Update VERSION for v7.2.0-rc4
[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 #define DEBUG_ALSA 0
38
39 struct pollhlp {
40 snd_pcm_t *handle;
41 struct pollfd *pfds;
42 int count;
43 int mask;
44 AudioState *s;
45 };
46
47 typedef struct ALSAVoiceOut {
48 HWVoiceOut hw;
49 snd_pcm_t *handle;
50 struct pollhlp pollhlp;
51 Audiodev *dev;
52 } ALSAVoiceOut;
53
54 typedef struct ALSAVoiceIn {
55 HWVoiceIn hw;
56 snd_pcm_t *handle;
57 struct pollhlp pollhlp;
58 Audiodev *dev;
59 } ALSAVoiceIn;
60
61 struct alsa_params_req {
62 int freq;
63 snd_pcm_format_t fmt;
64 int nchannels;
65 };
66
67 struct alsa_params_obt {
68 int freq;
69 AudioFormat fmt;
70 int endianness;
71 int nchannels;
72 snd_pcm_uframes_t samples;
73 };
74
75 static void G_GNUC_PRINTF (2, 3) alsa_logerr (int err, const char *fmt, ...)
76 {
77 va_list ap;
78
79 va_start (ap, fmt);
80 AUD_vlog (AUDIO_CAP, fmt, ap);
81 va_end (ap);
82
83 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
84 }
85
86 static void G_GNUC_PRINTF (3, 4) alsa_logerr2 (
87 int err,
88 const char *typ,
89 const char *fmt,
90 ...
91 )
92 {
93 va_list ap;
94
95 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
96
97 va_start (ap, fmt);
98 AUD_vlog (AUDIO_CAP, fmt, ap);
99 va_end (ap);
100
101 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
102 }
103
104 static void alsa_fini_poll (struct pollhlp *hlp)
105 {
106 int i;
107 struct pollfd *pfds = hlp->pfds;
108
109 if (pfds) {
110 for (i = 0; i < hlp->count; ++i) {
111 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
112 }
113 g_free (pfds);
114 }
115 hlp->pfds = NULL;
116 hlp->count = 0;
117 hlp->handle = NULL;
118 }
119
120 static void alsa_anal_close1 (snd_pcm_t **handlep)
121 {
122 int err = snd_pcm_close (*handlep);
123 if (err) {
124 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
125 }
126 *handlep = NULL;
127 }
128
129 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
130 {
131 alsa_fini_poll (hlp);
132 alsa_anal_close1 (handlep);
133 }
134
135 static int alsa_recover (snd_pcm_t *handle)
136 {
137 int err = snd_pcm_prepare (handle);
138 if (err < 0) {
139 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
140 return -1;
141 }
142 return 0;
143 }
144
145 static int alsa_resume (snd_pcm_t *handle)
146 {
147 int err = snd_pcm_resume (handle);
148 if (err < 0) {
149 alsa_logerr (err, "Failed to resume handle %p\n", handle);
150 return -1;
151 }
152 return 0;
153 }
154
155 static void alsa_poll_handler (void *opaque)
156 {
157 int err, count;
158 snd_pcm_state_t state;
159 struct pollhlp *hlp = opaque;
160 unsigned short revents;
161
162 count = poll (hlp->pfds, hlp->count, 0);
163 if (count < 0) {
164 dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
165 return;
166 }
167
168 if (!count) {
169 return;
170 }
171
172 /* XXX: ALSA example uses initial count, not the one returned by
173 poll, correct? */
174 err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
175 hlp->count, &revents);
176 if (err < 0) {
177 alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
178 return;
179 }
180
181 if (!(revents & hlp->mask)) {
182 trace_alsa_revents(revents);
183 return;
184 }
185
186 state = snd_pcm_state (hlp->handle);
187 switch (state) {
188 case SND_PCM_STATE_SETUP:
189 alsa_recover (hlp->handle);
190 break;
191
192 case SND_PCM_STATE_XRUN:
193 alsa_recover (hlp->handle);
194 break;
195
196 case SND_PCM_STATE_SUSPENDED:
197 alsa_resume (hlp->handle);
198 break;
199
200 case SND_PCM_STATE_PREPARED:
201 audio_run(hlp->s, "alsa run (prepared)");
202 break;
203
204 case SND_PCM_STATE_RUNNING:
205 audio_run(hlp->s, "alsa run (running)");
206 break;
207
208 default:
209 dolog ("Unexpected state %d\n", state);
210 }
211 }
212
213 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
214 {
215 int i, count, err;
216 struct pollfd *pfds;
217
218 count = snd_pcm_poll_descriptors_count (handle);
219 if (count <= 0) {
220 dolog ("Could not initialize poll mode\n"
221 "Invalid number of poll descriptors %d\n", count);
222 return -1;
223 }
224
225 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
226 if (!pfds) {
227 dolog ("Could not initialize poll mode\n");
228 return -1;
229 }
230
231 err = snd_pcm_poll_descriptors (handle, pfds, count);
232 if (err < 0) {
233 alsa_logerr (err, "Could not initialize poll mode\n"
234 "Could not obtain poll descriptors\n");
235 g_free (pfds);
236 return -1;
237 }
238
239 for (i = 0; i < count; ++i) {
240 if (pfds[i].events & POLLIN) {
241 qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
242 }
243 if (pfds[i].events & POLLOUT) {
244 trace_alsa_pollout(i, pfds[i].fd);
245 qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
246 }
247 trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
248
249 }
250 hlp->pfds = pfds;
251 hlp->count = count;
252 hlp->handle = handle;
253 hlp->mask = mask;
254 return 0;
255 }
256
257 static int alsa_poll_out (HWVoiceOut *hw)
258 {
259 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
260
261 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
262 }
263
264 static int alsa_poll_in (HWVoiceIn *hw)
265 {
266 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
267
268 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
269 }
270
271 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
272 {
273 switch (fmt) {
274 case AUDIO_FORMAT_S8:
275 return SND_PCM_FORMAT_S8;
276
277 case AUDIO_FORMAT_U8:
278 return SND_PCM_FORMAT_U8;
279
280 case AUDIO_FORMAT_S16:
281 if (endianness) {
282 return SND_PCM_FORMAT_S16_BE;
283 } else {
284 return SND_PCM_FORMAT_S16_LE;
285 }
286
287 case AUDIO_FORMAT_U16:
288 if (endianness) {
289 return SND_PCM_FORMAT_U16_BE;
290 } else {
291 return SND_PCM_FORMAT_U16_LE;
292 }
293
294 case AUDIO_FORMAT_S32:
295 if (endianness) {
296 return SND_PCM_FORMAT_S32_BE;
297 } else {
298 return SND_PCM_FORMAT_S32_LE;
299 }
300
301 case AUDIO_FORMAT_U32:
302 if (endianness) {
303 return SND_PCM_FORMAT_U32_BE;
304 } else {
305 return SND_PCM_FORMAT_U32_LE;
306 }
307
308 case AUDIO_FORMAT_F32:
309 if (endianness) {
310 return SND_PCM_FORMAT_FLOAT_BE;
311 } else {
312 return SND_PCM_FORMAT_FLOAT_LE;
313 }
314
315 default:
316 dolog ("Internal logic error: Bad audio format %d\n", fmt);
317 #ifdef DEBUG_AUDIO
318 abort ();
319 #endif
320 return SND_PCM_FORMAT_U8;
321 }
322 }
323
324 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
325 int *endianness)
326 {
327 switch (alsafmt) {
328 case SND_PCM_FORMAT_S8:
329 *endianness = 0;
330 *fmt = AUDIO_FORMAT_S8;
331 break;
332
333 case SND_PCM_FORMAT_U8:
334 *endianness = 0;
335 *fmt = AUDIO_FORMAT_U8;
336 break;
337
338 case SND_PCM_FORMAT_S16_LE:
339 *endianness = 0;
340 *fmt = AUDIO_FORMAT_S16;
341 break;
342
343 case SND_PCM_FORMAT_U16_LE:
344 *endianness = 0;
345 *fmt = AUDIO_FORMAT_U16;
346 break;
347
348 case SND_PCM_FORMAT_S16_BE:
349 *endianness = 1;
350 *fmt = AUDIO_FORMAT_S16;
351 break;
352
353 case SND_PCM_FORMAT_U16_BE:
354 *endianness = 1;
355 *fmt = AUDIO_FORMAT_U16;
356 break;
357
358 case SND_PCM_FORMAT_S32_LE:
359 *endianness = 0;
360 *fmt = AUDIO_FORMAT_S32;
361 break;
362
363 case SND_PCM_FORMAT_U32_LE:
364 *endianness = 0;
365 *fmt = AUDIO_FORMAT_U32;
366 break;
367
368 case SND_PCM_FORMAT_S32_BE:
369 *endianness = 1;
370 *fmt = AUDIO_FORMAT_S32;
371 break;
372
373 case SND_PCM_FORMAT_U32_BE:
374 *endianness = 1;
375 *fmt = AUDIO_FORMAT_U32;
376 break;
377
378 case SND_PCM_FORMAT_FLOAT_LE:
379 *endianness = 0;
380 *fmt = AUDIO_FORMAT_F32;
381 break;
382
383 case SND_PCM_FORMAT_FLOAT_BE:
384 *endianness = 1;
385 *fmt = AUDIO_FORMAT_F32;
386 break;
387
388 default:
389 dolog ("Unrecognized audio format %d\n", alsafmt);
390 return -1;
391 }
392
393 return 0;
394 }
395
396 static void alsa_dump_info (struct alsa_params_req *req,
397 struct alsa_params_obt *obt,
398 snd_pcm_format_t obtfmt,
399 AudiodevAlsaPerDirectionOptions *apdo)
400 {
401 dolog("parameter | requested value | obtained value\n");
402 dolog("format | %10d | %10d\n", req->fmt, obtfmt);
403 dolog("channels | %10d | %10d\n",
404 req->nchannels, obt->nchannels);
405 dolog("frequency | %10d | %10d\n", req->freq, obt->freq);
406 dolog("============================================\n");
407 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
408 apdo->buffer_length, apdo->period_length);
409 dolog("obtained: samples %ld\n", obt->samples);
410 }
411
412 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
413 {
414 int err;
415 snd_pcm_sw_params_t *sw_params;
416
417 snd_pcm_sw_params_alloca (&sw_params);
418
419 err = snd_pcm_sw_params_current (handle, sw_params);
420 if (err < 0) {
421 dolog ("Could not fully initialize DAC\n");
422 alsa_logerr (err, "Failed to get current software parameters\n");
423 return;
424 }
425
426 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
427 if (err < 0) {
428 dolog ("Could not fully initialize DAC\n");
429 alsa_logerr (err, "Failed to set software threshold to %ld\n",
430 threshold);
431 return;
432 }
433
434 err = snd_pcm_sw_params (handle, sw_params);
435 if (err < 0) {
436 dolog ("Could not fully initialize DAC\n");
437 alsa_logerr (err, "Failed to set software parameters\n");
438 return;
439 }
440 }
441
442 static int alsa_open(bool in, struct alsa_params_req *req,
443 struct alsa_params_obt *obt, snd_pcm_t **handlep,
444 Audiodev *dev)
445 {
446 AudiodevAlsaOptions *aopts = &dev->u.alsa;
447 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
448 snd_pcm_t *handle;
449 snd_pcm_hw_params_t *hw_params;
450 int err;
451 unsigned int freq, nchannels;
452 const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
453 snd_pcm_uframes_t obt_buffer_size;
454 const char *typ = in ? "ADC" : "DAC";
455 snd_pcm_format_t obtfmt;
456
457 freq = req->freq;
458 nchannels = req->nchannels;
459
460 snd_pcm_hw_params_alloca (&hw_params);
461
462 err = snd_pcm_open (
463 &handle,
464 pcm_name,
465 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
466 SND_PCM_NONBLOCK
467 );
468 if (err < 0) {
469 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
470 return -1;
471 }
472
473 err = snd_pcm_hw_params_any (handle, hw_params);
474 if (err < 0) {
475 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
476 goto err;
477 }
478
479 err = snd_pcm_hw_params_set_access (
480 handle,
481 hw_params,
482 SND_PCM_ACCESS_RW_INTERLEAVED
483 );
484 if (err < 0) {
485 alsa_logerr2 (err, typ, "Failed to set access type\n");
486 goto err;
487 }
488
489 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
490 if (err < 0) {
491 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
492 }
493
494 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
495 if (err < 0) {
496 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
497 goto err;
498 }
499
500 err = snd_pcm_hw_params_set_channels_near (
501 handle,
502 hw_params,
503 &nchannels
504 );
505 if (err < 0) {
506 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
507 req->nchannels);
508 goto err;
509 }
510
511 if (apdo->buffer_length) {
512 int dir = 0;
513 unsigned int btime = apdo->buffer_length;
514
515 err = snd_pcm_hw_params_set_buffer_time_near(
516 handle, hw_params, &btime, &dir);
517
518 if (err < 0) {
519 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
520 apdo->buffer_length);
521 goto err;
522 }
523
524 if (apdo->has_buffer_length && btime != apdo->buffer_length) {
525 dolog("Requested buffer time %" PRId32
526 " was rejected, using %u\n", apdo->buffer_length, btime);
527 }
528 }
529
530 if (apdo->period_length) {
531 int dir = 0;
532 unsigned int ptime = apdo->period_length;
533
534 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
535 &dir);
536
537 if (err < 0) {
538 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
539 apdo->period_length);
540 goto err;
541 }
542
543 if (apdo->has_period_length && ptime != apdo->period_length) {
544 dolog("Requested period time %" PRId32 " was rejected, using %d\n",
545 apdo->period_length, ptime);
546 }
547 }
548
549 err = snd_pcm_hw_params (handle, hw_params);
550 if (err < 0) {
551 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
552 goto err;
553 }
554
555 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
556 if (err < 0) {
557 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
558 goto err;
559 }
560
561 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
562 if (err < 0) {
563 alsa_logerr2 (err, typ, "Failed to get format\n");
564 goto err;
565 }
566
567 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
568 dolog ("Invalid format was returned %d\n", obtfmt);
569 goto err;
570 }
571
572 err = snd_pcm_prepare (handle);
573 if (err < 0) {
574 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
575 goto err;
576 }
577
578 if (!in && aopts->has_threshold && aopts->threshold) {
579 struct audsettings as = { .freq = freq };
580 alsa_set_threshold(
581 handle,
582 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
583 &as, aopts->threshold));
584 }
585
586 obt->nchannels = nchannels;
587 obt->freq = freq;
588 obt->samples = obt_buffer_size;
589
590 *handlep = handle;
591
592 if (DEBUG_ALSA || obtfmt != req->fmt ||
593 obt->nchannels != req->nchannels || obt->freq != req->freq) {
594 dolog ("Audio parameters for %s\n", typ);
595 alsa_dump_info(req, obt, obtfmt, apdo);
596 }
597
598 return 0;
599
600 err:
601 alsa_anal_close1 (&handle);
602 return -1;
603 }
604
605 static size_t alsa_buffer_get_free(HWVoiceOut *hw)
606 {
607 ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
608 snd_pcm_sframes_t avail;
609 size_t alsa_free, generic_free, generic_in_use;
610
611 avail = snd_pcm_avail_update(alsa->handle);
612 if (avail < 0) {
613 if (avail == -EPIPE) {
614 if (!alsa_recover(alsa->handle)) {
615 avail = snd_pcm_avail_update(alsa->handle);
616 }
617 }
618 if (avail < 0) {
619 alsa_logerr(avail,
620 "Could not obtain number of available frames\n");
621 avail = 0;
622 }
623 }
624
625 alsa_free = avail * hw->info.bytes_per_frame;
626 generic_free = audio_generic_buffer_get_free(hw);
627 generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
628 if (generic_in_use) {
629 /*
630 * This code can only be reached in the unlikely case that
631 * snd_pcm_avail_update() returned a larger number of frames
632 * than snd_pcm_writei() could write. Make sure that all
633 * remaining bytes in the generic buffer can be written.
634 */
635 alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
636 }
637
638 return alsa_free;
639 }
640
641 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
642 {
643 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
644 size_t pos = 0;
645 size_t len_frames = len / hw->info.bytes_per_frame;
646
647 while (len_frames) {
648 char *src = advance(buf, pos);
649 snd_pcm_sframes_t written;
650
651 written = snd_pcm_writei(alsa->handle, src, len_frames);
652
653 if (written <= 0) {
654 switch (written) {
655 case 0:
656 trace_alsa_wrote_zero(len_frames);
657 return pos;
658
659 case -EPIPE:
660 if (alsa_recover(alsa->handle)) {
661 alsa_logerr(written, "Failed to write %zu frames\n",
662 len_frames);
663 return pos;
664 }
665 trace_alsa_xrun_out();
666 continue;
667
668 case -ESTRPIPE:
669 /*
670 * stream is suspended and waiting for an application
671 * recovery
672 */
673 if (alsa_resume(alsa->handle)) {
674 alsa_logerr(written, "Failed to write %zu frames\n",
675 len_frames);
676 return pos;
677 }
678 trace_alsa_resume_out();
679 continue;
680
681 case -EAGAIN:
682 return pos;
683
684 default:
685 alsa_logerr(written, "Failed to write %zu frames from %p\n",
686 len, src);
687 return pos;
688 }
689 }
690
691 pos += written * hw->info.bytes_per_frame;
692 if (written < len_frames) {
693 break;
694 }
695 len_frames -= written;
696 }
697
698 return pos;
699 }
700
701 static void alsa_fini_out (HWVoiceOut *hw)
702 {
703 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
704
705 ldebug ("alsa_fini\n");
706 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
707 }
708
709 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
710 void *drv_opaque)
711 {
712 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
713 struct alsa_params_req req;
714 struct alsa_params_obt obt;
715 snd_pcm_t *handle;
716 struct audsettings obt_as;
717 Audiodev *dev = drv_opaque;
718
719 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
720 req.freq = as->freq;
721 req.nchannels = as->nchannels;
722
723 if (alsa_open(0, &req, &obt, &handle, dev)) {
724 return -1;
725 }
726
727 obt_as.freq = obt.freq;
728 obt_as.nchannels = obt.nchannels;
729 obt_as.fmt = obt.fmt;
730 obt_as.endianness = obt.endianness;
731
732 audio_pcm_init_info (&hw->info, &obt_as);
733 hw->samples = obt.samples;
734
735 alsa->pollhlp.s = hw->s;
736 alsa->handle = handle;
737 alsa->dev = dev;
738 return 0;
739 }
740
741 #define VOICE_CTL_PAUSE 0
742 #define VOICE_CTL_PREPARE 1
743 #define VOICE_CTL_START 2
744
745 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
746 {
747 int err;
748
749 if (ctl == VOICE_CTL_PAUSE) {
750 err = snd_pcm_drop (handle);
751 if (err < 0) {
752 alsa_logerr (err, "Could not stop %s\n", typ);
753 return -1;
754 }
755 } else {
756 err = snd_pcm_prepare (handle);
757 if (err < 0) {
758 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
759 return -1;
760 }
761 if (ctl == VOICE_CTL_START) {
762 err = snd_pcm_start(handle);
763 if (err < 0) {
764 alsa_logerr (err, "Could not start handle for %s\n", typ);
765 return -1;
766 }
767 }
768 }
769
770 return 0;
771 }
772
773 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
774 {
775 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
776 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
777
778 if (enable) {
779 bool poll_mode = apdo->try_poll;
780
781 ldebug("enabling voice\n");
782 if (poll_mode && alsa_poll_out(hw)) {
783 poll_mode = 0;
784 }
785 hw->poll_mode = poll_mode;
786 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
787 } else {
788 ldebug("disabling voice\n");
789 if (hw->poll_mode) {
790 hw->poll_mode = 0;
791 alsa_fini_poll(&alsa->pollhlp);
792 }
793 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
794 }
795 }
796
797 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
798 {
799 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
800 struct alsa_params_req req;
801 struct alsa_params_obt obt;
802 snd_pcm_t *handle;
803 struct audsettings obt_as;
804 Audiodev *dev = drv_opaque;
805
806 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
807 req.freq = as->freq;
808 req.nchannels = as->nchannels;
809
810 if (alsa_open(1, &req, &obt, &handle, dev)) {
811 return -1;
812 }
813
814 obt_as.freq = obt.freq;
815 obt_as.nchannels = obt.nchannels;
816 obt_as.fmt = obt.fmt;
817 obt_as.endianness = obt.endianness;
818
819 audio_pcm_init_info (&hw->info, &obt_as);
820 hw->samples = obt.samples;
821
822 alsa->pollhlp.s = hw->s;
823 alsa->handle = handle;
824 alsa->dev = dev;
825 return 0;
826 }
827
828 static void alsa_fini_in (HWVoiceIn *hw)
829 {
830 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
831
832 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
833 }
834
835 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
836 {
837 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
838 size_t pos = 0;
839
840 while (len) {
841 void *dst = advance(buf, pos);
842 snd_pcm_sframes_t nread;
843
844 nread = snd_pcm_readi(
845 alsa->handle, dst, len / hw->info.bytes_per_frame);
846
847 if (nread <= 0) {
848 switch (nread) {
849 case 0:
850 trace_alsa_read_zero(len);
851 return pos;
852
853 case -EPIPE:
854 if (alsa_recover(alsa->handle)) {
855 alsa_logerr(nread, "Failed to read %zu frames\n", len);
856 return pos;
857 }
858 trace_alsa_xrun_in();
859 continue;
860
861 case -EAGAIN:
862 return pos;
863
864 default:
865 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
866 len, dst);
867 return pos;
868 }
869 }
870
871 pos += nread * hw->info.bytes_per_frame;
872 len -= nread * hw->info.bytes_per_frame;
873 }
874
875 return pos;
876 }
877
878 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
879 {
880 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
881 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
882
883 if (enable) {
884 bool poll_mode = apdo->try_poll;
885
886 ldebug("enabling voice\n");
887 if (poll_mode && alsa_poll_in(hw)) {
888 poll_mode = 0;
889 }
890 hw->poll_mode = poll_mode;
891
892 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
893 } else {
894 ldebug ("disabling voice\n");
895 if (hw->poll_mode) {
896 hw->poll_mode = 0;
897 alsa_fini_poll(&alsa->pollhlp);
898 }
899 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
900 }
901 }
902
903 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
904 {
905 if (!apdo->has_try_poll) {
906 apdo->try_poll = true;
907 apdo->has_try_poll = true;
908 }
909 }
910
911 static void *alsa_audio_init(Audiodev *dev)
912 {
913 AudiodevAlsaOptions *aopts;
914 assert(dev->driver == AUDIODEV_DRIVER_ALSA);
915
916 aopts = &dev->u.alsa;
917 alsa_init_per_direction(aopts->in);
918 alsa_init_per_direction(aopts->out);
919
920 /*
921 * need to define them, as otherwise alsa produces no sound
922 * doesn't set has_* so alsa_open can identify it wasn't set by the user
923 */
924 if (!dev->u.alsa.out->has_period_length) {
925 /* 1024 frames assuming 44100Hz */
926 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
927 }
928 if (!dev->u.alsa.out->has_buffer_length) {
929 /* 4096 frames assuming 44100Hz */
930 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
931 }
932
933 /*
934 * OptsVisitor sets unspecified optional fields to zero, but do not depend
935 * on it...
936 */
937 if (!dev->u.alsa.in->has_period_length) {
938 dev->u.alsa.in->period_length = 0;
939 }
940 if (!dev->u.alsa.in->has_buffer_length) {
941 dev->u.alsa.in->buffer_length = 0;
942 }
943
944 return dev;
945 }
946
947 static void alsa_audio_fini (void *opaque)
948 {
949 }
950
951 static struct audio_pcm_ops alsa_pcm_ops = {
952 .init_out = alsa_init_out,
953 .fini_out = alsa_fini_out,
954 .write = alsa_write,
955 .buffer_get_free = alsa_buffer_get_free,
956 .run_buffer_out = audio_generic_run_buffer_out,
957 .enable_out = alsa_enable_out,
958
959 .init_in = alsa_init_in,
960 .fini_in = alsa_fini_in,
961 .read = alsa_read,
962 .run_buffer_in = audio_generic_run_buffer_in,
963 .enable_in = alsa_enable_in,
964 };
965
966 static struct audio_driver alsa_audio_driver = {
967 .name = "alsa",
968 .descr = "ALSA http://www.alsa-project.org",
969 .init = alsa_audio_init,
970 .fini = alsa_audio_fini,
971 .pcm_ops = &alsa_pcm_ops,
972 .can_be_default = 1,
973 .max_voices_out = INT_MAX,
974 .max_voices_in = INT_MAX,
975 .voice_size_out = sizeof (ALSAVoiceOut),
976 .voice_size_in = sizeof (ALSAVoiceIn)
977 };
978
979 static void register_audio_alsa(void)
980 {
981 audio_driver_register(&alsa_audio_driver);
982 }
983 type_init(register_audio_alsa);