hw/ide/core: Trivial typo fix
[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 }
282 else {
283 return SND_PCM_FORMAT_S16_LE;
284 }
285
286 case AUDIO_FORMAT_U16:
287 if (endianness) {
288 return SND_PCM_FORMAT_U16_BE;
289 }
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 }
298 else {
299 return SND_PCM_FORMAT_S32_LE;
300 }
301
302 case AUDIO_FORMAT_U32:
303 if (endianness) {
304 return SND_PCM_FORMAT_U32_BE;
305 }
306 else {
307 return SND_PCM_FORMAT_U32_LE;
308 }
309
310 case AUDIO_FORMAT_F32:
311 if (endianness) {
312 return SND_PCM_FORMAT_FLOAT_BE;
313 } else {
314 return SND_PCM_FORMAT_FLOAT_LE;
315 }
316
317 default:
318 dolog ("Internal logic error: Bad audio format %d\n", fmt);
319 #ifdef DEBUG_AUDIO
320 abort ();
321 #endif
322 return SND_PCM_FORMAT_U8;
323 }
324 }
325
326 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
327 int *endianness)
328 {
329 switch (alsafmt) {
330 case SND_PCM_FORMAT_S8:
331 *endianness = 0;
332 *fmt = AUDIO_FORMAT_S8;
333 break;
334
335 case SND_PCM_FORMAT_U8:
336 *endianness = 0;
337 *fmt = AUDIO_FORMAT_U8;
338 break;
339
340 case SND_PCM_FORMAT_S16_LE:
341 *endianness = 0;
342 *fmt = AUDIO_FORMAT_S16;
343 break;
344
345 case SND_PCM_FORMAT_U16_LE:
346 *endianness = 0;
347 *fmt = AUDIO_FORMAT_U16;
348 break;
349
350 case SND_PCM_FORMAT_S16_BE:
351 *endianness = 1;
352 *fmt = AUDIO_FORMAT_S16;
353 break;
354
355 case SND_PCM_FORMAT_U16_BE:
356 *endianness = 1;
357 *fmt = AUDIO_FORMAT_U16;
358 break;
359
360 case SND_PCM_FORMAT_S32_LE:
361 *endianness = 0;
362 *fmt = AUDIO_FORMAT_S32;
363 break;
364
365 case SND_PCM_FORMAT_U32_LE:
366 *endianness = 0;
367 *fmt = AUDIO_FORMAT_U32;
368 break;
369
370 case SND_PCM_FORMAT_S32_BE:
371 *endianness = 1;
372 *fmt = AUDIO_FORMAT_S32;
373 break;
374
375 case SND_PCM_FORMAT_U32_BE:
376 *endianness = 1;
377 *fmt = AUDIO_FORMAT_U32;
378 break;
379
380 case SND_PCM_FORMAT_FLOAT_LE:
381 *endianness = 0;
382 *fmt = AUDIO_FORMAT_F32;
383 break;
384
385 case SND_PCM_FORMAT_FLOAT_BE:
386 *endianness = 1;
387 *fmt = AUDIO_FORMAT_F32;
388 break;
389
390 default:
391 dolog ("Unrecognized audio format %d\n", alsafmt);
392 return -1;
393 }
394
395 return 0;
396 }
397
398 static void alsa_dump_info (struct alsa_params_req *req,
399 struct alsa_params_obt *obt,
400 snd_pcm_format_t obtfmt,
401 AudiodevAlsaPerDirectionOptions *apdo)
402 {
403 dolog("parameter | requested value | obtained value\n");
404 dolog("format | %10d | %10d\n", req->fmt, obtfmt);
405 dolog("channels | %10d | %10d\n",
406 req->nchannels, obt->nchannels);
407 dolog("frequency | %10d | %10d\n", req->freq, obt->freq);
408 dolog("============================================\n");
409 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
410 apdo->buffer_length, apdo->period_length);
411 dolog("obtained: samples %ld\n", obt->samples);
412 }
413
414 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
415 {
416 int err;
417 snd_pcm_sw_params_t *sw_params;
418
419 snd_pcm_sw_params_alloca (&sw_params);
420
421 err = snd_pcm_sw_params_current (handle, sw_params);
422 if (err < 0) {
423 dolog ("Could not fully initialize DAC\n");
424 alsa_logerr (err, "Failed to get current software parameters\n");
425 return;
426 }
427
428 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
429 if (err < 0) {
430 dolog ("Could not fully initialize DAC\n");
431 alsa_logerr (err, "Failed to set software threshold to %ld\n",
432 threshold);
433 return;
434 }
435
436 err = snd_pcm_sw_params (handle, sw_params);
437 if (err < 0) {
438 dolog ("Could not fully initialize DAC\n");
439 alsa_logerr (err, "Failed to set software parameters\n");
440 return;
441 }
442 }
443
444 static int alsa_open(bool in, struct alsa_params_req *req,
445 struct alsa_params_obt *obt, snd_pcm_t **handlep,
446 Audiodev *dev)
447 {
448 AudiodevAlsaOptions *aopts = &dev->u.alsa;
449 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
450 snd_pcm_t *handle;
451 snd_pcm_hw_params_t *hw_params;
452 int err;
453 unsigned int freq, nchannels;
454 const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
455 snd_pcm_uframes_t obt_buffer_size;
456 const char *typ = in ? "ADC" : "DAC";
457 snd_pcm_format_t obtfmt;
458
459 freq = req->freq;
460 nchannels = req->nchannels;
461
462 snd_pcm_hw_params_alloca (&hw_params);
463
464 err = snd_pcm_open (
465 &handle,
466 pcm_name,
467 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
468 SND_PCM_NONBLOCK
469 );
470 if (err < 0) {
471 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
472 return -1;
473 }
474
475 err = snd_pcm_hw_params_any (handle, hw_params);
476 if (err < 0) {
477 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
478 goto err;
479 }
480
481 err = snd_pcm_hw_params_set_access (
482 handle,
483 hw_params,
484 SND_PCM_ACCESS_RW_INTERLEAVED
485 );
486 if (err < 0) {
487 alsa_logerr2 (err, typ, "Failed to set access type\n");
488 goto err;
489 }
490
491 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
492 if (err < 0) {
493 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
494 }
495
496 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
497 if (err < 0) {
498 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
499 goto err;
500 }
501
502 err = snd_pcm_hw_params_set_channels_near (
503 handle,
504 hw_params,
505 &nchannels
506 );
507 if (err < 0) {
508 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
509 req->nchannels);
510 goto err;
511 }
512
513 if (apdo->buffer_length) {
514 int dir = 0;
515 unsigned int btime = apdo->buffer_length;
516
517 err = snd_pcm_hw_params_set_buffer_time_near(
518 handle, hw_params, &btime, &dir);
519
520 if (err < 0) {
521 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
522 apdo->buffer_length);
523 goto err;
524 }
525
526 if (apdo->has_buffer_length && btime != apdo->buffer_length) {
527 dolog("Requested buffer time %" PRId32
528 " was rejected, using %u\n", apdo->buffer_length, btime);
529 }
530 }
531
532 if (apdo->period_length) {
533 int dir = 0;
534 unsigned int ptime = apdo->period_length;
535
536 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
537 &dir);
538
539 if (err < 0) {
540 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
541 apdo->period_length);
542 goto err;
543 }
544
545 if (apdo->has_period_length && ptime != apdo->period_length) {
546 dolog("Requested period time %" PRId32 " was rejected, using %d\n",
547 apdo->period_length, ptime);
548 }
549 }
550
551 err = snd_pcm_hw_params (handle, hw_params);
552 if (err < 0) {
553 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
554 goto err;
555 }
556
557 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
558 if (err < 0) {
559 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
560 goto err;
561 }
562
563 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
564 if (err < 0) {
565 alsa_logerr2 (err, typ, "Failed to get format\n");
566 goto err;
567 }
568
569 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
570 dolog ("Invalid format was returned %d\n", obtfmt);
571 goto err;
572 }
573
574 err = snd_pcm_prepare (handle);
575 if (err < 0) {
576 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
577 goto err;
578 }
579
580 if (!in && aopts->has_threshold && aopts->threshold) {
581 struct audsettings as = { .freq = freq };
582 alsa_set_threshold(
583 handle,
584 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
585 &as, aopts->threshold));
586 }
587
588 obt->nchannels = nchannels;
589 obt->freq = freq;
590 obt->samples = obt_buffer_size;
591
592 *handlep = handle;
593
594 if (obtfmt != req->fmt ||
595 obt->nchannels != req->nchannels ||
596 obt->freq != req->freq) {
597 dolog ("Audio parameters for %s\n", typ);
598 alsa_dump_info(req, obt, obtfmt, apdo);
599 }
600
601 #ifdef DEBUG
602 alsa_dump_info(req, obt, obtfmt, pdo);
603 #endif
604 return 0;
605
606 err:
607 alsa_anal_close1 (&handle);
608 return -1;
609 }
610
611 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
612 {
613 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
614 size_t pos = 0;
615 size_t len_frames = len / hw->info.bytes_per_frame;
616
617 while (len_frames) {
618 char *src = advance(buf, pos);
619 snd_pcm_sframes_t written;
620
621 written = snd_pcm_writei(alsa->handle, src, len_frames);
622
623 if (written <= 0) {
624 switch (written) {
625 case 0:
626 trace_alsa_wrote_zero(len_frames);
627 return pos;
628
629 case -EPIPE:
630 if (alsa_recover(alsa->handle)) {
631 alsa_logerr(written, "Failed to write %zu frames\n",
632 len_frames);
633 return pos;
634 }
635 trace_alsa_xrun_out();
636 continue;
637
638 case -ESTRPIPE:
639 /*
640 * stream is suspended and waiting for an application
641 * recovery
642 */
643 if (alsa_resume(alsa->handle)) {
644 alsa_logerr(written, "Failed to write %zu frames\n",
645 len_frames);
646 return pos;
647 }
648 trace_alsa_resume_out();
649 continue;
650
651 case -EAGAIN:
652 return pos;
653
654 default:
655 alsa_logerr(written, "Failed to write %zu frames from %p\n",
656 len, src);
657 return pos;
658 }
659 }
660
661 pos += written * hw->info.bytes_per_frame;
662 if (written < len_frames) {
663 break;
664 }
665 len_frames -= written;
666 }
667
668 return pos;
669 }
670
671 static void alsa_fini_out (HWVoiceOut *hw)
672 {
673 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
674
675 ldebug ("alsa_fini\n");
676 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
677 }
678
679 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
680 void *drv_opaque)
681 {
682 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
683 struct alsa_params_req req;
684 struct alsa_params_obt obt;
685 snd_pcm_t *handle;
686 struct audsettings obt_as;
687 Audiodev *dev = drv_opaque;
688
689 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
690 req.freq = as->freq;
691 req.nchannels = as->nchannels;
692
693 if (alsa_open(0, &req, &obt, &handle, dev)) {
694 return -1;
695 }
696
697 obt_as.freq = obt.freq;
698 obt_as.nchannels = obt.nchannels;
699 obt_as.fmt = obt.fmt;
700 obt_as.endianness = obt.endianness;
701
702 audio_pcm_init_info (&hw->info, &obt_as);
703 hw->samples = obt.samples;
704
705 alsa->pollhlp.s = hw->s;
706 alsa->handle = handle;
707 alsa->dev = dev;
708 return 0;
709 }
710
711 #define VOICE_CTL_PAUSE 0
712 #define VOICE_CTL_PREPARE 1
713 #define VOICE_CTL_START 2
714
715 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
716 {
717 int err;
718
719 if (ctl == VOICE_CTL_PAUSE) {
720 err = snd_pcm_drop (handle);
721 if (err < 0) {
722 alsa_logerr (err, "Could not stop %s\n", typ);
723 return -1;
724 }
725 }
726 else {
727 err = snd_pcm_prepare (handle);
728 if (err < 0) {
729 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
730 return -1;
731 }
732 if (ctl == VOICE_CTL_START) {
733 err = snd_pcm_start(handle);
734 if (err < 0) {
735 alsa_logerr (err, "Could not start handle for %s\n", typ);
736 return -1;
737 }
738 }
739 }
740
741 return 0;
742 }
743
744 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
745 {
746 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
747 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
748
749 if (enable) {
750 bool poll_mode = apdo->try_poll;
751
752 ldebug("enabling voice\n");
753 if (poll_mode && alsa_poll_out(hw)) {
754 poll_mode = 0;
755 }
756 hw->poll_mode = poll_mode;
757 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
758 } else {
759 ldebug("disabling voice\n");
760 if (hw->poll_mode) {
761 hw->poll_mode = 0;
762 alsa_fini_poll(&alsa->pollhlp);
763 }
764 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
765 }
766 }
767
768 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
769 {
770 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
771 struct alsa_params_req req;
772 struct alsa_params_obt obt;
773 snd_pcm_t *handle;
774 struct audsettings obt_as;
775 Audiodev *dev = drv_opaque;
776
777 req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
778 req.freq = as->freq;
779 req.nchannels = as->nchannels;
780
781 if (alsa_open(1, &req, &obt, &handle, dev)) {
782 return -1;
783 }
784
785 obt_as.freq = obt.freq;
786 obt_as.nchannels = obt.nchannels;
787 obt_as.fmt = obt.fmt;
788 obt_as.endianness = obt.endianness;
789
790 audio_pcm_init_info (&hw->info, &obt_as);
791 hw->samples = obt.samples;
792
793 alsa->pollhlp.s = hw->s;
794 alsa->handle = handle;
795 alsa->dev = dev;
796 return 0;
797 }
798
799 static void alsa_fini_in (HWVoiceIn *hw)
800 {
801 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
802
803 alsa_anal_close (&alsa->handle, &alsa->pollhlp);
804 }
805
806 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
807 {
808 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
809 size_t pos = 0;
810
811 while (len) {
812 void *dst = advance(buf, pos);
813 snd_pcm_sframes_t nread;
814
815 nread = snd_pcm_readi(
816 alsa->handle, dst, len / hw->info.bytes_per_frame);
817
818 if (nread <= 0) {
819 switch (nread) {
820 case 0:
821 trace_alsa_read_zero(len);
822 return pos;
823
824 case -EPIPE:
825 if (alsa_recover(alsa->handle)) {
826 alsa_logerr(nread, "Failed to read %zu frames\n", len);
827 return pos;
828 }
829 trace_alsa_xrun_in();
830 continue;
831
832 case -EAGAIN:
833 return pos;
834
835 default:
836 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
837 len, dst);
838 return pos;
839 }
840 }
841
842 pos += nread * hw->info.bytes_per_frame;
843 len -= nread * hw->info.bytes_per_frame;
844 }
845
846 return pos;
847 }
848
849 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
850 {
851 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
852 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
853
854 if (enable) {
855 bool poll_mode = apdo->try_poll;
856
857 ldebug("enabling voice\n");
858 if (poll_mode && alsa_poll_in(hw)) {
859 poll_mode = 0;
860 }
861 hw->poll_mode = poll_mode;
862
863 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
864 } else {
865 ldebug ("disabling voice\n");
866 if (hw->poll_mode) {
867 hw->poll_mode = 0;
868 alsa_fini_poll(&alsa->pollhlp);
869 }
870 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
871 }
872 }
873
874 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
875 {
876 if (!apdo->has_try_poll) {
877 apdo->try_poll = true;
878 apdo->has_try_poll = true;
879 }
880 }
881
882 static void *alsa_audio_init(Audiodev *dev)
883 {
884 AudiodevAlsaOptions *aopts;
885 assert(dev->driver == AUDIODEV_DRIVER_ALSA);
886
887 aopts = &dev->u.alsa;
888 alsa_init_per_direction(aopts->in);
889 alsa_init_per_direction(aopts->out);
890
891 /*
892 * need to define them, as otherwise alsa produces no sound
893 * doesn't set has_* so alsa_open can identify it wasn't set by the user
894 */
895 if (!dev->u.alsa.out->has_period_length) {
896 /* 1024 frames assuming 44100Hz */
897 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
898 }
899 if (!dev->u.alsa.out->has_buffer_length) {
900 /* 4096 frames assuming 44100Hz */
901 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
902 }
903
904 /*
905 * OptsVisitor sets unspecified optional fields to zero, but do not depend
906 * on it...
907 */
908 if (!dev->u.alsa.in->has_period_length) {
909 dev->u.alsa.in->period_length = 0;
910 }
911 if (!dev->u.alsa.in->has_buffer_length) {
912 dev->u.alsa.in->buffer_length = 0;
913 }
914
915 return dev;
916 }
917
918 static void alsa_audio_fini (void *opaque)
919 {
920 }
921
922 static struct audio_pcm_ops alsa_pcm_ops = {
923 .init_out = alsa_init_out,
924 .fini_out = alsa_fini_out,
925 .write = alsa_write,
926 .run_buffer_out = audio_generic_run_buffer_out,
927 .enable_out = alsa_enable_out,
928
929 .init_in = alsa_init_in,
930 .fini_in = alsa_fini_in,
931 .read = alsa_read,
932 .enable_in = alsa_enable_in,
933 };
934
935 static struct audio_driver alsa_audio_driver = {
936 .name = "alsa",
937 .descr = "ALSA http://www.alsa-project.org",
938 .init = alsa_audio_init,
939 .fini = alsa_audio_fini,
940 .pcm_ops = &alsa_pcm_ops,
941 .can_be_default = 1,
942 .max_voices_out = INT_MAX,
943 .max_voices_in = INT_MAX,
944 .voice_size_out = sizeof (ALSAVoiceOut),
945 .voice_size_in = sizeof (ALSAVoiceIn)
946 };
947
948 static void register_audio_alsa(void)
949 {
950 audio_driver_register(&alsa_audio_driver);
951 }
952 type_init(register_audio_alsa);