Merge remote-tracking branch 'remotes/ericb/tags/pull-nbd-2021-01-20' into staging
[qemu.git] / audio / alsaaudio.c
1 /*
2 * QEMU ALSA audio driver
3 *
4 * Copyright (c) 2005 Vassili Karpov (malc)
5 *
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:
12 *
13 * The above copyright notice and this permission notice shall be included in
14 * all copies or substantial portions of the Software.
15 *
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.
23 */
24
25 #include "qemu/osdep.h"
26 #include <alsa/asoundlib.h>
27 #include "qemu/main-loop.h"
28 #include "qemu/module.h"
29 #include "audio.h"
30 #include "trace.h"
31
32 #pragma GCC diagnostic ignored "-Waddress"
33
34 #define AUDIO_CAP "alsa"
35 #include "audio_int.h"
36
37 struct pollhlp {
38 snd_pcm_t *handle;
39 struct pollfd *pfds;
40 int count;
41 int mask;
42 AudioState *s;
43 };
44
45 typedef struct ALSAVoiceOut {
46 HWVoiceOut hw;
47 snd_pcm_t *handle;
48 struct pollhlp pollhlp;
49 Audiodev *dev;
50 } ALSAVoiceOut;
51
52 typedef struct ALSAVoiceIn {
53 HWVoiceIn hw;
54 snd_pcm_t *handle;
55 struct pollhlp pollhlp;
56 Audiodev *dev;
57 } ALSAVoiceIn;
58
59 struct alsa_params_req {
60 int freq;
61 snd_pcm_format_t fmt;
62 int nchannels;
63 };
64
65 struct alsa_params_obt {
66 int freq;
67 AudioFormat fmt;
68 int endianness;
69 int nchannels;
70 snd_pcm_uframes_t samples;
71 };
72
73 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
74 {
75 va_list ap;
76
77 va_start (ap, fmt);
78 AUD_vlog (AUDIO_CAP, fmt, ap);
79 va_end (ap);
80
81 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
82 }
83
84 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
85 int err,
86 const char *typ,
87 const char *fmt,
88 ...
89 )
90 {
91 va_list ap;
92
93 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
94
95 va_start (ap, fmt);
96 AUD_vlog (AUDIO_CAP, fmt, ap);
97 va_end (ap);
98
99 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
100 }
101
102 static void alsa_fini_poll (struct pollhlp *hlp)
103 {
104 int i;
105 struct pollfd *pfds = hlp->pfds;
106
107 if (pfds) {
108 for (i = 0; i < hlp->count; ++i) {
109 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
110 }
111 g_free (pfds);
112 }
113 hlp->pfds = NULL;
114 hlp->count = 0;
115 hlp->handle = NULL;
116 }
117
118 static void alsa_anal_close1 (snd_pcm_t **handlep)
119 {
120 int err = snd_pcm_close (*handlep);
121 if (err) {
122 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
123 }
124 *handlep = NULL;
125 }
126
127 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
128 {
129 alsa_fini_poll (hlp);
130 alsa_anal_close1 (handlep);
131 }
132
133 static int alsa_recover (snd_pcm_t *handle)
134 {
135 int err = snd_pcm_prepare (handle);
136 if (err < 0) {
137 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
138 return -1;
139 }
140 return 0;
141 }
142
143 static int alsa_resume (snd_pcm_t *handle)
144 {
145 int err = snd_pcm_resume (handle);
146 if (err < 0) {
147 alsa_logerr (err, "Failed to resume handle %p\n", handle);
148 return -1;
149 }
150 return 0;
151 }
152
153 static void alsa_poll_handler (void *opaque)
154 {
155 int err, count;
156 snd_pcm_state_t state;
157 struct pollhlp *hlp = opaque;
158 unsigned short revents;
159
160 count = poll (hlp->pfds, hlp->count, 0);
161 if (count < 0) {
162 dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
163 return;
164 }
165
166 if (!count) {
167 return;
168 }
169
170 /* XXX: ALSA example uses initial count, not the one returned by
171 poll, correct? */
172 err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
173 hlp->count, &revents);
174 if (err < 0) {
175 alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
176 return;
177 }
178
179 if (!(revents & hlp->mask)) {
180 trace_alsa_revents(revents);
181 return;
182 }
183
184 state = snd_pcm_state (hlp->handle);
185 switch (state) {
186 case SND_PCM_STATE_SETUP:
187 alsa_recover (hlp->handle);
188 break;
189
190 case SND_PCM_STATE_XRUN:
191 alsa_recover (hlp->handle);
192 break;
193
194 case SND_PCM_STATE_SUSPENDED:
195 alsa_resume (hlp->handle);
196 break;
197
198 case SND_PCM_STATE_PREPARED:
199 audio_run(hlp->s, "alsa run (prepared)");
200 break;
201
202 case SND_PCM_STATE_RUNNING:
203 audio_run(hlp->s, "alsa run (running)");
204 break;
205
206 default:
207 dolog ("Unexpected state %d\n", state);
208 }
209 }
210
211 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
212 {
213 int i, count, err;
214 struct pollfd *pfds;
215
216 count = snd_pcm_poll_descriptors_count (handle);
217 if (count <= 0) {
218 dolog ("Could not initialize poll mode\n"
219 "Invalid number of poll descriptors %d\n", count);
220 return -1;
221 }
222
223 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
224 if (!pfds) {
225 dolog ("Could not initialize poll mode\n");
226 return -1;
227 }
228
229 err = snd_pcm_poll_descriptors (handle, pfds, count);
230 if (err < 0) {
231 alsa_logerr (err, "Could not initialize poll mode\n"
232 "Could not obtain poll descriptors\n");
233 g_free (pfds);
234 return -1;
235 }
236
237 for (i = 0; i < count; ++i) {
238 if (pfds[i].events & POLLIN) {
239 qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
240 }
241 if (pfds[i].events & POLLOUT) {
242 trace_alsa_pollout(i, pfds[i].fd);
243 qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
244 }
245 trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
246
247 }
248 hlp->pfds = pfds;
249 hlp->count = count;
250 hlp->handle = handle;
251 hlp->mask = mask;
252 return 0;
253 }
254
255 static int alsa_poll_out (HWVoiceOut *hw)
256 {
257 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
258
259 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
260 }
261
262 static int alsa_poll_in (HWVoiceIn *hw)
263 {
264 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
265
266 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
267 }
268
269 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
270 {
271 switch (fmt) {
272 case AUDIO_FORMAT_S8:
273 return SND_PCM_FORMAT_S8;
274
275 case AUDIO_FORMAT_U8:
276 return SND_PCM_FORMAT_U8;
277
278 case AUDIO_FORMAT_S16:
279 if (endianness) {
280 return SND_PCM_FORMAT_S16_BE;
281 } else {
282 return SND_PCM_FORMAT_S16_LE;
283 }
284
285 case AUDIO_FORMAT_U16:
286 if (endianness) {
287 return SND_PCM_FORMAT_U16_BE;
288 } else {
289 return SND_PCM_FORMAT_U16_LE;
290 }
291
292 case AUDIO_FORMAT_S32:
293 if (endianness) {
294 return SND_PCM_FORMAT_S32_BE;
295 } else {
296 return SND_PCM_FORMAT_S32_LE;
297 }
298
299 case AUDIO_FORMAT_U32:
300 if (endianness) {
301 return SND_PCM_FORMAT_U32_BE;
302 } else {
303 return SND_PCM_FORMAT_U32_LE;
304 }
305
306 case AUDIO_FORMAT_F32:
307 if (endianness) {
308 return SND_PCM_FORMAT_FLOAT_BE;
309 } else {
310 return SND_PCM_FORMAT_FLOAT_LE;
311 }
312
313 default:
314 dolog ("Internal logic error: Bad audio format %d\n", fmt);
315 #ifdef DEBUG_AUDIO
316 abort ();
317 #endif
318 return SND_PCM_FORMAT_U8;
319 }
320 }
321
322 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
323 int *endianness)
324 {
325 switch (alsafmt) {
326 case SND_PCM_FORMAT_S8:
327 *endianness = 0;
328 *fmt = AUDIO_FORMAT_S8;
329 break;
330
331 case SND_PCM_FORMAT_U8:
332 *endianness = 0;
333 *fmt = AUDIO_FORMAT_U8;
334 break;
335
336 case SND_PCM_FORMAT_S16_LE:
337 *endianness = 0;
338 *fmt = AUDIO_FORMAT_S16;
339 break;
340
341 case SND_PCM_FORMAT_U16_LE:
342 *endianness = 0;
343 *fmt = AUDIO_FORMAT_U16;
344 break;
345
346 case SND_PCM_FORMAT_S16_BE:
347 *endianness = 1;
348 *fmt = AUDIO_FORMAT_S16;
349 break;
350
351 case SND_PCM_FORMAT_U16_BE:
352 *endianness = 1;
353 *fmt = AUDIO_FORMAT_U16;
354 break;
355
356 case SND_PCM_FORMAT_S32_LE:
357 *endianness = 0;
358 *fmt = AUDIO_FORMAT_S32;
359 break;
360
361 case SND_PCM_FORMAT_U32_LE:
362 *endianness = 0;
363 *fmt = AUDIO_FORMAT_U32;
364 break;
365
366 case SND_PCM_FORMAT_S32_BE:
367 *endianness = 1;
368 *fmt = AUDIO_FORMAT_S32;
369 break;
370
371 case SND_PCM_FORMAT_U32_BE:
372 *endianness = 1;
373 *fmt = AUDIO_FORMAT_U32;
374 break;
375
376 case SND_PCM_FORMAT_FLOAT_LE:
377 *endianness = 0;
378 *fmt = AUDIO_FORMAT_F32;
379 break;
380
381 case SND_PCM_FORMAT_FLOAT_BE:
382 *endianness = 1;
383 *fmt = AUDIO_FORMAT_F32;
384 break;
385
386 default:
387 dolog ("Unrecognized audio format %d\n", alsafmt);
388 return -1;
389 }
390
391 return 0;
392 }
393
394 static void alsa_dump_info (struct alsa_params_req *req,
395 struct alsa_params_obt *obt,
396 snd_pcm_format_t obtfmt,
397 AudiodevAlsaPerDirectionOptions *apdo)
398 {
399 dolog("parameter | requested value | obtained value\n");
400 dolog("format | %10d | %10d\n", req->fmt, obtfmt);
401 dolog("channels | %10d | %10d\n",
402 req->nchannels, obt->nchannels);
403 dolog("frequency | %10d | %10d\n", req->freq, obt->freq);
404 dolog("============================================\n");
405 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
406 apdo->buffer_length, apdo->period_length);
407 dolog("obtained: samples %ld\n", obt->samples);
408 }
409
410 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
411 {
412 int err;
413 snd_pcm_sw_params_t *sw_params;
414
415 snd_pcm_sw_params_alloca (&sw_params);
416
417 err = snd_pcm_sw_params_current (handle, sw_params);
418 if (err < 0) {
419 dolog ("Could not fully initialize DAC\n");
420 alsa_logerr (err, "Failed to get current software parameters\n");
421 return;
422 }
423
424 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
425 if (err < 0) {
426 dolog ("Could not fully initialize DAC\n");
427 alsa_logerr (err, "Failed to set software threshold to %ld\n",
428 threshold);
429 return;
430 }
431
432 err = snd_pcm_sw_params (handle, sw_params);
433 if (err < 0) {
434 dolog ("Could not fully initialize DAC\n");
435 alsa_logerr (err, "Failed to set software parameters\n");
436 return;
437 }
438 }
439
440 static int alsa_open(bool in, struct alsa_params_req *req,
441 struct alsa_params_obt *obt, snd_pcm_t **handlep,
442 Audiodev *dev)
443 {
444 AudiodevAlsaOptions *aopts = &dev->u.alsa;
445 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
446 snd_pcm_t *handle;
447 snd_pcm_hw_params_t *hw_params;
448 int err;
449 unsigned int freq, nchannels;
450 const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
451 snd_pcm_uframes_t obt_buffer_size;
452 const char *typ = in ? "ADC" : "DAC";
453 snd_pcm_format_t obtfmt;
454
455 freq = req->freq;
456 nchannels = req->nchannels;
457
458 snd_pcm_hw_params_alloca (&hw_params);
459
460 err = snd_pcm_open (
461 &handle,
462 pcm_name,
463 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
464 SND_PCM_NONBLOCK
465 );
466 if (err < 0) {
467 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
468 return -1;
469 }
470
471 err = snd_pcm_hw_params_any (handle, hw_params);
472 if (err < 0) {
473 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
474 goto err;
475 }
476
477 err = snd_pcm_hw_params_set_access (
478 handle,
479 hw_params,
480 SND_PCM_ACCESS_RW_INTERLEAVED
481 );
482 if (err < 0) {
483 alsa_logerr2 (err, typ, "Failed to set access type\n");
484 goto err;
485 }
486
487 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
488 if (err < 0) {
489 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
490 }
491
492 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
493 if (err < 0) {
494 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
495 goto err;
496 }
497
498 err = snd_pcm_hw_params_set_channels_near (
499 handle,
500 hw_params,
501 &nchannels
502 );
503 if (err < 0) {
504 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
505 req->nchannels);
506 goto err;
507 }
508
509 if (apdo->buffer_length) {
510 int dir = 0;
511 unsigned int btime = apdo->buffer_length;
512
513 err = snd_pcm_hw_params_set_buffer_time_near(
514 handle, hw_params, &btime, &dir);
515
516 if (err < 0) {
517 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
518 apdo->buffer_length);
519 goto err;
520 }
521
522 if (apdo->has_buffer_length && btime != apdo->buffer_length) {
523 dolog("Requested buffer time %" PRId32
524 " was rejected, using %u\n", apdo->buffer_length, btime);
525 }
526 }
527
528 if (apdo->period_length) {
529 int dir = 0;
530 unsigned int ptime = apdo->period_length;
531
532 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
533 &dir);
534
535 if (err < 0) {
536 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
537 apdo->period_length);
538 goto err;
539 }
540
541 if (apdo->has_period_length && ptime != apdo->period_length) {
542 dolog("Requested period time %" PRId32 " was rejected, using %d\n",
543 apdo->period_length, ptime);
544 }
545 }
546
547 err = snd_pcm_hw_params (handle, hw_params);
548 if (err < 0) {
549 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
550 goto err;
551 }
552
553 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
554 if (err < 0) {
555 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
556 goto err;
557 }
558
559 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
560 if (err < 0) {
561 alsa_logerr2 (err, typ, "Failed to get format\n");
562 goto err;
563 }
564
565 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
566 dolog ("Invalid format was returned %d\n", obtfmt);
567 goto err;
568 }
569
570 err = snd_pcm_prepare (handle);
571 if (err < 0) {
572 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
573 goto err;
574 }
575
576 if (!in && aopts->has_threshold && aopts->threshold) {
577 struct audsettings as = { .freq = freq };
578 alsa_set_threshold(
579 handle,
580 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
581 &as, aopts->threshold));
582 }
583
584 obt->nchannels = nchannels;
585 obt->freq = freq;
586 obt->samples = obt_buffer_size;
587
588 *handlep = handle;
589
590 if (obtfmt != req->fmt ||
591 obt->nchannels != req->nchannels ||
592 obt->freq != req->freq) {
593 dolog ("Audio parameters for %s\n", typ);
594 alsa_dump_info(req, obt, obtfmt, apdo);
595 }
596
597 #ifdef DEBUG
598 alsa_dump_info(req, obt, obtfmt, apdo);
599 #endif
600 return 0;
601
602 err:
603 alsa_anal_close1 (&handle);
604 return -1;
605 }
606
607 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
608 {
609 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
610 size_t pos = 0;
611 size_t len_frames = len / hw->info.bytes_per_frame;
612
613 while (len_frames) {
614 char *src = advance(buf, pos);
615 snd_pcm_sframes_t written;
616
617 written = snd_pcm_writei(alsa->handle, src, len_frames);
618
619 if (written <= 0) {
620 switch (written) {
621 case 0:
622 trace_alsa_wrote_zero(len_frames);
623 return pos;
624
625 case -EPIPE:
626 if (alsa_recover(alsa->handle)) {
627 alsa_logerr(written, "Failed to write %zu frames\n",
628 len_frames);
629 return pos;
630 }
631 trace_alsa_xrun_out();
632 continue;
633
634 case -ESTRPIPE:
635 /*
636 * stream is suspended and waiting for an application
637 * recovery
638 */
639 if (alsa_resume(alsa->handle)) {
640 alsa_logerr(written, "Failed to write %zu frames\n",
641 len_frames);
642 return pos;
643 }
644 trace_alsa_resume_out();
645 continue;
646
647 case -EAGAIN:
648 return pos;
649
650 default:
651 alsa_logerr(written, "Failed to write %zu frames from %p\n",
652 len, src);
653 return pos;
654 }
655 }
656
657 pos += written * hw->info.bytes_per_frame;
658 if (written < len_frames) {
659 break;
660 }
661 len_frames -= written;
662 }
663
664 return pos;
665 }
666
667 static void alsa_fini_out (HWVoiceOut *hw)
668 {
669 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
670
671 ldebug ("alsa_fini\n");
672 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
673 }
674
675 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
676 void *drv_opaque)
677 {
678 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
679 struct alsa_params_req req;
680 struct alsa_params_obt obt;
681 snd_pcm_t *handle;
682 struct audsettings obt_as;
683 Audiodev *dev = drv_opaque;
684
685 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
686 req.freq = as->freq;
687 req.nchannels = as->nchannels;
688
689 if (alsa_open(0, &req, &obt, &handle, dev)) {
690 return -1;
691 }
692
693 obt_as.freq = obt.freq;
694 obt_as.nchannels = obt.nchannels;
695 obt_as.fmt = obt.fmt;
696 obt_as.endianness = obt.endianness;
697
698 audio_pcm_init_info (&hw->info, &obt_as);
699 hw->samples = obt.samples;
700
701 alsa->pollhlp.s = hw->s;
702 alsa->handle = handle;
703 alsa->dev = dev;
704 return 0;
705 }
706
707 #define VOICE_CTL_PAUSE 0
708 #define VOICE_CTL_PREPARE 1
709 #define VOICE_CTL_START 2
710
711 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
712 {
713 int err;
714
715 if (ctl == VOICE_CTL_PAUSE) {
716 err = snd_pcm_drop (handle);
717 if (err < 0) {
718 alsa_logerr (err, "Could not stop %s\n", typ);
719 return -1;
720 }
721 } else {
722 err = snd_pcm_prepare (handle);
723 if (err < 0) {
724 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
725 return -1;
726 }
727 if (ctl == VOICE_CTL_START) {
728 err = snd_pcm_start(handle);
729 if (err < 0) {
730 alsa_logerr (err, "Could not start handle for %s\n", typ);
731 return -1;
732 }
733 }
734 }
735
736 return 0;
737 }
738
739 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
740 {
741 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
742 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
743
744 if (enable) {
745 bool poll_mode = apdo->try_poll;
746
747 ldebug("enabling voice\n");
748 if (poll_mode && alsa_poll_out(hw)) {
749 poll_mode = 0;
750 }
751 hw->poll_mode = poll_mode;
752 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
753 } else {
754 ldebug("disabling voice\n");
755 if (hw->poll_mode) {
756 hw->poll_mode = 0;
757 alsa_fini_poll(&alsa->pollhlp);
758 }
759 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
760 }
761 }
762
763 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
764 {
765 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
766 struct alsa_params_req req;
767 struct alsa_params_obt obt;
768 snd_pcm_t *handle;
769 struct audsettings obt_as;
770 Audiodev *dev = drv_opaque;
771
772 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
773 req.freq = as->freq;
774 req.nchannels = as->nchannels;
775
776 if (alsa_open(1, &req, &obt, &handle, dev)) {
777 return -1;
778 }
779
780 obt_as.freq = obt.freq;
781 obt_as.nchannels = obt.nchannels;
782 obt_as.fmt = obt.fmt;
783 obt_as.endianness = obt.endianness;
784
785 audio_pcm_init_info (&hw->info, &obt_as);
786 hw->samples = obt.samples;
787
788 alsa->pollhlp.s = hw->s;
789 alsa->handle = handle;
790 alsa->dev = dev;
791 return 0;
792 }
793
794 static void alsa_fini_in (HWVoiceIn *hw)
795 {
796 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
797
798 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
799 }
800
801 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
802 {
803 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
804 size_t pos = 0;
805
806 while (len) {
807 void *dst = advance(buf, pos);
808 snd_pcm_sframes_t nread;
809
810 nread = snd_pcm_readi(
811 alsa->handle, dst, len / hw->info.bytes_per_frame);
812
813 if (nread <= 0) {
814 switch (nread) {
815 case 0:
816 trace_alsa_read_zero(len);
817 return pos;
818
819 case -EPIPE:
820 if (alsa_recover(alsa->handle)) {
821 alsa_logerr(nread, "Failed to read %zu frames\n", len);
822 return pos;
823 }
824 trace_alsa_xrun_in();
825 continue;
826
827 case -EAGAIN:
828 return pos;
829
830 default:
831 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
832 len, dst);
833 return pos;
834 }
835 }
836
837 pos += nread * hw->info.bytes_per_frame;
838 len -= nread * hw->info.bytes_per_frame;
839 }
840
841 return pos;
842 }
843
844 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
845 {
846 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
847 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
848
849 if (enable) {
850 bool poll_mode = apdo->try_poll;
851
852 ldebug("enabling voice\n");
853 if (poll_mode && alsa_poll_in(hw)) {
854 poll_mode = 0;
855 }
856 hw->poll_mode = poll_mode;
857
858 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
859 } else {
860 ldebug ("disabling voice\n");
861 if (hw->poll_mode) {
862 hw->poll_mode = 0;
863 alsa_fini_poll(&alsa->pollhlp);
864 }
865 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
866 }
867 }
868
869 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
870 {
871 if (!apdo->has_try_poll) {
872 apdo->try_poll = true;
873 apdo->has_try_poll = true;
874 }
875 }
876
877 static void *alsa_audio_init(Audiodev *dev)
878 {
879 AudiodevAlsaOptions *aopts;
880 assert(dev->driver == AUDIODEV_DRIVER_ALSA);
881
882 aopts = &dev->u.alsa;
883 alsa_init_per_direction(aopts->in);
884 alsa_init_per_direction(aopts->out);
885
886 /*
887 * need to define them, as otherwise alsa produces no sound
888 * doesn't set has_* so alsa_open can identify it wasn't set by the user
889 */
890 if (!dev->u.alsa.out->has_period_length) {
891 /* 1024 frames assuming 44100Hz */
892 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
893 }
894 if (!dev->u.alsa.out->has_buffer_length) {
895 /* 4096 frames assuming 44100Hz */
896 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
897 }
898
899 /*
900 * OptsVisitor sets unspecified optional fields to zero, but do not depend
901 * on it...
902 */
903 if (!dev->u.alsa.in->has_period_length) {
904 dev->u.alsa.in->period_length = 0;
905 }
906 if (!dev->u.alsa.in->has_buffer_length) {
907 dev->u.alsa.in->buffer_length = 0;
908 }
909
910 return dev;
911 }
912
913 static void alsa_audio_fini (void *opaque)
914 {
915 }
916
917 static struct audio_pcm_ops alsa_pcm_ops = {
918 .init_out = alsa_init_out,
919 .fini_out = alsa_fini_out,
920 .write = alsa_write,
921 .run_buffer_out = audio_generic_run_buffer_out,
922 .enable_out = alsa_enable_out,
923
924 .init_in = alsa_init_in,
925 .fini_in = alsa_fini_in,
926 .read = alsa_read,
927 .run_buffer_in = audio_generic_run_buffer_in,
928 .enable_in = alsa_enable_in,
929 };
930
931 static struct audio_driver alsa_audio_driver = {
932 .name = "alsa",
933 .descr = "ALSA http://www.alsa-project.org",
934 .init = alsa_audio_init,
935 .fini = alsa_audio_fini,
936 .pcm_ops = &alsa_pcm_ops,
937 .can_be_default = 1,
938 .max_voices_out = INT_MAX,
939 .max_voices_in = INT_MAX,
940 .voice_size_out = sizeof (ALSAVoiceOut),
941 .voice_size_in = sizeof (ALSAVoiceIn)
942 };
943
944 static void register_audio_alsa(void)
945 {
946 audio_driver_register(&alsa_audio_driver);
947 }
948 type_init(register_audio_alsa);