
-	return (0);
-}
diff --git a/usr/src/uts/sparc/audio/Makefile b/usr/src/uts/sparc/audio/Makefile
new file mode 100644
index 0000000000..2bb577fac6
--- /dev/null
+++ b/usr/src/uts/sparc/audio/Makefile
@@ -0,0 +1,80 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/sparc/audio/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audio support module.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audio
+OBJECTS		= $(AUDIO_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIO_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/impl
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/sparc/Makefile.sparc
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/sparc/Makefile.targ
diff --git a/usr/src/uts/sparc/audio1575/Makefile b/usr/src/uts/sparc/audio1575/Makefile
index 0abcdcdd1c..76706de77e 100644
--- a/usr/src/uts/sparc/audio1575/Makefile
+++ b/usr/src/uts/sparc/audio1575/Makefile
@@ -19,13 +19,11 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/audio1575/Makefile
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of the M1575 audio driver
 #	(audio1575) kernel module.
 #
@@ -41,10 +39,7 @@ MODULE		= audio1575
 OBJECTS		= $(AUDIO1575_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIO1575_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/sun/io/audio/sada/drv/audio1575
-WARLOCK_OUT 	= $(AUDIO1575_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audio1575
 
 #
 #	Include common rules.
@@ -57,9 +52,9 @@ include $(UTSBASE)/sparc/Makefile.sparc
 CFLAGS		+= $(CCVERBOSE)
 
 #
-#	Depends on misc/audiosup
+#	Depends on drv/audio and misc/ac97
 #
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/mixer -Nmisc/audiosup
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
 
 #
 #	Define targets
@@ -76,13 +71,6 @@ DEF_BUILDS	= $(DEF_BUILDSONLY64)
 CLEANLINTFILES	+= $(LINT32_FILES)
 
 #
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
-#
 #	Default build targets.
 #
 .KEEP_STATE:
@@ -92,10 +80,8 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS) 
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -109,51 +95,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-# NOTE: there will be warnings about q_lock which is the simulated
-# rwlock of the taskq framework
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_standalone \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audio1575_with_sada.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audio1575_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/sun/io/audio/sada/drv/audio1575/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files \
-	$(WLCMD_DIR)/audio1575.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audio1575.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audio810/Makefile b/usr/src/uts/sparc/audio810/Makefile
deleted file mode 100644
index a087988ac4..0000000000
--- a/usr/src/uts/sparc/audio810/Makefile
+++ /dev/null
@@ -1,155 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# uts/sparc/audio810/Makefile
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-#	This makefile drives the production of Metropolis audio driver
-#	(audio810) kernel module.
-#
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= audio810
-OBJECTS		= $(AUDIO810_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AUDIO810_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audio810
-WARLOCK_OUT 	= $(AUDIO810_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-# 
-#	Overrides, lint pass one enforcement 
-# 
-CFLAGS		+= $(CCVERBOSE)
-
-#
-#	Depends on misc/audiosup, misc/mixer
-#
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/mixer -Nmisc/audiosup
-
-#
-#	Define targets
-#
-ALL_TARGET	= $(BINARY) $(SRC_CONFILE)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE_64)
-
-#
-#       Overrides
-#
-ALL_BUILDS	= $(ALL_BUILDSONLY64)
-DEF_BUILDS	= $(DEF_BUILDSONLY64)
-CLEANLINTFILES	+= $(LINT32_FILES)
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		$(DEF_DEPS)
-
-all:		$(ALL_DEPS)
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-lint:		$(LINT_DEPS)
-
-modlintlib:	$(MODLINTLIB_DEPS) lint32
-
-clean.lint:	$(CLEAN_LINT_DEPS)
-
-install:	$(INSTALL_DEPS)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audio810_with_sada.wlcmd warlock_standalone
-	$(WARLOCK) -c $(WLCMD_DIR)/audio810_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audio810/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audio810.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audio810.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audiocs/Makefile b/usr/src/uts/sparc/audiocs/Makefile
index 7371d07fa8..f4fd53c27f 100644
--- a/usr/src/uts/sparc/audiocs/Makefile
+++ b/usr/src/uts/sparc/audiocs/Makefile
@@ -21,11 +21,9 @@
 #
 # uts/sparc/audiocs/Makefile
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of the Crystal 4231
 #	audio driver (audiocs) kernel module.
 #
@@ -42,10 +40,7 @@ MODULE		= audiocs
 OBJECTS		= $(AUDIOCS_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIOCS_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/sun/io/audio/sada/drv/audiocs
-WARLOCK_OUT 	= $(AUDIOCS_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/sun/io/audio/drv/audiocs
 
 #
 #	Include common rules.
@@ -65,16 +60,9 @@ INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
 CFLAGS		+= $(CCVERBOSE)
 
 #
-#	Depends on misc/audiosup and misc/mixer
+#	Depends on misc/audiosup
 #
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/audiosup -Nmisc/mixer
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
+LDFLAGS		+= -dy -Ndrv/audio
 
 #
 #	Default build targets.
@@ -86,10 +74,8 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS)
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -103,47 +89,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	warlock_standalone $(WLCMD_DIR)/audiocs_with_sada.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiocs_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/sun/io/audio/sada/drv/audiocs/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiocs.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiocs.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audioens/Makefile b/usr/src/uts/sparc/audioens/Makefile
new file mode 100644
index 0000000000..da0b7b2125
--- /dev/null
+++ b/usr/src/uts/sparc/audioens/Makefile
@@ -0,0 +1,82 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/sparc/audioens/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audioens driver.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audioens
+OBJECTS		= $(AUDIOENS_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIOENS_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audioens
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/sparc/Makefile.sparc
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+LDFLAGS		+= -dy -Ndrv/audio -Nmisc/ac97
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/sparc/Makefile.targ
diff --git a/usr/src/uts/sparc/audiohd/Makefile b/usr/src/uts/sparc/audiohd/Makefile
deleted file mode 100644
index 38c21c1f70..0000000000
--- a/usr/src/uts/sparc/audiohd/Makefile
+++ /dev/null
@@ -1,156 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# uts/sparc/audiohd/Makefile
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-#	This makefile drives the production of high definition audio
-#	driver (audiohd) kernel module.
-#
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= audiohd
-OBJECTS		= $(AUDIOHD_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AUDIOHD_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audiohd
-WARLOCK_OUT = $(AUDIOHD_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-# 
-#	Overrides, lint pass one enforcement 
-# 
-CFLAGS		+= $(CCVERBOSE)
-
-#
-#	Depends on misc/audiosup, misc/mixer
-#
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/mixer -Nmisc/audiosup
-
-#
-#	Define targets
-#
-ALL_TARGET	= $(BINARY) $(SRC_CONFILE)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE_64)
-
-#
-#       Overrides
-#
-ALL_BUILDS	= $(ALL_BUILDSONLY64)
-DEF_BUILDS	= $(DEF_BUILDSONLY64)
-CLEANLINTFILES	+= $(LINT32_FILES)
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-LINTTAGS	+= -erroff=E_ASSIGN_NARROW_CONV
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		$(DEF_DEPS)
-
-all:		$(ALL_DEPS)
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-lint:		$(LINT_DEPS)
-
-modlintlib:	$(MODLINTLIB_DEPS) lint32
-
-clean.lint:	$(CLEAN_LINT_DEPS)
-
-install:	$(INSTALL_DEPS)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files  \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audiohd_with_sada.wlcmd warlock_standalone
-	$(WARLOCK) -c $(WLCMD_DIR)/audiohd_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audiohd/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiohd.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiohd.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audioixp/Makefile b/usr/src/uts/sparc/audioixp/Makefile
deleted file mode 100644
index bace3949f0..0000000000
--- a/usr/src/uts/sparc/audioixp/Makefile
+++ /dev/null
@@ -1,117 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-# uts/sparc/audioixp/Makefile
-#
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
-#	This makefile drives the production of hight definition audio
-#	driver (audioixp) kernel module.
-#
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= audioixp 
-WARLOCK_OUT 	= $(AUDIOIXP_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		warlock
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audioixp_with_sada.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audioixp_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audioixp/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audioixp.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audioixp.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audiopci/Makefile b/usr/src/uts/sparc/audiopci/Makefile
new file mode 100644
index 0000000000..fcf95b3443
--- /dev/null
+++ b/usr/src/uts/sparc/audiopci/Makefile
@@ -0,0 +1,82 @@
+#
+# CDDL HEADER START
+#
+# The contents of this file are subject to the terms of the
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
+#
+# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+# or http://www.opensolaris.org/os/licensing.
+# See the License for the specific language governing permissions
+# and limitations under the License.
+#
+# When distributing Covered Code, include this CDDL HEADER in each
+# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+# If applicable, add the following below this CDDL HEADER, with the
+# fields enclosed by brackets "[]" replaced with your own identifying
+# information: Portions Copyright [yyyy] [name of copyright owner]
+#
+# CDDL HEADER END
+#
+#
+# uts/sparc/audiopci/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+# Use is subject to license terms.
+#
+#	This makefile drives the production of the audiopci driver.
+#
+
+#
+#	Path to the base of the uts directory tree (usually /usr/src/uts).
+#
+UTSBASE = ../..
+
+#
+#	Define the module and object file sets.
+#
+MODULE		= audiopci
+OBJECTS		= $(AUDIOPCI_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUDIOPCI_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audiopci
+
+#
+#	Include common rules.
+#
+include $(UTSBASE)/sparc/Makefile.sparc
+
+#
+#	Define targets
+#
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
+LINT_TARGET	= $(MODULE).lint
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
+
+LDFLAGS		+= -dy -Ndrv/audio
+
+#
+#	Default build targets.
+#
+.KEEP_STATE:
+
+def:		$(DEF_DEPS)
+
+all:		$(ALL_DEPS)
+
+clean:		$(CLEAN_DEPS)
+
+clobber:	$(CLOBBER_DEPS)
+
+lint:		$(LINT_DEPS)
+
+modlintlib:	$(MODLINTLIB_DEPS)
+
+clean.lint:	$(CLEAN_LINT_DEPS)
+
+install:	$(INSTALL_DEPS)
+
+#
+#	Include common targets.
+#
+include $(UTSBASE)/sparc/Makefile.targ
diff --git a/usr/src/uts/sparc/audiosup/Makefile b/usr/src/uts/sparc/audiosup/Makefile
deleted file mode 100644
index 47bbf5e7e1..0000000000
--- a/usr/src/uts/sparc/audiosup/Makefile
+++ /dev/null
@@ -1,128 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# uts/sparc/audiosup/Makefile
-#
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-#	This makefile drives the production of the audio support (audiosup)
-#	module.
-#
-#ident	"%Z%%M%	%I%	%E% SMI"
-
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= audiosup
-OBJECTS		= $(AUDIO_SUP_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AUDIO_SUP_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT = $(AUDIO_SUP_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
-
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-#
-#	Turn off -xspace so we get all optimizations.
-#
-SPACEFLAG	=
-
-#
-#	Define targets
-#
-ALL_TARGET	= $(BINARY)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
-
-# 
-#	Overrides, lint pass one enforcement 
-# 
-CFLAGS		+= $(CCVERBOSE)
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		$(DEF_DEPS)
-
-all:		$(ALL_DEPS)
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-lint:		$(LINT_DEPS)
-
-modlintlib:	$(MODLINTLIB_DEPS)
-
-clean.lint:	$(CLEAN_LINT_DEPS)
-
-install:	$(INSTALL_DEPS)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiosup.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiosup.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/framework/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/audiots/Makefile b/usr/src/uts/sparc/audiots/Makefile
index 94887e61d1..0bb9e744e7 100644
--- a/usr/src/uts/sparc/audiots/Makefile
+++ b/usr/src/uts/sparc/audiots/Makefile
@@ -19,13 +19,11 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/audiots/Makefile
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 #	This makefile drives the production of the T2 audio driver
 #	(audiots) kernel module.
 #
@@ -41,10 +39,7 @@ MODULE		= audiots
 OBJECTS		= $(AUDIOTS_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(AUDIOTS_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
-CONF_SRCDIR	= $(UTSBASE)/common/io/audio/sada/drv/audiots
-WARLOCK_OUT 	= $(AUDIOTS_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/drv/audiots
 
 #
 #	Include common rules.
@@ -59,7 +54,7 @@ CFLAGS		+= $(CCVERBOSE)
 #
 #	Depends on misc/audiosup
 #
-LDFLAGS		+= -dy -Nmisc/amsrc2 -Nmisc/mixer -Nmisc/audiosup
+LDFLAGS		+= -dy -N misc/ac97 -N drv/audio
 
 #
 #	Define targets
@@ -76,14 +71,6 @@ DEF_BUILDS	= $(DEF_BUILDSONLY64)
 CLEANLINTFILES	+= $(LINT32_FILES)
 
 #
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-LINTTAGS	+= -erroff=E_ASSIGN_NARROW_CONV
-
-#
 #	Default build targets.
 #
 .KEEP_STATE:
@@ -93,10 +80,8 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS)
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
 
 lint:		$(LINT_DEPS)
 
@@ -110,50 +95,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-# NOTE: there will be warnings about q_lock which is the simulated
-# rwlock of the taskq framework
-#
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_standalone \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
-	$(WLCMD_DIR)/audiots_with_sada.wlcmd warlock_standalone
-	$(WARLOCK) -c $(WLCMD_DIR)/audiots_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/drv/audiots/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone: $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/audiots.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/audiots.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/amsrc2/Makefile b/usr/src/uts/sparc/austr/Makefile
index 0d2672896c..f02bc322bb 100644
--- a/usr/src/uts/sparc/amsrc2/Makefile
+++ b/usr/src/uts/sparc/austr/Makefile
@@ -18,15 +18,14 @@
 #
 # CDDL HEADER END
 #
-# uts/sparc/amsrc2/Makefile
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# uts/sparc/austr/Makefile
+#
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#	This makefile drives the production of the audio mixer
-#	sample rate conversion routine #1 (amsrc2).
+#	This makefile drives the production of the audio streams module.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 #
 #	Path to the base of the uts directory tree (usually /usr/src/uts).
@@ -36,12 +35,11 @@ UTSBASE = ../..
 #
 #	Define the module and object file sets.
 #
-MODULE		= amsrc2
-OBJECTS		= $(AMSRC2_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(AMSRC2_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT	= $(AMSRC2_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
+MODULE		= austr
+OBJECTS		= $(AUSTR_OBJS:%=$(OBJS_DIR)/%)
+LINTS		= $(AUSTR_OBJS:%.o=$(LINTS_DIR)/%.ln)
+ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
+CONF_SRCDIR	= $(UTSBASE)/common/io/audio/impl
 
 #
 #	Include common rules.
@@ -49,33 +47,16 @@ WARLOCK_OK	= $(MODULE).ok
 include $(UTSBASE)/sparc/Makefile.sparc
 
 #
-#	Turn off -xspace so we get all optimizations.
+#	Depends on drv/audio
 #
-SPACEFLAG	=
+LDFLAGS		+= -dy -Ndrv/audio
 
 #
 #	Define targets
 #
-ALL_TARGET	= $(BINARY)
+ALL_TARGET	= $(BINARY) $(SRC_CONFFILE)
 LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
-
-#
-#       Overrides
-#
-ALL_BUILDS	= $(ALL_BUILDSONLY64)
-DEF_BUILDS	= $(DEF_BUILDSONLY64)
-CLEANLINTFILES	+= $(LINT32_FILES)
-
-# 
-#	Overrides, lint pass one enforcement 
-# 
-CFLAGS		+= $(CCVERBOSE)
-
-#
-#	Depends on misc/amsrc2, misc/audiosup, and misc/mixer
-#
-LDFLAGS		+= -dy -Nmisc/audiosup -Nmisc/mixer
+INSTALL_TARGET	= $(BINARY) $(ROOTMODULE) $(ROOT_CONFFILE)
 
 #
 #	Default build targets.
@@ -87,14 +68,12 @@ def:		$(DEF_DEPS)
 all:		$(ALL_DEPS)
 
 clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK) main.ll
 
 clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK) main.ll
 
 lint:		$(LINT_DEPS)
 
-modlintlib:	$(MODLINTLIB_DEPS) lint32
+modlintlib:	$(MODLINTLIB_DEPS)
 
 clean.lint:	$(CLEAN_LINT_DEPS)
 
@@ -104,27 +83,3 @@ install:	$(INSTALL_DEPS)
 #	Include common targets.
 #
 include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-WARLOCK		= warlock
-WLCC		= wlcc
-TOUCH		= touch
-TEST		= test
-
-#
-#	Warlock rules
-#
-
-warlock:	main.ll $(WARLOCK_OK)
-
-main.ll:	main.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ main.c
-
-$(WARLOCK_OK):	$(WARLOCK_OUT)
-	$(WARLOCK) $(WARLOCK_OUT) -l main.ll
-	$(TOUCH) $(WARLOCK_OK) 
-
-%.ll:	$(UTSBASE)/common/io/audio/sada/amsrc2/%.c main.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
diff --git a/usr/src/uts/sparc/mixer/Makefile b/usr/src/uts/sparc/mixer/Makefile
deleted file mode 100644
index 0f23f2e6b0..0000000000
--- a/usr/src/uts/sparc/mixer/Makefile
+++ /dev/null
@@ -1,156 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License (the "License").
-# You may not use this file except in compliance with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# uts/sparc/mixer/Makefile
-#
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-#	This makefile drives the production of the audio mixer (mixer)
-#	Audio Personality Module.
-#
-#ident	"%Z%%M%	%I%	%E% SMI"
-
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE = ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= mixer
-OBJECTS		= $(MIXER_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(MIXER_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_MISC_DIR)/$(MODULE)
-WARLOCK_OUT = $(MIXER_OBJS:%.o=%.ll)
-WARLOCK_OK	= $(MODULE).ok
-WLCMD_DIR       = $(UTSBASE)/common/io/warlock
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-#
-#	Turn off -xspace so we get all optimizations.
-#
-SPACEFLAG	=
-
-#
-#	Define targets
-#
-ALL_TARGET	= $(BINARY)
-LINT_TARGET	= $(MODULE).lint
-INSTALL_TARGET	= $(BINARY) $(ROOTMODULE)
-
-#
-#	Overrides, lint pass one enforcement
-#
-CFLAGS		+= $(CCVERBOSE)
-
-#
-# For now, disable these lint checks; maintainers should endeavor
-# to investigate and remove these for maximum lint coverage.
-# Please do not carry these forward to new Makefiles.
-#
-LINTTAGS	+= -erroff=E_SUSPICIOUS_COMPARISON
-LINTTAGS	+= -erroff=E_BAD_PTR_CAST_ALIGN
-LINTTAGS	+= -erroff=E_SUPPRESSION_DIRECTIVE_UNUSED
-LINTTAGS	+= -erroff=E_STATIC_UNUSED
-LINTTAGS	+= -erroff=E_PTRDIFF_OVERFLOW
-LINTTAGS	+= -erroff=E_ASSIGN_NARROW_CONV
-
-#
-#	Depends on misc/audiosup
-#
-LDFLAGS		+= -dy -Nmisc/audiosup
-
-#
-#	Default build targets.
-#
-.KEEP_STATE:
-
-def:		$(DEF_DEPS)
-
-all:		$(ALL_DEPS)
-
-clean:		$(CLEAN_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-clobber:	$(CLOBBER_DEPS)
-		$(RM) $(WARLOCK_OUT) $(WARLOCK_OK)
-
-lint:		$(LINT_DEPS)
-
-modlintlib:	$(MODLINTLIB_DEPS)
-
-clean.lint:	$(CLEAN_LINT_DEPS)
-
-install:	$(INSTALL_DEPS)
-
-#
-#	Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
-
-#
-#	Defines for local commands.
-#
-TEST		= test
-WLCC		= wlcc
-TOUCH		= touch
-WARLOCK 	= warlock
-
-#
-#	Warlock targets
-#
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
-
-warlock:	$(WARLOCK_OK)
-
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files \
-	warlock_audiosup.files warlock_amsrc2.files \
-	warlock_standalone $(WLCMD_DIR)/mixer_with_audiosup.wlcmd 
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer_with_audiosup.wlcmd $(WARLOCK_OUT) \
-	$(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer_with_audiosup.wlcmd $(WARLOCK_OUT) \
-	$(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
-	-l ../warlock/ddi_dki_impl.ll
-	$(TOUCH) $(WARLOCK_OK)
-
-%.ll: $(UTSBASE)/common/io/audio/sada/mixer/%.c
-	$(WLCC) $(CPPFLAGS) -DDEBUG -o $@ $<
-
-warlock_standalone:  $(WARLOCK_OUT) warlock_ddi.files $(WLCMD_DIR)/mixer.wlcmd
-	$(WARLOCK) -c $(WLCMD_DIR)/mixer.wlcmd $(WARLOCK_OUT) \
-	-l ../warlock/ddi_dki_impl.ll
-
-warlock_ddi.files:
-	@cd ../warlock; pwd; $(MAKE) warlock
-
-warlock_audiosup.files:
-	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/os/dacf.conf b/usr/src/uts/sparc/os/dacf.conf
index de9e672a68..199afd894f 100644
--- a/usr/src/uts/sparc/os/dacf.conf
+++ b/usr/src/uts/sparc/os/dacf.conf
@@ -19,14 +19,13 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 #
 # NOTICE: This file contains important KERNEL STATE.  Do not edit this file.
 # Its format and contents are subject to change in future releases of Solaris.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
 
 #
 # keyboard and mouse configuration rules
@@ -60,13 +59,6 @@ driver-minorname="su_pnp:mouse" consconfig_dacf:ms_config post-attach - pushmod=
 driver-minorname="ttymux:con" ttymux_dacf:ttymux_config post-attach -
 
 #
-# Configure and/or unconfigure usb audio devices.
-# usb_ac as a mux for hid and usb_as devices.
-#
-driver-minorname="usb_ac:mux" usb_ac_dacf:usb_audio_config post-attach -
-driver-minorname="usb_ac:mux" usb_ac_dacf:usb_audio_config pre-detach -
-
-#
 # Devices directly supporting the keyboard API need no device-specific module,
 # but do need to be linked to the console stream.
 #
diff --git a/usr/src/uts/sparc/usb_ac/Makefile b/usr/src/uts/sparc/usb_ac/Makefile
index 7e5ccf3df5..bf3840846c 100644
--- a/usr/src/uts/sparc/usb_ac/Makefile
+++ b/usr/src/uts/sparc/usb_ac/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/usb_ac/Makefile
@@ -40,7 +40,7 @@ OBJECTS		= $(USB_AC_OBJS:%=$(OBJS_DIR)/%)
 LINTS		= $(USB_AC_OBJS:%.o=$(LINTS_DIR)/%.ln)
 ROOTMODULE	= $(ROOT_DRV_DIR)/$(MODULE)
 CONF_SRCDIR	= $(UTSBASE)/common/io/usb/clients/audio/usb_ac
-WARLOCK_OUT	= $(USB_AC_OBJS:%.o=%.ll) usb_ac_dacf.ll
+WARLOCK_OUT	= $(USB_AC_OBJS:%.o=%.ll)
 WARLOCK_OK	= $(MODULE).ok
 WLCMD_DIR       = $(UTSBASE)/common/io/warlock
 
@@ -55,7 +55,7 @@ include $(UTSBASE)/sparc/Makefile.sparc
 CFLAGS += $(CCVERBOSE)
 
 #
-LDFLAGS	+= -dy -Nmisc/usba -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS	+= -dy -Nmisc/usba -Ndrv/audio
 
 #
 #	Define targets
@@ -111,9 +111,6 @@ USBA_FILES	= $(USBA_OBJS:%.o=../usba/%.ll)
 UHCI_FILES	= $(UHCI_OBJS:%.o=../uhci/%.ll)
 OHCI_FILES	= $(OHCI_OBJS:%.o=../ohci/%.ll)
 EHCI_FILES	= $(EHCI_OBJS:%.o=../ehci/%.ll)
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
 
 
 #
@@ -130,7 +127,6 @@ warlock_with_sada: $(WLCMD_DIR)/usb_ac_with_sada.wlcmd \
 	$(WARLOCK_OUT) \
 	warlock_audiosup.files warlock_amsrc2.files warlock_mixer.files
 	$(WARLOCK) -c $(WLCMD_DIR)/usb_ac_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
 	-l ../warlock/ddi_dki_impl.ll
 
 warlock_with_usba: $(WLCMD_DIR)/usb_ac_with_usba.wlcmd  $(WARLOCK_OUT) \
diff --git a/usr/src/uts/sparc/usb_ac_dacf/Makefile b/usr/src/uts/sparc/usb_ac_dacf/Makefile
deleted file mode 100644
index 16f567e113..0000000000
--- a/usr/src/uts/sparc/usb_ac_dacf/Makefile
+++ /dev/null
@@ -1,89 +0,0 @@
-#
-# CDDL HEADER START
-#
-# The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
-#
-# You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
-# or http://www.opensolaris.org/os/licensing.
-# See the License for the specific language governing permissions
-# and limitations under the License.
-#
-# When distributing Covered Code, include this CDDL HEADER in each
-# file and include the License file at usr/src/OPENSOLARIS.LICENSE.
-# If applicable, add the following below this CDDL HEADER, with the
-# fields enclosed by brackets "[]" replaced with your own identifying
-# information: Portions Copyright [yyyy] [name of copyright owner]
-#
-# CDDL HEADER END
-#
-#
-# uts/sparc/usb_ac_dacf/Makefile
-#
-# Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
-# Use is subject to license terms.
-#
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
-#	This makefile drives the production of the usb_ac_dacf module.
-#
-#
-#	Path to the base of the uts directory tree (usually /usr/src/uts).
-#
-UTSBASE	= ../..
-
-#
-#	Define the module and object file sets.
-#
-MODULE		= usb_ac_dacf
-OBJECTS		= $(USB_AC_DACF_OBJS:%=$(OBJS_DIR)/%)
-LINTS		= $(USB_AC_DACF_OBJS:%.o=$(LINTS_DIR)/%.ln)
-ROOTMODULE	= $(ROOT_DACF_DIR)/$(MODULE)
-
-#
-#	Include common rules.
-#
-include $(UTSBASE)/sparc/Makefile.sparc
-
-# 
-# lint pass one enforcement 
-# 
-CFLAGS += $(CCVERBOSE)
-
-# 
-# Dependencies
-#
-LDFLAGS += -dy -Nmisc/usba -Nmisc/amsrc2 -Nmisc/mixer -Nmisc/audiosup
-
-#
-#       Define targets
-#
-ALL_TARGET      = $(BINARY)
-LINT_TARGET     = $(MODULE).lint
-INSTALL_TARGET  = $(BINARY) $(ROOTMODULE)
-
-
-.KEEP_STATE:
-
-def:            $(DEF_DEPS)
-
-all:            $(ALL_DEPS)
-
-clean:          $(CLEAN_DEPS)
-
-clobber:        $(CLOBBER_DEPS)
-
-lint:           $(LINT_DEPS)
-
-modlintlib:     $(MODLINTLIB_DEPS)
-
-clean.lint:     $(CLEAN_LINT_DEPS)
-
-install:        $(INSTALL_DEPS)
-
-#
-#       Include common targets.
-#
-include $(UTSBASE)/sparc/Makefile.targ
diff --git a/usr/src/uts/sparc/usb_ah/Makefile b/usr/src/uts/sparc/usb_ah/Makefile
index cfc1d5b7fe..bd65518c1f 100644
--- a/usr/src/uts/sparc/usb_ah/Makefile
+++ b/usr/src/uts/sparc/usb_ah/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/usb_ah/Makefile
@@ -56,9 +56,9 @@ include $(UTSBASE)/sparc/Makefile.sparc
 CFLAGS += $(CCVERBOSE)
 
 #
-# depends on usba, hidparser and mixer related misc modules
+# depends on usba, hidparser and audio related misc modules
 #
-LDFLAGS	+= -dy -Nmisc/usba -Nmisc/hidparser -Nmisc/audiosup -Nmisc/mixer
+LDFLAGS	+= -dy -Nmisc/usba -Nmisc/hidparser -Ndrv/usb_ac
 
 #
 #	Define targets
diff --git a/usr/src/uts/sparc/usb_as/Makefile b/usr/src/uts/sparc/usb_as/Makefile
index e258576618..bc7db4c381 100644
--- a/usr/src/uts/sparc/usb_as/Makefile
+++ b/usr/src/uts/sparc/usb_as/Makefile
@@ -19,7 +19,7 @@
 # CDDL HEADER END
 #
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 # uts/sparc/usb_as/Makefile
@@ -56,9 +56,7 @@ include $(UTSBASE)/sparc/Makefile.sparc
 CFLAGS += $(CCVERBOSE)
 
 #
-# do not remove depends on amsrc2. If removed, amsrc2 gets unloaded when
-# usb_ac_dacf gets unloaded
-LDFLAGS		+= -dy -Nmisc/usba -Nmisc/audiosup -Nmisc/mixer -Nmisc/amsrc2
+LDFLAGS		+= -dy -Nmisc/usba -Ndrv/usb_ac
 
 #
 #	Define targets
@@ -114,9 +112,6 @@ USBA_FILES	= $(USBA_OBJS:%.o=../usba/%.ll)
 UHCI_FILES	= $(UHCI_OBJS:%.o=../uhci/%.ll)
 OHCI_FILES	= $(OHCI_OBJS:%.o=../ohci/%.ll)
 EHCI_FILES	= $(EHCI_OBJS:%.o=../ehci/%.ll)
-MIXER_FILES = $(MIXER_OBJS:%.o=$(UTSBASE)/sparc/mixer/%.ll)
-AUDIO_SUP_FILES = $(AUDIO_SUP_OBJS:%.o=$(UTSBASE)/sparc/audiosup/%.ll)
-AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
 
 
 #
@@ -124,16 +119,13 @@ AMSRC2_FILES = $(AMSRC2_OBJS:%.o=$(UTSBASE)/sparc/amsrc2/%.ll)
 #
 warlock: $(WARLOCK_OK)
 
-$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files \
-	warlock_audiosup.files warlock_mixer.files warlock_amsrc2.files \
+$(WARLOCK_OK):	$(WARLOCK_OUT) warlock_ddi.files warlock_audiosup.files \
 	warlock_standalone warlock_with_usba  warlock_with_sada
 	$(TOUCH) $(WARLOCK_OK)
 
 warlock_with_sada: $(WLCMD_DIR)/usb_as_with_sada.wlcmd \
-	$(WARLOCK_OUT) \
-	warlock_audiosup.files warlock_amsrc2.files warlock_mixer.files
+	$(WARLOCK_OUT) warlock_audiosup.files 
 	$(WARLOCK) -c $(WLCMD_DIR)/usb_as_with_sada.wlcmd $(WARLOCK_OUT) \
-	$(MIXER_FILES) $(AUDIO_SUP_FILES) $(AMSRC2_FILES) \
 	-l ../warlock/ddi_dki_impl.ll
 
 warlock_with_usba: $(WLCMD_DIR)/usb_as_with_usba.wlcmd $(WARLOCK_OUT) \
@@ -168,9 +160,3 @@ warlock_ddi.files:
 
 warlock_audiosup.files:
 	@cd ../audiosup; pwd; $(MAKE) warlock
-
-warlock_amsrc2.files:
-	@cd ../amsrc2; pwd; $(MAKE) warlock
-
-warlock_mixer.files:
-	@cd ../mixer; pwd; $(MAKE) warlock
diff --git a/usr/src/uts/sparc/warlock/Makefile b/usr/src/uts/sparc/warlock/Makefile
index 746c15d646..4cbe0e4e15 100644
--- a/usr/src/uts/sparc/warlock/Makefile
+++ b/usr/src/uts/sparc/warlock/Makefile
@@ -18,7 +18,7 @@
 #
 # CDDL HEADER END
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
 #	sparc architecture dependent
@@ -51,7 +51,7 @@ include $(UTSBASE)/sparc/Makefile.sparc
 #
 #	lock_lint rules
 #
-all:	warlock warlock.1394 warlock.audio warlock.ecpp warlock.scsi \
+all:	warlock warlock.1394 warlock.ecpp warlock.scsi \
 	warlock.smartcard warlock.usb warlock.ib warlock.sata warlock.wc
 
 warlock: $(MODULE).ok 
@@ -109,24 +109,6 @@ warlock.1394:
 warlock.ecpp:
 	@cd ../ecpp; $(MAKE) clean; $(MAKE) warlock
 
-warlock.audio:
-	@cd ../audiosup; $(MAKE) clean; $(MAKE) warlock
-	@cd ../mixer; $(MAKE) clean; $(MAKE) warlock
-	@cd ../amsrc2; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audio1575; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audio810; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audiohd; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audioixp; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audiocs; $(MAKE) clean; $(MAKE) warlock
-	@cd ../audiots; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_ac; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_as; $(MAKE) clean; $(MAKE) warlock
-	@cd ../usb_ah; $(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/sparc/amsrc1; $(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/sparc/audiovia823x; \
-		$(MAKE) clean; $(MAKE) warlock
-$(CLOSED_BUILD)	@cd $(CLOSED)/uts/sparc/audioens; $(MAKE) clean; $(MAKE) warlock
-
 warlock.smartcard:
 $(CLOSED_BUILD)	@cd $(CLOSED)/uts/sun4u/scmi2c; $(MAKE) clean; $(MAKE) warlock
 
diff --git a/usr/src/uts/sun/Makefile.files b/usr/src/uts/sun/Makefile.files
index b8da4818f3..9821c7e90b 100644
--- a/usr/src/uts/sun/Makefile.files
+++ b/usr/src/uts/sun/Makefile.files
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # uts/sun/Makefile.files
 #
 # This Makefile defines file modules and build rules for the directory
@@ -48,7 +46,6 @@ GENUNIX_OBJS +=								\
 #			Driver Modules
 #
 AUDIOCS_OBJS +=	audio_4231.o audio_4231_apcdma.o audio_4231_eb2dma.o
-AUDIO1575_OBJS += audio1575.o
 
 BPP_OBJS +=	bpp.o
 
diff --git a/usr/src/uts/sun/Makefile.rules b/usr/src/uts/sun/Makefile.rules
index cd583af7ac..22d0d3ec10 100644
--- a/usr/src/uts/sun/Makefile.rules
+++ b/usr/src/uts/sun/Makefile.rules
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-# ident	"%Z%%M%	%I%	%E% SMI"
-#
 # uts/sun/Makefile.rules
 #
 #	This Makefile defines all build rules for the directory uts/sun and
@@ -48,11 +46,7 @@ $(OBJS_DIR)/%.o:		$(UTSBASE)/sun/io/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
-$(OBJS_DIR)/%.o:		$(UTSBASE)/sun/io/audio/sada/drv/audiocs/%.c
-	$(COMPILE.c) -o $@ $<
-	$(CTFCONVERT_O)
-
-$(OBJS_DIR)/%.o:		$(UTSBASE)/sun/io/audio/sada/drv/audio1575/%.c
+$(OBJS_DIR)/%.o:		$(UTSBASE)/sun/io/audio/drv/audiocs/%.c
 	$(COMPILE.c) -o $@ $<
 	$(CTFCONVERT_O)
 
@@ -99,10 +93,7 @@ $(OBJS_DIR)/%.o:		$(UTSBASE)/common/io/scsi/adapters/%.c
 $(LINTS_DIR)/%.ln:		$(UTSBASE)/sun/io/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/sun/io/audio/sada/drv/audiocs/%.c
-	@($(LHEAD) $(LINT.c) $< $(LTAIL))
-
-$(LINTS_DIR)/%.ln:		$(UTSBASE)/sun/io/audio/sada/drv/audio1575/%.c
+$(LINTS_DIR)/%.ln:		$(UTSBASE)/sun/io/audio/drv/audiocs/%.c
 	@($(LHEAD) $(LINT.c) $< $(LTAIL))
 
 $(LINTS_DIR)/%.ln:		$(UTSBASE)/sun/io/eri/%.c
diff --git a/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231.c b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231.c
new file mode 100644
index 0000000000..5c1b6db44b
--- /dev/null
+++ b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231.c
@@ -0,0 +1,2406 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+
+/*
+ * audiocs Audio Driver
+ *
+ * This Audio Driver controls the Crystal CS4231 Codec used on many SPARC
+ * platforms. It does not support the CS4231 on Power PCs or x86 PCs. It
+ * does support two different DMA engines, the APC and EB2. The code for
+ * those DMA engines is split out and a well defined, but private, interface
+ * is used to control those DMA engines.
+ *
+ * For some reason setting the CS4231's registers doesn't always
+ * succeed.  Therefore every time we set a register we always read it
+ * back to make sure it was set. If not we wait a little while and
+ * then try again. This is all taken care of in the routines
+ * audiocs_put_index() and audiocs_sel_index() and the macros ORIDX()
+ * and ANDIDX(). We don't worry about the status register because it
+ * is cleared by writing anything to it.  So it doesn't matter what
+ * the value written is.
+ *
+ * This driver supports suspending and resuming. A suspend just stops playing
+ * and recording. The play DMA buffers end up getting thrown away, but when
+ * you shut down the machine there is a break in the audio anyway, so they
+ * won't be missed and it isn't worth the effort to save them. When we resume
+ * we always start playing and recording. If they aren't needed they get
+ * shut off by the mixer.
+ *
+ * Power management is supported by this driver.
+ *
+ *	NOTE: This module depends on drv/audio being loaded first.
+ */
+
+#include <sys/modctl.h>
+#include <sys/kmem.h>
+#include <sys/stropts.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include "audio_4231.h"
+
+/*
+ * Module linkage routines for the kernel
+ */
+static int audiocs_ddi_attach(dev_info_t *, ddi_attach_cmd_t);
+static int audiocs_ddi_detach(dev_info_t *, ddi_detach_cmd_t);
+static int audiocs_ddi_power(dev_info_t *, int, int);
+
+/*
+ * Entry point routine prototypes
+ */
+static int audiocs_open(void *, int, unsigned *, unsigned *, caddr_t *);
+static void audiocs_close(void *);
+static int audiocs_start(void *);
+static void audiocs_stop(void *);
+static int audiocs_format(void *);
+static int audiocs_channels(void *);
+static int audiocs_rate(void *);
+static uint64_t audiocs_count(void *);
+static void audiocs_sync(void *, unsigned);
+static size_t audiocs_qlen(void *);
+
+/*
+ * Control callbacks.
+ */
+static int audiocs_get_value(void *, uint64_t *);
+static int audiocs_set_ogain(void *, uint64_t);
+static int audiocs_set_igain(void *, uint64_t);
+static int audiocs_set_mgain(void *, uint64_t);
+static int audiocs_set_inputs(void *, uint64_t);
+static int audiocs_set_outputs(void *, uint64_t);
+static int audiocs_set_micboost(void *, uint64_t);
+
+/* Local Routines */
+static int audiocs_resume(dev_info_t *);
+static int audiocs_attach(dev_info_t *);
+static int audiocs_detach(dev_info_t *);
+static int audiocs_suspend(dev_info_t *);
+
+static void audiocs_destroy(CS_state_t *);
+static int audiocs_init_state(CS_state_t *);
+static int audiocs_chip_init(CS_state_t *);
+static int audiocs_alloc_engine(CS_state_t *, int);
+static void audiocs_free_engine(CS_engine_t *);
+static void audiocs_reset_engine(CS_engine_t *);
+static int audiocs_start_engine(CS_engine_t *);
+static void audiocs_stop_engine(CS_engine_t *);
+static void audiocs_get_ports(CS_state_t *);
+static void audiocs_configure_input(CS_state_t *);
+static void audiocs_configure_output(CS_state_t *);
+static CS_ctrl_t *audiocs_alloc_ctrl(CS_state_t *, uint32_t, uint64_t);
+static void audiocs_free_ctrl(CS_ctrl_t *);
+static int audiocs_add_controls(CS_state_t *);
+static void audiocs_del_controls(CS_state_t *);
+static void audiocs_power_up(CS_state_t *);
+static void audiocs_power_down(CS_state_t *);
+static int audiocs_poll_ready(CS_state_t *);
+#ifdef	DEBUG
+static void audiocs_put_index(CS_state_t *,  uint8_t, uint8_t, int);
+static void audiocs_sel_index(CS_state_t *, uint8_t, int);
+#define	SELIDX(s, idx)		audiocs_sel_index(s, idx, __LINE__)
+#define	PUTIDX(s, val, mask)	audiocs_put_index(s, val, mask, __LINE__)
+#else
+static void audiocs_put_index(CS_state_t *,  uint8_t, uint8_t);
+static void audiocs_sel_index(CS_state_t *, uint8_t);
+#define	SELIDX(s, idx)		audiocs_sel_index(s, idx)
+#define	PUTIDX(s, val, mask)	audiocs_put_index(s, val, mask)
+#endif
+
+#define	ORIDX(s, val, mask)						\
+	PUTIDX(s,							\
+	    (ddi_get8((handle), &CS4231_IDR) | (uint8_t)(val)),		\
+	    (uint8_t)(mask))
+
+#define	ANDIDX(s, val, mask)						\
+	PUTIDX(s, (ddi_get8((handle), &CS4231_IDR) & (uint8_t)(val)),	\
+	    (uint8_t)(mask))
+
+static audio_engine_ops_t audiocs_engine_ops = {
+	AUDIO_ENGINE_VERSION,
+	audiocs_open,
+	audiocs_close,
+	audiocs_start,
+	audiocs_stop,
+	audiocs_count,
+	audiocs_format,
+	audiocs_channels,
+	audiocs_rate,
+	audiocs_sync,
+	audiocs_qlen
+};
+
+#define	OUTPUT_SPEAKER		0
+#define	OUTPUT_HEADPHONES	1
+#define	OUTPUT_LINEOUT		2
+
+static const char *audiocs_outputs[] = {
+	AUDIO_PORT_SPEAKER,
+	AUDIO_PORT_HEADPHONES,
+	AUDIO_PORT_LINEOUT,
+	NULL
+};
+
+#define	INPUT_MIC		0
+#define	INPUT_LINEIN		1
+#define	INPUT_STEREOMIX		2
+#define	INPUT_CD		3
+
+static const char *audiocs_inputs[] = {
+	AUDIO_PORT_MIC,
+	AUDIO_PORT_LINEIN,
+	AUDIO_PORT_STEREOMIX,
+	AUDIO_PORT_CD,
+	NULL
+};
+
+/*
+ * Global variables, but viewable only by this file.
+ */
+
+/* play gain array, converts linear gain to 64 steps of log10 gain */
+static uint8_t cs4231_atten[] = {
+	0x3f,	0x3e,	0x3d,	0x3c,	0x3b,	/* [000] -> [004] */
+	0x3a,	0x39,	0x38,	0x37,	0x36,	/* [005] -> [009] */
+	0x35,	0x34,	0x33,	0x32,	0x31,	/* [010] -> [014] */
+	0x30,	0x2f,	0x2e,	0x2d,	0x2c,	/* [015] -> [019] */
+	0x2b,	0x2a,	0x29,	0x29,	0x28,	/* [020] -> [024] */
+	0x28,	0x27,	0x27,	0x26,	0x26,	/* [025] -> [029] */
+	0x25,	0x25,	0x24,	0x24,	0x23,	/* [030] -> [034] */
+	0x23,	0x22,	0x22,	0x21,	0x21,	/* [035] -> [039] */
+	0x20,	0x20,	0x1f,	0x1f,	0x1f,	/* [040] -> [044] */
+	0x1e,	0x1e,	0x1e,	0x1d,	0x1d,	/* [045] -> [049] */
+	0x1d,	0x1c,	0x1c,	0x1c,	0x1b,	/* [050] -> [054] */
+	0x1b,	0x1b,	0x1a,	0x1a,	0x1a,	/* [055] -> [059] */
+	0x1a,	0x19,	0x19,	0x19,	0x19,	/* [060] -> [064] */
+	0x18,	0x18,	0x18,	0x18,	0x17,	/* [065] -> [069] */
+	0x17,	0x17,	0x17,	0x16,	0x16,	/* [070] -> [074] */
+	0x16,	0x16,	0x16,	0x15,	0x15,	/* [075] -> [079] */
+	0x15,	0x15,	0x15,	0x14,	0x14,	/* [080] -> [084] */
+	0x14,	0x14,	0x14,	0x13,	0x13,	/* [085] -> [089] */
+	0x13,	0x13,	0x13,	0x12,	0x12,	/* [090] -> [094] */
+	0x12,	0x12,	0x12,	0x12,	0x11,	/* [095] -> [099] */
+	0x11,	0x11,	0x11,	0x11,	0x11,	/* [100] -> [104] */
+	0x10,	0x10,	0x10,	0x10,	0x10,	/* [105] -> [109] */
+	0x10,	0x0f,	0x0f,	0x0f,	0x0f,	/* [110] -> [114] */
+	0x0f,	0x0f,	0x0e,	0x0e,	0x0e,	/* [114] -> [119] */
+	0x0e,	0x0e,	0x0e,	0x0e,	0x0d,	/* [120] -> [124] */
+	0x0d,	0x0d,	0x0d,	0x0d,	0x0d,	/* [125] -> [129] */
+	0x0d,	0x0c,	0x0c,	0x0c,	0x0c,	/* [130] -> [134] */
+	0x0c,	0x0c,	0x0c,	0x0b,	0x0b,	/* [135] -> [139] */
+	0x0b,	0x0b,	0x0b,	0x0b,	0x0b,	/* [140] -> [144] */
+	0x0b,	0x0a,	0x0a,	0x0a,	0x0a,	/* [145] -> [149] */
+	0x0a,	0x0a,	0x0a,	0x0a,	0x09,	/* [150] -> [154] */
+	0x09,	0x09,	0x09,	0x09,	0x09,	/* [155] -> [159] */
+	0x09,	0x09,	0x08,	0x08,	0x08,	/* [160] -> [164] */
+	0x08,	0x08,	0x08,	0x08,	0x08,	/* [165] -> [169] */
+	0x08,	0x07,	0x07,	0x07,	0x07,	/* [170] -> [174] */
+	0x07,	0x07,	0x07,	0x07,	0x07,	/* [175] -> [179] */
+	0x06,	0x06,	0x06,	0x06,	0x06,	/* [180] -> [184] */
+	0x06,	0x06,	0x06,	0x06,	0x05,	/* [185] -> [189] */
+	0x05,	0x05,	0x05,	0x05,	0x05,	/* [190] -> [194] */
+	0x05,	0x05,	0x05,	0x05,	0x04,	/* [195] -> [199] */
+	0x04,	0x04,	0x04,	0x04,	0x04,	/* [200] -> [204] */
+	0x04,	0x04,	0x04,	0x04,	0x03,	/* [205] -> [209] */
+	0x03,	0x03,	0x03,	0x03,	0x03,	/* [210] -> [214] */
+	0x03,	0x03,	0x03,	0x03,	0x03,	/* [215] -> [219] */
+	0x02,	0x02,	0x02,	0x02,	0x02,	/* [220] -> [224] */
+	0x02,	0x02,	0x02,	0x02,	0x02,	/* [225] -> [229] */
+	0x02,	0x01,	0x01,	0x01,	0x01,	/* [230] -> [234] */
+	0x01,	0x01,	0x01,	0x01,	0x01,	/* [235] -> [239] */
+	0x01,	0x01,	0x01,	0x00,	0x00,	/* [240] -> [244] */
+	0x00,	0x00,	0x00,	0x00,	0x00,	/* [245] -> [249] */
+	0x00,	0x00,	0x00,	0x00,	0x00,	/* [250] -> [254] */
+	0x00					/* [255] */
+};
+
+/*
+ * STREAMS Structures
+ */
+
+/*
+ * DDI Structures
+ */
+
+/* Device operations structure */
+static struct dev_ops audiocs_dev_ops = {
+	DEVO_REV,			/* devo_rev */
+	0,				/* devo_refcnt */
+	NULL,				/* devo_getinfo */
+	nulldev,			/* devo_identify - obsolete */
+	nulldev,			/* devo_probe - not needed */
+	audiocs_ddi_attach,		/* devo_attach */
+	audiocs_ddi_detach,		/* devo_detach */
+	nodev,				/* devo_reset */
+	NULL,				/* devi_cb_ops */
+	NULL,				/* devo_bus_ops */
+	audiocs_ddi_power,		/* devo_power */
+	ddi_quiesce_not_supported,	/* devo_quiesce */
+};
+
+/* Linkage structure for loadable drivers */
+static struct modldrv audiocs_modldrv = {
+	&mod_driverops,		/* drv_modops */
+	CS4231_MOD_NAME,	/* drv_linkinfo */
+	&audiocs_dev_ops	/* drv_dev_ops */
+};
+
+/* Module linkage structure */
+static struct modlinkage audiocs_modlinkage = {
+	MODREV_1,			/* ml_rev */
+	(void *)&audiocs_modldrv,	/* ml_linkage */
+	NULL				/* NULL terminates the list */
+};
+
+
+/* *******  Loadable Module Configuration Entry Points  ********************* */
+
+/*
+ * _init()
+ *
+ * Description:
+ *	Implements _init(9E).
+ *
+ * Returns:
+ *	mod_install() status, see mod_install(9f)
+ */
+int
+_init(void)
+{
+	int	rv;
+
+	audio_init_ops(&audiocs_dev_ops, CS4231_NAME);
+
+	if ((rv = mod_install(&audiocs_modlinkage)) != 0) {
+		audio_fini_ops(&audiocs_dev_ops);
+	}
+
+	return (rv);
+}
+
+/*
+ * _fini()
+ *
+ * Description:
+ *	Implements _fini(9E).
+ *
+ * Returns:
+ *	mod_remove() status, see mod_remove(9f)
+ */
+int
+_fini(void)
+{
+	int	rv;
+
+	if ((rv = mod_remove(&audiocs_modlinkage)) == 0) {
+		audio_fini_ops(&audiocs_dev_ops);
+	}
+
+	return (rv);
+}
+
+/*
+ * _info()
+ *
+ * Description:
+ *	Implements _info(9E).
+ *
+ * Arguments:
+ *	modinfo *modinfop	Pointer to the opaque modinfo structure
+ *
+ * Returns:
+ *	mod_info() status, see mod_info(9f)
+ */
+int
+_info(struct modinfo *modinfop)
+{
+	return (mod_info(&audiocs_modlinkage, modinfop));
+}
+
+
+/* *******  Driver Entry Points  ******************************************** */
+
+/*
+ * audiocs_ddi_attach()
+ *
+ * Description:
+ *	Implement attach(9e).
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_attach_cmd_t cmd	Attach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audiocs_ddi_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_ATTACH:
+		return (audiocs_attach(dip));
+
+	case DDI_RESUME:
+		return (audiocs_resume(dip));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+/*
+ * audiocs_ddi_detach()
+ *
+ * Description:
+ *	Implement detach(9e).
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *	ddi_detach_cmd_t cmd	Detach command
+ *
+ * Returns:
+ *	DDI_SUCCESS		Success.
+ *	DDI_FAILURE		Failure.
+ */
+static int
+audiocs_ddi_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
+{
+	switch (cmd) {
+	case DDI_DETACH:
+		return (audiocs_detach(dip));
+
+	case DDI_SUSPEND:
+		return (audiocs_suspend(dip));
+
+	default:
+		return (DDI_FAILURE);
+	}
+}
+
+/*
+ * audiocs_ddi_power()
+ *
+ * Description:
+ *	Implements power(9E).
+ *
+ * Arguments:
+ *	def_info_t	*dip		Ptr to the device's dev_info structure
+ *	int		component	Which component to power up/down
+ *	int		level		The power level for the component
+ *
+ * Returns:
+ *	DDI_SUCCESS		Power level changed, we always succeed
+ */
+static int
+audiocs_ddi_power(dev_info_t *dip, int component, int level)
+{
+	CS_state_t		*state;
+
+	if (component != CS4231_COMPONENT)
+		return (DDI_FAILURE);
+
+	/* get the state structure */
+	state = ddi_get_driver_private(dip);
+
+	ASSERT(!mutex_owned(&state->cs_lock));
+
+	/* make sure we have some work to do */
+	mutex_enter(&state->cs_lock);
+
+	/*
+	 * We don't do anything if we're suspended.  Suspend/resume diddles
+	 * with power anyway.
+	 */
+	if (!state->cs_suspended) {
+
+		/* check the level change to see what we need to do */
+		if (level == CS4231_PWR_OFF && state->cs_powered) {
+
+			/* power down and save the state */
+			audiocs_power_down(state);
+			state->cs_powered = B_FALSE;
+
+		} else if (level == CS4231_PWR_ON && !state->cs_powered) {
+
+			/* power up */
+			audiocs_power_up(state);
+			state->cs_powered = B_TRUE;
+		}
+	}
+
+	mutex_exit(&state->cs_lock);
+
+	ASSERT(!mutex_owned(&state->cs_lock));
+
+	return (DDI_SUCCESS);
+}
+
+/* ******* Local Routines *************************************************** */
+
+static void
+audiocs_destroy(CS_state_t *state)
+{
+	if (state == NULL)
+		return;
+
+	/*
+	 * Unregister any interrupts. That way we can't get called by and
+	 * interrupt after the audio framework is removed.
+	 */
+	CS4231_DMA_REM_INTR(state);
+
+	for (int i = CS4231_PLAY; i <= CS4231_REC; i++) {
+		audiocs_free_engine(state->cs_engines[i]);
+	}
+	audiocs_del_controls(state);
+
+	/* free the kernel statistics structure */
+	if (state->cs_ksp) {
+		kstat_delete(state->cs_ksp);
+	}
+
+	if (state->cs_adev) {
+		audio_dev_free(state->cs_adev);
+	}
+
+	/* unmap the registers */
+	CS4231_DMA_UNMAP_REGS(state);
+
+	/* destroy the state mutex */
+	mutex_destroy(&state->cs_lock);
+	kmem_free(state, sizeof (*state));
+}
+
+/*
+ * audiocs_attach()
+ *
+ * Description:
+ *	Attach an instance of the CS4231 driver. This routine does the device
+ *	dependent attach tasks.  When it is complete it calls
+ *	audio_dev_register() to register with the framework.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audiocs_attach(dev_info_t *dip)
+{
+	CS_state_t		*state;
+	audio_dev_t		*adev;
+
+	/* allocate the state structure */
+	state = kmem_zalloc(sizeof (*state), KM_SLEEP);
+	state->cs_dip = dip;
+	ddi_set_driver_private(dip, state);
+
+	/* get the iblock cookie needed for interrupt context */
+	if (ddi_get_iblock_cookie(dip, 0, &state->cs_iblock) != DDI_SUCCESS) {
+		audio_dev_warn(NULL, "cannot get iblock cookie");
+		kmem_free(state, sizeof (*state));
+		return (DDI_FAILURE);
+	}
+
+	/* now fill it in, initialize the state mutexs first */
+	mutex_init(&state->cs_lock, NULL, MUTEX_DRIVER, state->cs_iblock);
+
+	/*
+	 * audio state initialization... should always succeed,
+	 * framework will message failure.
+	 */
+	if ((state->cs_adev = audio_dev_alloc(dip, 0)) == NULL) {
+		goto error;
+	}
+	adev = state->cs_adev;
+	audio_dev_set_description(adev, CS_DEV_CONFIG_ONBRD1);
+	audio_dev_add_info(adev, "Legacy codec: Crystal Semiconductor CS4231");
+
+	/* initialize the audio state structures */
+	if ((audiocs_init_state(state)) == DDI_FAILURE) {
+		audio_dev_warn(adev, "init_state() failed");
+		goto error;
+	}
+
+	/* initialize the audio chip */
+	if ((audiocs_chip_init(state)) == DDI_FAILURE) {
+		audio_dev_warn(adev, "chip_init() failed");
+		goto error;
+	}
+	/* chip init will have powered us up */
+	state->cs_powered = B_TRUE;
+
+	/* set up kernel statistics */
+	if ((state->cs_ksp = kstat_create(ddi_driver_name(dip),
+	    ddi_get_instance(dip), ddi_driver_name(dip),
+	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT)) != NULL) {
+		kstat_install(state->cs_ksp);
+	}
+
+	/* we're ready, set up the interrupt handler */
+	if (CS4231_DMA_ADD_INTR(state) != DDI_SUCCESS) {
+		goto error;
+	}
+
+	/* finally register with framework to kick everything off */
+	if (audio_dev_register(state->cs_adev) != DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "unable to register audio dev");
+	}
+
+	/* everything worked out, so report the device */
+	ddi_report_dev(dip);
+
+	return (DDI_SUCCESS);
+
+error:
+	audiocs_destroy(state);
+	return (DDI_FAILURE);
+}
+
+/*
+ * audiocs_resume()
+ *
+ * Description:
+ *	Resume a suspended device instance.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was initialized properly
+ *	DDI_FAILURE		The driver couldn't be initialized properly
+ */
+static int
+audiocs_resume(dev_info_t *dip)
+{
+	CS_state_t		*state;
+	audio_dev_t		*adev;
+
+	/* we've already allocated the state structure so get ptr */
+	state = ddi_get_driver_private(dip);
+	adev = state->cs_adev;
+
+	ASSERT(dip == state->cs_dip);
+	ASSERT(!mutex_owned(&state->cs_lock));
+
+	/* mark the Codec busy -- this should keep power(9e) away */
+	(void) pm_busy_component(state->cs_dip, CS4231_COMPONENT);
+
+	/* power it up */
+	audiocs_power_up(state);
+	state->cs_powered = B_TRUE;
+
+	/* initialize the audio chip */
+	if ((audiocs_chip_init(state)) == DDI_FAILURE) {
+		audio_dev_warn(adev, "chip_init() failed");
+		(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
+		return (DDI_FAILURE);
+	}
+
+	mutex_enter(&state->cs_lock);
+	state->cs_suspended = B_FALSE;
+
+	/* restore mixer settings */
+	audiocs_configure_output(state);
+	audiocs_configure_input(state);
+
+	for (int i = CS4231_PLAY; i <= CS4231_REC; i++) {
+		CS_engine_t	*eng = state->cs_engines[i];
+
+		audiocs_reset_engine(eng);
+		if (eng->ce_started) {
+			(void) audiocs_start_engine(eng);
+		} else {
+			audiocs_stop_engine(eng);
+		}
+	}
+	mutex_exit(&state->cs_lock);
+
+	/*
+	 * We have already powered up the chip, but this alerts the
+	 * framework to the fact.
+	 */
+	(void) pm_raise_power(dip, CS4231_COMPONENT, CS4231_PWR_ON);
+	(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_detach()
+ *
+ * Description:
+ *	Detach an instance of the CS4231 driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was detached
+ *	DDI_FAILURE		The driver couldn't be detached (busy)
+ */
+static int
+audiocs_detach(dev_info_t *dip)
+{
+	CS_state_t		*state;
+	audio_dev_t		*adev;
+	ddi_acc_handle_t	handle;
+
+	/* get the state structure */
+	state = ddi_get_driver_private(dip);
+	handle = CODEC_HANDLE;
+	adev = state->cs_adev;
+
+	/* don't detach if still in use */
+	if (audio_dev_unregister(adev) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	if (state->cs_powered) {
+		/*
+		 * Make sure the Codec and DMA engine are off.
+		 */
+		SELIDX(state, INTC_REG);
+		ANDIDX(state, ~(INTC_PEN|INTC_CEN), INTC_VALID_MASK);
+
+		/* make sure the DMA engine isn't going to do anything */
+		CS4231_DMA_RESET(state);
+
+		/*
+		 * power down the device, no reason to waste power without
+		 * a driver
+		 */
+		(void) pm_lower_power(dip, CS4231_COMPONENT, CS4231_PWR_OFF);
+	}
+
+	audiocs_destroy(state);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_suspend()
+ *
+ * Description:
+ *	Suspend an instance of the CS4231 driver.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's dev_info struct
+ *
+ * Returns:
+ *	DDI_SUCCESS		The driver was detached
+ *	DDI_FAILURE		The driver couldn't be detached
+ */
+static int
+audiocs_suspend(dev_info_t *dip)
+{
+	CS_state_t		*state;
+
+	/* get the state structure */
+	state = ddi_get_driver_private(dip);
+
+	ASSERT(!mutex_owned(&state->cs_lock));
+
+	mutex_enter(&state->cs_lock);
+
+	ASSERT(!state->cs_suspended);
+
+	if (state->cs_powered) {
+		/* stop playing and recording */
+		CS4231_DMA_STOP(state, state->cs_engines[CS4231_PLAY]);
+		CS4231_DMA_STOP(state, state->cs_engines[CS4231_REC]);
+
+		/* now we can power down the Codec */
+		audiocs_power_down(state);
+		state->cs_powered = B_FALSE;
+	}
+	state->cs_suspended = B_TRUE;	/* stop new ops */
+	mutex_exit(&state->cs_lock);
+
+	ASSERT(!mutex_owned(&state->cs_lock));
+	return (DDI_SUCCESS);
+}
+
+#define	PLAYCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_PLAY)
+#define	RECCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_REC)
+#define	MONCTL	(AUDIO_CTRL_FLAG_RW | AUDIO_CTRL_FLAG_MONITOR)
+#define	PCMVOL	(PLAYCTL | AUDIO_CTRL_FLAG_PCMVOL)
+#define	MAINVOL	(PLAYCTL | AUDIO_CTRL_FLAG_MAINVOL)
+#define	RECVOL	(RECCTL | AUDIO_CTRL_FLAG_RECVOL)
+#define	MONVOL	(MONCTL | AUDIO_CTRL_FLAG_MONVOL)
+
+/*
+ * audiocs_alloc_ctrl
+ *
+ * Description:
+ *	Allocates a control structure for the audio mixer.
+ *
+ * Arguments:
+ *	CS_state_t	*state		Device soft state.
+ *	uint32_t	num		Control number to allocate.
+ *	uint64_t	val		Initial value.
+ *
+ * Returns:
+ *	Pointer to newly allocated CS_ctrl_t structure.
+ */
+static CS_ctrl_t *
+audiocs_alloc_ctrl(CS_state_t *state, uint32_t num, uint64_t val)
+{
+	audio_ctrl_desc_t	desc;
+	audio_ctrl_wr_t		fn;
+	CS_ctrl_t		*cc;
+
+	cc = kmem_zalloc(sizeof (*cc), KM_SLEEP);
+	cc->cc_state = state;
+	cc->cc_num = num;
+
+	bzero(&desc, sizeof (desc));
+
+	switch (num) {
+	case CTL_VOLUME:
+		desc.acd_name = AUDIO_CTRL_ID_VOLUME;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = PCMVOL;
+		fn = audiocs_set_ogain;
+		break;
+
+	case CTL_IGAIN:
+		desc.acd_name = AUDIO_CTRL_ID_RECGAIN;
+		desc.acd_type = AUDIO_CTRL_TYPE_STEREO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = RECVOL;
+		fn = audiocs_set_igain;
+		break;
+
+	case CTL_MGAIN:
+		desc.acd_name = AUDIO_CTRL_ID_MONGAIN;
+		desc.acd_type = AUDIO_CTRL_TYPE_MONO;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 100;
+		desc.acd_flags = MONVOL;
+		fn = audiocs_set_mgain;
+		break;
+
+	case CTL_INPUTS:
+		desc.acd_name = AUDIO_CTRL_ID_RECSRC;
+		desc.acd_type = AUDIO_CTRL_TYPE_ENUM;
+		desc.acd_minvalue = state->cs_imask;
+		desc.acd_maxvalue = state->cs_imask;
+		desc.acd_flags = RECCTL;
+		for (int i = 0; audiocs_inputs[i]; i++) {
+			desc.acd_enum[i] = audiocs_inputs[i];
+		}
+		fn = audiocs_set_inputs;
+
+		break;
+
+	case CTL_OUTPUTS:
+		desc.acd_name = AUDIO_CTRL_ID_OUTPUTS;
+		desc.acd_type = AUDIO_CTRL_TYPE_ENUM;
+		desc.acd_minvalue = state->cs_omod;
+		desc.acd_maxvalue = state->cs_omask;
+		desc.acd_flags = PLAYCTL | AUDIO_CTRL_FLAG_MULTI;
+		for (int i = 0; audiocs_outputs[i]; i++) {
+			desc.acd_enum[i] = audiocs_outputs[i];
+		}
+		fn = audiocs_set_outputs;
+		break;
+
+	case CTL_MICBOOST:
+		desc.acd_name = AUDIO_CTRL_ID_MICBOOST;
+		desc.acd_type = AUDIO_CTRL_TYPE_BOOLEAN;
+		desc.acd_minvalue = 0;
+		desc.acd_maxvalue = 1;
+		desc.acd_flags = RECCTL;
+		fn = audiocs_set_micboost;
+		break;
+	}
+
+	cc->cc_val = val;
+	cc->cc_ctrl = audio_dev_add_control(state->cs_adev, &desc,
+	    audiocs_get_value, fn, cc);
+
+	return (cc);
+}
+
+/*
+ * audiocs_free_ctrl
+ *
+ * Description:
+ *	Frees a control and all resources associated with it.
+ *
+ * Arguments:
+ *	CS_ctrl_t	*cc	Pointer to control structure.
+ */
+static void
+audiocs_free_ctrl(CS_ctrl_t *cc)
+{
+	if (cc == NULL)
+		return;
+	if (cc->cc_ctrl)
+		audio_dev_del_control(cc->cc_ctrl);
+	kmem_free(cc, sizeof (*cc));
+}
+
+/*
+ * audiocs_add_controls
+ *
+ * Description:
+ *	Allocates and registers all controls for this device.
+ *
+ * Arguments:
+ *	CS_state_t	*state		Device soft state.
+ *
+ * Returns:
+ *	DDI_SUCCESS	All controls added and registered
+ *	DDI_FAILURE	At least one control was not added or registered.
+ */
+static int
+audiocs_add_controls(CS_state_t *state)
+{
+#define	ADD_CTRL(CTL, ID, VAL)						\
+	state->cs_##CTL = audiocs_alloc_ctrl(state, ID, VAL);		\
+	if (state->cs_##CTL == NULL) {					\
+		audio_dev_warn(state->cs_adev,				\
+		    "unable to allocate %s control", #ID);		\
+		return (DDI_FAILURE);					\
+	}
+
+	ADD_CTRL(ogain, CTL_VOLUME, 0x4b4b);
+	ADD_CTRL(igain, CTL_IGAIN, 0x3232);
+	ADD_CTRL(mgain, CTL_MGAIN, 0);
+	ADD_CTRL(micboost, CTL_MICBOOST, 0);
+	ADD_CTRL(outputs, CTL_OUTPUTS, (state->cs_omask & ~state->cs_omod) |
+	    (1U << OUTPUT_SPEAKER));
+	ADD_CTRL(inputs, CTL_INPUTS, (1U << INPUT_MIC));
+
+	mutex_enter(&state->cs_lock);
+	audiocs_configure_output(state);
+	audiocs_configure_input(state);
+	mutex_exit(&state->cs_lock);
+
+	return (DDI_SUCCESS);
+}
+/*
+ * audiocs_del_controls
+ *
+ * Description:
+ *	Unregisters and frees all controls for this device.
+ *
+ * Arguments:
+ *	CS_state_t	*state		Device soft state.
+ */
+void
+audiocs_del_controls(CS_state_t *state)
+{
+	audiocs_free_ctrl(state->cs_ogain);
+	audiocs_free_ctrl(state->cs_igain);
+	audiocs_free_ctrl(state->cs_mgain);
+	audiocs_free_ctrl(state->cs_micboost);
+	audiocs_free_ctrl(state->cs_inputs);
+	audiocs_free_ctrl(state->cs_outputs);
+}
+
+
+/*
+ * audiocs_chip_init()
+ *
+ * Description:
+ *	Power up the audio core, initialize the audio Codec, prepare the chip
+ *	for use.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS			Chip initialized and ready to use
+ *	DDI_FAILURE			Chip not initialized and not ready
+ */
+static int
+audiocs_chip_init(CS_state_t *state)
+{
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+
+	/* make sure we are powered up */
+	CS4231_DMA_POWER(state, CS4231_PWR_ON);
+
+	CS4231_DMA_RESET(state);
+
+	/* activate registers 16 -> 31 */
+	SELIDX(state, MID_REG);
+	ddi_put8(handle, &CS4231_IDR, MID_MODE2);
+
+	/* now figure out what version we have */
+	SELIDX(state, VID_REG);
+	if (ddi_get8(handle, &CS4231_IDR) & VID_A) {
+		state->cs_revA = B_TRUE;
+	} else {
+		state->cs_revA = B_FALSE;
+	}
+
+	/* get rid of annoying popping by muting the output channels */
+	SELIDX(state, LDACO_REG);
+	PUTIDX(state, LDACO_LDM | LDACO_MID_GAIN, LDAC0_VALID_MASK);
+	SELIDX(state, RDACO_REG);
+	PUTIDX(state, RDACO_RDM | RDACO_MID_GAIN, RDAC0_VALID_MASK);
+
+	/* initialize aux input channels to known gain values & muted */
+	SELIDX(state, LAUX1_REG);
+	PUTIDX(state, LAUX1_LX1M | LAUX1_UNITY_GAIN, LAUX1_VALID_MASK);
+	SELIDX(state, RAUX1_REG);
+	PUTIDX(state, RAUX1_RX1M | RAUX1_UNITY_GAIN, RAUX1_VALID_MASK);
+	SELIDX(state, LAUX2_REG);
+	PUTIDX(state, LAUX2_LX2M | LAUX2_UNITY_GAIN, LAUX2_VALID_MASK);
+	SELIDX(state, RAUX2_REG);
+	PUTIDX(state, RAUX2_RX2M | RAUX2_UNITY_GAIN, RAUX2_VALID_MASK);
+
+	/* initialize aux input channels to known gain values & muted */
+	SELIDX(state, LLIC_REG);
+	PUTIDX(state, LLIC_LLM | LLIC_UNITY_GAIN, LLIC_VALID_MASK);
+	SELIDX(state, RLIC_REG);
+	PUTIDX(state, RLIC_RLM | RLIC_UNITY_GAIN, RLIC_VALID_MASK);
+
+	/* program the sample rate, play and capture must be the same */
+	SELIDX(state, FSDF_REG | IAR_MCE);
+	PUTIDX(state, FS_48000 | PDF_LINEAR16NE | PDF_STEREO, FSDF_VALID_MASK);
+	SELIDX(state, FSDF_REG);
+	SELIDX(state, CDF_REG | IAR_MCE);
+	PUTIDX(state, CDF_LINEAR16NE | CDF_STEREO, CDF_VALID_MASK);
+	SELIDX(state, CDF_REG);
+
+	/*
+	 * Set up the Codec for playback and capture disabled, dual DMA, and
+	 * playback and capture DMA.
+	 */
+	SELIDX(state, (INTC_REG | IAR_MCE));
+	PUTIDX(state, INTC_DDC | INTC_PDMA | INTC_CDMA, INTC_VALID_MASK);
+
+	/* turn off the MCE bit */
+	SELIDX(state, LADCI_REG);
+
+	/* wait for the Codec before we continue XXX - do we need this? */
+	if (audiocs_poll_ready(state) == DDI_FAILURE) {
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * Turn on the output level bit to be 2.8 Vpp. Also, don't go to 0 on
+	 * underflow.
+	 */
+	SELIDX(state, AFE1_REG);
+	PUTIDX(state, AFE1_OLB, AFE1_VALID_MASK);
+
+	/* turn on the high pass filter if Rev A */
+	SELIDX(state, AFE2_REG);
+	if (state->cs_revA) {
+		PUTIDX(state, AFE2_HPF, AFE2_VALID_MASK);
+	} else {
+		PUTIDX(state, 0, AFE2_VALID_MASK);
+	}
+
+	/* clear the play and capture interrupt flags */
+	SELIDX(state, AFS_REG);
+	ddi_put8(handle, &CS4231_STATUS, (AFS_RESET_STATUS));
+
+	/* the play and record gains will be set by the audio mixer */
+
+	/* unmute the output */
+	SELIDX(state, LDACO_REG);
+	ANDIDX(state, ~LDACO_LDM, LDAC0_VALID_MASK);
+	SELIDX(state, RDACO_REG);
+	ANDIDX(state, ~RDACO_RDM, RDAC0_VALID_MASK);
+
+	/* unmute the mono speaker and mute mono in */
+	SELIDX(state, MIOC_REG);
+	PUTIDX(state, MIOC_MIM, MIOC_VALID_MASK);
+
+	/* clear the mode change bit */
+	SELIDX(state, RDACO_REG);
+
+	/* wait for the Codec before we continue XXX - do we need this? */
+	if (audiocs_poll_ready(state) == DDI_FAILURE) {
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_init_state()
+ *
+ * Description:
+ *	This routine initializes the audio driver's state structure and
+ *	maps in the registers. This also includes reading the properties.
+ *
+ *	CAUTION: This routine maps the registers and initializes a mutex.
+ *		 Failure cleanup is handled by cs4231_attach(). It is not
+ *		 handled locally by this routine.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS			State structure initialized
+ *	DDI_FAILURE			State structure not initialized
+ */
+static int
+audiocs_init_state(CS_state_t *state)
+{
+	audio_dev_t	*adev = state->cs_adev;
+	dev_info_t	*dip = state->cs_dip;
+	char		*prop_str;
+	char		*pm_comp[] = {
+				"NAME=audiocs audio device",
+				"0=off",
+				"1=on" };
+
+	/* set up the pm-components */
+	if (ddi_prop_update_string_array(DDI_DEV_T_NONE, dip,
+	    "pm-components", pm_comp, 3) != DDI_PROP_SUCCESS) {
+		audio_dev_warn(adev, "couldn't create pm-components property");
+		return (DDI_FAILURE);
+	}
+
+	/* figure out which DMA engine hardware we have */
+	if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
+	    "dma-model", &prop_str) == DDI_PROP_SUCCESS) {
+		if (strcmp(prop_str, "eb2dma") == 0) {
+			state->cs_dma_engine = EB2_DMA;
+			state->cs_dma_ops = &cs4231_eb2dma_ops;
+		} else {
+			state->cs_dma_engine = APC_DMA;
+			state->cs_dma_ops = &cs4231_apcdma_ops;
+		}
+		ddi_prop_free(prop_str);
+	} else {
+		state->cs_dma_engine = APC_DMA;
+		state->cs_dma_ops = &cs4231_apcdma_ops;
+	}
+
+	/* cs_regs, cs_eb2_regs and cs_handles filled in later */
+
+	/* most of what's left is filled in when the registers are mapped */
+
+	audiocs_get_ports(state);
+
+	/* Allocate engines, must be done before register mapping called  */
+	if ((audiocs_alloc_engine(state, CS4231_PLAY) != DDI_SUCCESS) ||
+	    (audiocs_alloc_engine(state, CS4231_REC) != DDI_SUCCESS)) {
+		return (DDI_FAILURE);
+	}
+
+	/* Map in the registers */
+	if (CS4231_DMA_MAP_REGS(state) == DDI_FAILURE) {
+		return (DDI_FAILURE);
+	}
+
+
+	/* Allocate and add controls, must be done *after* registers mapped */
+	if (audiocs_add_controls(state) != DDI_SUCCESS) {
+		return (DDI_FAILURE);
+	}
+
+	state->cs_suspended = B_FALSE;
+	state->cs_powered = B_FALSE;
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_get_ports()
+ *
+ * Description:
+ *	Get which audiocs h/w version we have and use this to
+ *	determine the input and output ports as well whether or not
+ *	the hardware has internal loopbacks or not. We also have three
+ *	different ways for the properties to be specified, which we
+ *	also need to worry about.
+ *
+ * Vers	Platform(s)	DMA eng.	audio-module**	loopback
+ * a    SS-4+/SS-5+	apcdma		no		no
+ * b	Ultra-1&2	apcdma		no		yes
+ * c	positron	apcdma		no		yes
+ * d	PPC - retired
+ * e	x86 - retired
+ * f	tazmo		eb2dma		Perigee		no
+ * g	tazmo		eb2dma		Quark		yes
+ * h	darwin+		eb2dma		no		N/A
+ *
+ * Vers	model~		aux1*		aux2*
+ * a	N/A		N/A		N/A
+ * b	N/A		N/A		N/A
+ * c	N/A		N/A		N/A
+ * d	retired
+ * e	retired
+ * f	SUNW,CS4231f	N/A		N/A
+ * g	SUNW,CS4231g	N/A		N/A
+ * h	SUNW,CS4231h	cdrom		none
+ *
+ * *   = Replaces internal-loopback for latest property type, can be
+ *	 set to "cdrom", "loopback", or "none".
+ *
+ * **  = For plugin audio modules only. Starting with darwin, this
+ *	 property is replaces by the model property.
+ *
+ * ~   = Replaces audio-module.
+ *
+ * +   = Has the capability of having a cable run from the internal
+ *	 CD-ROM to the audio device.
+ *
+ * N/A = Not applicable, the property wasn't created for early
+ *	 platforms, or the property has been retired.
+ *
+ * NOTE: Older tazmo and quark machines don't have the model property.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ */
+static void
+audiocs_get_ports(CS_state_t *state)
+{
+	dev_info_t	*dip = state->cs_dip;
+	audio_dev_t	*adev = state->cs_adev;
+	char		*prop_str;
+
+	/* First we set the common ports, etc. */
+	state->cs_omask = state->cs_omod =
+	    (1U << OUTPUT_SPEAKER) |
+	    (1U << OUTPUT_HEADPHONES) |
+	    (1U << OUTPUT_LINEOUT);
+	state->cs_imask =
+	    (1U << INPUT_MIC) |
+	    (1U << INPUT_LINEIN) |
+	    (1U << INPUT_STEREOMIX);
+
+	/* now we try the new "model" property */
+	if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
+	    "model", &prop_str) == DDI_PROP_SUCCESS) {
+		if (strcmp(prop_str, "SUNW,CS4231h") == 0) {
+			/* darwin */
+			audio_dev_set_version(adev, CS_DEV_VERSION_H);
+			state->cs_imask |= (1U << INPUT_CD);
+			state->cs_omod = (1U << OUTPUT_SPEAKER);
+		} else if (strcmp(prop_str, "SUNW,CS4231g") == 0) {
+			/* quark audio module */
+			audio_dev_set_version(adev, CS_DEV_VERSION_G);
+			/*
+			 * NB: This could do SUNVTS LOOPBACK, but we
+			 * don't support it for now... owing to no
+			 * support in framework.
+			 */
+		} else if (strcmp(prop_str, "SUNW,CS4231f") == 0) {
+			/* tazmo */
+			audio_dev_set_version(adev, CS_DEV_VERSION_F);
+		} else {
+			audio_dev_set_version(adev, prop_str);
+			audio_dev_warn(adev,
+			    "unknown audio model: %s, some parts of "
+			    "audio may not work correctly", prop_str);
+		}
+		ddi_prop_free(prop_str);	/* done with the property */
+	} else {	/* now try the older "audio-module" property */
+		if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip,
+		    DDI_PROP_DONTPASS, "audio-module", &prop_str) ==
+		    DDI_PROP_SUCCESS) {
+			switch (*prop_str) {
+			case 'Q':	/* quark audio module */
+				audio_dev_set_version(adev, CS_DEV_VERSION_G);
+				/* See quark comment above about SunVTS */
+				break;
+			case 'P':	/* tazmo */
+				audio_dev_set_version(adev, CS_DEV_VERSION_F);
+				break;
+			default:
+				audio_dev_set_version(adev, prop_str);
+				audio_dev_warn(adev,
+				    "unknown audio module: %s, some "
+				    "parts of audio may not work correctly",
+				    prop_str);
+				break;
+			}
+			ddi_prop_free(prop_str);	/* done with the prop */
+		} else {	/* now try heuristics, ;-( */
+			if (ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+			    DDI_PROP_DONTPASS, "internal-loopback", B_FALSE)) {
+				if (state->cs_dma_engine == EB2_DMA) {
+					audio_dev_set_version(adev,
+					    CS_DEV_VERSION_C);
+				} else {
+					audio_dev_set_version(adev,
+					    CS_DEV_VERSION_B);
+				}
+				/*
+				 * Again, we don't support SunVTS for these
+				 * boards, although we potentially could.
+				 */
+			} else {
+				audio_dev_set_version(adev, CS_DEV_VERSION_A);
+				state->cs_imask |= (1U << INPUT_CD);
+			}
+		}
+	}
+}
+
+/*
+ * audiocs_power_up()
+ *
+ * Description:
+ *	Power up the Codec and restore the codec's registers.
+ *
+ *	NOTE: We don't worry about locking since the only routines
+ *		that may call us are attach() and power() Both of
+ *		which should be the only threads in the driver.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ */
+static void
+audiocs_power_up(CS_state_t *state)
+{
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+	int			i;
+
+	/* turn on the Codec */
+	CS4231_DMA_POWER(state, CS4231_PWR_ON);
+
+	/* reset the DMA engine(s) */
+	CS4231_DMA_RESET(state);
+
+	/*
+	 * Reload the Codec's registers, the DMA engines will be
+	 * taken care of when play and record start up again. But
+	 * first enable registers 16 -> 31.
+	 */
+	SELIDX(state, MID_REG);
+	PUTIDX(state, state->cs_save[MID_REG], MID_VALID_MASK);
+
+	for (i = 0; i < CS4231_REGS; i++) {
+		/* restore Codec registers */
+		SELIDX(state, (i | IAR_MCE));
+		ddi_put8(handle, &CS4231_IDR, state->cs_save[i]);
+		drv_usecwait(500);	/* chip bug */
+	}
+	/* clear MCE bit */
+	SELIDX(state, 0);
+}
+
+/*
+ * audiocs_power_down()
+ *
+ * Description:
+ *	Power down the Codec and save the codec's registers.
+ *
+ *	NOTE: See the note in cs4231_power_up() about locking.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ */
+static void
+audiocs_power_down(CS_state_t *state)
+{
+	ddi_acc_handle_t	handle;
+	int			i;
+
+	handle = state->cs_handles.cs_codec_hndl;
+
+	/*
+	 * We are powering down, so we don't need to do a thing with
+	 * the DMA engines. However, we do need to save the Codec
+	 * registers.
+	 */
+
+	for (i = 0; i < CS4231_REGS; i++) {
+		/* save Codec regs */
+		SELIDX(state, i);
+		state->cs_save[i] = ddi_get8(handle, &CS4231_IDR);
+	}
+
+	/* turn off the Codec */
+	CS4231_DMA_POWER(state, CS4231_PWR_OFF);
+
+}	/* cs4231_power_down() */
+
+/*
+ * audiocs_configure_input()
+ *
+ * Description:
+ *	Configure input properties of the mixer (e.g. igain, ports).
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ */
+static void
+audiocs_configure_input(CS_state_t *state)
+{
+	uint8_t		l, r;
+	uint64_t	inputs;
+	uint64_t	micboost;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	if (state->cs_suspended)
+		return;
+
+	inputs = state->cs_inputs->cc_val;
+	micboost = state->cs_micboost->cc_val;
+	r = (state->cs_igain->cc_val & 0xff);
+	l = ((state->cs_igain->cc_val & 0xff00) >> 8);
+
+	/* rescale these for our atten array */
+	l = (((uint32_t)l * 255) / 100) & 0xff;
+	r = (((uint32_t)r * 255) / 100) & 0xff;
+
+	/* we downshift by 4 bits -- igain only has 16 possible values */
+	/* NB: that we do not scale here!  The SADA driver didn't do so. */
+	l = l >> 4;
+	r = r >> 4;
+
+	if (inputs & (1U << INPUT_MIC)) {
+		l |= LADCI_LMIC;
+		r |= RADCI_RMIC;
+	}
+	if (inputs & (1U << INPUT_LINEIN)) {
+		l |= LADCI_LLINE;
+		r |= RADCI_RLINE;
+	}
+	if (inputs & (1U << INPUT_CD)) {
+		/* note that SunVTS also uses this */
+		l |= LADCI_LAUX1;
+		r |= RADCI_RAUX1;
+	}
+	if (inputs & (1U << INPUT_STEREOMIX)) {
+		l |= LADCI_LLOOP;
+		r |= RADCI_RLOOP;
+	}
+	if (micboost) {
+		l |= LADCI_LMGE;
+		r |= RADCI_RMGE;
+	}
+
+	SELIDX(state, LADCI_REG);
+	PUTIDX(state, l, LADCI_VALID_MASK);
+
+	SELIDX(state, RADCI_REG);
+	PUTIDX(state, r, RADCI_VALID_MASK);
+}
+
+/*
+ * audiocs_configure_output()
+ *
+ * Description:
+ *	Configure output properties of the mixer (e.g. ogain, mgain).
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ */
+static void
+audiocs_configure_output(CS_state_t *state)
+{
+	uint64_t		outputs;
+	uint8_t			l, r;
+	uint8_t			rmute, lmute;
+	uint8_t			mgain;
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+
+	rmute = lmute = 0;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	if (state->cs_suspended)
+		return;
+
+	outputs = state->cs_outputs->cc_val;
+
+	/* port selection */
+	SELIDX(state, MIOC_REG);
+	if (outputs & (1U << OUTPUT_SPEAKER)) {
+		ANDIDX(state, ~MIOC_MONO_SPKR_MUTE, MIOC_VALID_MASK);
+	} else {
+		ORIDX(state, MIOC_MONO_SPKR_MUTE, MIOC_VALID_MASK);
+	}
+	SELIDX(state, PC_REG);
+	if (outputs & (1U << OUTPUT_HEADPHONES)) {
+		ANDIDX(state, ~PC_HEADPHONE_MUTE, PC_VALID_MASK);
+	} else {
+		ORIDX(state, PC_HEADPHONE_MUTE, PC_VALID_MASK);
+	}
+	SELIDX(state, PC_REG);
+	if (outputs & (1U << OUTPUT_LINEOUT)) {
+		ANDIDX(state, ~PC_LINE_OUT_MUTE, PC_VALID_MASK);
+	} else {
+		ORIDX(state, PC_LINE_OUT_MUTE, PC_VALID_MASK);
+	}
+
+	/* monitor gain */
+	mgain = cs4231_atten[((state->cs_mgain->cc_val * 255) / 100) & 0xff];
+	SELIDX(state, LC_REG);
+	if (mgain == 0) {
+		/* disable loopbacks when gain == 0 */
+		PUTIDX(state, LC_OFF, LC_VALID_MASK);
+	} else {
+		/* we use cs4231_atten[] to linearize attenuation */
+		PUTIDX(state, (mgain << 2) | LC_LBE, LC_VALID_MASK);
+	}
+
+	/* output gain */
+	l = ((state->cs_ogain->cc_val >> 8) & 0xff);
+	r = (state->cs_ogain->cc_val & 0xff);
+	if (l == 0) {
+		lmute = LDACO_LDM;
+	}
+	if (r == 0) {
+		rmute = RDACO_RDM;
+	}
+
+	/* rescale these for our atten array */
+	l = cs4231_atten[(((uint32_t)l * 255) / 100) & 0xff] | lmute;
+	r = cs4231_atten[(((uint32_t)r * 255) / 100) & 0xff] | rmute;
+
+	SELIDX(state, LDACO_REG);
+	PUTIDX(state, l, LDAC0_VALID_MASK);
+	SELIDX(state, RDACO_REG);
+	PUTIDX(state, r, RDAC0_VALID_MASK);
+}
+
+/*
+ * audiocs_get_value()
+ *
+ * Description:
+ *	Get a control value
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	*valp		Pointer to store value.
+ *
+ * Returns:
+ *	0		The Codec parameter has been retrieved.
+ */
+static int
+audiocs_get_value(void *arg, uint64_t *valp)
+{
+	CS_ctrl_t		*cc = arg;
+	CS_state_t		*state = cc->cc_state;
+
+	mutex_enter(&state->cs_lock);
+	*valp = cc->cc_val;
+	mutex_exit(&state->cs_lock);
+	return (0);
+}
+
+
+/*
+ * audiocs_set_ogain()
+ *
+ * Description:
+ *	Set the play gain.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The gain to set (both left and right)
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_ogain(void *arg, uint64_t val)
+{
+	CS_ctrl_t		*cc = arg;
+	CS_state_t		*state = cc->cc_state;
+
+	if ((val & ~0xffff) ||
+	    ((val & 0xff) > 100) ||
+	    (((val & 0xff00) >> 8) > 100))
+		return (EINVAL);
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = val;
+	audiocs_configure_output(state);
+	mutex_exit(&state->cs_lock);
+	return (0);
+}
+
+/*
+ * audiocs_set_micboost()
+ *
+ * Description:
+ *	Set the 20 dB microphone boost.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The 1 to enable, 0 to disable.
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_micboost(void *arg, uint64_t val)
+{
+	CS_ctrl_t	*cc = arg;
+	CS_state_t	*state = cc->cc_state;
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = val ? B_TRUE : B_FALSE;
+	audiocs_configure_input(state);
+	mutex_exit(&state->cs_lock);
+	return (0);
+}
+
+/*
+ * audiocs_set_igain()
+ *
+ * Description:
+ *	Set the record gain.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The gain to set (both left and right)
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_igain(void *arg, uint64_t val)
+{
+	CS_ctrl_t	*cc = arg;
+	CS_state_t	*state = cc->cc_state;
+
+	if ((val & ~0xffff) ||
+	    ((val & 0xff) > 100) ||
+	    (((val & 0xff00) >> 8) > 100))
+		return (EINVAL);
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = val;
+	audiocs_configure_input(state);
+	mutex_exit(&state->cs_lock);
+
+	return (0);
+}
+
+/*
+ * audiocs_set_inputs()
+ *
+ * Description:
+ *	Set the input ports.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The mask of output ports.
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_inputs(void *arg, uint64_t val)
+{
+	CS_ctrl_t	*cc = arg;
+	CS_state_t	*state = cc->cc_state;
+
+	if (val & ~(state->cs_imask))
+		return (EINVAL);
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = val;
+	audiocs_configure_input(state);
+	mutex_exit(&state->cs_lock);
+
+	return (0);
+}
+
+/*
+ * audiocs_set_outputs()
+ *
+ * Description:
+ *	Set the output ports.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The mask of input ports.
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_outputs(void *arg, uint64_t val)
+{
+	CS_ctrl_t	*cc = arg;
+	CS_state_t	*state = cc->cc_state;
+
+	if ((val & ~(state->cs_omod)) !=
+	    (state->cs_omask & ~state->cs_omod))
+		return (EINVAL);
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = val;
+	audiocs_configure_output(state);
+	mutex_exit(&state->cs_lock);
+
+	return (0);
+}
+
+/*
+ * audiocs_set_mgain()
+ *
+ * Description:
+ *	Set the monitor gain.
+ *
+ * Arguments:
+ *	void		*arg		The device's state structure
+ *	uint64_t	val		The gain to set (monoaural).)
+ *
+ * Returns:
+ *	0		The Codec parameter has been set
+ */
+static int
+audiocs_set_mgain(void *arg, uint64_t gain)
+{
+	CS_ctrl_t	*cc = arg;
+	CS_state_t	*state = cc->cc_state;
+
+	if (gain > 100)
+		return (EINVAL);
+
+	mutex_enter(&state->cs_lock);
+	cc->cc_val = gain;
+	audiocs_configure_output(state);
+	mutex_exit(&state->cs_lock);
+
+	return (0);
+}
+
+/*
+ * audiocs_open()
+ *
+ * Description:
+ *	Opens a DMA engine for use.
+ *
+ * Arguments:
+ *	void		*arg		The DMA engine to set up
+ *	int		flag		Open flags
+ *	unsigned	*fragfrp	Receives number of frames per fragment
+ *	unsigned	*nfragsp	Receives number of fragments
+ *	caddr_t		*bufp		Receives kernel data buffer
+ *
+ * Returns:
+ *	0	on success
+ *	errno	on failure
+ */
+static int
+audiocs_open(void *arg, int flag,
+    unsigned *fragfrp, unsigned *nfragsp, caddr_t *bufp)
+{
+	CS_engine_t	*eng = arg;
+	CS_state_t	*state = eng->ce_state;
+	dev_info_t	*dip = state->cs_dip;
+
+	_NOTE(ARGUNUSED(flag));
+
+	(void) pm_busy_component(dip, CS4231_COMPONENT);
+	if (pm_raise_power(dip, CS4231_COMPONENT, CS4231_PWR_ON) ==
+	    DDI_FAILURE) {
+
+		/* match the busy call above */
+		(void) pm_idle_component(dip, CS4231_COMPONENT);
+
+		audio_dev_warn(state->cs_adev, "power up failed");
+	}
+
+	eng->ce_started = B_FALSE;
+	eng->ce_count = 0;
+
+	*fragfrp = eng->ce_fragfr;
+	*nfragsp = CS4231_NFRAGS;
+	*bufp = eng->ce_kaddr;
+
+	mutex_enter(&state->cs_lock);
+	audiocs_reset_engine(eng);
+	mutex_exit(&state->cs_lock);
+	return (0);
+}
+
+/*
+ * audiocs_close()
+ *
+ * Description:
+ *	Closes an audio DMA engine that was previously opened.  Since
+ *	nobody is using it, we take this opportunity to possibly power
+ *	down the entire device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to shut down
+ */
+static void
+audiocs_close(void *arg)
+{
+	CS_engine_t	*eng = arg;
+	CS_state_t	*state = eng->ce_state;
+
+	mutex_enter(&state->cs_lock);
+	audiocs_stop_engine(eng);
+	eng->ce_started = B_FALSE;
+	mutex_exit(&state->cs_lock);
+
+	(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
+}
+
+/*
+ * audiocs_stop()
+ *
+ * Description:
+ *	This is called by the framework to stop an engine that is
+ *	transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to stop
+ */
+static void
+audiocs_stop(void *arg)
+{
+	CS_engine_t	*eng = arg;
+	CS_state_t	*state = eng->ce_state;
+
+	mutex_enter(&state->cs_lock);
+	if (eng->ce_started) {
+		audiocs_stop_engine(eng);
+		eng->ce_started = B_FALSE;
+	}
+	mutex_exit(&state->cs_lock);
+}
+
+/*
+ * audiocs_start()
+ *
+ * Description:
+ *	This is called by the framework to start an engine transferring data.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to start
+ *
+ * Returns:
+ *	0 	on success, an errno otherwise
+ */
+static int
+audiocs_start(void *arg)
+{
+	CS_engine_t	*eng = arg;
+	CS_state_t	*state = eng->ce_state;
+	int		rv = 0;
+
+	mutex_enter(&state->cs_lock);
+	if (!eng->ce_started) {
+		if (audiocs_start_engine(eng) == DDI_SUCCESS) {
+			eng->ce_started = B_TRUE;
+		} else {
+			rv = EIO;
+		}
+	}
+	mutex_exit(&state->cs_lock);
+	return (rv);
+}
+
+/*
+ * audiocs_format()
+ *
+ * Description:
+ *	Called by the framework to query the format of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	AUDIO_FORMAT_S16_NE
+ */
+static int
+audiocs_format(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (AUDIO_FORMAT_S16_NE);
+}
+
+/*
+ * audiocs_channels()
+ *
+ * Description:
+ *	Called by the framework to query the channels of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	2 (stereo)
+ */
+static int
+audiocs_channels(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (2);
+}
+
+/*
+ * audiocs_rates()
+ *
+ * Description:
+ *	Called by the framework to query the sample rate of the device.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	48000
+ */
+static int
+audiocs_rate(void *arg)
+{
+	_NOTE(ARGUNUSED(arg));
+
+	return (48000);
+}
+
+/*
+ * audiocs_count()
+ *
+ * Description:
+ *	This is called by the framework to get the engine's frame counter
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	frame count for current engine
+ */
+static uint64_t
+audiocs_count(void *arg)
+{
+	CS_engine_t	*eng = arg;
+	CS_state_t	*state = eng->ce_state;
+	uint64_t	val;
+
+	mutex_enter(&state->cs_lock);
+	val = eng->ce_count;
+	mutex_exit(&state->cs_lock);
+
+	return (val);
+}
+
+/*
+ * audiocs_sync()
+ *
+ * Description:
+ *	This is called by the framework to synchronize DMA caches.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to sync
+ */
+static void
+audiocs_sync(void *arg, unsigned nframes)
+{
+	CS_engine_t *eng = arg;
+	_NOTE(ARGUNUSED(nframes));
+
+	(void) ddi_dma_sync(eng->ce_dmah, 0, 0, eng->ce_syncdir);
+}
+
+/*
+ * audiocs_qlen()
+ *
+ * Description:
+ *	This is called by the framework to determine on-device queue length.
+ *
+ * Arguments:
+ *	void	*arg		The DMA engine to query
+ *
+ * Returns:
+ *	hardware queue length not reported by count (0 for this device)
+ */
+static size_t
+audiocs_qlen(void *arg)
+{
+	CS_engine_t	*eng = arg;
+
+	return (eng->ce_fragfr);
+}
+
+
+/*
+ * audiocs_reset_engine()
+ *
+ * Description:
+ *	This routine resets the DMA engine pareparing it for work.
+ *
+ * Arguments:
+ *	CS_engine_t	*engine		DMA engine to stop.
+ */
+void
+audiocs_reset_engine(CS_engine_t *eng)
+{
+	CS_state_t	*state = eng->ce_state;
+	uint8_t		mask;
+	uint8_t		value;
+	uint8_t		reg;
+
+	if (eng->ce_num == CS4231_PLAY) {
+		/* sample rate only set on play side */
+		value = FS_48000 | PDF_STEREO | PDF_LINEAR16NE;
+		reg = FSDF_REG;
+		mask = FSDF_VALID_MASK;
+	} else {
+		value = CDF_STEREO | CDF_LINEAR16NE;
+		reg = CDF_REG;
+		mask = CDF_VALID_MASK;
+	}
+
+	SELIDX(state, reg | IAR_MCE);
+	PUTIDX(state, value, mask);
+
+	(void) audiocs_poll_ready(state);
+
+	/* clear the mode change bit */
+	SELIDX(state, reg);
+}
+
+/*
+ * audiocs_alloc_engine()
+ *
+ * Description:
+ *	Allocates the DMA handles and the memory for the DMA engine.
+ *
+ * Arguments:
+ *	CS_state_t	*dip	Pointer to the device's soft state
+ *	int		num	Engine number, CS4231_PLAY or CS4231_REC.
+ *
+ * Returns:
+ *	DDI_SUCCESS		Engine initialized.
+ *	DDI_FAILURE		Engine not initialized.
+ */
+int
+audiocs_alloc_engine(CS_state_t *state, int num)
+{
+	char			*prop;
+	unsigned		caps;
+	int			dir;
+	int			rc;
+	audio_dev_t		*adev;
+	dev_info_t		*dip;
+	CS_engine_t		*eng;
+	uint_t			ccnt;
+	ddi_dma_cookie_t	dmac;
+
+	static ddi_device_acc_attr_t buf_attr = {
+		DDI_DEVICE_ATTR_V0,
+		DDI_NEVERSWAP_ACC,
+		DDI_STRICTORDER_ACC
+	};
+
+	adev = state->cs_adev;
+	dip = state->cs_dip;
+
+	eng = kmem_zalloc(sizeof (*eng), KM_SLEEP);
+	eng->ce_state = state;
+	eng->ce_started = B_FALSE;
+	eng->ce_num = num;
+
+	switch (num) {
+	case CS4231_REC:
+		prop = "record-interrupts";
+		dir = DDI_DMA_READ;
+		caps = ENGINE_INPUT_CAP;
+		eng->ce_syncdir = DDI_DMA_SYNC_FORKERNEL;
+		eng->ce_codec_en = INTC_CEN;
+		break;
+	case CS4231_PLAY:
+		prop = "play-interrupts";
+		dir = DDI_DMA_WRITE;
+		caps = ENGINE_OUTPUT_CAP;
+		eng->ce_syncdir = DDI_DMA_SYNC_FORDEV;
+		eng->ce_codec_en = INTC_PEN;
+		break;
+	default:
+		kmem_free(eng, sizeof (*eng));
+		audio_dev_warn(adev, "bad engine number (%d)!", num);
+		return (DDI_FAILURE);
+	}
+	state->cs_engines[num] = eng;
+
+	eng->ce_intrs = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
+	    DDI_PROP_DONTPASS, prop, CS4231_INTS);
+
+	/* make sure the values are good */
+	if (eng->ce_intrs < CS4231_MIN_INTS) {
+		audio_dev_warn(adev, "%s too low, %d, resetting to %d",
+		    prop, eng->ce_intrs, CS4231_INTS);
+		eng->ce_intrs = CS4231_INTS;
+	} else if (eng->ce_intrs > CS4231_MAX_INTS) {
+		audio_dev_warn(adev, "%s too high, %d, resetting to %d",
+		    prop, eng->ce_intrs, CS4231_INTS);
+		eng->ce_intrs = CS4231_INTS;
+	}
+
+	/*
+	 * Figure out how much space we need.  Sample rate is 48kHz, and
+	 * we need to store 8 chunks.  (Note that this means that low
+	 * interrupt frequencies will require more RAM.  We could probably
+	 * do some cleverness to use a more dynamic list.)
+	 */
+	eng->ce_fragfr = 48000 / eng->ce_intrs;
+	eng->ce_fragfr &= ~(64 - 1);	/* align @ 64B boundaries */
+	eng->ce_fragfr = max(eng->ce_fragfr, 64);
+	eng->ce_fragsz = eng->ce_fragfr * 4;	/* each frame is 4 bytes */
+	eng->ce_size = eng->ce_fragsz * CS4231_NFRAGS;
+
+	/* allocate dma handle */
+	rc = ddi_dma_alloc_handle(dip, CS4231_DMA_ATTR(state), DDI_DMA_SLEEP,
+	    NULL, &eng->ce_dmah);
+	if (rc != DDI_SUCCESS) {
+		audio_dev_warn(adev, "ddi_dma_alloc_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+	/* allocate DMA buffer */
+	rc = ddi_dma_mem_alloc(eng->ce_dmah, eng->ce_size, &buf_attr,
+	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &eng->ce_kaddr,
+	    &eng->ce_size, &eng->ce_acch);
+	if (rc == DDI_FAILURE) {
+		audio_dev_warn(adev, "dma_mem_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	/* bind DMA buffer */
+	rc = ddi_dma_addr_bind_handle(eng->ce_dmah, NULL,
+	    eng->ce_kaddr, eng->ce_size, dir | DDI_DMA_CONSISTENT,
+	    DDI_DMA_SLEEP, NULL, &dmac, &ccnt);
+	if ((rc != DDI_DMA_MAPPED) || (ccnt != 1)) {
+		audio_dev_warn(adev,
+		    "ddi_dma_addr_bind_handle failed: %d", rc);
+		return (DDI_FAILURE);
+	}
+
+	/* save off phys addresses for each frag */
+	for (int i = 0; i < CS4231_NFRAGS; i++) {
+		eng->ce_paddr[i] = dmac.dmac_address;
+		dmac.dmac_address += eng->ce_fragsz;
+	}
+
+	eng->ce_engine = audio_engine_alloc(&audiocs_engine_ops, caps);
+	if (eng->ce_engine == NULL) {
+		audio_dev_warn(adev, "audio_engine_alloc failed");
+		return (DDI_FAILURE);
+	}
+
+	audio_engine_set_private(eng->ce_engine, eng);
+	audio_dev_add_engine(adev, eng->ce_engine);
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_free_engine()
+ *
+ * Description:
+ *	This routine fress the engine and all associated resources.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng	Engine to free.
+ */
+void
+audiocs_free_engine(CS_engine_t *eng)
+{
+	CS_state_t	*state = eng->ce_state;
+	audio_dev_t	*adev = state->cs_adev;
+
+	if (eng == NULL)
+		return;
+	if (eng->ce_engine) {
+		audio_dev_remove_engine(adev, eng->ce_engine);
+		audio_engine_free(eng->ce_engine);
+	}
+	if (eng->ce_paddr[0]) {
+		(void) ddi_dma_unbind_handle(eng->ce_dmah);
+	}
+	if (eng->ce_acch) {
+		ddi_dma_mem_free(&eng->ce_acch);
+	}
+	if (eng->ce_dmah) {
+		ddi_dma_free_handle(&eng->ce_dmah);
+	}
+	kmem_free(eng, sizeof (*eng));
+}
+
+/*
+ * audiocs_start_port()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng		Port of DMA engine to start.
+ *
+ * Returns:
+ *	DDI_SUCCESS	DMA engine started.
+ *	DDI_FAILURE	DMA engine not started.
+ */
+int
+audiocs_start_engine(CS_engine_t *eng)
+{
+	CS_state_t		*state = eng->ce_state;
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	/*
+	 * If we are suspended, we can't touch hardware.
+	 */
+	if (state->cs_suspended)
+		return (DDI_SUCCESS);
+
+	/*
+	 * Start the DMA engine.
+	 */
+	if (CS4231_DMA_START(state, eng) != DDI_SUCCESS)
+		return (DDI_FAILURE);
+
+	/*
+	 * Start the codec.
+	 */
+	SELIDX(state, INTC_REG);
+	ORIDX(state, eng->ce_codec_en, INTC_VALID_MASK);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * audiocs_stop_engine()
+ *
+ * Description:
+ *	This routine stop the DMA engine.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng		DMA engine to stop.
+ */
+void
+audiocs_stop_engine(CS_engine_t *eng)
+{
+	CS_state_t	*state = eng->ce_state;
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	/*
+	 * If we are suspended, we can't touch hardware.
+	 */
+	if (state->cs_suspended)
+		return;
+
+	/*
+	 * Stop the DMA engine.
+	 */
+	CS4231_DMA_STOP(state, eng);
+
+	/*
+	 * Stop the codec.
+	 */
+	SELIDX(state, INTC_REG);
+	ANDIDX(state, ~(eng->ce_codec_en), INTC_VALID_MASK);
+}
+
+/*
+ * audiocs_poll_ready()
+ *
+ * Description:
+ *	This routine waits for the Codec to complete its initialization
+ *	sequence and is done with its autocalibration.
+ *
+ *	Early versions of the Codec have a bug that can take as long as
+ *	15 seconds to complete its initialization. For these cases we
+ *	use a timeout mechanism so we don't keep the machine locked up.
+ *
+ * Arguments:
+ *	CS_state_t	*state	The device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		The Codec is ready to continue
+ *	DDI_FAILURE		The Codec isn't ready to continue
+ */
+int
+audiocs_poll_ready(CS_state_t *state)
+{
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+	int			x = 0;
+	uint8_t			iar;
+	uint8_t			idr;
+
+	ASSERT(state->cs_regs != NULL);
+	ASSERT(handle != NULL);
+
+	/* wait for the chip to initialize itself */
+	iar = ddi_get8(handle, &CS4231_IAR);
+
+	while ((iar & IAR_INIT) && x++ < CS4231_TIMEOUT) {
+		drv_usecwait(50);
+		iar = ddi_get8(handle, &CS4231_IAR);
+	}
+
+	if (x >= CS4231_TIMEOUT) {
+		return (DDI_FAILURE);
+	}
+
+	x = 0;
+
+	/*
+	 * Now wait for the chip to complete its autocalibration.
+	 * Set the test register.
+	 */
+	SELIDX(state, ESI_REG);
+
+	idr = ddi_get8(handle, &CS4231_IDR);
+
+	while ((idr & ESI_ACI) && x++ < CS4231_TIMEOUT) {
+		drv_usecwait(50);
+		idr = ddi_get8(handle, &CS4231_IDR);
+	}
+
+	if (x >= CS4231_TIMEOUT) {
+		return (DDI_FAILURE);
+	}
+
+
+	return (DDI_SUCCESS);
+
+}
+
+/*
+ * audiocs_sel_index()
+ *
+ * Description:
+ *	Select a cs4231 register. The cs4231 has a hardware bug where a
+ *	register is not always selected the first time. We try and try
+ *	again until the proper register is selected or we time out and
+ *	print an error message.
+ *
+ * Arguments:
+ *	audiohdl_t	ahandle		Handle to this device
+ *	ddi_acc_handle_t handle		A handle to the device's registers
+ *	uint8_t		addr		The register address to program
+ *	int		reg		The register to select
+ *
+ * Returns:
+ *	void
+ */
+void
+#ifdef	DEBUG
+audiocs_sel_index(CS_state_t *state, uint8_t reg, int n)
+#else
+audiocs_sel_index(CS_state_t *state, uint8_t reg)
+#endif
+{
+	int			x;
+	uint8_t			T;
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+	uint8_t			*addr = &CS4231_IAR;
+
+	for (x = 0; x < CS4231_RETRIES; x++) {
+		ddi_put8(handle, addr, reg);
+		T = ddi_get8(handle, addr);
+		if (T == reg) {
+			break;
+		}
+		drv_usecwait(1000);
+	}
+
+	if (x == CS4231_RETRIES) {
+		audio_dev_warn(state->cs_adev,
+#ifdef	DEBUG
+		    "line %d: Couldn't select index (0x%02x 0x%02x)", n,
+#else
+		    "Couldn't select index (0x%02x 0x%02x)",
+#endif
+		    T, reg);
+		audio_dev_warn(state->cs_adev,
+		    "audio may not work correctly until it is stopped and "
+		    "restarted");
+	}
+}
+
+/*
+ * audiocs_put_index()
+ *
+ * Description:
+ *	Program a cs4231 register. The cs4231 has a hardware bug where a
+ *	register is not programmed properly the first time. We program a value,
+ *	then immediately read back the value and reprogram if nescessary.
+ *	We do this until the register is properly programmed or we time out and
+ *	print an error message.
+ *
+ * Arguments:
+ *	CS_state_t	state		Handle to this device
+ *	uint8_t		mask		Mask to not set reserved register bits
+ *	int		val		The value to program
+ *
+ * Returns:
+ *	void
+ */
+void
+#ifdef DEBUG
+audiocs_put_index(CS_state_t *state, uint8_t val, uint8_t mask, int n)
+#else
+audiocs_put_index(CS_state_t *state, uint8_t val, uint8_t mask)
+#endif
+{
+	int			x;
+	uint8_t			T;
+	ddi_acc_handle_t	handle = CODEC_HANDLE;
+	uint8_t			*addr = &CS4231_IDR;
+
+	val &= mask;
+
+	for (x = 0; x < CS4231_RETRIES; x++) {
+		ddi_put8(handle, addr, val);
+		T = ddi_get8(handle, addr);
+		if (T == val) {
+			break;
+		}
+		drv_usecwait(1000);
+	}
+
+	if (x == CS4231_RETRIES) {
+#ifdef DEBUG
+		audio_dev_warn(state->cs_adev,
+		    "line %d: Couldn't set value (0x%02x 0x%02x)", n, T, val);
+#else
+		audio_dev_warn(state->cs_adev,
+		    "Couldn't set value (0x%02x 0x%02x)", T, val);
+#endif
+		audio_dev_warn(state->cs_adev,
+		    "audio may not work correctly until it is stopped and "
+		    "restarted");
+	}
+}
diff --git a/usr/src/uts/sun/sys/audio/impl/audio_4231_impl.h b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231.h
index db378f458d..3f5cbe1135 100644
--- a/usr/src/uts/sun/sys/audio/impl/audio_4231_impl.h
+++ b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231.h
@@ -2,9 +2,8 @@
  * CDDL HEADER START
  *
  * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
  *
  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
  * or http://www.opensolaris.org/os/licensing.
@@ -20,32 +19,65 @@
  * CDDL HEADER END
  */
 /*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
  * Use is subject to license terms.
  */
 
-#ifndef	_SYS_AUDIO4231_IMPL_H
-#define	_SYS_AUDIO4231_IMPL_H
+#ifndef _AUDIO_4231_H
+#define	_AUDIO_4231_H
 
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
+#ifdef __cplusplus
 extern "C" {
 #endif
 
 /*
- * Implementation specific header file for the audiocs device driver.
+ * Header file for the audiocs device driver.
  */
 
-#ifdef _KERNEL
+/*
+ * Values returned by the AUDIO_GETDEV ioctl()
+ */
+#define	CS_DEV_NAME		"SUNW,CS4231"
+#define	CS_DEV_CONFIG_ONBRD1	"onboard1"
+#define	CS_DEV_VERSION		"a"	/* SS5 				*/
+#define	CS_DEV_VERSION_A	CS_DEV_VERSION
+#define	CS_DEV_VERSION_B	"b"	/* Electron - internal loopback	*/
+#define	CS_DEV_VERSION_C	"c"	/* Positron			*/
+#define	CS_DEV_VERSION_D	"d"	/* PowerPC - Retired		*/
+#define	CS_DEV_VERSION_E	"e"	/* x86 - Retired		*/
+#define	CS_DEV_VERSION_F	"f"	/* Tazmo			*/
+#define	CS_DEV_VERSION_G	"g"	/* Quark Audio Module		*/
+#define	CS_DEV_VERSION_H	"h"	/* Darwin			*/
 
 /*
- * Global routines.
+ * Driver supported configuration information
  */
-void cs4231_reg_select(audiohdl_t ahandle, ddi_acc_handle_t, uint8_t *,
-    uint8_t, int, char *);
-void cs4231_put8(audiohdl_t ahandle, ddi_acc_handle_t, uint8_t *, uint8_t,
-    uint8_t, int, char *);
+#define	CS4231_NAME		"audiocs"
+#define	CS4231_MOD_NAME		"CS4231 audio driver"
+
+#define	CS4231_INTS		(175)		/* default interrupt rate */
+#define	CS4231_MIN_INTS		(10)		/* minimum interrupt rate */
+#define	CS4231_MAX_INTS		(2000)		/* maximum interrupt rate */
+
+/*
+ * Implementation specific header file for the audiocs device driver.
+ */
+
+#ifdef _KERNEL
+
+enum {
+	CTL_VOLUME = 0,
+	CTL_IGAIN,
+	CTL_MGAIN,
+	CTL_INPUTS,
+	CTL_OUTPUTS,
+	CTL_MICBOOST,
+	CTL_NUM
+};
+
+typedef struct CS_engine CS_engine_t;
+typedef struct CS_ctrl CS_ctrl_t;
+typedef struct CS_state CS_state_t;
 
 /*
  * These are the registers for the APC DMA channel interface to the
@@ -56,7 +88,7 @@ void cs4231_put8(audiohdl_t ahandle, ddi_acc_handle_t, uint8_t *, uint8_t,
 struct cs4231_apc {
 	uint32_t 	dmacsr;		/* APC CSR */
 	uint32_t	lpad[3];	/* PAD */
-	uint32_t 	dmacva;		/* Captue Virtual Address */
+	uint32_t 	dmacva;		/* Capture Virtual Address */
 	uint32_t 	dmacc;		/* Capture Count */
 	uint32_t 	dmacnva;	/* Capture Next VAddress */
 	uint32_t 	dmacnc;		/* Capture next count */
@@ -67,7 +99,6 @@ struct cs4231_apc {
 };
 typedef struct cs4231_apc cs4231_apc_t;
 
-#define	APC_HANDLE	state->cs_handles.cs_codec_hndl
 #define	APC_DMACSR	state->cs_regs->apc.dmacsr
 #define	APC_DMACVA	state->cs_regs->apc.dmacva
 #define	APC_DMACC	state->cs_regs->apc.dmacc
@@ -140,10 +171,6 @@ struct cs4231_eb2regs {
 };
 typedef struct cs4231_eb2regs cs4231_eb2regs_t;
 
-#define	EB2_CODEC_HNDL	state->cs_handles.cs_codec_hndl
-#define	EB2_PLAY_HNDL	state->cs_handles.cs_eb2_play_hndl
-#define	EB2_REC_HNDL	state->cs_handles.cs_eb2_rec_hndl
-#define	EB2_AUXIO_HNDL	state->cs_handles.cs_eb2_auxio_hndl
 #define	EB2_PLAY_CSR	state->cs_eb2_regs.play->eb2csr
 #define	EB2_PLAY_ACR	state->cs_eb2_regs.play->eb2acr
 #define	EB2_PLAY_BCR	state->cs_eb2_regs.play->eb2bcr
@@ -189,16 +216,13 @@ typedef struct cs4231_eb2regs cs4231_eb2regs_t;
 #define	EB2_DEV_ID		0xf0000000u	/* Device ID -0x0C, R/O */
 
 #define	EB2_ID_VALUE		0xC0000000u	/* ID read from CSR */
-#define	EB2_PCLEAR_RESET_VALUE	(EB2_READ|EB2_EN_NEXT)
-#define	EB2_RCLEAR_RESET_VALUE	(EB2_WRITE|EB2_EN_NEXT)
+#define	EB2_PCLEAR_RESET_VALUE	(EB2_READ|EB2_EN_NEXT|EB2_EN_CNT)
+#define	EB2_RCLEAR_RESET_VALUE	(EB2_WRITE|EB2_EN_NEXT|EB2_EN_CNT)
 
-#define	EB2_PINTR_MASK		(EB2_INT_EN)
-#define	EB2_RINTR_MASK		(EB2_INT_EN)
-
-#define	EB2_PLAY_ENABLE		(EB2_PINTR_MASK|EB2_EN_DMA|EB2_EN_CNT|EB2_64|\
+#define	EB2_PLAY_ENABLE		(EB2_INT_EN|EB2_EN_DMA|EB2_EN_CNT|EB2_64|\
 					EB2_PCLEAR_RESET_VALUE)
 
-#define	EB2_REC_ENABLE		(EB2_RINTR_MASK|EB2_EN_DMA|EB2_EN_CNT|EB2_64|\
+#define	EB2_REC_ENABLE		(EB2_INT_EN|EB2_EN_DMA|EB2_EN_CNT|EB2_64|\
 					EB2_RCLEAR_RESET_VALUE)
 
 #define	EB2_FIFO_DRAIN		(EB2_DRAIN|EB2_CYC_PENDING)
@@ -206,22 +230,16 @@ typedef struct cs4231_eb2regs cs4231_eb2regs_t;
 /*
  * Misc. defines
  */
-#define	CS4231_IDNUM			(0x6175)
-#define	CS4231_MINPACKET		(0)
-#define	CS4231_MAXPACKET		(1*1024)
-#define	CS4231_HIWATER			(AM_MAX_QUEUED_MSGS_SIZE)
-#define	CS4231_LOWATER			(AM_MIN_QUEUED_MSGS_SIZE)
-#define	CS4231_MAX_CHANNELS		(200)		/* force max # chs */
 #define	CS4231_REGS			(32)
 #define	CS4231_NCOMPONENTS		(1)
 #define	CS4231_COMPONENT		(0)
 #define	CS4231_PWR_OFF			(0)
 #define	CS4231_PWR_ON			(1)
 #define	CS4231_TIMEOUT			(100000)
-#define	CS4231_NOT_SUSPENDED		(0)
-#define	CS4231_SUSPENDED		(~CS4231_NOT_SUSPENDED)
-#define	CS4231_BSIZE			(8*1024)
 #define	CS4231_300MS			(300*1000)
+#define	CS4231_PLAY			0
+#define	CS4231_REC			1
+#define	CS4231_NFRAGS			8
 
 /*
  * Supported dma engines and the ops vector
@@ -244,6 +262,7 @@ struct cs4231_pioregs {
 };
 typedef struct cs4231_pioregs cs4231_pioregs_t;
 
+
 struct cs4231_eb2 {
 	cs4231_eb2regs_t	*play;		/* play EB2 registers */
 	cs4231_eb2regs_t	*record;	/* record EB2 registers */
@@ -265,9 +284,6 @@ typedef struct cs4231_regs cs4231_regs_t;
 /*
  * Misc. state enumerations and structures
  */
-enum cs_cd {NO_INTERNAL_CD, INTERNAL_CD_ON_AUX1, INTERNAL_CD_ON_AUX2};
-typedef enum cs_cd cs_cd_e;
-
 struct cs4231_handle {
 	ddi_acc_handle_t	cs_codec_hndl;	/* CODEC handle, APC & EB2 */
 	ddi_acc_handle_t	cs_eb2_play_hndl; /* EB2 only, play handle */
@@ -275,6 +291,48 @@ struct cs4231_handle {
 	ddi_acc_handle_t	cs_eb2_auxio_hndl; /* EB2 only, auxio handle */
 };
 typedef struct cs4231_handle cs4231_handle_t;
+#define	CODEC_HANDLE	state->cs_handles.cs_codec_hndl
+#define	APC_HANDLE	state->cs_handles.cs_codec_hndl
+#define	EB2_PLAY_HNDL	state->cs_handles.cs_eb2_play_hndl
+#define	EB2_REC_HNDL	state->cs_handles.cs_eb2_rec_hndl
+#define	EB2_AUXIO_HNDL	state->cs_handles.cs_eb2_auxio_hndl
+
+/*
+ * CS_port_t - per port (playback or record) state
+ */
+struct CS_engine {
+	CS_state_t		*ce_state;
+	audio_engine_t		*ce_engine;
+	int			ce_num;
+	unsigned		ce_syncdir;
+	unsigned		ce_intrs;
+	unsigned		ce_fragfr;
+	unsigned		ce_fragsz;
+	unsigned		ce_nframes;
+	unsigned		ce_cfrag;
+	boolean_t		ce_started;
+	uint64_t		ce_count;
+
+	size_t			ce_size;
+	caddr_t			ce_kaddr;
+	ddi_dma_handle_t	ce_dmah;
+	ddi_acc_handle_t	ce_acch;
+	uint32_t		ce_paddr[CS4231_NFRAGS];
+
+	/* registers (EB2 only) */
+	ddi_acc_handle_t	ce_regsh;
+	cs4231_eb2regs_t	*ce_eb2regs;	/* EB2 registers */
+
+	/* codec enable */
+	uint8_t			ce_codec_en;
+};
+
+struct CS_ctrl {
+	CS_state_t		*cc_state;
+	audio_ctrl_t		*cc_ctrl;
+	uint32_t		cc_num;
+	uint64_t		cc_val;
+};
 
 /*
  * CS_state_t - per instance state and operation data
@@ -283,84 +341,39 @@ struct CS_state {
 	kmutex_t		cs_lock;	/* state protection lock */
 	kcondvar_t		cs_cv;		/* suspend/resume cond. var. */
 	ddi_iblock_cookie_t	cs_iblock;	/* iblock cookie */
-	uint_t			cs_flags;	/* flags */
 	kstat_t			*cs_ksp;	/* kernel statistics */
 	dev_info_t		*cs_dip;	/* used by cs4231_getinfo() */
-	audiohdl_t		cs_ahandle;	/* audio handle */
-	audio_info_t		cs_defaults;	/* default state for the dev */
-	am_ad_info_t		cs_ad_info;	/* audio device info state */
+	audio_dev_t		*cs_adev;	/* audio device state */
+
 	cs_dmae_types_e		cs_dma_engine;	/* dma engine for this h/w */
 	struct cs4231_dma_ops	*cs_dma_ops;	/* dma engine ops vector */
 	cs4231_regs_t		*cs_regs;	/* hardware registers */
 	cs4231_eb2_t		cs_eb2_regs;	/* eb2 DMA registers */
 	cs4231_handle_t		cs_handles;	/* hardware handles */
-	audio_device_t		cs_dev_info;	/* device info strings */
-	int			cs_instance;	/* device instance */
-	int			cs_suspended;	/* power management state */
-	int			cs_powered;	/* device powered up? */
-	int			cs_busy_cnt;	/* device busy count */
-	ddi_dma_attr_t		*cs_dma_attr;	/* dma attributes */
-	ddi_dma_handle_t	cs_ph[2];	/* play DMA handles */
-	ddi_dma_handle_t	cs_ch[2];	/* capture DMA handles */
-	ddi_dma_cookie_t	cs_pc[2];	/* play DMA cookies */
-	ddi_dma_cookie_t	cs_cc[2];	/* capture DMA cookies */
-	ddi_acc_handle_t	cs_pmh[2];	/* play DMA memory handles */
-	ddi_acc_handle_t	cs_cmh[2];	/* capture DMA memory handles */
-	size_t			cs_pml[2];	/* play DMA memory length */
-	size_t			cs_cml[2];	/* capture DMA memory length */
-	caddr_t			cs_pb[2];	/* play DMA buffers */
-	caddr_t			cs_cb[2];	/* capture DMA buffers */
-	int			cs_pcnt[2];	/* play count, in samples */
-	int			cs_ccnt[2];	/* capture count, in bytes */
-	int			cs_pbuf_toggle;	/* play DMA buffer toggle */
-	int			cs_cbuf_toggle;	/* capture DMA buffer toggle */
-	uint_t			cs_play_sr;	/* play sample rate */
-	uint_t			cs_play_ch;	/* play channels */
-	uint_t			cs_play_prec;	/* play precision */
-	uint_t			cs_play_enc;	/* play encoding */
-	uint_t			cs_record_sr;	/* record sample rate */
-	uint_t			cs_record_ch;	/* record channels */
-	uint_t			cs_record_prec;	/* record precision */
-	uint_t			cs_record_enc;	/* record encoding */
-	uint_t			cs_output_muted; /* output muted */
-	cs_cd_e			cs_cd_input_line; /* AUX1, AUX2 or none */
+
+	boolean_t		cs_suspended;	/* power management state */
+	boolean_t		cs_powered;	/* device powered up? */
+
+	CS_engine_t		*cs_engines[2];
+
 	boolean_t		cs_revA;	/* B_TRUE if Rev A CODEC */
-	boolean_t		cs_autocal;	/* auto calibrate if B_TRUE */
 	uint8_t			cs_save[CS4231_REGS];	/* PM reg. storage */
-	uint8_t			cs_save_ce;	/* save capture encoding */
-	uint8_t			cs_save_pe;	/* save play encoding */
+
+	/*
+	 * Control related fields.
+	 */
+	uint64_t		cs_imask;
+	uint64_t		cs_omask;
+	uint64_t		cs_omod;	/* modifiable ports */
+
+	CS_ctrl_t		*cs_ogain;
+	CS_ctrl_t		*cs_igain;
+	CS_ctrl_t		*cs_micboost;
+	CS_ctrl_t		*cs_mgain;
+	CS_ctrl_t		*cs_outputs;
+	CS_ctrl_t		*cs_inputs;
 };
-typedef struct CS_state CS_state_t;
 
-_NOTE(MUTEX_PROTECTS_DATA(CS_state::cs_lock, CS_state))
-_NOTE(READ_ONLY_DATA(CS_state::cs_dip))
-_NOTE(READ_ONLY_DATA(CS_state::cs_ahandle))
-_NOTE(READ_ONLY_DATA(CS_state::cs_instance))
-_NOTE(READ_ONLY_DATA(CS_state::cs_handles))
-_NOTE(SCHEME_PROTECTS_DATA("method", CS_state::cs_cb))
-_NOTE(SCHEME_PROTECTS_DATA("method", CS_state::cs_pb))
-_NOTE(SCHEME_PROTECTS_DATA("method", CS_state::cs_cbuf_toggle))
-_NOTE(SCHEME_PROTECTS_DATA("method", CS_state::cs_pbuf_toggle))
-
-/* CS_state.flags defines */
-#define	PDMA_ENGINE_INITIALIZED	0x0001u	/* play DMA engine initialized */
-#define	REC_INTR_PENDING	0x0002u	/* record interrupt pending */
-#define	APC_P_STOPPED		0x0004u	/* APC playing has been shutdown */
-#define	RDMA_ENGINE_INITIALIZED	0x0008u	/* record DMA engine initialized */
-#define	PLAY_ACTIVE		0x0010u	/* play active, for PM */
-
-#define	PLAY_DMA_HANDLE		state->cs_ph[state->cs_pbuf_toggle]
-#define	PLAY_DMA_COOKIE		state->cs_pc[state->cs_pbuf_toggle]
-#define	PLAY_DMA_BUF		state->cs_pb[state->cs_pbuf_toggle]
-#define	PLAY_DMA_MEM_HANDLE	state->cs_pmh[state->cs_pbuf_toggle]
-#define	PLAY_DMA_MEM_LENGTH	state->cs_pml[state->cs_pbuf_toggle]
-#define	PLAY_COUNT		state->cs_pcnt[state->cs_pbuf_toggle]
-#define	CAP_DMA_HANDLE		state->cs_ch[state->cs_cbuf_toggle]
-#define	CAP_DMA_COOKIE		state->cs_cc[state->cs_cbuf_toggle]
-#define	CAP_DMA_BUF		state->cs_cb[state->cs_cbuf_toggle]
-#define	CAP_DMA_MEM_HANDLE	state->cs_cmh[state->cs_cbuf_toggle]
-#define	CAP_DMA_MEM_LENGTH	state->cs_cml[state->cs_cbuf_toggle]
-#define	CAP_COUNT		state->cs_ccnt[state->cs_cbuf_toggle]
 #define	KIOP(X)			((kstat_intr_t *)(X->cs_ksp->ks_data))
 
 /*
@@ -368,17 +381,14 @@ _NOTE(SCHEME_PROTECTS_DATA("method", CS_state::cs_pbuf_toggle))
  */
 struct cs4231_dma_ops {
 	char	*dma_device;
-	int	(*cs_dma_map_regs)(dev_info_t *, CS_state_t *, size_t, size_t);
+	ddi_dma_attr_t	*cs_dma_attr;
+	int	(*cs_dma_map_regs)(CS_state_t *);
 	void	(*cs_dma_unmap_regs)(CS_state_t *);
 	void	(*cs_dma_reset)(CS_state_t *);
 	int	(*cs_dma_add_intr)(CS_state_t *);
-	void	(*cs_dma_rem_intr)(dev_info_t *);
-	int 	(*cs_dma_p_start)(CS_state_t *);
-	void	(*cs_dma_p_pause)(CS_state_t *);
-	void	(*cs_dma_p_restart)(CS_state_t *);
-	void	(*cs_dma_p_stop)(CS_state_t *);
-	int	(*cs_dma_r_start)(CS_state_t *);
-	void	(*cs_dma_r_stop)(CS_state_t *);
+	void	(*cs_dma_rem_intr)(CS_state_t *);
+	int 	(*cs_dma_start)(CS_engine_t *);
+	void	(*cs_dma_stop)(CS_engine_t *);
 	void	(*cs_dma_power)(CS_state_t *, int);
 };
 typedef struct cs4231_dma_ops cs4231_dma_ops_t;
@@ -386,42 +396,28 @@ typedef struct cs4231_dma_ops cs4231_dma_ops_t;
 extern cs4231_dma_ops_t cs4231_apcdma_ops;
 extern cs4231_dma_ops_t cs4231_eb2dma_ops;
 
-#define	CS4231_DMA_MAP_REGS(DIP, S, P, C)	\
-				((S)->cs_dma_ops->cs_dma_map_regs)(DIP, S, P, C)
+#define	CS4231_DMA_MAP_REGS(S)		((S)->cs_dma_ops->cs_dma_map_regs)(S)
 #define	CS4231_DMA_UNMAP_REGS(S)	((S)->cs_dma_ops->cs_dma_unmap_regs)(S)
 #define	CS4231_DMA_RESET(S)		((S)->cs_dma_ops->cs_dma_reset)(S)
 #define	CS4231_DMA_ADD_INTR(S)		((S)->cs_dma_ops->cs_dma_add_intr)(S)
-#define	CS4231_DMA_REM_INTR(DIP, S)	((S)->cs_dma_ops->cs_dma_rem_intr)(DIP)
-#define	CS4231_DMA_START_PLAY(S)	((S)->cs_dma_ops->cs_dma_p_start)(S)
-#define	CS4231_DMA_PAUSE_PLAY(S)	((S)->cs_dma_ops->cs_dma_p_pause)(S)
-#define	CS4231_DMA_RESTART_PLAY(S)	((S)->cs_dma_ops->cs_dma_p_restart)(S)
-#define	CS4231_DMA_STOP_PLAY(S)		((S)->cs_dma_ops->cs_dma_p_stop)(S)
-#define	CS4231_DMA_START_RECORD(S)	((S)->cs_dma_ops->cs_dma_r_start)(S)
-#define	CS4231_DMA_STOP_RECORD(S)	((S)->cs_dma_ops->cs_dma_r_stop)(S)
+#define	CS4231_DMA_REM_INTR(S)		((S)->cs_dma_ops->cs_dma_rem_intr)(S)
+#define	CS4231_DMA_START(S, E)		((S)->cs_dma_ops->cs_dma_start)(E)
+#define	CS4231_DMA_STOP(S, E)		((S)->cs_dma_ops->cs_dma_stop)(E)
 #define	CS4231_DMA_POWER(S, L)		((S)->cs_dma_ops->cs_dma_power)(S, L)
+#define	CS4231_DMA_ATTR(S)		((S)->cs_dma_ops->cs_dma_attr)
 
 /*
  * Useful bit twiddlers
  */
 #define	CS4231_RETRIES		10
 
-#define	OR_SET_BYTE(handle, addr, val, mask)				\
-	cs4231_put8((state->cs_ahandle), (handle), (uint8_t *)(addr),	\
-	    (ddi_get8((handle), (uint8_t *)(addr)) | (uint8_t)(val)),	\
-	    (uint8_t)(mask), (__LINE__), (thisfile));
-
 #define	OR_SET_WORD(handle, addr, val)					\
 	ddi_put32((handle), (uint_t *)(addr),				\
-		(ddi_get32((handle), (uint_t *)(addr)) | (uint_t)(val)));
-
-#define	AND_SET_BYTE(handle, addr, val, mask)				\
-	cs4231_put8((state->cs_ahandle), (handle), (uint8_t *)(addr),	\
-	    (ddi_get8((handle), (uint8_t *)(addr)) & (uint8_t)(val)),	\
-	    (uint8_t)(mask), (__LINE__), (thisfile));
+		(ddi_get32((handle), (uint_t *)(addr)) | (uint_t)(val)))
 
 #define	AND_SET_WORD(handle, addr, val)					\
 	ddi_put32((handle), (uint_t *)(addr),				\
-		(ddi_get32((handle), (uint_t *)(addr)) & (uint_t)(val)));
+		(ddi_get32((handle), (uint_t *)(addr)) & (uint_t)(val)))
 
 /*
  * CS4231 Register Set Definitions
@@ -532,6 +528,11 @@ extern cs4231_dma_ops_t cs4231_eb2dma_ops;
 #define	PDF_ADPCM4		0xa0	/* ADPCM, 4-bit, IMA compatible */
 #define	PDF_LINEAR16BE		0xc0	/* Linear, 16-bit signed, big endian */
 #define	FSDF_VALID_MASK		0xff	/* Valid bits mask */
+#ifdef	_BIG_ENDIAN
+#define	PDF_LINEAR16NE		PDF_LINEAR16BE
+#else
+#define	PDF_LINEAR16NE		PDF_LINEAR16LE
+#endif
 
 /* Index 09 - Interface Configuration, Mode 1&2 */
 #define	INTC_REG		0x09	/* Interrupt Configuration Register */
@@ -624,7 +625,7 @@ extern cs4231_dma_ops_t cs4231_eb2dma_ops;
 /* Index 22 and 23 are reserved */
 
 /* Index 24 - Alternate Feature Status, Mode 2 only */
-#define	AFS_REG			0x18	/* Alternate Feature Status Registure */
+#define	AFS_REG			0x18	/* Alternate Feature Status Register */
 #define	AFS_PU			0x01	/* Playback Underrun */
 #define	AFS_PO			0x02	/* Playback Overrun */
 #define	AFS_CO			0x04	/* Capture Overrun */
@@ -664,6 +665,11 @@ extern cs4231_dma_ops_t cs4231_eb2dma_ops;
 #define	CDF_ADPCM4		0xa0	/* ADPCM, 4-bit, IMA compatible */
 #define	CDF_LINEAR16BE		0xc0	/* Linear, 16-bit signed, big endian */
 #define	CDF_VALID_MASK		0xf0	/* Valid bits mask */
+#ifdef	_BIG_ENDIAN
+#define	CDF_LINEAR16NE		CDF_LINEAR16BE
+#else
+#define	CDF_LINEAR16NE		CDF_LINEAR16LE
+#endif
 
 /* Index 29 is reserved */
 
@@ -677,8 +683,8 @@ extern cs4231_dma_ops_t cs4231_eb2dma_ops;
 
 #endif	/* _KERNEL */
 
-#ifdef	__cplusplus
+#ifdef __cplusplus
 }
 #endif
 
-#endif	/* _SYS_AUDIO4231_IMPL_H */
+#endif	/* _AUDIO_4231_H */
diff --git a/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_apcdma.c b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_apcdma.c
new file mode 100644
index 0000000000..64b1cbe12a
--- /dev/null
+++ b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_apcdma.c
@@ -0,0 +1,567 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Platform specifc code for the APC DMA controller. The APC is an SBus
+ * IC that includes play and record DMA engines and an interface for
+ * the CS4231.
+ */
+
+#include <sys/systm.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include "audio_4231.h"
+
+/*
+ * Attribute structure for the APC, used to create DMA handles.
+ */
+static ddi_dma_attr_t apc_dma_attr = {
+	DMA_ATTR_V0,			/* version */
+	0x0000000000000000LL,		/* dlim_addr_lo */
+	0x00000000ffffffffLL,		/* dlim_addr_hi */
+	0x0000000000000fffLL,		/* DMA counter register */
+	0x0000000000000001LL,		/* DMA address alignment */
+	0x00000014,			/* 4 and 16 byte burst sizes */
+	0x00000001,			/* min effective DMA size */
+	0x0000000000000fffLL,		/* maximum transfer size, 8k */
+	0x000000000000ffffLL,		/* segment boundary, 32k */
+	0x00000001,			/* s/g list length, no s/g */
+	0x00000001,			/* granularity of device, don't care */
+	0				/* DMA flags */
+};
+
+static ddi_device_acc_attr_t acc_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_BE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * Local routines
+ */
+static uint_t apc_intr(caddr_t);
+static void apc_load_fragment(CS_engine_t *);
+
+/*
+ * DMA ops vector functions
+ */
+static int apc_map_regs(CS_state_t *);
+static void apc_unmap_regs(CS_state_t *);
+static void apc_reset(CS_state_t *);
+static int apc_add_intr(CS_state_t *);
+static void apc_rem_intr(CS_state_t *);
+static int apc_start_engine(CS_engine_t *);
+static void apc_stop_engine(CS_engine_t *);
+static void apc_power(CS_state_t *, int);
+
+cs4231_dma_ops_t cs4231_apcdma_ops = {
+	"APC DMA controller",
+	&apc_dma_attr,
+	apc_map_regs,
+	apc_unmap_regs,
+	apc_reset,
+	apc_add_intr,
+	apc_rem_intr,
+	apc_start_engine,
+	apc_stop_engine,
+	apc_power,
+};
+
+/*
+ * apc_map_regs()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use. It then binds each of the buffers to its
+ *	respective handle, getting a DMA cookie. Finally, the registers
+ *	are mapped in.
+ *
+ *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
+ *		memory. This means these calls will almost always succeed.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state structure
+ *
+ * Returns:
+ *	AUDIO_SUCCESS		Registers successfully mapped
+ *	AUDIO_FAILURE		Registers not successfully mapped
+ */
+static int
+apc_map_regs(CS_state_t *state)
+{
+	ddi_acc_handle_t	*handle = &APC_HANDLE;
+	dev_info_t		*dip = state->cs_dip;
+
+	/* map in the registers, getting a handle */
+	if (ddi_regs_map_setup(dip, 0, (caddr_t *)&state->cs_regs, 0,
+	    sizeof (cs4231_regs_t), &acc_attr, handle) != DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "ddi_regs_map_setup() failed");
+		return (DDI_FAILURE);
+	}
+
+	/* clear the CSR so we have all interrupts disabled */
+	ddi_put32(*handle, &APC_DMACSR, APC_CLEAR_RESET_VALUE);
+
+	return (DDI_SUCCESS);
+}	/* apc_map_regs() */
+
+/*
+ * apc_unmap_regs()
+ *
+ * Description:
+ *	This routine unmaps the Codec's and DMA engine's registers.
+ *	It must be idempotent.
+ *
+ * Arguments:
+ *	CS_state_t	*state	The device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+apc_unmap_regs(CS_state_t *state)
+{
+	if (APC_HANDLE)
+		ddi_regs_map_free(&APC_HANDLE);
+
+}	/* apc_unmap_regs() */
+
+/*
+ * apc_reset()
+ *
+ * Description:
+ *	Reset both the play and record DMA engines. The engines are left
+ *	with interrupts and the DMA engine disabled.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo structure
+ *	CS_state_t	*state	The device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+apc_reset(CS_state_t *state)
+{
+	ddi_acc_handle_t	handle = APC_HANDLE;
+
+	/*
+	 * The APC has a bug where the reset is not done
+	 * until you do the next pio to the APC. This
+	 * next write to the CSR causes the posted reset to
+	 * happen.
+	 */
+
+	ddi_put32(handle, &APC_DMACSR, APC_RESET);
+	ddi_put32(handle, &APC_DMACSR, APC_CLEAR_RESET_VALUE);
+
+}	/* apc_reset() */
+
+/*
+ * apc_add_intr()
+ *
+ * Description:
+ *	Register the APC interrupts with the kernel.
+ *
+ *	NOTE: This does NOT turn on interrupts.
+ *
+ *	CAUTION: While the interrupts are added, the Codec interrupts are
+ *		not enabled.
+ *
+ * Arguments:
+ *	CS_state_t	*state	Pointer to the device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+apc_add_intr(CS_state_t *state)
+{
+	dev_info_t	*dip = state->cs_dip;
+
+	/* first we make sure this isn't a high level interrupt */
+	if (ddi_intr_hilevel(dip, 0) != 0) {
+		audio_dev_warn(state->cs_adev,
+		    "unsupported high level interrupt");
+		return (DDI_FAILURE);
+	}
+
+	/* okay to register the interrupt */
+	if (ddi_add_intr(dip, 0, NULL, NULL, apc_intr, (caddr_t)state) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "bad interrupt specification");
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+
+}	/* apc_add_intr() */
+
+/*
+ * apc_rem_intr()
+ *
+ * Description:
+ *	Unregister the APC interrupts from the kernel.
+ *
+ *	CAUTION: While the interrupts are removed, the Codec interrupts are
+ *		not disabled, but then, they never should have been on in
+ *		the first place.
+ *
+ * Arguments:
+ *	CS_state_t	*state	Pointer to the device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+apc_rem_intr(CS_state_t *state)
+{
+	ddi_remove_intr(state->cs_dip, 0, NULL);
+}	/* apc_rem_intr() */
+
+/*
+ * apc_start_engine()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ *	For hard starts the DMA engine is started by programming the
+ *	Next Virtual Address and then the Next Counter twice, and
+ *	finally enabling the DMA engine.
+ *
+ *	NOTE: The state structure must be locked before this routine is called.
+ *
+ *	CAUTION: ?!? This routine doesn't start the Codec because the first
+ *		interrupt causes a recursive mutex_enter.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng	The engine to start
+ *
+ * Returns:
+ *	DDI_SUCCESS		The DMA engine was started
+ *	DDI_FAILURE		The DMA engine was not started
+ */
+static int
+apc_start_engine(CS_engine_t *eng)
+{
+	CS_state_t		*state = eng->ce_state;
+	ddi_acc_handle_t	handle = APC_HANDLE;
+	uint32_t		csr;
+	uint32_t		enable;
+	uint32_t		dirty;
+	int			x;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	if (eng->ce_num == CS4231_PLAY) {
+		enable = APC_PLAY_ENABLE;
+		dirty = APC_PD;
+	} else {
+		enable = APC_CAP_ENABLE;
+		dirty = APC_CD;
+	}
+
+	/* make sure it's okay to program the Next Address/Count registers */
+	csr = ddi_get32(handle, &APC_DMACSR);
+	for (x = 0; !(csr & dirty) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);	/* no reason to beat on the bus */
+		csr = ddi_get32(handle, &APC_DMACSR);
+	}
+	if (x >= CS4231_TIMEOUT) {
+		audio_dev_warn(state->cs_adev,
+		    "timeout waiting for engine, not started!");
+		return (DDI_FAILURE);
+	}
+
+	/*
+	 * Program the first fragment.
+	 */
+	apc_load_fragment(eng);
+
+	/*
+	 * Start the DMA engine, including interrupts.
+	 */
+	OR_SET_WORD(handle, &APC_DMACSR, enable);
+
+	/*
+	 * Program the second fragment.
+	 */
+	apc_load_fragment(eng);
+
+	return (DDI_SUCCESS);
+}
+
+/*
+ * apc_stop_engine()
+ *
+ * Description:
+ *	This routine stops the engine.
+ *
+ *	The DMA engine is stopped by using the CAP_ABORT bit.
+ *
+ *	NOTE: The state structure must be locked before this routine is called.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng	The engine to sotp
+ *
+ * Returns:
+ *	void
+ */
+static void
+apc_stop_engine(CS_engine_t *eng)
+{
+	CS_state_t		*state = eng->ce_state;
+	ddi_acc_handle_t	handle = APC_HANDLE;
+	uint32_t		reg;
+	uint32_t		intren;
+	uint32_t		abort;
+	uint32_t		drainbit;
+	uint32_t		disable;
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	if (eng->ce_num == CS4231_PLAY) {
+		intren = APC_PINTR_ENABLE;
+		abort = APC_P_ABORT;
+		drainbit = APC_PM;
+		disable = APC_PLAY_DISABLE;
+	} else {
+		intren = APC_CINTR_ENABLE;
+		abort = APC_C_ABORT;
+		drainbit = APC_CX;
+		disable = APC_CAP_DISABLE;
+	}
+
+	/* clear the interrupts so the ISR doesn't get involved */
+	AND_SET_WORD(handle, &APC_DMACSR, ~intren);
+
+	/* first, abort the DMA engine */
+	OR_SET_WORD(handle, &APC_DMACSR, abort);
+
+	/* wait for the pipeline to empty */
+	reg = ddi_get32(handle, &APC_DMACSR);
+	for (int x = 0; (!(reg & drainbit)) && (x < CS4231_TIMEOUT); x++) {
+		drv_usecwait(1);	/* don't beat on bus */
+		reg = ddi_get32(handle, &APC_DMACSR);
+	}
+
+	/* now clear the enable and abort bits */
+	AND_SET_WORD(handle, &APC_DMACSR, ~(abort|disable));
+}
+
+
+/*
+ * apc_power()
+ *
+ * Description:
+ *	This routine turns the Codec off by using the COD_PDWN bit in the
+ *	apc chip. To turn power on we have to reset the APC, which clears
+ *	the COD_PDWN bit. However, this is a settling bug in the APC which
+ *	requires the driver to delay quite a while before we may continue.
+ *	Since this is the first time this feature has actually been used
+ *	it isn't too surprising that it has some problems.
+ *
+ *	NOTE: The state structure must be locked when this routine is called.
+ *
+ * Arguments:
+ *	CS_state_t	*state		Ptr to the device's state structure
+ *	int		level		Power level to set
+ *
+ * Returns:
+ *	void
+ */
+static void
+apc_power(CS_state_t *state, int level)
+{
+	ddi_acc_handle_t	handle = APC_HANDLE;
+
+	if (level == CS4231_PWR_ON) {	/* turn power on */
+		AND_SET_WORD(handle, &APC_DMACSR, ~APC_COD_PDWN);
+		OR_SET_WORD(handle, &APC_DMACSR, APC_RESET);
+		AND_SET_WORD(handle, &APC_DMACSR, ~APC_RESET);
+
+		/*
+		 * wait for state change,
+		 */
+		delay(drv_usectohz(CS4231_300MS));
+	} else {	/* turn power off */
+		ASSERT(level == CS4231_PWR_OFF);
+		OR_SET_WORD(handle, &APC_DMACSR, APC_COD_PDWN);
+	}
+
+}	/* apc_power() */
+
+
+/* *******  Local Routines ************************************************** */
+
+/*
+ * apc_intr()
+ *
+ * Description:
+ *	APC interrupt service routine, which services both play and capture
+ *	interrupts. First we find out why there was an interrupt, then we
+ *	take the appropriate action.
+ *
+ *	Because this ISR deals with both play and record interrupts we have
+ *	to be careful to not lose an interrupt. So we service the record
+ *	interrupt first and save the incoming data until later. This is all
+ *	done without releasing the lock, thus there can be no race conditions.
+ *	Then we process the play interrupt. While processing the play interrupt
+ *	we have to release the lock. When this happens we send recorded data
+ *	to the mixer and then get the next chunk of data to play. If there
+ *	wasn't a play interrupt then we finish by sending the recorded data,
+ *	if any.
+ *
+ * Arguments:
+ *	caddr_t		T	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+apc_intr(caddr_t T)
+{
+	CS_state_t		*state = (void *)T;
+	ddi_acc_handle_t	handle = APC_HANDLE;
+	uint_t			csr;
+	int			rc = DDI_INTR_UNCLAIMED;
+
+	/* the state must be protected */
+	mutex_enter(&state->cs_lock);
+
+	if (state->cs_suspended) {
+		mutex_exit(&state->cs_lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	/* get the APC CSR */
+	csr = ddi_get32(handle, &APC_DMACSR);
+
+	/* make sure this device sent the interrupt */
+	if (!(csr & APC_IP)) {
+		if (csr & APC_PMI_EN) {
+			/*
+			 * Clear device generated interrupt while play is
+			 * active (Only seen while playing and insane mode
+			 * switching)
+			 */
+			mutex_exit(&state->cs_lock);
+			return (DDI_INTR_CLAIMED);
+		} else {
+			/* nope, this isn't our interrupt */
+			mutex_exit(&state->cs_lock);
+			return (DDI_INTR_UNCLAIMED);
+		}
+	}
+
+	/* clear all interrupts we captured this time */
+	ddi_put32(handle, &APC_DMACSR, csr);
+
+	if (csr & APC_CINTR_MASK) {
+		/* try to load the next record buffer */
+		apc_load_fragment(state->cs_engines[CS4231_REC]);
+		rc = DDI_INTR_CLAIMED;
+	}
+
+	if (csr & APC_PINTR_MASK) {
+		/* try to load the next play buffer */
+		apc_load_fragment(state->cs_engines[CS4231_PLAY]);
+		rc = DDI_INTR_CLAIMED;
+	}
+
+done:
+
+	/* APC error interrupt, not sure what to do here */
+	if (csr & APC_EI) {
+		audio_dev_warn(state->cs_adev, "error interrupt: 0x%x", csr);
+		rc = DDI_INTR_CLAIMED;
+	}
+
+	/* update the kernel interrupt statistics */
+	if (state->cs_ksp) {
+		if (rc == DDI_INTR_CLAIMED) {
+			KIOP(state)->intrs[KSTAT_INTR_HARD]++;
+		}
+	}
+
+	mutex_exit(&state->cs_lock);
+
+	if (csr & APC_CINTR_MASK) {
+		CS_engine_t	*eng = state->cs_engines[CS4231_REC];
+		if (eng->ce_started)
+			audio_engine_produce(eng->ce_engine);
+	}
+	if (csr & APC_PINTR_MASK) {
+		CS_engine_t	*eng = state->cs_engines[CS4231_PLAY];
+		if (eng->ce_started)
+			audio_engine_consume(eng->ce_engine);
+	}
+
+	return (rc);
+
+}	/* apc_intr() */
+
+static void
+apc_load_fragment(CS_engine_t *eng)
+{
+	CS_state_t		*state = eng->ce_state;
+	ddi_acc_handle_t	handle = APC_HANDLE;
+	uint32_t		dirty;
+	uint32_t		*nva;	/* next VA reg */
+	uint32_t		*nc;	/* next count reg */
+
+	if (eng->ce_num == CS4231_PLAY) {
+		dirty = APC_PD;
+		nva = &APC_DMAPNVA;
+		nc = &APC_DMAPNC;
+	} else {
+		dirty = APC_CD;
+		nva = &APC_DMACNVA;
+		nc = &APC_DMACNC;
+	}
+
+	/* if we can't load another address, then don't */
+	if ((ddi_get32(handle, &APC_DMACSR) & dirty) == 0) {
+		return;
+	}
+
+	/* read the NVA, as per APC document */
+	(void) ddi_get32(handle, nva);
+
+	/* write the address of the next fragment */
+	ddi_put32(handle, nva, eng->ce_paddr[eng->ce_cfrag]);
+
+	/* now program the NC reg., which enables the state machine */
+	ddi_put32(handle, nc, eng->ce_fragsz);
+
+	eng->ce_cfrag++;
+	eng->ce_cfrag %= CS4231_NFRAGS;
+	eng->ce_count += eng->ce_fragfr;
+}
diff --git a/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_eb2dma.c b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_eb2dma.c
new file mode 100644
index 0000000000..4c81bdd51b
--- /dev/null
+++ b/usr/src/uts/sun/io/audio/drv/audiocs/audio_4231_eb2dma.c
@@ -0,0 +1,573 @@
+/*
+ * CDDL HEADER START
+ *
+ * The contents of this file are subject to the terms of the
+ * Common Development and Distribution License (the "License").
+ * You may not use this file except in compliance with the License.
+ *
+ * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
+ * or http://www.opensolaris.org/os/licensing.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ *
+ * When distributing Covered Code, include this CDDL HEADER in each
+ * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
+ * If applicable, add the following below this CDDL HEADER, with the
+ * fields enclosed by brackets "[]" replaced with your own identifying
+ * information: Portions Copyright [yyyy] [name of copyright owner]
+ *
+ * CDDL HEADER END
+ */
+/*
+ * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
+ * Use is subject to license terms.
+ */
+
+/*
+ * Platform specifc code for the EB2 DMA controller. The EB2 is a PCI bus
+ * IC that includes play and record DMA engines and an interface for
+ * the CS4231.
+ */
+
+#include <sys/systm.h>
+#include <sys/ddi.h>
+#include <sys/sunddi.h>
+#include <sys/note.h>
+#include <sys/audio/audio_driver.h>
+#include "audio_4231.h"
+
+/*
+ * Attribute structure for the APC, used to create DMA handles.
+ */
+static ddi_dma_attr_t eb2_dma_attr = {
+	DMA_ATTR_V0,			/* version */
+	0x0000000000000000LL,		/* dlim_addr_lo */
+	0x00000000ffffffffLL,		/* dlim_addr_hi */
+	0x0000000000ffffffLL,		/* DMA counter register */
+	0x0000000000000001LL,		/* DMA address alignment */
+	0x00000074,			/* 4 and 16 byte burst sizes */
+	0x00000001,			/* min effective DMA size */
+	0x000000000000ffffLL,		/* maximum transfer size, 8k */
+	0x000000000000ffffLL,		/* segment boundary, 32k */
+	0x00000001,			/* s/g list length, no s/g */
+	0x00000001,			/* granularity of device, don't care */
+	0				/* DMA flags */
+};
+
+static ddi_device_acc_attr_t codec_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_BE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+static ddi_device_acc_attr_t eb2_attr = {
+	DDI_DEVICE_ATTR_V0,
+	DDI_STRUCTURE_LE_ACC,
+	DDI_STRICTORDER_ACC
+};
+
+/*
+ * Local routines
+ */
+static uint_t eb2_intr(caddr_t);
+static void eb2_load_fragment(CS_engine_t *);
+
+/*
+ * DMA ops vector functions
+ */
+static int eb2_map_regs(CS_state_t *);
+static void eb2_unmap_regs(CS_state_t *);
+static void eb2_reset(CS_state_t *);
+static int eb2_add_intr(CS_state_t *);
+static void eb2_rem_intr(CS_state_t *);
+static int eb2_start_engine(CS_engine_t *);
+static void eb2_stop_engine(CS_engine_t *);
+static void eb2_power(CS_state_t *, int);
+
+cs4231_dma_ops_t cs4231_eb2dma_ops = {
+	"EB2 DMA controller",
+	&eb2_dma_attr,
+	eb2_map_regs,
+	eb2_unmap_regs,
+	eb2_reset,
+	eb2_add_intr,
+	eb2_rem_intr,
+	eb2_start_engine,
+	eb2_stop_engine,
+	eb2_power,
+};
+
+/*
+ * eb2_map_regs()
+ *
+ * Description:
+ *	This routine allocates the DMA handles and the memory for the
+ *	DMA engines to use. It then binds each of the buffers to its
+ *	respective handle, getting a DMA cookie. Finally, the registers
+ *	are mapped in.
+ *
+ *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
+ *		memory. This means these calls will almost always succeed.
+ *
+ * Arguments:
+ *	CS_state_t	*state		The device's state
+ *
+ * Returns:
+ *	DDI_SUCCESS		Registers successfully mapped
+ *	DDI_FAILURE		Registers not successfully mapped
+ */
+static int
+eb2_map_regs(CS_state_t *state)
+{
+	dev_info_t	*dip = state->cs_dip;
+
+	/* now, map the codec */
+	if (ddi_regs_map_setup(dip, 0, (caddr_t *)&state->cs_regs, 0,
+	    sizeof (cs4231_pioregs_t), &codec_attr, &CODEC_HANDLE) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "failed mapping codec regs");
+		goto error;
+	}
+
+	/* next the play registers */
+	if (ddi_regs_map_setup(dip, 1, (caddr_t *)&state->cs_eb2_regs.play, 0,
+	    sizeof (cs4231_eb2regs_t), &eb2_attr, &EB2_PLAY_HNDL) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "failed mapping play regs");
+		goto error;
+	}
+	state->cs_engines[CS4231_PLAY]->ce_regsh = EB2_PLAY_HNDL;
+	state->cs_engines[CS4231_PLAY]->ce_eb2regs = state->cs_eb2_regs.play;
+
+	/* now the capture registers */
+	if (ddi_regs_map_setup(dip, 2, (caddr_t *)&state->cs_eb2_regs.record, 0,
+	    sizeof (cs4231_eb2regs_t), &eb2_attr, &EB2_REC_HNDL) !=
+	    DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "failed mapping rec regs");
+		goto error;
+	}
+	state->cs_engines[CS4231_REC]->ce_regsh = EB2_REC_HNDL;
+	state->cs_engines[CS4231_REC]->ce_eb2regs = state->cs_eb2_regs.record;
+
+	/* finally the auxio register */
+	if (ddi_regs_map_setup(dip, 3, (caddr_t *)&state->cs_eb2_regs.auxio, 0,
+	    sizeof (uint_t), &eb2_attr, &EB2_AUXIO_HNDL) != DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "failed mapping auxio reg");
+		goto error;
+	}
+
+	/* disable play and record interrupts */
+	ddi_put32(EB2_PLAY_HNDL, &EB2_PLAY_CSR, EB2_PCLEAR_RESET_VALUE);
+	ddi_put32(EB2_REC_HNDL, &EB2_REC_CSR, EB2_RCLEAR_RESET_VALUE);
+
+	return (DDI_SUCCESS);
+
+error:
+	eb2_unmap_regs(state);
+	return (DDI_FAILURE);
+
+}	/* eb2_map_regs() */
+
+/*
+ * eb2_unmap_regs()
+ *
+ * Description:
+ *	This routine unmaps the Codec's and DMA engine's registers.
+ *	It must be idempotent.
+ *
+ * Arguments:
+ *	CS_state_t	*state	The device's state
+ *
+ * Returns:
+ *	void
+ */
+static void
+eb2_unmap_regs(CS_state_t *state)
+{
+	if (CODEC_HANDLE)
+		ddi_regs_map_free(&CODEC_HANDLE);
+	if (EB2_PLAY_HNDL)
+		ddi_regs_map_free(&EB2_PLAY_HNDL);
+	if (EB2_REC_HNDL)
+		ddi_regs_map_free(&EB2_REC_HNDL);
+	if (EB2_AUXIO_HNDL)
+		ddi_regs_map_free(&EB2_AUXIO_HNDL);
+
+}	/* eb2_unmap_regs() */
+
+/*
+ * eb2_reset()
+ *
+ * Description:
+ *	Reset both the play and record DMA engines. The engines are left
+ *	with interrupts and the DMA engine disabled.
+ *
+ * Arguments:
+ *	dev_info_t	*dip	Pointer to the device's devinfo structure
+ *	CS_state_t	*state	The device's state structure
+ *
+ * Returns:
+ *	void
+ */
+static void
+eb2_reset(CS_state_t *state)
+{
+	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
+	ddi_acc_handle_t	rhandle = EB2_REC_HNDL;
+	uint_t			reg;
+	int			x;
+
+	/* start with the play side */
+	ddi_put32(phandle, &EB2_PLAY_CSR, EB2_RESET);
+	/* wait for play data to drain */
+	reg = ddi_get32(phandle, &EB2_PLAY_CSR);
+	for (x = 0; (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);	/* don't beat on the bus */
+		reg = ddi_get32(phandle, &EB2_PLAY_CSR);
+	}
+	/* clear the reset bit and program for chaining */
+	ddi_put32(phandle, &EB2_PLAY_CSR, EB2_PCLEAR_RESET_VALUE);
+
+	/* now do the record side and program for chaining */
+	ddi_put32(rhandle, &EB2_REC_CSR, EB2_RESET);
+	/* wait for record data to drain */
+	reg = ddi_get32(rhandle, &EB2_REC_CSR);
+	for (x = 0; (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);	/* don't beat on the bus */
+		reg = ddi_get32(rhandle, &EB2_REC_CSR);
+	}
+	/* clear the reset bit */
+	ddi_put32(rhandle, &EB2_REC_CSR, EB2_RCLEAR_RESET_VALUE);
+
+}	/* eb2_reset() */
+
+/*
+ * eb2_add_intr()
+ *
+ * Description:
+ *	Register the EB2 interrupts with the kernel.
+ *
+ *	NOTE: This does NOT turn on interrupts.
+ *
+ *	CAUTION: While the interrupts are added, the Codec interrupts are
+ *		not enabled.
+ *
+ * Arguments:
+ *	CS_state_t	*state	Pointer to the device's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		Interrupts added
+ *	DDI_FAILURE		Interrupts not added
+ */
+static int
+eb2_add_intr(CS_state_t *state)
+{
+	dev_info_t	*dip = state->cs_dip;
+
+	/* first we make sure these aren't high level interrupts */
+	if (ddi_intr_hilevel(dip, 0) != 0) {
+		audio_dev_warn(state->cs_adev, "unsupported hi level intr 0");
+		return (DDI_FAILURE);
+	}
+	if (ddi_intr_hilevel(dip, 1) != 0) {
+		audio_dev_warn(state->cs_adev, "unsupported hi level intr 1");
+		return (DDI_FAILURE);
+	}
+
+	/* okay to register the interrupts */
+	if (ddi_add_intr(dip, 0, NULL, NULL, eb2_intr,
+	    (caddr_t)state->cs_engines[CS4231_REC]) != DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "bad record interrupt spec");
+		return (DDI_FAILURE);
+	}
+
+	if (ddi_add_intr(dip, 1, NULL, NULL, eb2_intr,
+	    (caddr_t)state->cs_engines[CS4231_PLAY]) != DDI_SUCCESS) {
+		audio_dev_warn(state->cs_adev, "play interrupt spec");
+		ddi_remove_intr(dip, 0, NULL);
+		return (DDI_FAILURE);
+	}
+
+	return (DDI_SUCCESS);
+
+}	/* eb2_add_intr() */
+
+/*
+ * eb2_rem_intr()
+ *
+ * Description:
+ *	Unregister the EB2 interrupts from the kernel.
+ *
+ *	CAUTION: While the interrupts are removed, the Codec interrupts are
+ *		not disabled, but then, they never should have been on in
+ *		the first place.
+ *
+ * Arguments:
+ *	CS_state_t	*state	Pointer to the device's soft state
+ *
+ * Returns:
+ *	void
+ */
+static void
+eb2_rem_intr(CS_state_t *state)
+{
+	ddi_remove_intr(state->cs_dip, 0, NULL);
+	ddi_remove_intr(state->cs_dip, 1, NULL);
+}	/* eb2_rem_intr() */
+
+/*
+ * eb2_start_engine()
+ *
+ * Description:
+ *	This routine starts the DMA engine.
+ *
+ *	NOTE: The state structure must be locked before this routine is called.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng	The DMA engine's state structure
+ *
+ * Returns:
+ *	DDI_SUCCESS		The DMA engine was started
+ *	DDI_FAILURE		The DMA engine was not started
+ */
+static int
+eb2_start_engine(CS_engine_t *eng)
+{
+	CS_state_t		*state = eng->ce_state;
+	ddi_acc_handle_t	handle = eng->ce_regsh;
+	cs4231_eb2regs_t	*regs = eng->ce_eb2regs;
+	uint_t			csr;
+	int			x;
+	uint32_t		reset;
+	uint32_t		enable;
+
+	if (eng->ce_num == CS4231_PLAY) {
+		reset = EB2_PCLEAR_RESET_VALUE;
+		enable = EB2_PLAY_ENABLE;
+	} else {
+		reset = EB2_RCLEAR_RESET_VALUE;
+		enable = EB2_REC_ENABLE;
+	}
+
+	ASSERT(mutex_owned(&state->cs_lock));
+
+	/* reset the DMA engine so we have a good starting place */
+	OR_SET_WORD(handle, &regs->eb2csr, EB2_RESET);
+
+	/* wait for the FIFO to drain, it should be empty */
+	csr = ddi_get32(handle, &regs->eb2csr);
+	for (x = 0; (csr & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);	/* no reason to beat on the bus */
+		csr = ddi_get32(handle, &regs->eb2csr);
+	}
+	if (x >= CS4231_TIMEOUT) {
+		audio_dev_warn(state->cs_adev,
+		    "timeout waiting for engine, not started!");
+		return (DDI_FAILURE);
+	}
+
+	/* now clear the RESET and EN_DMA bits */
+	AND_SET_WORD(handle, &regs->eb2csr, ~(EB2_RESET|EB2_EN_DMA));
+
+	/* put into chaining mode, enable byte counts  */
+	OR_SET_WORD(handle, &regs->eb2csr, reset);
+
+	/*
+	 * Program the DMA engine.
+	 */
+	eb2_load_fragment(eng);
+
+	/*
+	 * Start playing before we load the next fragment.
+	 */
+	OR_SET_WORD(handle, &regs->eb2csr, enable);
+
+	/*
+	 * Program a 2nd fragment.
+	 */
+	eb2_load_fragment(eng);
+
+	return (DDI_SUCCESS);
+
+}	/* eb2_start_engine() */
+
+/*
+ * eb2_stop_engine()
+ *
+ * Description:
+ *	This routine stops the DMA engine.
+ *
+ *	NOTE: The state structure must be locked before this routine is called.
+ *
+ * Arguments:
+ *	CS_engine_t	*eng	The engine to stop
+ *
+ * Returns:
+ *	void
+ */
+static void
+eb2_stop_engine(CS_engine_t *eng)
+{
+	ddi_acc_handle_t	handle = eng->ce_regsh;
+	cs4231_eb2regs_t	*regs = eng->ce_eb2regs;
+	uint_t			csr;
+
+	/* shut off DMA and disable interrupts */
+	AND_SET_WORD(handle, &regs->eb2csr, ~(EB2_EN_DMA | EB2_INT_EN));
+
+	csr = ddi_get32(handle, &regs->eb2csr);
+	for (int x = 0; (csr & EB2_CYC_PENDING) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);
+		csr = ddi_get32(handle, &regs->eb2csr);
+	}
+
+	/* set the RESET bit to stop audio, also clear any TC interrupt */
+	OR_SET_WORD(handle, &regs->eb2csr, EB2_RESET | EB2_TC);
+
+	/* wait for the FIFO to drain */
+	csr = ddi_get32(handle, &regs->eb2csr);
+	for (int x = 0; (csr & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
+		drv_usecwait(1);		/* don't beat on the bus */
+		csr = ddi_get32(handle, &regs->eb2csr);
+	}
+
+	/* clear the RESET and EN_DMA bits */
+	AND_SET_WORD(handle, &regs->eb2csr, ~(EB2_RESET|EB2_EN_DMA));
+
+}	/* eb2_stop_engine() */
+
+/*
+ * eb2_power()
+ *
+ * Description:
+ *	This routine turns the Codec on or off using the auxio register
+ *	in the eb2 device (cheerio or rio). Fortunately we don't need
+ *	to delay like we do with the APC.
+ *
+ *	NOTE: The state structure must be locked when this routine is called.
+ *
+ * Arguments:
+ *	CS_state_t	*state		Ptr to the device's state structure
+ *	int		level		Power level to set
+ *
+ * Returns:
+ *	void
+ */
+static void
+eb2_power(CS_state_t *state, int level)
+{
+	ddi_acc_handle_t	xhandle = EB2_AUXIO_HNDL;
+
+	if (level == CS4231_PWR_ON) {	/* turn power on */
+		AND_SET_WORD(xhandle, EB2_AUXIO_REG, ~EB2_AUXIO_COD_PDWN);
+	} else {	/* turn power off */
+		OR_SET_WORD(xhandle, EB2_AUXIO_REG, EB2_AUXIO_COD_PDWN);
+	}
+
+}	/* eb2_power() */
+
+
+/* *******  Local Routines ************************************************** */
+
+/*
+ * eb2_intr()
+ *
+ * Description:
+ *	EB2 interrupt serivce routine. First we find out why there was an
+ *	interrupt, then we take the appropriate action.
+ *
+ * Arguments:
+ *	caddr_t		T	Pointer to the interrupting device's state
+ *				structure
+ *
+ * Returns:
+ *	DDI_INTR_CLAIMED	Interrupt claimed and processed
+ *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
+ */
+static uint_t
+eb2_intr(caddr_t T)
+{
+	CS_engine_t		*eng = (void *)T;
+	CS_state_t		*state = eng->ce_state;
+	cs4231_eb2regs_t	*regs = eng->ce_eb2regs;
+	ddi_acc_handle_t	handle = eng->ce_regsh;
+	uint32_t		csr;
+	boolean_t		doit = B_FALSE;
+
+	/* the state must be protected */
+	mutex_enter(&state->cs_lock);
+	if (state->cs_suspended) {
+		mutex_exit(&state->cs_lock);
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	/* get the EB2 CSR */
+	csr = ddi_get32(handle, &regs->eb2csr);
+
+	/* make sure this device sent the interrupt */
+	if (!(csr & EB2_INT_PEND)) {
+		mutex_exit(&state->cs_lock);
+		/* nope, this isn't our interrupt */
+		return (DDI_INTR_UNCLAIMED);
+	}
+
+	/* clear all interrupts we captured at this time */
+	ddi_put32(handle, &regs->eb2csr, (csr|EB2_TC));
+
+	if (csr & EB2_TC) {
+
+		/* try to load the next audio buffer */
+		eb2_load_fragment(eng);
+
+		/* if engine was started, then we want to consume later */
+		doit = eng->ce_started;
+
+	} else if (csr & EB2_ERR_PEND) {
+		audio_dev_warn(state->cs_adev, "error intr: 0x%x", csr);
+
+	} else {
+		audio_dev_warn(state->cs_adev, "unknown intr: 0x%x", csr);
+	}
+
+	/* update the kernel interrupt statisitcs */
+	if (state->cs_ksp) {
+		KIOP(state)->intrs[KSTAT_INTR_HARD]++;
+	}
+
+	mutex_exit(&state->cs_lock);
+
+	if (doit) {
+		if (eng->ce_num == CS4231_PLAY) {
+			audio_engine_consume(eng->ce_engine);
+		} else {
+			audio_engine_produce(eng->ce_engine);
+		}
+	}
+
+	return (DDI_INTR_CLAIMED);
+
+}	/* eb2_intr() */
+
+static void
+eb2_load_fragment(CS_engine_t *eng)
+{
+	ddi_acc_handle_t	handle = eng->ce_regsh;
+	cs4231_eb2regs_t	*regs = eng->ce_eb2regs;
+
+	/* if next address already loaded, then we're done */
+	if ((ddi_get32(handle, &regs->eb2csr) & EB2_NA_LOADED)) {
+		return;
+	}
+
+	/*
+	 * For eb2 we first program the Next Byte Count Register.
+	 */
+	ddi_put32(handle, &regs->eb2bcr, eng->ce_fragsz);
+
+	/* now program the Next Address Register */
+	ddi_put32(handle, &regs->eb2acr, eng->ce_paddr[eng->ce_cfrag]);
+
+	eng->ce_cfrag++;
+	eng->ce_cfrag %= CS4231_NFRAGS;
+	eng->ce_count += eng->ce_fragfr;
+}
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audiocs.conf b/usr/src/uts/sun/io/audio/drv/audiocs/audiocs.conf
index 984e9e7abf..60424b03f0 100644
--- a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audiocs.conf
+++ b/usr/src/uts/sun/io/audio/drv/audiocs/audiocs.conf
@@ -2,9 +2,8 @@
 # CDDL HEADER START
 #
 # The contents of this file are subject to the terms of the
-# Common Development and Distribution License, Version 1.0 only
-# (the "License").  You may not use this file except in compliance
-# with the License.
+# Common Development and Distribution License (the "License").
+# You may not use this file except in compliance with the License.
 #
 # You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 # or http://www.opensolaris.org/os/licensing.
@@ -20,25 +19,14 @@
 # CDDL HEADER END
 #
 #
-# Copyright 1999-2002 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-#
 # Configuration file for the audiocs audio driver.
 #
 #	WARNING: This is an UNSTABLE configuration file. Its contents
 #		may change at any time.
 #
-# mixer-mode determines the mode the mixer audio personality module
-# sets for this driver when it is loaded. The mixerctl(1) command may
-# be used to change modes on the fly.
-#
-#	mixer-mode=1;	<-- mixer enabled
-#	mixer-mode=0;	<-- mixer disabled, old audio(7I) behavior
-#
-
-mixer-mode=1;
 
 #
 # play-interrupts sets the number of interrupts per second when playing.
@@ -57,10 +45,3 @@ mixer-mode=1;
 #
 play-interrupts=175;
 record-interrupts=175;
-
-#
-# Uncomment reset-configuration to cause the audiocs driver's state to
-# be reset to the default when the driver is loaded. Otherwise this state
-# is retained across driver unload/reload cycles, but not across reboots.
-#
-#reset-configuration=1;
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audio1575/audio1575.c b/usr/src/uts/sun/io/audio/sada/drv/audio1575/audio1575.c
deleted file mode 100644
index 8a9c92de1c..0000000000
--- a/usr/src/uts/sun/io/audio/sada/drv/audio1575/audio1575.c
+++ /dev/null
@@ -1,4119 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-
-/*
- * audio1575 Audio Driver
- *
- * The driver is primarily targeted at providing audio support for
- * those sparc systems which use the Uli M1575 audio core
- * and the Analog Devices AD1981 codec.
- *
- * This driver uses the mixer Audio Personality Module to implement
- * audio(7I) and mixer(7I) semantics. Both play and record are single
- * streaming.
- *
- * The M1575 audio core, in AC'97 controller mode, has independent
- * channels for PCM in, PCM out, mic in, modem in, and modem out.
- *
- * The AC'97 controller is a PCI bus master with scatter/gather
- * support. Each channel has a DMA engine. Currently, we use only
- * the PCM in and PCM out channels. Each DMA engine uses one buffer
- * descriptor list. And the buffer descriptor list is an array of up
- * to 32 entries, each of which describes a data buffer. Each entry
- * contains a pointer to a data buffer, control bits, and the length
- * of the buffer being pointed to, where the length is expressed as
- * the number of samples. This, combined with the 16-bit sample size,
- * gives the actual physical length of the buffer.
- *
- * We use the BD list (buffer descriptor list) as a round-robin FIFO.
- * Both the software and hardware loop around the BD list. For playback,
- * the software writes to the buffers pointed by the BD entries of BD
- * list, and the hardware sends the data in the buffers out. For record,
- * the process is reversed. So we define the struct, m1575_sample_buf,
- * to handle the BD. The software uses the head, tail and avail fields of
- * this structure to manipulate the FIFO. The head field indicates the
- * first valid BD hardware can manipulate. The tail field indicates the
- * BD after the last valid BD. And the avail field indicates how many
- * buffers are available. There're also two hardware registers to index
- * the FIFO, the CIV (current index value) indicating the current BD the
- * hardware is transferring, and the LVI (last valid index) indicating
- * the last valid BD that contains proper data which the hardware should
- * not pass over. Each time a BD is processed, the hardware will issue an
- * interrupt. If the system is busy, there can be more than one BD to be
- * processed when the OS have chance to handle the interrupt. When an
- * interrupt generated, the interrupt handler will first reclaim the BD(s)
- * which had been transferred, which will be the limit [head, CIV-1], then
- * update the value of the head field to CIV, update the value of avail to
- * CIV - head. And then it will process avail BDs from tail, and set the
- * LVI to the last processed BD and update tail to LVI + 1, and update the
- * avail to 0.
- *
- * We allocate N blocks of DMA memory, say A and B, for every DMA
- * engine, and bind the first, Block A, to the even entries of BDL,
- * while bind the second to the odd entries. That's say, for each buffer
- * descriptor list of DMA engine, entry 0, 2,,, 30 would be bound to Block
- * A, and entry 1, 3,,, 31 would be bound to Block B. Take the playback as
- * an example. At the beginning of playback, we set the entry 0 and 1 to
- * point to block A and B separately, and tell the DMA engine that the last
- * valid entry is entry 1. So the DMA engine doesn't access the entries after
- * entry 1. When the first playback interrupt generated, we reclaim entry
- * 0, and fill the BD entry 2 with the address of Block A, then set the
- * entry 2 to be the last valid entry, and so on. So at any time there are
- * at most two entries available for per DMA engine.
- *
- * Every time we program AC97 codec, we save the value in codec_shadow[].
- * This means that register state information is saved for power management
- * shutdown (we'll support this later). When the codec is started back up
- * we use this saved state to restore codec's state in audio1575_chip_init()
- *
- * TODO: System power management is not yet supported by the driver.
- *
- * 	NOTE:
- * 	This driver depends on the misc/audiosup, misc/amsrc2 and
- * 	misc/mixer modules being loaded first.
- */
-
-#include <sys/types.h>
-#include <sys/modctl.h>
-#include <sys/kmem.h>
-#include <sys/conf.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/pci.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/ac97.h>
-#include <sys/audio/impl/audio1575_impl.h>
-#include <sys/audio/audio1575.h>
-#include <sys/cpuvar.h>
-
-/*
- * Module linkage routines for the kernel
- */
-static int audio1575_getinfo(dev_info_t *, ddi_info_cmd_t, void*, void**);
-static int audio1575_attach(dev_info_t *, ddi_attach_cmd_t);
-static int audio1575_detach(dev_info_t *, ddi_detach_cmd_t);
-
-/*
- * Entry point routine prototypes
- */
-static int audio1575_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int audio1575_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int audio1575_ad_start_play(audiohdl_t, int);
-static void audio1575_ad_pause_play(audiohdl_t, int);
-static void audio1575_ad_stop_play(audiohdl_t, int);
-static int audio1575_ad_start_record(audiohdl_t, int);
-static void audio1575_ad_stop_record(audiohdl_t, int);
-
-/*
- * interrupt handler
- */
-static uint_t	audio1575_intr(caddr_t);
-
-/*
- * Local Routine Prototypes
- */
-static int audio1575_init_ac97(audio1575_state_t *, int);
-static int audio1575_reset_ac97(audio1575_state_t *);
-static int audio1575_codec_sync(audio1575_state_t *);
-static int audio1575_write_ac97(audio1575_state_t *, int, uint16_t);
-static int audio1575_read_ac97(audio1575_state_t *, int, uint16_t *);
-static int audio1575_and_ac97(audio1575_state_t *, int, uint16_t);
-static int audio1575_or_ac97(audio1575_state_t *, int, uint16_t);
-static int audio1575_chip_init(audio1575_state_t *, int);
-static int audio1575_init_state(audio1575_state_t *, dev_info_t *);
-static int audio1575_map_regs(dev_info_t *, audio1575_state_t *);
-static void audio1575_unmap_regs(audio1575_state_t *);
-static int audio1575_alloc_sample_buf(audio1575_state_t *, int, int);
-static void audio1575_free_sample_buf(audio1575_state_t *, int);
-static void audio1575_dma_stop(audio1575_state_t *);
-static int audio1575_fill_play_buf(audio1575_state_t *);
-static void audio1575_reclaim_play_buf(audio1575_state_t *);
-static int audio1575_prepare_record_buf(audio1575_state_t *);
-static void audio1575_reclaim_record_buf(audio1575_state_t *);
-static int audio1575_set_gain(audio1575_state_t *, int, int, int);
-static int audio1575_set_port(audio1575_state_t *, int, int);
-static int audio1575_set_monitor_gain(audio1575_state_t *, int);
-static void audio1575_pci_enable(audio1575_state_t *);
-static void audio1575_pci_disable(audio1575_state_t *);
-static int audio1575_dma_pause(audio1575_state_t *, int);
-static int audio1575_dma_resume(audio1575_state_t *, int);
-static int audio1575_dma_reset(audio1575_state_t *, int);
-
-/*
- * Global variables, but used only by this file.
- */
-
-/* anchor for soft state structures */
-static void	*audio1575_statep;
-
-/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
-static char	*audio1575_name = M1575_NAME;
-
-/*
- * STREAMS structures
- */
-
-/* STREAMS driver id and limit value struct */
-static struct module_info audio1575_modinfo = {
-	M1575_IDNUM,		/* module ID number */
-	M1575_NAME,		/* module name */
-	M1575_MINPACKET,	/* minimum packet size */
-	M1575_MAXPACKET,	/* maximum packet size */
-	M1575_HIWATER,		/* high water mark */
-	M1575_LOWATER,		/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit audio1575_rqueue = {
-	audio_sup_rput,		/* put procedure */
-	audio_sup_rsvc,		/* service procedure */
-	audio_sup_open,		/* open procedure */
-	audio_sup_close,	/* close procedure */
-	NULL,			/* unused */
-	&audio1575_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit audio1575_wqueue = {
-	audio_sup_wput,		/* write procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&audio1575_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab audio1575_str_info = {
-	&audio1575_rqueue,	/* read queue */
-	&audio1575_wqueue,	/* write queue */
-	NULL,			/* mux lower read queue */
-	NULL,			/* mux lower write queue */
-};
-
-/*
- * DDI Structures
- */
-
-/* Entry points structure */
-static struct cb_ops audio1575_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&audio1575_str_info,	/* cb_str */
-	D_NEW | D_MP | D_64BIT, /* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_awrite */
-};
-
-/* Device operations structure */
-static struct dev_ops audio1575_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	audio1575_getinfo,	/* devo_getinfo */
-	nulldev,		/* devo_identify - obsolete */
-	nulldev,		/* devo_probe */
-	audio1575_attach,	/* devo_attach */
-	audio1575_detach,	/* devo_detach */
-	nodev,			/* devo_reset */
-	&audio1575_cb_ops,	/* devi_cb_ops */
-	NULL,			/* devo_bus_ops */
-	NULL,			/* devo_power */
-	ddi_quiesce_not_supported,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv audio1575_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	M1575_MOD_NAME,		/* drv_linkinfo */
-	&audio1575_dev_ops,	/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage audio1575_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&audio1575_modldrv,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-static uint_t audio1575_mixer_srs[] = {
-	M1575_SAMPR5510,
-	M1575_SAMPR48000,
-	0
-};
-
-static uint_t audio1575_compat_srs [] = {
-	M1575_SAMPR8000, M1575_SAMPR9600, M1575_SAMPR11025,
-	M1575_SAMPR16000, M1575_SAMPR18900, M1575_SAMPR22050,
-	M1575_SAMPR27420, M1575_SAMPR32000, M1575_SAMPR33075,
-	M1575_SAMPR37800, M1575_SAMPR44100, M1575_SAMPR48000,
-	0
-};
-
-static am_ad_sample_rates_t audio1575_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	audio1575_mixer_srs
-};
-
-static am_ad_sample_rates_t audio1575_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	audio1575_compat_srs
-};
-
-static uint_t audio1575_channels[] = {
-	AUDIO_CHANNELS_STEREO,
-	0
-};
-
-static am_ad_cap_comb_t audio1575_combinations[] = {
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
-
-/* AD1981B Equalization Biquad IIR Filter coefficient address table */
-static m1575_biquad_t filters[] = {
-	{0x1b, 0x0f}, {0x1a, 0x00}, {0x19, 0x00}, {0x1d, 0x00}, {0x1c, 0x00},
-	{0x20, 0x0f}, {0x1f, 0x00}, {0x1e, 0x00}, {0x22, 0x00}, {0x21, 0x00},
-	{0x25, 0x0f}, {0x24, 0x00}, {0x23, 0x00}, {0x27, 0x00}, {0x26, 0x00},
-	{0x2a, 0x0f}, {0x29, 0x00}, {0x28, 0x00}, {0x2c, 0x00}, {0x2b, 0x00},
-	{0x2f, 0x0f}, {0x2e, 0x00}, {0x2d, 0x00}, {0x31, 0x00}, {0x30, 0x00},
-	{0x34, 0x0f}, {0x33, 0x00}, {0x32, 0x00}, {0x36, 0x00}, {0x35, 0x00},
-	{0x39, 0x0f}, {0x38, 0x00}, {0x37, 0x00}, {0x3b, 0x00}, {0x3a, 0x00},
-};
-
-/*
- * device access attributes for register mapping
- */
-static struct ddi_device_acc_attr dev_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_LE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-/*
- * DMA attributes of buffer descriptor list
- */
-static ddi_dma_attr_t bdlist_dma_attr = {
-	DMA_ATTR_V0,	/* version */
-	0x0000000000000000LL,		/* dlim_addr_lo */
-	0x00000000ffffffffLL,		/* dlim_addr_hi */
-	0x000000000000ffffLL,		/* DMA counter register - 64 bits */
-	0x0000000000000008LL,		/* DMA address align must be 8-bytes */
-	0x0000003c,			/* 1 through 64 byte burst sizes */
-	0x00000008,			/* min xfer DMA size BDList entry */
-	0x00000000000ffffLL,		/* max xfer size, 64K */
-	0x000000000001fffLL,		/* seg, set to PAGESIZE */
-	0x00000001,			/* s/g list length, no s/g */
-	0x00000008,			/* granularity of device minxfer */
-	0				/* DMA flags use virtual address */
-};
-
-/*
- * DMA attributes of buffers to be used to receive/send audio data
- */
-static ddi_dma_attr_t	sample_buf_dma_attr = {
-	DMA_ATTR_V0,
-	0x0000000000000000LL,		/* dlim_addr_lo */
-	0x00000000ffffffffLL,		/* dlim_addr_hi */
-	0x000000000001fffeLL,		/* DMA counter register - 16 bits */
-	0x0000000000000002LL,		/* DMA address align 2-byte boundary */
-	0x0000003c,			/* 1 through 60 byte burst sizes */
-	0x00000004,			/* min xfer DMA size BDList entry */
-	0x000000000001ffffLL,		/* max xfer size, 64K */
-	0x000000000001ffffLL,		/* seg, set to 64K */
-	0x00000001,			/* s/g list length, no s/g */
-	0x00000004,			/* granularity of device minxfer */
-	0				/* DMA flags use virtual address */
-};
-
-static am_ad_entry_t audio1575_entry = {
-	NULL,				/* ad_setup() */
-	NULL,				/* ad_teardown() */
-	audio1575_ad_set_config,	/* ad_set_config() */
-	audio1575_ad_set_format,	/* ad_set_format() */
-	audio1575_ad_start_play,	/* ad_start_play() */
-	audio1575_ad_pause_play,	/* ad_pause_play() */
-	audio1575_ad_stop_play,		/* ad_stop_play() */
-	audio1575_ad_start_record,	/* ad_start_record() */
-	audio1575_ad_stop_record,	/* ad_stop_record() */
-	NULL,				/* ad_ioctl() */
-	NULL				/* ad_iocdata() */
-};
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *	This is how access is initially given to all the static structures.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int	error;
-
-	ATRACE("audio1575 _init() entering", NULL);
-
-	if ((error = ddi_soft_state_init(&audio1575_statep,
-	    sizeof (audio1575_state_t), 1)) != 0) {
-		ATRACE("audio1575 ddi_soft_state_init() failure",
-		    error);
-
-		return (error);
-	}
-
-	if ((error = mod_install(&audio1575_modlinkage)) != 0) {
-		ddi_soft_state_fini(&audio1575_statep);
-	}
-
-	ATRACE("audio1575 _init() returning", error);
-
-	return (error);
-
-}	/* _init() */
-
-/*
- * _fini()
- *
- * Description:
- *	Module de-initialization, called when the driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int		error;
-
-	ATRACE("audio1575 _fini() entering", audio1575_statep);
-
-	if ((error = mod_remove(&audio1575_modlinkage)) != 0) {
-
-		return (error);
-	}
-
-	ddi_soft_state_fini(&audio1575_statep);
-
-	ATRACE("audio1575 _fini() returning", error);
-
-	return (0);
-
-}	/* _fini() */
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns information about the driver.
- *
- * Arguments:
- *	modinfo		*modinfop	Pointer to the opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int	error;
-
-	ATRACE("audio1575 _info() entering", NULL);
-
-	error = mod_info(&audio1575_modlinkage, modinfop);
-
-	ATRACE("audio1575 _info() returning", error);
-
-	return (error);
-
-}	/* _info() */
-
-
-/* ******************* Driver Entry Points ********************************* */
-
-/*
- * audio1575_getinfo()
- */
-static int
-audio1575_getinfo(dev_info_t *dip, ddi_info_cmd_t cmd, void *arg, void **result)
-{
-	audio1575_state_t	*statep;
-	int 			instance;
-	int 			error = DDI_FAILURE;
-
-	ATRACE("audio1575_getinfo() entering", dip);
-
-	switch (cmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((statep = ddi_get_soft_state(audio1575_statep,
-		    instance)) != NULL) {
-			*result = statep->m1575_dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
-		break;
-
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void *)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-
-	default:
-		break;
-	}
-
-	ATRACE("audio1575_getinfo() returning", error);
-
-	return (error);
-
-}	/* audio1575_getinfo() */
-
-/*
- * audio1575_attach()
- *
- * Description:
- *	Attach an instance of the audio1575 driver. This routine does the
- * 	device dependent attach tasks. When it is completed, it calls
- *	audio_sup_register() and am_attach() so they may do their work.
- *
- *	NOTE: mutex_init() no longer needs a name string, so set
- *		to NULL to save kernel space.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd	Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-static int
-audio1575_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	audio1575_state_t	*statep;
-	uint32_t		intrsr;
-	uint32_t		intrcr;
-	int 			instance;
-	audio_sup_reg_data_t	data;
-
-	ATRACE("audio1575_attach() entering", dip);
-
-	instance = ddi_get_instance(dip);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		ATRACE("audio1575_attach() DDI_ATTACH", NULL);
-		break;
-	case DDI_RESUME:
-		ATRACE("audio1575_attach() DDI_RESUME", NULL);
-
-		/* we've already allocated the state structure so get ptr */
-		if ((statep = ddi_get_soft_state(audio1575_statep, instance)) ==
-		    NULL) {
-			audio_sup_log(NULL, CE_WARN,
-			    "!%s%d: attach() RESUME get soft state failure",
-			    audio1575_name, instance);
-
-			return (DDI_FAILURE);
-		}
-
-		ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-		mutex_enter(&statep->m1575_intr_mutex);
-
-		ASSERT((statep->m1575_flags & M1575_DMA_SUSPENDED) ==
-		    M1575_DMA_SUSPENDED);
-
-		statep->m1575_flags &= ~M1575_DMA_SUSPENDED;
-
-		mutex_exit(&statep->m1575_intr_mutex);
-
-		/*
-		 * Restore the audio1575 chip's state and
-		 * put the DMA Engines in a known state
-		 */
-		if ((audio1575_chip_init(statep, M1575_INIT_RESTORE)) ==
-		    AUDIO_FAILURE) {
-			ATRACE("audio1575_attach()chip init failure",
-			    DDI_FAILURE);
-
-			return (DDI_FAILURE);
-		}
-
-		/*
-		 * Start playing and recording, if not needed they'll stop
-		 * on their own. But, we don't start them if the hardware has
-		 * failed.
-		 */
-		if (audio_sup_restore_state(statep->m1575_ahandle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(statep->m1575_ahandle, CE_WARN,
-			    "!attach() audio restart failure");
-		}
-
-		ATRACE("audio1575_attach() DDI_RESUME done", NULL);
-
-		ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-		return (DDI_SUCCESS);
-
-	default:
-		ATRACE("audio1575_attach() unknown command", cmd);
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() unknown command: 0x%x",
-		    audio1575_name, instance, cmd);
-
-		return (DDI_FAILURE);
-	}
-
-	/* allocate the soft state structure */
-	if (ddi_soft_state_zalloc(audio1575_statep, instance) != DDI_SUCCESS) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() soft state allocate failure",
-		    audio1575_name, instance);
-		goto error_state;
-	}
-
-	/* get the state structure */
-	if ((statep = ddi_get_soft_state(audio1575_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() ddi_get_soft_state() failure",
-		    audio1575_name, instance);
-		goto error_state;
-	}
-
-	statep->m1575_inst = instance;
-
-	/* call audiosup module registration routine */
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-
-	/* register the driver with the audio support module */
-	if ((statep->m1575_ahandle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() audio_sup_register() failure",
-		    audio1575_name, instance);
-		goto error_state;
-	}
-
-	/* map in the audio registers */
-	if (audio1575_map_regs(dip, statep) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!attach() couldn't map registers");
-		goto error_audiosup;
-	}
-
-	/* Enable PCI I/O and Memory Spaces */
-	audio1575_pci_enable(statep);
-
-	/*
-	 * clear the interrupt control and status register
-	 * READ/WRITE/READ workaround required
-	 * for buggy hardware
-	 */
-
-	intrcr = 0L;
-	M1575_AM_PUT32(M1575_INTRCR_REG, intrcr);
-	intrcr = M1575_AM_GET32(M1575_INTRCR_REG);
-
-	intrsr = M1575_AM_GET32(M1575_INTRSR_REG);
-	M1575_AM_PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
-	intrsr = M1575_AM_GET32(M1575_INTRSR_REG);
-
-	/* initialize the audio state structures */
-	if ((audio1575_init_state(statep, dip)) != AUDIO_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!attach() init state structure failure");
-		goto error_pci_disable;
-	}
-
-	/* from here on, must destroy the mutex on error */
-	mutex_init(&statep->m1575_intr_mutex, NULL, MUTEX_DRIVER,
-	    DDI_INTR_PRI(statep->m1575_intr_pri));
-
-	/* allocate play and record sample buffers */
-	if ((audio1575_alloc_sample_buf(statep, M1575_DMA_PCM_OUT,
-	    statep->m1575_play_buf_size) == AUDIO_FAILURE) ||
-	    (audio1575_alloc_sample_buf(statep, M1575_DMA_PCM_IN,
-	    statep->m1575_record_buf_size) == AUDIO_FAILURE)) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!attach() couldn't alloc sample buffers");
-		goto error_destroy;
-	}
-
-	/* initialize audio controller and AC97 codec */
-	if (audio1575_chip_init(statep, M1575_INIT_NO_RESTORE) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle,  CE_WARN,
-		    "!attach() failure to init chip");
-		goto error_dealloc;
-	}
-
-	/* save private state */
-	audio_sup_set_private(statep->m1575_ahandle, statep);
-
-	/* call the mixer attach() routine */
-	if (am_attach(statep->m1575_ahandle, cmd, &statep->m1575_ad_info) !=
-	    AUDIO_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!attach() am_attach() failure");
-		goto error_dealloc;
-	}
-
-	/* set up kernel statistics */
-	if ((statep->m1575_ksp = kstat_create(M1575_NAME, instance,
-	    M1575_NAME, "controller", KSTAT_TYPE_INTR, 1,
-	    KSTAT_FLAG_PERSISTENT)) != NULL) {
-		kstat_install(statep->m1575_ksp);
-	}
-
-	/* now add our interrupt handler */
-	if (ddi_intr_add_handler(statep->m1575_h_table[0],
-	    (ddi_intr_handler_t *)audio1575_intr, (caddr_t)statep, NULL)) {
-		audio_sup_log(statep->m1575_ahandle,
-		    CE_WARN, "!attach() ddi_intr_add_handler() failure");
-		ATRACE("audio1575_attach() ddi_intr_add_handler() failure",
-		    NULL);
-		goto error_kstat;
-	}
-
-	/* Enable PCI Interrupts */
-	M1575_PCI_PUT8(M1575_PCIMISC_REG, M1575_PCIMISC_INTENB);
-
-	/* enable audio interrupts */
-	if (ddi_intr_enable(statep->m1575_h_table[0]) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!attach() - ddi_intr_enable() failure");
-		ATRACE("audio1575_attach() ddi_add_intr failure", NULL);
-		goto error_intr_enable;
-	}
-
-	/* everything worked out, so report the device */
-	ddi_report_dev(dip);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_attach() returning DDI_SUCCESS", NULL);
-
-	return (DDI_SUCCESS);
-
-error_intr_enable:
-	ATRACE("audio1575_attach() error_intr_enable", NULL)
-	/* Disable PCI Interrupts */
-	M1575_PCI_PUT8(M1575_PCIMISC_REG, 0);
-	(void) ddi_intr_remove_handler(statep->m1575_h_table[0]);
-
-error_kstat:
-	ATRACE("audio1575_attach() error_kstat", NULL);
-	if (statep->m1575_ksp) {
-		kstat_delete(statep->m1575_ksp);
-	}
-
-	(void) am_detach(statep->m1575_ahandle, DDI_DETACH);
-
-error_dealloc:
-	ATRACE("audio1575_attach() error_dealloc", NULL);
-	audio1575_free_sample_buf(statep, M1575_DMA_PCM_IN);
-	audio1575_free_sample_buf(statep, M1575_DMA_PCM_OUT);
-
-error_destroy:
-	ATRACE("audio1575_attach() error_destroy", statep);
-	(void) ddi_intr_free(statep->m1575_h_table[0]);
-	kmem_free(statep->m1575_h_table, sizeof (ddi_intr_handle_t));
-	mutex_destroy(&statep->m1575_intr_mutex);
-
-error_pci_disable:
-	ATRACE("audio1575_attach() error_pci_disable", NULL)
-	audio1575_pci_disable(statep);
-
-error_unmap:
-	ATRACE("audio1575_attach() error_unmap", NULL);
-	audio1575_unmap_regs(statep);
-
-error_audiosup:
-	ATRACE("audio1575_attach() error_audiosup", statep);
-	(void) audio_sup_unregister(statep->m1575_ahandle);
-
-error_state:
-	ATRACE("audio1575_attach() error_state", statep);
-	ddi_soft_state_free(audio1575_statep, instance);
-
-	ATRACE("audio1575_attach() returning DDI_FAILURE", NULL);
-
-	return (DDI_FAILURE);
-
-}	/* audio1575_attach() */
-
-/*
- * audio1575_detach()
- *
- * Description:
- *	Detach an instance of the audio1575 driver. After the Codec is detached
- *	we call am_detach() and audio_sup_register() so they may do their work.
- *
- * Arguments:
- *	dev_info_t		*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t	cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS	The driver was detached
- *	DDI_FAILURE	The driver couldn't be detached
- */
-static int
-audio1575_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
-{
-	audio1575_state_t	*statep;
-	int			instance;
-
-	instance = ddi_get_instance(dip);
-
-	ATRACE("audio1575_detach() entering", instance);
-
-	if ((statep = ddi_get_soft_state(audio1575_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: detach() get soft state failure",
-		    audio1575_name, instance);
-		ATRACE("audio1575_detach() get soft state failure", instance);
-
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-	case DDI_SUSPEND:
-		ATRACE("audio1575_detach() DDI_SUSPEND", NULL);
-
-		mutex_enter(&statep->m1575_intr_mutex);
-
-		ASSERT((statep->m1575_flags & M1575_DMA_SUSPENDED) !=
-		    M1575_DMA_SUSPENDED);
-
-		/* stop all new operations */
-		statep->m1575_flags |= M1575_DMA_SUSPENDED;
-
-		/*
-		 * stop all DMA operations
-		 */
-		(void) audio1575_dma_stop(statep);
-
-		/*
-		 * Save the controller state. The Codec's state is
-		 * already in codec_shadow[].
-		 */
-		if (audio_sup_save_state(statep->m1575_ahandle,
-		    AUDIO_ALL_DEVICES, AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(statep->m1575_ahandle, CE_WARN,
-			    "!detach() audio save failure");
-		}
-
-		mutex_exit(&statep->m1575_intr_mutex);
-
-		ATRACE("audio1575_detach() SUSPEND successful", statep);
-
-		ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-		return (DDI_SUCCESS);
-
-	default:
-		ATRACE("audio1575_detach() unknown command", cmd);
-
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!%s%d: detach() unknown command: 0x%x",
-		    audio1575_name, instance, cmd);
-
-		return (DDI_FAILURE);
-	}
-
-	/* stop DMA engines */
-	mutex_enter(&statep->m1575_intr_mutex);
-	audio1575_dma_stop(statep);
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	/* disable interrupts */
-	(void) ddi_intr_disable(statep->m1575_h_table[0]);
-
-	/* Remove the interrupt handler */
-	(void) ddi_intr_remove_handler(statep->m1575_h_table[0]);
-
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* reset the AD1981B codec */
-	M1575_AM_PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
-
-	/* turn off the AC_LINK clock */
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 0);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 4);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 0);
-
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	/* detach audio mixer */
-	(void) am_detach(statep->m1575_ahandle, cmd);
-
-	/*
-	 * call the audio support module's detach routine to remove this
-	 * driver completely from the audio driver architecture.
-	 */
-	(void) audio_sup_unregister(statep->m1575_ahandle);
-
-	/* free the interrupt handle */
-	(void) ddi_intr_free(statep->m1575_h_table[0]);
-
-	/* free memory */
-	kmem_free(statep->m1575_h_table, sizeof (ddi_intr_handle_t));
-
-	/* free DMA memory */
-	audio1575_free_sample_buf(statep, M1575_DMA_PCM_IN);
-	audio1575_free_sample_buf(statep, M1575_DMA_PCM_OUT);
-
-	/* free the kernel statistics structure */
-	if (statep->m1575_ksp) {
-		kstat_delete(statep->m1575_ksp);
-	}
-
-	/* Disable PCI I/O and Memory Spaces */
-	audio1575_pci_disable(statep);
-
-	/* unmap the registers */
-	audio1575_unmap_regs(statep);
-
-	/* destroy the state mutex */
-	mutex_destroy(&statep->m1575_intr_mutex);
-
-	/* free the memory for the state pointer */
-	ddi_soft_state_free(audio1575_statep, instance);
-
-	ATRACE("audio1575_detach() returning success", NULL);
-
-	return (DDI_SUCCESS);
-
-}	/* audio1575_detach */
-
-/*
- * audio1575_intr()
- *
- * Description:
- *	Interrupt service routine for both play and record. For play we
- *	get the next buffers worth of audio. For record we send it on to
- *	the mixer.
- *
- *	Each of buffer descriptor has a field IOC(interrupt on completion)
- *	When both this and the IOC bit of correspondent dma control register
- *	is set, it means that the controller should issue an interrupt upon
- *	completion of this buffer. Note that in the clearing of the interrupts
- *	below that the PCM IN and PCM out interrupts ar cleared by their
- *	respective control registers and not by writing a '1' to the INTRSR
- *	the interrupt status register. Only CPRINTR,SPINTR,and GPIOINTR
- *	require a '1' written to the INTRSR register to clear those
- *	interrupts. See comments below.
- *
- * Arguments:
- *	caddr_t		arg	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-audio1575_intr(caddr_t arg)
-{
-	audio1575_state_t	*statep = (audio1575_state_t *)arg;
-	uint32_t		intrsr;
-	uint16_t		sr;
-
-	ATRACE("in audio1575_intr()", statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	intrsr = M1575_AM_GET32(M1575_INTRSR_REG);
-
-	/* check if device is interrupting */
-	if (intrsr == 0) {
-		if (statep->m1575_ksp) {
-			/* increment the spurious ino5 interrupt cnt */
-			M1575_KIOP(statep)->intrs[KSTAT_INTR_SPURIOUS]++;
-		}
-
-		mutex_exit(&statep->m1575_intr_mutex);
-
-		ATRACE("audio1575_intr() not our interrupt", intrsr);
-
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	/*
-	 * The Uli M1575 generates an interrupt for each interrupt
-	 * type. therefore we only process one interrupt type
-	 * per invocation of the audio1575_intr() routine.
-	 * WARNING: DO NOT attempt to optimize this by looping
-	 * until the INTRSR register is clear as this will
-	 * generate spurious ino5 interrupts.
-	 */
-	if (intrsr & M1575_INTRSR_PCMIINTR) {
-		/* Clear PCM IN interrupt */
-		ATRACE("audio1575_intr() PCMIINTR", intrsr);
-		sr = M1575_AM_GET16(M1575_PCMISR_REG);
-		M1575_AM_PUT16(M1575_PCMISR_REG, sr & M1575_STATUS_CLR);
-		/*
-		 * Note: This interrupt is not cleared by writing a '1'
-		 * to the M1575_INTRSR_REG according to the M1575 Super I/O
-		 * data sheet on page 189.
-		 */
-
-		if (statep->m1575_flags & M1575_DMA_RECD_STARTED) {
-			audio1575_reclaim_record_buf(statep);
-			(void) audio1575_prepare_record_buf(statep);
-		}
-	} else if (intrsr & M1575_INTRSR_PCMOINTR) {
-		/* Clear PCM OUT interrupt */
-		ATRACE("audio1575_intr() PCMOINTR", intrsr);
-		sr = M1575_AM_GET16(M1575_PCMOSR_REG);
-		M1575_AM_PUT16(M1575_PCMOSR_REG, sr & M1575_STATUS_CLR);
-		/*
-		 * Note: This interrupt is not cleared by writing a '1'
-		 * to the M1575_INTRSR_REG according to the M1575 Super I/O
-		 * data sheet on page 189.
-		 */
-
-		if (statep->m1575_flags & M1575_DMA_PLAY_STARTED) {
-			audio1575_reclaim_play_buf(statep);
-			(void) audio1575_fill_play_buf(statep);
-		}
-	} else if (intrsr & M1575_INTRSR_SPRINTR) {
-		/* Clear Status Register Available Interrupt */
-		ATRACE("audio1575_intr() SPRINTR", intrsr);
-		M1575_AM_PUT32(M1575_INTRSR_REG, M1575_INTRSR_SPRINTR);
-
-	} else if (intrsr & M1575_INTRSR_CPRINTR) {
-		/* Clear Command Register Available Interrupt */
-		ATRACE("audio1575_intr() CPRINTR", intrsr);
-		M1575_AM_PUT32(M1575_INTRSR_REG, M1575_INTRSR_CPRINTR);
-	} else if (intrsr & M1575_INTRSR_GPIOINTR) {
-		/* Clear General Purpose I/O Register Interrupt */
-		ATRACE("audio1575_intr() GPIOINTR", intrsr);
-		M1575_AM_PUT32(M1575_INTRSR_REG, M1575_INTRSR_GPIOINTR);
-	} else {
-		/* Clear Unknown Interrupt */
-		ATRACE("audio1575_intr() Unknown Interrupt", intrsr);
-		M1575_AM_PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
-	}
-
-	/* update the kernel interrupt statistics */
-	if (statep->m1575_ksp) {
-		M1575_KIOP(statep)->intrs[KSTAT_INTR_HARD]++;
-	}
-
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ATRACE("audio1575_intr() done", statep);
-
-	return (DDI_INTR_CLAIMED);
-
-}	/* audio1575_intr() */
-
-
-/* *********************** Mixer Entry Point Routines ******************* */
-
-/*
- * audio1575_ad_set_config()
- *
- * Description:
- *	This routine is used to set new Codec parameters, except the data
- *	format which has it's own routine. If the Codec doesn't support a
- *	particular parameter and it is asked to set it then we return
- *	AUDIO_FAILURE.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *	int		command		The configuration to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		Argument #1
- *	int		arg2		Argument #2, not always needed
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set,
- *				or the parameter couldn't be set
- */
-static int
-audio1575_ad_set_config(audiohdl_t ahandle, int stream, int command,
-    int dir, int arg1, int arg2)
-{
-	audio1575_state_t	*statep;
-	int 			rc = AUDIO_FAILURE;
-
-	ATRACE("audio1575_ad_set_config() entering", stream);
-	ATRACE("audio1575_ad_set_config() command", command);
-	ATRACE("audio1575_ad_set_config() dir", dir);
-	ATRACE("audio1575_ad_set_config() arg1", arg1);
-	ATRACE("audio1575_ad_set_config() arg2", arg2);
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/*
-	 * CAUTION: From here on we must goto done to exit.
-	 */
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, of balance on gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	arg1 --> gain
-		 * 	arg2 --> channel #, 0 == left, 1 == right
-		 */
-		rc = audio1575_set_gain(statep, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of in ports, as well as which ports
-		 * are modifiable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 * 	arg1 --> port bit pattern
-		 * 	arg2 --> not used
-		 */
-		rc = audio1575_set_port(statep, dir, arg1);
-		break;
-
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 * 	AUDIO_MIN_GAIN <= gain <= AUDIO_MAX_GAIN
-		 *
-		 * 	dir ---> N/A
-		 *	arg1 --> gain
-		 * 	arg2 --> not used
-		 */
-		rc = audio1575_set_monitor_gain(statep, arg1);
-		break;
-
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == mute, 0 == enable
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* mute */
-			(void) audio1575_or_ac97(statep,
-			    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-			(void) audio1575_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, HPVR_MUTE);
-			(void) audio1575_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MMVR_MUTE);
-			rc = AUDIO_SUCCESS;
-		} else {	/* not muted */
-			/* by setting the port we unmute only active ports */
-			rc = audio1575_set_port(statep,
-			    AUDIO_PLAY, statep->m1575_output_port);
-		}
-
-		break;
-
-	case AM_MIC_BOOST:
-		/*
-		 * Enable or disable the mic's 20 dB boost preamplifier.
-		 *
-		 * 	dir ---> N/A
-		 * 	arg1 --> ~0 == enable, 0 == disabled
-		 * 	arg2 --> not used
-		 */
-		if (arg1) {	/* enable */
-			(void) audio1575_or_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER, MICVR_20dB_BOOST);
-			statep->m1575_ad_info.ad_add_mode |=
-			    AM_ADD_MODE_MIC_BOOST;
-		} else {	/* disable */
-			(void) audio1575_and_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER,
-			    (uint16_t)~MICVR_20dB_BOOST);
-			statep->m1575_ad_info.ad_add_mode &=
-			    ~AM_ADD_MODE_MIC_BOOST;
-		}
-
-		rc = AUDIO_SUCCESS;
-		break;
-
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 *
-		 * AM_SET_GAIN_BAL
-		 * AM_SET_MONO_MIC
-		 * AM_BASS_BOOST
-		 * AM_MID_BOOST
-		 * AM_TREBLE_BOOST
-		 * AM_LOUDNESS
-		 */
-		ATRACE("audio1575_ad_set_config() unsupported cmd", command);
-		break;
-	}
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ATRACE("audio1575_ad_set_config() returning", rc);
-
-	return (rc);
-
-}	/* audio1575_ad_set_config() */
-
-/*
- * audio1575_ad_set_format()
- *
- * Description:
- *	This routine is used to set a new audio control data format.
- *	We only support 16 bit signed linear.
- *
- * Arguments:
- * 	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD
- *	int		sample_rate	Data sample rate
- *	int		channels	Number of channels, 1 or 2
- *	int		precision	Bits per sample, 16
- *	int		encoding	Encoding method, linear
- *
- * Returns:
- *	AUDIO_SUCCESS	The Codec data format has been set
- *	AUDIO_FAILURE	The Codec data format has not been set, or the
- *			data format couldn't be set
- */
-static int
-audio1575_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-    int sample_rate, int channels, int precision, int encoding)
-{
-	audio1575_state_t	*statep;
-	int			rc = AUDIO_FAILURE;
-
-	ASSERT(precision == AUDIO_PRECISION_16);
-	ASSERT(channels == AUDIO_CHANNELS_STEREO);
-
-	ATRACE("audio1575_ad_set_format() entering", stream);
-	ATRACE("audio1575_ad_set_format() dir", dir);
-	ATRACE("audio1575_ad_set_format() sample_rate", sample_rate);
-	ATRACE("audio1575_ad_set_format() channels", channels);
-	ATRACE("audio1575_ad_set_format() precision", precision);
-	ATRACE("audio1575_ad_set_format() encoding", encoding);
-
-	if (encoding != AUDIO_ENCODING_LINEAR) {
-		ATRACE("audio1575_ad_set_format() bad encoding", encoding);
-		return (rc);
-	}
-
-	/* get the soft state structure */
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	switch (sample_rate) {
-	case M1575_SAMPR8000:
-	case M1575_SAMPR9600:
-	case M1575_SAMPR11025:
-	case M1575_SAMPR16000:
-	case M1575_SAMPR18900:
-	case M1575_SAMPR22050:
-	case M1575_SAMPR27420:
-	case M1575_SAMPR32000:
-	case M1575_SAMPR33075:
-	case M1575_SAMPR37800:
-	case M1575_SAMPR44100:
-	case M1575_SAMPR48000:
-		rc = AUDIO_SUCCESS;
-		break;
-
-	default:
-		ATRACE("audio1575_ad_set_format() bad sample rate",
-		    sample_rate);
-		goto done;
-	}
-
-	switch (dir) {
-	case AUDIO_PLAY:
-		rc = audio1575_write_ac97(statep,
-		    AC97_EXTENDED_FRONT_DAC_RATE_REGISTER, sample_rate);
-		statep->m1575_psample_rate = sample_rate;
-		statep->m1575_pchannels = channels;
-		statep->m1575_pprecision = precision;
-		break;
-
-	case AUDIO_RECORD:
-		rc = audio1575_write_ac97(statep,
-		    AC97_EXTENDED_LR_DAC_RATE_REGISTER, sample_rate);
-		statep->m1575_csample_rate = sample_rate;
-		statep->m1575_cchannels = channels;
-		statep->m1575_cprecision = precision;
-		break;
-
-	default:
-		rc = AUDIO_FAILURE;
-		ATRACE("audio1575_set_format() bad audio dir", dir);
-		break;
-	}
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ATRACE("audio1575_ad_set_format() returning", rc);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	return (rc);
-
-}	/* audio1575_ad_set_format() */
-
-/*
- * audio1575_ad_start_play()
- *
- * Description:
- *	This routine starts the playback DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Play not started/restarted, no audio to play
- */
-static int
-audio1575_ad_start_play(audiohdl_t ahandle, int stream)
-{
-	audio1575_state_t	*statep;
-	m1575_sample_buf_t	*buf;
-	uint8_t			pcmocr;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE("audio1575_ad_start_play() stream", stream);
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* get ptr to play buffer */
-	buf = &statep->m1575_play_buf;
-
-	/* we are already PLAYING audio */
-	if (statep->m1575_flags & M1575_DMA_PLAY_STARTED) {
-		rc = AUDIO_SUCCESS;
-		goto done;
-	}
-
-	/* If play was PAUSED then Start Playing */
-	if (statep->m1575_flags & M1575_DMA_PLAY_PAUSED) {
-		if ((audio1575_dma_resume(statep, M1575_DMA_PCM_OUT)) ==
-		    AUDIO_FAILURE) {
-			statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-			goto done;
-		}
-
-		statep->m1575_flags &= ~M1575_DMA_PLAY_PAUSED;
-		statep->m1575_flags |= M1575_DMA_PLAY_STARTED;
-		rc = AUDIO_SUCCESS;
-		goto done;
-	}
-
-	/* we are here for the first time to play audio */
-	if (!buf->io_started) {
-		/* Prepare the DMA Engine */
-		if ((audio1575_dma_pause(statep, M1575_DMA_PCM_OUT)) ==
-		    AUDIO_FAILURE) {
-			statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-			goto done;
-		}
-		if ((audio1575_dma_reset(statep, M1575_DMA_PCM_OUT)) ==
-		    AUDIO_FAILURE) {
-			statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-			goto done;
-		}
-		/* set last valid index to zero	*/
-		M1575_AM_PUT8(M1575_PCMOLVIV_REG, 0);
-
-		/* set the buffer base */
-		M1575_AM_PUT32(M1575_PCMOBDBAR_REG,
-		    statep->m1575_bdl_phys_pout);
-		buf->head = 0;
-		buf->tail = 0;
-		buf->avail = M1575_PLAY_BUFS;
-
-		/* Set the IOCE bit */
-		pcmocr = M1575_AM_GET8(M1575_PCMOCR_REG);
-		pcmocr |= M1575_PCMOCR_IOCE;
-		M1575_AM_PUT8(M1575_PCMOCR_REG, pcmocr);
-	}
-
-	if ((audio1575_fill_play_buf(statep)) ==
-	    AUDIO_FAILURE) {
-		statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-	} else {
-		buf->io_started = B_TRUE;
-		statep->m1575_flags |= M1575_DMA_PLAY_STARTED;
-		rc = AUDIO_SUCCESS;
-	}
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_ad_start_play() returning", rc);
-
-	return (rc);
-
-}	/* audio1575_ad_start_play() */
-
-/*
- * audio1575_ad_pause_play()
- *
- * Description:
- *	This routine pauses the play DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- * 	void
- */
-static void
-audio1575_ad_pause_play(audiohdl_t ahandle, int stream)
-{
-	audio1575_state_t	*statep;
-	uint32_t		dmacr;
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ATRACE("audio1575_ad_pause_play() entering", ahandle);
-	ATRACE("audio1575_ad_pause_play() stream", stream);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* do nothing if not running */
-	if ((statep->m1575_flags & M1575_DMA_PLAY_STARTED) == 0) {
-		dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-		if ((dmacr & M1575_DMACR_PCMOSTART) !=
-		    M1575_DMACR_PCMOSTART) {
-			ATRACE("audio1575_pause_play() DMA engine "
-			    "already stopped", statep);
-			goto done;
-		}
-	}
-
-	/* Stop the DMA and set DMA pause flag */
-	if ((audio1575_dma_pause(statep, M1575_DMA_PCM_OUT)) ==
-	    AUDIO_FAILURE) {
-		statep->m1575_flags &= ~M1575_DMA_PLAY_PAUSED;
-		ATRACE("audio1575_ad_pause_play() failure", NULL);
-		goto done;
-	}
-
-	statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-	statep->m1575_flags |= M1575_DMA_PLAY_PAUSED;
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ATRACE("audio1575_ad_pause_play() returning", NULL);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-}	/* audio1575_ad_pause_play() */
-
-/*
- * audio1575_ad_stop_play()
- *
- * Description:
- *	This routine stops the playback DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream Codecs,
- *					which is not how we program the device
- *					for now.
- *
- * Returns:
- *	void
- */
-static void
-audio1575_ad_stop_play(audiohdl_t ahandle, int stream)
-{
-	audio1575_state_t	*statep;
-	m1575_sample_buf_t	*buf;
-
-	ATRACE("audio1575_ad_stop_play() entering", ahandle);
-	ATRACE("audio1575_ad_stop_play() stream", stream);
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* reset the DMA Play engine */
-	if ((audio1575_dma_reset(statep, M1575_DMA_PCM_OUT)) ==
-	    AUDIO_FAILURE) {
-		ATRACE("audio1575_ad_stop_play() failure", NULL);
-		goto done;
-	}
-
-	/* Clear the PCM Out Control Register */
-	M1575_AM_PUT8(M1575_PCMOCR_REG, 0);
-
-	/* clear the play started and paused flags */
-	buf = &statep->m1575_play_buf;
-	buf->io_started = B_FALSE;
-	statep->m1575_flags &= ~(M1575_DMA_PLAY_STARTED | \
-	    M1575_DMA_PLAY_PAUSED | M1575_DMA_PLAY_EMPTY);
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_ad_stop_play() returning", NULL);
-
-}	/* audio1575_ad_stop_play() */
-
-/*
- * audio1575_ad_start_record()
- *
- * Description:
- *	This routine starts the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which isn't going to apply for record
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Record not started
- */
-static int
-audio1575_ad_start_record(audiohdl_t ahandle, int stream)
-{
-	audio1575_state_t	*statep;
-	m1575_sample_buf_t	*buf;
-	uint8_t			pcmicr;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE("audio1575_ad_start_record() entering", ahandle);
-	ATRACE("audio1575_ad_start_record() stream", stream);
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* get our record buffer ptr */
-	buf = &statep->m1575_record_buf;
-
-	if (statep->m1575_flags & M1575_DMA_RECD_STARTED) {
-		rc = AUDIO_SUCCESS;
-		goto done;
-	}
-
-	/* Prepare DMA engine */
-	if ((audio1575_dma_reset(statep, M1575_DMA_PCM_IN)) ==
-	    AUDIO_FAILURE) {
-		statep->m1575_flags &= ~M1575_DMA_RECD_STARTED;
-		goto done;
-	}
-
-	/* DMA Engine reset was successful */
-	buf->head = 0;
-	buf->tail = 0;
-	buf->avail = M1575_REC_BUFS;
-
-	/* Set the IOCE bit */
-	pcmicr = M1575_AM_GET8(M1575_PCMICR_REG);
-	pcmicr |= M1575_PCMICR_IOCE;
-	M1575_AM_PUT8(M1575_PCMICR_REG, pcmicr);
-
-	/* set last valid index to 0 */
-	M1575_AM_PUT8(M1575_PCMILVIV_REG, 0);
-
-	/* setup the Base Address Register */
-	M1575_AM_PUT32(M1575_PCMIBDBAR_REG, statep->m1575_bdl_phys_pin);
-
-	if ((audio1575_prepare_record_buf(statep)) ==
-	    AUDIO_SUCCESS) {
-		statep->m1575_flags |= M1575_DMA_RECD_STARTED;
-		rc = AUDIO_SUCCESS;
-	} else {
-		statep->m1575_flags &= ~M1575_DMA_RECD_STARTED;
-	}
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_ad_start_record() returning", rc);
-
-	return (rc);
-
-}	/* audio1575_ad_start_record() */
-
-/*
- * audio1575_ad_stop_record()
- *
- * Description:
- *	This routine stops the PCM in DMA engine
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle for this driver
- *	int		stream		Stream number for multi-stream
- *					Codecs, which isn't going to apply
- *					for record
- *
- * Returns:
- *	void
- */
-static void
-audio1575_ad_stop_record(audiohdl_t ahandle, int stream)
-{
-	audio1575_state_t	*statep;
-	m1575_sample_buf_t	*buf;
-
-	ATRACE("audio1575_ad_stop_record() entering", ahandle);
-	ATRACE("audio1575_ad_stop_record() stream", stream);
-
-	statep = audio_sup_get_private(ahandle);
-	ASSERT(statep);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	/* reset the DMA input registers */
-	if ((audio1575_dma_reset(statep, M1575_DMA_PCM_IN)) ==
-	    AUDIO_FAILURE) {
-		ATRACE("audio1575_ad_stop_record() failure", NULL);
-		goto done;
-	}
-
-	/* clear the PCM In Control Register */
-	M1575_AM_PUT8(M1575_PCMICR_REG, 0);
-
-	/* clear the record started flag */
-	statep->m1575_flags &= ~M1575_DMA_RECD_STARTED;
-	buf = &statep->m1575_record_buf;
-	buf->io_started = B_FALSE;
-
-done:
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	ASSERT(!mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_ad_stop_record() returning", NULL);
-
-}	/* audio1575_ad_stop_record() */
-
-
-/* *********************** Local Routines *************************** */
-
-/*
- * audio1575_init_state()
- *
- * Description:
- *	This routine initializes the audio driver's state structure
- *
- *	CAUTION: This routine cannot allocate resources, unless it frees
- *		them before returning from an error. Also, error_destroy:
- *		in audio1575_attach() would need to be fixed as well.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	dev_info_t		*dip		Pointer to the device's
- *						dev_info struct
- * Returns:
- *	AUDIO_SUCCESS		State structure initialized
- *	AUDIO_FAILURE		State structure not initialized
- */
-static int
-audio1575_init_state(audio1575_state_t *statep, dev_info_t *dip)
-{
-	int 		rints;
-	int 		pints;
-	int		count = 0;
-	int		actual = 0;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("audio1575_init_state() entering", NULL);
-
-	statep->m1575_dip = dip;
-
-	/* default to 5 bit volume */
-	statep->m1575_vol_bits_mask = 5;
-
-	/* see if we support internal CDROM */
-	statep->m1575_cdrom = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "cdrom", 0);
-
-	/* get the mode from the .conf file */
-	if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "mixer-mode", AM_MIXER_MODE)) {
-		statep->m1575_mode = AM_MIXER_MODE;
-	} else {
-		statep->m1575_mode = AM_COMPAT_MODE;
-	}
-
-	/* get play interrupts */
-	pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "play-interrupts", M1575_INTS);
-	if (pints > M1575_MAX_INTS) {
-		ATRACE("audio1575_init_state() "
-		    "play interrupt rate too high, resetting", pints);
-		audio_sup_log(statep->m1575_ahandle, CE_NOTE,
-		    "init_state() "
-		    "play interrupt rate set too high, %d, resetting to %d",
-		    pints, M1575_INTS);
-		pints = M1575_INTS;
-	} else if (pints < M1575_MIN_INTS) {
-		ATRACE("audio1575_init_state() "
-		    "play interrupt rate too low, resetting", pints);
-		audio_sup_log(statep->m1575_ahandle, CE_NOTE,
-		    "init_state() "
-		    "play interrupt rate set too low, %d, resetting to %d",
-		    pints, M1575_INTS);
-		pints = M1575_INTS;
-	}
-
-	/* get record interrupts */
-	rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "record-interrupts", M1575_INTS);
-	if (rints > M1575_MAX_INTS) {
-		ATRACE("audio1575_init_state() "
-		    "record interrupt rate too high, resetting", rints);
-		audio_sup_log(statep->m1575_ahandle, CE_NOTE,
-		    "init_state() "
-		    "record interrupt rate set too high, %d, resetting to %d",
-		    rints, M1575_INTS);
-		rints = M1575_INTS;
-	} else if (rints < M1575_MIN_INTS) {
-		ATRACE("audio1575_init_state() "
-		    "record interrupt rate too low, resetting", rints);
-		audio_sup_log(statep->m1575_ahandle, CE_NOTE,
-		    "init_state() "
-		    "record interrupt rate set too low, %d "
-		    "resetting to %d", rints, M1575_INTS);
-		rints = M1575_INTS;
-	}
-
-	/* get supported interrupt types */
-	rc = ddi_intr_get_supported_types(dip, &statep->m1575_intr_type);
-
-	if ((rc != DDI_SUCCESS) ||
-	    (!(statep->m1575_intr_type & DDI_INTR_TYPE_FIXED))) {
-		audio_sup_log(statep->m1575_ahandle,
-		    CE_WARN, "!init_state() Fixed type interrupts not "
-		    "supported");
-		ATRACE("audio1575_init_state() fixed type INTR not supported",
-		    NULL);
-		goto error_intr;
-	}
-
-	/* make sure we only have one fixed type interrupt */
-	rc = ddi_intr_get_nintrs(dip, DDI_INTR_TYPE_FIXED, &count);
-
-	if ((rc != DDI_SUCCESS) || (count != 1)) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!init_state() no fixed interrupts");
-		ATRACE("audio1575_init_state() no fixed interrupts",
-		    NULL);
-		goto error_intr;
-	}
-
-	/* allocate interrupt table */
-	statep->m1575_h_table = kmem_zalloc(sizeof (ddi_intr_handle_t),
-	    KM_SLEEP);
-
-	rc = ddi_intr_alloc(dip, statep->m1575_h_table, DDI_INTR_TYPE_FIXED, 0,
-	    count, &actual, DDI_INTR_ALLOC_NORMAL);
-	if ((rc != DDI_SUCCESS) || (actual != 1)) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!init_state() ddi_intr_alloc() failure");
-		ATRACE("audio1575_init_state() ddi_intr_alloc failure", actual);
-		goto error_alloc;
-	}
-
-	ASSERT(count == actual);
-	/* Get the interrupt priority for initializing the m1575_intr_mutex */
-	if ((ddi_intr_get_pri(statep->m1575_h_table[0],
-	    &statep->m1575_intr_pri)) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!init_state() ddi_intr_get_pri() failure");
-		ATRACE("audio1575_init_state() ddi_intr_get_pri() failure",
-		    NULL);
-		goto error_free;
-	}
-
-	/* test for a high level interrupt */
-	if (statep->m1575_intr_pri >= ddi_intr_get_hilevel_pri()) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!init_state() unsupported high level interrupt");
-		ATRACE("audio1575_init_state() high level interrupt not "
-		    "supported", NULL);
-		goto error_free;
-	}
-
-	/* fill in the device default state */
-	statep->m1575_defaults.play.sample_rate = M1575_DEFAULT_SR;
-	statep->m1575_defaults.play.channels = M1575_DEFAULT_CH;
-	statep->m1575_defaults.play.precision = M1575_DEFAULT_PREC;
-	statep->m1575_defaults.play.encoding = M1575_DEFAULT_ENC;
-	statep->m1575_defaults.play.gain = M1575_DEFAULT_PGAIN;
-	statep->m1575_defaults.play.port = AUDIO_SPEAKER;
-	statep->m1575_defaults.play.avail_ports =
-	    AUDIO_SPEAKER | AUDIO_LINE_OUT | AUDIO_HEADPHONE;
-	statep->m1575_defaults.play.mod_ports =
-	    AUDIO_SPEAKER | AUDIO_LINE_OUT | AUDIO_HEADPHONE;
-	statep->m1575_defaults.play.buffer_size = M1575_BSIZE;
-	statep->m1575_defaults.play.balance = M1575_DEFAULT_BAL;
-	statep->m1575_defaults.record.sample_rate = M1575_DEFAULT_SR;
-	statep->m1575_defaults.record.channels = M1575_DEFAULT_CH;
-	statep->m1575_defaults.record.precision = M1575_DEFAULT_PREC;
-	statep->m1575_defaults.record.encoding = M1575_DEFAULT_ENC;
-	statep->m1575_defaults.record.gain = M1575_DEFAULT_RGAIN;
-	statep->m1575_defaults.record.port = AUDIO_MICROPHONE;
-	statep->m1575_defaults.record.avail_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->m1575_defaults.record.mod_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	statep->m1575_defaults.record.buffer_size = M1575_BSIZE;
-	statep->m1575_defaults.record.balance = M1575_DEFAULT_BAL;
-
-	statep->m1575_defaults.monitor_gain = M1575_DEFAULT_MONITOR_GAIN;
-	statep->m1575_defaults.output_muted = B_FALSE;
-	statep->m1575_defaults.ref_cnt = B_FALSE;
-	statep->m1575_defaults.hw_features =
-	    AUDIO_HWFEATURE_DUPLEX|AUDIO_HWFEATURE_PLAY|
-	    AUDIO_HWFEATURE_IN2OUT|AUDIO_HWFEATURE_RECORD;
-	statep->m1575_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	if (statep->m1575_cdrom) {
-		statep->m1575_defaults.record.avail_ports |= AUDIO_CD;
-		statep->m1575_defaults.record.mod_ports |= AUDIO_CD;
-	}
-
-	statep->m1575_psample_rate = statep->m1575_defaults.play.sample_rate;
-	statep->m1575_pchannels = statep->m1575_defaults.play.channels;
-	statep->m1575_pprecision = statep->m1575_defaults.play.precision;
-	statep->m1575_csample_rate = statep->m1575_defaults.record.sample_rate;
-	statep->m1575_cchannels = statep->m1575_defaults.record.channels;
-	statep->m1575_cprecision = statep->m1575_defaults.record.precision;
-
-	/*
-	 * fill in the ad_info structure
-	 */
-	statep->m1575_ad_info.ad_mode = statep->m1575_mode;
-	statep->m1575_ad_info.ad_int_vers = AM_VERSION;
-	statep->m1575_ad_info.ad_add_mode = NULL;
-	statep->m1575_ad_info.ad_codec_type = AM_TRAD_CODEC;
-	statep->m1575_ad_info.ad_defaults = &statep->m1575_defaults;
-	statep->m1575_ad_info.ad_play_comb = audio1575_combinations;
-	statep->m1575_ad_info.ad_rec_comb = audio1575_combinations;
-	statep->m1575_ad_info.ad_entry = &audio1575_entry;
-	statep->m1575_ad_info.ad_dev_info = &statep->m1575_dev_info;
-	statep->m1575_ad_info.ad_diag_flags = AM_DIAG_INTERNAL_LOOP;
-	statep->m1575_ad_info.ad_diff_flags =
-	    AM_DIFF_SR | AM_DIFF_CH | AM_DIFF_PREC | AM_DIFF_ENC;
-	statep->m1575_ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	statep->m1575_ad_info.ad_misc_flags = AM_MISC_RP_EXCL;
-	statep->m1575_ad_info.ad_misc_flags |= AM_MISC_MONO_DUP;
-	statep->m1575_ad_info.ad_translate_flags = AM_MISC_8_R_TRANSLATE;
-	statep->m1575_ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	statep->m1575_ad_info.ad_play.ad_mixer_srs =
-	    audio1575_mixer_sample_rates;
-	statep->m1575_ad_info.ad_play.ad_compat_srs =
-	    audio1575_compat_sample_rates;
-	statep->m1575_ad_info.ad_play.ad_conv = &am_src2;
-	statep->m1575_ad_info.ad_play.ad_sr_info = NULL;
-	statep->m1575_ad_info.ad_play.ad_chs = audio1575_channels;
-	statep->m1575_ad_info.ad_play.ad_int_rate = pints;
-	statep->m1575_ad_info.ad_play.ad_max_chs = M1575_MAX_OUT_CHANNELS;
-	statep->m1575_ad_info.ad_play.ad_bsize = M1575_BSIZE;
-
-	/* record capabilities */
-	statep->m1575_ad_info.ad_record.ad_mixer_srs =
-	    audio1575_mixer_sample_rates;
-	statep->m1575_ad_info.ad_record.ad_compat_srs =
-	    audio1575_compat_sample_rates;
-	statep->m1575_ad_info.ad_record.ad_conv = &am_src2;
-	statep->m1575_ad_info.ad_record.ad_sr_info = NULL;
-	statep->m1575_ad_info.ad_record.ad_chs = audio1575_channels;
-	statep->m1575_ad_info.ad_record.ad_int_rate = rints;
-	statep->m1575_ad_info.ad_record.ad_max_chs = M1575_MAX_CHANNELS;
-	statep->m1575_ad_info.ad_record.ad_bsize = M1575_BSIZE;
-
-	/* fill in device info strings */
-	(void) strcpy(statep->m1575_dev_info.name, M1575_DEV_NAME);
-	(void) strcpy(statep->m1575_dev_info.config, M1575_DEV_CONFIG);
-	(void) strcpy(statep->m1575_dev_info.version, M1575_DEV_VERSION);
-
-	/* compute play and record buffer sizes */
-	statep->m1575_play_buf_size = M1575_SAMPR48000 *
-	    AUDIO_CHANNELS_STEREO * (AUDIO_PRECISION_16 >>
-	    AUDIO_PRECISION_SHIFT) / pints;
-	statep->m1575_play_buf_size += M1575_MOD_SIZE -
-	    (statep->m1575_play_buf_size % M1575_MOD_SIZE);
-	statep->m1575_record_buf_size = M1575_SAMPR48000 *
-	    AUDIO_CHANNELS_STEREO * (AUDIO_PRECISION_16 >>
-	    AUDIO_PRECISION_SHIFT) / rints;
-	statep->m1575_record_buf_size += M1575_MOD_SIZE -
-	    (statep->m1575_record_buf_size % M1575_MOD_SIZE);
-
-	statep->m1575_flags = 0;
-
-	ATRACE("audio1575_init_state() returning", NULL);
-
-	return (AUDIO_SUCCESS);
-
-error_free:
-	(void) ddi_intr_free(statep->m1575_h_table[0]);
-error_alloc:
-	kmem_free(statep->m1575_h_table, sizeof (ddi_intr_handle_t));
-error_intr:
-	ATRACE("audio1575_init_state() failure", NULL);
-
-	return (AUDIO_FAILURE);
-
-}	/* audio1575_init_state() */
-
-/*
- * audio1575_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. Finally, the registers are mapped in.
- *
- *	CAUTION: Make sure all errors call audio_sup_log().
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-audio1575_map_regs(dev_info_t *dip, audio1575_state_t *statep)
-{
-	ddi_dma_cookie_t	cookie;
-	uint_t			count;
-
-	ATRACE("in audio1575_map_regs() entering", statep);
-
-	statep->m1575_res_flags = 0;
-
-	/* Check for fault management capabilities */
-	if (DDI_FM_ACC_ERR_CAP(ddi_fm_capable(dip))) {
-		dev_attr.devacc_attr_access = DDI_FLAGERR_ACC;
-	}
-
-	/* map the M1575 Audio PCI Cfg Space */
-	if ((ddi_regs_map_setup(dip, M1575_AUDIO_PCICFG_SPACE,
-	    (caddr_t *)&statep->m1575_pci_regs, 0, 0, &dev_attr,
-	    &statep->m1575_pci_regs_handle)) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() PCI Config mapping failure");
-		ATRACE("audio1575_map_regs() PCI Cfg Space failure", NULL);
-		goto error;
-	}
-	statep->m1575_res_flags |= M1575_RS_PCI_REGS;
-
-	/* map the M1575 Audio registers in PCI IO Space */
-	if ((ddi_regs_map_setup(dip, M1575_AUDIO_IO_SPACE,
-	    (caddr_t *)&statep->m1575_am_regs, 0, 0, &dev_attr,
-	    &statep->m1575_am_regs_handle)) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() Audio IO mapping failure");
-		ATRACE("audio1575_map_regs() PCI IO Space failure", NULL);
-		goto error;
-	}
-	statep->m1575_res_flags |= M1575_RS_AM_REGS;
-
-	/* map the M1575 Audio registers in PCI MEM32 Space */
-	if ((ddi_regs_map_setup(dip, M1575_AUDIO_MEM_SPACE,
-	    (caddr_t *)&statep->m1575_bm_regs, 0, 0, &dev_attr,
-	    &statep->m1575_bm_regs_handle)) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() Audio MEM32 mapping failure");
-		ATRACE("audio1575_map_regs() PCI MEM32 Space failure", NULL);
-		goto error;
-	}
-	statep->m1575_res_flags |= M1575_RS_BM_REGS;
-
-	/*
-	 * Here we allocate DMA memory for the buffer descriptor list.
-	 * we allocate adjacent DMA memory for all DMA engines.
-	 */
-	if (ddi_dma_alloc_handle(dip, &bdlist_dma_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &statep->m1575_bdl_dma_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() ddi_dma_alloc_handle "
-		    "(bdlist) failure");
-		ATRACE("audio1575_map_regs() ddi_dma_alloc_handle() failure",
-		    NULL);
-		goto error;
-	}
-	statep->m1575_res_flags |= M1575_RS_DMA_BDL_HANDLE;
-
-	/*
-	 * we allocate all buffer descriptors lists in contiguous dma memory.
-	 */
-	if (ddi_dma_mem_alloc(statep->m1575_bdl_dma_handle,
-	    sizeof (m1575_bd_entry_t) * M1575_BD_NUMS * 2,
-	    &dev_attr, DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    (caddr_t *)&statep->m1575_bdl_virtual, &statep->m1575_bdl_size,
-	    &statep->m1575_bdl_acc_handle) != DDI_SUCCESS) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() ddi_dma_mem_alloc(bdlist) failure");
-		ATRACE("audio1575_map_regs() ddi_dma_mem_alloc(bdlist) failure",
-		    NULL);
-		goto error;
-	}
-	statep->m1575_res_flags |= M1575_RS_DMA_BDL_MEM;
-
-	if (ddi_dma_addr_bind_handle(statep->m1575_bdl_dma_handle, NULL,
-	    (caddr_t)statep->m1575_bdl_virtual, statep->m1575_bdl_size,
-	    DDI_DMA_RDWR|DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL, &cookie,
-	    &count) != DDI_DMA_MAPPED) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!audio1575_map_regs() addr_bind_handle failure");
-		ATRACE("audio1575_map_regs() ddi_dma_addr_bind_handle failure",
-		    NULL);
-		goto error;
-	}
-
-	statep->m1575_bdl_virt_pin =
-	    (m1575_bd_entry_t *)(statep->m1575_bdl_virtual);
-	statep->m1575_bdl_virt_pout =
-	    statep->m1575_bdl_virt_pin + M1575_BD_NUMS;
-	statep->m1575_bdl_phys_pin = (uint32_t)(cookie.dmac_address);
-	statep->m1575_bdl_phys_pout = statep->m1575_bdl_phys_pin +
-	    sizeof (m1575_bd_entry_t) * M1575_BD_NUMS;
-	statep->m1575_res_flags |= M1575_RS_DMA_BDL_BIND;
-
-	ATRACE("audio1575_map_regs() returning", AUDIO_SUCCESS);
-
-	return (AUDIO_SUCCESS);
-
-error:
-	audio1575_unmap_regs(statep);
-
-	ATRACE("audio1575_map_regs() returning AUDIO_FAILURE", NULL);
-
-	return (AUDIO_FAILURE);
-
-}	/* audio1575_map_regs() */
-
-/*
- * audio1575_unmap_regs()
- *
- * Description:
- *	This routine unbinds the play and record DMA handles, frees
- *	the DMA buffers and the unmaps control registers.
- *
- * Arguments:
- *	audio1575_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_unmap_regs(audio1575_state_t *statep)
-{
-	if (statep->m1575_res_flags & M1575_RS_DMA_BDL_BIND) {
-		statep->m1575_res_flags &= ~M1575_RS_DMA_BDL_BIND;
-		(void) ddi_dma_unbind_handle(statep->m1575_bdl_dma_handle);
-	}
-
-	if (statep->m1575_res_flags & M1575_RS_DMA_BDL_MEM) {
-		statep->m1575_res_flags &= ~M1575_RS_DMA_BDL_MEM;
-		ddi_dma_mem_free(&statep->m1575_bdl_acc_handle);
-	}
-
-	if (statep->m1575_res_flags & M1575_RS_DMA_BDL_HANDLE) {
-		statep->m1575_res_flags &= ~M1575_RS_DMA_BDL_HANDLE;
-		ddi_dma_free_handle(&statep->m1575_bdl_dma_handle);
-	}
-
-	if (statep->m1575_res_flags & M1575_RS_BM_REGS) {
-		statep->m1575_res_flags &= ~M1575_RS_BM_REGS;
-		ddi_regs_map_free(&statep->m1575_bm_regs_handle);
-	}
-
-	if (statep->m1575_res_flags & M1575_RS_AM_REGS) {
-		statep->m1575_res_flags &= ~M1575_RS_AM_REGS;
-		ddi_regs_map_free(&statep->m1575_am_regs_handle);
-	}
-
-	if (statep->m1575_res_flags & M1575_RS_PCI_REGS) {
-		statep->m1575_res_flags &= ~M1575_RS_PCI_REGS;
-		pci_config_teardown(&statep->m1575_pci_regs_handle);
-	}
-
-}	/* audio1575_unmap_regs() */
-
-/*
- * audio1575_alloc_sample_buf()
- *
- * Description:
- *	This routine allocates DMA buffers for the sample buffer. It
- *	allocates two DMA chunks (buffers) to the specified DMA engine
- *	(sample buffer structure). The two data chunks will be bound
- *	to the buffer descriptor entries of corresponding buffer
- *	descriptor list, and be used to transfer audio sample data to
- *	and from the audio controller.
- *
- * Arguments:
- *	audio1575_state_t	*state	The device's state structure
- *	int			which	Which sample buffer, PCM in or PCM out
- *					M1575_DMA_PCM_IN ---PCM in DMA engine
- *					M1575_DMA_PCM_OUT---PCM out DMA engine
- *	int			len	The length of the DMA buffers
- *
- * Returns:
- *	AUDIO_SUCCESS	 Allocating DMA buffers successfully
- *	AUDIO_FAILURE	 Failed to allocate dma buffers
- */
-static int
-audio1575_alloc_sample_buf(audio1575_state_t *statep, int which, int len)
-{
-	m1575_sample_buf_t	*buf;
-	m1575_bdlist_chunk_t	*chunk;
-	ddi_dma_cookie_t	cookie;
-	uint_t			count;
-	int			i;
-	int			j;
-	int			handle_cnt = 0;
-	int			buf_cnt = 0;
-	int			bind_cnt = 0;
-
-	ATRACE("audio1575_alloc_sample_buf() entering", which);
-
-	switch (which) {
-	case M1575_DMA_PCM_OUT:
-		buf = &statep->m1575_play_buf;
-		for (i = 0; i < M1575_PLAY_BUFS; i++) {
-			chunk = &(buf->chunk[i]);
-			if (ddi_dma_alloc_handle(statep->m1575_dip,
-			    &sample_buf_dma_attr,
-			    DDI_DMA_SLEEP, NULL,
-			    &chunk->dma_handle) != DDI_SUCCESS) {
-				goto error;
-			}
-			handle_cnt++;
-
-			if (ddi_dma_mem_alloc(chunk->dma_handle, len,
-			    &dev_attr, DDI_DMA_STREAMING, DDI_DMA_SLEEP,
-			    NULL, &chunk->data_buf,
-			    &chunk->real_len, &chunk->acc_handle) !=
-			    DDI_SUCCESS) {
-				goto error;
-			}
-			buf_cnt++;
-
-			if (ddi_dma_addr_bind_handle(chunk->dma_handle,
-			    NULL, chunk->data_buf, chunk->real_len,
-			    DDI_DMA_WRITE|DDI_DMA_STREAMING,
-			    DDI_DMA_SLEEP, NULL, &cookie, &count) !=
-			    DDI_DMA_MAPPED) {
-				goto error;
-			}
-			bind_cnt++;
-
-			chunk->addr_phy = (uint32_t)cookie.dmac_address;
-		}
-		break;
-
-	case M1575_DMA_PCM_IN:
-		buf = &statep->m1575_record_buf;
-		for (i = 0; i < M1575_REC_BUFS; i++) {
-			chunk = &(buf->chunk[i]);
-			if (ddi_dma_alloc_handle(statep->m1575_dip,
-			    &sample_buf_dma_attr, DDI_DMA_SLEEP,
-			    NULL, &chunk->dma_handle) !=
-			    DDI_SUCCESS) {
-				goto error;
-			}
-			handle_cnt++;
-
-			if (ddi_dma_mem_alloc(chunk->dma_handle, len,
-			    &dev_attr, DDI_DMA_STREAMING, DDI_DMA_SLEEP,
-			    NULL, &chunk->data_buf, &chunk->real_len,
-			    &chunk->acc_handle) != DDI_SUCCESS) {
-				goto error;
-			}
-			buf_cnt++;
-
-			if (ddi_dma_addr_bind_handle(chunk->dma_handle, NULL,
-			    chunk->data_buf, chunk->real_len,
-			    DDI_DMA_READ|DDI_DMA_STREAMING, DDI_DMA_SLEEP,
-			    NULL, &cookie, &count) != DDI_DMA_MAPPED) {
-				goto error;
-			}
-			bind_cnt++;
-
-			chunk->addr_phy = (uint32_t)cookie.dmac_address;
-		}
-		break;
-
-	default:
-		ATRACE("audio1575_alloc_sample_buf() unknown buffer type",
-		    which);
-
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("audio1575_alloc_sample_buf() returning", AUDIO_SUCCESS);
-
-	return (AUDIO_SUCCESS);
-
-error:
-	for (j = 0; j < bind_cnt; j++) {
-		(void) ddi_dma_unbind_handle((buf->chunk[j].dma_handle));
-	}
-	for (j = 0; j < buf_cnt; j++) {
-		ddi_dma_mem_free(&(buf->chunk[j].acc_handle));
-	}
-	for (j = 0; j < handle_cnt; j++) {
-		ddi_dma_free_handle(&(buf->chunk[j].dma_handle));
-	}
-
-	ATRACE("audio1575_alloc_sample_buf() returning", AUDIO_FAILURE);
-
-	return (AUDIO_FAILURE);
-
-}	/* audio1575_alloc_sample_buf() */
-
-/*
- * audio1575_free_sample_buf()
- *
- * Description:
- *	This routine frees the DMA buffers of the sample buffer. The DMA
- *	buffers were allocated by calling audio1575_alloc_sample_buf().
- *
- * Arguments:
- *	audio1575_state_t	*state	The device's state structure
- *	int			which	Which sample buffer, PCM in or PCM out
- *
- * Returns:
- *	void
- */
-static void
-audio1575_free_sample_buf(audio1575_state_t *statep, int which)
-{
-	m1575_sample_buf_t	*buf;
-	m1575_bdlist_chunk_t	*chunk;
-	int			i;
-
-	ATRACE("audio1575_free_sample_buf() entering", statep);
-
-	switch (which) {
-	case M1575_DMA_PCM_IN:
-		buf = &statep->m1575_record_buf;
-		for (i = 0; i < M1575_REC_BUFS; i++) {
-			chunk = &(buf->chunk[i]);
-			(void) ddi_dma_unbind_handle(chunk->dma_handle);
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-		}
-		break;
-
-	case M1575_DMA_PCM_OUT:
-		buf = &statep->m1575_play_buf;
-		for (i = 0; i < M1575_PLAY_BUFS; i++) {
-			chunk = &(buf->chunk[i]);
-			(void) ddi_dma_unbind_handle(chunk->dma_handle);
-			ddi_dma_mem_free(&chunk->acc_handle);
-			ddi_dma_free_handle(&chunk->dma_handle);
-		}
-		break;
-
-	default:
-		ATRACE("audio1575_free_sample() unknown buffer type",
-		    which);
-		break;
-	}
-
-	ATRACE("audio1575_free_sample_buf() returning", NULL);
-
-}	/* audio1575_free_sample_buf() */
-
-/*
- * audio1575_reclaim_play_buf()
- *
- * Description:
- *	When the audio controller finishes fetching the data from DMA
- *	buffers, this routine will be called by interrupt handler to
- *	reclaim the DMA buffers.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_reclaim_play_buf(audio1575_state_t *statep)
-{
-	m1575_sample_buf_t	*buf;
-	int16_t			pcmociv;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	/* get the play buf ptr */
-	buf = &statep->m1575_play_buf;
-
-	/* get out current index value */
-	pcmociv	= M1575_AM_GET8(M1575_PCMOCIV_REG);
-
-	while (buf->head != pcmociv) {
-		buf->avail++;
-		buf->head++;
-		buf->head &= M1575_BD_MSK;
-	}
-
-}	/* audio1575_reclaim_play_buf() */
-
-/*
- * audio1575_init_ac97()
- *
- * Description:
- *	This routine initializes the AC97 codec.
- *	The AC97 codec registers are programmed from codec_shadow[].
- *	If we are not doing a restore, we initialize codec_shadow[], otherwise
- *	we use the current values of shadow
- *
- * Arguments:
- *	audio1575_state_t	*state	The device's state structure
- *	int			restore	If M1575_INIT_RESTORE then
- *					restore	from codec_shadow[]
- * Returns:
- *	AUDIO_SUCCESS	The hardware was initialized properly
- *	AUDIO_FAILURE	The hardware couldn't be initialized properly
- */
-static int
-audio1575_init_ac97(audio1575_state_t *statep, int restore)
-{
-	uint16_t	sr;
-	uint16_t	*shadow;
-	uint16_t	tmp;
-	uint16_t	addr;
-	int 		i;
-	clock_t		ticks;
-
-	ATRACE("audio1575_init_ac97() entering", NULL);
-
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	ticks = drv_usectohz(AD1981_POWERON_DELAY_USEC);
-
-	/* AC97 register reset */
-	if (audio1575_reset_ac97(statep) != AUDIO_SUCCESS) {
-		ATRACE("audio1575_ac97_init() codec not reset", NULL);
-		mutex_exit(&statep->m1575_intr_mutex);
-		return (AUDIO_FAILURE);
-	}
-
-	/* turn on the AD1981B codec power and wait for analog i/o be ready */
-	(void) audio1575_read_ac97(statep,
-	    AC97_POWERDOWN_CTRL_STAT_REGISTER, &sr);
-	(void) audio1575_write_ac97(statep,
-	    AC97_POWERDOWN_CTRL_STAT_REGISTER, sr & 0x00ff);
-
-	/*
-	 * Wait 1 sec for the analog section to power up
-	 * checking every 10 ms.
-	 */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		(void) audio1575_read_ac97(statep,
-		    AC97_POWERDOWN_CTRL_STAT_REGISTER, &sr);
-		if ((sr & PCSR_POWERD_UP) == PCSR_POWERD_UP) {
-			break;
-		}
-#ifndef __lock_lint
-		delay(ticks);
-#endif
-	}
-
-	/* return failure if the codec did not power up */
-	if (i >= M1575_LOOP_CTR) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!failure to power up the AC97 Codec");
-		ATRACE("audio1575_ac97_init() codec not powered up", sr);
-		mutex_exit(&statep->m1575_intr_mutex);
-		return (AUDIO_FAILURE);
-	}
-
-	/* point to our codec shadow registers */
-	shadow = statep->m1575_codec_shadow;
-
-	if (restore == M1575_INIT_NO_RESTORE) {
-		for (i = 0; i < M1575_LAST_AC_REG; i += 2) {
-			(void) audio1575_read_ac97(statep, i,
-			    &(shadow[M1575_CODEC_REG(i)]));
-		}
-
-		/* 02h - set master line out volume, muted, 0dB */
-		shadow[M1575_CODEC_REG(AC97_MASTER_VOLUME_REGISTER)] =
-		    MVR_MUTE;
-
-		/* 04h - set alternate line out volume, muted, 0dB */
-		shadow[M1575_CODEC_REG(AC97_HEADPHONE_VOLUME_REGISTER)] =
-		    HPVR_MUTE;
-
-		/* 06h - set master mono volume, muted, 0dB */
-		shadow[M1575_CODEC_REG(AC97_MONO_MASTER_VOLUME_REGSITER)] =
-		    MMVR_MUTE | MMVR_0dB_ATTEN;
-
-		/* 0ch - set phone input, mute, 0dB attenuation */
-		shadow[M1575_CODEC_REG(AC97_PHONE_VOLUME_REGISTER)] =
-		    PVR_MUTE|PVR_0dB_GAIN;
-
-		/*
-		 * 0eh - set mic input, mute, +20dB attenuation
-		 * actually this is 30dB when MICVR_20dB_Boost
-		 * is set. (see misc. register 0x76 setting
-		 * of MIC_30dB_GAIN below.)
-		 */
-		shadow[M1575_CODEC_REG(AC97_MIC_VOLUME_REGISTER)] =
-		    MICVR_MUTE | MICVR_20dB_BOOST | MICVR_0dB_GAIN;
-		statep->m1575_ad_info.ad_add_mode |= AM_ADD_MODE_MIC_BOOST;
-
-		/* 10h - set line input, mute, 0dB attenuation */
-		shadow[M1575_CODEC_REG(AC97_LINE_IN_VOLUME_REGISTER)] =
-		    LIVR_MUTE|LIVR_RIGHT_0dB_GAIN|LIVR_LEFT_0dB_GAIN;
-
-		/* 12h - set cd input, mute, 0dB attenuation */
-		shadow[M1575_CODEC_REG(AC97_CD_VOLUME_REGISTER)] =
-		    CDVR_MUTE|CDVR_RIGHT_0dB_GAIN|CDVR_LEFT_0dB_GAIN;
-
-		/* 16h - set aux input, mute, 0dB attenuation */
-		shadow[M1575_CODEC_REG(AC97_AUX_VOLUME_REGISTER)] =
-		    AUXVR_MUTE|AUXVR_RIGHT_0dB_GAIN|AUXVR_LEFT_0dB_GAIN;
-
-		/* 18h - set PCM out input, NOT muted, 0dB gain */
-		shadow[M1575_CODEC_REG(AC97_PCM_OUT_VOLUME_REGISTER)] =
-		    PCMOVR_RIGHT_0dB_GAIN|PCMOVR_LEFT_0dB_GAIN;
-
-		/* 1ah - set input device as mic */
-		shadow[M1575_CODEC_REG(AC97_RECORD_SELECT_CTRL_REGISTER)] =
-		    RSCR_R_MIC|RSCR_L_MIC;
-
-		/* 1ch - set record gain to 0dB and not muted */
-		shadow[M1575_CODEC_REG(AC97_RECORD_GAIN_REGISTER)] =
-		    RGR_RIGHT_0db_GAIN|RGR_LEFT_0db_GAIN;
-
-		/* 20h - set GP register, mic 1, everything else off */
-		shadow[M1575_CODEC_REG(AC97_GENERAL_PURPOSE_REGISTER)] =
-		    GPR_MS_MIC1|GPR_MONO_MIX_IN;
-
-		/* 28h - set to use Primary Codec channels 1 & 2 */
-		shadow[M1575_CODEC_REG(AC97_EXTENDED_AUDIO_REGISTER)] =
-		    EAR_PRIMARY_CODEC;
-
-		/* 2ah - set to use Primary Codec channels 1 & 2 */
-		shadow[M1575_CODEC_REG(AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER)]
-		    = 0;
-
-		/* 2ch - PCM Front DAC Sample Rate */
-		shadow[M1575_CODEC_REG(AC97_EXTENDED_FRONT_DAC_RATE_REGISTER)] =
-		    AC97_SAMPLE_RATE_48000;
-
-		/* 32h - PCM ADC Sample Rate */
-		shadow[M1575_CODEC_REG(AC97_EXTENDED_LR_DAC_RATE_REGISTER)] =
-		    AC97_SAMPLE_RATE_48000;
-
-		/* 64h - Mixer ADC Gain Register */
-		shadow[M1575_CODEC_REG(AC97_MIXER_ADC_GAIN_REGISTER)] =
-		    RGR_RIGHT_0db_GAIN|RGR_LEFT_0db_GAIN|RGR_MUTE;
-
-		/* 76h - Misc. Control Bit  Register */
-		shadow[M1575_CODEC_REG(AC97_MISC_CONTROL_BIT_REGISTER)] =
-		    MIC_30dB_GAIN | C1MIC;
-	}
-
-	/* Now we set the AC97 codec registers to the saved values */
-	for (i = 2; i <= M1575_LAST_AC_REG; i += 2) {
-		(void) audio1575_write_ac97(statep, i,
-		    shadow[M1575_CODEC_REG(i)]);
-	}
-
-	/*
-	 * Now we setup the EQ register to scale 16 bit linear pcm
-	 * values into 20 bit DAC values.
-	 */
-
-	/* 60h - EQ Control Register */
-
-	/*
-	 * Here we set up the 6 biquad IIR filters to use
-	 * coefficient a0 as our scaling factor. The equation for
-	 * this filter is:
-	 *
-	 * Y(n) = a0(a1(y(n-1))+a2(y(n-2))+b2(x(n-2))+b1(x(n-1)))
-	 * we use the following coefficient values:
-	 *
-	 *  a0 = 0x000f;
-	 *  a1 = 0x0001;
-	 *  a2 = 0x0001;
-	 *  b1 = 0x0001;
-	 *  b2 = 0x0001;
-	 */
-	for (i = 0; i < AD1981_MAX_FILTERS; i++) {
-		addr = filters[i].addr | AD1981_EQCTRL_SYM | AD1981_EQCTRL_EQM;
-		tmp = filters[i].coeff;
-		(void) audio1575_write_ac97(statep, AD1981_EQCTRL_REG, addr);
-		(void) audio1575_write_ac97(statep, AD1981_EQDATA_REG, tmp);
-	}
-
-	/* The AD1981B Codec only implements 5 bit volume register */
-	(void) audio1575_read_ac97(statep, AC97_MASTER_VOLUME_REGISTER, &tmp);
-	if ((tmp & 0x7fff) != (MVR_RIGHT_MASK | MVR_LEFT_MASK)) {
-		statep->m1575_vol_bits_mask = 6;
-	}
-
-	/* resume the master volume to mute */
-	(void) audio1575_write_ac97(statep, AC97_MASTER_VOLUME_REGISTER,
-	    MVR_MUTE);
-
-	/*
-	 * if the codec chip do support variable sample rate,
-	 * we set the sample rate to 48K
-	 */
-	(void) audio1575_read_ac97(statep, AC97_EXTENDED_AUDIO_REGISTER, &tmp);
-	if (!(tmp & EAR_VRA)) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!AD8191B codec does not support VRA");
-		ATRACE("audio1575_ac97_init() AD1981 EAR_VRA not set", tmp);
-		mutex_exit(&statep->m1575_intr_mutex);
-
-		return (AUDIO_FAILURE);
-
-	} else {
-		/* set variable sample rate mode */
-		(void) audio1575_write_ac97(statep,
-		    AC97_EXTENDED_AUDIO_STAT_CTRL_REGISTER, EASCR_VRA);
-	}
-
-	ATRACE("audio1575_ac97_init() sample rate",
-	    statep->m1575_defaults.play.sample_rate);
-
-	ATRACE("audio1575_ac97_init() returning", AUDIO_SUCCESS);
-
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_init_ac97() */
-
-/*
- * audio1575_chip_init()
- *
- * Description:
- *	This routine initializes the M1575 AC97 audio controller and the AC97
- *	codec.	The AC97 codec registers are programmed from codec_shadow[].
- *	If we are not doing a restore, we initialize codec_shadow[], otherwise
- *	we use the current values of shadow.	This routine expects that the
- *	PCI IO and Memory spaces have been mapped and enabled already.
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			restore		If M1575_INIT_RESTORE then
- *						restore	from codec_shadow[]
- * Returns:
- *	AUDIO_SUCCESS	The hardware was initialized properly
- *	AUDIO_FAILURE	The hardware couldn't be initialized properly
- */
-static int
-audio1575_chip_init(audio1575_state_t *statep, int restore)
-{
-	uint32_t	scr;
-	uint32_t	ssr;
-	uint32_t	rtsr;
-	uint32_t	intfcr;
-	int 		i;
-	int		j;
-	int		rc;
-	uint8_t		clk_detect;
-	clock_t		ticks;
-
-	ATRACE("audio1575_chip_init() entering", NULL);
-
-	mutex_enter(&statep->m1575_intr_mutex);
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	ticks = drv_usectohz(M1575_LOOP_CTR);
-
-	/*
-	 * SADA only supports stereo, so we set the channel bits
-	 * to "00" to select 2 channels.
-	 * will also set the following:
-	 *
-	 * Disable double rate enable
-	 * no SPDIF output selected
-	 * 16 bit audio record mode
-	 * 16 bit pcm out mode
-	 * PCM Out 6 chan mode FL FR CEN BL BR LFE
-	 * PCM Out 2 channel mode (00)
-	 */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		/* Reset the AC97 Codec	and default to 2 channel 16 bit mode */
-		M1575_AM_PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
-#ifndef __lock_lint
-		delay(ticks<<1);
-#endif
-		/* Read the System Status Reg */
-		ssr = M1575_AM_GET32(M1575_SSR_REG);
-
-		/* make sure and release the blocked reset bit */
-		if (ssr & M1575_SSR_RSTBLK) {
-			ATRACE("audio1575_chip_init()! RSTBLK detected", ssr);
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr |= M1575_INTFCR_RSTREL;
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-#ifndef __lock_lint
-			delay(ticks);
-#endif
-			/* Read the System Status Reg */
-			ssr = M1575_AM_GET32(M1575_SSR_REG);
-
-			/* make sure and release the blocked reset bit */
-			if (ssr & M1575_SSR_RSTBLK) {
-				ATRACE("audio1575_chip_init()! RSTBLK failure",
-				    ssr);
-				mutex_exit(&statep->m1575_intr_mutex);
-				return (AUDIO_FAILURE);
-			}
-
-			/* Reset the controller */
-			M1575_AM_PUT32(M1575_SCR_REG, M1575_SCR_COLDRST);
-#ifndef __lock_lint
-			delay(ticks);
-#endif
-		}
-
-		/* according AC'97 spec, wait for codec reset */
-		for (j = 0; j < M1575_LOOP_CTR; j++) {
-			scr = M1575_AM_GET32(M1575_SCR_REG);
-			if ((scr & M1575_SCR_COLDRST) == 0) {
-				break;
-			}
-#ifndef __lock_lint
-			delay(ticks);
-#endif
-		}
-
-		/* codec reset failed */
-		if (j >= M1575_LOOP_CTR) {
-			audio_sup_log(statep->m1575_ahandle, CE_WARN,
-			    "!failure to reset codec");
-			ATRACE("audio1575_chip_init() "
-			    "!AC97 COLDRST failure", scr);
-			mutex_exit(&statep->m1575_intr_mutex);
-			return (AUDIO_FAILURE);
-		}
-
-		/*
-		 * Wait for FACRDY First codec ready. The hardware can
-		 * provide the state of
-		 * codec ready bit on SDATA_IN[0] and as reflected in
-		 * the Recv Tag Slot Reg.
-		 */
-		rtsr = M1575_AM_GET32(M1575_RTSR_REG);
-		if (rtsr & M1575_RTSR_FACRDY) {
-			break;
-		} else { /* reset the status and wait for new status to set */
-			rtsr |= M1575_RTSR_FACRDY;
-			M1575_AM_PUT32(M1575_RTSR_REG, rtsr);
-			drv_usecwait(10);
-		}
-	}
-
-	/* if we could not reset the AC97 codec then report failure */
-	if (i >= M1575_LOOP_CTR) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!chip_init() no codec ready signal received");
-		ATRACE("audio1575_chip_init() AC97 Codec init "
-		    "failure - BAD Board", rtsr);
-		mutex_exit(&statep->m1575_intr_mutex);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* Magic code from ULi to Turn on the AC_LINK clock */
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 0);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 4);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 0);
-	(void) M1575_PCI_GET8(M1575_PCIACD_REG);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 2);
-	M1575_PCI_PUT8(M1575_PCIACD_REG, 0);
-	clk_detect = M1575_PCI_GET8(M1575_PCIACD_REG);
-
-	if (clk_detect != 1) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!chip_init() No AC97 Clock Detected");
-		ATRACE("audio1575_chip_init() no AC97 Clock detected",
-		    clk_detect);
-		mutex_exit(&statep->m1575_intr_mutex);
-		return (AUDIO_FAILURE);
-	}
-
-	/* Magic code from Uli to Init FIFO1 and FIFO2 */
-	M1575_AM_PUT32(M1575_FIFOCR1_REG, 0x81818181);
-	M1575_AM_PUT32(M1575_FIFOCR2_REG, 0x81818181);
-	M1575_AM_PUT32(M1575_FIFOCR3_REG, 0x81818181);
-
-	/* Disable SPDIF Output */
-	M1575_AM_PUT8(M1575_CSPOCR_REG, 0);
-
-	/* Disable Everyone */
-	M1575_AM_PUT32(M1575_INTFCR_REG, 0L);
-
-	audio1575_dma_stop(statep);
-
-	mutex_exit(&statep->m1575_intr_mutex);
-
-	/* now initialize the AC97 codec */
-	rc = audio1575_init_ac97(statep, restore);
-
-	ATRACE("audio1575_chip_init() returning", rc);
-
-	return (rc);
-
-}	/* audio1575_chip_init() */
-
-/*
- * audio1575_dma_stop()
- *
- * Description:
- *	This routine is used to put each DMA engine into the quiet state.
- *
- * Arguments:
- *	audio1575_state_t *statep	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_dma_stop(audio1575_state_t *statep)
-{
-	uint32_t	dmacr;
-	uint32_t	intrsr;
-	uint32_t	intrcr;
-	uint16_t	sr;
-	int		i;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	ATRACE("audio1575_dma_stop() entering", NULL);
-
-	/* pause bus master (needed for the following reset register) */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-		dmacr |= M1575_DMACR_PAUSE_ALL;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-		if (dmacr & M1575_DMACR_PAUSE_ALL) {
-			break;
-		}
-		drv_usecwait(10);
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-	}
-
-	if (i >= M1575_LOOP_CTR) {
-		audio_sup_log(statep->m1575_ahandle, CE_WARN,
-		    "!dma_stop() failed to stop DMA engines");
-		ATRACE("audio1575_dma_stop() failure "
-		    "to stop DMA engines", dmacr);
-		return;
-	}
-
-	/* Pause bus master (needed for the following reset register) */
-	M1575_AM_PUT8(M1575_PCMICR_REG, 0);
-	M1575_AM_PUT8(M1575_PCMOCR_REG, 0);
-	M1575_AM_PUT8(M1575_MICICR_REG, 0);
-	M1575_AM_PUT8(M1575_CSPOCR_REG, 0);
-	M1575_AM_PUT8(M1575_PCMI2CR_RR, 0);
-	M1575_AM_PUT8(M1575_MICI2CR_RR, 0);
-
-	/* Reset the bus master registers for all DMA engines */
-	M1575_AM_PUT8(M1575_PCMICR_REG, M1575_PCMICR_RR);
-	M1575_AM_PUT8(M1575_PCMOCR_REG, M1575_PCMOCR_RR);
-	M1575_AM_PUT8(M1575_MICICR_REG, M1575_MICICR_RR);
-	M1575_AM_PUT8(M1575_CSPOCR_REG, M1575_CSPOCR_RR);
-	M1575_AM_PUT8(M1575_PCMI2CR_REG, M1575_PCMI2CR_RR);
-	M1575_AM_PUT8(M1575_MICI2CR_REG, M1575_MICI2CR_RR);
-
-	/* Reset FIFOS */
-	M1575_AM_PUT32(M1575_FIFOCR1_REG, 0x81818181);
-	M1575_AM_PUT32(M1575_FIFOCR2_REG, 0x81818181);
-	M1575_AM_PUT32(M1575_FIFOCR3_REG, 0x81818181);
-
-	/* Clear Interrupts */
-	sr = M1575_AM_GET16(M1575_PCMISR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_PCMISR_REG, sr);
-
-	sr = M1575_AM_GET16(M1575_PCMOSR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_PCMOSR_REG, sr);
-
-	sr = M1575_AM_GET16(M1575_MICISR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_MICISR_REG, sr);
-
-	sr = M1575_AM_GET16(M1575_CSPOSR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_CSPOSR_REG, sr);
-
-	sr = M1575_AM_GET16(M1575_PCMI2SR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_PCMI2SR_REG, sr);
-
-	sr = M1575_AM_GET16(M1575_MICI2SR_REG) | M1575_STATUS_CLR;
-	M1575_AM_PUT16(M1575_MICI2SR_REG, sr);
-
-	/*
-	 * clear the interrupt control and status register
-	 * READ/WRITE/READ workaround required
-	 * for buggy hardware
-	 */
-
-	intrcr = 0L;
-	M1575_AM_PUT32(M1575_INTRCR_REG, intrcr);
-	intrcr = M1575_AM_GET32(M1575_INTRCR_REG);
-
-	intrsr = M1575_AM_GET32(M1575_INTRSR_REG);
-	M1575_AM_PUT32(M1575_INTRSR_REG, (intrsr & M1575_INTR_MASK));
-	intrsr = M1575_AM_GET32(M1575_INTRSR_REG);
-
-	/* clear all flags except DMA suspend flag */
-	statep->m1575_flags &= M1575_DMA_SUSPENDED;
-
-	ATRACE("audio1575_stop_dma() returning", NULL);
-
-}	/* audio1575_dma_stop() */
-
-/*
- * audio1575_set_gain()
- *
- * Description:
- *	Set the play/record gain.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD, if
- *						direction is important
- *	int			arg1		The gain to set
- *	int			arg2		The channel, 0 == left
- *						or 1 == right
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audio1575_set_gain(audio1575_state_t *statep, int dir, int gain, int channel)
-{
-	uint16_t	tmp;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	if (gain > AUDIO_MAX_GAIN) {
-		gain = AUDIO_MAX_GAIN;
-	} else if (gain < AUDIO_MIN_GAIN) {
-		gain = AUDIO_MIN_GAIN;
-	}
-
-	switch (dir) {
-	case AUDIO_PLAY:
-		/*
-		 * For play we use PCM so all volumes change with just
-		 * one write. This way we get line out, headphone and
-		 * internal speaker in one shot.
-		 *
-		 * The AC97 Codec goes from -34.5 dB (11111) to 0 dB
-		 * (01000) to +12.0 dB (00000). We turn gain into attenuation
-		 * The AD1981B codec uses attenuation instead of gain as well.
-		 * abs(-34.5dB)+12.dB = 46.5dB of dynamic range/1.5dB per bit
-		 * gives a range of 0x00-0x1f (0-31) values.
-		 */
-
-		/* read the play gain register */
-		(void) audio1575_read_ac97(statep, AC97_PCM_OUT_VOLUME_REGISTER,
-		    &tmp);
-		tmp &= ~PCMOVR_MUTE;
-
-		if (channel == 0) {	/* left channel */
-			tmp &= ~PCMOVR_LEFT_GAIN_MASK;
-			tmp |= (((AUDIO_MAX_GAIN - gain) >>
-			    M1575_GAIN_SHIFT3) << M1575_BYTE_SHIFT) &
-			    PCMOVR_LEFT_GAIN_MASK;
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			tmp &= ~PCMOVR_RIGHT_GAIN_MASK;
-			tmp |= ((AUDIO_MAX_GAIN - gain) >>
-			    M1575_GAIN_SHIFT3) & PCMOVR_RIGHT_GAIN_MASK;
-		}
-
-		/* update the play gain register */
-		(void) audio1575_write_ac97(statep,
-		    AC97_PCM_OUT_VOLUME_REGISTER, tmp);
-		(void) audio1575_read_ac97(statep, AC97_PCM_OUT_VOLUME_REGISTER,
-		    &tmp);
-		break;
-
-	case AUDIO_RECORD:
-		/*
-		 * For record we use the master record gain with all
-		 * of the inputs set to 0dB.
-		 * The AC97 Codec goes from 0 dB (0000) to +22.5 dB
-		 * (1111) 22.5 dB/1.5dB per bit = range of 0x0-0xf;
-		 * Note: this is gain not attenuation.
-		 */
-
-		/* read the record gain register */
-		(void) audio1575_read_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, &tmp);
-
-		if (channel == 0) {	/* left channel */
-			tmp &= ~RGR_LEFT_MASK;
-			tmp |= (gain << M1575_GAIN_SHIFT4) & RGR_LEFT_MASK;
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			tmp &= ~RGR_RIGHT_MASK;
-			tmp |= (gain >> M1575_GAIN_SHIFT4) & RGR_RIGHT_MASK;
-		}
-
-		/* update the record gain register */
-		(void) audio1575_write_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, tmp);
-		(void) audio1575_read_ac97(statep,
-		    AC97_RECORD_GAIN_REGISTER, &tmp);
-		break;
-
-	default:
-		ATRACE("audio1575_set_gain() Unknown audio direction", dir);
-
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_set_gain() */
-
-/*
- * audio1575_set_port()
- *
- * Description:
- *	Set the play/record port.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *						which is not how we program
- *						the device for now.
- *	int			dir		AUDIO_PLAY or AUDIO_RECORD,
- *						if direction is important
- *	int			port		The port to set:
- *				AUDIO_SPEAKER	output to built-in speaker
- *				AUDIO_HEADPHONE	output to headphone
- *				AUDIO_LINE_OUT	output to line out
- *
- *				AUDIO_MICROPHONE	input from microphone
- *				AUDIO_LINE_IN		input from line in
- *				AUDIO_CD 		input from internal CD
- *				AUDIO_CODEC_LOOPB_IN	input from Codec
- *							internal loopback
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The port could not been set
- */
-static int
-audio1575_set_port(audio1575_state_t *statep, int dir, int port)
-{
-	uint16_t	tmp = 0;
-	uint16_t	reg = 0;
-	uint32_t	intfcr = 0;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	switch (dir) {
-	case AUDIO_PLAY: /* output port */
-		if (port == M1575_PORT_UNMUTE) {
-			port = statep->m1575_output_port;
-		}
-
-		if (port & AUDIO_SPEAKER) {
-			(void) audio1575_and_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_SPEAKER;
-		} else {
-			(void) audio1575_or_ac97(statep,
-			    AC97_MONO_MASTER_VOLUME_REGSITER, MVR_MUTE);
-		}
-
-		if (port & AUDIO_LINE_OUT) {
-			(void) audio1575_and_ac97(statep,
-			    AC97_MASTER_VOLUME_REGISTER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_LINE_OUT;
-		} else {
-			(void) audio1575_or_ac97(statep,
-			    AC97_MASTER_VOLUME_REGISTER, MVR_MUTE);
-		}
-
-		if (port & AUDIO_HEADPHONE) {
-			(void) audio1575_and_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER,
-			    (uint16_t)~MVR_MUTE);
-			tmp |= AUDIO_HEADPHONE;
-		} else {
-			(void) audio1575_or_ac97(statep,
-			    AC97_HEADPHONE_VOLUME_REGISTER, MVR_MUTE);
-		}
-
-		intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-		intfcr |= (M1575_INTFCR_PCMOENB);
-		M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-
-		ATRACE("audio1575_set_port() out port", tmp);
-		statep->m1575_output_port = tmp;
-		if (tmp != port) {
-			ATRACE("audio1575_set_port() bad out port", port);
-
-			return (AUDIO_FAILURE);
-		}
-		break;
-
-	case AUDIO_RECORD: /* input port */
-		switch (port) {
-		case AUDIO_NONE:
-			/* set to an unused input */
-			tmp = RSCR_R_PHONE | RSCR_L_PHONE;
-			ATRACE("audio1575_set_port() AUDIO_NONE", NULL);
-
-			/* mute the master record input */
-			(void) audio1575_or_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER, RGR_MUTE);
-
-			if (statep->m1575_monitor_gain) {
-				if (statep->m1575_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_CD) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER, CDVR_MUTE);
-				}
-			}
-
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr &= ~(M1575_INTFCR_PCMIENB|M1575_INTFCR_MICENB);
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-			break;
-
-		case AUDIO_MICROPHONE:
-			/* set to the mic input */
-			tmp = RSCR_R_MIC | RSCR_L_MIC;
-			if (statep->m1575_monitor_gain) {
-				if (statep->m1575_input_port == AUDIO_LINE_IN) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_LINE_IN_VOLUME_REGISTER,
-					    LIVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_CD) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-			(void) audio1575_write_ac97(statep,
-			    AC97_MIC_VOLUME_REGISTER,
-			    statep->m1575_monitor_gain);
-			}
-
-			/* Enable the MIC input on AC Link Channel 3 */
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr |= (M1575_INTFCR_PCMIENB);
-			intfcr &= ~(M1575_INTFCR_MICISEL|M1575_INTFCR_MICENB |
-			    M1575_INTFCR_MICI2SEL| M1575_INTFCR_MICI2ENB);
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-			break;
-
-		case AUDIO_LINE_IN:
-			/* set to the line in input */
-			tmp = RSCR_R_LINE_IN | RSCR_L_LINE_IN;
-			ATRACE("audio1575_set_port() LINE_IN", tmp);
-			/* see if we need to update monitor loopback */
-			if (statep->m1575_monitor_gain) {
-				if (statep->m1575_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_CD) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER,
-					    CDVR_MUTE);
-				}
-				(void) audio1575_write_ac97(statep,
-				    AC97_LINE_IN_VOLUME_REGISTER,
-				    statep->m1575_monitor_gain);
-			}
-
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr &= ~(M1575_INTFCR_MICENB);
-			intfcr |= (M1575_INTFCR_PCMIENB);
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			break;
-
-		case AUDIO_CD:
-			/* set to the line in input */
-			tmp = RSCR_R_CD|RSCR_L_CD;
-			/* see if we need to update monitor loopback */
-			if (statep->m1575_monitor_gain) {
-				reg = AC97_LINE_IN_VOLUME_REGISTER;
-				if (statep->m1575_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_LINE_IN) {
-					(void) audio1575_or_ac97(statep,
-					    reg, LIVR_MUTE);
-				}
-				(void) audio1575_write_ac97(statep,
-				    AC97_CD_VOLUME_REGISTER,
-				    statep->m1575_monitor_gain);
-			}
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr &= ~(M1575_INTFCR_MICENB);
-			intfcr |= (M1575_INTFCR_PCMIENB);
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-			break;
-
-		case AUDIO_CODEC_LOOPB_IN:
-			/* set to the loopback input */
-			tmp = RSCR_R_STEREO_MIX | RSCR_L_STEREO_MIX;
-			ATRACE("audio1575_set_port() LPBK", tmp);
-			if (statep->m1575_monitor_gain) {
-				if (statep->m1575_input_port == AUDIO_LINE_IN) {
-				(void) audio1575_or_ac97(statep,
-				    AC97_LINE_IN_VOLUME_REGISTER, LIVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_MICROPHONE) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_MIC_VOLUME_REGISTER,
-					    MICVR_MUTE);
-				} else if (statep->m1575_input_port ==
-				    AUDIO_CD) {
-					(void) audio1575_or_ac97(statep,
-					    AC97_CD_VOLUME_REGISTER, CDVR_MUTE);
-				}
-			}
-			intfcr = M1575_AM_GET32(M1575_INTFCR_REG);
-			intfcr &= ~(M1575_INTFCR_MICENB | M1575_INTFCR_MICISEL);
-			intfcr |= (M1575_INTFCR_PCMIENB);
-			M1575_AM_PUT32(M1575_INTFCR_REG, intfcr);
-			break;
-
-		default:
-			ATRACE("audio1575_set_port bad in port", port);
-
-			return (AUDIO_FAILURE);
-		}
-
-		/* select the input */
-		(void) audio1575_write_ac97(statep,
-		    AC97_RECORD_SELECT_CTRL_REGISTER, tmp);
-
-		if ((port != AUDIO_NONE) &&
-		    (statep->m1575_codec_shadow[M1575_CODEC_REG(
-		    AC97_RECORD_GAIN_REGISTER)] & RGR_MUTE)) {
-			(void) audio1575_and_ac97(statep,
-			    AC97_RECORD_GAIN_REGISTER,
-			    (uint16_t)~RGR_MUTE);
-		}
-		statep->m1575_input_port = port;
-		break;
-	}
-
-	ATRACE("audio1575_set_port() returning", port);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_set_port() */
-
-/*
- * audio1575_set_monitor_gain()
- *
- * Description:
- *	Set the monitor gain.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			gain		The gain to set
- *
- * Returns:
- * 	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-audio1575_set_monitor_gain(audio1575_state_t *statep, int gain)
-{
-	uint16_t	tmp_short;
-	int		rc = AUDIO_SUCCESS;
-
-	ATRACE("audio1575_set_monitor_gain() entering", statep);
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (gain == 0) {
-		/* disable loopbacks when gain == 0 */
-		tmp_short = MVR_MUTE;
-	} else {
-		/* Adjust the value of gain to the requirement of AC'97 */
-		tmp_short = (((AUDIO_MAX_GAIN - gain) >> M1575_GAIN_SHIFT3) <<
-		    M1575_BYTE_SHIFT) & PCMOVR_LEFT_GAIN_MASK;
-		tmp_short |= ((AUDIO_MAX_GAIN - gain) >> M1575_GAIN_SHIFT3) &
-		    PCMOVR_RIGHT_GAIN_MASK;
-	}
-
-	switch (statep->m1575_input_port) {
-	case AUDIO_NONE:
-		/*
-		 * It is possible to set the value of gain before any input
-		 * is selected. So, we just save the gain and then return
-		 * SUCCESS.
-		 */
-		break;
-
-	case AUDIO_MICROPHONE:
-		tmp_short |= statep->m1575_codec_shadow[M1575_CODEC_REG(
-		    AC97_MIC_VOLUME_REGISTER)] & MICVR_20dB_BOOST;
-		(void) audio1575_write_ac97(statep,
-		    AC97_MIC_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_LINE_IN:
-		(void) audio1575_write_ac97(statep,
-		    AC97_LINE_IN_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CD:
-		(void) audio1575_write_ac97(statep,
-		    AC97_CD_VOLUME_REGISTER, tmp_short);
-		break;
-
-	case AUDIO_CODEC_LOOPB_IN:
-		/* we already are setting the loopback, so done */
-		goto done;
-
-	default:
-		/* this should never happen! */
-		ATRACE("audio1575_ad_set_config() monitor gain bad device",
-		    statep->m1575_input_port);
-		rc = AUDIO_FAILURE;
-		goto done;
-	}
-
-	if (gain == 0) {
-		statep->m1575_monitor_gain = 0;
-	} else {
-		statep->m1575_monitor_gain = tmp_short;
-	}
-
-done:
-	ATRACE("audio1575_set_monitor_gain() returning", rc);
-
-	return (rc);
-
-}	/* audio1575_set_monitor_gain() */
-
-/*
- * audio1575_codec_sync()
- *
- * Description:
- *	Serialize access to the AC97 audio mixer registers.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Ready for an I/O access to the codec
- *	AUDIO_FAILURE		An I/O access is currently in progress, can't
- *				perform another I/O access.
- */
-static int
-audio1575_codec_sync(audio1575_state_t *statep)
-{
-	int 		i;
-	int 		j;
-	uint32_t	casr;
-	uint32_t	cspsr;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	/* do the Uli Shuffle ... */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		/* Read the semaphore */
-		casr = M1575_AM_GET32(M1575_CASR_REG);
-		/* loop till we own the semaphore */
-		if ((casr & 1) == 0) {
-			for (j = 0; j < M1575_LOOP_CTR; j++) {
-				/* Wait for CWRSUCC 0x8 */
-				cspsr = M1575_AM_GET32(M1575_CSPSR_REG);
-				if ((cspsr & M1575_CSPSR_SUCC) ==
-				    M1575_CSPSR_SUCC) {
-					return (AUDIO_SUCCESS);
-				}
-				drv_usecwait(1);
-			}
-		}
-		drv_usecwait(10);
-	}
-
-	return (AUDIO_FAILURE);
-
-}	/* audio1575_codec_sync() */
-
-/*
- * audio1575_and_ac97()
- *
- * Description:
- *	Logically AND the value with the specified ac97 codec register
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to AND
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio1575_and_ac97(audio1575_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t	tmp;
-
-	if ((audio1575_write_ac97(statep, reg, data &
-	    statep->m1575_codec_shadow[M1575_CODEC_REG(reg)])) ==
-	    AUDIO_FAILURE) {
-		return (AUDIO_FAILURE);
-	}
-
-	(void) audio1575_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_and_ac97() */
-
-/*
- * audio1575_or_ac97()
- *
- * Description:
- *	Logically OR the value with the specified ac97 codec register
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The value to OR
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio1575_or_ac97(audio1575_state_t *statep, int reg, uint16_t data)
-{
-	uint16_t	tmp;
-
-	if ((audio1575_write_ac97(statep, reg, data |
-	    statep->m1575_codec_shadow[M1575_CODEC_REG(reg)])) ==
-	    AUDIO_FAILURE) {
-		return (AUDIO_FAILURE);
-	}
-
-	(void) audio1575_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_or_ac97() */
-
-/*
- * audio1575_write_ac97()
- *
- * Description:
- *	Set the specific AC97 Codec register.
- *
- * Arguments:
- *	audio1575_state_t	 *state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		data		The data want to be set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set
- */
-static int
-audio1575_write_ac97(audio1575_state_t *statep, int reg, uint16_t data)
-{
-	uint32_t	cspsr;
-	uint16_t	tmp;
-	int		i;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	if (audio1575_codec_sync(statep) != AUDIO_SUCCESS) {
-		return (AUDIO_FAILURE);
-	}
-
-	/* write the data to WRITE to the lo word of the CPR register */
-	M1575_AM_PUT16(M1575_CPR_REG, data);
-
-	/* write the address to WRITE to the hi word of the CPR register */
-	M1575_AM_PUT16(M1575_CPR_REG+2, reg);
-
-	/* wait until command is completed sucessfully */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		/* Wait for Write Ready	0x01 */
-		cspsr = M1575_AM_GET32(M1575_CSPSR_REG);
-		if ((cspsr & M1575_CSPSR_WRRDY) == M1575_CSPSR_WRRDY) {
-			break;
-		}
-		drv_usecwait(1);
-	}
-
-	if (i >= M1575_LOOP_CTR) {
-		ATRACE("audio1575_write_ac97() failure", i);
-
-		return (AUDIO_FAILURE);
-	}
-
-	(void) audio1575_read_ac97(statep, reg, &tmp);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_write_ac97() */
-
-/*
- * audio1575_read_ac97()
- *
- * Description:
- *	Get the specific AC97 Codec register. It also updates codec_shadow[]
- *	with the register value.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *	int			reg		AC97 register number
- *	uint16_t		*data		The data to be returned
- *
- * Returns:
- *	AUDIO_SUCCESS		Reading the codec register successfully
- *	AUDIO_FAILURE		Failed to read the register
- */
-static int
-audio1575_read_ac97(audio1575_state_t *statep, int reg, uint16_t *data)
-{
-	uint32_t	cspsr;
-	uint16_t	addr = 0;
-	int		i;
-
-	ATRACE("audio1575_read_ac97() entering", NULL);
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	if ((audio1575_codec_sync(statep)) != AUDIO_SUCCESS) {
-		*data = 0xffff;
-		ATRACE("audio1575_read_ac97() sync failure", NULL);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * at this point we has the CASR semaphore
-	 * and the codec is r/w ready
-	 * OR in the READ opcode into the address field
-	 */
-
-	addr = (reg | M1575_CPR_READ);
-
-	/* write the address to READ to the hi word of the CPR register */
-	M1575_AM_PUT16(M1575_CPR_REG+2, addr);
-
-	/* wait until command is completed sucessfully */
-	for (i = 0; i < M1575_LOOP_CTR; i++) {
-		/* Wait for Read Ready	0x02 */
-		cspsr = M1575_AM_GET32(M1575_CSPSR_REG);
-		if ((cspsr & M1575_CSPSR_RDRDY) == M1575_CSPSR_RDRDY) {
-			break;
-		}
-		drv_usecwait(1);
-	}
-
-	if (i >= M1575_LOOP_CTR) {
-		*data = 0xffff;
-		ATRACE("audio1575_read_ac97() CSPSR NOT READY", cspsr);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* read back the data and address */
-	*data = M1575_AM_GET16(M1575_SPR_REG);
-	addr = M1575_AM_GET16(M1575_SPR_REG+2);
-
-	if (addr != reg) {
-		ATRACE("audio1575_read_ac97() bad read address", addr);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* store new value in codec shadow register */
-	statep->m1575_codec_shadow[M1575_CODEC_REG(reg)] = *data;
-
-	ATRACE("audio1575_read_ac97() returning successfully", NULL);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_read_ac97() */
-
-/*
- * audio1575_reset_ac97()
- *
- * Description:
- *	Reset AC97 Codec register.
- *
- * Arguments:
- *	audio1575_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Reset the codec successfully
- *	AUDIO_FAILURE		Failed to reset the codec
- */
-static int
-audio1575_reset_ac97(audio1575_state_t *statep)
-{
-	uint16_t	tmp;
-
-	if (audio1575_read_ac97(statep,
-	    AC97_POWERDOWN_CTRL_STAT_REGISTER, &tmp) != AUDIO_SUCCESS) {
-
-		return (AUDIO_FAILURE);
-	}
-
-	if (audio1575_write_ac97(statep, AC97_RESET_REGISTER, 0xffff) !=
-	    AUDIO_SUCCESS) {
-
-		return (AUDIO_FAILURE);
-	}
-
-	if (audio1575_read_ac97(statep, AC97_RESET_REGISTER, &tmp) !=
-	    AUDIO_SUCCESS) {
-
-		return (AUDIO_FAILURE);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_reset_ac97() */
-
-/*
- * audio1575_fill_play_buf()
- *
- * Description:
- *	This routine is called by m1575_ad_start_play() and the interrupt
- *	handler. It fills playback samples into the DMA memory, sets the
- *	BDL entries, and starts the playback DMA engine.
- *	the m1575_intr_mutex must be held on entry.
- *
- * Arguments:
- *	audio1575_state_t *statep	The device's state structure
- *
- * Returns:
- * 	AUDIO_SUCCESS		Starting PCM out engine successfully
- * 	AUDIO_FAILURE		Failed to start PCM out engine.
- */
-static int
-audio1575_fill_play_buf(audio1575_state_t *statep)
-{
-	m1575_bdlist_chunk_t	*chunk;
-	m1575_sample_buf_t	*buf;
-	m1575_bd_entry_t	*bdesc;
-	int			samples;
-	uint32_t		dmacr;
-	int			rs;
-	int			bufcount = 0;
-	int			rc = AUDIO_SUCCESS;
-
-	ASSERT(mutex_owned(&statep->m1575_intr_mutex));
-
-	/* get the play buffer pointer */
-	buf = &statep->m1575_play_buf;
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	/* compute number of samples */
-	samples = statep->m1575_psample_rate * statep->m1575_pchannels /
-	    statep->m1575_ad_info.ad_play.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	while (buf->avail > 0) {
-		chunk = &(buf->chunk[buf->tail & M1575_PLAY_BUF_MSK]);
-		mutex_exit(&statep->m1575_intr_mutex);
-		rs = am_get_audio(statep->m1575_ahandle,
-		    (char *)(chunk->data_buf), AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&statep->m1575_intr_mutex);
-
-		if (((statep->m1575_flags & M1575_DMA_PLAY_STARTED) == 0) &&
-		    (buf->io_started)) {
-			audio_sup_log(statep->m1575_ahandle, CE_WARN,
-			    "!fill_play_buf() Err:PLAY started and IO started");
-			ATRACE("audio1575_fill_play_buf() Err:PLAY STARTED "
-			    "and IO STARTED", NULL);
-
-			return (AUDIO_FAILURE);
-		}
-		/* no more samples to play */
-		if (rs <= 0) {
-			if (statep->m1575_flags & M1575_DMA_PLAY_EMPTY) {
-				if (bufcount != 0) {
-					break;
-				}
-				/*
-				 * Clear the flag so if audio is restarted while
-				 * in am_play_shutdown() we can detect it and
-				 * not mess things up.
-				 */
-				statep->m1575_flags &= ~M1575_DMA_PLAY_STARTED;
-
-				/* shutdown the mixer */
-				mutex_exit(&statep->m1575_intr_mutex);
-				am_play_shutdown(statep->m1575_ahandle, NULL);
-				mutex_enter(&statep->m1575_intr_mutex);
-
-				/*
-				 * Make sure playing wasn't restarted when lock
-				 * lost if reopened, should return success
-				 */
-				if (statep->m1575_flags &
-				    M1575_DMA_PLAY_STARTED) {
-					return (AUDIO_SUCCESS);
-				}
-
-				/* Finished playing, then stop it */
-				rc = audio1575_dma_reset(statep,
-				    M1575_DMA_PCM_OUT);
-				if (rc == AUDIO_FAILURE) {
-					ATRACE("audio1575_fill_play_buf() "
-					    "dma_reset failure", rc);
-					return (rc);
-				}
-
-				/* clear the PCM Out ctrl reg */
-				M1575_AM_PUT8(M1575_PCMOCR_REG, 0);
-				buf->io_started = B_FALSE;
-
-				/* clr the flags getting ready for next start */
-				statep->m1575_flags &= ~(M1575_DMA_PLAY_PAUSED |
-				    M1575_DMA_PLAY_EMPTY);
-
-				/* return the value for m1575_ad_start_play() */
-				return (AUDIO_FAILURE);
-
-			} else {
-				/* M1575_DMA_PLAY_EMPTY */
-				statep->m1575_flags |= M1575_DMA_PLAY_EMPTY;
-			}
-		} else {
-			/* !M1575_DMA_PLAY_EMPTY */
-			statep->m1575_flags &= ~M1575_DMA_PLAY_EMPTY;
-			bufcount++;
-			(void) ddi_dma_sync(chunk->dma_handle, 0, rs << 1,
-			    DDI_DMA_SYNC_FORDEV);
-		}
-
-		/* put the samples into buffer descriptor list entry */
-		bdesc = &(statep->m1575_bdl_virt_pout[buf->tail]);
-		bdesc->buf_base = chunk->addr_phy;
-		bdesc->buf_len = (uint16_t)rs;
-		bdesc->buf_cmd |= IOC;
-		M1575_AM_PUT8(M1575_PCMOLVIV_REG, buf->tail);
-		buf->tail++;
-		buf->tail &= M1575_BD_MSK;
-		buf->avail--;
-	}
-
-	/* start PCM out engine */
-	dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-	if (buf->avail < M1575_PLAY_BUFS &&
-	    !(dmacr & M1575_DMACR_PCMOSTART)) {
-		dmacr |= M1575_DMACR_PCMOSTART;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		ATRACE("audio1575_fill_play_buf() PLAY DMA STARTED", dmacr);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_fill_play_buf() */
-
-/*
- * audio1575_prepare_record_buf()
- *
- * Description:
- *	This routine is called by audio1575_ad_start_record(). It prepares DMA
- *	memory for PCM in engine, sets the buffer descriptor entries for PCM
- *	in engine, and starts PCM in engine for recording.
- *
- * Arguments:
- *	audio1575_state_t *statep	 The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Started PCM in engine successfully
- *	AUDIO_FAILURE		Failed to start PCM in engine.
- */
-static int
-audio1575_prepare_record_buf(audio1575_state_t *statep)
-{
-	m1575_bdlist_chunk_t	*chunk;
-	m1575_sample_buf_t	*buf;
-	m1575_bd_entry_t	*bdesc;
-	uint32_t		dmacr;
-	int			samples;
-
-	/* get the record buf ptr */
-	buf = &statep->m1575_record_buf;
-
-	if (buf->avail == 0) {
-		return (AUDIO_SUCCESS);
-	}
-
-	samples = statep->m1575_csample_rate * statep->m1575_cchannels /
-	    statep->m1575_ad_info.ad_record.ad_int_rate;
-
-	/* if not an even number of samples we panic! */
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples & 1) != 0) {
-		samples++;
-	}
-
-	statep->m1575_csamples = samples;
-
-	while (buf->avail > 0) {
-		chunk = &buf->chunk[buf->tail & M1575_REC_BUF_MSK];
-		bdesc = &(statep->m1575_bdl_virt_pin[buf->tail]);
-		bdesc->buf_base = chunk->addr_phy;
-		bdesc->buf_len = (uint16_t)samples;
-		bdesc->buf_cmd |= IOC;
-		M1575_AM_PUT8(M1575_PCMILVIV_REG, buf->tail);
-		buf->tail++;
-		buf->tail &= M1575_BD_MSK;
-		buf->avail--;
-	}
-
-	if (!buf->io_started) {
-		buf->io_started = B_TRUE;
-	}
-
-	dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-	if ((buf->avail < M1575_REC_BUFS) && !(dmacr & M1575_DMACR_PCMISTART)) {
-		/* start PCM In engine */
-		dmacr |= M1575_DMACR_PCMISTART;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		ATRACE("audio1575_prepare_record_buf() "
-		    "RECORD DMA STARTED", dmacr);
-	}
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_prepare_record_buf() */
-
-/*
- * audio1575_reclaim_record_buf()
- *
- * Description:
- *	This routine is called by the interrupt handler. It sends the PCM
- *	samples (record data) up to the mixer module by calling am_send_audio(),
- *	and reclaims the buffer descriptor entries for PCM in engine.
- *
- * Arguments:
- *	audio1575_state_t *statep	 The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_reclaim_record_buf(audio1575_state_t *statep)
-{
-	m1575_bdlist_chunk_t 	*chunk;
-	m1575_sample_buf_t	*buf;
-	int16_t			pcmiciv;
-	int			samples;
-
-	/* get record buf ptr */
-	buf = &statep->m1575_record_buf;
-
-	/* get our current index value */
-	pcmiciv = M1575_AM_GET8(M1575_PCMICIV_REG);
-
-	/* get number of samples */
-	samples = statep->m1575_csamples;
-
-	/* While we have record buffers to process */
-	while ((buf->head != pcmiciv && buf->avail < M1575_REC_BUFS)) {
-		chunk = &buf->chunk[buf->head & M1575_REC_BUF_MSK];
-		(void) ddi_dma_sync(chunk->dma_handle, 0,
-		    chunk->real_len, DDI_DMA_SYNC_FORCPU);
-		mutex_exit(&statep->m1575_intr_mutex);
-		am_send_audio(statep->m1575_ahandle, chunk->data_buf,
-		    AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&statep->m1575_intr_mutex);
-		buf->avail++;
-		buf->head++;
-		buf->head &= M1575_BD_MSK;
-		if ((statep->m1575_flags & M1575_DMA_RECD_STARTED) == 0) {
-			break;
-		}
-	}
-
-}	/* audio1575_reclaim_record_buf() */
-
-/*
- * audio1575_pci_enable()
- *
- * Description:
- *	This routine Enables all PCI IO and MEMORY accesses
- *
- * Arguments:
- *	audio1575_state_t *statep	 The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_pci_enable(audio1575_state_t *statep)
-{
-	uint16_t pcics_reg;
-
-	ATRACE("audio1575_pci_enable() entering", statep);
-
-	pcics_reg = pci_config_get16(statep->m1575_pci_regs_handle,
-	    PCI_CONF_COMM);
-	ATRACE("audio1575_pci_enable() PCICS Reg ", pcics_reg);
-	pcics_reg |= (PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
-	pci_config_put16(statep->m1575_pci_regs_handle, PCI_CONF_COMM,
-	    pcics_reg);
-
-	ATRACE("audio1575_pci_enable() returning", pcics_reg);
-
-}	/* audio1575_pci_enable() */
-
-/*
- * audio1575_pci_disable()
- *
- * Description:
- *	This routine Disables all PCI IO and MEMORY accesses
- *
- * Arguments:
- *	audio1575_state_t *statep	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-audio1575_pci_disable(audio1575_state_t *statep)
-{
-	uint16_t pcics_reg;
-
-	ATRACE("audio1575_pci_disable() entering", statep);
-
-	pcics_reg = pci_config_get16(statep->m1575_pci_regs_handle,
-	    PCI_CONF_COMM);
-	pcics_reg &= ~(PCI_COMM_IO | PCI_COMM_MAE | PCI_COMM_ME);
-	pci_config_put16(statep->m1575_pci_regs_handle, PCI_CONF_COMM,
-	    pcics_reg);
-
-	ATRACE("audio1575_pci_disable() returning", pcics_reg);
-
-}	/* audio1575_pci_disable() */
-
-/*
- * audio1575_dma_pause()
- *
- * Description:
- *	This routine pauses DMA on a particular channel
- *	It does not reset andy registers so play/record can be resumed.
- *
- * Arguments:
- *	audio1575_state_t *statep	The device's state structure
- *	int 	chan			The DMA channel PCM_IN,PCM_OUT,MIC_IN
- *
- * Returns:
- *	AUDIO_SUCCESS		DMA paused successfully
- *	AUDIO_FAILURE		DMA failed to pause
- */
-static int
-audio1575_dma_pause(audio1575_state_t *statep, int chan)
-{
-	uint32_t dmacr;
-
-	ATRACE("audio1575_dma_pause() entering", NULL);
-
-	dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-
-	switch (chan) {
-	case M1575_DMA_PCM_IN:
-		dmacr |= M1575_DMACR_PCMIPAUSE;
-		break;
-
-	case M1575_DMA_PCM_OUT:
-		dmacr |= M1575_DMACR_PCMOPAUSE;
-		break;
-
-	default:
-		ATRACE("audio1575_dma_pause() bad channel", chan);
-
-		return (AUDIO_FAILURE);
-	}
-
-	M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-
-	ATRACE("audio1575_dma_pause() returning", AUDIO_SUCCESS);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_dma_pause() */
-
-/*
- * audio1575_dma_resume()
- *
- * Description:
- *	This routine resumes DMA on a particular channel
- *	It does not resume if the BDL list is empty.
- *
- * Arguments:
- *	audio1575_state_t	*statep	The device's state structure
- *	int 	chan		The DMA channel
- *				PCM_IN,PCM_OUT,MIC_IN
- *
- * Returns:
- *	AUDIO_SUCCESS		DMA resumed successfully
- *	AUDIO_FAILURE		DMA failed to resume
- */
-static int
-audio1575_dma_resume(audio1575_state_t *statep, int chan)
-{
-	uint32_t dmacr;
-	uint32_t fifocr1;
-
-	ATRACE("audio1575_dma_resume() entering", NULL);
-
-	dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-
-	switch (chan) {
-	case M1575_DMA_PCM_IN:
-		/* ULi says do fifo resets here */
-		fifocr1 = M1575_AM_GET32(M1575_FIFOCR1_REG);
-		fifocr1 |= M1575_FIFOCR1_PCMIRST;
-		M1575_AM_PUT32(M1575_FIFOCR1_REG, fifocr1);
-		dmacr &= ~M1575_DMACR_PCMIPAUSE;
-		dmacr |= M1575_DMACR_PCMISTART;
-		break;
-
-	case M1575_DMA_PCM_OUT:
-		dmacr &= ~M1575_DMACR_PCMOPAUSE;
-		dmacr |= M1575_DMACR_PCMOSTART;
-		break;
-	default:
-		ATRACE("audio1575_dma_resume() bad channel", chan);
-
-		return (AUDIO_FAILURE);
-	}
-
-	M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-
-	ATRACE("audio1575_dma_resume() returning", AUDIO_SUCCESS);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_dma_resume() */
-
-/*
- * audio1575_dma reset()
- *
- * Description:
- *	This routine resets the DMA on a particular channel
- *	All DMA registers are RESET.
- *
- * Arguments:
- *	audio1575_state_t *statep	The device's state structure
- *	int chan			The DMA channel PCM_IN,PCM_OUT,MIC_IN
- *
- * Returns:
- *	AUDIO_SUCCESS		DMA reset successfully
- *	AUDIO_FAILURE		DMA failed to reset
- */
-static int
-audio1575_dma_reset(audio1575_state_t *statep, int chan)
-{
-	uint8_t		cr;
-	uint32_t	fifocr1;
-	uint32_t	dmacr;
-
-	ATRACE("audio1575_dma_reset() entering", NULL);
-
-	dmacr = M1575_AM_GET32(M1575_DMACR_REG);
-	fifocr1 = M1575_AM_GET32(M1575_FIFOCR1_REG);
-
-	switch (chan) {
-	case M1575_DMA_PCM_IN:
-		/* Uli FIFO madness ... */
-		fifocr1 = M1575_AM_GET32(M1575_FIFOCR1_REG);
-		fifocr1 |= M1575_FIFOCR1_PCMIRST;
-		M1575_AM_PUT32(M1575_FIFOCR1_REG, fifocr1);
-		dmacr |= M1575_DMACR_PCMIPAUSE;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		cr = M1575_AM_GET8(M1575_PCMICR_REG);
-		cr |= M1575_PCMICR_RR;
-		M1575_AM_PUT8(M1575_PCMICR_REG, cr);
-		dmacr &= ~M1575_DMACR_PCMIPAUSE;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		break;
-
-	case M1575_DMA_PCM_OUT:
-		/* Uli FIFO madness ... */
-		fifocr1 = M1575_AM_GET32(M1575_FIFOCR1_REG);
-		fifocr1 |= M1575_FIFOCR1_PCMORST;
-		M1575_AM_PUT32(M1575_FIFOCR1_REG, fifocr1);
-		dmacr |= M1575_DMACR_PCMOPAUSE;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		cr = M1575_AM_GET8(M1575_PCMOCR_REG);
-		cr |= M1575_PCMOCR_RR;
-		M1575_AM_PUT8(M1575_PCMOCR_REG, cr);
-		dmacr &= ~M1575_DMACR_PCMOPAUSE;
-		M1575_AM_PUT32(M1575_DMACR_REG, dmacr);
-		break;
-
-	default:
-		ATRACE("audio1575_dma_reset() bad channel", chan);
-
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("audio1575_dma_reset() returning", AUDIO_SUCCESS);
-
-	return (AUDIO_SUCCESS);
-
-}	/* audio1575_dma_reset() */
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231.c b/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231.c
deleted file mode 100644
index af340d964d..0000000000
--- a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231.c
+++ /dev/null
@@ -1,3020 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-
-/*
- * audiocs Audio Driver
- *
- * This Audio Driver controls the Crystal CS4231 Codec used on many SPARC
- * platforms. It does not support the CS4231 on Power PCs or x86 PCs. It
- * does support two different DMA engines, the APC and EB2. The code for
- * those DMA engines is split out and a well defined, but private, interface
- * is used to control those DMA engines.
- *
- * For some reason setting the CS4231's registers doesn't always succeed.
- * Therefore every time we set a register we always read it back to make
- * sure it was set. If not we wait a little while and then try again. This
- * is all taken care of in the routines cs4231_put8() and cs4231_reg_select()
- * and the macros OR_SET_BYTE() and AND_SET_BYTE(). We don't worry about
- * the status register because it is cleared by writing anything to it.
- * So it doesn't matter what the value written is.
- *
- * This driver uses the mixer Audio Personality Module to implement audio(7I)
- * and mixer(7I) semantics. Unfortunately this is a single stream Codec,
- * forcing the mixer to do sample rate conversion.
- *
- * This driver supports suspending and resuming. A suspend just stops playing
- * and recording. The play DMA buffers end up getting thrown away, but when
- * you shut down the machine there is a break in the audio anyway, so they
- * won't be missed and it isn't worth the effort to save them. When we resume
- * we always start playing and recording. If they aren't needed they get
- * shut off by the mixer.
- *
- * System power management is supported by this driver. To facilitate
- * this feature the routines audiocs_set_busy() and audiocs_set_idle()
- * are provided.
- *	audiocs_set_busy() is called at the beginning of all audiocs_ad_*()
- *	entry point routines. It blocks if the driver is being suspended.
- *	Once it unblocks it increments a busy count and raises power.
- *	Once this busy count is incremented any calls to suspend the driver
- *	will block until the count goes back to zero.
- *
- *	audiocs_set_idle() is called at the end of all audiocs_ad_*() entry
- *	points. It decrements the busy count. Once that count reaches zero
- *	it wakes up a sleeping suspend.
- *
- * Component power management is also supported by this driver. As long as
- * the busy count raised by audiocs_set_busy() is non-zero or audio is
- * actively playing or recording power can not be lowered.
- *
- * The ad_start_play()/record() routines call pm_busy_component() so that
- * as long as playing/recording is going on the device won't be powered down.
- * The ad_stop_play()/record() routines call pm_idle_component() so that when
- * the busy count goes to 0 the device will be powered down.
- *
- *	NOTE: This module depends on the misc/audiosup and misc/mixer modules
- *		being loaded first.
- */
-
-#include <sys/modctl.h>
-#include <sys/kmem.h>
-#include <sys/stropts.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src2.h>
-#include <sys/audio/impl/audio_4231_impl.h>
-#include <sys/audio/audio_4231.h>
-
-/*
- * Global routines.
- */
-int cs4231_poll_ready(CS_state_t *);
-
-/*
- * Module linkage routines for the kernel
- */
-static int cs4231_getinfo(dev_info_t *, ddi_info_cmd_t, void *, void **);
-static int cs4231_attach(dev_info_t *, ddi_attach_cmd_t);
-static int cs4231_detach(dev_info_t *, ddi_detach_cmd_t);
-static int cs4231_power(dev_info_t *, int, int);
-
-/*
- * Entry point routine prototypes
- */
-static int cs4231_ad_set_config(audiohdl_t, int, int, int, int, int);
-static int cs4231_ad_set_format(audiohdl_t, int, int, int, int, int, int);
-static int cs4231_ad_start_play(audiohdl_t, int);
-static void cs4231_ad_pause_play(audiohdl_t, int);
-static void cs4231_ad_stop_play(audiohdl_t, int);
-static int cs4231_ad_start_record(audiohdl_t, int);
-static void cs4231_ad_stop_record(audiohdl_t, int);
-
-/* Local Routines */
-static int cs4231_init_state(CS_state_t *, dev_info_t *);
-static int cs4231_chip_init(CS_state_t *);
-static void cs4231_get_ports(CS_state_t *, dev_info_t *);
-static int cs4231_set_port(CS_state_t *, int, int);
-static int cs4231_set_gain(CS_state_t *, int, int, int, int);
-static int cs4231_set_monitor_gain(CS_state_t *, int);
-static int cs4231_set_busy(CS_state_t *);
-static void cs4231_set_idle(CS_state_t *);
-static void cs4231_power_up(CS_state_t *);
-static void cs4231_power_down(CS_state_t *);
-
-/*
- * Global variables, but viewable only by this file.
- */
-
-/* anchor for soft state structures */
-static void *cs_statep;
-
-/* driver name, so we don't have to call ddi_driver_name() or hard code strs */
-static char *audiocs_name = CS4231_NAME;
-
-/* File name for the cs4231_put8() and cs4231_reg_select() routines */
-static char *thisfile = __FILE__;
-
-static uint_t cs_mixer_srs[] = {
-	CS4231_SAMPR5510, CS4231_SAMPR48000, 0
-};
-
-static uint_t cs_compat_srs[] = {
-	CS4231_SAMPR5510, CS4231_SAMPR6620, CS4231_SAMPR8000,
-	CS4231_SAMPR9600, CS4231_SAMPR11025, CS4231_SAMPR16000,
-	CS4231_SAMPR18900, CS4231_SAMPR22050, CS4231_SAMPR27420,
-	CS4231_SAMPR32000, CS4231_SAMPR33075, CS4231_SAMPR37800,
-	CS4231_SAMPR44100, CS4231_SAMPR48000, 0
-};
-
-static am_ad_sample_rates_t cs_mixer_sample_rates = {
-	MIXER_SRS_FLAG_SR_LIMITS,
-	cs_mixer_srs
-};
-
-static am_ad_sample_rates_t cs_compat_sample_rates = {
-	MIXER_SRS_FLAG_SR_NOT_LIMITS,
-	cs_compat_srs
-};
-
-static uint_t cs_channels[] = {
-	AUDIO_CHANNELS_MONO, AUDIO_CHANNELS_STEREO, 0
-};
-
-static am_ad_cap_comb_t cs_combinations[] = {
-	{ AUDIO_PRECISION_8, AUDIO_ENCODING_LINEAR },
-	{ AUDIO_PRECISION_8, AUDIO_ENCODING_ULAW },
-	{ AUDIO_PRECISION_8, AUDIO_ENCODING_ALAW },
-	{ AUDIO_PRECISION_16, AUDIO_ENCODING_LINEAR },
-	{ 0 }
-};
-
-static am_ad_entry_t cs_entry = {
-	NULL,			/* ad_setup() */
-	NULL,			/* ad_teardown() */
-	cs4231_ad_set_config,	/* ad_set_config() */
-	cs4231_ad_set_format,	/* ad_set_format() */
-	cs4231_ad_start_play,	/* ad_start_play() */
-	cs4231_ad_pause_play,	/* ad_pause_play() */
-	cs4231_ad_stop_play,	/* ad_stop_play() */
-	cs4231_ad_start_record,	/* ad_start_record() */
-	cs4231_ad_stop_record,	/* ad_stop_record() */
-	NULL,			/* ad_ioctl() */
-	NULL			/* ad_iocdata() */
-};
-
-/* play gain array, converts linear gain to 64 steps of log10 gain */
-static uint8_t cs4231_atten[] = {
-	0x3f,	0x3e,	0x3d,	0x3c,	0x3b,	/* [000] -> [004] */
-	0x3a,	0x39,	0x38,	0x37,	0x36,	/* [005] -> [009] */
-	0x35,	0x34,	0x33,	0x32,	0x31,	/* [010] -> [014] */
-	0x30,	0x2f,	0x2e,	0x2d,	0x2c,	/* [015] -> [019] */
-	0x2b,	0x2a,	0x29,	0x29,	0x28,	/* [020] -> [024] */
-	0x28,	0x27,	0x27,	0x26,	0x26,	/* [025] -> [029] */
-	0x25,	0x25,	0x24,	0x24,	0x23,	/* [030] -> [034] */
-	0x23,	0x22,	0x22,	0x21,	0x21,	/* [035] -> [039] */
-	0x20,	0x20,	0x1f,	0x1f,	0x1f,	/* [040] -> [044] */
-	0x1e,	0x1e,	0x1e,	0x1d,	0x1d,	/* [045] -> [049] */
-	0x1d,	0x1c,	0x1c,	0x1c,	0x1b,	/* [050] -> [054] */
-	0x1b,	0x1b,	0x1a,	0x1a,	0x1a,	/* [055] -> [059] */
-	0x1a,	0x19,	0x19,	0x19,	0x19,	/* [060] -> [064] */
-	0x18,	0x18,	0x18,	0x18,	0x17,	/* [065] -> [069] */
-	0x17,	0x17,	0x17,	0x16,	0x16,	/* [070] -> [074] */
-	0x16,	0x16,	0x16,	0x15,	0x15,	/* [075] -> [079] */
-	0x15,	0x15,	0x15,	0x14,	0x14,	/* [080] -> [084] */
-	0x14,	0x14,	0x14,	0x13,	0x13,	/* [085] -> [089] */
-	0x13,	0x13,	0x13,	0x12,	0x12,	/* [090] -> [094] */
-	0x12,	0x12,	0x12,	0x12,	0x11,	/* [095] -> [099] */
-	0x11,	0x11,	0x11,	0x11,	0x11,	/* [100] -> [104] */
-	0x10,	0x10,	0x10,	0x10,	0x10,	/* [105] -> [109] */
-	0x10,	0x0f,	0x0f,	0x0f,	0x0f,	/* [110] -> [114] */
-	0x0f,	0x0f,	0x0e,	0x0e,	0x0e,	/* [114] -> [119] */
-	0x0e,	0x0e,	0x0e,	0x0e,	0x0d,	/* [120] -> [124] */
-	0x0d,	0x0d,	0x0d,	0x0d,	0x0d,	/* [125] -> [129] */
-	0x0d,	0x0c,	0x0c,	0x0c,	0x0c,	/* [130] -> [134] */
-	0x0c,	0x0c,	0x0c,	0x0b,	0x0b,	/* [135] -> [139] */
-	0x0b,	0x0b,	0x0b,	0x0b,	0x0b,	/* [140] -> [144] */
-	0x0b,	0x0a,	0x0a,	0x0a,	0x0a,	/* [145] -> [149] */
-	0x0a,	0x0a,	0x0a,	0x0a,	0x09,	/* [150] -> [154] */
-	0x09,	0x09,	0x09,	0x09,	0x09,	/* [155] -> [159] */
-	0x09,	0x09,	0x08,	0x08,	0x08,	/* [160] -> [164] */
-	0x08,	0x08,	0x08,	0x08,	0x08,	/* [165] -> [169] */
-	0x08,	0x07,	0x07,	0x07,	0x07,	/* [170] -> [174] */
-	0x07,	0x07,	0x07,	0x07,	0x07,	/* [175] -> [179] */
-	0x06,	0x06,	0x06,	0x06,	0x06,	/* [180] -> [184] */
-	0x06,	0x06,	0x06,	0x06,	0x05,	/* [185] -> [189] */
-	0x05,	0x05,	0x05,	0x05,	0x05,	/* [190] -> [194] */
-	0x05,	0x05,	0x05,	0x05,	0x04,	/* [195] -> [199] */
-	0x04,	0x04,	0x04,	0x04,	0x04,	/* [200] -> [204] */
-	0x04,	0x04,	0x04,	0x04,	0x03,	/* [205] -> [209] */
-	0x03,	0x03,	0x03,	0x03,	0x03,	/* [210] -> [214] */
-	0x03,	0x03,	0x03,	0x03,	0x03,	/* [215] -> [219] */
-	0x02,	0x02,	0x02,	0x02,	0x02,	/* [220] -> [224] */
-	0x02,	0x02,	0x02,	0x02,	0x02,	/* [225] -> [229] */
-	0x02,	0x01,	0x01,	0x01,	0x01,	/* [230] -> [234] */
-	0x01,	0x01,	0x01,	0x01,	0x01,	/* [235] -> [239] */
-	0x01,	0x01,	0x01,	0x00,	0x00,	/* [240] -> [244] */
-	0x00,	0x00,	0x00,	0x00,	0x00,	/* [245] -> [249] */
-	0x00,	0x00,	0x00,	0x00,	0x00,	/* [250] -> [254] */
-	0x00					/* [255] */
-};
-
-/*
- * STREAMS Structures
- */
-
-/* STREAMS driver id and limit value structure */
-static struct module_info cs4231_modinfo = {
-	CS4231_IDNUM,		/* module ID number */
-	CS4231_NAME,		/* module name */
-	CS4231_MINPACKET,	/* minimum packet size */
-	CS4231_MAXPACKET,	/* maximum packet size */
-	CS4231_HIWATER,		/* high water mark */
-	CS4231_LOWATER		/* low water mark */
-};
-
-/* STREAMS queue processing procedures structures */
-/* read queue */
-static struct qinit cs4231_rqueue = {
-	audio_sup_rput,		/* put procedure */
-	audio_sup_rsvc,		/* service procedure */
-	audio_sup_open,		/* open procedure */
-	audio_sup_close,	/* close procedure */
-	NULL,			/* unused */
-	&cs4231_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* write queue */
-static struct qinit cs4231_wqueue = {
-	audio_sup_wput,		/* put procedure */
-	audio_sup_wsvc,		/* service procedure */
-	NULL,			/* open procedure */
-	NULL,			/* close procedure */
-	NULL,			/* unused */
-	&cs4231_modinfo,	/* module parameters */
-	NULL			/* module statistics */
-};
-
-/* STREAMS entity declaration structure */
-static struct streamtab cs4231_str_info = {
-	&cs4231_rqueue,		/* read queue */
-	&cs4231_wqueue,		/* write queue */
-	NULL,			/* mux lower read queue */
-	NULL,			/* mux lower write queue */
-};
-
-/*
- * DDI Structures
- */
-
-/* Entry points structure */
-static struct cb_ops cs4231_cb_ops = {
-	nulldev,		/* cb_open */
-	nulldev,		/* cb_close */
-	nodev,			/* cb_strategy */
-	nodev,			/* cb_print */
-	nodev,			/* cb_dump */
-	nodev,			/* cb_read */
-	nodev,			/* cb_write */
-	nodev,			/* cb_ioctl */
-	nodev,			/* cb_devmap */
-	nodev,			/* cb_mmap */
-	nodev,			/* cb_segmap */
-	nochpoll,		/* cb_chpoll */
-	ddi_prop_op,		/* cb_prop_op */
-	&cs4231_str_info,	/* cb_str */
-	D_NEW|D_MP|D_64BIT,	/* cb_flag */
-	CB_REV,			/* cb_rev */
-	nodev,			/* cb_aread */
-	nodev,			/* cb_arwite */
-};
-
-/* Device operations structure */
-static struct dev_ops cs4231_dev_ops = {
-	DEVO_REV,		/* devo_rev */
-	0,			/* devo_refcnt */
-	cs4231_getinfo,		/* devo_getinfo */
-	nulldev,		/* devo_identify - obsolete */
-	nulldev,		/* devo_probe - not needed */
-	cs4231_attach,		/* devo_attach */
-	cs4231_detach,		/* devo_detach */
-	nodev,			/* devo_reset */
-	&cs4231_cb_ops,		/* devi_cb_ops */
-	NULL,			/* devo_bus_ops */
-	cs4231_power,		/* devo_power */
-	ddi_quiesce_not_supported,	/* devo_quiesce */
-};
-
-/* Linkage structure for loadable drivers */
-static struct modldrv cs4231_modldrv = {
-	&mod_driverops,		/* drv_modops */
-	CS4231_MOD_NAME,	/* drv_linkinfo */
-	&cs4231_dev_ops		/* drv_dev_ops */
-};
-
-/* Module linkage structure */
-static struct modlinkage cs4231_modlinkage = {
-	MODREV_1,			/* ml_rev */
-	(void *)&cs4231_modldrv,	/* ml_linkage */
-	NULL				/* NULL terminates the list */
-};
-
-
-/* *******  Loadable Module Configuration Entry Points  ********************* */
-
-/*
- * _init()
- *
- * Description:
- *	Driver initialization, called when driver is first loaded.
- *	This is how access is initially given to all the static structures.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	ddi_soft_state_init() status, see ddi_soft_state_init(9f), or
- *	mod_install() status, see mod_install(9f)
- */
-int
-_init(void)
-{
-	int		error;
-
-	ATRACE("in audiocs _init()", 0);
-
-	/* initialize the soft state */
-	if ((error = ddi_soft_state_init(&cs_statep, sizeof (CS_state_t), 0)) !=
-	    0) {
-		ATRACE("audiocs ddi_soft_state_init() failed", cs_statep);
-		return (error);
-	}
-
-	if ((error = mod_install(&cs4231_modlinkage)) != 0) {
-		ddi_soft_state_fini(&cs_statep);
-	}
-
-	ATRACE("audiocs _init() cs_statep", cs_statep);
-	ATRACE_32("audiocs _init() returning", error);
-
-	return (error);
-}
-
-/*
- * _fini()
- *
- * Description:
- *	Module de-initialization, called when the driver is to be unloaded.
- *
- * Arguments:
- *	None
- *
- * Returns:
- *	mod_remove() status, see mod_remove(9f)
- */
-int
-_fini(void)
-{
-	int		error;
-
-	ATRACE("in audiocs _fini()", cs_statep);
-
-	if ((error = mod_remove(&cs4231_modlinkage)) != 0) {
-		return (error);
-	}
-
-	/* free the soft state internal structures */
-	ddi_soft_state_fini(&cs_statep);
-
-	ATRACE_32("audiocs _fini() returning", error);
-
-	return (0);
-}
-
-/*
- * _info()
- *
- * Description:
- *	Module information, returns infomation about the driver.
- *
- * Arguments:
- *	modinfo *modinfop	Pointer to the opaque modinfo structure
- *
- * Returns:
- *	mod_info() status, see mod_info(9f)
- */
-int
-_info(struct modinfo *modinfop)
-{
-	int		error;
-
-	ATRACE("in audiocs _info()", 0);
-
-	error = mod_info(&cs4231_modlinkage, modinfop);
-
-	ATRACE_32("audiocs _info() returning", error);
-
-	return (error);
-}
-
-
-/* *******  Driver Entry Points  ******************************************** */
-/*
- * cs4231_getinfo()
- */
-/*ARGSUSED*/
-static int
-cs4231_getinfo(dev_info_t *dip, ddi_info_cmd_t infocmd,
-	void *arg, void **result)
-{
-	CS_state_t	*state;
-	int error = DDI_FAILURE;
-	int instance;
-
-	switch (infocmd) {
-	case DDI_INFO_DEVT2DEVINFO:
-		instance = audio_sup_devt_to_instance((dev_t)arg);
-		if ((state = ddi_get_soft_state(cs_statep,
-		    instance)) != NULL) {
-			*result = state->cs_dip;
-			error = DDI_SUCCESS;
-		} else {
-			*result = NULL;
-		}
-		break;
-	case DDI_INFO_DEVT2INSTANCE:
-		*result = (void *)(uintptr_t)
-		    audio_sup_devt_to_instance((dev_t)arg);
-		error = DDI_SUCCESS;
-		break;
-	default:
-		break;
-	}
-
-	return (error);
-}
-
-/*
- * cs4231_attach()
- *
- * Description:
- *	Attach an instance of the CS4231 driver. This routine does the device
- *	dependent attach tasks. When it is complete it calls
- *	audio_sup_register() and am_attach() so they may do their work.
- *
- *	NOTE: mutex_init() no longer needs a name string, so set
- *		to NULL to save kernel space.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_attach_cmd_t cmd	Attach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was initialized properly
- *	DDI_FAILURE		The driver couldn't be initialized properly
- */
-static int
-cs4231_attach(dev_info_t *dip, ddi_attach_cmd_t cmd)
-{
-	CS_state_t		*state;
-	audiohdl_t		ahandle;
-	audio_sup_reg_data_t	data;
-	int			instance;
-
-	ATRACE("in cs_attach()", dip);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("cs_attach() instance", instance);
-	ATRACE("cs_attach() cs_statep", cs_statep);
-
-	switch (cmd) {
-	case DDI_ATTACH:
-		break;
-	case DDI_RESUME:
-		ATRACE("cs_attach() DDI_RESUME", NULL);
-
-		/* we've already allocated the state structure so get ptr */
-		if ((state = ddi_get_soft_state(cs_statep, instance)) == NULL) {
-			audio_sup_log(NULL, CE_WARN,
-			    "!%s%d: attach() RESUME get soft state failed",
-			    audiocs_name, instance);
-			return (DDI_FAILURE);
-		}
-
-		ASSERT(dip == state->cs_dip);
-		ASSERT(!mutex_owned(&state->cs_lock));
-
-		ahandle = state->cs_ahandle;
-
-		/* power up the Codec */
-		ASSERT(state->cs_powered == CS4231_PWR_OFF);
-		(void) pm_busy_component(state->cs_dip, CS4231_COMPONENT);
-		if (pm_raise_power(dip, CS4231_COMPONENT, CS4231_PWR_ON)
-		    == DDI_FAILURE) {
-			/* match the busy call above */
-			(void) pm_idle_component(state->cs_dip,
-			    CS4231_COMPONENT);
-			audio_sup_log(ahandle, CE_WARN,
-			    "!attach() DDI_RESUME failed");
-			return (DDI_FAILURE);
-		}
-		mutex_enter(&state->cs_lock);
-
-		ASSERT(mutex_owned(&state->cs_lock));
-		ASSERT(state->cs_suspended == CS4231_SUSPENDED);
-
-		state->cs_suspended = CS4231_NOT_SUSPENDED;
-
-		cv_broadcast(&state->cs_cv);	/* let entry points continue */
-
-		mutex_exit(&state->cs_lock);
-
-		/* now restart playing and recording */
-		if (audio_sup_restore_state(ahandle, AUDIO_ALL_DEVICES,
-		    AUDIO_BOTH) == AUDIO_FAILURE) {
-			audio_sup_log(ahandle, CE_WARN,
-			    "!attach() audio restart failed");
-		}
-
-		/* we're no longer busy */
-		ASSERT(state->cs_powered == CS4231_PWR_ON);
-		(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
-
-		ATRACE("cs_attach() DDI_RESUME succeeded", NULL);
-		ASSERT(!mutex_owned(&state->cs_lock));
-
-		return (DDI_SUCCESS);
-	default:
-		audio_sup_log(NULL, CE_NOTE,
-		    "!%s%d: attach() unknown command 0x%x", audiocs_name,
-		    instance, cmd);
-		return (DDI_FAILURE);
-	}
-
-	/* allocate the state structure */
-	if (ddi_soft_state_zalloc(cs_statep, instance) == DDI_FAILURE) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() soft state allocate failed", audiocs_name,
-		    instance);
-		return (DDI_FAILURE);
-	}
-
-	/*
-	 * WARNING: From here on all errors require that we free memory,
-	 *	including the state structure.
-	 */
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(cs_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: attach() get soft state failed", audiocs_name,
-		    instance);
-		goto error_mem;
-	}
-
-	/* call audiosup module registration routine */
-	ATRACE("cs_attach() calling audio_sup_register()", NULL);
-	data.asrd_version = AUDIOSUP_VERSION;
-	data.asrd_key = NULL;
-	if ((state->cs_ahandle = audio_sup_register(dip, &data)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: cs4231_attach() audio_sup_register() failed",
-		    audiocs_name, instance);
-		goto error_mem;
-	}
-
-	ahandle = state->cs_ahandle;
-
-	/* initialize the audio state structures */
-	if ((cs4231_init_state(state, dip)) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!attach() init_state() failed");
-		goto error_audiosup;
-	}
-
-	/* initialize the audio chip */
-	ATRACE("cs_attach() calling chip_init()", NULL);
-	if ((cs4231_chip_init(state)) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN, "!attach() chip_init() failed");
-		goto error_destroy;
-	}
-
-	/* save private state */
-	audio_sup_set_private(ahandle, state);
-
-	/* call the mixer attach() routine */
-	ATRACE("cs_attach() calling am_attach()", &state->cs_ad_info);
-	if (am_attach(ahandle, cmd, &state->cs_ad_info) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN, "!attach() am_attach() failed");
-		goto error_destroy;
-	}
-
-	/* set up kernel statistics */
-	if ((state->cs_ksp = kstat_create(audiocs_name, instance, audiocs_name,
-	    "controller", KSTAT_TYPE_INTR, 1, KSTAT_FLAG_PERSISTENT)) != NULL) {
-		kstat_install(state->cs_ksp);
-	}
-
-	/* we're ready, set up the interrupt handler */
-	ATRACE("cs_attach() calling DMA_ADD_INTR()", state);
-	if (CS4231_DMA_ADD_INTR(state) != AUDIO_SUCCESS) {
-		ATRACE("cs_attach() DMA_ADD_INTR() failed", state);
-		goto error_kstat;
-	}
-
-	/* everything worked out, so report the device */
-	ddi_report_dev(dip);
-
-	ATRACE("cs_attach() returning success", state);
-
-	return (DDI_SUCCESS);
-
-error_kstat:
-	ATRACE("cs_attach() error_kstat", state);
-	if (state->cs_ksp) {
-		kstat_delete(state->cs_ksp);
-	}
-
-	(void) am_detach(ahandle, DDI_DETACH);
-
-error_destroy:
-	ATRACE("cs_attach() error_destroy", state);
-	CS4231_DMA_UNMAP_REGS(state);
-	mutex_destroy(&state->cs_lock);
-	cv_destroy(&state->cs_cv);
-
-error_audiosup:
-	ATRACE("cs_attach() error_audiosup", state);
-	(void) audio_sup_unregister(ahandle);
-
-error_mem:
-	ATRACE("cs_attach() error_mem", state);
-	ddi_soft_state_free(cs_statep, instance);
-
-	ATRACE("cs_attach() returning failure", NULL);
-
-	return (DDI_FAILURE);
-
-}	/* cs4231_attach() */
-
-/*
- * cs4231_detach()
- *
- * Description:
- *	Detach an instance of the CS4231 driver. After the Codec is detached
- *	we call am_detach() and audio_sup_unregister() so they may do their
- *	work.
- *
- *	Power management is pretty simple. If active we fail, otherwise
- *	we save the Codec state.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's dev_info struct
- *	ddi_detach_cmd_t cmd	Detach command
- *
- * Returns:
- *	DDI_SUCCESS		The driver was detached
- *	DDI_FAILURE		The driver couldn't be detached
- */
-static int
-cs4231_detach(dev_info_t *dip, ddi_detach_cmd_t cmd)
-{
-	CS_state_t		*state;
-	audiohdl_t		ahandle;
-	ddi_acc_handle_t	handle;
-	int			instance;
-
-	ATRACE_32("in cs_detach()", cmd);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("cs_detach() instance", instance);
-	ATRACE("cs_detach() cs_statep", cs_statep);
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(cs_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: detach() get soft state failed", audiocs_name,
-		    instance);
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	handle = state->cs_handles.cs_codec_hndl;
-	ahandle = state->cs_ahandle;
-
-	switch (cmd) {
-	case DDI_DETACH:
-		break;
-	case DDI_SUSPEND:
-		ATRACE("cs_detach() DDI_SUSPEND", NULL);
-
-		mutex_enter(&state->cs_lock);
-
-		ASSERT(state->cs_suspended == CS4231_NOT_SUSPENDED);
-		state->cs_suspended = CS4231_SUSPENDED;	/* stop new ops */
-
-		/* wait for current operations to complete */
-		while (state->cs_busy_cnt != 0) {
-			cv_wait(&state->cs_cv, &state->cs_lock);
-		}
-
-		if (state->cs_powered == CS4231_PWR_ON) {
-			/* stop playing and recording */
-			CS4231_DMA_STOP_RECORD(state);
-			CS4231_DMA_STOP_PLAY(state);
-
-			/* now we can power down the Codec */
-			cs4231_power_down(state);
-
-			if (audio_sup_save_state(ahandle, AUDIO_ALL_DEVICES,
-			    AUDIO_BOTH) == AUDIO_FAILURE) {
-				audio_sup_log(ahandle, CE_WARN,
-				    "!detach() audio save failed");
-			}
-		}
-		mutex_exit(&state->cs_lock);
-
-		ATRACE("cs_detach() SUSPEND successful", state);
-		ASSERT(!mutex_owned(&state->cs_lock));
-		return (DDI_SUCCESS);
-	default:
-		ATRACE_32("cs_detach() unknown command failure", cmd);
-		audio_sup_log(ahandle, CE_NOTE,
-		    "!detach() unknown command 0x%x", cmd);
-		return (DDI_FAILURE);
-	}
-
-	if (state->cs_powered == CS4231_PWR_ON) {
-		/*
-		 * Make sure the Codec and DMA engine are off.
-		 */
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR, INTC_REG,
-		    __LINE__, thisfile);
-		AND_SET_BYTE(handle, &CS4231_IDR, ~(INTC_PEN|INTC_CEN),
-		    INTC_VALID_MASK);
-
-		/* make sure the DMA engine isn't going to do anything */
-		CS4231_DMA_RESET(state);
-
-		/*
-		 * power down the device, no reason to waste power without
-		 * a driver
-		 */
-		(void) pm_lower_power(state->cs_dip, CS4231_COMPONENT,
-		    CS4231_PWR_OFF);
-	}
-
-	/*
-	 * unregister the interrupt. That way we can't get called by and
-	 * interrupt after the audio framework is removed.
-	 */
-	ATRACE("cs_detach() calling DMA_REM_INTR", state);
-	CS4231_DMA_REM_INTR(dip, state);
-
-	/*
-	 * Call the mixer detach routine to tear down the mixer before
-	 * we lose the hardware.
-	 */
-	ATRACE("cs_detach() calling am_detach()", dip);
-	if (am_detach(ahandle, cmd) == AUDIO_FAILURE) {
-		ATRACE_32("cs_detach() am_detach() failed", cmd);
-		return (DDI_FAILURE);
-	}
-	ATRACE("cs_detach() calling audio_sup_unregister()", dip);
-	if (audio_sup_unregister(ahandle) == AUDIO_FAILURE) {
-		ATRACE_32("cs_detach() audio_sup_unregister() failed", cmd);
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(state->cs_busy_cnt == 0);
-
-	/* unmap the registers */
-	CS4231_DMA_UNMAP_REGS(state);
-
-	/* free the kernel statistics structure */
-	if (state->cs_ksp) {
-		kstat_delete(state->cs_ksp);
-	}
-	state->cs_ksp = NULL;
-
-	/* destroy the state mutex */
-	mutex_destroy(&state->cs_lock);
-	cv_destroy(&state->cs_cv);
-
-	/* free the memory for the state pointer */
-	ddi_soft_state_free(cs_statep, instance);
-
-	ATRACE("cs_detach() returning success", cs_statep);
-
-	return (DDI_SUCCESS);
-
-}	/* cs4231_detach() */
-
-/*
- * cs4231_power()
- *
- * Description:
- *	This routine is used to turn the power to the Codec on and off.
- *	The different DMA engines have different ways to turn on/off the
- *	power to the Codec. Therefore we call the DMA engine specific code
- *	to do the work, if we need to make a change.
- *
- *	If the level is CS4231_PWR_OFF then we call cs4231_power_down(). If the
- *	level is CS4231_PWR_ON then we call cs4231_power_up().
- *
- *	This routine doesn't stop or restart play and record. Other routines
- *	are responsible for that.
- *
- * Arguments:
- *	def_info_t	*dip		Ptr to the device's dev_info structure
- *	int		component	Which component to power up/down
- *	int		level		The power level for the component
- *
- * Returns:
- *	DDI_SUCCESS		Power level changed, we always succeed
- */
-static int
-cs4231_power(dev_info_t *dip, int component, int level)
-{
-	CS_state_t		*state;
-	int			instance;
-	int			rc = DDI_FAILURE;
-
-	ATRACE("in cs_power()", dip);
-	ATRACE("cs_power() cs_statep", cs_statep);
-	ASSERT(component == 0);
-
-	instance = ddi_get_instance(dip);
-	ATRACE_32("cs_power() instance", instance);
-
-	/* get the state structure */
-	if ((state = ddi_get_soft_state(cs_statep, instance)) == NULL) {
-		audio_sup_log(NULL, CE_WARN,
-		    "!%s%d: power() get soft state failed", audiocs_name,
-		    instance);
-		return (DDI_FAILURE);
-	}
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* make sure we have some work to do */
-	mutex_enter(&state->cs_lock);
-
-	/* check the level change to see what we need to do */
-	if (level == CS4231_PWR_OFF && state->cs_powered == CS4231_PWR_ON) {
-
-		/* don't power off if we're busy */
-		if (state->cs_busy_cnt) {
-			/* device is busy, so don't power off */
-			mutex_exit(&state->cs_lock);
-
-			/* reset the timer */
-			(void) pm_idle_component(dip, CS4231_COMPONENT);
-
-			ATRACE("cs_power() power off failed, busy",
-			    state->cs_busy_cnt);
-			ASSERT(rc == DDI_FAILURE);
-
-			goto done;
-		}
-		/* power down and save the state */
-		cs4231_power_down(state);
-
-	} else if (level == CS4231_PWR_ON &&
-	    state->cs_powered == CS4231_PWR_OFF) {
-
-		/* power up */
-		cs4231_power_up(state);
-
-#ifdef DEBUG
-	} else {
-		ATRACE_32("cs_power() no change to make", level);
-#endif
-	}
-
-	mutex_exit(&state->cs_lock);
-
-	rc = DDI_SUCCESS;
-
-done:
-	ASSERT(!mutex_owned(&state->cs_lock));
-	ATRACE("cs4231_power() done", level);
-
-	return (rc);
-
-}	/* cs4231_power() */
-
-
-/* *******  Audio Driver Entry Point Routines ******************************* */
-
-/*
- * cs4231_ad_pause_play()
- *
- * Description:
- *	This routine pauses the play DMA engine.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-cs4231_ad_pause_play(audiohdl_t ahandle, int stream)
-{
-	CS_state_t		*state;
-
-	ATRACE("in cs_ad_pause_play()", ahandle);
-	ATRACE("cs_ad_pause_play() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!pause_play() set_busy() failed");
-		return;
-	}
-
-	/* we need to protect the state structure */
-	mutex_enter(&state->cs_lock);
-
-	ATRACE("cs_ad_pause_play() calling DMA_PAUSE_PLAY()", state);
-	CS4231_DMA_PAUSE_PLAY(state);
-	ATRACE("cs_ad_pause_play() DMA_PAUSE_PLAY() returned", state);
-
-	mutex_exit(&state->cs_lock);
-
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE("cs_ad_pause_play() returning", state);
-
-}	/* cs4231_ad_pause_play() */
-
-/*
- * cs4231_ad_set_config()
- *
- * Description:
- *	This routine is used to set new Codec parameters, except the data
- *	format which has it's own routine. If the Codec doesn't support a
- *	particular parameter and it is asked to set it then we return
- *	AUDIO_FAILURE.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *	int		command		The configuration to set
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		arg1		Argument #1
- *	int		arg2		Argument #2, not always needed
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The Codec parameter has not been set, or the
- *				parameter couldn't be set
- */
-/*ARGSUSED*/
-static int
-cs4231_ad_set_config(audiohdl_t ahandle, int stream, int command, int dir,
-	int arg1, int arg2)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE_32("in cs_ad_set_config()", command);
-	ATRACE_32("cs_ad_set_config() stream", stream);
-	ATRACE_32("cs_ad_set_config() command", command);
-	ATRACE_32("cs_ad_set_config() dir", dir);
-	ATRACE_32("cs_ad_set_config() arg1", arg1);
-	ATRACE_32("cs_ad_set_config() arg2", arg2);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* wait on suspend, power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!set_config() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	/* CAUTION: From here on we must goto done to exit. */
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	switch (command) {
-	case AM_SET_GAIN:
-		/*
-		 * Set the gain for a channel. The audio mixer calculates the
-		 * impact, if any, on the channel's gain.
-		 *
-		 *	0 <= gain <= AUDIO_MAX_GAIN
-		 *
-		 *	arg1 --> gain
-		 *	arg2 --> channel #, 0 = left, 1 = right
-		 */
-
-		rc = cs4231_set_gain(state, stream, dir, arg1, arg2);
-		break;
-
-	case AM_SET_PORT:
-		/*
-		 * Enable/disable the input or output ports. The audio mixer
-		 * enforces exclusiveness of ports, as well as which ports
-		 * are modifyable. We just turn on the ports that match the
-		 * bits.
-		 *
-		 *	arg1 --> port bit pattern
-		 *	arg2 --> not used
-		 */
-
-		rc = cs4231_set_port(state, dir, arg1);
-		break;
-
-	case AM_SET_MONITOR_GAIN:
-		/*
-		 * Set the loopback monitor gain.
-		 *
-		 *	0 <= gain <= AUDIO_MAX_GAIN
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> gain
-		 *	arg2 --> not used
-		 */
-
-		rc = cs4231_set_monitor_gain(state, arg1);
-		break;
-
-	case AM_OUTPUT_MUTE:
-		/*
-		 * Mute or enable the output.
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> ~0 = mute, 0 = unmute
-		 *	arg2 --> not used
-		 */
-		mutex_enter(&state->cs_lock);
-
-		if (arg1) {
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    LDACO_REG, __LINE__, thisfile);
-			OR_SET_BYTE(handle, &CS4231_IDR, LDACO_LDM,
-			    LDAC0_VALID_MASK);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    RDACO_REG, __LINE__, thisfile);
-			OR_SET_BYTE(handle, &CS4231_IDR, RDACO_RDM,
-			    RDAC0_VALID_MASK);
-			state->cs_output_muted = B_TRUE;
-		} else { /* Unmute */
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    LDACO_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~LDACO_LDM,
-			    LDAC0_VALID_MASK);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    RDACO_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~RDACO_RDM,
-			    RDAC0_VALID_MASK);
-			state->cs_output_muted = B_FALSE;
-		}
-		mutex_exit(&state->cs_lock);
-
-		rc = AUDIO_SUCCESS;
-		goto done;
-
-	case AM_MIC_BOOST:
-		/*
-		 * Enable or disable the mic's 20 dB boost preamplifier.
-		 *
-		 *	dir ---> N/A
-		 *	arg1 --> ~0 == enable, 0 == disabled
-		 *	arg2 --> not used
-		 */
-		mutex_enter(&state->cs_lock);
-		if (arg1) {
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    LADCI_REG, __LINE__, thisfile);
-			OR_SET_BYTE(handle, &CS4231_IDR, LADCI_LMGE,
-			    LADCI_VALID_MASK);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    RADCI_REG, __LINE__, thisfile);
-			OR_SET_BYTE(handle, &CS4231_IDR, RADCI_RMGE,
-			    RADCI_VALID_MASK);
-			state->cs_ad_info.ad_add_mode |= AM_ADD_MODE_MIC_BOOST;
-		} else {
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    LADCI_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~LADCI_LMGE,
-			    LADCI_VALID_MASK);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    RADCI_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~RADCI_RMGE,
-			    RADCI_VALID_MASK);
-			state->cs_ad_info.ad_add_mode &= ~AM_ADD_MODE_MIC_BOOST;
-		}
-		mutex_exit(&state->cs_lock);
-
-		rc = AUDIO_SUCCESS;
-		goto done;
-
-	default:
-		/*
-		 * We let default catch commands we don't support, as well
-		 * as bad commands.
-		 */
-		ATRACE_32("cs_ad_set_config() unsupported command", command);
-		goto done;
-	}
-
-done:
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE_32("cs4231_ad_set_config() returning", rc);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (rc);
-
-}	/* cs4231_ad_set_config() */
-
-/*
- * cs4231_ad_set_format()
- *
- * Description:
- *	This routine is used to set a new Codec data format.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		sample_rate	Data sample rate
- *	int		channels	Number of channels, 1 or 2
- *	int		precision	Bits per sample, 8 or 16
- *	int		encoding	Encoding method, u-law, A-law and linear
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec data format has been set
- *	AUDIO_FAILURE		The Codec data format has not been set, or the
- *				data format couldn't be set
- */
-/*ARGSUSED*/
-static int
-cs4231_ad_set_format(audiohdl_t ahandle, int stream, int dir,
-	int sample_rate, int channels, int precision, int encoding)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-	uint8_t			mask;
-	uint8_t			value;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE_32("in cs_ad_set_format()", sample_rate);
-	ATRACE("cs_ad_set_format() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* wait on suspend, power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!set_format() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/*
-	 * CAUTION: From here on we must goto done to exit.
-	 */
-
-	if (dir == AUDIO_PLAY) {	/* sample rate set on play side only */
-		switch (sample_rate) {
-		case CS4231_SAMPR5510:		value = FS_5510; break;
-		case CS4231_SAMPR6620:		value = FS_6620; break;
-		case CS4231_SAMPR8000:		value = FS_8000; break;
-		case CS4231_SAMPR9600:		value = FS_9600; break;
-		case CS4231_SAMPR11025:		value = FS_11025; break;
-		case CS4231_SAMPR16000:		value = FS_16000; break;
-		case CS4231_SAMPR18900:		value = FS_18900; break;
-		case CS4231_SAMPR22050:		value = FS_22050; break;
-		case CS4231_SAMPR27420:		value = FS_27420; break;
-		case CS4231_SAMPR32000:		value = FS_32000; break;
-		case CS4231_SAMPR33075:		value = FS_33075; break;
-		case CS4231_SAMPR37800:		value = FS_37800; break;
-		case CS4231_SAMPR44100:		value = FS_44100; break;
-		case CS4231_SAMPR48000:		value = FS_48000; break;
-		default:
-			ATRACE_32("cs_ad_set_format() bad sample rate",
-			    sample_rate);
-			goto done;
-		}
-	} else {
-		value = 0;
-	}
-
-	/* if not mono then must be stereo, i.e., the default */
-	if (channels == AUDIO_CHANNELS_STEREO) {
-		ATRACE_32("cs_ad_set_format() STEREO", channels);
-		value |= PDF_STEREO;
-	} else if (channels != AUDIO_CHANNELS_MONO) {
-		ATRACE_32("cs_ad_set_format() bad # of channels", channels);
-		goto done;
-#ifdef DEBUG
-	} else {
-		ATRACE_32("cs_ad_set_format() MONO", channels);
-#endif
-	}
-
-	if (precision == AUDIO_PRECISION_8) {
-		ATRACE_32("cs_ad_set_format() 8-bit", precision);
-		switch (encoding) {
-		case AUDIO_ENCODING_ULAW:
-			value |= PDF_ULAW8;
-			break;
-		case AUDIO_ENCODING_ALAW:
-			value |= PDF_ALAW8;
-			break;
-		case AUDIO_ENCODING_LINEAR:
-			value |= PDF_LINEAR8;
-			break;
-		default:
-			goto done;
-		}
-	} else {	/* 16 bit, default, and there is only one choice */
-		ATRACE_32("cs_ad_set_format() 16-bit", precision);
-		if (encoding != AUDIO_ENCODING_LINEAR) {
-			goto done;
-		}
-
-		value |= PDF_LINEAR16BE;
-	}
-
-	mutex_enter(&state->cs_lock);
-	if (dir == AUDIO_PLAY) {	/* play side */
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    (FSDF_REG | IAR_MCE), __LINE__, thisfile);
-		ATRACE_8("cs_ad_set_format() programming FSDF_REG", value);
-		state->cs_play_sr = sample_rate;
-		state->cs_play_ch = channels;
-		state->cs_play_prec = precision;
-		state->cs_play_enc = encoding;
-		state->cs_save_pe = value;
-		mask = FSDF_VALID_MASK;
-	} else {			/* capture side */
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    (CDF_REG | IAR_MCE), __LINE__, thisfile);
-		ATRACE_8("cs_ad_set_format() programming CDF_REG", value);
-		state->cs_record_sr = sample_rate;
-		state->cs_record_ch = channels;
-		state->cs_record_prec = precision;
-		state->cs_record_enc = encoding;
-		state->cs_save_ce = value;
-		mask = CDF_VALID_MASK;
-	}
-
-	cs4231_put8(ahandle, handle, &CS4231_IDR, value, mask,
-	    __LINE__, thisfile);
-
-	(void) cs4231_poll_ready(state);
-
-	/* clear the mode change bit */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, FSDF_REG, __LINE__,
-	    thisfile);
-	mutex_exit(&state->cs_lock);
-
-	ATRACE_32("cs_ad_set_format() returning", sample_rate);
-
-	rc = AUDIO_SUCCESS;
-
-done:
-	/* we're no longer busy */
-	cs4231_set_idle(state);
-
-	ATRACE_32("cs_ad_set_format() returning", rc);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (rc);
-
-}	/* cs4231_ad_set_format() */
-
-/*
- * cs4231_ad_start_play()
- *
- * Description:
- *	This routine starts the play DMA engine. It checks to make sure the
- *	DMA engine is off before it does anything, otherwise it may mess
- *	things up.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *
- * Returns:
- *	AUDIO_SUCCESS		Playing started/restarted
- *	AUDIO_FAILURE		Audio not restarted, no audio to play
- */
-/*ARGSUSED*/
-static int
-cs4231_ad_start_play(audiohdl_t ahandle, int stream)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-	int			rc;
-
-	ATRACE("in cs_ad_start_play()", ahandle);
-	ATRACE("cs_ad_start_play() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!start_play() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* we need to protect the state structure */
-	mutex_enter(&state->cs_lock);
-	ASSERT(state->cs_powered == CS4231_PWR_ON);
-
-	/* see if we are already playing */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, INTC_REG, __LINE__,
-	    thisfile);
-	if (INTC_PEN & ddi_get8(handle, &CS4231_IDR)) {
-		mutex_exit(&state->cs_lock);
-		ATRACE("cs_ad_start_play() already playing", NULL);
-		rc = AUDIO_SUCCESS;
-		goto done;
-	}
-
-	if (state->cs_flags & PDMA_ENGINE_INITIALIZED) {
-		ATRACE("cs_ad_start_play() calling DMA_RESTART_PLAY()", state);
-		CS4231_DMA_RESTART_PLAY(state);
-		ATRACE_32("cs_ad_start_play() DMA_RESTART_PLAY() returned", 0);
-		rc = AUDIO_SUCCESS;
-	} else {
-		/*
-		 * The newer versions of the EB2 DMA engine reset on a non-even
-		 * sample boundary. Then when it restarts it'll be in mid sample
-		 * which results in loud static. When we start again we reload
-		 * the format register, which resets the Codec, starting on an
-		 * even boundary, and thus no static. We end up doing this for
-		 * the APC DMA engine as well, but it's harmless.
-		 *
-		 * CAUTION: Don't do this for record. It causes SunVTS to
-		 *	fail. Also, do not reset the DMA engine if record is
-		 *	active. This occasionally upsets everything.
-		 */
-		if (!(state->cs_flags & RDMA_ENGINE_INITIALIZED)) {
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    (FSDF_REG | IAR_MCE), __LINE__, thisfile);
-			cs4231_put8(ahandle, handle, &CS4231_IDR,
-			    state->cs_save_pe, FSDF_VALID_MASK,
-			    __LINE__, thisfile);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    FSDF_REG, __LINE__, thisfile);
-			ATRACE("cs_ad_start_play() play DMA engine reset",
-			    state);
-		}
-
-		ATRACE("cs_ad_start_play() calling DMA_START_PLAY()", state);
-		rc = CS4231_DMA_START_PLAY(state);
-		ATRACE_32("cs_ad_start_play() DMA_START_PLAY() returned", rc);
-
-		if (rc == AUDIO_SUCCESS) {
-			ATRACE("cs_ad_start_play() programming Codec to play",
-			    state);
-			cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-			    INTC_REG, __LINE__, thisfile);
-			OR_SET_BYTE(handle, &CS4231_IDR, INTC_PEN,
-			    INTC_VALID_MASK);
-
-			ATRACE_8("cs_ad_start_play() Codec INTC_REG",
-			    ddi_get8(handle, &CS4231_IDR));
-			(void) pm_busy_component(state->cs_dip,
-			    CS4231_COMPONENT);
-			state->cs_flags |= PLAY_ACTIVE;
-#ifdef DEBUG
-		} else {
-			ATRACE("cs_ad_start_play() Codec not started", rc);
-#endif
-		}
-
-	}
-
-	mutex_exit(&state->cs_lock);
-
-done:
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE("cs4231_ad_start_play() returning", rc);
-
-	return (rc);
-
-}	/* cs4231_ad_start_play() */
-
-/*
- * cs4231_ad_stop_play()
- *
- * Description:
- *	This routine stops the play DMA engine.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-cs4231_ad_stop_play(audiohdl_t ahandle, int stream)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-
-	ATRACE("cs_ad_stop_play() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!stop_play() set_busy() failed");
-		return;
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* we need to protect the state structure */
-	mutex_enter(&state->cs_lock);
-
-	ATRACE_8("cs_ad_stop_play() Codec INTC_REG",
-	    ddi_get8(handle, &CS4231_IDR));
-
-	/* stop the play DMA engine */
-	ATRACE("cs_ad_stop_play() calling DMA_STOP_PLAY()", state);
-	CS4231_DMA_STOP_PLAY(state);
-	ATRACE("cs_ad_stop_play() DMA_STOP_PLAY() returned", state);
-	/* DMA_STOP() returns with the PEN cleared */
-
-	if (state->cs_flags & PLAY_ACTIVE) {
-		state->cs_flags &= ~PLAY_ACTIVE;
-		(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
-	}
-
-	mutex_exit(&state->cs_lock);
-
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE("cs4231_ad_stop_play() returning", state);
-
-}	/* cs4231_ad_stop_play() */
-
-/*
- * cs4231_ad_start_record()
- *
- * Description:
- *	This routine starts the record DMA engine. It checks to make sure the
- *	DMA engine is off before it does anything, otherwise it may mess
- *	things up.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *
- * Returns:
- *	AUDIO_SUCCESS		Recording successfully started
- *	AUDIO_FAILURE		Recording not successfully started
- */
-/*ARGSUSED*/
-static int
-cs4231_ad_start_record(audiohdl_t ahandle, int stream)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE("cs_ad_start_record() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!start_record() set_busy() failed");
-		return (AUDIO_FAILURE);
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* we need to protect the state structure */
-	mutex_enter(&state->cs_lock);
-	ASSERT(state->cs_powered == CS4231_PWR_ON);
-
-	/* see if we are already recording */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, INTC_REG, __LINE__,
-	    thisfile);
-	if (INTC_CEN & ddi_get8(handle, &CS4231_IDR)) {
-		mutex_exit(&state->cs_lock);
-		ATRACE("cs_ad_start_record() already recording", NULL);
-		rc = AUDIO_SUCCESS;
-		goto done;
-	}
-
-	/*
-	 * Enable record DMA on the Codec, do NOT reprogram the Codec as
-	 * done for play. This will cause SunVTS to fail.
-	 */
-	ATRACE("cs_ad_start_record() calling DMA_START_RECORD()", state);
-	rc = CS4231_DMA_START_RECORD(state);
-	ATRACE("cs_ad_start_record() DMA_START_RECORD() returned", rc);
-
-	if (rc == AUDIO_SUCCESS) {
-		ATRACE("cs_ad_start_record() programming Codec to rec.", state);
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR, INTC_REG,
-		    __LINE__, thisfile);
-		OR_SET_BYTE(handle, &CS4231_IDR, INTC_CEN, INTC_VALID_MASK);
-
-		ATRACE_8("cs_ad_start_record() Codec INTC_REG",
-		    ddi_get8(handle, &CS4231_IDR));
-
-		(void) pm_busy_component(state->cs_dip, CS4231_COMPONENT);
-
-#ifdef DEBUG
-	} else {
-		ATRACE("cs_ad_start_record() Codec not started", rc);
-#endif
-	}
-
-	mutex_exit(&state->cs_lock);
-
-done:
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE("cs4231_ad_start_record() returning", rc);
-
-	return (rc);
-
-}	/* cs4231_ad_start_record() */
-
-/*
- * cs4231_ad_stop_record()
- *
- * Description:
- *	This routine stops the record DMA engine.
- *
- *	The stream argument is ignored because this isn't a multi-stream Codec.
- *
- *	NOTE: This routine must be called with the state unlocked.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *
- * Returns:
- *	void
- */
-/*ARGSUSED*/
-static void
-cs4231_ad_stop_record(audiohdl_t ahandle, int stream)
-{
-	CS_state_t		*state;
-	ddi_acc_handle_t	handle;
-
-	ATRACE("cs_ad_stop_record() cs_statep", cs_statep);
-
-	/* get the state structure */
-	state = audio_sup_get_private(ahandle);
-	ASSERT(state);
-
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* power up and mark as busy */
-	if (cs4231_set_busy(state) == AUDIO_FAILURE) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!stop_record() set_busy() failed");
-		return;
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* we need to protect the state structure */
-	mutex_enter(&state->cs_lock);
-
-	/* stop the record DMA engine and clear the active flag */
-	ATRACE("cs_ad_stop_record() calling DMA_STOP_RECORD()", state);
-	CS4231_DMA_STOP_RECORD(state);
-	ATRACE("cs_ad_stop_record() DMA_STOP_RECORD() returned", state);
-
-	ATRACE("cs_ad_stop_record() programming Codec to rec.", state);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_CEN, INTC_VALID_MASK);
-
-	ATRACE_8("cs_ad_stop_record() Codec INTC_REG",
-	    ddi_get8(handle, &CS4231_IDR));
-
-	(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
-
-	mutex_exit(&state->cs_lock);
-
-	/* need an idle for the busy above */
-	cs4231_set_idle(state);
-
-	ATRACE("cs4231_ad_stop_record() returning", state);
-
-}       /* cs4231_ad_stop_record() */
-
-
-/* ******* Local Routines *************************************************** */
-
-/*
- * cs4231_chip_init()
- *
- * Description:
- *	Power up the audio core, initialize the audio Codec, prepare the chip
- *	for use.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS			Chip initialized and ready to use
- *	AUDIO_FAILURE			Chip not initialized and not ready
- */
-static int
-cs4231_chip_init(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle;
-	audiohdl_t		ahandle = state->cs_ahandle;
-
-	/* make sure we are powered up */
-	mutex_enter(&state->cs_lock);
-	CS4231_DMA_POWER(state, CS4231_PWR_ON);
-	mutex_exit(&state->cs_lock);
-
-	ATRACE("cs_attach() calling DMA_RESET()", state);
-	CS4231_DMA_RESET(state);
-
-	/* no autocalibrate */
-	state->cs_autocal = B_FALSE;
-
-	/* initialize the Codec */
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* activate registers 16 -> 31 */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, MID_REG,
-	    __LINE__, thisfile);
-	ddi_put8(handle, &CS4231_IDR, MID_MODE2);
-
-	/* now figure out what version we have */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, VID_REG,
-	    __LINE__, thisfile);
-	if (ddi_get8(handle, &CS4231_IDR) & VID_A) {
-		ATRACE("cs_attach() revA", state);
-		state->cs_revA = B_TRUE;
-	} else {
-		ATRACE("cs_attach() !revA", state);
-		state->cs_revA = B_FALSE;
-	}
-
-	/* get rid of annoying popping by muting the output channels */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LDACO_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (LDACO_LDM | LDACO_MID_GAIN), LDAC0_VALID_MASK, __LINE__,
-	    thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RDACO_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (RDACO_RDM | RDACO_MID_GAIN), RDAC0_VALID_MASK, __LINE__,
-	    thisfile);
-
-	/* initialize aux input channels to known gain values & muted */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LAUX1_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (LAUX1_LX1M | LAUX1_UNITY_GAIN), LAUX1_VALID_MASK, __LINE__,
-	    thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RAUX1_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (RAUX1_RX1M | RAUX1_UNITY_GAIN), RAUX1_VALID_MASK, __LINE__,
-	    thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LAUX2_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (LAUX2_LX2M | LAUX2_UNITY_GAIN), LAUX2_VALID_MASK, __LINE__,
-	    thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RAUX2_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (RAUX2_RX2M | RAUX2_UNITY_GAIN), RAUX2_VALID_MASK, __LINE__,
-	    thisfile);
-
-	/* initialize aux input channels to known gain values & muted */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LLIC_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, (LLIC_LLM | LLIC_UNITY_GAIN),
-	    LLIC_VALID_MASK, __LINE__, thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RLIC_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, (RLIC_RLM | RLIC_UNITY_GAIN),
-	    RLIC_VALID_MASK, __LINE__, thisfile);
-
-	/* program the sample rate, play and capture must be the same */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-	    (FSDF_REG | IAR_MCE), __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR,
-	    (FS_8000 | PDF_ULAW8 | PDF_MONO), FSDF_VALID_MASK,
-	    __LINE__, thisfile);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-	    (CDF_REG | IAR_MCE), __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, (CDF_ULAW8 | CDF_MONO),
-	    CDF_VALID_MASK, __LINE__, thisfile);
-
-	/*
-	 * Set up the Codec for playback and capture disabled, dual DMA, and
-	 * playback and capture DMA. Also, set autocal if we are supposed to.
-	 */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-	    (INTC_REG | IAR_MCE), __LINE__, thisfile);
-	if (state->cs_autocal == B_TRUE) {
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    (INTC_ACAL|INTC_DDC|INTC_PDMA|INTC_CDMA), INTC_VALID_MASK,
-		    __LINE__, thisfile);
-	} else {
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    (INTC_DDC | INTC_PDMA | INTC_CDMA), INTC_VALID_MASK,
-		    __LINE__, thisfile);
-	}
-
-	/* turn off the MCE bit */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LADCI_REG,
-	    __LINE__, thisfile);
-
-	/* wait for the Codec before we continue XXX - do we need this? */
-	if (cs4231_poll_ready(state) == AUDIO_FAILURE) {
-		ATRACE("cs_attach() poll_ready() #1 failed", state);
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Turn on the output level bit to be 2.8 Vpp. Also, don't go to 0 on
-	 * underflow.
-	 */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, AFE1_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, AFE1_OLB, AFE1_VALID_MASK,
-	    __LINE__, thisfile);
-
-	/* turn on the high pass filter if Rev A */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, AFE2_REG,
-	    __LINE__, thisfile);
-	if (state->cs_revA) {
-		cs4231_put8(ahandle, handle, &CS4231_IDR, AFE2_HPF,
-		    AFE2_VALID_MASK, __LINE__, thisfile);
-	} else {
-		cs4231_put8(ahandle, handle, &CS4231_IDR, 0,
-		    AFE2_VALID_MASK, __LINE__, thisfile);
-	}
-
-	/* clear the play and capture interrupt flags */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, AFS_REG,
-	    __LINE__, thisfile);
-	ddi_put8(handle, &CS4231_STATUS, (AFS_RESET_STATUS));
-
-	/* the play and record gains will be set by the audio mixer */
-
-	/* unmute the output */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LDACO_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~LDACO_LDM, LDAC0_VALID_MASK);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RDACO_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~RDACO_RDM, RDAC0_VALID_MASK);
-
-	/* unmute the mono speaker and mute mono in */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, MIOC_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, MIOC_MIM, MIOC_VALID_MASK,
-	    __LINE__, thisfile);
-
-	/* clear the mode change bit */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, RDACO_REG,
-	    __LINE__, thisfile);
-
-	/* wait for the Codec before we continue XXX - do we need this? */
-	if (cs4231_poll_ready(state) == AUDIO_FAILURE) {
-		ATRACE("attach() poll_ready() #2 failed", state);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("cs_attach() chip initialized", state);
-	return (AUDIO_SUCCESS);
-
-}	/* cs4231_chip_init() */
-
-/*
- * audiocs_init_state()
- *
- * Description:
- *	This routine initializes the audio driver's state structure and
- *	maps in the registers. This also includes reading the properties.
- *
- *	CAUTION: This routine maps the registers and initializes a mutex.
- *		 Failure cleanup is handled by cs4231_attach(). It is not
- *		 handled locally by this routine.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *	dev_info_t	*dip		Pointer to the device's dev_info struct
- *
- * Returns:
- *	AUDIO_SUCCESS			State structure initialized
- *	AUDIO_FAILURE			State structure not initialized
- */
-static int
-cs4231_init_state(CS_state_t *state, dev_info_t *dip)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	char			*prop_str;
-	char			*pm_comp[] = {
-					"NAME=audiocs audio device",
-					"0=off",
-					"1=on" };
-	int			cs4231_pints;
-	int			cs4231_rints;
-	size_t			cbuf_size;
-	size_t			pbuf_size;
-	int			instance;
-
-	ATRACE("in cs_init_state()", dip);
-	instance = ddi_get_instance(dip);
-	ATRACE_32("cs_attach() instance", instance);
-
-	/*
-	 * get the play and record interrupts per second,
-	 * look for either cs4231_XXXX or XXXX-interrupts.
-	 */
-	if ((cs4231_pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "cs4231_pints", -1)) == -1) {
-		cs4231_pints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-		    DDI_PROP_DONTPASS, "play-interrupts", CS4231_INTS);
-	}
-	ATRACE_32("cs_init_state() play interrupts per sec", cs4231_pints);
-
-	if ((cs4231_rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "cs4231_rints", -1)) == -1) {
-		cs4231_rints = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-		    DDI_PROP_DONTPASS, "record-interrupts", CS4231_INTS);
-	}
-	ATRACE_32("cs_init_state() record interrupts per sec", cs4231_rints);
-
-	if (cs4231_pints < CS4231_MIN_INTS) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "attach() play interrupt rate set too low: %d, resetting"
-		    " to %d", cs4231_pints, CS4231_INTS);
-		cs4231_pints = CS4231_INTS;
-	} else if (cs4231_pints > CS4231_MAX_INTS) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "attach() play interrupt rate set too high: %d, resetting"
-		    " to %d", cs4231_pints, CS4231_INTS);
-		cs4231_pints = CS4231_INTS;
-	}
-
-	if (cs4231_rints < CS4231_MIN_INTS) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "attach() record interrupt rate set too low: %d, resetting"
-		    " to %d", cs4231_rints, CS4231_INTS);
-		cs4231_rints = CS4231_INTS;
-	} else if (cs4231_rints > CS4231_MAX_INTS) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "attach() record interrupt rate set too high: %d, resetting"
-		    " to %d", cs4231_rints, CS4231_INTS);
-		cs4231_rints = CS4231_INTS;
-	}
-
-	/*
-	 * Figure out the largest transfer size for the DMA engine. Then
-	 * map in the CS4231 and the DMA registers and reset the DMA engine.
-	 */
-	pbuf_size = CS4231_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / cs4231_pints;
-	cbuf_size = CS4231_SAMPR48000 * AUDIO_CHANNELS_STEREO *
-	    (AUDIO_PRECISION_16 >> AUDIO_PRECISION_SHIFT) / cs4231_rints;
-
-	/* get the mode from the .conf file */
-
-	if ((state->cs_ad_info.ad_mode = ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-	    DDI_PROP_DONTPASS, "cs4231_mode", -1)) == -1) {
-		if (ddi_prop_get_int(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-		    "mixer-mode", AM_MIXER_MODE)) {
-			state->cs_ad_info.ad_mode = AM_MIXER_MODE;
-		} else {
-			state->cs_ad_info.ad_mode = AM_COMPAT_MODE;
-		}
-	}
-	ATRACE_32("cs_init_state() setting mode", state->cs_ad_info.ad_mode);
-
-	/* set up the pm-components */
-	if (ddi_prop_update_string_array(DDI_DEV_T_NONE, dip,
-	    "pm-components", pm_comp, 3) != DDI_PROP_SUCCESS) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!init_state() couldn't create component");
-		return (AUDIO_FAILURE);
-	}
-
-	/* save the device info pointer */
-	state->cs_dip = dip;
-
-	/* get the iblock cookie needed for interrupt context */
-	if (ddi_get_iblock_cookie(dip, (uint_t)0, &state->cs_iblock) !=
-	    DDI_SUCCESS) {
-		audio_sup_log(ahandle, CE_WARN,
-		    "!init_state() cannot get iblock cookie");
-		return (AUDIO_FAILURE);
-	}
-
-	/* now fill it in, initialize the state mutexs first */
-	mutex_init(&state->cs_lock, NULL, MUTEX_DRIVER, state->cs_iblock);
-	cv_init(&state->cs_cv, NULL, CV_DRIVER, NULL);
-
-	/* fill in the device default state */
-	state->cs_defaults.play.sample_rate = CS4231_DEFAULT_SR;
-	state->cs_defaults.play.channels = CS4231_DEFAULT_CH;
-	state->cs_defaults.play.precision = CS4231_DEFAULT_PREC;
-	state->cs_defaults.play.encoding = CS4231_DEFAULT_ENC;
-	state->cs_defaults.play.gain = CS4231_DEFAULT_PGAIN;
-	state->cs_defaults.play.port = AUDIO_SPEAKER;
-	state->cs_defaults.play.buffer_size = CS4231_BSIZE;
-	state->cs_defaults.play.balance = CS4231_DEFAULT_BAL;
-	state->cs_defaults.record.sample_rate = CS4231_DEFAULT_SR;
-	state->cs_defaults.record.channels = CS4231_DEFAULT_CH;
-	state->cs_defaults.record.precision = CS4231_DEFAULT_PREC;
-	state->cs_defaults.record.encoding = CS4231_DEFAULT_ENC;
-	state->cs_defaults.record.gain = CS4231_DEFAULT_PGAIN;
-	state->cs_defaults.record.port = AUDIO_MICROPHONE;
-	state->cs_defaults.record.buffer_size = CS4231_BSIZE;
-	state->cs_defaults.record.balance = CS4231_DEFAULT_BAL;
-	state->cs_defaults.monitor_gain = CS4231_DEFAULT_MONITOR_GAIN;
-	state->cs_defaults.output_muted = B_FALSE;
-	state->cs_defaults.hw_features = AUDIO_HWFEATURE_DUPLEX|
-	    AUDIO_HWFEATURE_IN2OUT|AUDIO_HWFEATURE_PLAY|AUDIO_HWFEATURE_RECORD;
-	state->cs_defaults.sw_features = AUDIO_SWFEATURE_MIXER;
-
-	/* fill in the ad_info structure */
-	state->cs_ad_info.ad_int_vers = AM_VERSION;
-
-	state->cs_ad_info.ad_add_mode = 0;
-	state->cs_ad_info.ad_codec_type = AM_TRAD_CODEC;
-	state->cs_ad_info.ad_defaults = &state->cs_defaults;
-	state->cs_ad_info.ad_play_comb = cs_combinations;
-	state->cs_ad_info.ad_rec_comb = cs_combinations;
-	state->cs_ad_info.ad_entry = &cs_entry;
-	state->cs_ad_info.ad_dev_info = &state->cs_dev_info;
-	state->cs_ad_info.ad_diag_flags = 0;
-	state->cs_ad_info.ad_diff_flags = AM_DIFF_CH|AM_DIFF_PREC|AM_DIFF_ENC;
-	state->cs_ad_info.ad_assist_flags = AM_ASSIST_MIC;
-	state->cs_ad_info.ad_misc_flags = AM_MISC_RP_EXCL|AM_MISC_MONO_DUP;
-	state->cs_ad_info.ad_translate_flags =
-	    AM_MISC_8_P_TRANSLATE|AM_MISC_8_R_TRANSLATE;
-	state->cs_ad_info.ad_num_mics = 1;
-
-	/* play capabilities */
-	state->cs_ad_info.ad_play.ad_mixer_srs = cs_mixer_sample_rates;
-	state->cs_ad_info.ad_play.ad_compat_srs = cs_compat_sample_rates;
-	state->cs_ad_info.ad_play.ad_conv = &am_src2;
-	state->cs_ad_info.ad_play.ad_sr_info = NULL;
-	state->cs_ad_info.ad_play.ad_chs = cs_channels;
-	state->cs_ad_info.ad_play.ad_int_rate = cs4231_pints;
-	state->cs_ad_info.ad_play.ad_max_chs = CS4231_MAX_CHANNELS;
-	state->cs_ad_info.ad_play.ad_bsize = CS4231_BSIZE;
-
-	/* record capabilities */
-	state->cs_ad_info.ad_record.ad_mixer_srs = cs_mixer_sample_rates;
-	state->cs_ad_info.ad_record.ad_compat_srs = cs_compat_sample_rates;
-	state->cs_ad_info.ad_record.ad_conv = &am_src2;
-	state->cs_ad_info.ad_record.ad_sr_info = NULL;
-	state->cs_ad_info.ad_record.ad_chs = cs_channels;
-	state->cs_ad_info.ad_record.ad_int_rate = cs4231_rints;
-	state->cs_ad_info.ad_record.ad_max_chs = CS4231_MAX_CHANNELS;
-	state->cs_ad_info.ad_record.ad_bsize = CS4231_BSIZE;
-
-	/* figure out which DMA engine hardware we have */
-	if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "dma-model", &prop_str) == DDI_PROP_SUCCESS) {
-		if (strcmp(prop_str, "eb2dma") == 0) {
-			ATRACE("cs_attach() eb2dma", state);
-			state->cs_dma_engine = EB2_DMA;
-			state->cs_dma_ops = &cs4231_eb2dma_ops;
-		} else {
-			ATRACE("cs_attach() apcdma", state);
-			state->cs_dma_engine = APC_DMA;
-			state->cs_dma_ops = &cs4231_apcdma_ops;
-		}
-		ddi_prop_free(prop_str);
-	} else {
-		ATRACE("cs_attach() no prop apcdma", state);
-		state->cs_dma_engine = APC_DMA;
-		state->cs_dma_ops = &cs4231_apcdma_ops;
-	}
-
-	/* cs_regs, cs_eb2_regs and cs_handles filled in later */
-
-	(void) strcpy(&state->cs_dev_info.name[0], CS_DEV_NAME);
-	/* always set to onboard1, not really correct, but very high runner */
-	(void) strcpy(&state->cs_dev_info.config[0], CS_DEV_CONFIG_ONBRD1);
-	/* version filled in below */
-
-	/* most of what's left is filled in when the registers are mapped */
-
-	ATRACE("cs_init_state() calling get_ports()", state);
-	cs4231_get_ports(state, dip);
-
-	/* Map in the registers */
-	if (CS4231_DMA_MAP_REGS(dip, state, pbuf_size, cbuf_size) ==
-	    AUDIO_FAILURE) {
-		goto error_regs;
-	}
-
-	state->cs_instance = ddi_get_instance(dip);
-	state->cs_suspended = CS4231_NOT_SUSPENDED;
-	state->cs_powered = CS4231_PWR_OFF;
-	state->cs_busy_cnt = 0;
-
-	return (AUDIO_SUCCESS);
-
-error_regs:
-	ATRACE("cs_init_state() error_regs", state);
-	mutex_destroy(&state->cs_lock);
-
-	ATRACE("cs_init_state() returning failure", NULL);
-
-	return (AUDIO_FAILURE);
-
-}	/* cs4231_init_state */
-
-/*
- * cs4231_get_ports()
- *
- * Description:
- *	Get which audiocs h/w version we have and use this to
- *	determine the input and output ports as well whether or not
- *	the hardware has internal loopbacks or not. We also have three
- *	different ways for the properties to be specified, which we
- *	also need to worry about.
- *
- * Vers	Platform(s)	DMA eng.	audio-module**	loopback
- * a    SS-4+/SS-5+	apcdma		no		no
- * b	Ultra-1&2	apcdma		no		yes
- * c	positron	apcdma		no		yes
- * d	PPC - retired
- * e	x86 - retired
- * f	tazmo		eb2dma		Perigee		no
- * g	tazmo		eb2dma		Quark		yes
- * h	darwin+		eb2dma		no		N/A
- *
- * Vers	model~		aux1*		aux2*
- * a	N/A		N/A		N/A
- * b	N/A		N/A		N/A
- * c	N/A		N/A		N/A
- * d	retired
- * e	retired
- * f	SUNW,CS4231f	N/A		N/A
- * g	SUNW,CS4231g	N/A		N/A
- * h	SUNW,CS4231h	cdrom		none
- *
- * *   = Replaces internal-loopback for latest property type, can be
- *	 set to "cdrom", "loopback", or "none".
- *
- * **  = For plugin audio modules only. Starting with darwin, this
- *	 property is replaces by the model property.
- *
- * ~   = Replaces audio-module.
- *
- * +   = Has the capability of having a cable run from the internal
- *	 CD-ROM to the audio device.
- *
- * N/A = Not applicable, the property wasn't created for early
- *	 platforms, or the property has been retired.
- *
- * NOTE: Older tazmo and quark machines don't have the model property.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *	dev_info_t	*dip		Pointer to the device's dev_info struct
- *
- * Returns:
- *	void
- */
-static void
-cs4231_get_ports(CS_state_t *state, dev_info_t *dip)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	char			*prop_str;
-
-	ATRACE("Beginning cs4231_get_ports()", NULL);
-
-	/* First we set the common ports, etc. */
-	state->cs_defaults.play.avail_ports =
-	    AUDIO_SPEAKER|AUDIO_HEADPHONE|AUDIO_LINE_OUT;
-	state->cs_defaults.play.mod_ports =
-	    AUDIO_SPEAKER|AUDIO_HEADPHONE|AUDIO_LINE_OUT;
-	state->cs_defaults.record.avail_ports =
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	state->cs_defaults.record.mod_ports |=
-	    AUDIO_MICROPHONE|AUDIO_LINE_IN|AUDIO_CODEC_LOOPB_IN;
-	state->cs_cd_input_line = NO_INTERNAL_CD;
-
-	/* now we try the new "model" property */
-	if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip, DDI_PROP_DONTPASS,
-	    "model", &prop_str) == DDI_PROP_SUCCESS) {
-		if (strcmp(prop_str, "SUNW,CS4231h") == 0) {
-			/* darwin */
-			ATRACE("cs_attach() NEW - darwin", state);
-			(void) strcpy(&state->cs_dev_info.version[0],
-			    CS_DEV_VERSION_H);
-			state->cs_defaults.record.avail_ports |=
-			    AUDIO_INTERNAL_CD_IN;
-			state->cs_defaults.play.mod_ports =
-			    AUDIO_SPEAKER;
-			state->cs_defaults.record.avail_ports |= AUDIO_CD;
-			state->cs_defaults.record.mod_ports |= AUDIO_CD;
-			state->cs_cd_input_line = INTERNAL_CD_ON_AUX1;
-		} else if (strcmp(prop_str, "SUNW,CS4231g") == 0) {
-			/* quark audio module */
-			ATRACE("cs_attach() NEW - quark", state);
-			(void) strcpy(&state->cs_dev_info.version[0],
-			    CS_DEV_VERSION_G);
-			state->cs_defaults.record.avail_ports |= AUDIO_SUNVTS;
-			state->cs_defaults.record.mod_ports |= AUDIO_SUNVTS;
-		} else if (strcmp(prop_str, "SUNW,CS4231f") == 0) {
-			/* tazmo */
-			ATRACE("cs_attach() NEW - tazmo", state);
-			(void) strcpy(&state->cs_dev_info.version[0],
-			    CS_DEV_VERSION_F);
-		} else {
-			ATRACE("cs_attach() NEW - unknown", state);
-			(void) strcpy(&state->cs_dev_info.version[0], "?");
-			audio_sup_log(ahandle, CE_NOTE,
-			    "!attach() unknown audio model: %s, some parts of "
-			    "audio may not work correctly",
-			    (prop_str ? prop_str : "unknown"));
-		}
-		ddi_prop_free(prop_str);	/* done with the property */
-	} else {	/* now try the older "audio-module" property */
-		if (ddi_prop_lookup_string(DDI_DEV_T_ANY, dip,
-		    DDI_PROP_DONTPASS, "audio-module", &prop_str) ==
-		    DDI_PROP_SUCCESS) {
-			switch (*prop_str) {
-			case 'Q':	/* quark audio module */
-				ATRACE("cs_attach() OLD - quark", state);
-				(void) strcpy(&state->cs_dev_info.version[0],
-				    CS_DEV_VERSION_G);
-				state->cs_defaults.record.avail_ports |=
-				    AUDIO_SUNVTS;
-				state->cs_defaults.record.mod_ports |=
-				    AUDIO_SUNVTS;
-				break;
-			case 'P':	/* tazmo */
-				ATRACE("cs_attach() OLD - tazmo", state);
-				(void) strcpy(&state->cs_dev_info.version[0],
-				    CS_DEV_VERSION_F);
-				break;
-			default:
-				ATRACE("cs_attach() OLD - unknown", state);
-				(void) strcpy(&state->cs_dev_info.version[0],
-				    "?");
-				audio_sup_log(ahandle, CE_NOTE,
-				    "!attach() unknown audio module: %s, some "
-				    "parts of audio may not work correctly",
-				    (prop_str ? prop_str : "unknown"));
-				break;
-			}
-			ddi_prop_free(prop_str);	/* done with the prop */
-		} else {	/* now try heuristics, ;-( */
-			if (ddi_prop_get_int(DDI_DEV_T_ANY, dip,
-			    DDI_PROP_DONTPASS, "internal-loopback", B_FALSE)) {
-				if (state->cs_dma_engine == EB2_DMA) {
-					ATRACE("cs_attach() OLD - C", state);
-					(void) strcpy(
-					    &state->cs_dev_info.version[0],
-					    CS_DEV_VERSION_C);
-				} else {
-					ATRACE("cs_attach() OLD - B", state);
-					(void) strcpy(
-					    &state->cs_dev_info.version[0],
-					    CS_DEV_VERSION_B);
-				}
-				state->cs_defaults.record.avail_ports |=
-				    AUDIO_SUNVTS;
-				state->cs_defaults.record.mod_ports |=
-				    AUDIO_SUNVTS;
-			} else {
-				ATRACE("cs_attach() ANCIENT - A", state);
-				(void) strcpy(&state->cs_dev_info.version[0],
-				    CS_DEV_VERSION_A);
-				state->cs_defaults.record.avail_ports |=
-				    AUDIO_INTERNAL_CD_IN;
-				state->cs_defaults.record.mod_ports |=
-				    AUDIO_INTERNAL_CD_IN;
-				state->cs_cd_input_line = INTERNAL_CD_ON_AUX1;
-			}
-		}
-	}
-
-	ATRACE("Finished in cs4231_get_ports()", NULL);
-
-}	/* cs4231_get_ports() */
-
-/*
- * cs4231_power_up()
- *
- * Description:
- *	Power up the Codec and restore the codec's registers.
- *
- *	NOTE: Like the audiots driver, we don't worry about locking since
- *		the only routines that may call us are attach() and power()
- *		Both of which should be the only threads in the driver.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-cs4231_power_up(CS_state_t *state)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	ddi_acc_handle_t	handle;
-	int			i;
-
-	ASSERT(mutex_owned(&state->cs_lock));
-	ATRACE("in cs4231_power_up()", NULL);
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/* turn on the Codec */
-	CS4231_DMA_POWER(state, CS4231_PWR_ON);
-
-	/* reset the DMA engine(s) */
-	CS4231_DMA_RESET(state);
-
-	/*
-	 * Reload the Codec's registers, the DMA engines will be
-	 * taken care of when play and record start up again. But
-	 * first enable registers 16 -> 31.
-	 */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, MID_REG,
-	    __LINE__, thisfile);
-	cs4231_put8(ahandle, handle, &CS4231_IDR, state->cs_save[MID_REG],
-	    MID_VALID_MASK, __LINE__, thisfile);
-
-	for (i = 0; i < CS4231_REGS; i++) {
-		/* restore Codec registers */
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    (i | IAR_MCE), __LINE__, thisfile);
-		ddi_put8(handle, &CS4231_IDR, state->cs_save[i]);
-		drv_usecwait(500);	/* chip bug */
-	}
-	/* clear MCE bit */
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, 0, __LINE__,
-	    thisfile);
-
-	ASSERT(state->cs_powered == CS4231_PWR_ON);
-
-	ATRACE("cs_power() power up successful", state);
-	ASSERT(mutex_owned(&state->cs_lock));
-
-}	/* cs4231_power_up() */
-
-/*
- * cs4231_power_down()
- *
- * Description:
- *	Power down the Codec and save the codec's registers.
- *
- *	NOTE: See the note in cs4231_power_up() about locking.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-cs4231_power_down(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle;
-	int			i;
-
-	ASSERT(mutex_owned(&state->cs_lock));
-	ATRACE("in cs4231_power_down()", NULL);
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	/*
-	 * We are powering down, so we don't need to do a thing with
-	 * the DMA engines. However, we do need to save the Codec
-	 * registers.
-	 */
-
-	for (i = 0; i < CS4231_REGS; i++) {
-		/* save Codec regs */
-		cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, i,
-		    __LINE__, thisfile);
-		state->cs_save[i] = ddi_get8(handle, &CS4231_IDR);
-	}
-
-	/* turn off the Codec */
-	CS4231_DMA_POWER(state, CS4231_PWR_OFF);
-
-	ASSERT(state->cs_powered == CS4231_PWR_OFF);
-
-	ATRACE("cs_power() power down successful", state);
-	ASSERT(mutex_owned(&state->cs_lock));
-
-}	/* cs4231_power_down() */
-
-/*
- * cs4231_set_gain()
- *
- * Description:
- *	Set the play or record gain.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *	int		stream		Stream number for multi-stream Codecs,
- *					which this isn't, so just ignore
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		gain		The gain to set
- *	int		channels	Number of channels, 1 or 2
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-/*ARGSUSED*/
-static int
-cs4231_set_gain(CS_state_t *state, int stream, int dir, int gain,
-    int channel)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	ddi_acc_handle_t	handle;
-	uint8_t			tmp_value;
-	int			rc = AUDIO_FAILURE;
-
-	ATRACE("in cs_set_gain()", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	if (dir == AUDIO_PLAY) {	/* play gain */
-		mutex_enter(&state->cs_lock);
-		if (channel == 0) {	/* left channel */
-			cs4231_reg_select(ahandle, handle,
-			    &CS4231_IAR, LDACO_REG, __LINE__, thisfile);
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			cs4231_reg_select(ahandle, handle,
-			    &CS4231_IAR, RDACO_REG, __LINE__, thisfile);
-		}
-		/* NOTE: LDAC0_VALID_MASK == RDAC0_VALID_MASK, so either ok */
-
-		/* we use cs4231_atten[] to linearize attenuation */
-		if (state->cs_output_muted || gain == 0) {
-			/* mute the output */
-			cs4231_put8(ahandle, handle, &CS4231_IDR,
-			    (cs4231_atten[gain]|LDACO_LDM), LDAC0_VALID_MASK,
-			    __LINE__, thisfile);
-			/* NOTE: LDACO_LDM == RDACO_LDM, so either ok */
-		} else {
-			cs4231_put8(ahandle, handle, &CS4231_IDR,
-			    cs4231_atten[gain], LDAC0_VALID_MASK,
-			    __LINE__, thisfile);
-			ATRACE("cs_set_gain() play gain set",
-			    cs4231_atten[gain]);
-
-		}
-		rc = AUDIO_SUCCESS;
-		mutex_exit(&state->cs_lock);
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-
-		mutex_enter(&state->cs_lock);
-		if (channel == 0) {	/* left channel */
-			cs4231_reg_select(ahandle, handle,
-			    &CS4231_IAR, LADCI_REG, __LINE__, thisfile);
-			tmp_value = ddi_get8(handle, &CS4231_IDR) &
-			    ~LADCI_GAIN_MASK;
-		} else {		/* right channel */
-			ASSERT(channel == 1);
-			cs4231_reg_select(ahandle, handle,
-			    &CS4231_IAR, RADCI_REG, __LINE__, thisfile);
-			tmp_value = ddi_get8(handle, &CS4231_IDR) &
-			    ~RADCI_GAIN_MASK;
-		}
-		/* NOTE: LADCI_VALID_MASK == RADCI_VALID_MASK, so either ok */
-
-		/* we shift right by 4 to go from 8-bit to 4-bit gain */
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    (tmp_value|(gain >> 4)), LADCI_VALID_MASK,
-		    __LINE__, thisfile);
-		ATRACE("cs_set_gain() record gain set",
-		    (tmp_value|(gain >> 4)));
-		rc = AUDIO_SUCCESS;
-		mutex_exit(&state->cs_lock);
-	}
-
-	ATRACE("cs_set_gain() returning", rc);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (rc);
-
-}	/* cs4231_set_gain() */
-
-/*
- * cs4231_set_port()
- *
- * Description:
- *	Set the play/record port.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *	int		dir		AUDIO_PLAY or AUDIO_RECORD, if
- *					direction is important
- *	int		port		The port to set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The port could not been set
- */
-static int
-cs4231_set_port(CS_state_t *state, int dir, int port)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	ddi_acc_handle_t	handle;
-	uint8_t			tmp_value;
-	int			rc = AUDIO_SUCCESS;
-
-	ATRACE("in cs_set_port()", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	handle = state->cs_handles.cs_codec_hndl;
-
-	if (dir == AUDIO_PLAY) {	/* output port(s) */
-		/* figure out which output port(s) to turn on */
-		tmp_value = 0;
-
-		mutex_enter(&state->cs_lock);
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    MIOC_REG, __LINE__, thisfile);
-		if (port & AUDIO_SPEAKER) {
-			AND_SET_BYTE(handle, &CS4231_IDR, ~MIOC_MONO_SPKR_MUTE,
-			    MIOC_VALID_MASK);
-			tmp_value |= AUDIO_SPEAKER;
-		} else {
-			OR_SET_BYTE(handle, &CS4231_IDR, MIOC_MONO_SPKR_MUTE,
-			    MIOC_VALID_MASK);
-		}
-
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    PC_REG, __LINE__, thisfile);
-		if (port & AUDIO_HEADPHONE) {
-			AND_SET_BYTE(handle, &CS4231_IDR, ~PC_HEADPHONE_MUTE,
-			    PC_VALID_MASK);
-			tmp_value |= AUDIO_HEADPHONE;
-		} else {
-			OR_SET_BYTE(handle, &CS4231_IDR, PC_HEADPHONE_MUTE,
-			    PC_VALID_MASK);
-		}
-
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    PC_REG, __LINE__, thisfile);
-		if (port & AUDIO_LINE_OUT) {
-			AND_SET_BYTE(handle, &CS4231_IDR, ~PC_LINE_OUT_MUTE,
-			    PC_VALID_MASK);
-			tmp_value |= AUDIO_LINE_OUT;
-		} else {
-			OR_SET_BYTE(handle, &CS4231_IDR, PC_LINE_OUT_MUTE,
-			    PC_VALID_MASK);
-		}
-		mutex_exit(&state->cs_lock);
-
-		ATRACE_32("cs_ad_set_config() set out port", tmp_value);
-
-		if (tmp_value != (port & 0x0ff)) {
-			ATRACE_32("cs_ad_set_config() bad out port", port);
-			rc = AUDIO_FAILURE;
-			goto done;
-		}
-	} else {
-		ASSERT(dir == AUDIO_RECORD);
-
-		/*
-		 * Figure out which input port to set. Fortunately
-		 * the left and right port bit patterns are the same.
-		 */
-		switch (port) {
-		case AUDIO_NONE:
-			tmp_value = 0;
-			break;
-		case AUDIO_MICROPHONE:
-			tmp_value = LADCI_LMIC;
-			break;
-		case AUDIO_LINE_IN:
-			tmp_value = LADCI_LLINE;
-			break;
-		case AUDIO_CD:
-			tmp_value = LADCI_LAUX1;
-			break;
-		case AUDIO_CODEC_LOOPB_IN:
-			tmp_value = LADCI_LLOOP;
-			break;
-		case AUDIO_SUNVTS:
-			tmp_value = LADCI_LAUX1;
-			break;
-		default:
-			/* unknown or inclusive input ports */
-			ATRACE_32("cs_ad_set_config() bad in port", port);
-			rc = AUDIO_FAILURE;
-			goto done;
-		}
-
-		mutex_enter(&state->cs_lock);
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    LADCI_REG, __LINE__, thisfile);
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    (ddi_get8(handle, &CS4231_IDR) & ~LADCI_IN_MASK)|tmp_value,
-		    LADCI_VALID_MASK, __LINE__, thisfile);
-		cs4231_reg_select(ahandle, handle, &CS4231_IAR,
-		    RADCI_REG, __LINE__, thisfile);
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    (ddi_get8(handle, &CS4231_IDR) & ~RADCI_IN_MASK)|tmp_value,
-		    RADCI_VALID_MASK, __LINE__, thisfile);
-		mutex_exit(&state->cs_lock);
-	}
-
-done:
-	ATRACE("cs_set_port() returning", rc);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (rc);
-
-}	/* cs4231_set_port() */
-
-/*
- * cs4231_set_monitor_gain()
- *
- * Description:
- *	Set the monitor gain.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *	int		gain		The gain to set
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec parameter has been set
- *	AUDIO_FAILURE		The gain has not been set
- */
-static int
-cs4231_set_monitor_gain(CS_state_t *state, int gain)
-{
-	audiohdl_t		ahandle = state->cs_ahandle;
-	ddi_acc_handle_t	handle;
-	int			rc = AUDIO_SUCCESS;
-
-	ATRACE("in cs_set_monitor gain()", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	if (gain > AUDIO_MAX_GAIN) {	/* sanity check */
-		gain = AUDIO_MAX_GAIN;
-	}
-
-	handle = state->cs_handles.cs_codec_hndl;
-	mutex_enter(&state->cs_lock);
-	cs4231_reg_select(ahandle, handle, &CS4231_IAR, LC_REG,
-	    __LINE__, thisfile);
-
-	if (gain == 0) {
-		/* disable loopbacks when gain == 0 */
-		cs4231_put8(ahandle, handle, &CS4231_IDR, LC_OFF,
-		    LC_VALID_MASK, __LINE__, thisfile);
-	} else {
-		/* we use cs4231_atten[] to linearize attenuation */
-		cs4231_put8(ahandle, handle, &CS4231_IDR,
-		    ((cs4231_atten[gain] << 2) | LC_LBE), LC_VALID_MASK,
-		    __LINE__, thisfile);
-	}
-	mutex_exit(&state->cs_lock);
-
-	ATRACE("cs_set_monitor_gain() returning", rc);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (rc);
-
-}	/* cs4231_set_monitor_gain() */
-
-/*
- * cs4231_set_busy()
- *
- * Description:
- *	This routine is called whenever a routine needs to guarantee
- *	that it will not be suspended or the power removed by the power
- *	manager. It will also block any routine while a suspend is
- *	going on.
- *
- *	CAUTION: This routine cannot be called by routines that will
- *		block. Otherwise DDI_SUSPEND will be blocked for a
- *		long time. And that is the wrong thing to do.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS			Set busy and powered up
- *	AUDIO_FAILURE			Couldn't power up, so not busy
- */
-static int
-cs4231_set_busy(CS_state_t *state)
-{
-	ATRACE("in cs4231_set_busy()", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&state->cs_lock);
-
-	/* block if we are going to be suspended */
-	while (state->cs_suspended == CS4231_SUSPENDED) {
-		cv_wait(&state->cs_cv, &state->cs_lock);
-	}
-
-	/*
-	 * Okay, we aren't going to be suspended yet, so mark as busy.
-	 * This will keep us from being suspended when we release the lock.
-	 */
-	ASSERT(state->cs_busy_cnt >= 0);
-	state->cs_busy_cnt++;
-
-	/* now can release the lock before we raise the power */
-	mutex_exit(&state->cs_lock);
-
-	/*
-	 * Mark as busy before we ask for power to be raised. This removes
-	 * the race condtion between the call to cs4231_power() and our call
-	 * to raise power. After we raise power we immediately mark as idle
-	 * so the count is still good.
-	 */
-	(void) pm_busy_component(state->cs_dip, CS4231_COMPONENT);
-	if (pm_raise_power(state->cs_dip, CS4231_COMPONENT, CS4231_PWR_ON) ==
-	    DDI_FAILURE) {
-		/* match the busy call above */
-		(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
-
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!%s%d:set_busy() power up failed",
-		    audiocs_name, state->cs_instance);
-
-		mutex_enter(&state->cs_lock);
-		state->cs_busy_cnt--;		/* restore busy count */
-		if (state->cs_busy_cnt == 0) {
-			/* let DDI_SUSPEND continue */
-			cv_broadcast(&state->cs_cv);
-		}
-		mutex_exit(&state->cs_lock);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* power is up and we are marked as busy, so we are done */
-
-	ATRACE("audiocs_set_busy() done", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	return (AUDIO_SUCCESS);
-
-}	/* cs4231_set_busy() */
-
-/*
- * cs4231_set_idle()
- *
- * Description:
- *	This routine reduces the busy count. It then does a cv_broadcast()
- *	if the count is 0 so a waiting DDI_SUSPEND will continue forward.
- *	It ends by resetting the power management timer.
- *
- *	We don't do anything with power because the routine that is no longer
- *	busy either doesn't need the hardware, or we are playing or recording
- *	so the power won't come down anyway.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-cs4231_set_idle(CS_state_t *state)
-{
-	ATRACE("in cs4231_set_idle()", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-	/* get the lock so we are safe */
-	mutex_enter(&state->cs_lock);
-
-	ASSERT(state->cs_suspended == CS4231_NOT_SUSPENDED);
-
-	/* decrement the busy count */
-	state->cs_busy_cnt--;
-
-	/* if no longer busy, then we wake up a waiting SUSPEND */
-	if (state->cs_busy_cnt == 0) {
-		cv_broadcast(&state->cs_cv);
-	}
-
-	/* we're done, so unlock */
-	mutex_exit(&state->cs_lock);
-
-	/* reset the timer */
-	(void) pm_idle_component(state->cs_dip, CS4231_COMPONENT);
-
-	ATRACE("cs4231_set_idle() done", state);
-	ASSERT(!mutex_owned(&state->cs_lock));
-
-}	/* cs4231_set_idle() */
-
-
-/* *******  Global Local Routines ******************************************* */
-
-/*
- * cs4231_poll_ready()
- *
- * Description:
- *	This routine waits for the Codec to complete its initialization
- *	sequence and is done with its autocalibration.
- *
- *	Early versions of the Codec have a bug that can take as long as
- *	15 seconds to complete its initialization. For these cases we
- *	use a timeout mechanism so we don't keep the machine locked up.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		The Codec is ready to continue
- *	AUDIO_FAILURE		The Codec isn't ready to continue
- */
-int
-cs4231_poll_ready(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = state->cs_handles.cs_codec_hndl;
-	int			x = 0;
-	uint8_t			iar;
-	uint8_t			idr;
-
-	ATRACE("in cs_poll_ready()", state);
-
-	ASSERT(state->cs_regs != NULL);
-	ASSERT(handle != NULL);
-
-	/* wait for the chip to initialize itself */
-	iar = ddi_get8(handle, &CS4231_IAR);
-
-	while ((iar & IAR_INIT) && x++ < CS4231_TIMEOUT) {
-		drv_usecwait(50);
-		iar = ddi_get8(handle, &CS4231_IAR);
-	}
-
-	if (x >= CS4231_TIMEOUT) {
-		ATRACE("cs_poll_ready() timeout #1", state);
-		return (AUDIO_FAILURE);
-	}
-
-	x = 0;
-
-	/*
-	 * Now wait for the chip to complete its autocalibration.
-	 * Set the test register.
-	 */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, ESI_REG,
-	    __LINE__, thisfile);
-
-	idr = ddi_get8(handle, &CS4231_IDR);
-
-	while ((idr & ESI_ACI) && x++ < CS4231_TIMEOUT) {
-		drv_usecwait(50);
-		idr = ddi_get8(handle, &CS4231_IDR);
-	}
-
-	if (x >= CS4231_TIMEOUT) {
-		ATRACE("cs_poll_ready() timeout #2", state);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("cs_poll_ready() returning", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* cs4231_poll_ready() */
-
-/*
- * cs4231_reg_select()
- *
- * Description:
- *	Select a cs4231 register. The cs4231 has a hardware bug where a
- *	register is not always selected the first time. We try and try
- *	again until the proper register is selected or we time out and
- *	print an error message.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	ddi_acc_handle_t handle		A handle to the device's registers
- *	uint8_t		addr		The register address to program
- *	int		reg		The register to select
- *	int		line		The line number where this function was
- *					called
- *	char *		*thefile	The name of the c file that called this
- *					function
- *
- * Returns:
- *	void
- */
-void
-cs4231_reg_select(audiohdl_t ahandle, ddi_acc_handle_t handle, uint8_t *addr,
-    uint8_t reg, int line, char *thefile)
-{
-	int		x;
-	uint8_t		T;
-
-	for (x = 0; x < CS4231_RETRIES; x++) {
-		ddi_put8(handle, addr, reg);
-		T = ddi_get8(handle, addr);
-		if (T == reg) {
-			break;
-		}
-		drv_usecwait(1000);
-	}
-
-	if (x == CS4231_RETRIES) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "!Couldn't select register (%s, Line #%d 0x%02x 0x%02x)",
-		    thefile, line, T, reg);
-		audio_sup_log(ahandle, CE_CONT,
-		    "!audio may not work correctly until it is stopped and "
-		    "restarted\n");
-	}
-
-}	/* cs4231_reg_select() */
-
-/*
- * cs4231_put8()
- *
- * Description:
- *	Program a cs4231 register. The cs4231 has a hardware bug where a
- *	register is not programmed properly the first time. We program a value,
- *	then immediately read back the value and reprogram if nescessary.
- *	We do this until the register is properly programmed or we time out and
- *	print an error message.
- *
- * Arguments:
- *	audiohdl_t	ahandle		Handle to this device
- *	ddi_acc_handle_t handle		A handle to the device's registers
- *	uint8_t		addr		The register address to program
- *	uint8_t		mask		Mask to not set reserved register bits
- *	int		val		The value to program
- *	int		line		The line number where this function was
- *					called
- *	char *		*thefile	The name of the c file that called this
- *					function
- *
- * Returns:
- *	void
- */
-void
-cs4231_put8(audiohdl_t ahandle, ddi_acc_handle_t handle, uint8_t *addr,
-    uint8_t val, uint8_t mask, int line, char *thefile)
-{
-	int		x;
-	uint8_t		T;
-
-	val &= mask;
-
-	for (x = 0; x < CS4231_RETRIES; x++) {
-		ddi_put8(handle, addr, val);
-		T = ddi_get8(handle, addr);
-		if (T == val) {
-			break;
-		}
-		drv_usecwait(1000);
-	}
-
-	if (x == CS4231_RETRIES) {
-		audio_sup_log(ahandle, CE_NOTE,
-		    "!Couldn't set value (%s, Line #%d 0x%02x 0x%02x)",
-		    thefile, line, T, val);
-		audio_sup_log(ahandle, CE_CONT,
-		    "!audio may not work correctly until it is stopped and "
-		    "restarted\n");
-	}
-
-}	/* cs4231_put8() */
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_apcdma.c b/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_apcdma.c
deleted file mode 100644
index 6f55c57d35..0000000000
--- a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_apcdma.c
+++ /dev/null
@@ -1,1332 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * Platform specifc code for the APC DMA controller. The APC is an SBus
- * IC that includes play and record DMA engines and an interface for
- * the CS4231.
- */
-
-#include <sys/systm.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src1.h>
-#include <sys/audio/impl/audio_4231_impl.h>
-#include <sys/audio/audio_4231.h>
-
-/*
- * Attribute structure for the APC, used to create DMA handles.
- */
-static ddi_dma_attr_t apc_attr = {
-	DMA_ATTR_V0,			/* version */
-	0x0000000000000000LL,		/* dlim_addr_lo */
-	0x00000000ffffffffLL,		/* dlim_addr_hi */
-	0x0000000000000fffLL,		/* DMA counter register */
-	0x0000000000000001LL,		/* DMA address alignment */
-	0x00000014,			/* 4 and 16 byte burst sizes */
-	0x00000001,			/* min effective DMA size */
-	0x0000000000000fffLL,		/* maximum transfer size, 8k */
-	0x000000000000ffffLL,		/* segment boundary, 32k */
-	0x00000001,			/* s/g list length, no s/g */
-	0x00000001,			/* granularity of device, don't care */
-	0				/* DMA flags */
-};
-
-static ddi_device_acc_attr_t acc_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_BE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-/*
- * Local routines
- */
-static uint_t apc_intr(caddr_t);
-static int apc_program_play(CS_state_t *, int, caddr_t *, int);
-
-/*
- * DMA ops vector functions
- */
-static int apc_map_regs(dev_info_t *, CS_state_t *, size_t, size_t);
-static void apc_unmap_regs(CS_state_t *);
-static void apc_reset(CS_state_t *);
-static int apc_add_intr(CS_state_t *);
-static void apc_rem_intr(dev_info_t *);
-static int apc_p_start(CS_state_t *);
-static void apc_p_pause(CS_state_t *);
-static void apc_p_restart(CS_state_t *);
-static void apc_p_stop(CS_state_t *);
-static int apc_r_start(CS_state_t *);
-static void apc_r_stop(CS_state_t *);
-static void apc_power(CS_state_t *, int);
-
-cs4231_dma_ops_t cs4231_apcdma_ops = {
-	"APC DMA controller",
-	apc_map_regs,
-	apc_unmap_regs,
-	apc_reset,
-	apc_add_intr,
-	apc_rem_intr,
-	apc_p_start,
-	apc_p_pause,
-	apc_p_restart,
-	apc_p_stop,
-	apc_r_start,
-	apc_r_stop,
-	apc_power
-};
-
-/* File name for the cs4231_put8() and cs4231_reg_select() routines */
-static char *thisfile = __FILE__;
-
-/*
- * apc_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. It then binds each of the buffers to its
- *	respective handle, getting a DMA cookie. Finally, the registers
- *	are mapped in.
- *
- *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
- *		memory. This means these calls will almost always succeed.
- *
- * Arguments:
- *	dev_info_t	*dip		Pointer to the device's devinfo
- *	CS_state_t	*state		The device's state structure
- *	size_t		pbuf_size	The size of the play DMA buffers to
- *					allocate, we need two
- *	size_t		cbuf_size	The size of the capture DMA buffers to
- *					allocate, we need two
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-apc_map_regs(dev_info_t *dip, CS_state_t *state, size_t pbuf_size,
-	size_t cbuf_size)
-{
-	ddi_acc_handle_t	*handle = &APC_HANDLE;
-	uint_t			dma_cookie_count;
-	int			rc;
-
-	ATRACE("in apc_map_regs()", state);
-
-	/* allocate two handles for play and two for record */
-	if ((rc = ddi_dma_alloc_handle(dip, &apc_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ph[0])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_alloc_handle(0) failed: %d", rc);
-		goto error;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &apc_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ph[1])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_alloc_handle(1) failed: %d", rc);
-		goto error_ph0;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &apc_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ch[0])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_alloc_handle(2) failed: %d", rc);
-		goto error_ph1;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &apc_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ch[1])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_alloc_handle(3) failed: %d", rc);
-		goto error_rh0;
-	}
-
-	/* allocate the four DMA buffers, two for play and two for record */
-	if (ddi_dma_mem_alloc(state->cs_ph[0], pbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_pb[0], &state->cs_pml[0],
-	    &state->cs_pmh[0]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_mem_alloc(0) failed");
-		goto error_rh1;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ph[1], pbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_pb[1], &state->cs_pml[1],
-	    &state->cs_pmh[1]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_mem_alloc(1) failed");
-		goto error_pb0;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ch[0], cbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_cb[0], &state->cs_cml[0],
-	    &state->cs_cmh[0]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_mem_alloc(2) failed");
-		goto error_pb1;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ch[1], cbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_cb[1], &state->cs_cml[1],
-	    &state->cs_cmh[1]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_mem_alloc(3) failed");
-		goto error_rb0;
-	}
-
-	/* bind each of the buffers to a DMA handle */
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ph[0], (struct as *)0,
-	    state->cs_pb[0], state->cs_pml[0], DDI_DMA_WRITE,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->cs_pc[0],
-	    &dma_cookie_count)) != DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_addr_bind_handle(0) failed: %d",
-		    rc);
-		goto error_rb1;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ph[1], (struct as *)0,
-	    state->cs_pb[1], state->cs_pml[1], DDI_DMA_WRITE,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->cs_pc[1],
-	    &dma_cookie_count)) != DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_addr_bind_handle(1) failed: %d",
-		    rc);
-		goto error_pc0;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ch[0], (struct as *)0,
-	    state->cs_cb[0], state->cs_cml[0], DDI_DMA_READ,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->cs_cc[0],
-	    &dma_cookie_count)) != DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_addr_bind_handle(2) failed: %d",
-		    rc);
-		goto error_pc1;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ch[1], (struct as *)0,
-	    state->cs_cb[1], state->cs_cml[1], DDI_DMA_READ,
-	    DDI_DMA_SLEEP, (caddr_t)0, &state->cs_cc[1],
-	    &dma_cookie_count)) != DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_dma_addr_bind_handle(3) failed: %d",
-		    rc);
-		goto error_rc0;
-	}
-	ASSERT(dma_cookie_count == 1);
-
-	/* map in the registers, getting a handle */
-	if (ddi_regs_map_setup(dip, 0, (caddr_t *)&state->cs_regs, 0,
-	    sizeof (cs4231_regs_t), &acc_attr, handle) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_map_regs() ddi_regs_map_setup() failed: %d", rc);
-		goto error_rc1;
-	}
-
-	/* clear the CSR so we have all interrupts disabled */
-	ddi_put32(*handle, &APC_DMACSR, APC_CLEAR_RESET_VALUE);
-
-	/* let the state structure know about the attributes */
-	state->cs_dma_attr = &apc_attr;
-
-	/* we start with play and record buffers 0 */
-	state->cs_pbuf_toggle = 0;
-	state->cs_cbuf_toggle = 0;
-
-	ATRACE("apc_map_regs() returning success", state);
-
-	return (AUDIO_SUCCESS);
-
-error_rc1:
-	(void) ddi_dma_unbind_handle(state->cs_ch[1]);
-error_rc0:
-	(void) ddi_dma_unbind_handle(state->cs_ch[0]);
-error_pc1:
-	(void) ddi_dma_unbind_handle(state->cs_ph[1]);
-error_pc0:
-	(void) ddi_dma_unbind_handle(state->cs_ph[0]);
-error_rb1:
-	ddi_dma_mem_free(&state->cs_cmh[1]);
-error_rb0:
-	ddi_dma_mem_free(&state->cs_cmh[0]);
-error_pb1:
-	ddi_dma_mem_free(&state->cs_pmh[1]);
-error_pb0:
-	ddi_dma_mem_free(&state->cs_pmh[0]);
-error_rh1:
-	ddi_dma_free_handle(&state->cs_ch[1]);
-error_rh0:
-	ddi_dma_free_handle(&state->cs_ch[0]);
-error_ph1:
-	ddi_dma_free_handle(&state->cs_ph[1]);
-error_ph0:
-	ddi_dma_free_handle(&state->cs_ph[0]);
-error:
-	ATRACE("apc_map_regs() returning failure", state);
-
-	return (AUDIO_FAILURE);
-
-}	/* apc_map_regs() */
-
-/*
- * apc_unmap_regs()
- *
- * Description:
- *	This routine unbinds the DMA cookies, frees the DMA buffers,
- *	deallocated the DMA handles, and finally unmaps the Codec's and
- *	DMA engine's registers.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_unmap_regs(CS_state_t *state)
-{
-	ATRACE("in apc_unmap_regs()", state);
-
-	(void) ddi_dma_unbind_handle(state->cs_ph[0]);
-	(void) ddi_dma_unbind_handle(state->cs_ph[1]);
-	(void) ddi_dma_unbind_handle(state->cs_ch[0]);
-	(void) ddi_dma_unbind_handle(state->cs_ch[1]);
-
-	ddi_dma_mem_free(&state->cs_pmh[0]);
-	ddi_dma_mem_free(&state->cs_pmh[1]);
-	ddi_dma_mem_free(&state->cs_cmh[0]);
-	ddi_dma_mem_free(&state->cs_cmh[1]);
-
-	ddi_dma_free_handle(&state->cs_ph[0]);
-	ddi_dma_free_handle(&state->cs_ph[1]);
-	ddi_dma_free_handle(&state->cs_ch[0]);
-	ddi_dma_free_handle(&state->cs_ch[1]);
-
-	ddi_regs_map_free(&APC_HANDLE);
-
-	ATRACE("apc_unmap_regs() returning", state);
-
-}	/* apc_unmap_regs() */
-
-/*
- * apc_reset()
- *
- * Description:
- *	Reset both the play and record DMA engines. The engines are left
- *	with interrupts and the DMA engine disabled.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo structure
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_reset(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-
-	ATRACE("in apc_reset()", state);
-
-	/*
-	 * The APC has a bug where the reset is not done
-	 * until you do the next pio to the APC. This
-	 * next write to the CSR causes the posted reset to
-	 * happen.
-	 */
-
-	ddi_put32(handle, &APC_DMACSR, APC_RESET);
-	ddi_put32(handle, &APC_DMACSR, APC_CLEAR_RESET_VALUE);
-
-	ATRACE("apc_reset() returning", state);
-
-}	/* apc_reset() */
-
-/*
- * apc_add_intr()
- *
- * Description:
- *	Register the APC interrupts with the kernel.
- *
- *	NOTE: This does NOT turn on interrupts.
- *
- *	CAUTION: While the interrupts are added, the Codec interrupts are
- *		not enabled.
- *
- * Arguments:
- *	CS_state_t	*state	Pointer to the device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-apc_add_intr(CS_state_t *state)
-{
-	dev_info_t	*dip = state->cs_dip;
-
-	ATRACE("in apc_add_intr()", state);
-
-	/* first we make sure this isn't a high level interrupt */
-	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_add_intr() unsupported high level interrupt");
-		return (AUDIO_FAILURE);
-	}
-
-	/* okay to register the interrupt */
-	if (ddi_add_intr(dip, 0, (ddi_iblock_cookie_t *)NULL,
-	    (ddi_idevice_cookie_t *)NULL, apc_intr,
-	    (caddr_t)state) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!apc_add_intr() bad interrupt specification");
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("apc_add_intr() returning successful", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* apc_add_intr() */
-
-/*
- * apc_rem_intr()
- *
- * Description:
- *	Unregister the APC interrupts from the kernel.
- *
- *	CAUTION: While the interrupts are removed, the Codec interrupts are
- *		not disabled, but then, they never should have been on in
- *		the first place.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo
- *
- * Returns:
- *	void
- */
-static void
-apc_rem_intr(dev_info_t *dip)
-{
-	ATRACE("in apc_rem_intr()", dip);
-
-	ddi_remove_intr(dip, 0, NULL);
-
-	ATRACE("apc_rem_intr() returning", dip);
-
-}	/* apc_rem_intr() */
-
-/*
- * apc_p_start()
- *
- * Description:
- *	This routine starts the play DMA engine. This includes "hard" starts
- *	where the DMA engine's registers need to be loaded as well as starting
- *	after a pause.
- *
- *	For hard starts the DMA engine is started by programming the Play Next
- *	Virtual Address and then the Play Next Counter twice, and finally
- *	enabling the play DMA engine.
- *
- *	Starting after a pause is much eaiser, we just turn on the DMA GO bit.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- *	CAUTION: This routine doesn't start the Codec because the first
- *		interrupt causes a recursive mutex_enter.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Successfully started
- *	AUDIO_FAILURE		Start failed
- */
-static int
-apc_p_start(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	int			i;
-	int			samples;
-
-	ATRACE("in apc_p_start()", state);
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* figure out the number of samples we want */
-	samples = state->cs_play_sr * state->cs_play_ch /
-	    state->cs_ad_info.ad_play.ad_int_rate;
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples % state->cs_play_ch) == 1) {
-		ATRACE("apc_p_start() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-	ATRACE("apc_p_start() samples per interrupt", samples);
-
-	/* try to load 2 buffers worth to get things started */
-	for (i = 0; i < 2; i++) {
-		ATRACE_32("apc_p_start() calling apc_program_play()", samples);
-		PLAY_COUNT = samples *
-		    (state->cs_play_prec >> AUDIO_PRECISION_SHIFT);
-		if (apc_program_play(state, samples, NULL, 0) <= 0) {
-			ATRACE("apc_p_start() "
-			    "apc_program_play() returns no samples", state);
-			break;
-		}
-	}
-
-	/*
-	 * If there wasn't anything to get then we aren't active. This can
-	 * happen if an AUDIO_DRAIN was issued with no audio in the stream.
-	 */
-	if (i == 0) {
-		ATRACE("apc_p_start() nothing to do", state);
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Even with just one DMA buffer loaded we can still play some audio.
-	 * Read the Codec Status Register so that we clear the Error and
-	 * Initialization register, Index 11, so we'll know when we get
-	 * a play underrun in the ISR. Then enable the play DMA engine,
-	 * including interrupts.
-	 */
-#ifdef DEBUG
-	ATRACE_8("apc_p_start() Read the Codec status reg, clearing Index 11",
-	    ddi_get8(handle, &CS4231_STATUS));
-#else
-	(void) ddi_get8(handle, &CS4231_STATUS); /* we only need a read */
-#endif
-
-	ATRACE("apc_p_start() turn on play DMA engine", state);
-	OR_SET_WORD(handle, &APC_DMACSR, APC_PLAY_ENABLE);
-
-	state->cs_flags |= PDMA_ENGINE_INITIALIZED;
-
-	ATRACE("apc_p_start() returning", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* apc_p_start() */
-
-/*
- * apc_p_pause()
- *
- * Description:
- *	This routine pauses the play DMA engine. Buffers, FIFO, etc. are NOT
- *	lost.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_p_pause(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in apc_p_pause()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* clear the PDMA_GO bit, pausing the DMA engine */
-	AND_SET_WORD(handle, &APC_DMACSR, ~APC_PDMA_GO);
-
-	/* we wait for the Codec FIFO to underrun */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, ESI_REG,
-	    __LINE__, thisfile);
-	creg = ddi_get8(handle, &CS4231_IDR);
-	for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		creg = ddi_get8(handle, &CS4231_IDR);
-	}
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_PEN, INTC_VALID_MASK);
-
-	ATRACE("apc_p_pause() returning", state);
-
-}	/* apc_p_pause() */
-
-/*
- * apc_p_restart()
- *
- * Description:
- *	This routine restarts the play DMA engine after pauseing. Buffers,
- *	FIFO, etc. Must be programmed and valid.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_p_restart(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-
-	ATRACE("in apc_p_restart()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* start the Codec */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	OR_SET_BYTE(handle, &CS4231_IDR, INTC_PEN, INTC_VALID_MASK);
-
-	/* set the PDMA_GO bit, restarting the DMA engine */
-	OR_SET_WORD(handle, &APC_DMACSR, APC_PDMA_GO);
-
-	ATRACE("apc_p_restart() returning", state);
-
-}	/* apc_p_restart() */
-
-/*
- * apc_p_stop()
- *
- * Description:
- *	This routine stops the play DMA engine.
- *
- *	The DMA engine is stopped by using the PLAY_ABORT bit.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_p_stop(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	uint_t			reg;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in apc_p_stop()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* Set stopped flag */
-	state->cs_flags |= APC_P_STOPPED;
-
-	/* set the play abort bit to stop playing audio */
-	OR_SET_WORD(handle, &APC_DMACSR, APC_P_ABORT)
-
-	/* wait for the pipeline to empty */
-	reg = ddi_get32(handle, &APC_DMACSR);
-	for (x = 0; !(reg & APC_PM) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		reg = ddi_get32(handle, &APC_DMACSR);
-	}
-
-#ifdef DEBUG
-	if (x >= CS4231_TIMEOUT) {
-		ATRACE("apc_p_stop() timeout", state);
-	}
-#endif
-
-	/* now clear the play enable and abort bits */
-	AND_SET_WORD(handle, &APC_DMACSR, ~(APC_PDMA_GO|APC_P_ABORT));
-
-	/* we wait for the Codec FIFO to underrun */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, ESI_REG,
-	    __LINE__, thisfile);
-	creg = ddi_get8(handle, &CS4231_IDR);
-	for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		creg = ddi_get8(handle, &CS4231_IDR);
-	}
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_PEN, INTC_VALID_MASK);
-
-	state->cs_flags &= ~PDMA_ENGINE_INITIALIZED;
-
-	ATRACE("apc_p_stop() returning", state);
-
-}	/* apc_p_stop() */
-
-/*
- * apc_r_start()
- *
- * Description:
- *	This routine starts the record DMA engine. The DMA engine is never
- *	paused for record, so a pause is equivalent to a stop. Thus all starts
- *	are hard starts.
- *
- *	For hard starts the DMA engine is started by programming the Record
- *	Next Virtual Address and then the Record Next Counter twice, and
- *	finally enabling the record DMA engine.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- *	CAUTION: This routine doesn't start the Codec because the first
- *		interrupt causes a recursive mutex_enter.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		The capture DMA was started
- *	AUDIO_FAILURE		The capture DMA was not started
- */
-static int
-apc_r_start(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	uint_t			csr;
-	int			bytes;
-	int			i;
-	int			samples;
-	int			x;
-
-	ATRACE("in apc_r_start()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* figure out the number of samples to capture */
-	samples = state->cs_record_sr * state->cs_record_ch /
-	    state->cs_ad_info.ad_record.ad_int_rate;
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples % state->cs_record_ch) == 1) {
-		ATRACE("apc_r_start() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-	ATRACE("apc_r_start() samples per interrupt", samples);
-
-	/* now convert the number of samples to the "size" in bytes */
-	ASSERT(state->cs_record_prec == AUDIO_PRECISION_8 ||
-	    state->cs_record_prec == AUDIO_PRECISION_16);
-	bytes = samples * (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-	ATRACE("apc_r_start() DMA count", bytes);
-
-	/* make sure it's okay to program the Next Address/Count registers */
-	csr = ddi_get32(handle, &APC_DMACSR);
-	ATRACE_32("apc_r_start() CSR", csr);
-	for (x = 0; !(csr & APC_CD) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* no reason to beat on the bus */
-		csr = ddi_get32(handle, &APC_DMACSR);
-	}
-	if (x >= CS4231_TIMEOUT) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE, "!apc_r_start() "
-		    "timeout waiting for Codec, record not started!");
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Program the DMA engine with both buffers. We MUST do both buffers
-	 * otherwise CAP_COUNT isn't going to set both byte counts.
-	 */
-	for (i = 0; i < 2; i++) {
-		/* read the CNVA, as per APC document */
-		(void) ddi_get32(handle, &APC_DMACNVA);
-
-		/* sync the DMA buffer before it is going to be used */
-		ATRACE("apc_r_start() dma buffer sync", state);
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!apc_r_start() ddi_dma_sync() failed, recording "
-			    "stopped");
-
-			/* send a cap. abort, this leaves the DMA engine ok */
-			OR_SET_WORD(handle, &APC_DMACSR, APC_C_ABORT);
-
-			/* wait for the pipeline to empty */
-			csr = ddi_get32(handle, &APC_DMACSR);
-			for (x = 0; !(csr & APC_CX) && x < CS4231_TIMEOUT;
-			    x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				csr = ddi_get32(handle, &APC_DMACSR);
-			}
-
-			/* now clear it */
-			AND_SET_WORD(handle, &APC_DMACSR, ~APC_C_ABORT);
-
-			return (AUDIO_FAILURE);
-		}
-
-		ATRACE_32("apc_r_start() next address",
-		    (uint_t)CAP_DMA_COOKIE.dmac_address);
-		ddi_put32(handle, &APC_DMACNVA,
-		    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-		/* now program the CNC reg., which enables the state machine */
-		ATRACE_32("apc_r_start() next count", (uint_t)(bytes));
-		ddi_put32(handle, &APC_DMACNC, (uint_t)(bytes));
-		CAP_COUNT = bytes;
-
-		/* now get ready for the next DMA buffer */
-		AUDIO_TOGGLE(state->cs_cbuf_toggle);
-		ATRACE_32("apc_r_start() new toggle", state->cs_cbuf_toggle);
-	}
-
-	state->cs_flags |= RDMA_ENGINE_INITIALIZED;
-
-	/* start the DMA engine */
-	OR_SET_WORD(handle, &APC_DMACSR, APC_CAP_ENABLE);
-	ATRACE("apc_r_start() returning success", 0);
-
-	return (AUDIO_SUCCESS);
-
-}	/* apc_r_start() */
-
-/*
- * apc_r_stop()
- *
- * Description:
- *	This routine stops the record DMA engine. It then sends any collected
- *	data to the audio mixer.
- *
- *	The DMA engine is stopped by using the CAP_ABORT bit.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-apc_r_stop(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	uint_t			reg;
-	int			samples;
-	int			x;
-
-	ATRACE("in apc_r_stop()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* clear the record interrupts so the ISR doesn't get involved */
-	AND_SET_WORD(handle, &APC_DMACSR, ~APC_CINTR_ENABLE);
-
-	/* first, abort the record DMA engine */
-	OR_SET_WORD(handle, &APC_DMACSR, APC_C_ABORT);
-
-	/* wait for the pipeline to empty */
-	reg = ddi_get32(handle, &APC_DMACSR);
-	for (x = 0; !(reg & APC_CX) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		reg = ddi_get32(handle, &APC_DMACSR);
-	}
-
-#ifdef DEBUG
-	if (x >= CS4231_TIMEOUT) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "apc_r_stop() timeout, record buffer flushed");
-	}
-#endif
-
-	/* clear the CXI interrupt bit, but CXI interrupt didn't go out */
-	OR_SET_WORD(handle, &APC_DMACSR, APC_CXI_EN);
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_CEN, INTC_VALID_MASK);
-
-	/* figure how many samples were recorded */
-	samples = (CAP_COUNT - ddi_get32(handle, &APC_DMACC)) /
-	    (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-
-	/* send the captured audio to the mixer */
-	if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0, DDI_DMA_SYNC_FORCPU) ==
-	    DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!apc_r_stop() ddi_dma_sync() failed, recorded audio lost");
-	} else {
-		mutex_exit(&state->cs_lock);
-		am_send_audio(state->cs_ahandle, CAP_DMA_BUF, AUDIO_NO_CHANNEL,
-		    samples);
-		mutex_enter(&state->cs_lock);
-	}
-
-	AND_SET_WORD(handle, &APC_DMACSR, ~(APC_C_ABORT|APC_CAP_DISABLE));
-
-	/* check that the pipeline is empty */
-	reg = ddi_get32(handle, &APC_DMACSR);
-	for (x = 0; !(reg & APC_CD) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		reg = ddi_get32(handle, &APC_DMACSR);
-	}
-
-#ifdef DEBUG
-	if (x >= CS4231_TIMEOUT) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "apc_r_stop() buffer pipeline not empty");
-	}
-#endif
-
-	state->cs_flags &= ~RDMA_ENGINE_INITIALIZED;
-
-	ATRACE("apc_r_stop() returning", state);
-
-}	/* apc_r_stop() */
-
-/*
- * apc_power()
- *
- * Description:
- *	This routine turns the Codec off by using the COD_PDWN bit in the
- *	apc chip. To turn power on we have to reset the APC, which clears
- *	the COD_PDWN bit. However, this is a settling bug in the APC which
- *	requires the driver to delay quite a while before we may continue.
- *	Since this is the first time this feature has actually been used
- *	it isn't too surprising that it has some problems.
- *
- *	NOTE: The state structure must be locked when this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state		Ptr to the device's state structure
- *	int		level		Power level to set
- *
- * Returns:
- *	void
- */
-static void
-apc_power(CS_state_t *state, int level)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-
-	ATRACE("in apc_power()", state);
-	ATRACE_32("apc_power() level", level);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	if (level == CS4231_PWR_ON) {	/* turn power on */
-		AND_SET_WORD(handle, &APC_DMACSR, ~APC_COD_PDWN);
-		OR_SET_WORD(handle, &APC_DMACSR, APC_RESET);
-		AND_SET_WORD(handle, &APC_DMACSR, ~APC_RESET);
-
-		/*
-		 * wait for state change,
-		 * __lock_lint tells warlock not to flag this delay()
-		 */
-#ifndef __lock_lint
-		delay(drv_usectohz(CS4231_300MS));
-#endif
-		state->cs_powered = CS4231_PWR_ON;
-	} else {	/* turn power off */
-		ASSERT(level == CS4231_PWR_OFF);
-		OR_SET_WORD(handle, &APC_DMACSR, APC_COD_PDWN);
-		state->cs_powered = CS4231_PWR_OFF;
-	}
-
-	ATRACE_32("apc_power() done", ddi_get32(handle, &APC_DMACSR));
-
-}	/* apc_power() */
-
-
-/* *******  Local Routines ************************************************** */
-
-/*
- * apc_intr()
- *
- * Description:
- *	APC interrupt service routine, which services both play and capture
- *	interrupts. First we find out why there was an interrupt, then we
- *	take the appropriate action.
- *
- *	Because this ISR deals with both play and record interrupts we have
- *	to be careful to not lose an interrupt. So we service the record
- *	interrupt first and save the incoming data until later. This is all
- *	done without releasing the lock, thus there can be no race conditions.
- *	Then we process the play interrupt. While processing the play interrupt
- *	we have to release the lock. When this happens we send recorded data
- *	to the mixer and then get the next chunk of data to play. If there
- *	wasn't a play interrupt then we finish by sending the recorded data,
- *	if any.
- *
- * Arguments:
- *	caddr_t		T	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-apc_intr(caddr_t T)
-{
-	CS_state_t		*state = (CS_state_t *)T;
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	caddr_t			rec_buf = NULL;
-	uint_t			csr;
-	uint_t			reg;
-	int			rc = DDI_INTR_UNCLAIMED;
-	int			rec_samples;
-	int			samples;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in apc_intr()", state);
-
-	/* the state must be protected */
-	mutex_enter(&state->cs_lock);
-
-	/* get the APC CSR */
-	csr = ddi_get32(handle, &APC_DMACSR);
-	ATRACE_32("apc_intr() interrupt CSR", csr);
-
-	/* make sure this device sent the interrupt */
-	if (!(csr & APC_IP)) {
-		/* Interrupt received when stopping play */
-		if (state->cs_flags & APC_P_STOPPED) {
-			/* Clear stopped flag */
-			state->cs_flags &= ~APC_P_STOPPED;
-			mutex_exit(&state->cs_lock);
-			return (DDI_INTR_CLAIMED);
-		} else if (csr & APC_PMI_EN) {
-			/*
-			 * Clear device generated interrupt while play is
-			 * active (Only seen while playing and insane mode
-			 * switching)
-			 */
-			mutex_exit(&state->cs_lock);
-			return (DDI_INTR_CLAIMED);
-		} else {
-			/* nope, this isn't our interrupt */
-			mutex_exit(&state->cs_lock);
-			ATRACE_32("apc_intr() device didn't send interrupt",
-			    csr);
-			return (DDI_INTR_UNCLAIMED);
-		}
-	}
-
-	/* Clear stopped flag */
-	state->cs_flags &= ~APC_P_STOPPED;
-
-	/* clear all interrupts we captured this time */
-	ddi_put32(handle, &APC_DMACSR, csr);
-	ATRACE_32("apc_intr() csr after clear", ddi_get32(handle, &APC_DMACSR));
-
-	if (csr & APC_CINTR_MASK) {
-		ATRACE("apc_intr(R) record interrupt", state);
-
-		/* sync DMA memory before sending it to the audio mixer */
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORCPU) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!apc_intr(R) ddi_dma_sync(#1) failed, recorded "
-			    "audio lost");
-		} else {
-			/* figure how many samples were recorded */
-			ASSERT(ddi_get32(handle, &APC_DMACSR) & APC_CD);
-			samples = CAP_COUNT /
-			    (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-
-			/* save the data for when we free the lock */
-			rec_buf = CAP_DMA_BUF;
-			rec_samples = samples;
-		}
-
-		/* sync the DMA buffer before it is going to be reused */
-		ATRACE("apc_intr(R) dma buffer sync", state);
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!apc_intr(R) ddi_dma_sync(#2) failed, recording "
-			    "disabled");
-
-			/* send a play abort, this leaves the DMA engine ok */
-			OR_SET_WORD(handle, &APC_DMACSR, APC_C_ABORT);
-
-			/* now clear it */
-			AND_SET_WORD(handle, &APC_DMACSR, ~APC_C_ABORT);
-		} else {
-			ATRACE_32("apc_intr(R) next address",
-			    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-			/* read the CNVA, as per APC document */
-			(void) ddi_get32(handle, &APC_DMACNVA);
-
-			/* now program the DMA buffer into the Next Address */
-			ddi_put32(handle, &APC_DMACNVA,
-			    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-			/* program the CNC reg., which enables the state mach */
-			ATRACE_32("apc_intr(R) next count", (uint_t)CAP_COUNT);
-			ddi_put32(handle, &APC_DMACNC, (uint_t)CAP_COUNT);
-
-			/* now get ready for the next DMA buffer */
-			AUDIO_TOGGLE(state->cs_cbuf_toggle);
-			ATRACE_32("apc_intr(R) new toggle",
-			    state->cs_cbuf_toggle);
-
-		}
-		/* we always claim the interrupt, even if DMA sync failed */
-		rc = DDI_INTR_CLAIMED;
-	}
-
-	if (csr & APC_PINTR_MASK) {
-		/* figure out the number of samples we want */
-		samples = PLAY_COUNT /
-		    (state->cs_play_prec >> AUDIO_PRECISION_SHIFT);
-		ATRACE_32("apc_intr(P) samples to get", samples);
-
-		/* try to load the next audio buffer, even if pipe is empty */
-		samples = apc_program_play(state, samples, &rec_buf,
-		    rec_samples);
-		ATRACE_32("apc_intr(P) samples apc_program_play() returned",
-		    samples);
-
-		if (samples <= 0 && (csr & APC_PM)) {
-			/*
-			 * There isn't any more data to play, so wait for
-			 * the pipe and the Codec FIFO to empty. Then turn
-			 * off the play DMA engine by aborting. Also, we should
-			 * note that simple play interrupts with no samples
-			 * are ignored, but acknowledged. We wait for the pipe
-			 * to empty before we declare the DMA engine is empty.
-			 */
-			ATRACE_32("apc_intr(P) no more data, wait for FIFO",
-			    samples);
-
-			/*
-			 * We wait for the Codec FIFO to underrun, this
-			 * implies that the APC pipe is also empty.
-			 */
-			cs4231_reg_select(state->cs_ahandle, handle,
-			    &CS4231_IAR, ESI_REG, __LINE__, thisfile);
-			creg = ddi_get8(handle, &CS4231_IDR);
-			for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT;
-			    x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				creg = ddi_get8(handle, &CS4231_IDR);
-			}
-
-			/*
-			 * Clear the flag so if audio is restarted while in
-			 * am_play_shutdown() we can detect it and not mess
-			 * things up.
-			 */
-			state->cs_flags &= ~PDMA_ENGINE_INITIALIZED;
-
-			/* now shutdown the play stream */
-			ATRACE("apc_intr(P) shutdown play stream", state);
-			mutex_exit(&state->cs_lock);
-
-			/* send the captured audio to the mixer ASAP */
-			if (rec_buf) {
-				am_send_audio(state->cs_ahandle, rec_buf,
-				    AUDIO_NO_CHANNEL, rec_samples);
-				rec_buf = NULL;
-			}
-
-			am_play_shutdown(state->cs_ahandle, NULL);
-
-			mutex_enter(&state->cs_lock);
-
-			/* make sure playing wasn't restarted when lock lost */
-			if (state->cs_flags & PDMA_ENGINE_INITIALIZED) {
-				/* yes, it was, so we're done */
-				ATRACE("apc_intr() restart after shutdown", 0);
-				rc = DDI_INTR_CLAIMED;
-				goto done;
-			}
-
-			/* reset the DMA engine, putting it in a known state */
-			OR_SET_WORD(handle, &APC_DMACSR, APC_P_ABORT);
-
-			/* wait for the pipeline to empty */
-			reg = ddi_get32(handle, &APC_DMACSR);
-			for (x = 0; !(reg & APC_PD) && x < CS4231_TIMEOUT;
-			    x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				reg = ddi_get32(handle, &APC_DMACSR);
-			}
-
-			/* clear the abort bit */
-			AND_SET_WORD(handle, &APC_DMACSR, ~APC_P_ABORT);
-
-			/* disable the Codec */
-			cs4231_reg_select(state->cs_ahandle, handle,
-			    &CS4231_IAR, INTC_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_PEN,
-			    INTC_VALID_MASK);
-
-			/* and reset the status */
-			ddi_put8(handle, &CS4231_STATUS, STATUS_RESET);
-
-			ATRACE("apc_intr(P) Play DMA engine off", state);
-		}
-		rc = DDI_INTR_CLAIMED;
-		ATRACE_32("apc_intr(P) done", rc);
-	}
-
-done:
-
-	/* APC error interrupt, not sure what to do here */
-	if (csr & APC_EI) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!apc_intr() error interrupt: 0x%x", csr);
-		rc = DDI_INTR_CLAIMED;
-	}
-
-	/* update the kernel interrupt statistics */
-	if (state->cs_ksp) {
-		if (rc == DDI_INTR_CLAIMED) {
-			KIOP(state)->intrs[KSTAT_INTR_HARD]++;
-		}
-	}
-
-	mutex_exit(&state->cs_lock);
-
-	/* one last chance to send the captured audio to the mixer */
-	if (rec_buf) {
-		am_send_audio(state->cs_ahandle, rec_buf, AUDIO_NO_CHANNEL,
-		    rec_samples);
-	}
-
-	ATRACE_32("apc_intr() returning", rc);
-
-	return (rc);
-
-}	/* apc_intr() */
-
-/*
- * apc_program_play()
- *
- * Description:
- *	This routine is used by apc_p_start() and apc_intr() to program
- *	the play DMA engine with the next buffer full of audio.
- *
- * Arguments:
- *	CS_state_t	*state		The device's state pointer
- *	int		samples		The number of samples to be retrieved
- *					from the mixer
- *	caddr_t		*rec_buf	Ptr to data buffer for record
- *	int		rec_samples	Num samples in record data buffer
- *
- * Returns:
- *	0			The buffer wasn't programmed, no audio
- *	> 0			The buffer was programmed
- *	AUDIO_FAILURE		The buffer wasn't programmed
- */
-static int
-apc_program_play(CS_state_t *state, int samples, caddr_t *rec_buf,
-	int rec_samples)
-{
-	ddi_acc_handle_t	handle = APC_HANDLE;
-	uint_t			precision;
-	int			rc;
-
-	ATRACE_32("in apc_program_play()", samples);
-
-	/* we need the precision to calculate the next count correctly */
-	precision = state->cs_play_prec >> AUDIO_PRECISION_SHIFT;
-	ATRACE_32("apc_program_play() precision", precision);
-
-	/* send record audio, then get the first buffer's worth of audio */
-	mutex_exit(&state->cs_lock);
-
-	/* send the captured audio to the mixer */
-	if (rec_buf && *rec_buf) {
-		am_send_audio(state->cs_ahandle, *rec_buf, AUDIO_NO_CHANNEL,
-		    rec_samples);
-		*rec_buf = NULL;
-	}
-
-	rc = am_get_audio(state->cs_ahandle, PLAY_DMA_BUF, AUDIO_NO_CHANNEL,
-	    samples);
-
-	mutex_enter(&state->cs_lock);
-	ATRACE_32("apc_program_play() am_get_audio() returned", rc);
-
-	if (rc == AUDIO_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!apc_program_play() am_get_audio() failed");
-		return (AUDIO_FAILURE);
-	} else if (rc == 0) {
-		ATRACE("apc_program_play() am_get_audio() returned 0 samples",
-		    state);
-		return (0);
-	}
-
-	/* sync the DMA buffer before it is going to be used */
-	ATRACE("apc_program_play() dma buffer sync", state);
-	if (ddi_dma_sync(PLAY_DMA_HANDLE, (off_t)0, (size_t)(rc * precision),
-	    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!apc_program_play() ddi_dma_sync(#2) failed, audio lost");
-
-		/* send a play abort, this leaves the DMA engine ok */
-		OR_SET_WORD(handle, &APC_DMACSR, APC_P_ABORT);
-		/* now clear it */
-		AND_SET_WORD(handle, &APC_DMACSR, ~APC_P_ABORT);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* program the PNVA */
-	ATRACE_32("apc_program_play() next address",
-	    (uint_t)PLAY_DMA_COOKIE.dmac_address);
-	ddi_put32(handle, &APC_DMAPNVA, (uint_t)PLAY_DMA_COOKIE.dmac_address);
-
-	/* now program the PNC register, which enables the state machine */
-	ATRACE_32("apc_program_play() next count", (uint_t)(rc * precision));
-	ddi_put32(handle, &APC_DMAPNC, (uint_t)(rc * precision));
-
-	/* now get ready for the next time we need a DMA buffer */
-	AUDIO_TOGGLE(state->cs_pbuf_toggle);
-	ATRACE_32("apc_program_play() new toggle", state->cs_pbuf_toggle);
-
-	ATRACE_32("apc_program_play() returning", rc);
-
-	return (rc);
-
-}	/* apc_program_play() */
diff --git a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_eb2dma.c b/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_eb2dma.c
deleted file mode 100644
index 9349d1f45a..0000000000
--- a/usr/src/uts/sun/io/audio/sada/drv/audiocs/audio_4231_eb2dma.c
+++ /dev/null
@@ -1,1459 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2004 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-/*
- * Platform specifc code for the EB2 DMA controller. The EB2 is a PCI bus
- * IC that includes play and record DMA engines and an interface for
- * the CS4231.
- */
-
-#include <sys/systm.h>
-#include <sys/ddi.h>
-#include <sys/sunddi.h>
-#include <sys/note.h>
-#include <sys/audio.h>
-#include <sys/audiovar.h>
-#include <sys/audio/audio_support.h>
-#include <sys/audio/audio_src.h>
-#include <sys/audio/audio_trace.h>
-#include <sys/mixer.h>
-#include <sys/audio/audio_mixer.h>
-#include <sys/audio/am_src1.h>
-#include <sys/audio/impl/audio_4231_impl.h>
-#include <sys/audio/audio_4231.h>
-
-/*
- * Attribute structure for the APC, used to create DMA handles.
- */
-static ddi_dma_attr_t eb2_attr = {
-	DMA_ATTR_V0,			/* version */
-	0x0000000000000000LL,		/* dlim_addr_lo */
-	0x00000000ffffffffLL,		/* dlim_addr_hi */
-	0x0000000000ffffffLL,		/* DMA counter register */
-	0x0000000000000001LL,		/* DMA address alignment */
-	0x00000074,			/* 4 and 16 byte burst sizes */
-	0x00000001,			/* min effective DMA size */
-	0x000000000000ffffLL,		/* maximum transfer size, 8k */
-	0x000000000000ffffLL,		/* segment boundary, 32k */
-	0x00000001,			/* s/g list length, no s/g */
-	0x00000001,			/* granularity of device, don't care */
-	0				/* DMA flags */
-};
-
-static ddi_device_acc_attr_t acc_attr = {
-	DDI_DEVICE_ATTR_V0,
-	DDI_STRUCTURE_BE_ACC,
-	DDI_STRICTORDER_ACC
-};
-
-/*
- * Local routines
- */
-static uint_t eb2_rec_intr(caddr_t);
-static uint_t eb2_play_intr(caddr_t);
-static int eb2_program_play(CS_state_t *, int);
-
-/*
- * DMA ops vector functions
- */
-static int eb2_map_regs(dev_info_t *, CS_state_t *, size_t, size_t);
-static void eb2_unmap_regs(CS_state_t *);
-static void eb2_reset(CS_state_t *);
-static int eb2_add_intr(CS_state_t *);
-static void eb2_rem_intr(dev_info_t *);
-static int eb2_p_start(CS_state_t *);
-static void eb2_p_pause(CS_state_t *);
-static void eb2_p_restart(CS_state_t *);
-static void eb2_p_stop(CS_state_t *);
-static int eb2_r_start(CS_state_t *);
-static void eb2_r_stop(CS_state_t *);
-static void eb2_power(CS_state_t *, int);
-
-cs4231_dma_ops_t cs4231_eb2dma_ops = {
-	"EB2 DMA controller",
-	eb2_map_regs,
-	eb2_unmap_regs,
-	eb2_reset,
-	eb2_add_intr,
-	eb2_rem_intr,
-	eb2_p_start,
-	eb2_p_pause,
-	eb2_p_restart,
-	eb2_p_stop,
-	eb2_r_start,
-	eb2_r_stop,
-	eb2_power
-};
-
-/* File name for the cs4231_put8() and cs4231_reg_select() routines */
-static char *thisfile = __FILE__;
-
-/*
- * eb2_map_regs()
- *
- * Description:
- *	This routine allocates the DMA handles and the memory for the
- *	DMA engines to use. It then binds each of the buffers to its
- *	respective handle, getting a DMA cookie. Finally, the registers
- *	are mapped in.
- *
- *	NOTE: All of the ddi_dma_... routines sleep if they cannot get
- *		memory. This means these calls will almost always succeed.
- *
- * Arguments:
- *	dev_info_t	*dip		Pointer to the device's devinfo
- *	CS_state_t	*state		The device's state
- *	int		pbuf_size	The size of the play DMA buffers to
- *					allocate, we need two
- *	int		cbuf_size	The size of the capture DMA buffers to
- *					allocate, we need two
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-eb2_map_regs(dev_info_t *dip, CS_state_t *state, size_t pbuf_size,
-	size_t cbuf_size)
-{
-	uint_t		dma_cookie_count;
-	int		rc;
-
-	ATRACE("in eb2_map_regs()", state);
-
-	/* allocate two handles for play and two for record */
-	if ((rc = ddi_dma_alloc_handle(dip, &eb2_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ph[0])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_alloc_handle(0) failed: %d", rc);
-		goto error;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &eb2_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ph[1])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_alloc_handle(1) failed: %d", rc);
-		goto error_ph0;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &eb2_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ch[0])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_alloc_handle(2) failed: %d", rc);
-		goto error_ph1;
-	}
-	if ((rc = ddi_dma_alloc_handle(dip, &eb2_attr, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_ch[1])) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_alloc_handle(3) failed: %d", rc);
-		goto error_rh0;
-	}
-
-	/* allocate the four DMA buffers, two for play and two for record */
-	if (ddi_dma_mem_alloc(state->cs_ph[0], pbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_pb[0], &state->cs_pml[0],
-	    &state->cs_pmh[0]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_mem_alloc(0) failed");
-		goto error_rh1;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ph[1], pbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_pb[1], &state->cs_pml[1],
-	    &state->cs_pmh[1]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_mem_alloc(1) failed");
-		goto error_pb0;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ch[0], cbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_cb[0], &state->cs_cml[0],
-	    &state->cs_cmh[0]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_mem_alloc(2) failed");
-		goto error_pb1;
-	}
-	if (ddi_dma_mem_alloc(state->cs_ch[1], cbuf_size, &acc_attr,
-	    DDI_DMA_CONSISTENT, DDI_DMA_SLEEP, NULL,
-	    &state->cs_cb[1], &state->cs_cml[1],
-	    &state->cs_cmh[1]) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_mem_alloc(3) failed");
-		goto error_rb0;
-	}
-
-	/* bind each of the buffers to a DMA handle */
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ph[0], (struct as *)0,
-	    state->cs_pb[0], pbuf_size, DDI_DMA_WRITE, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_pc[0], &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_addr_bind_handle(0) failed: %d",
-		    rc);
-		goto error_rb1;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ph[1], (struct as *)0,
-	    state->cs_pb[1], pbuf_size, DDI_DMA_WRITE, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_pc[1], &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_addr_bind_handle(1) failed: %d",
-		    rc);
-		goto error_pc0;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ch[0], (struct as *)0,
-	    state->cs_cb[0], cbuf_size, DDI_DMA_READ, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_cc[0], &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_addr_bind_handle(2) failed: %d",
-		    rc);
-		goto error_pc1;
-	}
-	ASSERT(dma_cookie_count == 1);
-	if ((rc = ddi_dma_addr_bind_handle(state->cs_ch[1], (struct as *)0,
-	    state->cs_cb[1], cbuf_size, DDI_DMA_READ, DDI_DMA_SLEEP,
-	    (caddr_t)0, &state->cs_cc[1], &dma_cookie_count)) !=
-	    DDI_DMA_MAPPED) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_dma_addr_bind_handle(3) failed: %d",
-		    rc);
-		goto error_rc0;
-	}
-	ASSERT(dma_cookie_count == 1);
-
-	/* now, map the codec */
-	if (ddi_regs_map_setup(dip, 0, (caddr_t *)&state->cs_regs, 0,
-	    sizeof (cs4231_pioregs_t), &acc_attr,
-	    &EB2_CODEC_HNDL) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_regs_map_setup() codec failed: %d",
-		    rc);
-		goto error_rc1;
-	}
-
-	/* next the play registers */
-	acc_attr.devacc_attr_endian_flags = DDI_STRUCTURE_LE_ACC;
-	if (ddi_regs_map_setup(dip, 1,
-	    (caddr_t *)&state->cs_eb2_regs.play, 0,
-	    sizeof (cs4231_eb2regs_t), &acc_attr,
-		&EB2_PLAY_HNDL) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_regs_map_setup() play registers "
-		    "failed: %d", rc);
-		goto error_dev1;
-	}
-
-	ASSERT(acc_attr.devacc_attr_endian_flags == DDI_STRUCTURE_LE_ACC);
-
-	/* now the capture registers */
-	if (ddi_regs_map_setup(dip, 2,
-	    (caddr_t *)&state->cs_eb2_regs.record, 0,
-	    sizeof (cs4231_eb2regs_t), &acc_attr,
-	    &EB2_REC_HNDL) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_regs_map_setup() record registers "
-		    "failed: %d", rc);
-		goto error_dev2;
-	}
-
-	/* finally the auxio register */
-	if (ddi_regs_map_setup(dip, 3,
-	    (caddr_t *)&state->cs_eb2_regs.auxio, 0,
-	    sizeof (uint_t), &acc_attr,
-	    &EB2_AUXIO_HNDL) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_map_regs() ddi_regs_map_setup() audio register "
-		    "failed: %d", rc);
-		goto error_dev2;
-	}
-
-	/* disable play and record interrupts */
-	ddi_put32(EB2_PLAY_HNDL, &EB2_PLAY_CSR, EB2_PCLEAR_RESET_VALUE);
-	ddi_put32(EB2_REC_HNDL, &EB2_PLAY_CSR, EB2_RCLEAR_RESET_VALUE);
-
-	/* let the state structure know about the attributes */
-	state->cs_dma_attr = &eb2_attr;
-
-	/* we start with play and record buffers 0 */
-	state->cs_pbuf_toggle = 0;
-	state->cs_cbuf_toggle = 0;
-
-	ATRACE("eb2_map_regs() returning success", state);
-
-	return (AUDIO_SUCCESS);
-
-error_dev2:
-	ddi_regs_map_free(&EB2_REC_HNDL);
-error_dev1:
-	ddi_regs_map_free(&EB2_PLAY_HNDL);
-error_dev0:
-	ddi_regs_map_free(&EB2_CODEC_HNDL);
-error_rc1:
-	(void) ddi_dma_unbind_handle(state->cs_ch[1]);
-error_rc0:
-	(void) ddi_dma_unbind_handle(state->cs_ch[0]);
-error_pc1:
-	(void) ddi_dma_unbind_handle(state->cs_ph[1]);
-error_pc0:
-	(void) ddi_dma_unbind_handle(state->cs_ph[0]);
-error_rb1:
-	ddi_dma_mem_free(&state->cs_cmh[1]);
-error_rb0:
-	ddi_dma_mem_free(&state->cs_cmh[0]);
-error_pb1:
-	ddi_dma_mem_free(&state->cs_pmh[1]);
-error_pb0:
-	ddi_dma_mem_free(&state->cs_pmh[0]);
-error_rh1:
-	ddi_dma_free_handle(&state->cs_ch[1]);
-error_rh0:
-	ddi_dma_free_handle(&state->cs_ch[0]);
-error_ph1:
-	ddi_dma_free_handle(&state->cs_ph[1]);
-error_ph0:
-	ddi_dma_free_handle(&state->cs_ph[0]);
-error:
-	ATRACE("eb2_map_regs() returning failure", state);
-
-	return (AUDIO_FAILURE);
-
-}	/* eb2_map_regs() */
-
-/*
- * eb2_unmap_regs()
- *
- * Description:
- *	This routine unbinds the DMA cookies, frees the DMA buffers,
- *	deallocated the DAM handles, and finally unmaps the Codec's and
- *	DMA engine's registers.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state
- *
- * Returns:
- *	void
- */
-static void
-eb2_unmap_regs(CS_state_t *state)
-{
-	ATRACE("in eb2_unmap_regs()", state);
-
-	(void) ddi_dma_unbind_handle(state->cs_ph[0]);
-	(void) ddi_dma_unbind_handle(state->cs_ph[1]);
-	(void) ddi_dma_unbind_handle(state->cs_ch[0]);
-	(void) ddi_dma_unbind_handle(state->cs_ch[1]);
-
-	ddi_dma_mem_free(&state->cs_pmh[0]);
-	ddi_dma_mem_free(&state->cs_pmh[1]);
-	ddi_dma_mem_free(&state->cs_cmh[0]);
-	ddi_dma_mem_free(&state->cs_cmh[1]);
-
-	ddi_dma_free_handle(&state->cs_ph[0]);
-	ddi_dma_free_handle(&state->cs_ph[1]);
-	ddi_dma_free_handle(&state->cs_ch[0]);
-	ddi_dma_free_handle(&state->cs_ch[1]);
-
-	ddi_regs_map_free(&EB2_CODEC_HNDL);
-	ddi_regs_map_free(&EB2_PLAY_HNDL);
-	ddi_regs_map_free(&EB2_REC_HNDL);
-	ddi_regs_map_free(&EB2_AUXIO_HNDL);
-
-	ATRACE("eb2_unmap_regs() returning", state);
-
-}	/* eb2_unmap_regs() */
-
-/*
- * eb2_reset()
- *
- * Description:
- *	Reset both the play and record DMA engines. The engines are left
- *	with interrupts and the DMA engine disabled.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo structure
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-eb2_reset(CS_state_t *state)
-{
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-	ddi_acc_handle_t	rhandle = EB2_REC_HNDL;
-	uint_t			reg;
-	int			x;
-
-	ATRACE("in eb2_reset()", state);
-
-	/* start with the play side */
-	ddi_put32(phandle, &EB2_PLAY_CSR, EB2_RESET);
-	/* wait for play data to drain */
-	reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-	for (x = 0; !(reg & EB2_DRAIN) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on the bus */
-		reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-	}
-	/* clear the reset bit and program for chaining */
-	ddi_put32(phandle, &EB2_PLAY_CSR, EB2_PCLEAR_RESET_VALUE);
-
-	/* now do the record side and program for chaining */
-	ddi_put32(rhandle, &EB2_REC_CSR, EB2_RESET);
-	/* wait for record data to drain */
-	reg = ddi_get32(rhandle, &EB2_REC_CSR);
-	for (x = 0; !(reg & EB2_DRAIN) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on the bus */
-		reg = ddi_get32(rhandle, &EB2_REC_CSR);
-	}
-	/* clear the reset bit */
-	ddi_put32(rhandle, &EB2_REC_CSR, EB2_RCLEAR_RESET_VALUE);
-
-	ATRACE("eb2_reset() returning", state);
-
-}	/* eb2_reset() */
-
-/*
- * eb2_add_intr()
- *
- * Description:
- *	Register the EB2 interrupts with the kernel.
- *
- *	NOTE: This does NOT turn on interrupts.
- *
- *	CAUTION: While the interrupts are added, the Codec interrupts are
- *		not enabled.
- *
- * Arguments:
- *	CS_state_t	*state	Pointer to the device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Registers successfully mapped
- *	AUDIO_FAILURE		Registers not successfully mapped
- */
-static int
-eb2_add_intr(CS_state_t *state)
-{
-	dev_info_t	*dip = state->cs_dip;
-
-	ATRACE("in eb2_add_intr()", state);
-
-	/* first we make sure these aren't high level interrupts */
-	if (ddi_intr_hilevel(dip, 0) != 0) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_add_intr() unsupported high level interrupt 0");
-		return (AUDIO_FAILURE);
-	}
-	if (ddi_intr_hilevel(dip, 1) != 0) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_add_intr() unsupported high level interrupt 1");
-		return (AUDIO_FAILURE);
-	}
-
-	/* okay to register the interrupts */
-	if (ddi_add_intr(dip, 0, (ddi_iblock_cookie_t *)0,
-	    (ddi_idevice_cookie_t *)0, eb2_rec_intr,
-	    (caddr_t)state) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_add_intr() bad record interrupt specification");
-		return (AUDIO_FAILURE);
-	}
-
-	if (ddi_add_intr(dip, 1, (ddi_iblock_cookie_t *)0,
-	    (ddi_idevice_cookie_t *)0, eb2_play_intr,
-	    (caddr_t)state) != DDI_SUCCESS) {
-		audio_sup_log(state->cs_ahandle, CE_WARN,
-		    "!eb2_add_intr() bad play interrupt specification");
-		ddi_remove_intr(dip, 0, NULL);
-		return (AUDIO_FAILURE);
-	}
-
-	ATRACE("eb2_add_intr() returning successful", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* eb2_add_intr() */
-
-/*
- * eb2_rem_intr()
- *
- * Description:
- *	Unregister the EB2 interrupts from the kernel.
- *
- *	CAUTION: While the interrupts are removed, the Codec interrupts are
- *		not disabled, but then, they never should have been on in
- *		the first place.
- *
- * Arguments:
- *	dev_info_t	*dip	Pointer to the device's devinfo
- *
- * Returns:
- *	void
- */
-static void
-eb2_rem_intr(dev_info_t *dip)
-{
-	ATRACE("in eb2_rem_intr()", dip);
-
-	ddi_remove_intr(dip, 0, NULL);
-	ddi_remove_intr(dip, 1, NULL);
-
-	ATRACE("eb2_rem_intr() returning", dip);
-
-}	/* eb2_rem_intr() */
-
-/*
- * eb2_p_start()
- *
- * Description:
- *	This routine starts the play DMA engine. This includes "hard" starts
- *	where the DMA engine's registers need to be loaded as well as starting
- *	after a pause;
- *
- *	For hard starts the DMA engine is started by programming the Play Next
- *	Byte Register and the Play Next Address Register twice, and finally
- *	enabling the DMA play engine.
- *
- *	Starting after a pause is much eaiser, we just turn on the EN_DMA bit.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- *	CAUTION: This routine doesn't start the Codec because the first
- *		interrupt causes a recursive mutex_enter.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		Successfully started
- *	AUDIO_FAILURE		Start failed
- */
-static int
-eb2_p_start(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = EB2_CODEC_HNDL;
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-	int			i;
-	int			samples;
-
-	ATRACE("in eb2_p_start()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* figure out the number of samples we want */
-	samples = state->cs_play_sr * state->cs_play_ch /
-	    state->cs_ad_info.ad_play.ad_int_rate;
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples % state->cs_play_ch) == 1) {
-		ATRACE("eb2_p_start() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-	ATRACE("eb2_p_start() samples per interrupt", samples);
-
-	/* try to load 2 buffers worth to get things started */
-	for (i = 0; i < 2; i++) {
-		ATRACE_32("eb2_p_start() calling eb2_program_play()", samples);
-		PLAY_COUNT = samples *
-		    (state->cs_play_prec >> AUDIO_PRECISION_SHIFT);
-		if (eb2_program_play(state, samples) <= 0) {
-			ATRACE("eb2_p_start() "
-			    "eb2_program_play() returns no samples", state);
-			break;
-		}
-
-		if (i == 0) {
-			/* start playing before we load the next address */
-			ATRACE("eb2_p_start() turn on play DMA engine", state);
-			OR_SET_WORD(phandle, &EB2_PLAY_CSR, EB2_PLAY_ENABLE);
-		}
-	}
-
-	/*
-	 * If there wasn't anything to get then we aren't active. This can
-	 * happen if an AUDIO_DRAIN was issued with no audio in the stream.
-	 */
-	if (i == 0) {
-		ATRACE("eb2_p_start() nothing to do", state);
-		return (AUDIO_FAILURE);
-	}
-
-	/*
-	 * Even with just one DMA buffer loaded we can still play some audio.
-	 * Read the Codec Status Register so that we clear the Error and
-	 * Initialization register, Index 11, so we'll know when we get
-	 * a play underrun in the ISR. Then enable the play DMA engine,
-	 * including interrupts.
-	 */
-#ifdef DEBUG
-	ATRACE_8("eb2_p_start() Read the Codec status reg, clearing Index 11",
-	    ddi_get8(handle, &CS4231_STATUS));
-#else
-	(void) ddi_get8(handle, &CS4231_STATUS); /* we only need a read */
-#endif
-
-	state->cs_flags |= PDMA_ENGINE_INITIALIZED;
-
-	ATRACE("eb2_p_start() returning", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* eb2_p_start() */
-
-/*
- * eb2_pause()
- *
- * Description:
- *	This routine pauses the play DMA engine. Buffers, FIFO, etc. are NOT
- *	lost.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-eb2_p_pause(CS_state_t *state)
-{
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in eb2_p_pause()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* clear the EN_DMA bit, pausing the engine */
-	AND_SET_WORD(phandle, &EB2_PLAY_CSR, ~EB2_EN_DMA);
-
-	/* we wait for the Codec FIFO to underrun */
-	cs4231_reg_select(state->cs_ahandle, phandle, &CS4231_IAR, ESI_REG,
-	    __LINE__, thisfile);
-	creg = ddi_get8(phandle, &CS4231_IDR);
-	for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		creg = ddi_get8(phandle, &CS4231_IDR);
-	}
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, phandle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(phandle, &CS4231_IDR, ~INTC_PEN, INTC_VALID_MASK);
-
-	ATRACE("eb2_p_pause() returning", state);
-
-}	/* eb2_p_pause() */
-
-/*
- * eb2_p_restart()
- *
- * Description:
- *	This routine restarts the play DMA engine after pausing. Buffers,
- *	FIFO, etc. MUST be programmed and valid.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-eb2_p_restart(CS_state_t *state)
-{
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-
-	ATRACE("in eb2_p_restart()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* start the Codec */
-	cs4231_reg_select(state->cs_ahandle, phandle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	OR_SET_BYTE(phandle, &CS4231_IDR, INTC_PEN, INTC_VALID_MASK);
-
-	/* set the EN_DMA bit, restarting the engine */
-	OR_SET_WORD(phandle, &EB2_PLAY_CSR, EB2_EN_DMA);
-
-	ATRACE("eb2_p_restart() returning", state);
-
-}	/* eb2_p_restart() */
-
-/*
- * eb2_p_stop()
- *
- * Description:
- *	This routine stops the play DMA engine.
- *
- *	The DMA engine is stopped by using the play RESET bit.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-eb2_p_stop(CS_state_t *state)
-{
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-	uint_t			reg;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in eb2_p_stop()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* clear the play interrupts so the ISR doesn't get involved */
-	AND_SET_WORD(phandle, &EB2_PLAY_CSR, ~EB2_PINTR_MASK);
-
-	/* set the play RESET bit to stop playing audio */
-	OR_SET_WORD(phandle, &EB2_PLAY_CSR, EB2_RESET);
-
-	/* wait for the FIFO to drain */
-	reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-	for (x = 0; (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on the bus */
-		reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-	}
-#ifdef DEBUG
-	if (x >= CS4231_TIMEOUT) {
-		ATRACE("eb2_p_stop() timeout", state);
-	}
-#endif
-
-	/* now clear the RESET and EN_DMA bits */
-	AND_SET_WORD(phandle, &EB2_PLAY_CSR, ~(EB2_RESET|EB2_EN_DMA));
-
-	/* we wait for the Codec FIFO to underrun */
-	cs4231_reg_select(state->cs_ahandle, phandle, &CS4231_IAR, ESI_REG,
-	    __LINE__, thisfile);
-	creg = ddi_get8(phandle, &CS4231_IDR);
-	for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* don't beat on bus */
-		creg = ddi_get8(phandle, &CS4231_IDR);
-	}
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, phandle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(phandle, &CS4231_IDR, ~INTC_PEN, INTC_VALID_MASK);
-
-	state->cs_flags &= ~PDMA_ENGINE_INITIALIZED;
-
-	ATRACE("eb2_p_stop() returning", state);
-
-}	/* eb2_p_stop() */
-
-/*
- * eb2_r_start()
- *
- * Description:
- *	This routine starts the record DMA engine. The DMA engine is never
- *	paused for record, so a puse is equivalent to a stop. Thus all starts
- *	are hard starts.
- *
- *	For hard starts the DMA engine is started by programming the Record
- *	NExt Byte Register and then the Record Next Address Register twice,
- *	and finally enabling the record DMA engine.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- *	CAUTION: This routine doesn't start the Codec because the first
- *		interrupt causes a recursive mutex_enter.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	AUDIO_SUCCESS		The capture DMA was started
- *	AUDIO_FAILURE		The capture DMA was not started
- */
-static int
-eb2_r_start(CS_state_t *state)
-{
-	ddi_acc_handle_t	rhandle = EB2_REC_HNDL;
-	uint_t			csr;
-	int			bytes;
-	int			i;
-	int			samples;
-	int			x;
-
-	ATRACE("in eb2_r_start()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* figure out the number of samples to capture */
-	samples = state->cs_record_sr * state->cs_record_ch /
-	    state->cs_ad_info.ad_record.ad_int_rate;
-	/* if stereo & sr = 11025 & ints = 50 then 441 samples, bad! - so fix */
-	if ((samples % state->cs_record_ch) == 1) {
-		ATRACE("eb2_r_start() samples not mod", samples);
-		/* need to adjust */
-		samples++;
-	}
-	ATRACE("eb2_r_start() samples per interrupt", samples);
-
-	/* now convert the number of samples to the "size" in bytes */
-	ASSERT(state->cs_record_prec == AUDIO_PRECISION_8 ||
-	    state->cs_record_prec == AUDIO_PRECISION_16);
-	bytes = samples * (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-	ATRACE("eb2_r_start() DMA count", bytes);
-
-	/* reset the DMA engine so we have a good starting place */
-	OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_RESET);
-
-	/* wait for the FIFO to drain, it should be empty */
-	csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	for (x = 0; (csr & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);	/* no reason to beat on the bus */
-		csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	}
-	if (x >= CS4231_TIMEOUT) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_r_start() timeout waiting for Codec, record not "
-		    "started!");
-		return (AUDIO_FAILURE);
-	}
-
-	/* now clear the RESET and EN_DMA bits */
-	AND_SET_WORD(rhandle, &EB2_REC_CSR, ~(EB2_RESET|EB2_EN_DMA));
-
-	/* put into chaining mode */
-	OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_RCLEAR_RESET_VALUE);
-
-	/*
-	 * Program the DMA engine with both buffers. We MUST do both buffers
-	 * otherwise CAP_COUNT isn't going to set both byte counts.
-	 */
-	for (i = 0; i < 2; i++) {
-		/* sync the DMA buffer before it is going to be used */
-		ATRACE("eb2_r_start() dma buffer sync", state);
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!eb2_r_start() ddi_dma_sync() failed, recording "
-			    "stopped");
-
-			/* reset the DMA engine */
-			OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_RESET);
-
-			/* wait for the FIFO to drain, it should be empty */
-			csr = ddi_get32(rhandle, &EB2_REC_CSR);
-			for (x = 0;
-			    (csr & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				csr = ddi_get32(rhandle, &EB2_REC_CSR);
-			}
-
-			/* now clear the RESET and EN_DMA bits */
-			AND_SET_WORD(rhandle, &EB2_REC_CSR,
-			    ~(EB2_RESET|EB2_EN_DMA));
-
-			return (AUDIO_FAILURE);
-		}
-
-		/* for eb2 we first program the Next Byte Count Register */
-		ATRACE_32("eb2_r_start() next count", (uint_t)bytes);
-		ddi_put32(rhandle, &EB2_REC_BCR, (uint_t)bytes);
-		CAP_COUNT = bytes;
-		ATRACE_32("eb2_r_start() samples to send", samples);
-
-		/* now program the Next Address Register - starts state mach */
-		ATRACE_32("eb2_r_start() next address",
-		    (uint_t)CAP_DMA_COOKIE.dmac_address);
-		ddi_put32(rhandle, &EB2_REC_ACR,
-		    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-		if (i == 0) {
-			/* start the DMA engine before loading the next addr */
-			ATRACE("eb2_r_start() turn on rec DMA engine", state);
-			OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_REC_ENABLE);
-		}
-
-		/* get ready for the next DMA buffer */
-		AUDIO_TOGGLE(state->cs_cbuf_toggle);
-		ATRACE_32("eb2_r_start() new toggle", state->cs_cbuf_toggle);
-	}
-
-	state->cs_flags |= RDMA_ENGINE_INITIALIZED;
-
-	ATRACE("eb2_r_start() returning", state);
-
-	return (AUDIO_SUCCESS);
-
-}	/* eb2_r_start() */
-
-/*
- * eb2_r_stop()
- *
- * Description:
- *	This routine stops the record DMA engine. It then sends any collected
- *	data to the audio mixer.
- *
- *	NOTE: The state structure must be locked before this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state structure
- *
- * Returns:
- *	void
- */
-static void
-eb2_r_stop(CS_state_t *state)
-{
-	ddi_acc_handle_t	handle = EB2_CODEC_HNDL;
-	ddi_acc_handle_t	rhandle = EB2_REC_HNDL;
-	uint_t			reg;
-	uint_t			csr;
-	int			samples;
-	int			x;
-
-	ATRACE("in eb2_r_stop()", state);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	/* stop the Codec */
-	cs4231_reg_select(state->cs_ahandle, handle, &CS4231_IAR, INTC_REG,
-	    __LINE__, thisfile);
-	AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_CEN, INTC_VALID_MASK);
-
-	csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	for (x = 0; (csr & EB2_CYC_PENDING) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);
-		csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	}
-
-	csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	if (csr & EB2_INT_PEND) {
-		state->cs_flags |= REC_INTR_PENDING;
-	}
-
-	/* set the record RESET bit to stop recording audio */
-	OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_RESET);
-
-	/* wait for the FIFO to drain */
-	reg = ddi_get32(rhandle, &EB2_REC_CSR);
-	for (x = 0; (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-		drv_usecwait(1);		/* don't beat on the bus */
-		reg = ddi_get32(rhandle, &EB2_REC_CSR);
-	}
-
-#ifdef DEBUG
-	if (x >= CS4231_TIMEOUT) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "audiocs: eb2_r_stop() timeout, record buffer flushed");
-	}
-#endif
-
-	/* clear the RESET and EN_DMA bits */
-	AND_SET_WORD(rhandle, &EB2_REC_CSR, ~(EB2_RESET|EB2_EN_DMA));
-
-	/* figure how many samples were recorded */
-	samples = (CAP_COUNT - ddi_get32(rhandle, &EB2_REC_BCR)) /
-	    (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-
-	samples -= samples % state->cs_record_ch;
-
-	/* send the captured audio to the mixer */
-	if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0, DDI_DMA_SYNC_FORCPU) ==
-	    DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_r_stop() ddi_dma_sync() failed, recorded audio lost");
-	} else {
-		mutex_exit(&state->cs_lock);
-		am_send_audio(state->cs_ahandle, CAP_DMA_BUF,
-		    AUDIO_NO_CHANNEL, samples);
-		mutex_enter(&state->cs_lock);
-	}
-
-	state->cs_flags &= ~RDMA_ENGINE_INITIALIZED;
-
-	ATRACE("eb2_r_stop() returning", state);
-
-}	/* eb2_r_stop() */
-
-/*
- * eb2_power()
- *
- * Description:
- *	This routine turns the Codec on or off using the auxio register
- *	in the eb2 device (cheerio or rio). Fortunately we don't need
- *	to delay like we do with the APC.
- *
- *	NOTE: The state structure must be locked when this routine is called.
- *
- * Arguments:
- *	CS_state_t	*state		Ptr to the device's state structure
- *	int		level		Power level to set
- *
- * Returns:
- *	void
- */
-static void
-eb2_power(CS_state_t *state, int level)
-{
-	ddi_acc_handle_t	xhandle = EB2_AUXIO_HNDL;
-
-	ATRACE("in eb2_power()", state);
-	ATRACE_32("eb2_power() level", level);
-
-	ASSERT(mutex_owned(&state->cs_lock));
-
-	if (level == CS4231_PWR_ON) {	/* turn power on */
-		AND_SET_WORD(xhandle, EB2_AUXIO_REG, ~EB2_AUXIO_COD_PDWN);
-		state->cs_powered = CS4231_PWR_ON;
-	} else {	/* turn power off */
-		OR_SET_WORD(xhandle, EB2_AUXIO_REG, EB2_AUXIO_COD_PDWN);
-		state->cs_powered = CS4231_PWR_OFF;
-	}
-
-	ATRACE_32("eb2_power() done", ddi_get32(xhandle, EB2_AUXIO_REG));
-
-}	/* eb2_power() */
-
-
-/* *******  Local Routines ************************************************** */
-
-/*
- * eb2_rec_intr()
- *
- * Description:
- *	EB2 record interrupt serivce routine. First we find out why there was
- *	an interrupt, then we take the appropriate action
- *
- * Arguments:
- *	caddr_t		T	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-eb2_rec_intr(caddr_t T)
-{
-	CS_state_t		*state = (CS_state_t *)T;
-	ddi_acc_handle_t	rhandle = EB2_REC_HNDL;
-	uint_t			csr;
-	uint_t			reg;
-	int			rc = DDI_INTR_UNCLAIMED;
-	int			samples;
-	int			x;
-
-	ATRACE("in eb2_rec_intr()", state);
-
-	/* the state must be protected */
-	mutex_enter(&state->cs_lock);
-
-	/* get the EB2 record CSR */
-	csr = ddi_get32(rhandle, &EB2_REC_CSR);
-	ATRACE_32("eb2_rec_intr() intrrupt CSR", csr);
-
-	/* make sure this device sent the interrupt */
-	if (!(csr & EB2_INT_PEND)) {
-		/* Interrupt that's still being serviced */
-		if (state->cs_flags & REC_INTR_PENDING) {
-			rc = DDI_INTR_CLAIMED;
-			goto done;
-		}
-
-		/* nope, this isn't our interrupt */
-		ATRACE_32("eb2_rec_intr() device didn't send interrupt", csr);
-		if (state->cs_ksp) {
-			KIOP(state)->intrs[KSTAT_INTR_SPURIOUS]++;
-		}
-		mutex_exit(&state->cs_lock);
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	/* clear all interrupts we captured at this time */
-	ddi_put32(rhandle, &EB2_REC_CSR, (csr|EB2_TC));
-	ATRACE_32("eb2_rec_intr() csr after clear",
-	    ddi_get32(rhandle, &EB2_REC_CSR));
-
-	if (csr & EB2_TC) {
-		ATRACE("eb2_rec_intr() record interrupt", state);
-
-		/* sync DMA memory before sending it to the audio mixer */
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORCPU) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!eb2_rec_intr() ddi_dma_sync(#1) failed, recorded "
-			    "audio lost");
-		} else {
-			/* figure how many samples were recorded */
-			samples = CAP_COUNT /
-			    (state->cs_record_prec >> AUDIO_PRECISION_SHIFT);
-
-			/* send the captured audio to the mixer */
-			mutex_exit(&state->cs_lock);
-			am_send_audio(state->cs_ahandle, CAP_DMA_BUF,
-			    AUDIO_NO_CHANNEL, samples);
-			mutex_enter(&state->cs_lock);
-		}
-
-		/* sync the DMA buffer before it is going to be reused */
-		ATRACE("eb2_rec_intr() dma buffer sync", state);
-		if (ddi_dma_sync(CAP_DMA_HANDLE, (off_t)0, 0,
-		    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-			audio_sup_log(state->cs_ahandle, CE_NOTE,
-			    "!eb2_rec_intr() ddi_dma_sync(#2) failed, recording"
-			    " disabled");
-
-			/* reset the DMA engine */
-			OR_SET_WORD(rhandle, &EB2_REC_CSR, EB2_RESET);
-
-			/* wait for the FIFO to drain, it should be empty */
-			reg = ddi_get32(rhandle, &EB2_REC_CSR);
-			for (x = 0;
-			    (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				reg = ddi_get32(rhandle, &EB2_REC_CSR);
-			}
-
-			/* now clear the RESET and EN_DMA bits */
-			AND_SET_WORD(rhandle, &EB2_REC_CSR,
-			    ~(EB2_RESET|EB2_EN_DMA));
-		} else {
-			ATRACE_32("eb2_rec_intr() next address",
-			    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-			/* first program the Next Byte Count Register */
-			ATRACE_32("eb2_rec_intr() next count",
-			    (uint_t)CAP_COUNT);
-			ddi_put32(rhandle, &EB2_REC_BCR, (uint_t)CAP_COUNT);
-
-			/* now program the Next Add. Reg. - starts state mach */
-			ATRACE_32("eb2_rec_intr() next address",
-			    (uint_t)CAP_DMA_COOKIE.dmac_address);
-			ddi_put32(rhandle, &EB2_REC_ACR,
-			    (uint_t)CAP_DMA_COOKIE.dmac_address);
-
-			/* get ready for the next DMA buffer */
-			AUDIO_TOGGLE(state->cs_cbuf_toggle);
-			ATRACE_32("eb2_rec_intr() new toggle",
-			    state->cs_cbuf_toggle);
-		}
-
-		/* we always claim the interrupt, even if DMA sync failed */
-		rc = DDI_INTR_CLAIMED;
-	} else if (csr & EB2_ERR_PEND) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_rec_intr() error interrupt: 0x%x", csr);
-		rc = DDI_INTR_CLAIMED;
-	} else {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_rec_intr() unknown interrupt: 0x%x", csr);
-		rc = DDI_INTR_CLAIMED;
-	}
-
-done:
-	state->cs_flags &= ~REC_INTR_PENDING;
-
-	/* update the kernel interrupt statistics */
-	if (state->cs_ksp) {
-		if (rc == DDI_INTR_CLAIMED) {
-			KIOP(state)->intrs[KSTAT_INTR_HARD]++;
-		} else {
-			KIOP(state)->intrs[KSTAT_INTR_SPURIOUS]++;
-		}
-	}
-
-	mutex_exit(&state->cs_lock);
-
-	ATRACE_32("eb2_rec_intr() returning", rc);
-
-	return (rc);
-
-}	/* eb2_rec_intr() */
-
-/*
- * eb2_play_intr()
- *
- * Description:
- *	EB2 play interrupt serivce routine. First we find out why there was an
- *	interrupt, then we take the appropriate action.
- *
- * Arguments:
- *	caddr_t		T	Pointer to the interrupting device's state
- *				structure
- *
- * Returns:
- *	DDI_INTR_CLAIMED	Interrupt claimed and processed
- *	DDI_INTR_UNCLAIMED	Interrupt not claimed, and thus ignored
- */
-static uint_t
-eb2_play_intr(caddr_t T)
-{
-	CS_state_t		*state = (CS_state_t *)T;
-	ddi_acc_handle_t	handle = EB2_CODEC_HNDL;
-	ddi_acc_handle_t	phandle = EB2_PLAY_HNDL;
-	uint_t			csr;
-	uint_t			reg;
-	int			rc = DDI_INTR_UNCLAIMED;
-	int			samples;
-	int			x;
-	uint8_t			creg;
-
-	ATRACE("in eb2_play_intr()", state);
-
-	/* the state must be protected */
-	mutex_enter(&state->cs_lock);
-
-	/* get the EB2 play CSR */
-	csr = ddi_get32(phandle, &EB2_PLAY_CSR);
-	ATRACE_32("eb2_play_intr() interrupt CSR", csr);
-
-	/* make sure this device sent the interrupt */
-	if (!(csr & EB2_INT_PEND)) {
-		mutex_exit(&state->cs_lock);
-		/* nope, this isn't our interrupt */
-		ATRACE_32("eb2_play_intr() device didn't send interrupt", csr);
-		return (DDI_INTR_UNCLAIMED);
-	}
-
-	/* clear all interrupts we captured at this time */
-	ddi_put32(phandle, &EB2_PLAY_CSR, (csr|EB2_TC));
-	ATRACE_32("eb2_play_intr() csr after clear",
-	    ddi_get32(phandle, &EB2_PLAY_CSR));
-
-	if (csr & EB2_TC) {
-		samples = PLAY_COUNT /
-		    (state->cs_play_prec >> AUDIO_PRECISION_SHIFT);
-		ATRACE_32("eb2_play_intr() samples to get", samples);
-
-		/* try to load the next audio buffer */
-		samples = eb2_program_play(state, samples);
-		ATRACE_32("eb2_play_intr() samples eb2_program_play() returned",
-		    samples);
-
-		if (samples <= 0 && !(csr & EB2_A_LOADED)) {
-			/*
-			 * There isn't any more data to play, so wait for
-			 * the the Codec FIFO to empty. Then turn off the
-			 * play DMA engine by reseting it. Also, we should
-			 * note that a play interrupt with an ADDRESS loaded
-			 * but no samples is ignored, but acknowledged. We
-			 * wait for the ADDRESS to not be loaded before we
-			 * declare the DMA engine empty.
-			 */
-			ATRACE_32("eb2_play_intr() no more data, wait for "
-			    "FIFO", samples);
-
-			/*
-			 * We wait for the Codec FIFO to underrun. Unlike the
-			 * APC, we wouldn't be here if the pipe wasn't already
-			 * empty.
-			 */
-			cs4231_reg_select(state->cs_ahandle, handle,
-			    &CS4231_IAR, ESI_REG, __LINE__, thisfile);
-			creg = ddi_get8(handle, &CS4231_IDR);
-			for (x = 0; !(creg & ESI_PUR) && x < CS4231_TIMEOUT;
-			    x++) {
-				drv_usecwait(1);	/* don't beat on bus */
-				creg = ddi_get8(handle, &CS4231_IDR);
-			}
-
-			/*
-			 * Clear the flag so if audio is restarted while in
-			 * am_play_shutdown() we can detect it and not mess
-			 * things up.
-			 */
-			state->cs_flags &= ~PDMA_ENGINE_INITIALIZED;
-
-			/* now shutdown the play stream */
-			ATRACE("eb2_intr() shutdown play stream", state);
-			mutex_exit(&state->cs_lock);
-			am_play_shutdown(state->cs_ahandle, NULL);
-			mutex_enter(&state->cs_lock);
-
-			/* make sure playing wasn't restarted when lock lost */
-			if (state->cs_flags & PDMA_ENGINE_INITIALIZED) {
-				/* yes, it was, so we're done */
-				ATRACE("eb2_intr() restart after shutdown", 0);
-				rc = DDI_INTR_CLAIMED;
-				goto done;
-			}
-
-			/*
-			 * Nope, play was not restarted so reset the
-			 * DMA eng, putting it into a known state
-			 */
-			OR_SET_WORD(phandle, &EB2_PLAY_CSR, EB2_RESET);
-
-			/* wait for the FIFO to drain, which it should be */
-			reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-			for (x = 0;
-			    (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-				drv_usecwait(1); /* don't beat on the bus */
-				reg = ddi_get32(phandle, &EB2_PLAY_CSR);
-			}
-
-			/* clear the reset bit */
-			AND_SET_WORD(phandle, &EB2_PLAY_CSR, ~EB2_RESET);
-
-			/* disable the Codec */
-			cs4231_reg_select(state->cs_ahandle, handle,
-			    &CS4231_IAR, INTC_REG, __LINE__, thisfile);
-			AND_SET_BYTE(handle, &CS4231_IDR, ~INTC_PEN,
-			    INTC_VALID_MASK);
-
-			/* and reset the status */
-			ddi_put8(handle, &CS4231_STATUS, STATUS_RESET);
-
-			ATRACE("eb2_play_intr() Play DMA engine off", state);
-		}
-		rc = DDI_INTR_CLAIMED;
-		ATRACE_32("eb2_play_intr() done", rc);
-	} else if (csr & EB2_ERR_PEND) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_play_intr() error interrupt: 0x%x", csr);
-		rc = DDI_INTR_CLAIMED;
-	} else {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_play_intr() unknown interrupt: 0x%x", csr);
-		rc = DDI_INTR_CLAIMED;
-	}
-
-done:
-
-	/* update the kernel interrupt statisitcs */
-	if (state->cs_ksp) {
-		if (rc == DDI_INTR_CLAIMED) {
-			KIOP(state)->intrs[KSTAT_INTR_HARD]++;
-		}
-	}
-
-	mutex_exit(&state->cs_lock);
-
-	ATRACE_32("eb2_play_intr() returning", rc);
-
-	return (rc);
-
-}	/* eb2_play_intr() */
-
-/*
- * eb2_program_play()
- *
- * Description:
- *	This routine is used by eb2_p_start() and eb2_play_intr() to program
- *	the play DMA engine with the next buffer full of audio.
- *
- * Arguments:
- *	CS_state_t	*state	The device's state pointer
- *	int		samples	The number of samples to be retrieved from the
- *				mixer
- *
- * Returns:
- *	0			The buffer wasn't programmed, no audio
- *	> 0			The buffer was programmed
- *	AUDIO_FAILURE		The buffer wasn't programmed
- */
-static int
-eb2_program_play(CS_state_t *state, int samples)
-{
-	ddi_acc_handle_t	handle = EB2_CODEC_HNDL;
-	uint_t			reg;
-	uint_t			precision;
-	int			x;
-	int			rc;
-
-	ATRACE_32("in eb2_program_play()", samples);
-
-	/* we need the precision to calculate the next count correctly */
-	precision = state->cs_play_prec >> AUDIO_PRECISION_SHIFT;
-	ATRACE_32("eb2_program_play() precision", precision);
-
-	/* get the first buffer's worth of audio */
-	mutex_exit(&state->cs_lock);
-	rc = am_get_audio(state->cs_ahandle, PLAY_DMA_BUF, AUDIO_NO_CHANNEL,
-	    samples);
-	mutex_enter(&state->cs_lock);
-	ATRACE_32("eb2_program_play() am_get_audio() returned", rc);
-
-	if (rc == AUDIO_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_program_play() am_get_audio() failed");
-		return (AUDIO_FAILURE);
-	} else if (rc == 0) {
-		ATRACE("eb2_program_play() am_get_audio() returned 0 samples",
-		    state);
-		return (0);
-	}
-
-	/* sync the DMA buffer before it is going to be used */
-	ATRACE("eb2_program_play() dma buffer sync", state);
-	if (ddi_dma_sync(PLAY_DMA_HANDLE, (off_t)0, (size_t)(rc * precision),
-	    DDI_DMA_SYNC_FORDEV) == DDI_FAILURE) {
-		audio_sup_log(state->cs_ahandle, CE_NOTE,
-		    "!eb2_program_play() ddi_dma_sync(#2) failed");
-
-		/* reset the DMA engine, this leaves the DMA engine ok */
-		OR_SET_WORD(handle, &EB2_PLAY_CSR, EB2_RESET);
-		/* wait for play data to drain */
-		reg = ddi_get32(handle, &EB2_PLAY_CSR);
-		for (x = 0; (reg & EB2_FIFO_DRAIN) && x < CS4231_TIMEOUT; x++) {
-			drv_usecwait(1);	/* don't beat on the bus */
-			reg = ddi_get32(handle, &EB2_PLAY_CSR);
-		}
-		/* clear the reset bit */
-		ddi_put32(handle, &EB2_PLAY_CSR, EB2_PCLEAR_RESET_VALUE);
-
-		return (AUDIO_FAILURE);
-	}
-
-	/* make sure we are in NEXT mode */
-	OR_SET_WORD(handle, &EB2_PLAY_CSR, EB2_EN_NEXT|EB2_EN_CNT);
-
-	/* program the Next Byte Count Register */
-	ATRACE_32("eb2_program_play() next count", (uint_t)(rc * precision));
-	ddi_put32(handle, &EB2_PLAY_BCR, (uint_t)(rc * precision));
-
-	/* now program the Next Address Register */
-	ATRACE_32("eb2_program_play() next address",
-	    (uint_t)PLAY_DMA_COOKIE.dmac_address);
-	ddi_put32(handle, &EB2_PLAY_ACR, (uint_t)PLAY_DMA_COOKIE.dmac_address);
-
-	/* now get ready for the next time we need a DMA buffer */
-	AUDIO_TOGGLE(state->cs_pbuf_toggle);
-	ATRACE_32("eb2_program_play() new toggle", state->cs_pbuf_toggle);
-
-	ATRACE_32("eb2_program_play() returning", rc);
-
-	return (rc);
-
-}	/* eb2_program_play() */
diff --git a/usr/src/uts/sun/sys/Makefile b/usr/src/uts/sun/sys/Makefile
index 38a2f0bfba..f4672bce99 100644
--- a/usr/src/uts/sun/sys/Makefile
+++ b/usr/src/uts/sun/sys/Makefile
@@ -20,11 +20,9 @@
 #
 
 #
-# Copyright 2008 Sun Microsystems, Inc.  All rights reserved.
+# Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 # Use is subject to license terms.
 #
-#ident	"%Z%%M%	%I%	%E% SMI"
-#
 # include global definitions
 include ../../../Makefile.master
 
@@ -49,10 +47,6 @@ stp4020_reg.h		stp4020_var.h \
 ttymux.h		ttymuxuser.h \
 zsdev.h
 
-AUDIOHDRS= \
-audio_4231.h \
-audio1575.h
-
 SCSIADHDRS= \
 espcmd.h	espreg.h	espvar.h	fascmd.h	fasdma.h \
 fasreg.h	fasvar.h	ifpio.h		sfvar.h
@@ -73,7 +67,6 @@ ROOTDIRS=	$(ROOTDIR) \
 
 ROOTHDRS= $(HDRS:%=$(ROOTDIR)/%)
 $(CLOSED_BUILD)ROOTHDRS += $(CLOSED_HDRS:%=$(ROOTDIR)/%)
-ROOTAUDHDRS= $(AUDIOHDRS:%=$(ROOTDIR)/audio/%)
 ROOTFCHDRS= $(FCHDRS:%=$(ROOTDIR)/fc4/%)
 ROOTSCSIADHDRS= $(SCSIADHDRS:%=$(ROOTDIR)/scsi/adapters/%)
 $(CLOSED_BUILD)ROOTSCSIADHDRS += \
@@ -113,7 +106,6 @@ scsi/targets/%.check:	scsi/targets/%.h
 
 CHECKHDRS= \
 	$(HDRS:%.h=%.check) \
-	$(AUDIOHDRS:%.h=audio/%.check) \
 	$(FCHDRS:%.h=fc4/%.check) \
 	$(SCSIADHDRS:%.h=scsi/adapters/%.check) \
 	$(SCSITARGHDRS:%.h=scsi/targets/%.check)
diff --git a/usr/src/uts/sun/sys/audio/audio1575.h b/usr/src/uts/sun/sys/audio/audio1575.h
deleted file mode 100644
index 0daab8ee76..0000000000
--- a/usr/src/uts/sun/sys/audio/audio1575.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2005 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIO1575_H_
-#define	_SYS_AUDIO1575_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/*
- * Header file for the audio1575 device driver
- */
-
-/*
- * Values returned by the AUDIO_GETDEV ioctl()
- */
-#define	M1575_DEV_NAME			"SUNW,audio1575"
-#define	M1575_DEV_CONFIG		"onboard1"
-#define	M1575_DEV_VERSION		"a"
-
-/*
- * Driver supported configuration information
- */
-#define	M1575_NAME			"audio1575"
-#define	M1575_MOD_NAME			"mixer audio driver"
-#define	M1575_SAMPR5510			(5510)
-#define	M1575_SAMPR6620			(6620)
-#define	M1575_SAMPR8000			(8000)
-#define	M1575_SAMPR9600			(9600)
-#define	M1575_SAMPR11025		(11025)
-#define	M1575_SAMPR16000		(16000)
-#define	M1575_SAMPR18900		(18900)
-#define	M1575_SAMPR22050		(22050)
-#define	M1575_SAMPR27420		(27420)
-#define	M1575_SAMPR32000		(32000)
-#define	M1575_SAMPR33075		(33075)
-#define	M1575_SAMPR37800		(37800)
-#define	M1575_SAMPR44100		(44100)
-#define	M1575_SAMPR48000		(48000)
-
-#define	M1575_DEFAULT_SR		M1575_SAMPR8000
-#define	M1575_DEFAULT_CH		AUDIO_CHANNELS_MONO
-#define	M1575_DEFAULT_PREC		AUDIO_PRECISION_8
-#define	M1575_DEFAULT_ENC		AUDIO_ENCODING_ULAW
-#define	M1575_DEFAULT_PGAIN		(AUDIO_MAX_GAIN * 3/4)
-#define	M1575_DEFAULT_RGAIN		(127)
-#define	M1575_DEFAULT_MONITOR_GAIN	(0)
-#define	M1575_DEFAULT_BAL		AUDIO_MID_BALANCE
-#define	M1575_INTS			(175)	/* default interrupt rate */
-#define	M1575_MIN_INTS			(25)	/* minimum interrupt rate */
-#define	M1575_MAX_INTS			(5000)	/* maximum interrupt rate */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif /* _SYS_AUDIO1575_H_ */
diff --git a/usr/src/uts/sun/sys/audio/audio_4231.h b/usr/src/uts/sun/sys/audio/audio_4231.h
deleted file mode 100644
index 1f66c8cd1e..0000000000
--- a/usr/src/uts/sun/sys/audio/audio_4231.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License, Version 1.0 only
- * (the "License").  You may not use this file except in compliance
- * with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright (c) 1999-2001 by Sun Microsystems, Inc.
- * All rights reserved.
- */
-
-#ifndef _SYS_AUDIO_4231_H
-#define	_SYS_AUDIO_4231_H
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Header file for the audiocs device driver.
- */
-
-/*
- * Values returned by the AUDIO_GETDEV ioctl()
- */
-#define	CS_DEV_NAME		"SUNW,CS4231"
-#define	CS_DEV_CONFIG_ONBRD1	"onboard1"
-#define	CS_DEV_VERSION		"a"	/* SS5 				*/
-#define	CS_DEV_VERSION_A	CS_DEV_VERSION
-#define	CS_DEV_VERSION_B	"b"	/* Electron - internal loopback	*/
-#define	CS_DEV_VERSION_C	"c"	/* Positron			*/
-#define	CS_DEV_VERSION_D	"d"	/* PowerPC - Retired		*/
-#define	CS_DEV_VERSION_E	"e"	/* x86 - Retired		*/
-#define	CS_DEV_VERSION_F	"f"	/* Tazmo			*/
-#define	CS_DEV_VERSION_G	"g"	/* Quark Audio Module		*/
-#define	CS_DEV_VERSION_H	"h"	/* Darwin			*/
-
-/*
- * Driver supported configuration information
- */
-#define	CS4231_NAME		"audiocs"
-#define	CS4231_MOD_NAME		"CS4231 mixer audio driver"
-
-#define	CS4231_SAMPR5510	(5510)
-#define	CS4231_SAMPR6620	(6620)
-#define	CS4231_SAMPR8000	(8000)
-#define	CS4231_SAMPR9600	(9600)
-#define	CS4231_SAMPR11025	(11025)
-#define	CS4231_SAMPR16000	(16000)
-#define	CS4231_SAMPR18900	(18900)
-#define	CS4231_SAMPR22050	(22050)
-#define	CS4231_SAMPR27420	(27420)
-#define	CS4231_SAMPR32000	(32000)
-#define	CS4231_SAMPR33075	(33075)
-#define	CS4231_SAMPR37800	(37800)
-#define	CS4231_SAMPR44100	(44100)
-#define	CS4231_SAMPR48000	(48000)
-
-#define	CS4231_DEFAULT_SR	CS4231_SAMPR8000
-#define	CS4231_DEFAULT_CH	AUDIO_CHANNELS_MONO
-#define	CS4231_DEFAULT_PREC	AUDIO_PRECISION_8
-#define	CS4231_DEFAULT_ENC	AUDIO_ENCODING_ULAW
-#define	CS4231_DEFAULT_PGAIN	AUDIO_MID_GAIN
-#define	CS4231_DEFAULT_RGAIN	AUDIO_MID_GAIN
-#define	CS4231_DEFAULT_MONITOR_GAIN	(0)
-#define	CS4231_DEFAULT_BAL	AUDIO_MID_BALANCE	/* MUST be mid */
-#define	CS4231_INTS		(175)		/* default interrupt rate */
-#define	CS4231_MIN_INTS		(10)		/* minimum interrupt rate */
-#define	CS4231_MAX_INTS		(2000)		/* maximum interrupt rate */
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO_4231_H */
diff --git a/usr/src/uts/sun/sys/audio/impl/audio1575_impl.h b/usr/src/uts/sun/sys/audio/impl/audio1575_impl.h
deleted file mode 100644
index 0e7397aa6d..0000000000
--- a/usr/src/uts/sun/sys/audio/impl/audio1575_impl.h
+++ /dev/null
@@ -1,766 +0,0 @@
-/*
- * CDDL HEADER START
- *
- * The contents of this file are subject to the terms of the
- * Common Development and Distribution License (the "License").
- * You may not use this file except in compliance with the License.
- *
- * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
- * or http://www.opensolaris.org/os/licensing.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- *
- * When distributing Covered Code, include this CDDL HEADER in each
- * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
- * If applicable, add the following below this CDDL HEADER, with the
- * fields enclosed by brackets "[]" replaced with your own identifying
- * information: Portions Copyright [yyyy] [name of copyright owner]
- *
- * CDDL HEADER END
- */
-/*
- * Copyright 2007 Sun Microsystems, Inc.  All rights reserved.
- * Use is subject to license terms.
- */
-
-#ifndef	_SYS_AUDIO1575_IMPL_H_
-#define	_SYS_AUDIO1575_IMPL_H_
-
-#pragma ident	"%Z%%M%	%I%	%E% SMI"
-
-#ifdef	__cplusplus
-extern "C" {
-#endif
-
-/*
- * Implementation specific header file for the audio1575 device driver.
- */
-
-#ifdef _KERNEL
-
-/* Misc. defines */
-#define	M1575_IDNUM				(0x5455)
-#define	M1575_CONFIG_DEVICE_ID			M1575_IDNUM
-#define	M1575_CONFIG_VENDOR_ID			(0x10b9)
-#define	M1575_CONFIG_SUBSYSTEM_ID		M1575_IDNUM
-#define	M1575_CONFIG_SUBSYSTEM_VENDOR_ID	M1575_CONFIG_VENDOR_ID
-#define	M1575_AUDIO_PCICFG_SPACE	(0)
-#define	M1575_AUDIO_IO_SPACE		(1)
-#define	M1575_AUDIO_MEM_SPACE		(2)
-
-#define	M1575_MINPACKET			(0)
-#define	M1575_MAXPACKET			(1*1024)
-#define	M1575_HIWATER			(64*1024)
-#define	M1575_LOWATER			(32*1024)
-#define	M1575_LOOP_CTR			(100)
-
-/* Gain and attenuation shift values */
-#define	M1575_GAIN_SHIFT3		(3)
-#define	M1575_GAIN_SHIFT4		(4)
-#define	M1575_BYTE_SHIFT		(8)
-
-/* audio direction */
-#define	M1575_DMA_PCM_IN		(1)
-#define	M1575_DMA_PCM_OUT		(2)
-
-/* last AC97 saved register */
-#define	M1575_LAST_AC_REG		(0x3a)
-
-/* Restore audio flags */
-#define	M1575_INIT_RESTORE		(0)
-#define	M1575_INIT_NO_RESTORE		~M1575_INIT_RESTORE
-
-/* AC97 codec shadow reg to index macro */
-#define	M1575_CODEC_REG(r)		((r) >> 1)
-
-/* play and record sample buffer counts */
-#define	M1575_PLAY_BUFS			(2)
-#define	M1575_PLAY_BUF_MSK		(M1575_PLAY_BUFS - 1)
-#define	M1575_REC_BUFS			(4)
-#define	M1575_REC_BUF_MSK		(M1575_REC_BUFS - 1)
-
-/* Buffer Descriptor  List defines */
-#define	M1575_BD_NUMS			(32)
-#define	M1575_BD_MSK			(M1575_BD_NUMS - 1)
-#define	M1575_BD_SIZE	\
-	(M1575_BD_NUMS * sizeof (m1575_bd_entry_t))
-
-/* default buffer size */
-#define	M1575_BSIZE			(8*1024)
-#define	M1575_MOD_SIZE			(16)
-#define	M1575_PLAY_BUF_SZ		(1024)
-#define	M1575_RECORD_BUF_SZ		(1024)
-#define	M1575_BUF_MIN			(512)
-#define	M1575_BUF_MAX			(8192)
-
-/* Audio channel defines */
-#define	M1575_MAX_CHANNELS		(32)
-#define	M1575_MAX_HW_CHANNELS		(6)
-#define	M1575_MAX_IN_CHANNELS		(1)
-#define	M1575_MAX_OUT_CHANNELS	\
-	(M1575_MAX_HW_CHANNELS - M1575_MAX_IN_CHANNELS)
-#define	M1575_INPUT_STREAM		1
-#define	M1575_PORT_UNMUTE		0xffffffff
-
-/* kstat interrupt counter define */
-#define	M1575_KIOP(X)	((kstat_intr_t *)(X->m1575_ksp->ks_data))
-
-/* AD1981B Specific Definitions */
-#define	AC97_MISC_CONTROL_BIT_REGISTER	0x76
-#define	MIC_20dB_GAIN	0x0000
-#define	MIC_10dB_GAIN	0x0001
-#define	MIC_30dB_GAIN	0x0010
-#define	C2MIC		0x0040
-#define	C1MIC		0x0000
-#define	AC97_MIXER_ADC_GAIN_REGISTER	0x64
-#define	MIXER_GAIN_MUTE	0x8000
-#define	MIXER_0db_GAIN_	0x0000
-/* Check for poweron status every 10 ms */
-#define	AD1981_POWERON_DELAY_USEC	10000
-
-/* PCI CFG SPACE REGISTERS for Audio (Device 29, Function 0) */
-#define	M1575_PCIID_REG		0x00	/* Vendor ID 32 */
-#define	M1575_PCICS_REG		0x04	/* Cmd & Status 32 */
-#define	M1575_PCIREV_REG	0x08	/* Class Code & RevId 32 */
-#define	M1575_PCILT_REG		0x0C	/* PCI latency 32 */
-#define	M1575_PCIIO_REG		0x10	/* PCI IOBASE 32 */
-#define	M1575_PCIMEM_REG	0x14	/* PCI MEMBASE 32 */
-#define	M1575_PCIINT_REG	0x3C	/* PCI INT Pin & Line 32 */
-#define	M1575_PCIPMR_REG	0x42	/* Power Capabilities 16 */
-#define	M1575_PCIPMCSR_REG	0x44	/* Power Cmd & Status 16 */
-#define	M1575_PCISCCR_REG	0x48	/* System Cfg Cntrl 16 */
-#define	M1575_PCIAPMUCR1_REG	0x54	/* Add. PMU Cntrl Reg  8 */
-#define	M1575_PCISCRR_REG	0x57	/* Scratch Reg  8 */
-#define	M1575_PCIMISC_REG	0x58	/* Misc Reg  8 */
-#define	M1575_PCIGCC_REG	0x59	/* Global Clk Control 16 */
-#define	M1575_PCIACD_REG	0x5C	/* AC97 Codec Detect  8 */
-#define	M1575_PCIMISC_REG	0x58	/* Misc Reg  8 */
-#define	M1575_PCIGCLK_REG	0x59	/* Misc Reg  8 */
-#define	M1575_PCIMSICTRL_REG	0x62	/* MSI Control Reg 16 */
-#define	M1575_PCIMSIADDR_REG	0x64	/* MSI Address Reg 32 */
-#define	M1575_PCIMSIDATA_REG	0x68	/* MSI Data Reg 16 */
-#define	M1575_PCIMSIMASK_REG	0x6C	/* MSI Data Reg 32 */
-#define	M1575_PCIMSIPEND_REG	0x70	/* MSI Pend Reg 32 */
-
-/* Bit definitions for PCI AC97 Clk detect Reg */
-#define	M1575_PCIACD_CLKDET	0x01
-#define	M1575_PCIMISC_INTENB	0x40
-#define	M1575_PCIINT_LINE	0x05
-
-/* Base Line Audio I/O Memory Registers */
-#define	M1575_SCR_REG		0x00	/* System Control Reg 32 */
-#define	M1575_SSR_REG		0x04	/* System System Reg 32 */
-#define	M1575_DMACR_REG		0x08	/* DMA Control Reg 32 */
-#define	M1575_FIFOCR1_REG	0x0C	/* FIFO 1 Control Reg 32 */
-#define	M1575_INTFCR_REG	0x10	/* Interface Ctrl Reg 32 */
-#define	M1575_INTRCR_REG	0x14	/* Interrupt Ctrl Reg 32 */
-#define	M1575_INTRSR_REG	0x18	/* Interrupt Status Reg 32 */
-#define	M1575_FIFOCR2_REG	0x1C	/* FIFO 2 Control Reg 32 */
-#define	M1575_CPR_REG		0x20	/* Cmd Port Reg 32 */
-#define	M1575_SPR_REG		0x24	/* Status Port Reg 32 */
-#define	M1575_FIFOCR3_REG	0x2C	/* FIFO 3 Control Reg 32 */
-#define	M1575_TTSR_REG		0x30	/* Tx Tag Slot Reg 32 */
-#define	M1575_RTSR_REG		0x34	/* Rx Tag Slot Reg 32 */
-#define	M1575_CSPSR_REG		0x38	/* CSP Status Reg 32 */
-#define	M1575_CASR_REG		0x3C	/* Codec Access Sem Reg 32 */
-
-/* PCM IN Registers */
-#define	M1575_PCMIBDBAR_REG	0x40 	/* 32 */
-#define	M1575_PCMICIV_REG	0x44 	/* 8 */
-#define	M1575_PCMILVIV_REG	0x45 	/* 8 */
-#define	M1575_PCMISR_REG	0x46 	/* 16 */
-#define	M1575_PCMIPICB_REG	0x48 	/* 16 */
-#define	M1575_PCMICR_REG	0x4B 	/* 8 */
-
-/* PCM OUT Registers */
-#define	M1575_PCMOBDBAR_REG	0x50 	/* 32 */
-#define	M1575_PCMOCIV_REG	0x54 	/* 8 */
-#define	M1575_PCMOLVIV_REG	0x55 	/* 8 */
-#define	M1575_PCMOSR_REG	0x56 	/* 16 */
-#define	M1575_PCMOPICB_REG	0x58 	/* 16 */
-#define	M1575_PCMOCR_REG	0x5B 	/* 8 */
-
-/* MIC In Registers */
-#define	M1575_MICIBDBAR_REG	0x60 	/* 32 */
-#define	M1575_MICICIV_REG	0x64 	/* 8 */
-#define	M1575_MICILVIV_REG	0x65 	/* 8 */
-#define	M1575_MICISR_REG	0x66 	/* 16 */
-#define	M1575_MICIPICB_REG	0x68 	/* 16 */
-#define	M1575_MICICR_REG	0x6B 	/* 8 */
-
-/* SPIDOF Registers */
-#define	M1575_CSPOBDBAR_REG	0x70 	/* 32 */
-#define	M1575_CSPOCIV_REG	0x74 	/* 8 */
-#define	M1575_CSPOLVIV_REG	0x75 	/* 8 */
-#define	M1575_CSPOSR_REG	0x76 	/* 16 */
-#define	M1575_CSPOPICB_REG	0x78 	/* 16 */
-#define	M1575_CSPOCR_REG	0x7B 	/* 8 */
-
-/* PCM IN2 Registers */
-#define	M1575_PCMI2BDBAR_REG	0xd0 	/* 32 */
-#define	M1575_PCMI2CIV_REG	0xd4 	/* 8 */
-#define	M1575_PCMI2LVIV_REG	0xd5 	/* 8 */
-#define	M1575_PCMI2SR_REG	0xd6 	/* 16 */
-#define	M1575_PCMI2PICB_REG	0xd8 	/* 16 */
-#define	M1575_PCMI2CR_REG	0xdB 	/* 8 */
-
-/* MIC2 IN2 Registers */
-#define	M1575_MICI2BDBAR_REG	0xe0 	/* 32 */
-#define	M1575_MICI2CIV_REG	0xe4 	/* 8 */
-#define	M1575_MICI2LVIV_REG	0xe5 	/* 8 */
-#define	M1575_MICI2SR_REG	0xe6 	/* 16 */
-#define	M1575_MICI2PICB_REG	0xe8 	/* 16 */
-#define	M1575_MICI2CR_REG	0xeB 	/* 8 */
-
-/* Bits of FIFO Control Register1 */
-#define	M1575_FIFOCR1_CSPORST	0x80000000 /* SPDIF Out Reset */
-#define	M1575_FIFOCR1_MICIRST	0x00800000 /* MIC In Reset */
-#define	M1575_FIFOCR1_PCMORST	0x00008000 /* PCM Out Reset */
-#define	M1575_FIFOCR1_PCMIRST	0x00000080 /* PCM In Reset */
-
-/* Bits of FIFO Control Register2 */
-#define	M1575_FIFOCR2_SPORST	0x80000000 /* SPDIF Out FIFO Reset */
-#define	M1575_FIFOCR2_SPIRST	0x00800000 /* SPDIF In  FIFO Reset */
-#define	M1575_FIFOCR2_LFEORST	0x00008000 /* LFE Out FIFO Reset */
-#define	M1575_FIFOCR2_CENORST	0x00000080 /* CENTER Out Reset */
-
-/* Bits of FIFO Control Register3 */
-#define	M1575_FIFOCR3_PCMI2RST	0x00800000 /* PCM In2 FIFO  Reset */
-#define	M1575_FIFOCR3_MICI2RST	0x00008000 /* MIC In2 FIFO Reset */
-#define	M1575_FIFOCR3_I2SIRST	0x00000080 /* I2S In FIFO Reset */
-
-/* Bits of DMA Control Register */
-#define	M1575_DMACR_PCMISTART	0x00000001
-#define	M1575_DMACR_PCMOSTART	0x00000002
-#define	M1575_DMACR_MICISTART	0x00000004
-#define	M1575_DMACR_CSPOSTART	0x00000008
-#define	M1575_DMACR_CENOSTART	0x00000010
-#define	M1575_DMACR_LFEOSTART	0x00000020
-#define	M1575_DMACR_SPISTART	0x00000040
-#define	M1575_DMACR_SPOSTART	0x00000080
-#define	M1575_DMACR_I2SISTART	0x00000100
-#define	M1575_DMACR_PCMI2START	0x00000200
-#define	M1575_DMACR_MICI2START	0x00000400
-#define	M1575_DMACR_PCMIPAUSE	0x00010000
-#define	M1575_DMACR_PCMOPAUSE	0x00020000
-#define	M1575_DMACR_MICIPAUSE	0x00040000
-#define	M1575_DMACR_CSPOPAUSE	0x00080000
-#define	M1575_DMACR_CENOPAUSE	0x00100000
-#define	M1575_DMACR_LFEOPAUSE	0x00200000
-#define	M1575_DMACR_SPIPAUSE	0x00400000
-#define	M1575_DMACR_SPOPAUSE	0x00800000
-#define	M1575_DMACR_I2SIPAUSE	0x01000000
-#define	M1575_DMACR_PCMI2PAUSE	0x02000000
-#define	M1575_DMACR_MICI2PAUSE	0x04000000
-
-#define	M1575_DMACR_PAUSE_ALL	0x07ff0000
-
-/* Bits of INTRSR Interrupt Status Register */
-#define	M1575_INTRSR_GPIOINTR	0x0000002
-#define	M1575_INTRSR_SPRINTR	0x0000020
-#define	M1575_INTRSR_CPRINTR	0x0000080
-#define	M1575_INTRSR_PCMIINTR   0x0010000
-#define	M1575_INTRSR_PCMOINTR 	0x0020000
-#define	M1575_INTRSR_MICIINTR  	0x0040000
-#define	M1575_INTRSR_CSPOINTR  	0x0080000
-#define	M1575_INTRSR_CENOINTR  	0x0100000
-#define	M1575_INTRSR_LFEOINTR  	0x0200000
-#define	M1575_INTRSR_SPIINTR	0x0400000
-#define	M1575_INTRSR_SPOINTR	0x0800000
-#define	M1575_INTRSR_I2SIINTR	0x1000000
-#define	M1575_INTRSR_PCMI2INTR	0x2000000
-#define	M1575_INTRSR_MICI2INTR	0x4000000
-
-#define	M1575_INTR_MASK (M1575_INTRSR_GPIOINTR |\
-	M1575_INTRSR_SPRINTR  |\
-	M1575_INTRSR_CPRINTR  |\
-	M1575_INTRSR_PCMIINTR |\
-	M1575_INTRSR_PCMOINTR |\
-	M1575_INTRSR_MICIINTR |\
-	M1575_INTRSR_CSPOINTR |\
-	M1575_INTRSR_CENOINTR |\
-	M1575_INTRSR_LFEOINTR |\
-	M1575_INTRSR_SPIINTR  |\
-	M1575_INTRSR_SPOINTR  |\
-	M1575_INTRSR_I2SIINTR |\
-	M1575_INTRSR_PCMI2INTR|\
-	M1575_INTRSR_MICI2INTR)
-
-#define	M1575_UNUSED_INTR_MASK (M1575_INTRSR_GPIOINTR |\
-	M1575_INTRSR_SPRINTR  |\
-	M1575_INTRSR_CPRINTR  |\
-	M1575_INTRSR_MICIINTR |\
-	M1575_INTRSR_CSPOINTR |\
-	M1575_INTRSR_CENOINTR |\
-	M1575_INTRSR_LFEOINTR |\
-	M1575_INTRSR_SPIINTR  |\
-	M1575_INTRSR_SPOINTR  |\
-	M1575_INTRSR_I2SIINTR |\
-	M1575_INTRSR_PCMI2INTR|\
-	M1575_INTRSR_MICI2INTR)
-
-/* Defines a generic clear for all MIC and PCM Status Registers */
-#define	M1575_STATUS_CLR	0x001e
-
-/* Defines a generic RESET for all MIC and PCM Control Registers */
-#define	M1575_CR_RR		0x02
-#define	M1575_SR_DMACS		0x01
-
-/* Bits of PCM In Status Register */
-#define	M1575_PCMISR_DMACS	0x01 	/* DMACS=0 if DMA Engine is IDLE */
-#define	M1575_PCMISR_CELV	0x02
-#define	M1575_PCMISR_LVBCI	0x04
-#define	M1575_PCMISR_BCIS	0x08
-#define	M1575_PCMISR_FIFOE	0x10
-
-/* Bits in PCM In Control Register */
-#define	M1575_PCMICR_RR		0x02	/* Reset */
-#define	M1575_PCMICR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
-#define	M1575_PCMICR_IOCE	0x10	/* Intr On Completion Enable */
-
-/* Bits of PCM Out Status Register */
-#define	M1575_PCMOSR_DMACS	0x01	/* DMACS=0 if DMA Engine is IDLE */
-#define	M1575_PCMOSR_CELV	0x02
-#define	M1575_PCMOSR_LVBCI	0x04
-#define	M1575_PCMOSR_BCIS	0x08
-#define	M1575_PCMOSR_FIFOE	0x10
-
-/* Bits in PCM Out Control Register */
-#define	M1575_PCMOCR_RR		0x02	/* Reset */
-#define	M1575_PCMOCR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
-#define	M1575_PCMOCR_IOCE	0x10	/* Intr On Completion Enable */
-
-/* Bits of MIC In Status Register */
-#define	M1575_MICISR_DMACS	0x01	/* DMACS=0 if DMA Engine is IDLE */
-#define	M1575_MICISR_CELV	0x02
-#define	M1575_MICISR_LVBCI	0x04
-#define	M1575_MICISR_BCIS	0x08
-#define	M1575_MICISR_FIFOE	0x10
-
-/* Bits in PCM In Control Register */
-#define	M1575_MICICR_RR		0x02	/* Reset */
-#define	M1575_MICICR_LVBIE	0x04	/* Last valid Buffer Intr Enable */
-#define	M1575_MICICR_IOCE	0x10	/* Intr On Completion Enable */
-
-/* Bits in System Control  Register */
-#define	M1575_SCR_WARMRST	0x00000001
-#define	M1575_SCR_COLDRST	0x00000002
-#define	M1575_SCR_DRENT		0x40000000
-#define	M1575_SCR_MSTRST	0x80000000
-
-/* Bits in System Status Register */
-#define	M1575_SSR_RSTBLK	0x00000002
-#define	M1575_SSR_FACCS_MSK	0x00000018
-#define	M1575_SSR_SCID		0x00000040
-
-/* Bits in Command Port Register */
-#define	M1575_CPR_ACSCS		0x0100 /* Audio Codec for cmd 1=codec 2 */
-#define	M1575_CPR_READ		0x0080
-
-/* Bits in Cmd Status Port Register */
-#define	M1575_CSPSR_SUCC	0x08	/* cmd successful */
-#define	M1575_CSPSR_RDRDY	0x02	/* ready for read cmd */
-#define	M1575_CSPSR_WRRDY	0x01	/* ready for write cmd */
-#define	M1575_PCMI2CR_RR	0x02	/* Reset */
-#define	M1575_MICI2CR_RR	0x02	/* Reset */
-#define	M1575_CSPOCR_RR		0x02	/* Reset */
-
-/* Bits in  Interface  Control  Register */
-#define	M1575_INTFCR_RSTREL		0x02000000
-#define	M1575_INTFCR_RSTBLK		0x00200000
-#define	M1575_INTFCR_MICENB		0x00100000
-#define	M1575_INTFCR_PCMIENB		0x00080000
-#define	M1575_INTFCR_MICI2ENB		0x00040000
-#define	M1575_INTFCR_PCMI2ENB		0x00020000
-#define	M1575_INTFCR_MICI2SEL		0x00008000
-#define	M1575_INTFCR_MICISEL		0x00004000
-#define	M1575_INTFCR_PCMOENB		0x00000002
-
-#define	M1575_INTRCR_CPRINTR		0x00000080
-#define	M1575_INTRCR_SPRINTR		0x00000020
-#define	M1575_INTRCR_GPIOINTR		0x00000002
-
-/* Bits of Recv Tag Slot Register */
-#define	M1575_RTSR_SACRDY	0x20u	/* 2nd Audio Codec Rdy */
-#define	M1575_RTSR_FACRDY	0x80u	/* 1st Audio Codec Rdy */
-
-/* Semaphore busy */
-#define	M1575_CASR_SEMBSY	0x80000000
-
-/* AD1981 codec vendor ID */
-#define	AD1981_VID1		0x4144
-#define	AD1981_VID2		0x5374
-
-/* AD1981B Codec Registers */
-#define	AD1981_RESET_REG	0x00
-#define	AD1981_MSTVOL_REG	0x02
-#define	AD1981_HPHVOL_REG	0x04
-#define	AD1981_MONOVOL_REG	0x06
-#define	AD1981_MICVOL_REG	0x0E
-#define	AD1981_CDVOL_REG	0x12
-#define	AD1981_PCMOVOL_REG	0x18
-#define	AD1981_RECSEL_REG	0x1A
-#define	AD1981_RECGAIN_REG	0x1C
-#define	AD1981_GENPUR_REG	0x20
-#define	AD1981_PWRCSR_REG	0x26
-#define	AD1981_EXTID_REG	0x28
-#define	AD1981_EXTCSR_REG	0x2A
-#define	AD1981_PCMDAC_REG	0x2C
-#define	AD1981_PCMADC_REG	0x32
-#define	AD1981_EQCTRL_REG	0x60
-#define	AD1981_EQDATA_REG	0x62
-#define	AD1981_MIXVOL_REG	0x64
-#define	AD1981_MISCTRL_REG	0x76
-#define	AD1981_VNDID1_REG	0x7C
-#define	AD1981_VNDID2_REG	0x7E
-
-/* AD1981B Biquad filter definitions */
-#define	AD1981_MAX_FILTERS	35
-#define	AD1981_EQCTRL_EQM	0x8000
-#define	AD1981_EQCTRL_SYM	0x0080
-
-/*
- * Equalizer Biquad Filter Coefficient Address offsets
- */
-struct m1575_biquad {
-	uint16_t addr;
-	uint16_t coeff;
-};
-typedef struct m1575_biquad m1575_biquad_t;
-
-/*
- * chunk buffer
- */
-struct m1575_bdlist_chunk {
-	caddr_t	 data_buf;		/* virtual address of buffer */
-	uint32_t addr_phy;		/* physical address of buffer */
-	ddi_dma_handle_t dma_handle;	/* dma handle */
-	ddi_acc_handle_t acc_handle;	/* access handle */
-	size_t real_len;		/* real len */
-};
-typedef struct m1575_bdlist_chunk m1575_bdlist_chunk_t;
-
-/*
- * sample buffer
- */
-struct m1575_sample_buf {
-	boolean_t io_started;	/* start/stop state for play/record */
-	int avail;		/* the number of available chunk(s) */
-	uint8_t tail;		/* For CPU, 1st available BD entry */
-	uint8_t head;		/* For CPU, 1st BD entry to reclaim */
-	m1575_bdlist_chunk_t chunk[M1575_BD_NUMS];
-};
-typedef struct m1575_sample_buf m1575_sample_buf_t;
-
-/*
- * buffer descripter list entry, see M1575 datasheet
- */
-#define	IOC 0x8000
-#define	BUP 0x4000
-
-struct m1575_bd_entry {
-	uint32_t buf_base;	/* the address of the buffer */
-	uint16_t buf_len;	/* the number of samples */
-	uint16_t buf_cmd;
-};
-typedef struct m1575_bd_entry m1575_bd_entry_t;
-
-/*
- * PCI config space register layout
- */
-struct audio1575_pci_regs {
-    uint32_t	vendor_dev_id;			/* 00h - 03h */
-    uint32_t	cmd_status_reg;			/* 04h - 07h */
-    uint32_t	class_code_rev_id;		/* 08h - 0bh */
-    uint32_t	bist_cache;			/* 0ch - 0fh */
-    uint32_t	io_base;			/* 10h - 13h */
-    uint32_t	mem_base;			/* 14h - 17h */
-    uint32_t	rsvd1[4];			/* 18h - 28h */
-    uint32_t	sub_ids;			/* 2ch - 2fh */
-    uint32_t	rsvd2;				/* 30h - 33h */
-    uint32_t	cap_ptr;			/* 34h - 37h */
-    uint32_t	rsvd3;				/* 38h - 3bh */
-    uint32_t	intr_line;			/* 3ch - 3fh */
-    uint16_t	cap_id_next;			/* 40h - 41h */
-    uint16_t	pm_cap;				/* 42h - 43h */
-    uint16_t	pm_csr;				/* 44h - 45h */
-    uint8_t	pm_csrbse;			/* 46h - 46h */
-    uint8_t	data_reg;			/* 47h - 47h */
-    uint16_t  	sccr_reg;			/* 48h - 49h */
-    uint16_t  	subvendor_id;			/* 50h - 51h */
-    uint16_t  	subdevice_id;			/* 52h - 53h */
-    uint8_t	apmucr1_reg;			/* 54h	*/
-    uint8_t	apmucr2_reg;			/* 55h 	*/
-    uint8_t	itec;				/* 56h	*/
-    uint8_t	scrr_reg;			/* 57h	*/
-    uint8_t	misc_reg;			/* 58h	*/
-    uint8_t	gcc_reg;			/* 59h	*/
-    uint16_t	rsvd5;				/* 5ah - 5bh */
-    uint8_t	ac97acd_reg;			/* 5ch	*/
-    uint8_t	rsvd6[3];			/* 5dh - 5fh */
-    uint8_t	msi_capid;			/* 60h	*/
-    uint8_t	msi_next;			/* 61h	*/
-    uint16_t	msi_ctrl;			/* 62h - 63h */
-    uint32_t	msi_addr;			/* 64h - 67h */
-    uint16_t	msi_data;			/* 68h - 69h */
-    uint16_t	rsvd7;				/* 6ah - 6bh */
-    uint32_t	msi_mask;			/* 6ch - 6fh */
-    uint32_t	msi_pend;			/* 70h - 73h */
-    uint16_t	rsrvd8[39];			/* 74h - c1h */
-    uint16_t	smodem_devid;			/* c2h - c3h */
-};
-typedef struct audio1575_pci_regs audio1575_pci_regs_t;
-
-/*
- * M1575 audio register layout
- */
-struct audio1575_audio_regs {
-	uint32_t	scr_reg;		/* 00h - 03h */
-	uint32_t	ssr_reg;		/* 04h - 07h */
-	uint32_t	dmacr_reg;		/* 08h - 0bh */
-	uint32_t	fifocr1_reg;		/* 0ch - 0fh */
-	uint32_t	intfcr_reg;		/* 10h - 13h */
-	uint32_t	intrcr_reg;		/* 14h - 17h */
-	uint32_t	intrsr_reg;		/* 18h - 1bh */
-	uint32_t	fifocr2_reg;		/* 1ch - 1fh */
-	uint32_t	cpr_reg;		/* 20h - 23h */
-	uint32_t	spr_reg;		/* 24h - 27h */
-	uint32_t	rsvd1;			/* 28h - 2bh */
-	uint32_t	fifocr3_reg;		/* 2ch - 2fh */
-	uint32_t	ttsr_reg;		/* 30h - 33h */
-	uint32_t	rtsr_reg;		/* 34h - 37h */
-	uint32_t	cspsr_reg;		/* 38h - 3bh */
-	uint32_t	casr_reg;		/* 3ch - 3fh */
-	uint32_t	pcmibdbar_reg;		/* 40h - 43h */
-	uint8_t		pcmciv_reg;		/* 44h - 44h */
-	uint8_t		pcmilviv_reg;		/* 45h - 45h */
-	uint16_t	pcmisr_reg;		/* 46h - 47h */
-	uint16_t	pcmipicb_reg;		/* 48h - 49h */
-	uint8_t		rsvd2;			/* 4ah - 4ah */
-	uint8_t		pcmicr_reg;		/* 4bh - 4bh */
-	uint32_t	rsvd3;			/* 4ch - 4fh */
-	uint32_t	pcmobdbar_reg;		/* 50h - 53h */
-	uint8_t		pcmociv_reg;		/* 54h - 54h */
-	uint8_t		pcmolviv_reg;		/* 55h - 55h */
-	uint16_t	pcmosr_reg;		/* 56h - 57h */
-	uint16_t	pcmopicb_reg;		/* 58h - 59h */
-	uint8_t		rsvd4;			/* 5ah - 5ah */
-	uint8_t		pcmocr_reg;		/* 5bh - 5bh */
-	uint32_t	rsvd5;			/* 5ch - 5fh */
-	uint32_t	micibdbar_reg;		/* 60h - 63h */
-	uint8_t		miciciv_reg;		/* 64h - 64h */
-	uint8_t		micilviv_reg;		/* 65h - 65h */
-	uint16_t	micisr_reg;		/* 66h - 67h */
-	uint16_t	micipicb_reg;		/* 68h - 69h */
-	uint8_t		rsvd6;			/* 6ah - 6ah */
-	uint8_t		micicr_reg;		/* 6bh - 6bh */
-	uint32_t	rsvd7;			/* 6ch - 6fh */
-	uint32_t	cspobdbar_reg;		/* 70h - 53h */
-	uint8_t		cspociv_reg;		/* 74h - 74h */
-	uint8_t		cspolviv_reg;		/* 75h - 75h */
-	uint16_t	csposr_reg;		/* 76h - 77h */
-	uint16_t	cspopicb_reg;		/* 78h - 79h */
-	uint8_t		rsvd8;			/* 7ah - 7ah */
-	uint8_t		cspocr_reg;		/* 7bh - 7bh */
-	uint32_t	rsvd9[21];		/* 7ch - cfh */
-	uint32_t	pcmi2bdbar_reg;		/* d0h - d3h */
-	uint8_t		pcmi2civ_reg;		/* d4h - d4h */
-	uint8_t		pcmi2lviv_reg;		/* d5h - d5h */
-	uint16_t	pcmi2sr_reg;		/* d6h - d7h */
-	uint16_t	pcmi2picb_reg;		/* d8h - d9h */
-	uint8_t		rsvd10;			/* dah - dah */
-	uint8_t		pcmi2cr_reg;		/* dbh - dbh */
-	uint32_t	rsvd11;			/* dch - dfh */
-	uint32_t	mici2bdbar_reg;		/* e0h - e3h */
-	uint8_t		mici2civ_reg;		/* e4h - e4h */
-	uint8_t		mici2lviv_reg;		/* e5h - e5h */
-	uint16_t	mici2sr_reg;		/* e6h - e7h */
-	uint16_t	mici2picb_reg;		/* e8h - e9h */
-	uint8_t		rsvd12;			/* eah - eah */
-	uint8_t		mici2cr_reg;		/* ebh - ebh */
-	uint32_t	rsvd13;			/* ech - efh */
-	uint32_t	hvcsr_reg;		/* f0h - f3h */
-	uint32_t	rsvd14[3];		/* f4h - ffh */
-};
-typedef struct audio1575_audio_regs audio1575_audio_regs_t;
-
-/*
- * audio1575_state_t per instance state and operation data
- */
-struct audio1575_state	{
-	kmutex_t		m1575_intr_mutex;	/* intr mutex */
-	dev_info_t		*m1575_dip;		/* dev instance ptr */
-	int			m1575_inst;		/* dev instance */
-	int			m1575_intr_type;	/* intr type */
-	uint_t			m1575_intr_pri;		/* intr priority */
-	ddi_intr_handle_t	*m1575_h_table;		/* intr table ptr */
-	audiohdl_t		m1575_ahandle;		/* audio handle */
-	am_ad_info_t		m1575_ad_info;		/* audio device info */
-	uint16_t		m1575_codec_shadow[64]; /* shadow AC97 regs */
-	ddi_acc_handle_t	m1575_pci_regs_handle;	/* pci config space */
-	ddi_acc_handle_t	m1575_am_regs_handle;	/* audio i/o regs */
-	ddi_acc_handle_t	m1575_bm_regs_handle;	/* audio mem regs */
-	audio1575_pci_regs_t	*m1575_pci_regs;	/* base of pci regs */
-	audio1575_audio_regs_t	*m1575_am_regs;		/* base of i/o regs */
-	audio1575_audio_regs_t	*m1575_bm_regs;		/* base of mem regs */
-	ddi_dma_handle_t	m1575_bdl_dma_handle;	/* for BDL */
-	ddi_acc_handle_t	m1575_bdl_acc_handle;	/* acc handle of BDL */
-	void			*m1575_bdl_virtual;	/* virt addr of BDL */
-	size_t			m1575_bdl_size;		/* real len of BDL */
-	m1575_bd_entry_t	*m1575_bdl_virt_pin;  	/* vaddr PCMIN BDL */
-	m1575_bd_entry_t	*m1575_bdl_virt_pout;	/* vaddr PCMOUT BDL */
-	uint32_t		m1575_bdl_phys_pin;	/* physadr PCMIN BDL */
-	uint32_t		m1575_bdl_phys_pout;	/* physadr PCMOUT BDL */
-	audio_info_t		m1575_defaults;		/* default states */
-	audio_device_t		m1575_dev_info;		/* audio device info */
-	uint16_t		m1575_vol_bits_mask;	/* volume ctrl bits */
-	kstat_t			*m1575_ksp;		/* kernel statistics */
-	uint32_t		m1575_flags;		/* state flags */
-	uint_t			m1575_cdrom;		/* 1= present, 0 not */
-	uint_t			m1575_mode; 		/* MIXER/COMPAT_MODE */
-	uint_t			m1575_psample_rate;	/* play sample rate */
-	uint_t			m1575_pchannels; 	/* play channels */
-	uint_t			m1575_pprecision; 	/* play precision */
-	uint_t			m1575_csample_rate;	/* record sample rate */
-	uint_t			m1575_cchannels;	/* record channels */
-	uint_t			m1575_cprecision;	/* record precision */
-	uint_t			m1575_output_port;	/* current out port */
-	uint_t			m1575_input_port;	/* current input port */
-	uint_t			m1575_monitor_gain;	/* monitor gain */
-	int			m1575_csamples;		/* pcmin samples/int */
-	int			m1575_psamples;		/* pcmout samples/int */
-	uint32_t		m1575_res_flags;	/* resource flags */
-	m1575_sample_buf_t	m1575_play_buf;		/* buf for playback */
-	int			m1575_play_buf_size; 	/* size of in buf */
-	m1575_sample_buf_t	m1575_record_buf;	/* buffer for record */
-	int			m1575_record_buf_size; 	/* size of in buffer */
-};
-typedef struct audio1575_state audio1575_state_t;
-
-/* audio1575_state_t.flags defines */
-#define	M1575_DMA_PLAY_STARTED		0x0001u	/* play DMA eng. initialized */
-#define	M1575_DMA_PLAY_PAUSED		0x0002u	/* play DMA engine paused */
-#define	M1575_DMA_PLAY_EMPTY		0x0004u	/* play DMA engine empty */
-#define	M1575_DMA_RECD_STARTED		0x0010u	/* record DMA engine started */
-#define	M1575_DMA_SUSPENDED		0x0020u	/* DMA suspended flag */
-
-/* bits of audio1575_state_t.m1575_res_flags */
-#define	M1575_RS_PCI_REGS		0x0001
-#define	M1575_RS_AM_REGS		0x0002
-#define	M1575_RS_BM_REGS		0x0004
-#define	M1575_RS_DMA_BDL_HANDLE		0x0008
-#define	M1575_RS_DMA_BDL_MEM		0x0010
-#define	M1575_RS_DMA_BDL_BIND		0x0020
-
-/* PCI Config register macros */
-#define	M1575_PCI_GET8(reg)						\
-	ddi_get8(statep->m1575_pci_regs_handle,				\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)))
-
-#define	M1575_PCI_GET16(reg)						\
-	ddi_get16(statep->m1575_pci_regs_handle,			\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)))
-
-#define	M1575_PCI_GET32(reg)						\
-	ddi_get32(statep->m1575_pci_regs_handle,			\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)))
-
-#define	M1575_PCI_PUT8(reg, val)					\
-	ddi_put8(statep->m1575_pci_regs_handle,				\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)), (val))
-
-#define	M1575_PCI_PUT16(reg, val)					\
-	ddi_put16(statep->m1575_pci_regs_handle,			\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)), (val))
-
-#define	M1575_PCI_PUT32(reg, val)					\
-	ddi_put32(statep->m1575_pci_regs_handle,			\
-	    (void *)((char *)statep->m1575_pci_regs + (reg)), (val))
-
-/* audio i/o register macros */
-#define	M1575_AM_GET8(reg)						\
-	ddi_get8(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)))
-
-#define	M1575_AM_GET16(reg)						\
-	ddi_get16(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)))
-
-#define	M1575_AM_GET32(reg)						\
-	ddi_get32(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)))
-
-#define	M1575_AM_PUT8(reg, val)						\
-	ddi_put8(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)), (val))
-
-#define	M1575_AM_PUT16(reg, val)					\
-	ddi_put16(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)), (val))
-
-#define	M1575_AM_PUT32(reg, val)					\
-	ddi_put32(statep->m1575_am_regs_handle,				\
-	    (void *)((char *)statep->m1575_am_regs + (reg)), (val))
-
-/* audio memory bus master registers */
-#define	M1575_BM_GET8(reg)						\
-	ddi_get8(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)))
-
-#define	M1575_BM_GET16(reg)						\
-	ddi_get16(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)))
-
-#define	M1575_BM_GET32(reg)						\
-	ddi_get32(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)))
-
-#define	M1575_BM_PUT8(reg, val) {					\
-	uint8_t	__T;							\
-	ddi_put8(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)), (val));	\
-	__T = ddi_get8(statep->m1575_bm_regs_handle,			\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)));		\
-	if (__T != val) {						\
-		cmn_err(CE_NOTE, "audio1575: couldn't set "		\
-		    "value (%d 0x%02x 0x%02x)", __LINE__, __T, val);	\
-		    cmn_err(CE_CONT, "audio may not work "		\
-		    "correctly until it is stopped and restarted");	\
-	}								\
-}
-
-#define	M1575_BM_PUT16(reg, val) {					\
-	uint16_t	__T;						\
-	ddi_put16(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)), (val));	\
-	__T = ddi_get16(statep->m1575_bm_regs_handle,			\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)));		\
-	if (__T != val) {						\
-		cmn_err(CE_NOTE, "audio1575: couldn't set "		\
-		    "value (%d 0x%02x 0x%02x)", __LINE__, __T, val);	\
-		    cmn_err(CE_CONT, "audio may not work "		\
-		    "correctly until it is stopped and restarted");	\
-	}								\
-}
-
-#define	M1575_BM_PUT32(reg, val) {					\
-	uint32_t	__T;						\
-	ddi_put32(statep->m1575_bm_regs_handle,				\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)), (val));	\
-	__T = ddi_get32(statep->m1575_bm_regs_handle,			\
-	    (void *)((char *)statep->m1575_bm_regs + (reg)));		\
-	if (__T != val) {						\
-		cmn_err(CE_NOTE, "audio1575: couldn't set "		\
-		    "value (%d 0x%02x 0x%02x)", __LINE__, __T, val);	\
-		    cmn_err(CE_CONT, "audio may not work "		\
-		    "correctly until it is stopped and restarted");	\
-	}								\
-}
-
-#endif	/* _KERNEL */
-
-#ifdef	__cplusplus
-}
-#endif
-
-#endif	/* _SYS_AUDIO1575_IMPL_H_ */