Update to 4.18, provided by Ben Collver in pkg/14630.

Changes are unknown. Package changes:
* the c-preprocessor should now correctly chose whether to build big-endian
  or little-endian specific code
* added support for NetBSD native audio API

2 files changed, 329 insertions, 0 deletions

diff --git a/audio/csound-bath/files/rtNetBSD.c b/audio/csound-bath/files/rtNetBSD.c
new file mode 100644
index 00000000000..abc066f9d89
--- /dev/null
+++ b/audio/csound-bath/files/rtNetBSD.c
@@ -0,0 +1,309 @@
+/* 
+   --- rtNetBSD.c ---
+
+   Native NetBSD audio support.
+
+   Ben Collver, collver@linuxfreemail.com
+*/
+
+#include <sys/types.h>
+#include <sys/audioio.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <unistd.h>
+
+#include "cs.h"
+#include "soundio.h"
+#include "rtNetBSD.h"
+
+static int dspfd;
+
+static  int     ishift = 0, oshift = 0, oMaxLag;
+extern  long    nrecs;
+        long    inrecs;
+extern  OPARMS  O;
+extern int Linefd;
+#ifdef PIPES
+extern FILE* Linepipe;
+#  define _pclose pclose
+#endif
+
+static int getshift(int dsize)  /* turn sample- or frame-size into shiftsize */
+{
+  switch(dsize) {
+  case 1:  return(0);
+  case 2:  return(1);
+  case 4:  return(2);
+  case 8:  return(3);
+  default: die("rtaudio: illegal dsize");
+	return(-1);		/* Not reached */
+        }
+}
+  
+
+void display_audio_info(
+	int fd,
+	const char *message,
+	audio_info_t *info)
+{
+	audio_encoding_t encoding;
+	char *encoding_name;
+
+	encoding.index = info->play.encoding;
+	if (ioctl(fd, AUDIO_GETENC, &encoding) == -1) {
+		asprintf(&encoding_name, "%d", info->play.encoding);
+	} else {
+		encoding_name = strdup(encoding.name);
+	}
+	printf("%-9s\n", message);
+	printf("         info.play.precision = %d\n", info->play.precision);
+	printf("         info.play.encoding = %s\n", encoding_name);
+	printf("         info.play.channels = %d\n", info->play.channels);
+	printf("         info.play.sample_rate = %d\n", info->play.sample_rate);
+	printf("         info.blocksize = %d\n\n", info->blocksize);
+	free(encoding_name);
+}
+
+void setsndparms(
+	int fd,
+	int format,
+	int nchnls,
+	MYFLT esr, 
+	unsigned bufsiz)
+{
+	audio_info_t info;
+	int count;
+	int frag_size;
+
+	AUDIO_INITINFO(&info);
+
+	switch (format) {
+	case AE_UNCH:
+		info.play.precision = 8;
+		info.play.encoding = AUDIO_ENCODING_ULINEAR;
+		break;
+	case AE_CHAR:
+		info.play.precision = 8;
+		info.play.encoding = AUDIO_ENCODING_SLINEAR;
+		break;
+	case AE_ULAW:
+		info.play.precision = 8;
+		info.play.encoding = AUDIO_ENCODING_ULAW;
+		break;
+	case AE_ALAW:
+		info.play.precision = 8;
+		info.play.encoding = AUDIO_ENCODING_ALAW;
+		break;
+	case AE_SHORT:
+		/* audio(4) says AUDIO_ENCODING_SLINEAR uses platform's
+		   byte order, so the below will automatically be BE on
+		   a BE system, and LE on a LE system.  Good/Bad/Ok? */
+
+		info.play.precision = 16;
+		info.play.encoding = AUDIO_ENCODING_SLINEAR;
+		break;
+	case AE_LONG:
+		info.play.precision = 32;
+		info.play.encoding = AUDIO_ENCODING_SLINEAR;
+		break;
+	case AE_FLOAT:
+		die("NetBSD audio does not support floating-point samples");
+	default:
+		die("unknown sample format");
+	}
+
+	info.play.channels = nchnls;
+	info.play.sample_rate = (unsigned int) esr;
+
+	/* set DMA buffer fragment size to Csound's output buffer size */
+	/*
+	 * The or'ing is ABSOLUTELY crucial to obtaining quick response
+	 * of csound to MIDI input - Please don't touch this line unless
+	 * you know you can do better (in terms of MIDI response)
+	 * [nicb@axnet.it]
+	 */
+	frag_size = 16;
+	count = 4;
+	while (frag_size < bufsiz && count < 18) {
+		frag_size <<= 1;
+		count++;
+	}
+	count |= 0x0020000; /* Larry Troxler's Idea */
+	info.blocksize = frag_size;
+
+	/* from ossaudio.c ioctl SNDCTL_DSP_SETFRAGMENT */
+	info.hiwat = ((unsigned)count >> 16) & 0x7fff;
+	if (info.hiwat == 0)
+		info.hiwat = 65536;
+
+	if (ioctl(fd, AUDIO_SETINFO, &info) == -1) {
+		display_audio_info(fd, "requested", &info);
+		if (ioctl(fd, AUDIO_GETINFO, &info) != -1)
+			display_audio_info(fd, "got", &info);
+		die("unable to configure soundcard");
+	}
+}
+
+int find_mixer_label(int fd, int class, const char *name)
+{
+	int i;
+	int mclass;
+	int retval = -1;
+	mixer_devinfo_t info;
+
+        for (i = 0; ; i++) {
+                info.index = i;
+                if (ioctl(fd, AUDIO_MIXER_DEVINFO, &info) < 0)
+                        break;
+		mclass = info.mixer_class;
+		if ((info.index == mclass || class == mclass) &&
+			!strcmp(info.label.name, name))
+		{
+			retval = i;
+		}
+        }
+	return retval;
+}
+
+void setvolume(unsigned volume)
+{
+	int fd;
+	int i;
+	int output_class;
+	int vol_output;
+	mixer_devinfo_t info;
+	mixer_ctrl_t value;
+
+	/* volume must be between 0 and 255 */
+
+	if ((fd = open(NETBSD_MIXER, O_WRONLY)) == -1)
+		die("unable to open soundcard mixer for setting volume");
+
+	output_class = find_mixer_label(fd, 0, "outputs");
+	vol_output = find_mixer_label(fd, output_class, "master");
+
+	if (vol_output == -1)
+		die("Could not find mixer control for audio output.");
+
+
+	info.index = vol_output;
+	ioctl(fd, AUDIO_MIXER_DEVINFO, &info);
+
+	value.dev = vol_output;
+	value.type = info.type;
+	value.un.value.num_channels = 2;
+	if (ioctl(fd, AUDIO_MIXER_READ, &value) < 0) {
+		value.un.value.num_channels = 1;
+		if (ioctl(fd, AUDIO_MIXER_READ, &value) < 0)
+			die("unable to read mixer on soundcard"); 
+	}
+
+	value.un.value.level[0] = 192;
+	value.un.value.level[1] = 192;
+	if (ioctl(fd, AUDIO_MIXER_WRITE, &value) < 0) {
+		die("unable to set output volume on soundcard"); 
+	}
+}
+
+
+void NetBSD_open(int nchnls, int dsize, MYFLT esr, int scale, int audio_mode)
+{
+	int dup;
+	int wbufsiz;
+	int audio_props;
+	audio_device_t device_info;
+#ifdef USE_SETSCHEDULER
+	extern void setscheduler(void);
+#endif
+
+	oMaxLag = O.oMaxLag;        /* import DAC setting from command line   */
+	if (oMaxLag <= 0)           /* if DAC sampframes ndef in command line */
+		oMaxLag = IODACSAMPS;     /* use the default value */
+	wbufsiz = oMaxLag * O.insampsiz;
+	switch (audio_mode) {
+	case NETBSD_RECORD:
+		if ((dspfd = open(NETBSD_SAMPLER, O_RDONLY)) == -1)
+			die("error while opening soundcard for audio input");
+		setsndparms(dspfd, O.informat, nchnls, esr, wbufsiz);
+		ishift = getshift(dsize);  
+		break;
+	case NETBSD_PLAY:
+		if ((dspfd = open(NETBSD_SAMPLER, O_WRONLY)) == -1) {
+			perror("foo");
+			die("error while opening soundcard for audio output");
+		}
+		setsndparms(dspfd, O.outformat, nchnls, esr, wbufsiz);
+		/* 'oshift' is not currently used by the Linux driver, but... */
+		oshift = getshift(nchnls * dsize);
+		break;
+	case NETBSD_DUPLEX:
+		if ((dspfd = open(NETBSD_SAMPLER, O_RDWR)) == -1) 
+			die("error during soundcard duplex mode query:");
+		ioctl(dspfd, AUDIO_GETPROPS, &audio_props);
+		if (!(audio_props | AUDIO_PROP_FULLDUPLEX))
+			die("hardware does not support full duplex mode");
+		setsndparms(dspfd, O.outformat, nchnls, esr, wbufsiz);
+		if (ioctl(dspfd, AUDIO_SETFD, audio_props) == -1)
+			die("error setting hardware to full duplex mode");
+		/* are these functions both required? */
+		oshift = getshift(nchnls * dsize);
+		ishift = getshift(dsize);
+		break;
+	default:
+		fprintf(stderr, "mode specification error in NetBSD_open: ");
+		fprintf(stderr, "unknown argument %d\n", audio_mode);
+		exit(1);
+	}
+
+	ioctl(dspfd, AUDIO_GETDEV, &device_info);
+	fprintf(stderr, "NetBSD audio info: %s, %s, %s\n",
+		device_info.name,
+		device_info.version,
+		device_info.config);
+#ifdef USE_SETSCHEDULER
+	setscheduler();
+#endif
+}
+
+int rtrecord(char *inbuf, int nbytes) /* get samples from ADC */
+{
+    /*  J. Mohr  1995 Oct 17 */
+    if ( (nbytes = read(dspfd, inbuf, nbytes)) == -1 )
+      die("error while reading DSP device for audio input");
+    return(nbytes);
+}
+
+void rtplay(char *outbuf, int nbytes) /* put samples to DAC  */
+    /* N.B. This routine serves as a THROTTLE in Csound Realtime Performance, */
+    /* delaying the actual writes and return until the hardware output buffer */
+    /* passes a sample-specific THRESHOLD.  If the I/O BLOCKING functionality */
+    /* is implemented ACCURATELY by the vendor-supplied audio-library write,  */
+    /* that is sufficient.  Otherwise, requires some kind of IOCTL from here. */
+    /* This functionality is IMPORTANT when other realtime I/O is occurring,  */
+    /* such as when external MIDI data is being collected from a serial port. */
+    /* Since Csound polls for MIDI input at the software synthesis K-rate     */
+    /* (the resolution of all software-synthesized events), the user can      */
+    /* eliminate MIDI jitter by requesting that both be made synchronous with */
+    /* the above audio I/O blocks, i.e. by setting -b to some 1 or 2 K-prds.  */
+{
+        long sampframes = nbytes >> oshift;
+        /*  J. Mohr  1995 Oct 17 */
+        if (write(dspfd, outbuf, nbytes) < nbytes)
+            printf("/dev/audio: couldn't write all bytes requested\n");
+        nrecs++;
+}
+
+void rtclose(void)              /* close the I/O device entirely  */
+{                               /* called only when both complete */
+    /*  J. Mohr  1995 Oct 17 */
+    if (close(dspfd) == -1)
+      die("unable to close DSP device");
+    if (O.Linein) {
+#ifdef PIPES
+      if (O.Linename[0]=='|') _pclose(Linepipe);
+      else
+#endif
+        if (strcmp(O.Linename, "stdin")!=0) close(Linefd);
+    }
+}
diff --git a/audio/csound-bath/files/rtNetBSD.h b/audio/csound-bath/files/rtNetBSD.h
new file mode 100644
index 00000000000..c57ac466386
--- /dev/null
+++ b/audio/csound-bath/files/rtNetBSD.h
@@ -0,0 +1,20 @@
+/* rtNetBSD.h */
+
+#ifndef _RTNETBSD_INCLUDED
+#define _RTNETBSD_INCLUDED
+
+#define NETBSD_RECORD 0
+#define NETBSD_PLAY   1
+#define NETBSD_DUPLEX 2
+
+#undef USE_SETSCHEDULER 
+
+#define NETBSD_MIXER "/dev/mixer0"
+#define NETBSD_SAMPLER "/dev/sound0"
+
+#define recopen(a,b,c,d) NetBSD_open(a,b,c,d,NETBSD_DUPLEX)
+#define playopen(a,b,c,d) NetBSD_open(a,b,c,d,NETBSD_PLAY)
+
+extern void NetBSD_open(int, int, MYFLT, int, int);
+
+#endif /* _RTNETBSD_INCLUDED */