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Diffstat (limited to 'desiredata/src/s_audio_alsa.c')
-rw-r--r--desiredata/src/s_audio_alsa.c554
1 files changed, 0 insertions, 554 deletions
diff --git a/desiredata/src/s_audio_alsa.c b/desiredata/src/s_audio_alsa.c
deleted file mode 100644
index b812fba0..00000000
--- a/desiredata/src/s_audio_alsa.c
+++ /dev/null
@@ -1,554 +0,0 @@
-/* Copyright (c) 1997-2003 Guenter Geiger, Miller Puckette, Larry Troxler, Winfried Ritsch, Karl MacMillan, and others.
-* For information on usage and redistribution, and for a DISCLAIMER OF ALL
-* WARRANTIES, see the file, "LICENSE.txt," in this distribution. */
-
-/* this file inputs and outputs audio using the ALSA API available on linux. */
-
-/* support for ALSA pcmv2 api by Karl MacMillan<karlmac@peabody.jhu.edu> */
-/* support for ALSA MMAP noninterleaved by Winfried Ritsch, IEM */
-
-#include <alsa/asoundlib.h>
-
-#include "desire.h"
-using namespace desire;
-#include <errno.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <sched.h>
-#include <sys/mman.h>
-#include "s_audio_alsa.h"
-
-/* Defines */
-#define DEBUG(x) x
-#define DEBUG2(x) {x;}
-
-/* needed for alsa 0.9 compatibility: */
-#if (SND_LIB_MAJOR < 1)
-#define ALSAAPI9
-#endif
-
-//static void alsa_close_audio();
-static void alsa_checkiosync();
-static void alsa_numbertoname(int iodev, char *devname, int nchar);
-static int alsa_jittermax;
-static void alsa_close_audio();
-#define ALSA_DEFJITTERMAX 3
-
- /* don't assume we can turn all 31 bits when doing float-to-fix;
- otherwise some audio drivers (e.g. Midiman/ALSA) wrap around. */
-#define FMAX 0x7ffff000
-#define CLIP32(x) (((x)>FMAX)?FMAX:((x) < -FMAX)?-FMAX:(x))
-
-static char *alsa_snd_buf;
-static int alsa_snd_bufsize;
-static int alsa_buf_samps;
-static snd_pcm_status_t *alsa_status;
-static int alsa_usemmap;
-t_alsa alsai,alsao;
-
-static void check_error(int err, const char *why) {if (err<0) error("%s: %s", why, snd_strerror(err));}
-
-static int alsaio_canmmap(t_alsa_dev *dev) {
- snd_pcm_hw_params_t *hw_params;
- int err1, err2;
- snd_pcm_hw_params_alloca(&hw_params);
- err1 = snd_pcm_hw_params_any(dev->a_handle, hw_params);
- if (err1 < 0) {
- check_error(err1,"Broken configuration: no configurations available");
- return 0;
- }
- err1 = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
- if (err1 < 0) {
- err2 = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
- } else err2 = -1;
-#if 0
- error("err 1 %d (%s), err2 %d (%s)", err1, snd_strerror(err1), err2, snd_strerror(err2));
-#endif
- return err1<0 && err2>=0;
-}
-
-static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, int nfrags, int frag_size) {
- int bufsizeforthis, err;
- snd_pcm_hw_params_t* hw_params;
- unsigned int tmp_uint;
- snd_pcm_uframes_t tmp_snd_pcm_uframes;
- if (sys_verbose) {
- if (out) post("configuring sound output...");
- else post("configuring sound input...");
- }
- /* set hardware parameters... */
- snd_pcm_hw_params_alloca(&hw_params);
- /* get the default params */
- err = snd_pcm_hw_params_any(dev->a_handle, hw_params);
- check_error(err, "snd_pcm_hw_params_any");
- /* try to set interleaved access */
- err = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
- if (err < 0) return -1;
- check_error(err, "snd_pcm_hw_params_set_access");
- /* Try to set 32 bit format first */
- err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params, SND_PCM_FORMAT_S32);
- if (err<0) {
- error("PD-ALSA: 32 bit format not available - using 16");
- err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,SND_PCM_FORMAT_S16);
- check_error(err, "snd_pcm_hw_params_set_format");
- dev->a_sampwidth = 2;
- } else dev->a_sampwidth = 4;
- if (sys_verbose) post("Sample width set to %d bytes", dev->a_sampwidth);
- /* set the subformat */
- err = snd_pcm_hw_params_set_subformat(dev->a_handle, hw_params, SND_PCM_SUBFORMAT_STD);
- check_error(err, "snd_pcm_hw_params_set_subformat");
- /* set the number of channels */
- tmp_uint = *channels;
- err = snd_pcm_hw_params_set_channels_min(dev->a_handle, hw_params, &tmp_uint);
- check_error(err, "snd_pcm_hw_params_set_channels");
- if (tmp_uint != (unsigned)*channels) post("ALSA: set input channels to %d", tmp_uint);
- *channels = tmp_uint;
- dev->a_channels = *channels;
- /* set the sampling rate */
- err = snd_pcm_hw_params_set_rate_min(dev->a_handle, hw_params, (unsigned int *)rate, 0);
- check_error(err, "snd_pcm_hw_params_set_rate_min (input)");
-#if 0
- err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir);
- post("input sample rate %d", err);
-#endif
- /* set the period - ie frag size */
- /* LATER try this to get a recommended period size...
- right now, it trips an assertion failure in ALSA lib */
-#ifdef ALSAAPI9
- err = snd_pcm_hw_params_set_period_size_near(dev->a_handle, hw_params, (snd_pcm_uframes_t)frag_size, 0);
-#else
- tmp_snd_pcm_uframes = frag_size;
- err = snd_pcm_hw_params_set_period_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes, 0);
-#endif
- check_error(err, "snd_pcm_hw_params_set_period_size_near (input)");
- /* set the number of periods - ie numfrags */
-#ifdef ALSAAPI9
- err = snd_pcm_hw_params_set_periods_near(dev->a_handle, hw_params, nfrags, 0);
-#else
- tmp_uint = nfrags;
- err = snd_pcm_hw_params_set_periods_near(dev->a_handle, hw_params, &tmp_uint, 0);
-#endif
- check_error(err, "snd_pcm_hw_params_set_periods_near (input)");
- /* set the buffer size */
-#ifdef ALSAAPI9
- err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle, hw_params, nfrags * frag_size);
-#else
- tmp_snd_pcm_uframes = nfrags * frag_size;
- err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes);
-#endif
- check_error(err, "snd_pcm_hw_params_set_buffer_size_near (input)");
- err = snd_pcm_hw_params(dev->a_handle, hw_params);
- check_error(err, "snd_pcm_hw_params (input)");
- /* set up the buffer */
- bufsizeforthis = sys_dacblocksize * dev->a_sampwidth * *channels;
- if (alsa_snd_buf) {
- if (alsa_snd_bufsize < bufsizeforthis) {
- if (!(alsa_snd_buf = (char *)realloc(alsa_snd_buf, bufsizeforthis))) {error("out of memory"); return 0;}
- memset(alsa_snd_buf, 0, bufsizeforthis);
- alsa_snd_bufsize = bufsizeforthis;
- }
- } else {
- if (!(alsa_snd_buf = (char *)malloc(bufsizeforthis))) {error("out of memory"); return 0;}
- memset(alsa_snd_buf, 0, bufsizeforthis);
- alsa_snd_bufsize = bufsizeforthis;
- }
- return 1;
-}
-
-/* return 0 on success */
-static int alsa_open_audio(
-int naudioindev, int * audioindev, int nchindev, int * chindev,
-int naudiooutdev, int *audiooutdev, int nchoutdev, int *choutdev, int rate, int dummy) {
- int err, inchans = 0, outchans = 0;
- char devname[512];
- int frag_size = (sys_blocksize ? sys_blocksize : ALSA_DEFFRAGSIZE);
- int nfrags, i;
- nfrags = int(sys_schedadvance * (float)rate / (1e6 * frag_size));
- /* save our belief as to ALSA's buffer size for later */
- alsa_buf_samps = nfrags * frag_size;
- alsai.ndev = alsao.ndev = 0;
- alsa_jittermax = ALSA_DEFJITTERMAX;
- if (sys_verbose) post("audio buffer set to %d", (int)(0.001 * sys_schedadvance));
- for (int i=0; i<naudioindev; i++) {
- alsa_numbertoname(audioindev[i], devname, 512);
- err = snd_pcm_open(&alsai.dev[alsai.ndev].a_handle, devname, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
- check_error(err, "snd_pcm_open (input)");
- if (err<0) continue;
- alsai.dev[alsai.ndev].a_devno = audioindev[i];
- snd_pcm_nonblock(alsai.dev[alsai.ndev].a_handle, 1);
- if (sys_verbose) post("opened input device name %s", devname);
- alsai.ndev++;
- }
- for (int i=0; i<naudiooutdev; i++) {
- alsa_numbertoname(audiooutdev[i], devname, 512);
- err = snd_pcm_open(&alsao.dev[alsao.ndev].a_handle, devname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
- check_error(err, "snd_pcm_open (output)");
- if (err<0) continue;
- alsao.dev[alsao.ndev].a_devno = audiooutdev[i];
- snd_pcm_nonblock(alsao.dev[alsao.ndev].a_handle, 1);
- alsao.ndev++;
- }
- if (!alsai.ndev && !alsao.ndev) goto blewit;
- /* If all the open devices support mmap_noninterleaved, let's call Wini's code in s_audio_alsamm.c */
- alsa_usemmap = 1;
- for (int i=0; i<alsai.ndev; i++) if (!alsaio_canmmap(&alsai.dev[i])) alsa_usemmap = 0;
- for (int i=0; i<alsao.ndev; i++) if (!alsaio_canmmap(&alsao.dev[i])) alsa_usemmap = 0;
- if (alsa_usemmap) {
- post("using mmap audio interface");
- if (alsamm_open_audio(rate)) goto blewit; else return 0;
- }
- for (int i=0; i<alsai.ndev; i++) {
- int channels = chindev[i];
- if (alsaio_setup(&alsai.dev[i], 0, &channels, &rate, nfrags, frag_size) < 0) goto blewit;
- inchans += channels;
- }
- for (int i=0; i<alsao.ndev; i++) {
- int channels = choutdev[i];
- if (alsaio_setup(&alsao.dev[i], 1, &channels, &rate, nfrags, frag_size) < 0) goto blewit;
- outchans += channels;
- }
- if (!inchans && !outchans) goto blewit;
- for (int i=0; i<alsai.ndev; i++) snd_pcm_prepare(alsai.dev[i].a_handle);
- for (int i=0; i<alsao.ndev; i++) snd_pcm_prepare(alsao.dev[i].a_handle);
- /* if duplex we can link the channels so they start together; however j is not used, so wtf */
- for (int i=0; i<alsai.ndev; i++) {
- //for (int j=0; j<alsao.ndev; j++) {
- if (alsai.dev[i].a_devno == alsao.dev[i].a_devno) {
- snd_pcm_link(alsai.dev[i].a_handle,alsao.dev[i].a_handle);
- }
- //}
- }
- /* allocate the status variables */
- if (!alsa_status) {
- err = snd_pcm_status_malloc(&alsa_status);
- check_error(err, "snd_pcm_status_malloc");
- }
- /* fill the buffer with silence */
- memset(alsa_snd_buf, 0, alsa_snd_bufsize);
- if (outchans) {
- i = (frag_size * nfrags)/sys_dacblocksize + 1;
- while (i--) {
- for (int i=0; i<alsao.ndev; i++)
- snd_pcm_writei(alsao.dev[i].a_handle, alsa_snd_buf, sys_dacblocksize);
- }
- } else if (inchans) {
- for (int i=0; i<alsai.ndev; i++)
- if ((err = snd_pcm_start(alsai.dev[i].a_handle)) < 0) check_error(err, "input start failed");
- }
- return 0;
-blewit:
- sys_inchannels = 0;
- sys_outchannels = 0;
- alsa_close_audio();
- return 1;
-}
-
-static void alsa_close_audio() {
- int err;
- if (alsa_usemmap) {alsamm_close_audio(); return;}
- for (int i=0; i<alsai.ndev; i++) {
- err = snd_pcm_close(alsai.dev[i].a_handle);
- check_error(err, "snd_pcm_close (input)");
- }
- for (int i=0; i<alsao.ndev; i++) {
- err = snd_pcm_close(alsao.dev[i].a_handle);
- check_error(err, "snd_pcm_close (output)");
- }
- alsai.ndev = alsao.ndev = 0;
-}
-
-int alsa_send_dacs() {
-#ifdef DEBUG_ALSA_XFER
- static int xferno = 0;
- static int callno = 0;
-#endif
- static double timenow;
- double timelast;
- t_sample *fp1, *fp2;
- int j, k, iodev, result, ch;
- int chansintogo, chansouttogo;
- unsigned int transfersize;
- if (alsa_usemmap) return alsamm_send_dacs();
- if (!alsai.ndev && !alsao.ndev) return SENDDACS_NO;
- chansintogo = sys_inchannels;
- chansouttogo = sys_outchannels;
- transfersize = sys_dacblocksize;
- timelast = timenow;
- timenow = sys_getrealtime();
-#ifdef DEBUG_ALSA_XFER
- if (timenow - timelast > 0.050) post("(%d)", int(1000 * (timenow - timelast)));
- callno++;
-#endif
- alsa_checkiosync(); /* check I/O are in sync and data not late */
- for (int i=0; i<alsai.ndev; i++) {
- snd_pcm_status(alsai.dev[i].a_handle, alsa_status);
- if (snd_pcm_status_get_avail(alsa_status) < transfersize) return SENDDACS_NO;
- }
- for (int i=0; i<alsao.ndev; i++) {
- snd_pcm_status(alsao.dev[i].a_handle, alsa_status);
- if (snd_pcm_status_get_avail(alsa_status) < transfersize) return SENDDACS_NO;
- }
- /* do output */
- fp1 = sys_soundout; ch = 0;
- for (int iodev=0; iodev<alsao.ndev; iodev++) {
- int thisdevchans = alsao.dev[iodev].a_channels;
- int chans = min(chansouttogo,thisdevchans);
- chansouttogo -= chans;
- int i;
- if (alsao.dev[iodev].a_sampwidth == 4) {
- for (i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
- fp2 = fp1;
- for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans, fp2++) {
- float s1 = *fp2 * INT32_MAX;
- ((t_alsa_sample32 *)alsa_snd_buf)[j] = CLIP32(int(s1));
- }
- }
- for (; i<thisdevchans; i++, ch++)
- for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans) ((t_alsa_sample32 *)alsa_snd_buf)[j] = 0;
- } else {
- for (i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
- fp2=fp1;
- for (int j=ch, k=sys_dacblocksize; k--; j += thisdevchans, fp2++) {
- int s = int(*fp2 * 32767.);
- if (s > 32767) s = 32767; else if (s < -32767) s = -32767;
- ((t_alsa_sample16 *)alsa_snd_buf)[j] = s;
- }
- }
- for (; i < thisdevchans; i++, ch++)
- for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans) ((t_alsa_sample16 *)alsa_snd_buf)[j] = 0;
- }
- result = snd_pcm_writei(alsao.dev[iodev].a_handle, alsa_snd_buf, transfersize);
- if (result != (int)transfersize) {
- #ifdef DEBUG_ALSA_XFER
- if (result >= 0 || errno == EAGAIN) post("ALSA: write returned %d of %d", result, transfersize);
- else error("ALSA: write: %s", snd_strerror(errno));
- post("inputcount %d, outputcount %d, outbufsize %d",
- inputcount, outputcount, (ALSA_EXTRABUFFER + sys_advance_samples) * alsao.dev[iodev].a_sampwidth * outchannels);
- #endif
- sys_log_error(ERR_DACSLEPT);
- return SENDDACS_NO;
- }
- /* zero out the output buffer */
- memset(sys_soundout, 0, sys_dacblocksize * sizeof(*sys_soundout) * sys_outchannels);
- if (sys_getrealtime() - timenow > 0.002) {
- #ifdef DEBUG_ALSA_XFER
- post("output %d took %d msec", callno, int(1000 * (timenow - timelast)));
- #endif
- timenow = sys_getrealtime();
- sys_log_error(ERR_DACSLEPT);
- }
- }
- /* do input */
- for (iodev = 0, fp1 = sys_soundin, ch = 0; iodev < alsai.ndev; iodev++) {
- int thisdevchans = alsai.dev[iodev].a_channels;
- int chans = (chansintogo < thisdevchans ? chansintogo : thisdevchans);
- chansouttogo -= chans;
- result = snd_pcm_readi(alsai.dev[iodev].a_handle, alsa_snd_buf, transfersize);
- if (result < (int)transfersize) {
-#ifdef DEBUG_ALSA_XFER
- if (result<0) error("snd_pcm_read %d %d: %s", callno, xferno, snd_strerror(errno));
- else post("snd_pcm_read %d %d returned only %d", callno, xferno, result);
- post("inputcount %d, outputcount %d, inbufsize %d",
- inputcount, outputcount, (ALSA_EXTRABUFFER + sys_advance_samples) * alsai.dev[iodev].a_sampwidth * inchannels);
-#endif
- sys_log_error(ERR_ADCSLEPT);
- return SENDDACS_NO;
- }
- if (alsai.dev[iodev].a_sampwidth == 4) {
- for (int i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
- for (j = ch, k = sys_dacblocksize, fp2 = fp1; k--; j += thisdevchans, fp2++)
- *fp2 = (float) ((t_alsa_sample32 *)alsa_snd_buf)[j] * (1./ INT32_MAX);
- }
- } else {
- for (int i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
- for (j = ch, k = sys_dacblocksize, fp2 = fp1; k--; j += thisdevchans, fp2++)
- *fp2 = (float) ((t_alsa_sample16 *)alsa_snd_buf)[j] * 3.051850e-05;
- }
- }
- }
-#ifdef DEBUG_ALSA_XFER
- xferno++;
-#endif
- if (sys_getrealtime() - timenow > 0.002) {
-#ifdef DEBUG_ALSA_XFER
- post("routine took %d msec", int(1000 * (sys_getrealtime() - timenow)));
-#endif
- sys_log_error(ERR_ADCSLEPT);
- }
- return SENDDACS_YES;
-}
-
-void alsa_printstate() {
- int result, i=0;
- snd_pcm_sframes_t indelay, outdelay;
- if (sys_audioapi != API_ALSA) {
- error("restart-audio: implemented for ALSA only.");
- return;
- }
- if (sys_inchannels) {
- result = snd_pcm_delay(alsai.dev[i].a_handle, &indelay);
- if (result<0) error("snd_pcm_delay 1 failed"); else post( "in delay %d", int( indelay));
- }
- if (sys_outchannels) {
- result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
- if (result<0) error("snd_pcm_delay 2 failed"); else post("out delay %d", int(outdelay));
- }
- post("sum %ld (%ld mod 64)", indelay + outdelay, (indelay+outdelay)%64);
- post("buf samples %d", alsa_buf_samps);
-}
-
-
-void alsa_putzeros(int iodev, int n) {
- memset(alsa_snd_buf, 0, alsao.dev[iodev].a_sampwidth * sys_dacblocksize * alsao.dev[iodev].a_channels);
- for (int i=0; i<n; i++) snd_pcm_writei(alsao.dev[iodev].a_handle, alsa_snd_buf, sys_dacblocksize);
-}
-
-void alsa_getzeros(int iodev, int n) {
- for (int i=0; i<n; i++) snd_pcm_readi(alsai.dev[iodev].a_handle, alsa_snd_buf, sys_dacblocksize);
-}
-
-/* call this only if both input and output are open */
-static void alsa_checkiosync() {
- int result, giveup = 1000, alreadylogged = 0;
- snd_pcm_sframes_t minphase, maxphase, thisphase, outdelay;
- while (1) {
- if (giveup-- <= 0) {post("tried but couldn't sync A/D/A"); alsa_jittermax += 1; return;}
- minphase = 0x7fffffff;
- maxphase = -0x7fffffff;
- for (int i=0; i<alsao.ndev; i++) {
- result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
- if (result < 0) {
- snd_pcm_prepare(alsao.dev[i].a_handle);
- result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
- }
- if (result<0) {
- error("output snd_pcm_delay failed: %s", snd_strerror(result));
- if (snd_pcm_status(alsao.dev[i].a_handle, alsa_status)<0) error("output snd_pcm_status failed");
- else post("astate %d", snd_pcm_status_get_state(alsa_status));
- return;
- }
- thisphase = alsa_buf_samps - outdelay;
- if (thisphase < minphase) minphase = thisphase;
- if (thisphase > maxphase) maxphase = thisphase;
- if (outdelay < 0)
- sys_log_error(ERR_DATALATE), alreadylogged = 1;
- }
- for (int i=0; i<alsai.ndev; i++) {
- result = snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
- if (result < 0) {
- snd_pcm_prepare(alsai.dev[i].a_handle);
- result = snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
- }
- if (result < 0) {
- error("output snd_pcm_delay failed: %s", snd_strerror(result));
- if (snd_pcm_status(alsao.dev[i].a_handle, alsa_status) < 0) error("output snd_pcm_status failed");
- else post("astate %d", snd_pcm_status_get_state(alsa_status));
- return;
- }
- if (thisphase < minphase) minphase = thisphase;
- if (thisphase > maxphase) maxphase = thisphase;
- }
- /* the "correct" position is for all the phases to be exactly equal;
- but since we only make corrections sys_dacblocksize samples at a time,
- we just ask that the spread be not more than 3/4 of a block. */
- if (maxphase <= minphase + (alsa_jittermax * (sys_dacblocksize / 4))) break;
- if (!alreadylogged) sys_log_error(ERR_RESYNC), alreadylogged = 1;
- for (int i=0; i<alsao.ndev; i++) {
- result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
- if (result < 0) break;
- thisphase = alsa_buf_samps - outdelay;
- if (thisphase > minphase + sys_dacblocksize) {
- alsa_putzeros(i,1);
-#if DEBUGSYNC
- post("putz %d %d", (int)thisphase, (int)minphase);
-#endif
- }
- }
- for (int i=0; i<alsai.ndev; i++) {
- result = snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
- if (result < 0) break;
- if (thisphase > minphase + sys_dacblocksize) {
- alsa_getzeros(i, 1);
-#if DEBUGSYNC
- post("getz %d %d", (int)thisphase, (int)minphase);
-#endif
- }
- }
- }
-#if DEBUGSYNC
- if (alreadylogged) post("done");
-#endif
-}
-
-static int alsa_nnames = 0;
-static char **alsa_names = 0;
-
-void alsa_adddev(char *name) {
- if (alsa_nnames) alsa_names = (char **)t_resizebytes(alsa_names, alsa_nnames*sizeof(char *), (alsa_nnames+1)*sizeof(char *));
- else alsa_names = (char **)t_getbytes(sizeof(char *));
- alsa_names[alsa_nnames] = gensym(name)->s_name;
- alsa_nnames++;
-}
-
-static void alsa_numbertoname(int devno, char *devname, int nchar) {
- int ndev = 0, cardno = -1;
- while (!snd_card_next(&cardno) && cardno >= 0) ndev++;
- if (devno < 2*ndev) {
- if (devno & 1) snprintf(devname, nchar, "plughw:%d", devno/2);
- else snprintf(devname, nchar, "hw:%d", devno/2);
- } else if (devno <2*ndev + alsa_nnames)
- snprintf(devname, nchar, "%s", alsa_names[devno - 2*ndev]);
- else snprintf(devname, nchar, "???");
-}
-
-/* For each hardware card found, we list two devices, the "hard" and
- "plug" one. The card scan is derived from portaudio code. */
-static void alsa_getdevs(char *indevlist, int *nindevs, char *outdevlist, int *noutdevs, int *canmulti, int maxndev, int devdescsize) {
- int ndev = 0, cardno = -1, i, j;
- *canmulti = 2; /* supports multiple devices */
- while (!snd_card_next(&cardno) && cardno >= 0) {
- snd_ctl_t *ctl;
- snd_ctl_card_info_t *info;
- char devname[80];
- char *desc;
- if (2*ndev + 2 > maxndev) break;
- /* apparently, "cardno" is just a counter; but check that here */
- if (ndev != cardno) post("oops: ALSA cards not reported in order?");
- sprintf(devname, "hw:%d", cardno);
- /* post("try %s..", devname); */
- if (snd_ctl_open(&ctl, devname, 0) >= 0) {
- snd_ctl_card_info_malloc(&info);
- snd_ctl_card_info(ctl, info);
- desc = strdup(snd_ctl_card_info_get_name(info));
- snd_ctl_card_info_free(info);
- } else {
- error("ALSA card scan error");
- desc = strdup("???");
- }
- sprintf(indevlist + 2*ndev * devdescsize, "%s (hardware)", desc);
- sprintf(indevlist + (2*ndev+1) * devdescsize, "%s (plug-in)", desc);
- sprintf(outdevlist + 2*ndev * devdescsize, "%s (hardware)", desc);
- sprintf(outdevlist + (2*ndev+1) * devdescsize, "%s (plug-in)", desc);
- ndev++;
- free(desc);
- }
- for (i=0, j=2*ndev; i<alsa_nnames; i++, j++) {
- if (j >= maxndev) break;
- snprintf(indevlist + j * devdescsize, devdescsize, "%s", alsa_names[i]);
- }
- *nindevs = *noutdevs = j;
-}
-
-struct t_audioapi alsa_api = {
- alsa_open_audio,
- alsa_close_audio,
- alsa_send_dacs,
- alsa_getdevs,
-};
generated by cgit v1.2.1 (git 2.18.0) at 2024-07-01 06:10:38 +0000