]>
Commit | Line | Data |
---|---|---|
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 | #include <alsa/asoundlib.h> | |
25 | #include "qemu-common.h" | |
26 | #include "qemu/main-loop.h" | |
27 | #include "audio.h" | |
28 | ||
29 | #if QEMU_GNUC_PREREQ(4, 3) | |
30 | #pragma GCC diagnostic ignored "-Waddress" | |
31 | #endif | |
32 | ||
33 | #define AUDIO_CAP "alsa" | |
34 | #include "audio_int.h" | |
35 | ||
36 | struct pollhlp { | |
37 | snd_pcm_t *handle; | |
38 | struct pollfd *pfds; | |
39 | int count; | |
40 | int mask; | |
41 | }; | |
42 | ||
43 | typedef struct ALSAVoiceOut { | |
44 | HWVoiceOut hw; | |
45 | int wpos; | |
46 | int pending; | |
47 | void *pcm_buf; | |
48 | snd_pcm_t *handle; | |
49 | struct pollhlp pollhlp; | |
50 | } ALSAVoiceOut; | |
51 | ||
52 | typedef struct ALSAVoiceIn { | |
53 | HWVoiceIn hw; | |
54 | snd_pcm_t *handle; | |
55 | void *pcm_buf; | |
56 | struct pollhlp pollhlp; | |
57 | } ALSAVoiceIn; | |
58 | ||
59 | static struct { | |
60 | int size_in_usec_in; | |
61 | int size_in_usec_out; | |
62 | const char *pcm_name_in; | |
63 | const char *pcm_name_out; | |
64 | unsigned int buffer_size_in; | |
65 | unsigned int period_size_in; | |
66 | unsigned int buffer_size_out; | |
67 | unsigned int period_size_out; | |
68 | unsigned int threshold; | |
69 | ||
70 | int buffer_size_in_overridden; | |
71 | int period_size_in_overridden; | |
72 | ||
73 | int buffer_size_out_overridden; | |
74 | int period_size_out_overridden; | |
75 | int verbose; | |
76 | } conf = { | |
77 | .buffer_size_out = 4096, | |
78 | .period_size_out = 1024, | |
79 | .pcm_name_out = "default", | |
80 | .pcm_name_in = "default", | |
81 | }; | |
82 | ||
83 | struct alsa_params_req { | |
84 | int freq; | |
85 | snd_pcm_format_t fmt; | |
86 | int nchannels; | |
87 | int size_in_usec; | |
88 | int override_mask; | |
89 | unsigned int buffer_size; | |
90 | unsigned int period_size; | |
91 | }; | |
92 | ||
93 | struct alsa_params_obt { | |
94 | int freq; | |
95 | audfmt_e fmt; | |
96 | int endianness; | |
97 | int nchannels; | |
98 | snd_pcm_uframes_t samples; | |
99 | }; | |
100 | ||
101 | static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) | |
102 | { | |
103 | va_list ap; | |
104 | ||
105 | va_start (ap, fmt); | |
106 | AUD_vlog (AUDIO_CAP, fmt, ap); | |
107 | va_end (ap); | |
108 | ||
109 | AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); | |
110 | } | |
111 | ||
112 | static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( | |
113 | int err, | |
114 | const char *typ, | |
115 | const char *fmt, | |
116 | ... | |
117 | ) | |
118 | { | |
119 | va_list ap; | |
120 | ||
121 | AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); | |
122 | ||
123 | va_start (ap, fmt); | |
124 | AUD_vlog (AUDIO_CAP, fmt, ap); | |
125 | va_end (ap); | |
126 | ||
127 | AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); | |
128 | } | |
129 | ||
130 | static void alsa_fini_poll (struct pollhlp *hlp) | |
131 | { | |
132 | int i; | |
133 | struct pollfd *pfds = hlp->pfds; | |
134 | ||
135 | if (pfds) { | |
136 | for (i = 0; i < hlp->count; ++i) { | |
137 | qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); | |
138 | } | |
139 | g_free (pfds); | |
140 | } | |
141 | hlp->pfds = NULL; | |
142 | hlp->count = 0; | |
143 | hlp->handle = NULL; | |
144 | } | |
145 | ||
146 | static void alsa_anal_close1 (snd_pcm_t **handlep) | |
147 | { | |
148 | int err = snd_pcm_close (*handlep); | |
149 | if (err) { | |
150 | alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); | |
151 | } | |
152 | *handlep = NULL; | |
153 | } | |
154 | ||
155 | static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) | |
156 | { | |
157 | alsa_fini_poll (hlp); | |
158 | alsa_anal_close1 (handlep); | |
159 | } | |
160 | ||
161 | static int alsa_recover (snd_pcm_t *handle) | |
162 | { | |
163 | int err = snd_pcm_prepare (handle); | |
164 | if (err < 0) { | |
165 | alsa_logerr (err, "Failed to prepare handle %p\n", handle); | |
166 | return -1; | |
167 | } | |
168 | return 0; | |
169 | } | |
170 | ||
171 | static int alsa_resume (snd_pcm_t *handle) | |
172 | { | |
173 | int err = snd_pcm_resume (handle); | |
174 | if (err < 0) { | |
175 | alsa_logerr (err, "Failed to resume handle %p\n", handle); | |
176 | return -1; | |
177 | } | |
178 | return 0; | |
179 | } | |
180 | ||
181 | static void alsa_poll_handler (void *opaque) | |
182 | { | |
183 | int err, count; | |
184 | snd_pcm_state_t state; | |
185 | struct pollhlp *hlp = opaque; | |
186 | unsigned short revents; | |
187 | ||
188 | count = poll (hlp->pfds, hlp->count, 0); | |
189 | if (count < 0) { | |
190 | dolog ("alsa_poll_handler: poll %s\n", strerror (errno)); | |
191 | return; | |
192 | } | |
193 | ||
194 | if (!count) { | |
195 | return; | |
196 | } | |
197 | ||
198 | /* XXX: ALSA example uses initial count, not the one returned by | |
199 | poll, correct? */ | |
200 | err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, | |
201 | hlp->count, &revents); | |
202 | if (err < 0) { | |
203 | alsa_logerr (err, "snd_pcm_poll_descriptors_revents"); | |
204 | return; | |
205 | } | |
206 | ||
207 | if (!(revents & hlp->mask)) { | |
208 | if (conf.verbose) { | |
209 | dolog ("revents = %d\n", revents); | |
210 | } | |
211 | return; | |
212 | } | |
213 | ||
214 | state = snd_pcm_state (hlp->handle); | |
215 | switch (state) { | |
216 | case SND_PCM_STATE_SETUP: | |
217 | alsa_recover (hlp->handle); | |
218 | break; | |
219 | ||
220 | case SND_PCM_STATE_XRUN: | |
221 | alsa_recover (hlp->handle); | |
222 | break; | |
223 | ||
224 | case SND_PCM_STATE_SUSPENDED: | |
225 | alsa_resume (hlp->handle); | |
226 | break; | |
227 | ||
228 | case SND_PCM_STATE_PREPARED: | |
229 | audio_run ("alsa run (prepared)"); | |
230 | break; | |
231 | ||
232 | case SND_PCM_STATE_RUNNING: | |
233 | audio_run ("alsa run (running)"); | |
234 | break; | |
235 | ||
236 | default: | |
237 | dolog ("Unexpected state %d\n", state); | |
238 | } | |
239 | } | |
240 | ||
241 | static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) | |
242 | { | |
243 | int i, count, err; | |
244 | struct pollfd *pfds; | |
245 | ||
246 | count = snd_pcm_poll_descriptors_count (handle); | |
247 | if (count <= 0) { | |
248 | dolog ("Could not initialize poll mode\n" | |
249 | "Invalid number of poll descriptors %d\n", count); | |
250 | return -1; | |
251 | } | |
252 | ||
253 | pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); | |
254 | if (!pfds) { | |
255 | dolog ("Could not initialize poll mode\n"); | |
256 | return -1; | |
257 | } | |
258 | ||
259 | err = snd_pcm_poll_descriptors (handle, pfds, count); | |
260 | if (err < 0) { | |
261 | alsa_logerr (err, "Could not initialize poll mode\n" | |
262 | "Could not obtain poll descriptors\n"); | |
263 | g_free (pfds); | |
264 | return -1; | |
265 | } | |
266 | ||
267 | for (i = 0; i < count; ++i) { | |
268 | if (pfds[i].events & POLLIN) { | |
269 | err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, | |
270 | NULL, hlp); | |
271 | } | |
272 | if (pfds[i].events & POLLOUT) { | |
273 | if (conf.verbose) { | |
274 | dolog ("POLLOUT %d %d\n", i, pfds[i].fd); | |
275 | } | |
276 | err = qemu_set_fd_handler (pfds[i].fd, NULL, | |
277 | alsa_poll_handler, hlp); | |
278 | } | |
279 | if (conf.verbose) { | |
280 | dolog ("Set handler events=%#x index=%d fd=%d err=%d\n", | |
281 | pfds[i].events, i, pfds[i].fd, err); | |
282 | } | |
283 | ||
284 | if (err) { | |
285 | dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n", | |
286 | pfds[i].events, i, pfds[i].fd, err); | |
287 | ||
288 | while (i--) { | |
289 | qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); | |
290 | } | |
291 | g_free (pfds); | |
292 | return -1; | |
293 | } | |
294 | } | |
295 | hlp->pfds = pfds; | |
296 | hlp->count = count; | |
297 | hlp->handle = handle; | |
298 | hlp->mask = mask; | |
299 | return 0; | |
300 | } | |
301 | ||
302 | static int alsa_poll_out (HWVoiceOut *hw) | |
303 | { | |
304 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
305 | ||
306 | return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); | |
307 | } | |
308 | ||
309 | static int alsa_poll_in (HWVoiceIn *hw) | |
310 | { | |
311 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
312 | ||
313 | return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); | |
314 | } | |
315 | ||
316 | static int alsa_write (SWVoiceOut *sw, void *buf, int len) | |
317 | { | |
318 | return audio_pcm_sw_write (sw, buf, len); | |
319 | } | |
320 | ||
321 | static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) | |
322 | { | |
323 | switch (fmt) { | |
324 | case AUD_FMT_S8: | |
325 | return SND_PCM_FORMAT_S8; | |
326 | ||
327 | case AUD_FMT_U8: | |
328 | return SND_PCM_FORMAT_U8; | |
329 | ||
330 | case AUD_FMT_S16: | |
331 | if (endianness) { | |
332 | return SND_PCM_FORMAT_S16_BE; | |
333 | } | |
334 | else { | |
335 | return SND_PCM_FORMAT_S16_LE; | |
336 | } | |
337 | ||
338 | case AUD_FMT_U16: | |
339 | if (endianness) { | |
340 | return SND_PCM_FORMAT_U16_BE; | |
341 | } | |
342 | else { | |
343 | return SND_PCM_FORMAT_U16_LE; | |
344 | } | |
345 | ||
346 | case AUD_FMT_S32: | |
347 | if (endianness) { | |
348 | return SND_PCM_FORMAT_S32_BE; | |
349 | } | |
350 | else { | |
351 | return SND_PCM_FORMAT_S32_LE; | |
352 | } | |
353 | ||
354 | case AUD_FMT_U32: | |
355 | if (endianness) { | |
356 | return SND_PCM_FORMAT_U32_BE; | |
357 | } | |
358 | else { | |
359 | return SND_PCM_FORMAT_U32_LE; | |
360 | } | |
361 | ||
362 | default: | |
363 | dolog ("Internal logic error: Bad audio format %d\n", fmt); | |
364 | #ifdef DEBUG_AUDIO | |
365 | abort (); | |
366 | #endif | |
367 | return SND_PCM_FORMAT_U8; | |
368 | } | |
369 | } | |
370 | ||
371 | static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, | |
372 | int *endianness) | |
373 | { | |
374 | switch (alsafmt) { | |
375 | case SND_PCM_FORMAT_S8: | |
376 | *endianness = 0; | |
377 | *fmt = AUD_FMT_S8; | |
378 | break; | |
379 | ||
380 | case SND_PCM_FORMAT_U8: | |
381 | *endianness = 0; | |
382 | *fmt = AUD_FMT_U8; | |
383 | break; | |
384 | ||
385 | case SND_PCM_FORMAT_S16_LE: | |
386 | *endianness = 0; | |
387 | *fmt = AUD_FMT_S16; | |
388 | break; | |
389 | ||
390 | case SND_PCM_FORMAT_U16_LE: | |
391 | *endianness = 0; | |
392 | *fmt = AUD_FMT_U16; | |
393 | break; | |
394 | ||
395 | case SND_PCM_FORMAT_S16_BE: | |
396 | *endianness = 1; | |
397 | *fmt = AUD_FMT_S16; | |
398 | break; | |
399 | ||
400 | case SND_PCM_FORMAT_U16_BE: | |
401 | *endianness = 1; | |
402 | *fmt = AUD_FMT_U16; | |
403 | break; | |
404 | ||
405 | case SND_PCM_FORMAT_S32_LE: | |
406 | *endianness = 0; | |
407 | *fmt = AUD_FMT_S32; | |
408 | break; | |
409 | ||
410 | case SND_PCM_FORMAT_U32_LE: | |
411 | *endianness = 0; | |
412 | *fmt = AUD_FMT_U32; | |
413 | break; | |
414 | ||
415 | case SND_PCM_FORMAT_S32_BE: | |
416 | *endianness = 1; | |
417 | *fmt = AUD_FMT_S32; | |
418 | break; | |
419 | ||
420 | case SND_PCM_FORMAT_U32_BE: | |
421 | *endianness = 1; | |
422 | *fmt = AUD_FMT_U32; | |
423 | break; | |
424 | ||
425 | default: | |
426 | dolog ("Unrecognized audio format %d\n", alsafmt); | |
427 | return -1; | |
428 | } | |
429 | ||
430 | return 0; | |
431 | } | |
432 | ||
433 | static void alsa_dump_info (struct alsa_params_req *req, | |
434 | struct alsa_params_obt *obt, | |
435 | snd_pcm_format_t obtfmt) | |
436 | { | |
437 | dolog ("parameter | requested value | obtained value\n"); | |
438 | dolog ("format | %10d | %10d\n", req->fmt, obtfmt); | |
439 | dolog ("channels | %10d | %10d\n", | |
440 | req->nchannels, obt->nchannels); | |
441 | dolog ("frequency | %10d | %10d\n", req->freq, obt->freq); | |
442 | dolog ("============================================\n"); | |
443 | dolog ("requested: buffer size %d period size %d\n", | |
444 | req->buffer_size, req->period_size); | |
445 | dolog ("obtained: samples %ld\n", obt->samples); | |
446 | } | |
447 | ||
448 | static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) | |
449 | { | |
450 | int err; | |
451 | snd_pcm_sw_params_t *sw_params; | |
452 | ||
453 | snd_pcm_sw_params_alloca (&sw_params); | |
454 | ||
455 | err = snd_pcm_sw_params_current (handle, sw_params); | |
456 | if (err < 0) { | |
457 | dolog ("Could not fully initialize DAC\n"); | |
458 | alsa_logerr (err, "Failed to get current software parameters\n"); | |
459 | return; | |
460 | } | |
461 | ||
462 | err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); | |
463 | if (err < 0) { | |
464 | dolog ("Could not fully initialize DAC\n"); | |
465 | alsa_logerr (err, "Failed to set software threshold to %ld\n", | |
466 | threshold); | |
467 | return; | |
468 | } | |
469 | ||
470 | err = snd_pcm_sw_params (handle, sw_params); | |
471 | if (err < 0) { | |
472 | dolog ("Could not fully initialize DAC\n"); | |
473 | alsa_logerr (err, "Failed to set software parameters\n"); | |
474 | return; | |
475 | } | |
476 | } | |
477 | ||
478 | static int alsa_open (int in, struct alsa_params_req *req, | |
479 | struct alsa_params_obt *obt, snd_pcm_t **handlep) | |
480 | { | |
481 | snd_pcm_t *handle; | |
482 | snd_pcm_hw_params_t *hw_params; | |
483 | int err; | |
484 | int size_in_usec; | |
485 | unsigned int freq, nchannels; | |
486 | const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; | |
487 | snd_pcm_uframes_t obt_buffer_size; | |
488 | const char *typ = in ? "ADC" : "DAC"; | |
489 | snd_pcm_format_t obtfmt; | |
490 | ||
491 | freq = req->freq; | |
492 | nchannels = req->nchannels; | |
493 | size_in_usec = req->size_in_usec; | |
494 | ||
495 | snd_pcm_hw_params_alloca (&hw_params); | |
496 | ||
497 | err = snd_pcm_open ( | |
498 | &handle, | |
499 | pcm_name, | |
500 | in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, | |
501 | SND_PCM_NONBLOCK | |
502 | ); | |
503 | if (err < 0) { | |
504 | alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); | |
505 | return -1; | |
506 | } | |
507 | ||
508 | err = snd_pcm_hw_params_any (handle, hw_params); | |
509 | if (err < 0) { | |
510 | alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); | |
511 | goto err; | |
512 | } | |
513 | ||
514 | err = snd_pcm_hw_params_set_access ( | |
515 | handle, | |
516 | hw_params, | |
517 | SND_PCM_ACCESS_RW_INTERLEAVED | |
518 | ); | |
519 | if (err < 0) { | |
520 | alsa_logerr2 (err, typ, "Failed to set access type\n"); | |
521 | goto err; | |
522 | } | |
523 | ||
524 | err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); | |
525 | if (err < 0 && conf.verbose) { | |
526 | alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); | |
527 | } | |
528 | ||
529 | err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); | |
530 | if (err < 0) { | |
531 | alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); | |
532 | goto err; | |
533 | } | |
534 | ||
535 | err = snd_pcm_hw_params_set_channels_near ( | |
536 | handle, | |
537 | hw_params, | |
538 | &nchannels | |
539 | ); | |
540 | if (err < 0) { | |
541 | alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", | |
542 | req->nchannels); | |
543 | goto err; | |
544 | } | |
545 | ||
546 | if (nchannels != 1 && nchannels != 2) { | |
547 | alsa_logerr2 (err, typ, | |
548 | "Can not handle obtained number of channels %d\n", | |
549 | nchannels); | |
550 | goto err; | |
551 | } | |
552 | ||
553 | if (req->buffer_size) { | |
554 | unsigned long obt; | |
555 | ||
556 | if (size_in_usec) { | |
557 | int dir = 0; | |
558 | unsigned int btime = req->buffer_size; | |
559 | ||
560 | err = snd_pcm_hw_params_set_buffer_time_near ( | |
561 | handle, | |
562 | hw_params, | |
563 | &btime, | |
564 | &dir | |
565 | ); | |
566 | obt = btime; | |
567 | } | |
568 | else { | |
569 | snd_pcm_uframes_t bsize = req->buffer_size; | |
570 | ||
571 | err = snd_pcm_hw_params_set_buffer_size_near ( | |
572 | handle, | |
573 | hw_params, | |
574 | &bsize | |
575 | ); | |
576 | obt = bsize; | |
577 | } | |
578 | if (err < 0) { | |
579 | alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n", | |
580 | size_in_usec ? "time" : "size", req->buffer_size); | |
581 | goto err; | |
582 | } | |
583 | ||
584 | if ((req->override_mask & 2) && (obt - req->buffer_size)) | |
585 | dolog ("Requested buffer %s %u was rejected, using %lu\n", | |
586 | size_in_usec ? "time" : "size", req->buffer_size, obt); | |
587 | } | |
588 | ||
589 | if (req->period_size) { | |
590 | unsigned long obt; | |
591 | ||
592 | if (size_in_usec) { | |
593 | int dir = 0; | |
594 | unsigned int ptime = req->period_size; | |
595 | ||
596 | err = snd_pcm_hw_params_set_period_time_near ( | |
597 | handle, | |
598 | hw_params, | |
599 | &ptime, | |
600 | &dir | |
601 | ); | |
602 | obt = ptime; | |
603 | } | |
604 | else { | |
605 | int dir = 0; | |
606 | snd_pcm_uframes_t psize = req->period_size; | |
607 | ||
608 | err = snd_pcm_hw_params_set_period_size_near ( | |
609 | handle, | |
610 | hw_params, | |
611 | &psize, | |
612 | &dir | |
613 | ); | |
614 | obt = psize; | |
615 | } | |
616 | ||
617 | if (err < 0) { | |
618 | alsa_logerr2 (err, typ, "Failed to set period %s to %d\n", | |
619 | size_in_usec ? "time" : "size", req->period_size); | |
620 | goto err; | |
621 | } | |
622 | ||
623 | if (((req->override_mask & 1) && (obt - req->period_size))) | |
624 | dolog ("Requested period %s %u was rejected, using %lu\n", | |
625 | size_in_usec ? "time" : "size", req->period_size, obt); | |
626 | } | |
627 | ||
628 | err = snd_pcm_hw_params (handle, hw_params); | |
629 | if (err < 0) { | |
630 | alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); | |
631 | goto err; | |
632 | } | |
633 | ||
634 | err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); | |
635 | if (err < 0) { | |
636 | alsa_logerr2 (err, typ, "Failed to get buffer size\n"); | |
637 | goto err; | |
638 | } | |
639 | ||
640 | err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); | |
641 | if (err < 0) { | |
642 | alsa_logerr2 (err, typ, "Failed to get format\n"); | |
643 | goto err; | |
644 | } | |
645 | ||
646 | if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { | |
647 | dolog ("Invalid format was returned %d\n", obtfmt); | |
648 | goto err; | |
649 | } | |
650 | ||
651 | err = snd_pcm_prepare (handle); | |
652 | if (err < 0) { | |
653 | alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); | |
654 | goto err; | |
655 | } | |
656 | ||
657 | if (!in && conf.threshold) { | |
658 | snd_pcm_uframes_t threshold; | |
659 | int bytes_per_sec; | |
660 | ||
661 | bytes_per_sec = freq << (nchannels == 2); | |
662 | ||
663 | switch (obt->fmt) { | |
664 | case AUD_FMT_S8: | |
665 | case AUD_FMT_U8: | |
666 | break; | |
667 | ||
668 | case AUD_FMT_S16: | |
669 | case AUD_FMT_U16: | |
670 | bytes_per_sec <<= 1; | |
671 | break; | |
672 | ||
673 | case AUD_FMT_S32: | |
674 | case AUD_FMT_U32: | |
675 | bytes_per_sec <<= 2; | |
676 | break; | |
677 | } | |
678 | ||
679 | threshold = (conf.threshold * bytes_per_sec) / 1000; | |
680 | alsa_set_threshold (handle, threshold); | |
681 | } | |
682 | ||
683 | obt->nchannels = nchannels; | |
684 | obt->freq = freq; | |
685 | obt->samples = obt_buffer_size; | |
686 | ||
687 | *handlep = handle; | |
688 | ||
689 | if (conf.verbose && | |
690 | (obtfmt != req->fmt || | |
691 | obt->nchannels != req->nchannels || | |
692 | obt->freq != req->freq)) { | |
693 | dolog ("Audio parameters for %s\n", typ); | |
694 | alsa_dump_info (req, obt, obtfmt); | |
695 | } | |
696 | ||
697 | #ifdef DEBUG | |
698 | alsa_dump_info (req, obt, obtfmt); | |
699 | #endif | |
700 | return 0; | |
701 | ||
702 | err: | |
703 | alsa_anal_close1 (&handle); | |
704 | return -1; | |
705 | } | |
706 | ||
707 | static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle) | |
708 | { | |
709 | snd_pcm_sframes_t avail; | |
710 | ||
711 | avail = snd_pcm_avail_update (handle); | |
712 | if (avail < 0) { | |
713 | if (avail == -EPIPE) { | |
714 | if (!alsa_recover (handle)) { | |
715 | avail = snd_pcm_avail_update (handle); | |
716 | } | |
717 | } | |
718 | ||
719 | if (avail < 0) { | |
720 | alsa_logerr (avail, | |
721 | "Could not obtain number of available frames\n"); | |
722 | return -1; | |
723 | } | |
724 | } | |
725 | ||
726 | return avail; | |
727 | } | |
728 | ||
729 | static void alsa_write_pending (ALSAVoiceOut *alsa) | |
730 | { | |
731 | HWVoiceOut *hw = &alsa->hw; | |
732 | ||
733 | while (alsa->pending) { | |
734 | int left_till_end_samples = hw->samples - alsa->wpos; | |
735 | int len = audio_MIN (alsa->pending, left_till_end_samples); | |
736 | char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift); | |
737 | ||
738 | while (len) { | |
739 | snd_pcm_sframes_t written; | |
740 | ||
741 | written = snd_pcm_writei (alsa->handle, src, len); | |
742 | ||
743 | if (written <= 0) { | |
744 | switch (written) { | |
745 | case 0: | |
746 | if (conf.verbose) { | |
747 | dolog ("Failed to write %d frames (wrote zero)\n", len); | |
748 | } | |
749 | return; | |
750 | ||
751 | case -EPIPE: | |
752 | if (alsa_recover (alsa->handle)) { | |
753 | alsa_logerr (written, "Failed to write %d frames\n", | |
754 | len); | |
755 | return; | |
756 | } | |
757 | if (conf.verbose) { | |
758 | dolog ("Recovering from playback xrun\n"); | |
759 | } | |
760 | continue; | |
761 | ||
762 | case -ESTRPIPE: | |
763 | /* stream is suspended and waiting for an | |
764 | application recovery */ | |
765 | if (alsa_resume (alsa->handle)) { | |
766 | alsa_logerr (written, "Failed to write %d frames\n", | |
767 | len); | |
768 | return; | |
769 | } | |
770 | if (conf.verbose) { | |
771 | dolog ("Resuming suspended output stream\n"); | |
772 | } | |
773 | continue; | |
774 | ||
775 | case -EAGAIN: | |
776 | return; | |
777 | ||
778 | default: | |
779 | alsa_logerr (written, "Failed to write %d frames from %p\n", | |
780 | len, src); | |
781 | return; | |
782 | } | |
783 | } | |
784 | ||
785 | alsa->wpos = (alsa->wpos + written) % hw->samples; | |
786 | alsa->pending -= written; | |
787 | len -= written; | |
788 | } | |
789 | } | |
790 | } | |
791 | ||
792 | static int alsa_run_out (HWVoiceOut *hw, int live) | |
793 | { | |
794 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
795 | int decr; | |
796 | snd_pcm_sframes_t avail; | |
797 | ||
798 | avail = alsa_get_avail (alsa->handle); | |
799 | if (avail < 0) { | |
800 | dolog ("Could not get number of available playback frames\n"); | |
801 | return 0; | |
802 | } | |
803 | ||
804 | decr = audio_MIN (live, avail); | |
805 | decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); | |
806 | alsa->pending += decr; | |
807 | alsa_write_pending (alsa); | |
808 | return decr; | |
809 | } | |
810 | ||
811 | static void alsa_fini_out (HWVoiceOut *hw) | |
812 | { | |
813 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
814 | ||
815 | ldebug ("alsa_fini\n"); | |
816 | alsa_anal_close (&alsa->handle, &alsa->pollhlp); | |
817 | ||
818 | if (alsa->pcm_buf) { | |
819 | g_free (alsa->pcm_buf); | |
820 | alsa->pcm_buf = NULL; | |
821 | } | |
822 | } | |
823 | ||
824 | static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) | |
825 | { | |
826 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
827 | struct alsa_params_req req; | |
828 | struct alsa_params_obt obt; | |
829 | snd_pcm_t *handle; | |
830 | struct audsettings obt_as; | |
831 | ||
832 | req.fmt = aud_to_alsafmt (as->fmt, as->endianness); | |
833 | req.freq = as->freq; | |
834 | req.nchannels = as->nchannels; | |
835 | req.period_size = conf.period_size_out; | |
836 | req.buffer_size = conf.buffer_size_out; | |
837 | req.size_in_usec = conf.size_in_usec_out; | |
838 | req.override_mask = | |
839 | (conf.period_size_out_overridden ? 1 : 0) | | |
840 | (conf.buffer_size_out_overridden ? 2 : 0); | |
841 | ||
842 | if (alsa_open (0, &req, &obt, &handle)) { | |
843 | return -1; | |
844 | } | |
845 | ||
846 | obt_as.freq = obt.freq; | |
847 | obt_as.nchannels = obt.nchannels; | |
848 | obt_as.fmt = obt.fmt; | |
849 | obt_as.endianness = obt.endianness; | |
850 | ||
851 | audio_pcm_init_info (&hw->info, &obt_as); | |
852 | hw->samples = obt.samples; | |
853 | ||
854 | alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift); | |
855 | if (!alsa->pcm_buf) { | |
856 | dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", | |
857 | hw->samples, 1 << hw->info.shift); | |
858 | alsa_anal_close1 (&handle); | |
859 | return -1; | |
860 | } | |
861 | ||
862 | alsa->handle = handle; | |
863 | return 0; | |
864 | } | |
865 | ||
866 | #define VOICE_CTL_PAUSE 0 | |
867 | #define VOICE_CTL_PREPARE 1 | |
868 | #define VOICE_CTL_START 2 | |
869 | ||
870 | static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) | |
871 | { | |
872 | int err; | |
873 | ||
874 | if (ctl == VOICE_CTL_PAUSE) { | |
875 | err = snd_pcm_drop (handle); | |
876 | if (err < 0) { | |
877 | alsa_logerr (err, "Could not stop %s\n", typ); | |
878 | return -1; | |
879 | } | |
880 | } | |
881 | else { | |
882 | err = snd_pcm_prepare (handle); | |
883 | if (err < 0) { | |
884 | alsa_logerr (err, "Could not prepare handle for %s\n", typ); | |
885 | return -1; | |
886 | } | |
887 | if (ctl == VOICE_CTL_START) { | |
888 | err = snd_pcm_start(handle); | |
889 | if (err < 0) { | |
890 | alsa_logerr (err, "Could not start handle for %s\n", typ); | |
891 | return -1; | |
892 | } | |
893 | } | |
894 | } | |
895 | ||
896 | return 0; | |
897 | } | |
898 | ||
899 | static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) | |
900 | { | |
901 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
902 | ||
903 | switch (cmd) { | |
904 | case VOICE_ENABLE: | |
905 | { | |
906 | va_list ap; | |
907 | int poll_mode; | |
908 | ||
909 | va_start (ap, cmd); | |
910 | poll_mode = va_arg (ap, int); | |
911 | va_end (ap); | |
912 | ||
913 | ldebug ("enabling voice\n"); | |
914 | if (poll_mode && alsa_poll_out (hw)) { | |
915 | poll_mode = 0; | |
916 | } | |
917 | hw->poll_mode = poll_mode; | |
918 | return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE); | |
919 | } | |
920 | ||
921 | case VOICE_DISABLE: | |
922 | ldebug ("disabling voice\n"); | |
923 | if (hw->poll_mode) { | |
924 | hw->poll_mode = 0; | |
925 | alsa_fini_poll (&alsa->pollhlp); | |
926 | } | |
927 | return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE); | |
928 | } | |
929 | ||
930 | return -1; | |
931 | } | |
932 | ||
933 | static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as) | |
934 | { | |
935 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
936 | struct alsa_params_req req; | |
937 | struct alsa_params_obt obt; | |
938 | snd_pcm_t *handle; | |
939 | struct audsettings obt_as; | |
940 | ||
941 | req.fmt = aud_to_alsafmt (as->fmt, as->endianness); | |
942 | req.freq = as->freq; | |
943 | req.nchannels = as->nchannels; | |
944 | req.period_size = conf.period_size_in; | |
945 | req.buffer_size = conf.buffer_size_in; | |
946 | req.size_in_usec = conf.size_in_usec_in; | |
947 | req.override_mask = | |
948 | (conf.period_size_in_overridden ? 1 : 0) | | |
949 | (conf.buffer_size_in_overridden ? 2 : 0); | |
950 | ||
951 | if (alsa_open (1, &req, &obt, &handle)) { | |
952 | return -1; | |
953 | } | |
954 | ||
955 | obt_as.freq = obt.freq; | |
956 | obt_as.nchannels = obt.nchannels; | |
957 | obt_as.fmt = obt.fmt; | |
958 | obt_as.endianness = obt.endianness; | |
959 | ||
960 | audio_pcm_init_info (&hw->info, &obt_as); | |
961 | hw->samples = obt.samples; | |
962 | ||
963 | alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); | |
964 | if (!alsa->pcm_buf) { | |
965 | dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", | |
966 | hw->samples, 1 << hw->info.shift); | |
967 | alsa_anal_close1 (&handle); | |
968 | return -1; | |
969 | } | |
970 | ||
971 | alsa->handle = handle; | |
972 | return 0; | |
973 | } | |
974 | ||
975 | static void alsa_fini_in (HWVoiceIn *hw) | |
976 | { | |
977 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
978 | ||
979 | alsa_anal_close (&alsa->handle, &alsa->pollhlp); | |
980 | ||
981 | if (alsa->pcm_buf) { | |
982 | g_free (alsa->pcm_buf); | |
983 | alsa->pcm_buf = NULL; | |
984 | } | |
985 | } | |
986 | ||
987 | static int alsa_run_in (HWVoiceIn *hw) | |
988 | { | |
989 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
990 | int hwshift = hw->info.shift; | |
991 | int i; | |
992 | int live = audio_pcm_hw_get_live_in (hw); | |
993 | int dead = hw->samples - live; | |
994 | int decr; | |
995 | struct { | |
996 | int add; | |
997 | int len; | |
998 | } bufs[2] = { | |
999 | { .add = hw->wpos, .len = 0 }, | |
1000 | { .add = 0, .len = 0 } | |
1001 | }; | |
1002 | snd_pcm_sframes_t avail; | |
1003 | snd_pcm_uframes_t read_samples = 0; | |
1004 | ||
1005 | if (!dead) { | |
1006 | return 0; | |
1007 | } | |
1008 | ||
1009 | avail = alsa_get_avail (alsa->handle); | |
1010 | if (avail < 0) { | |
1011 | dolog ("Could not get number of captured frames\n"); | |
1012 | return 0; | |
1013 | } | |
1014 | ||
1015 | if (!avail) { | |
1016 | snd_pcm_state_t state; | |
1017 | ||
1018 | state = snd_pcm_state (alsa->handle); | |
1019 | switch (state) { | |
1020 | case SND_PCM_STATE_PREPARED: | |
1021 | avail = hw->samples; | |
1022 | break; | |
1023 | case SND_PCM_STATE_SUSPENDED: | |
1024 | /* stream is suspended and waiting for an application recovery */ | |
1025 | if (alsa_resume (alsa->handle)) { | |
1026 | dolog ("Failed to resume suspended input stream\n"); | |
1027 | return 0; | |
1028 | } | |
1029 | if (conf.verbose) { | |
1030 | dolog ("Resuming suspended input stream\n"); | |
1031 | } | |
1032 | break; | |
1033 | default: | |
1034 | if (conf.verbose) { | |
1035 | dolog ("No frames available and ALSA state is %d\n", state); | |
1036 | } | |
1037 | return 0; | |
1038 | } | |
1039 | } | |
1040 | ||
1041 | decr = audio_MIN (dead, avail); | |
1042 | if (!decr) { | |
1043 | return 0; | |
1044 | } | |
1045 | ||
1046 | if (hw->wpos + decr > hw->samples) { | |
1047 | bufs[0].len = (hw->samples - hw->wpos); | |
1048 | bufs[1].len = (decr - (hw->samples - hw->wpos)); | |
1049 | } | |
1050 | else { | |
1051 | bufs[0].len = decr; | |
1052 | } | |
1053 | ||
1054 | for (i = 0; i < 2; ++i) { | |
1055 | void *src; | |
1056 | struct st_sample *dst; | |
1057 | snd_pcm_sframes_t nread; | |
1058 | snd_pcm_uframes_t len; | |
1059 | ||
1060 | len = bufs[i].len; | |
1061 | ||
1062 | src = advance (alsa->pcm_buf, bufs[i].add << hwshift); | |
1063 | dst = hw->conv_buf + bufs[i].add; | |
1064 | ||
1065 | while (len) { | |
1066 | nread = snd_pcm_readi (alsa->handle, src, len); | |
1067 | ||
1068 | if (nread <= 0) { | |
1069 | switch (nread) { | |
1070 | case 0: | |
1071 | if (conf.verbose) { | |
1072 | dolog ("Failed to read %ld frames (read zero)\n", len); | |
1073 | } | |
1074 | goto exit; | |
1075 | ||
1076 | case -EPIPE: | |
1077 | if (alsa_recover (alsa->handle)) { | |
1078 | alsa_logerr (nread, "Failed to read %ld frames\n", len); | |
1079 | goto exit; | |
1080 | } | |
1081 | if (conf.verbose) { | |
1082 | dolog ("Recovering from capture xrun\n"); | |
1083 | } | |
1084 | continue; | |
1085 | ||
1086 | case -EAGAIN: | |
1087 | goto exit; | |
1088 | ||
1089 | default: | |
1090 | alsa_logerr ( | |
1091 | nread, | |
1092 | "Failed to read %ld frames from %p\n", | |
1093 | len, | |
1094 | src | |
1095 | ); | |
1096 | goto exit; | |
1097 | } | |
1098 | } | |
1099 | ||
1100 | hw->conv (dst, src, nread); | |
1101 | ||
1102 | src = advance (src, nread << hwshift); | |
1103 | dst += nread; | |
1104 | ||
1105 | read_samples += nread; | |
1106 | len -= nread; | |
1107 | } | |
1108 | } | |
1109 | ||
1110 | exit: | |
1111 | hw->wpos = (hw->wpos + read_samples) % hw->samples; | |
1112 | return read_samples; | |
1113 | } | |
1114 | ||
1115 | static int alsa_read (SWVoiceIn *sw, void *buf, int size) | |
1116 | { | |
1117 | return audio_pcm_sw_read (sw, buf, size); | |
1118 | } | |
1119 | ||
1120 | static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) | |
1121 | { | |
1122 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
1123 | ||
1124 | switch (cmd) { | |
1125 | case VOICE_ENABLE: | |
1126 | { | |
1127 | va_list ap; | |
1128 | int poll_mode; | |
1129 | ||
1130 | va_start (ap, cmd); | |
1131 | poll_mode = va_arg (ap, int); | |
1132 | va_end (ap); | |
1133 | ||
1134 | ldebug ("enabling voice\n"); | |
1135 | if (poll_mode && alsa_poll_in (hw)) { | |
1136 | poll_mode = 0; | |
1137 | } | |
1138 | hw->poll_mode = poll_mode; | |
1139 | ||
1140 | return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START); | |
1141 | } | |
1142 | ||
1143 | case VOICE_DISABLE: | |
1144 | ldebug ("disabling voice\n"); | |
1145 | if (hw->poll_mode) { | |
1146 | hw->poll_mode = 0; | |
1147 | alsa_fini_poll (&alsa->pollhlp); | |
1148 | } | |
1149 | return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE); | |
1150 | } | |
1151 | ||
1152 | return -1; | |
1153 | } | |
1154 | ||
1155 | static void *alsa_audio_init (void) | |
1156 | { | |
1157 | return &conf; | |
1158 | } | |
1159 | ||
1160 | static void alsa_audio_fini (void *opaque) | |
1161 | { | |
1162 | (void) opaque; | |
1163 | } | |
1164 | ||
1165 | static struct audio_option alsa_options[] = { | |
1166 | { | |
1167 | .name = "DAC_SIZE_IN_USEC", | |
1168 | .tag = AUD_OPT_BOOL, | |
1169 | .valp = &conf.size_in_usec_out, | |
1170 | .descr = "DAC period/buffer size in microseconds (otherwise in frames)" | |
1171 | }, | |
1172 | { | |
1173 | .name = "DAC_PERIOD_SIZE", | |
1174 | .tag = AUD_OPT_INT, | |
1175 | .valp = &conf.period_size_out, | |
1176 | .descr = "DAC period size (0 to go with system default)", | |
1177 | .overriddenp = &conf.period_size_out_overridden | |
1178 | }, | |
1179 | { | |
1180 | .name = "DAC_BUFFER_SIZE", | |
1181 | .tag = AUD_OPT_INT, | |
1182 | .valp = &conf.buffer_size_out, | |
1183 | .descr = "DAC buffer size (0 to go with system default)", | |
1184 | .overriddenp = &conf.buffer_size_out_overridden | |
1185 | }, | |
1186 | { | |
1187 | .name = "ADC_SIZE_IN_USEC", | |
1188 | .tag = AUD_OPT_BOOL, | |
1189 | .valp = &conf.size_in_usec_in, | |
1190 | .descr = | |
1191 | "ADC period/buffer size in microseconds (otherwise in frames)" | |
1192 | }, | |
1193 | { | |
1194 | .name = "ADC_PERIOD_SIZE", | |
1195 | .tag = AUD_OPT_INT, | |
1196 | .valp = &conf.period_size_in, | |
1197 | .descr = "ADC period size (0 to go with system default)", | |
1198 | .overriddenp = &conf.period_size_in_overridden | |
1199 | }, | |
1200 | { | |
1201 | .name = "ADC_BUFFER_SIZE", | |
1202 | .tag = AUD_OPT_INT, | |
1203 | .valp = &conf.buffer_size_in, | |
1204 | .descr = "ADC buffer size (0 to go with system default)", | |
1205 | .overriddenp = &conf.buffer_size_in_overridden | |
1206 | }, | |
1207 | { | |
1208 | .name = "THRESHOLD", | |
1209 | .tag = AUD_OPT_INT, | |
1210 | .valp = &conf.threshold, | |
1211 | .descr = "(undocumented)" | |
1212 | }, | |
1213 | { | |
1214 | .name = "DAC_DEV", | |
1215 | .tag = AUD_OPT_STR, | |
1216 | .valp = &conf.pcm_name_out, | |
1217 | .descr = "DAC device name (for instance dmix)" | |
1218 | }, | |
1219 | { | |
1220 | .name = "ADC_DEV", | |
1221 | .tag = AUD_OPT_STR, | |
1222 | .valp = &conf.pcm_name_in, | |
1223 | .descr = "ADC device name" | |
1224 | }, | |
1225 | { | |
1226 | .name = "VERBOSE", | |
1227 | .tag = AUD_OPT_BOOL, | |
1228 | .valp = &conf.verbose, | |
1229 | .descr = "Behave in a more verbose way" | |
1230 | }, | |
1231 | { /* End of list */ } | |
1232 | }; | |
1233 | ||
1234 | static struct audio_pcm_ops alsa_pcm_ops = { | |
1235 | .init_out = alsa_init_out, | |
1236 | .fini_out = alsa_fini_out, | |
1237 | .run_out = alsa_run_out, | |
1238 | .write = alsa_write, | |
1239 | .ctl_out = alsa_ctl_out, | |
1240 | ||
1241 | .init_in = alsa_init_in, | |
1242 | .fini_in = alsa_fini_in, | |
1243 | .run_in = alsa_run_in, | |
1244 | .read = alsa_read, | |
1245 | .ctl_in = alsa_ctl_in, | |
1246 | }; | |
1247 | ||
1248 | struct audio_driver alsa_audio_driver = { | |
1249 | .name = "alsa", | |
1250 | .descr = "ALSA http://www.alsa-project.org", | |
1251 | .options = alsa_options, | |
1252 | .init = alsa_audio_init, | |
1253 | .fini = alsa_audio_fini, | |
1254 | .pcm_ops = &alsa_pcm_ops, | |
1255 | .can_be_default = 1, | |
1256 | .max_voices_out = INT_MAX, | |
1257 | .max_voices_in = INT_MAX, | |
1258 | .voice_size_out = sizeof (ALSAVoiceOut), | |
1259 | .voice_size_in = sizeof (ALSAVoiceIn) | |
1260 | }; |