Replace gcc variadic macro extension with C99 version (missed one)
[qemu/mini2440.git] / audio / alsaaudio.c
blob2f6c764dca30c7e2215fc5ab1edb89262b40bb7f
1 /*
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
22 * THE SOFTWARE.
24 #include <alsa/asoundlib.h>
25 #include "qemu-common.h"
26 #include "audio.h"
28 #define AUDIO_CAP "alsa"
29 #include "audio_int.h"
31 typedef struct ALSAVoiceOut {
32 HWVoiceOut hw;
33 void *pcm_buf;
34 snd_pcm_t *handle;
35 } ALSAVoiceOut;
37 typedef struct ALSAVoiceIn {
38 HWVoiceIn hw;
39 snd_pcm_t *handle;
40 void *pcm_buf;
41 } ALSAVoiceIn;
43 static struct {
44 int size_in_usec_in;
45 int size_in_usec_out;
46 const char *pcm_name_in;
47 const char *pcm_name_out;
48 unsigned int buffer_size_in;
49 unsigned int period_size_in;
50 unsigned int buffer_size_out;
51 unsigned int period_size_out;
52 unsigned int threshold;
54 int buffer_size_in_overridden;
55 int period_size_in_overridden;
57 int buffer_size_out_overridden;
58 int period_size_out_overridden;
59 int verbose;
60 } conf = {
61 .buffer_size_out = 1024,
62 .pcm_name_out = "default",
63 .pcm_name_in = "default",
66 struct alsa_params_req {
67 int freq;
68 snd_pcm_format_t fmt;
69 int nchannels;
70 int size_in_usec;
71 int override_mask;
72 unsigned int buffer_size;
73 unsigned int period_size;
76 struct alsa_params_obt {
77 int freq;
78 audfmt_e fmt;
79 int endianness;
80 int nchannels;
81 snd_pcm_uframes_t samples;
84 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
86 va_list ap;
88 va_start (ap, fmt);
89 AUD_vlog (AUDIO_CAP, fmt, ap);
90 va_end (ap);
92 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
95 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
96 int err,
97 const char *typ,
98 const char *fmt,
99 ...
102 va_list ap;
104 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
106 va_start (ap, fmt);
107 AUD_vlog (AUDIO_CAP, fmt, ap);
108 va_end (ap);
110 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
113 static void alsa_anal_close (snd_pcm_t **handlep)
115 int err = snd_pcm_close (*handlep);
116 if (err) {
117 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
119 *handlep = NULL;
122 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
124 return audio_pcm_sw_write (sw, buf, len);
127 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
129 switch (fmt) {
130 case AUD_FMT_S8:
131 return SND_PCM_FORMAT_S8;
133 case AUD_FMT_U8:
134 return SND_PCM_FORMAT_U8;
136 case AUD_FMT_S16:
137 return SND_PCM_FORMAT_S16_LE;
139 case AUD_FMT_U16:
140 return SND_PCM_FORMAT_U16_LE;
142 case AUD_FMT_S32:
143 return SND_PCM_FORMAT_S32_LE;
145 case AUD_FMT_U32:
146 return SND_PCM_FORMAT_U32_LE;
148 default:
149 dolog ("Internal logic error: Bad audio format %d\n", fmt);
150 #ifdef DEBUG_AUDIO
151 abort ();
152 #endif
153 return SND_PCM_FORMAT_U8;
157 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
158 int *endianness)
160 switch (alsafmt) {
161 case SND_PCM_FORMAT_S8:
162 *endianness = 0;
163 *fmt = AUD_FMT_S8;
164 break;
166 case SND_PCM_FORMAT_U8:
167 *endianness = 0;
168 *fmt = AUD_FMT_U8;
169 break;
171 case SND_PCM_FORMAT_S16_LE:
172 *endianness = 0;
173 *fmt = AUD_FMT_S16;
174 break;
176 case SND_PCM_FORMAT_U16_LE:
177 *endianness = 0;
178 *fmt = AUD_FMT_U16;
179 break;
181 case SND_PCM_FORMAT_S16_BE:
182 *endianness = 1;
183 *fmt = AUD_FMT_S16;
184 break;
186 case SND_PCM_FORMAT_U16_BE:
187 *endianness = 1;
188 *fmt = AUD_FMT_U16;
189 break;
191 case SND_PCM_FORMAT_S32_LE:
192 *endianness = 0;
193 *fmt = AUD_FMT_S32;
194 break;
196 case SND_PCM_FORMAT_U32_LE:
197 *endianness = 0;
198 *fmt = AUD_FMT_U32;
199 break;
201 case SND_PCM_FORMAT_S32_BE:
202 *endianness = 1;
203 *fmt = AUD_FMT_S32;
204 break;
206 case SND_PCM_FORMAT_U32_BE:
207 *endianness = 1;
208 *fmt = AUD_FMT_U32;
209 break;
211 default:
212 dolog ("Unrecognized audio format %d\n", alsafmt);
213 return -1;
216 return 0;
219 static void alsa_dump_info (struct alsa_params_req *req,
220 struct alsa_params_obt *obt)
222 dolog ("parameter | requested value | obtained value\n");
223 dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
224 dolog ("channels | %10d | %10d\n",
225 req->nchannels, obt->nchannels);
226 dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
227 dolog ("============================================\n");
228 dolog ("requested: buffer size %d period size %d\n",
229 req->buffer_size, req->period_size);
230 dolog ("obtained: samples %ld\n", obt->samples);
233 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
235 int err;
236 snd_pcm_sw_params_t *sw_params;
238 snd_pcm_sw_params_alloca (&sw_params);
240 err = snd_pcm_sw_params_current (handle, sw_params);
241 if (err < 0) {
242 dolog ("Could not fully initialize DAC\n");
243 alsa_logerr (err, "Failed to get current software parameters\n");
244 return;
247 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
248 if (err < 0) {
249 dolog ("Could not fully initialize DAC\n");
250 alsa_logerr (err, "Failed to set software threshold to %ld\n",
251 threshold);
252 return;
255 err = snd_pcm_sw_params (handle, sw_params);
256 if (err < 0) {
257 dolog ("Could not fully initialize DAC\n");
258 alsa_logerr (err, "Failed to set software parameters\n");
259 return;
263 static int alsa_open (int in, struct alsa_params_req *req,
264 struct alsa_params_obt *obt, snd_pcm_t **handlep)
266 snd_pcm_t *handle;
267 snd_pcm_hw_params_t *hw_params;
268 int err;
269 int size_in_usec;
270 unsigned int freq, nchannels;
271 const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
272 snd_pcm_uframes_t obt_buffer_size;
273 const char *typ = in ? "ADC" : "DAC";
274 snd_pcm_format_t obtfmt;
276 freq = req->freq;
277 nchannels = req->nchannels;
278 size_in_usec = req->size_in_usec;
280 snd_pcm_hw_params_alloca (&hw_params);
282 err = snd_pcm_open (
283 &handle,
284 pcm_name,
285 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
286 SND_PCM_NONBLOCK
288 if (err < 0) {
289 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
290 return -1;
293 err = snd_pcm_hw_params_any (handle, hw_params);
294 if (err < 0) {
295 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
296 goto err;
299 err = snd_pcm_hw_params_set_access (
300 handle,
301 hw_params,
302 SND_PCM_ACCESS_RW_INTERLEAVED
304 if (err < 0) {
305 alsa_logerr2 (err, typ, "Failed to set access type\n");
306 goto err;
309 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
310 if (err < 0 && conf.verbose) {
311 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
314 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
315 if (err < 0) {
316 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
317 goto err;
320 err = snd_pcm_hw_params_set_channels_near (
321 handle,
322 hw_params,
323 &nchannels
325 if (err < 0) {
326 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
327 req->nchannels);
328 goto err;
331 if (nchannels != 1 && nchannels != 2) {
332 alsa_logerr2 (err, typ,
333 "Can not handle obtained number of channels %d\n",
334 nchannels);
335 goto err;
338 if (req->buffer_size) {
339 unsigned long obt;
341 if (size_in_usec) {
342 int dir = 0;
343 unsigned int btime = req->buffer_size;
345 err = snd_pcm_hw_params_set_buffer_time_near (
346 handle,
347 hw_params,
348 &btime,
349 &dir
351 obt = btime;
353 else {
354 snd_pcm_uframes_t bsize = req->buffer_size;
356 err = snd_pcm_hw_params_set_buffer_size_near (
357 handle,
358 hw_params,
359 &bsize
361 obt = bsize;
363 if (err < 0) {
364 alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
365 size_in_usec ? "time" : "size", req->buffer_size);
366 goto err;
369 if ((req->override_mask & 2) && (obt - req->buffer_size))
370 dolog ("Requested buffer %s %u was rejected, using %lu\n",
371 size_in_usec ? "time" : "size", req->buffer_size, obt);
374 if (req->period_size) {
375 unsigned long obt;
377 if (size_in_usec) {
378 int dir = 0;
379 unsigned int ptime = req->period_size;
381 err = snd_pcm_hw_params_set_period_time_near (
382 handle,
383 hw_params,
384 &ptime,
385 &dir
387 obt = ptime;
389 else {
390 int dir = 0;
391 snd_pcm_uframes_t psize = req->period_size;
393 err = snd_pcm_hw_params_set_period_size_near (
394 handle,
395 hw_params,
396 &psize,
397 &dir
399 obt = psize;
402 if (err < 0) {
403 alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
404 size_in_usec ? "time" : "size", req->period_size);
405 goto err;
408 if ((req->override_mask & 1) && (obt - req->period_size))
409 dolog ("Requested period %s %u was rejected, using %lu\n",
410 size_in_usec ? "time" : "size", req->period_size, obt);
413 err = snd_pcm_hw_params (handle, hw_params);
414 if (err < 0) {
415 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
416 goto err;
419 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
420 if (err < 0) {
421 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
422 goto err;
425 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
426 if (err < 0) {
427 alsa_logerr2 (err, typ, "Failed to get format\n");
428 goto err;
431 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
432 dolog ("Invalid format was returned %d\n", obtfmt);
433 goto err;
436 err = snd_pcm_prepare (handle);
437 if (err < 0) {
438 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
439 goto err;
442 if (!in && conf.threshold) {
443 snd_pcm_uframes_t threshold;
444 int bytes_per_sec;
446 bytes_per_sec = freq << (nchannels == 2);
448 switch (obt->fmt) {
449 case AUD_FMT_S8:
450 case AUD_FMT_U8:
451 break;
453 case AUD_FMT_S16:
454 case AUD_FMT_U16:
455 bytes_per_sec <<= 1;
456 break;
458 case AUD_FMT_S32:
459 case AUD_FMT_U32:
460 bytes_per_sec <<= 2;
461 break;
464 threshold = (conf.threshold * bytes_per_sec) / 1000;
465 alsa_set_threshold (handle, threshold);
468 obt->nchannels = nchannels;
469 obt->freq = freq;
470 obt->samples = obt_buffer_size;
472 *handlep = handle;
474 if (conf.verbose &&
475 (obt->fmt != req->fmt ||
476 obt->nchannels != req->nchannels ||
477 obt->freq != req->freq)) {
478 dolog ("Audio paramters for %s\n", typ);
479 alsa_dump_info (req, obt);
482 #ifdef DEBUG
483 alsa_dump_info (req, obt);
484 #endif
485 return 0;
487 err:
488 alsa_anal_close (&handle);
489 return -1;
492 static int alsa_recover (snd_pcm_t *handle)
494 int err = snd_pcm_prepare (handle);
495 if (err < 0) {
496 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
497 return -1;
499 return 0;
502 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
504 snd_pcm_sframes_t avail;
506 avail = snd_pcm_avail_update (handle);
507 if (avail < 0) {
508 if (avail == -EPIPE) {
509 if (!alsa_recover (handle)) {
510 avail = snd_pcm_avail_update (handle);
514 if (avail < 0) {
515 alsa_logerr (avail,
516 "Could not obtain number of available frames\n");
517 return -1;
521 return avail;
524 static int alsa_run_out (HWVoiceOut *hw)
526 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
527 int rpos, live, decr;
528 int samples;
529 uint8_t *dst;
530 struct st_sample *src;
531 snd_pcm_sframes_t avail;
533 live = audio_pcm_hw_get_live_out (hw);
534 if (!live) {
535 return 0;
538 avail = alsa_get_avail (alsa->handle);
539 if (avail < 0) {
540 dolog ("Could not get number of available playback frames\n");
541 return 0;
544 decr = audio_MIN (live, avail);
545 samples = decr;
546 rpos = hw->rpos;
547 while (samples) {
548 int left_till_end_samples = hw->samples - rpos;
549 int len = audio_MIN (samples, left_till_end_samples);
550 snd_pcm_sframes_t written;
552 src = hw->mix_buf + rpos;
553 dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
555 hw->clip (dst, src, len);
557 while (len) {
558 written = snd_pcm_writei (alsa->handle, dst, len);
560 if (written <= 0) {
561 switch (written) {
562 case 0:
563 if (conf.verbose) {
564 dolog ("Failed to write %d frames (wrote zero)\n", len);
566 goto exit;
568 case -EPIPE:
569 if (alsa_recover (alsa->handle)) {
570 alsa_logerr (written, "Failed to write %d frames\n",
571 len);
572 goto exit;
574 if (conf.verbose) {
575 dolog ("Recovering from playback xrun\n");
577 continue;
579 case -EAGAIN:
580 goto exit;
582 default:
583 alsa_logerr (written, "Failed to write %d frames to %p\n",
584 len, dst);
585 goto exit;
589 rpos = (rpos + written) % hw->samples;
590 samples -= written;
591 len -= written;
592 dst = advance (dst, written << hw->info.shift);
593 src += written;
597 exit:
598 hw->rpos = rpos;
599 return decr;
602 static void alsa_fini_out (HWVoiceOut *hw)
604 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
606 ldebug ("alsa_fini\n");
607 alsa_anal_close (&alsa->handle);
609 if (alsa->pcm_buf) {
610 qemu_free (alsa->pcm_buf);
611 alsa->pcm_buf = NULL;
615 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
617 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
618 struct alsa_params_req req;
619 struct alsa_params_obt obt;
620 snd_pcm_t *handle;
621 struct audsettings obt_as;
623 req.fmt = aud_to_alsafmt (as->fmt);
624 req.freq = as->freq;
625 req.nchannels = as->nchannels;
626 req.period_size = conf.period_size_out;
627 req.buffer_size = conf.buffer_size_out;
628 req.size_in_usec = conf.size_in_usec_out;
629 req.override_mask = !!conf.period_size_out_overridden
630 | (!!conf.buffer_size_out_overridden << 1);
632 if (alsa_open (0, &req, &obt, &handle)) {
633 return -1;
636 obt_as.freq = obt.freq;
637 obt_as.nchannels = obt.nchannels;
638 obt_as.fmt = obt.fmt;
639 obt_as.endianness = obt.endianness;
641 audio_pcm_init_info (&hw->info, &obt_as);
642 hw->samples = obt.samples;
644 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
645 if (!alsa->pcm_buf) {
646 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
647 hw->samples, 1 << hw->info.shift);
648 alsa_anal_close (&handle);
649 return -1;
652 alsa->handle = handle;
653 return 0;
656 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
658 int err;
660 if (pause) {
661 err = snd_pcm_drop (handle);
662 if (err < 0) {
663 alsa_logerr (err, "Could not stop %s\n", typ);
664 return -1;
667 else {
668 err = snd_pcm_prepare (handle);
669 if (err < 0) {
670 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
671 return -1;
675 return 0;
678 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
680 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
682 switch (cmd) {
683 case VOICE_ENABLE:
684 ldebug ("enabling voice\n");
685 return alsa_voice_ctl (alsa->handle, "playback", 0);
687 case VOICE_DISABLE:
688 ldebug ("disabling voice\n");
689 return alsa_voice_ctl (alsa->handle, "playback", 1);
692 return -1;
695 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
697 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
698 struct alsa_params_req req;
699 struct alsa_params_obt obt;
700 snd_pcm_t *handle;
701 struct audsettings obt_as;
703 req.fmt = aud_to_alsafmt (as->fmt);
704 req.freq = as->freq;
705 req.nchannels = as->nchannels;
706 req.period_size = conf.period_size_in;
707 req.buffer_size = conf.buffer_size_in;
708 req.size_in_usec = conf.size_in_usec_in;
709 req.override_mask = !!conf.period_size_in_overridden
710 | (!!conf.buffer_size_in_overridden << 1);
712 if (alsa_open (1, &req, &obt, &handle)) {
713 return -1;
716 obt_as.freq = obt.freq;
717 obt_as.nchannels = obt.nchannels;
718 obt_as.fmt = obt.fmt;
719 obt_as.endianness = obt.endianness;
721 audio_pcm_init_info (&hw->info, &obt_as);
722 hw->samples = obt.samples;
724 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
725 if (!alsa->pcm_buf) {
726 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
727 hw->samples, 1 << hw->info.shift);
728 alsa_anal_close (&handle);
729 return -1;
732 alsa->handle = handle;
733 return 0;
736 static void alsa_fini_in (HWVoiceIn *hw)
738 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
740 alsa_anal_close (&alsa->handle);
742 if (alsa->pcm_buf) {
743 qemu_free (alsa->pcm_buf);
744 alsa->pcm_buf = NULL;
748 static int alsa_run_in (HWVoiceIn *hw)
750 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
751 int hwshift = hw->info.shift;
752 int i;
753 int live = audio_pcm_hw_get_live_in (hw);
754 int dead = hw->samples - live;
755 int decr;
756 struct {
757 int add;
758 int len;
759 } bufs[2] = {
760 { hw->wpos, 0 },
761 { 0, 0 }
763 snd_pcm_sframes_t avail;
764 snd_pcm_uframes_t read_samples = 0;
766 if (!dead) {
767 return 0;
770 avail = alsa_get_avail (alsa->handle);
771 if (avail < 0) {
772 dolog ("Could not get number of captured frames\n");
773 return 0;
776 if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
777 avail = hw->samples;
780 decr = audio_MIN (dead, avail);
781 if (!decr) {
782 return 0;
785 if (hw->wpos + decr > hw->samples) {
786 bufs[0].len = (hw->samples - hw->wpos);
787 bufs[1].len = (decr - (hw->samples - hw->wpos));
789 else {
790 bufs[0].len = decr;
793 for (i = 0; i < 2; ++i) {
794 void *src;
795 struct st_sample *dst;
796 snd_pcm_sframes_t nread;
797 snd_pcm_uframes_t len;
799 len = bufs[i].len;
801 src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
802 dst = hw->conv_buf + bufs[i].add;
804 while (len) {
805 nread = snd_pcm_readi (alsa->handle, src, len);
807 if (nread <= 0) {
808 switch (nread) {
809 case 0:
810 if (conf.verbose) {
811 dolog ("Failed to read %ld frames (read zero)\n", len);
813 goto exit;
815 case -EPIPE:
816 if (alsa_recover (alsa->handle)) {
817 alsa_logerr (nread, "Failed to read %ld frames\n", len);
818 goto exit;
820 if (conf.verbose) {
821 dolog ("Recovering from capture xrun\n");
823 continue;
825 case -EAGAIN:
826 goto exit;
828 default:
829 alsa_logerr (
830 nread,
831 "Failed to read %ld frames from %p\n",
832 len,
835 goto exit;
839 hw->conv (dst, src, nread, &nominal_volume);
841 src = advance (src, nread << hwshift);
842 dst += nread;
844 read_samples += nread;
845 len -= nread;
849 exit:
850 hw->wpos = (hw->wpos + read_samples) % hw->samples;
851 return read_samples;
854 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
856 return audio_pcm_sw_read (sw, buf, size);
859 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
861 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
863 switch (cmd) {
864 case VOICE_ENABLE:
865 ldebug ("enabling voice\n");
866 return alsa_voice_ctl (alsa->handle, "capture", 0);
868 case VOICE_DISABLE:
869 ldebug ("disabling voice\n");
870 return alsa_voice_ctl (alsa->handle, "capture", 1);
873 return -1;
876 static void *alsa_audio_init (void)
878 return &conf;
881 static void alsa_audio_fini (void *opaque)
883 (void) opaque;
886 static struct audio_option alsa_options[] = {
887 {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
888 "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
889 {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
890 "DAC period size (0 to go with system default)",
891 &conf.period_size_out_overridden, 0},
892 {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
893 "DAC buffer size (0 to go with system default)",
894 &conf.buffer_size_out_overridden, 0},
896 {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
897 "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
898 {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
899 "ADC period size (0 to go with system default)",
900 &conf.period_size_in_overridden, 0},
901 {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
902 "ADC buffer size (0 to go with system default)",
903 &conf.buffer_size_in_overridden, 0},
905 {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
906 "(undocumented)", NULL, 0},
908 {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
909 "DAC device name (for instance dmix)", NULL, 0},
911 {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
912 "ADC device name", NULL, 0},
914 {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
915 "Behave in a more verbose way", NULL, 0},
917 {NULL, 0, NULL, NULL, NULL, 0}
920 static struct audio_pcm_ops alsa_pcm_ops = {
921 alsa_init_out,
922 alsa_fini_out,
923 alsa_run_out,
924 alsa_write,
925 alsa_ctl_out,
927 alsa_init_in,
928 alsa_fini_in,
929 alsa_run_in,
930 alsa_read,
931 alsa_ctl_in
934 struct audio_driver alsa_audio_driver = {
935 INIT_FIELD (name = ) "alsa",
936 INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
937 INIT_FIELD (options = ) alsa_options,
938 INIT_FIELD (init = ) alsa_audio_init,
939 INIT_FIELD (fini = ) alsa_audio_fini,
940 INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
941 INIT_FIELD (can_be_default = ) 1,
942 INIT_FIELD (max_voices_out = ) INT_MAX,
943 INIT_FIELD (max_voices_in = ) INT_MAX,
944 INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
945 INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)