2 * QEMU ALSA audio driver
4 * Copyright (c) 2005 Vassili Karpov (malc)
6 * Permission is hereby granted, free of charge, to any person obtaining a copy
7 * of this software and associated documentation files (the "Software"), to deal
8 * in the Software without restriction, including without limitation the rights
9 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
10 * copies of the Software, and to permit persons to whom the Software is
11 * furnished to do so, subject to the following conditions:
13 * The above copyright notice and this permission notice shall be included in
14 * all copies or substantial portions of the Software.
16 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
17 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
18 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
19 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
20 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
21 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
24 #include <alsa/asoundlib.h>
25 #include "qemu-common.h"
26 #include "qemu-char.h"
29 #if QEMU_GNUC_PREREQ(4, 3)
30 #pragma GCC diagnostic ignored "-Waddress"
33 #define AUDIO_CAP "alsa"
34 #include "audio_int.h"
42 typedef struct ALSAVoiceOut {
46 struct pollhlp pollhlp;
49 typedef struct ALSAVoiceIn {
53 struct pollhlp pollhlp;
59 const char *pcm_name_in;
60 const char *pcm_name_out;
61 unsigned int buffer_size_in;
62 unsigned int period_size_in;
63 unsigned int buffer_size_out;
64 unsigned int period_size_out;
65 unsigned int threshold;
67 int buffer_size_in_overridden;
68 int period_size_in_overridden;
70 int buffer_size_out_overridden;
71 int period_size_out_overridden;
74 .buffer_size_out = 1024,
75 .pcm_name_out = "default",
76 .pcm_name_in = "default",
79 struct alsa_params_req {
85 unsigned int buffer_size;
86 unsigned int period_size;
89 struct alsa_params_obt {
94 snd_pcm_uframes_t samples;
97 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
102 AUD_vlog (AUDIO_CAP, fmt, ap);
105 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
108 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
117 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
120 AUD_vlog (AUDIO_CAP, fmt, ap);
123 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
126 static void alsa_fini_poll (struct pollhlp *hlp)
129 struct pollfd *pfds = hlp->pfds;
132 for (i = 0; i < hlp->count; ++i) {
133 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
142 static void alsa_anal_close1 (snd_pcm_t **handlep)
144 int err = snd_pcm_close (*handlep);
146 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
151 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
153 alsa_fini_poll (hlp);
154 alsa_anal_close1 (handlep);
157 static int alsa_recover (snd_pcm_t *handle)
159 int err = snd_pcm_prepare (handle);
161 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
167 static int alsa_resume (snd_pcm_t *handle)
169 int err = snd_pcm_resume (handle);
171 alsa_logerr (err, "Failed to resume handle %p\n", handle);
177 static void alsa_poll_handler (void *opaque)
180 snd_pcm_state_t state;
181 struct pollhlp *hlp = opaque;
182 unsigned short revents;
184 count = poll (hlp->pfds, hlp->count, 0);
186 dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
194 /* XXX: ALSA example uses initial count, not the one returned by
196 err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
197 hlp->count, &revents);
199 alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
203 if (!(revents & POLLOUT)) {
205 dolog ("revents = %d\n", revents);
210 state = snd_pcm_state (hlp->handle);
212 case SND_PCM_STATE_XRUN:
213 alsa_recover (hlp->handle);
216 case SND_PCM_STATE_SUSPENDED:
217 alsa_resume (hlp->handle);
220 case SND_PCM_STATE_PREPARED:
221 audio_run ("alsa run (prepared)");
224 case SND_PCM_STATE_RUNNING:
225 audio_run ("alsa run (running)");
229 dolog ("Unexpected state %d\n", state);
233 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp)
238 count = snd_pcm_poll_descriptors_count (handle);
240 dolog ("Could not initialize poll mode\n"
241 "Invalid number of poll descriptors %d\n", count);
245 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
247 dolog ("Could not initialize poll mode\n");
251 err = snd_pcm_poll_descriptors (handle, pfds, count);
253 alsa_logerr (err, "Could not initialize poll mode\n"
254 "Could not obtain poll descriptors\n");
259 for (i = 0; i < count; ++i) {
260 if (pfds[i].events & POLLIN) {
261 err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
264 if (pfds[i].events & POLLOUT) {
266 dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
268 err = qemu_set_fd_handler (pfds[i].fd, NULL,
269 alsa_poll_handler, hlp);
272 dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
273 pfds[i].events, i, pfds[i].fd, err);
277 dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
278 pfds[i].events, i, pfds[i].fd, err);
281 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
289 hlp->handle = handle;
293 static int alsa_poll_out (HWVoiceOut *hw)
295 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
297 return alsa_poll_helper (alsa->handle, &alsa->pollhlp);
300 static int alsa_poll_in (HWVoiceIn *hw)
302 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
304 return alsa_poll_helper (alsa->handle, &alsa->pollhlp);
307 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
309 return audio_pcm_sw_write (sw, buf, len);
312 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
316 return SND_PCM_FORMAT_S8;
319 return SND_PCM_FORMAT_U8;
322 return SND_PCM_FORMAT_S16_LE;
325 return SND_PCM_FORMAT_U16_LE;
328 return SND_PCM_FORMAT_S32_LE;
331 return SND_PCM_FORMAT_U32_LE;
334 dolog ("Internal logic error: Bad audio format %d\n", fmt);
338 return SND_PCM_FORMAT_U8;
342 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
346 case SND_PCM_FORMAT_S8:
351 case SND_PCM_FORMAT_U8:
356 case SND_PCM_FORMAT_S16_LE:
361 case SND_PCM_FORMAT_U16_LE:
366 case SND_PCM_FORMAT_S16_BE:
371 case SND_PCM_FORMAT_U16_BE:
376 case SND_PCM_FORMAT_S32_LE:
381 case SND_PCM_FORMAT_U32_LE:
386 case SND_PCM_FORMAT_S32_BE:
391 case SND_PCM_FORMAT_U32_BE:
397 dolog ("Unrecognized audio format %d\n", alsafmt);
404 static void alsa_dump_info (struct alsa_params_req *req,
405 struct alsa_params_obt *obt)
407 dolog ("parameter | requested value | obtained value\n");
408 dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
409 dolog ("channels | %10d | %10d\n",
410 req->nchannels, obt->nchannels);
411 dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
412 dolog ("============================================\n");
413 dolog ("requested: buffer size %d period size %d\n",
414 req->buffer_size, req->period_size);
415 dolog ("obtained: samples %ld\n", obt->samples);
418 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
421 snd_pcm_sw_params_t *sw_params;
423 snd_pcm_sw_params_alloca (&sw_params);
425 err = snd_pcm_sw_params_current (handle, sw_params);
427 dolog ("Could not fully initialize DAC\n");
428 alsa_logerr (err, "Failed to get current software parameters\n");
432 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
434 dolog ("Could not fully initialize DAC\n");
435 alsa_logerr (err, "Failed to set software threshold to %ld\n",
440 err = snd_pcm_sw_params (handle, sw_params);
442 dolog ("Could not fully initialize DAC\n");
443 alsa_logerr (err, "Failed to set software parameters\n");
448 static int alsa_open (int in, struct alsa_params_req *req,
449 struct alsa_params_obt *obt, snd_pcm_t **handlep)
452 snd_pcm_hw_params_t *hw_params;
455 unsigned int freq, nchannels;
456 const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
457 snd_pcm_uframes_t obt_buffer_size;
458 const char *typ = in ? "ADC" : "DAC";
459 snd_pcm_format_t obtfmt;
462 nchannels = req->nchannels;
463 size_in_usec = req->size_in_usec;
465 snd_pcm_hw_params_alloca (&hw_params);
470 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
474 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
478 err = snd_pcm_hw_params_any (handle, hw_params);
480 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
484 err = snd_pcm_hw_params_set_access (
487 SND_PCM_ACCESS_RW_INTERLEAVED
490 alsa_logerr2 (err, typ, "Failed to set access type\n");
494 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
495 if (err < 0 && conf.verbose) {
496 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
499 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
501 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
505 err = snd_pcm_hw_params_set_channels_near (
511 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
516 if (nchannels != 1 && nchannels != 2) {
517 alsa_logerr2 (err, typ,
518 "Can not handle obtained number of channels %d\n",
523 if (req->buffer_size) {
528 unsigned int btime = req->buffer_size;
530 err = snd_pcm_hw_params_set_buffer_time_near (
539 snd_pcm_uframes_t bsize = req->buffer_size;
541 err = snd_pcm_hw_params_set_buffer_size_near (
549 alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
550 size_in_usec ? "time" : "size", req->buffer_size);
554 if ((req->override_mask & 2) && (obt - req->buffer_size))
555 dolog ("Requested buffer %s %u was rejected, using %lu\n",
556 size_in_usec ? "time" : "size", req->buffer_size, obt);
559 if (req->period_size) {
564 unsigned int ptime = req->period_size;
566 err = snd_pcm_hw_params_set_period_time_near (
576 snd_pcm_uframes_t psize = req->period_size;
578 err = snd_pcm_hw_params_set_period_size_near (
588 alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
589 size_in_usec ? "time" : "size", req->period_size);
593 if ((req->override_mask & 1) && (obt - req->period_size))
594 dolog ("Requested period %s %u was rejected, using %lu\n",
595 size_in_usec ? "time" : "size", req->period_size, obt);
598 err = snd_pcm_hw_params (handle, hw_params);
600 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
604 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
606 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
610 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
612 alsa_logerr2 (err, typ, "Failed to get format\n");
616 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
617 dolog ("Invalid format was returned %d\n", obtfmt);
621 err = snd_pcm_prepare (handle);
623 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
627 if (!in && conf.threshold) {
628 snd_pcm_uframes_t threshold;
631 bytes_per_sec = freq << (nchannels == 2);
649 threshold = (conf.threshold * bytes_per_sec) / 1000;
650 alsa_set_threshold (handle, threshold);
653 obt->nchannels = nchannels;
655 obt->samples = obt_buffer_size;
660 (obt->fmt != req->fmt ||
661 obt->nchannels != req->nchannels ||
662 obt->freq != req->freq)) {
663 dolog ("Audio paramters for %s\n", typ);
664 alsa_dump_info (req, obt);
668 alsa_dump_info (req, obt);
673 alsa_anal_close1 (&handle);
677 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
679 snd_pcm_sframes_t avail;
681 avail = snd_pcm_avail_update (handle);
683 if (avail == -EPIPE) {
684 if (!alsa_recover (handle)) {
685 avail = snd_pcm_avail_update (handle);
691 "Could not obtain number of available frames\n");
699 static int alsa_run_out (HWVoiceOut *hw)
701 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
702 int rpos, live, decr;
705 struct st_sample *src;
706 snd_pcm_sframes_t avail;
708 live = audio_pcm_hw_get_live_out (hw);
713 avail = alsa_get_avail (alsa->handle);
715 dolog ("Could not get number of available playback frames\n");
719 decr = audio_MIN (live, avail);
723 int left_till_end_samples = hw->samples - rpos;
724 int len = audio_MIN (samples, left_till_end_samples);
725 snd_pcm_sframes_t written;
727 src = hw->mix_buf + rpos;
728 dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
730 hw->clip (dst, src, len);
733 written = snd_pcm_writei (alsa->handle, dst, len);
739 dolog ("Failed to write %d frames (wrote zero)\n", len);
744 if (alsa_recover (alsa->handle)) {
745 alsa_logerr (written, "Failed to write %d frames\n",
750 dolog ("Recovering from playback xrun\n");
755 /* stream is suspended and waiting for an
756 application recovery */
757 if (alsa_resume (alsa->handle)) {
758 alsa_logerr (written, "Failed to write %d frames\n",
763 dolog ("Resuming suspended output stream\n");
771 alsa_logerr (written, "Failed to write %d frames to %p\n",
777 rpos = (rpos + written) % hw->samples;
780 dst = advance (dst, written << hw->info.shift);
790 static void alsa_fini_out (HWVoiceOut *hw)
792 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
794 ldebug ("alsa_fini\n");
795 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
798 qemu_free (alsa->pcm_buf);
799 alsa->pcm_buf = NULL;
803 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
805 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
806 struct alsa_params_req req;
807 struct alsa_params_obt obt;
809 struct audsettings obt_as;
811 req.fmt = aud_to_alsafmt (as->fmt);
813 req.nchannels = as->nchannels;
814 req.period_size = conf.period_size_out;
815 req.buffer_size = conf.buffer_size_out;
816 req.size_in_usec = conf.size_in_usec_out;
818 (conf.period_size_out_overridden ? 1 : 0) |
819 (conf.buffer_size_out_overridden ? 2 : 0);
821 if (alsa_open (0, &req, &obt, &handle)) {
825 obt_as.freq = obt.freq;
826 obt_as.nchannels = obt.nchannels;
827 obt_as.fmt = obt.fmt;
828 obt_as.endianness = obt.endianness;
830 audio_pcm_init_info (&hw->info, &obt_as);
831 hw->samples = obt.samples;
833 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
834 if (!alsa->pcm_buf) {
835 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
836 hw->samples, 1 << hw->info.shift);
837 alsa_anal_close1 (&handle);
841 alsa->handle = handle;
845 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
850 err = snd_pcm_drop (handle);
852 alsa_logerr (err, "Could not stop %s\n", typ);
857 err = snd_pcm_prepare (handle);
859 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
867 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
871 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
874 poll_mode = va_arg (ap, int);
879 ldebug ("enabling voice\n");
880 if (poll_mode && alsa_poll_out (hw)) {
883 hw->poll_mode = poll_mode;
884 return alsa_voice_ctl (alsa->handle, "playback", 0);
887 ldebug ("disabling voice\n");
888 return alsa_voice_ctl (alsa->handle, "playback", 1);
894 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
896 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
897 struct alsa_params_req req;
898 struct alsa_params_obt obt;
900 struct audsettings obt_as;
902 req.fmt = aud_to_alsafmt (as->fmt);
904 req.nchannels = as->nchannels;
905 req.period_size = conf.period_size_in;
906 req.buffer_size = conf.buffer_size_in;
907 req.size_in_usec = conf.size_in_usec_in;
909 (conf.period_size_in_overridden ? 1 : 0) |
910 (conf.buffer_size_in_overridden ? 2 : 0);
912 if (alsa_open (1, &req, &obt, &handle)) {
916 obt_as.freq = obt.freq;
917 obt_as.nchannels = obt.nchannels;
918 obt_as.fmt = obt.fmt;
919 obt_as.endianness = obt.endianness;
921 audio_pcm_init_info (&hw->info, &obt_as);
922 hw->samples = obt.samples;
924 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
925 if (!alsa->pcm_buf) {
926 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
927 hw->samples, 1 << hw->info.shift);
928 alsa_anal_close1 (&handle);
932 alsa->handle = handle;
936 static void alsa_fini_in (HWVoiceIn *hw)
938 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
940 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
943 qemu_free (alsa->pcm_buf);
944 alsa->pcm_buf = NULL;
948 static int alsa_run_in (HWVoiceIn *hw)
950 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
951 int hwshift = hw->info.shift;
953 int live = audio_pcm_hw_get_live_in (hw);
954 int dead = hw->samples - live;
960 { .add = hw->wpos, .len = 0 },
961 { .add = 0, .len = 0 }
963 snd_pcm_sframes_t avail;
964 snd_pcm_uframes_t read_samples = 0;
970 avail = alsa_get_avail (alsa->handle);
972 dolog ("Could not get number of captured frames\n");
977 snd_pcm_state_t state;
979 state = snd_pcm_state (alsa->handle);
981 case SND_PCM_STATE_PREPARED:
984 case SND_PCM_STATE_SUSPENDED:
985 /* stream is suspended and waiting for an application recovery */
986 if (alsa_resume (alsa->handle)) {
987 dolog ("Failed to resume suspended input stream\n");
991 dolog ("Resuming suspended input stream\n");
996 dolog ("No frames available and ALSA state is %d\n", state);
1002 decr = audio_MIN (dead, avail);
1007 if (hw->wpos + decr > hw->samples) {
1008 bufs[0].len = (hw->samples - hw->wpos);
1009 bufs[1].len = (decr - (hw->samples - hw->wpos));
1015 for (i = 0; i < 2; ++i) {
1017 struct st_sample *dst;
1018 snd_pcm_sframes_t nread;
1019 snd_pcm_uframes_t len;
1023 src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1024 dst = hw->conv_buf + bufs[i].add;
1027 nread = snd_pcm_readi (alsa->handle, src, len);
1033 dolog ("Failed to read %ld frames (read zero)\n", len);
1038 if (alsa_recover (alsa->handle)) {
1039 alsa_logerr (nread, "Failed to read %ld frames\n", len);
1043 dolog ("Recovering from capture xrun\n");
1053 "Failed to read %ld frames from %p\n",
1061 hw->conv (dst, src, nread, &nominal_volume);
1063 src = advance (src, nread << hwshift);
1066 read_samples += nread;
1072 hw->wpos = (hw->wpos + read_samples) % hw->samples;
1073 return read_samples;
1076 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
1078 return audio_pcm_sw_read (sw, buf, size);
1081 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1085 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
1088 poll_mode = va_arg (ap, int);
1093 ldebug ("enabling voice\n");
1094 if (poll_mode && alsa_poll_in (hw)) {
1097 hw->poll_mode = poll_mode;
1099 return alsa_voice_ctl (alsa->handle, "capture", 0);
1102 ldebug ("disabling voice\n");
1103 if (hw->poll_mode) {
1105 alsa_fini_poll (&alsa->pollhlp);
1107 return alsa_voice_ctl (alsa->handle, "capture", 1);
1113 static void *alsa_audio_init (void)
1118 static void alsa_audio_fini (void *opaque)
1123 static struct audio_option alsa_options[] = {
1125 .name = "DAC_SIZE_IN_USEC",
1126 .tag = AUD_OPT_BOOL,
1127 .valp = &conf.size_in_usec_out,
1128 .descr = "DAC period/buffer size in microseconds (otherwise in frames)"
1131 .name = "DAC_PERIOD_SIZE",
1133 .valp = &conf.period_size_out,
1134 .descr = "DAC period size (0 to go with system default)",
1135 .overriddenp = &conf.period_size_out_overridden
1138 .name = "DAC_BUFFER_SIZE",
1140 .valp = &conf.buffer_size_out,
1141 .descr = "DAC buffer size (0 to go with system default)",
1142 .overriddenp = &conf.buffer_size_out_overridden
1145 .name = "ADC_SIZE_IN_USEC",
1146 .tag = AUD_OPT_BOOL,
1147 .valp = &conf.size_in_usec_in,
1149 "ADC period/buffer size in microseconds (otherwise in frames)"
1152 .name = "ADC_PERIOD_SIZE",
1154 .valp = &conf.period_size_in,
1155 .descr = "ADC period size (0 to go with system default)",
1156 .overriddenp = &conf.period_size_in_overridden
1159 .name = "ADC_BUFFER_SIZE",
1161 .valp = &conf.buffer_size_in,
1162 .descr = "ADC buffer size (0 to go with system default)",
1163 .overriddenp = &conf.buffer_size_in_overridden
1166 .name = "THRESHOLD",
1168 .valp = &conf.threshold,
1169 .descr = "(undocumented)"
1174 .valp = &conf.pcm_name_out,
1175 .descr = "DAC device name (for instance dmix)"
1180 .valp = &conf.pcm_name_in,
1181 .descr = "ADC device name"
1185 .tag = AUD_OPT_BOOL,
1186 .valp = &conf.verbose,
1187 .descr = "Behave in a more verbose way"
1189 { /* End of list */ }
1192 static struct audio_pcm_ops alsa_pcm_ops = {
1193 .init_out = alsa_init_out,
1194 .fini_out = alsa_fini_out,
1195 .run_out = alsa_run_out,
1196 .write = alsa_write,
1197 .ctl_out = alsa_ctl_out,
1199 .init_in = alsa_init_in,
1200 .fini_in = alsa_fini_in,
1201 .run_in = alsa_run_in,
1203 .ctl_in = alsa_ctl_in,
1206 struct audio_driver alsa_audio_driver = {
1208 .descr = "ALSA http://www.alsa-project.org",
1209 .options = alsa_options,
1210 .init = alsa_audio_init,
1211 .fini = alsa_audio_fini,
1212 .pcm_ops = &alsa_pcm_ops,
1213 .can_be_default = 1,
1214 .max_voices_out = INT_MAX,
1215 .max_voices_in = INT_MAX,
1216 .voice_size_out = sizeof (ALSAVoiceOut),
1217 .voice_size_in = sizeof (ALSAVoiceIn)