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Diffstat (limited to 'audio/csound/files/rtNetBSD.c')
-rw-r--r--audio/csound/files/rtNetBSD.c309
1 files changed, 309 insertions, 0 deletions
diff --git a/audio/csound/files/rtNetBSD.c b/audio/csound/files/rtNetBSD.c
new file mode 100644
index 00000000000..abc066f9d89
--- /dev/null
+++ b/audio/csound/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);
+ }
+}
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