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1d14ffa9 FB |
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 "vl.h" | |
26 | ||
27 | #define AUDIO_CAP "alsa" | |
28 | #include "audio_int.h" | |
29 | ||
30 | typedef struct ALSAVoiceOut { | |
31 | HWVoiceOut hw; | |
32 | void *pcm_buf; | |
33 | snd_pcm_t *handle; | |
34 | int can_pause; | |
35 | int was_enabled; | |
36 | } ALSAVoiceOut; | |
37 | ||
38 | typedef struct ALSAVoiceIn { | |
39 | HWVoiceIn hw; | |
40 | snd_pcm_t *handle; | |
41 | void *pcm_buf; | |
42 | int can_pause; | |
43 | } ALSAVoiceIn; | |
44 | ||
45 | static struct { | |
46 | int size_in_usec_in; | |
47 | int size_in_usec_out; | |
48 | const char *pcm_name_in; | |
49 | const char *pcm_name_out; | |
50 | unsigned int buffer_size_in; | |
51 | unsigned int period_size_in; | |
52 | unsigned int buffer_size_out; | |
53 | unsigned int period_size_out; | |
54 | unsigned int threshold; | |
55 | ||
56 | int buffer_size_in_overriden; | |
57 | int period_size_in_overriden; | |
58 | ||
59 | int buffer_size_out_overriden; | |
60 | int period_size_out_overriden; | |
61 | } conf = { | |
62 | #ifdef HIGH_LATENCY | |
63 | .size_in_usec_in = 1, | |
64 | .size_in_usec_out = 1, | |
65 | #endif | |
66 | .pcm_name_out = "hw:0,0", | |
67 | .pcm_name_in = "hw:0,0", | |
68 | #ifdef HIGH_LATENCY | |
69 | .buffer_size_in = 400000, | |
70 | .period_size_in = 400000 / 4, | |
71 | .buffer_size_out = 400000, | |
72 | .period_size_out = 400000 / 4, | |
73 | #else | |
74 | #define DEFAULT_BUFFER_SIZE 1024 | |
75 | #define DEFAULT_PERIOD_SIZE 256 | |
76 | .buffer_size_in = DEFAULT_BUFFER_SIZE, | |
77 | .period_size_in = DEFAULT_PERIOD_SIZE, | |
78 | .buffer_size_out = DEFAULT_BUFFER_SIZE, | |
79 | .period_size_out = DEFAULT_PERIOD_SIZE, | |
80 | .buffer_size_in_overriden = 0, | |
81 | .buffer_size_out_overriden = 0, | |
82 | .period_size_in_overriden = 0, | |
83 | .period_size_out_overriden = 0, | |
84 | #endif | |
85 | .threshold = 0 | |
86 | }; | |
87 | ||
88 | struct alsa_params_req { | |
89 | int freq; | |
90 | audfmt_e fmt; | |
91 | int nchannels; | |
92 | unsigned int buffer_size; | |
93 | unsigned int period_size; | |
94 | }; | |
95 | ||
96 | struct alsa_params_obt { | |
97 | int freq; | |
98 | audfmt_e fmt; | |
99 | int nchannels; | |
100 | int can_pause; | |
c0fe3827 | 101 | snd_pcm_uframes_t samples; |
1d14ffa9 FB |
102 | }; |
103 | ||
104 | static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) | |
105 | { | |
106 | va_list ap; | |
107 | ||
108 | va_start (ap, fmt); | |
109 | AUD_vlog (AUDIO_CAP, fmt, ap); | |
110 | va_end (ap); | |
111 | ||
112 | AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); | |
113 | } | |
114 | ||
115 | static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( | |
116 | int err, | |
117 | const char *typ, | |
118 | const char *fmt, | |
119 | ... | |
120 | ) | |
121 | { | |
122 | va_list ap; | |
123 | ||
c0fe3827 | 124 | AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); |
1d14ffa9 FB |
125 | |
126 | va_start (ap, fmt); | |
127 | AUD_vlog (AUDIO_CAP, fmt, ap); | |
128 | va_end (ap); | |
129 | ||
130 | AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); | |
131 | } | |
132 | ||
133 | static void alsa_anal_close (snd_pcm_t **handlep) | |
134 | { | |
135 | int err = snd_pcm_close (*handlep); | |
136 | if (err) { | |
137 | alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); | |
138 | } | |
139 | *handlep = NULL; | |
140 | } | |
141 | ||
142 | static int alsa_write (SWVoiceOut *sw, void *buf, int len) | |
143 | { | |
144 | return audio_pcm_sw_write (sw, buf, len); | |
145 | } | |
146 | ||
147 | static int aud_to_alsafmt (audfmt_e fmt) | |
148 | { | |
149 | switch (fmt) { | |
150 | case AUD_FMT_S8: | |
151 | return SND_PCM_FORMAT_S8; | |
152 | ||
153 | case AUD_FMT_U8: | |
154 | return SND_PCM_FORMAT_U8; | |
155 | ||
156 | case AUD_FMT_S16: | |
157 | return SND_PCM_FORMAT_S16_LE; | |
158 | ||
159 | case AUD_FMT_U16: | |
160 | return SND_PCM_FORMAT_U16_LE; | |
161 | ||
162 | default: | |
163 | dolog ("Internal logic error: Bad audio format %d\n", fmt); | |
164 | #ifdef DEBUG_AUDIO | |
165 | abort (); | |
166 | #endif | |
167 | return SND_PCM_FORMAT_U8; | |
168 | } | |
169 | } | |
170 | ||
171 | static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) | |
172 | { | |
173 | switch (alsafmt) { | |
174 | case SND_PCM_FORMAT_S8: | |
175 | *endianness = 0; | |
176 | *fmt = AUD_FMT_S8; | |
177 | break; | |
178 | ||
179 | case SND_PCM_FORMAT_U8: | |
180 | *endianness = 0; | |
181 | *fmt = AUD_FMT_U8; | |
182 | break; | |
183 | ||
184 | case SND_PCM_FORMAT_S16_LE: | |
185 | *endianness = 0; | |
186 | *fmt = AUD_FMT_S16; | |
187 | break; | |
188 | ||
189 | case SND_PCM_FORMAT_U16_LE: | |
190 | *endianness = 0; | |
191 | *fmt = AUD_FMT_U16; | |
192 | break; | |
193 | ||
194 | case SND_PCM_FORMAT_S16_BE: | |
195 | *endianness = 1; | |
196 | *fmt = AUD_FMT_S16; | |
197 | break; | |
198 | ||
199 | case SND_PCM_FORMAT_U16_BE: | |
200 | *endianness = 1; | |
201 | *fmt = AUD_FMT_U16; | |
202 | break; | |
203 | ||
204 | default: | |
205 | dolog ("Unrecognized audio format %d\n", alsafmt); | |
206 | return -1; | |
207 | } | |
208 | ||
209 | return 0; | |
210 | } | |
211 | ||
c0fe3827 | 212 | #if defined DEBUG_MISMATCHES || defined DEBUG |
1d14ffa9 FB |
213 | static void alsa_dump_info (struct alsa_params_req *req, |
214 | struct alsa_params_obt *obt) | |
215 | { | |
216 | dolog ("parameter | requested value | obtained value\n"); | |
217 | dolog ("format | %10d | %10d\n", req->fmt, obt->fmt); | |
218 | dolog ("channels | %10d | %10d\n", | |
219 | req->nchannels, obt->nchannels); | |
220 | dolog ("frequency | %10d | %10d\n", req->freq, obt->freq); | |
221 | dolog ("============================================\n"); | |
222 | dolog ("requested: buffer size %d period size %d\n", | |
223 | req->buffer_size, req->period_size); | |
c0fe3827 | 224 | dolog ("obtained: samples %ld\n", obt->samples); |
1d14ffa9 FB |
225 | } |
226 | #endif | |
227 | ||
228 | static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) | |
229 | { | |
230 | int err; | |
231 | snd_pcm_sw_params_t *sw_params; | |
232 | ||
233 | snd_pcm_sw_params_alloca (&sw_params); | |
234 | ||
235 | err = snd_pcm_sw_params_current (handle, sw_params); | |
236 | if (err < 0) { | |
c0fe3827 | 237 | dolog ("Could not fully initialize DAC\n"); |
1d14ffa9 FB |
238 | alsa_logerr (err, "Failed to get current software parameters\n"); |
239 | return; | |
240 | } | |
241 | ||
242 | err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); | |
243 | if (err < 0) { | |
c0fe3827 | 244 | dolog ("Could not fully initialize DAC\n"); |
1d14ffa9 FB |
245 | alsa_logerr (err, "Failed to set software threshold to %ld\n", |
246 | threshold); | |
247 | return; | |
248 | } | |
249 | ||
250 | err = snd_pcm_sw_params (handle, sw_params); | |
251 | if (err < 0) { | |
c0fe3827 | 252 | dolog ("Could not fully initialize DAC\n"); |
1d14ffa9 FB |
253 | alsa_logerr (err, "Failed to set software parameters\n"); |
254 | return; | |
255 | } | |
256 | } | |
257 | ||
258 | static int alsa_open (int in, struct alsa_params_req *req, | |
259 | struct alsa_params_obt *obt, snd_pcm_t **handlep) | |
260 | { | |
261 | snd_pcm_t *handle; | |
262 | snd_pcm_hw_params_t *hw_params; | |
263 | int err, freq, nchannels; | |
264 | const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; | |
265 | unsigned int period_size, buffer_size; | |
266 | snd_pcm_uframes_t obt_buffer_size; | |
267 | const char *typ = in ? "ADC" : "DAC"; | |
268 | ||
269 | freq = req->freq; | |
270 | period_size = req->period_size; | |
271 | buffer_size = req->buffer_size; | |
272 | nchannels = req->nchannels; | |
273 | ||
274 | snd_pcm_hw_params_alloca (&hw_params); | |
275 | ||
276 | err = snd_pcm_open ( | |
277 | &handle, | |
278 | pcm_name, | |
279 | in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, | |
280 | SND_PCM_NONBLOCK | |
281 | ); | |
282 | if (err < 0) { | |
283 | alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); | |
284 | return -1; | |
285 | } | |
286 | ||
287 | err = snd_pcm_hw_params_any (handle, hw_params); | |
288 | if (err < 0) { | |
289 | alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); | |
290 | goto err; | |
291 | } | |
292 | ||
293 | err = snd_pcm_hw_params_set_access ( | |
294 | handle, | |
295 | hw_params, | |
296 | SND_PCM_ACCESS_RW_INTERLEAVED | |
297 | ); | |
298 | if (err < 0) { | |
299 | alsa_logerr2 (err, typ, "Failed to set access type\n"); | |
300 | goto err; | |
301 | } | |
302 | ||
303 | err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); | |
304 | if (err < 0) { | |
305 | alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); | |
306 | goto err; | |
307 | } | |
308 | ||
309 | err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); | |
310 | if (err < 0) { | |
311 | alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); | |
312 | goto err; | |
313 | } | |
314 | ||
315 | err = snd_pcm_hw_params_set_channels_near ( | |
316 | handle, | |
317 | hw_params, | |
318 | &nchannels | |
319 | ); | |
320 | if (err < 0) { | |
321 | alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", | |
322 | req->nchannels); | |
323 | goto err; | |
324 | } | |
325 | ||
326 | if (nchannels != 1 && nchannels != 2) { | |
327 | alsa_logerr2 (err, typ, | |
328 | "Can not handle obtained number of channels %d\n", | |
329 | nchannels); | |
330 | goto err; | |
331 | } | |
332 | ||
333 | if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) { | |
334 | if (!buffer_size) { | |
335 | buffer_size = DEFAULT_BUFFER_SIZE; | |
336 | period_size= DEFAULT_PERIOD_SIZE; | |
337 | } | |
338 | } | |
339 | ||
340 | if (buffer_size) { | |
341 | if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) { | |
342 | if (period_size) { | |
343 | err = snd_pcm_hw_params_set_period_time_near ( | |
344 | handle, | |
345 | hw_params, | |
346 | &period_size, | |
c0fe3827 FB |
347 | 0 |
348 | ); | |
1d14ffa9 FB |
349 | if (err < 0) { |
350 | alsa_logerr2 (err, typ, | |
351 | "Failed to set period time %d\n", | |
352 | req->period_size); | |
353 | goto err; | |
354 | } | |
355 | } | |
356 | ||
357 | err = snd_pcm_hw_params_set_buffer_time_near ( | |
358 | handle, | |
359 | hw_params, | |
360 | &buffer_size, | |
c0fe3827 FB |
361 | 0 |
362 | ); | |
1d14ffa9 FB |
363 | |
364 | if (err < 0) { | |
365 | alsa_logerr2 (err, typ, | |
366 | "Failed to set buffer time %d\n", | |
367 | req->buffer_size); | |
368 | goto err; | |
369 | } | |
370 | } | |
371 | else { | |
372 | int dir; | |
373 | snd_pcm_uframes_t minval; | |
374 | ||
375 | if (period_size) { | |
376 | minval = period_size; | |
377 | dir = 0; | |
378 | ||
379 | err = snd_pcm_hw_params_get_period_size_min ( | |
380 | hw_params, | |
381 | &minval, | |
382 | &dir | |
383 | ); | |
384 | if (err < 0) { | |
385 | alsa_logerr ( | |
386 | err, | |
c0fe3827 | 387 | "Could not get minmal period size for %s\n", |
1d14ffa9 FB |
388 | typ |
389 | ); | |
390 | } | |
391 | else { | |
392 | if (period_size < minval) { | |
393 | if ((in && conf.period_size_in_overriden) | |
394 | || (!in && conf.period_size_out_overriden)) { | |
395 | dolog ("%s period size(%d) is less " | |
396 | "than minmal period size(%ld)\n", | |
397 | typ, | |
398 | period_size, | |
399 | minval); | |
400 | } | |
401 | period_size = minval; | |
402 | } | |
403 | } | |
404 | ||
405 | err = snd_pcm_hw_params_set_period_size ( | |
406 | handle, | |
407 | hw_params, | |
408 | period_size, | |
409 | 0 | |
410 | ); | |
411 | if (err < 0) { | |
412 | alsa_logerr2 (err, typ, "Failed to set period size %d\n", | |
413 | req->period_size); | |
414 | goto err; | |
415 | } | |
416 | } | |
417 | ||
418 | minval = buffer_size; | |
419 | err = snd_pcm_hw_params_get_buffer_size_min ( | |
420 | hw_params, | |
421 | &minval | |
422 | ); | |
423 | if (err < 0) { | |
c0fe3827 | 424 | alsa_logerr (err, "Could not get minmal buffer size for %s\n", |
1d14ffa9 FB |
425 | typ); |
426 | } | |
427 | else { | |
428 | if (buffer_size < minval) { | |
429 | if ((in && conf.buffer_size_in_overriden) | |
430 | || (!in && conf.buffer_size_out_overriden)) { | |
431 | dolog ( | |
432 | "%s buffer size(%d) is less " | |
433 | "than minimal buffer size(%ld)\n", | |
434 | typ, | |
435 | buffer_size, | |
436 | minval | |
437 | ); | |
438 | } | |
439 | buffer_size = minval; | |
440 | } | |
441 | } | |
442 | ||
443 | err = snd_pcm_hw_params_set_buffer_size ( | |
444 | handle, | |
445 | hw_params, | |
446 | buffer_size | |
447 | ); | |
448 | if (err < 0) { | |
449 | alsa_logerr2 (err, typ, "Failed to set buffer size %d\n", | |
450 | req->buffer_size); | |
451 | goto err; | |
452 | } | |
453 | } | |
454 | } | |
455 | else { | |
c0fe3827 | 456 | dolog ("warning: Buffer size is not set\n"); |
1d14ffa9 FB |
457 | } |
458 | ||
459 | err = snd_pcm_hw_params (handle, hw_params); | |
460 | if (err < 0) { | |
461 | alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); | |
462 | goto err; | |
463 | } | |
464 | ||
465 | err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); | |
466 | if (err < 0) { | |
467 | alsa_logerr2 (err, typ, "Failed to get buffer size\n"); | |
468 | goto err; | |
469 | } | |
470 | ||
471 | err = snd_pcm_prepare (handle); | |
472 | if (err < 0) { | |
c0fe3827 | 473 | alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); |
1d14ffa9 FB |
474 | goto err; |
475 | } | |
476 | ||
477 | obt->can_pause = snd_pcm_hw_params_can_pause (hw_params); | |
478 | if (obt->can_pause < 0) { | |
c0fe3827 | 479 | alsa_logerr (err, "Could not get pause capability for %s\n", typ); |
1d14ffa9 FB |
480 | obt->can_pause = 0; |
481 | } | |
482 | ||
483 | if (!in && conf.threshold) { | |
484 | snd_pcm_uframes_t threshold; | |
485 | int bytes_per_sec; | |
486 | ||
487 | bytes_per_sec = freq | |
488 | << (nchannels == 2) | |
489 | << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); | |
490 | ||
491 | threshold = (conf.threshold * bytes_per_sec) / 1000; | |
492 | alsa_set_threshold (handle, threshold); | |
493 | } | |
494 | ||
495 | obt->fmt = req->fmt; | |
496 | obt->nchannels = nchannels; | |
497 | obt->freq = freq; | |
c0fe3827 | 498 | obt->samples = obt_buffer_size; |
1d14ffa9 FB |
499 | *handlep = handle; |
500 | ||
c0fe3827 | 501 | #if defined DEBUG_MISMATCHES || defined DEBUG |
1d14ffa9 FB |
502 | if (obt->fmt != req->fmt || |
503 | obt->nchannels != req->nchannels || | |
504 | obt->freq != req->freq) { | |
1d14ffa9 FB |
505 | dolog ("Audio paramters mismatch for %s\n", typ); |
506 | alsa_dump_info (req, obt); | |
1d14ffa9 | 507 | } |
c0fe3827 | 508 | #endif |
1d14ffa9 FB |
509 | |
510 | #ifdef DEBUG | |
511 | alsa_dump_info (req, obt); | |
512 | #endif | |
513 | return 0; | |
514 | ||
515 | err: | |
516 | alsa_anal_close (&handle); | |
517 | return -1; | |
518 | } | |
519 | ||
520 | static int alsa_recover (snd_pcm_t *handle) | |
521 | { | |
522 | int err = snd_pcm_prepare (handle); | |
523 | if (err < 0) { | |
524 | alsa_logerr (err, "Failed to prepare handle %p\n", handle); | |
525 | return -1; | |
526 | } | |
527 | return 0; | |
528 | } | |
529 | ||
530 | static int alsa_run_out (HWVoiceOut *hw) | |
531 | { | |
532 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
533 | int rpos, live, decr; | |
534 | int samples; | |
535 | uint8_t *dst; | |
536 | st_sample_t *src; | |
537 | snd_pcm_sframes_t avail; | |
538 | ||
539 | live = audio_pcm_hw_get_live_out (hw); | |
540 | if (!live) { | |
541 | return 0; | |
542 | } | |
543 | ||
544 | avail = snd_pcm_avail_update (alsa->handle); | |
545 | if (avail < 0) { | |
546 | if (avail == -EPIPE) { | |
547 | if (!alsa_recover (alsa->handle)) { | |
548 | avail = snd_pcm_avail_update (alsa->handle); | |
549 | if (avail >= 0) { | |
550 | goto ok; | |
551 | } | |
552 | } | |
553 | } | |
554 | ||
c0fe3827 | 555 | alsa_logerr (avail, "Could not get amount free space\n"); |
1d14ffa9 FB |
556 | return 0; |
557 | } | |
558 | ||
559 | ok: | |
560 | decr = audio_MIN (live, avail); | |
561 | samples = decr; | |
562 | rpos = hw->rpos; | |
563 | while (samples) { | |
564 | int left_till_end_samples = hw->samples - rpos; | |
565 | int convert_samples = audio_MIN (samples, left_till_end_samples); | |
566 | snd_pcm_sframes_t written; | |
567 | ||
568 | src = hw->mix_buf + rpos; | |
569 | dst = advance (alsa->pcm_buf, rpos << hw->info.shift); | |
570 | ||
571 | hw->clip (dst, src, convert_samples); | |
572 | ||
4787c71d FB |
573 | while (convert_samples) { |
574 | written = snd_pcm_writei (alsa->handle, dst, convert_samples); | |
575 | ||
576 | if (written < 0) { | |
577 | switch (written) { | |
578 | case -EPIPE: | |
579 | if (!alsa_recover (alsa->handle)) { | |
580 | continue; | |
581 | } | |
582 | dolog ("Failed to write %d frames to %p, " | |
583 | "handle %p not prepared\n", | |
584 | convert_samples, | |
585 | dst, | |
586 | alsa->handle); | |
587 | goto exit; | |
588 | ||
589 | case -EAGAIN: | |
590 | continue; | |
591 | ||
592 | default: | |
593 | alsa_logerr (written, "Failed to write %d frames to %p\n", | |
594 | convert_samples, dst); | |
595 | goto exit; | |
1d14ffa9 | 596 | } |
1d14ffa9 | 597 | } |
1d14ffa9 | 598 | |
4787c71d FB |
599 | mixeng_clear (src, written); |
600 | rpos = (rpos + written) % hw->samples; | |
601 | samples -= written; | |
602 | convert_samples -= written; | |
603 | dst = advance (dst, written << hw->info.shift); | |
604 | src += written; | |
605 | } | |
1d14ffa9 FB |
606 | } |
607 | ||
608 | exit: | |
609 | hw->rpos = rpos; | |
610 | return decr; | |
611 | } | |
612 | ||
613 | static void alsa_fini_out (HWVoiceOut *hw) | |
614 | { | |
615 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
616 | ||
617 | ldebug ("alsa_fini\n"); | |
618 | alsa_anal_close (&alsa->handle); | |
619 | ||
620 | if (alsa->pcm_buf) { | |
621 | qemu_free (alsa->pcm_buf); | |
622 | alsa->pcm_buf = NULL; | |
623 | } | |
624 | } | |
625 | ||
c0fe3827 | 626 | static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
1d14ffa9 FB |
627 | { |
628 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
629 | struct alsa_params_req req; | |
630 | struct alsa_params_obt obt; | |
631 | audfmt_e effective_fmt; | |
632 | int endianness; | |
633 | int err; | |
634 | snd_pcm_t *handle; | |
c0fe3827 | 635 | audsettings_t obt_as; |
1d14ffa9 | 636 | |
c0fe3827 FB |
637 | req.fmt = aud_to_alsafmt (as->fmt); |
638 | req.freq = as->freq; | |
639 | req.nchannels = as->nchannels; | |
1d14ffa9 FB |
640 | req.period_size = conf.period_size_out; |
641 | req.buffer_size = conf.buffer_size_out; | |
642 | ||
643 | if (alsa_open (0, &req, &obt, &handle)) { | |
644 | return -1; | |
645 | } | |
646 | ||
647 | err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); | |
648 | if (err) { | |
649 | alsa_anal_close (&handle); | |
650 | return -1; | |
651 | } | |
652 | ||
c0fe3827 FB |
653 | obt_as.freq = obt.freq; |
654 | obt_as.nchannels = obt.nchannels; | |
655 | obt_as.fmt = effective_fmt; | |
656 | ||
1d14ffa9 FB |
657 | audio_pcm_init_info ( |
658 | &hw->info, | |
c0fe3827 | 659 | &obt_as, |
1d14ffa9 FB |
660 | audio_need_to_swap_endian (endianness) |
661 | ); | |
662 | alsa->can_pause = obt.can_pause; | |
c0fe3827 | 663 | hw->samples = obt.samples; |
1d14ffa9 | 664 | |
c0fe3827 | 665 | alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift); |
1d14ffa9 | 666 | if (!alsa->pcm_buf) { |
4787c71d FB |
667 | dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", |
668 | hw->samples, 1 << hw->info.shift); | |
1d14ffa9 FB |
669 | alsa_anal_close (&handle); |
670 | return -1; | |
671 | } | |
672 | ||
673 | alsa->handle = handle; | |
674 | alsa->was_enabled = 0; | |
675 | return 0; | |
676 | } | |
677 | ||
678 | static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) | |
679 | { | |
680 | int err; | |
681 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; | |
682 | ||
683 | switch (cmd) { | |
684 | case VOICE_ENABLE: | |
685 | ldebug ("enabling voice\n"); | |
686 | audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples); | |
687 | if (alsa->can_pause) { | |
688 | /* Why this was_enabled madness is needed at all?? */ | |
689 | if (alsa->was_enabled) { | |
690 | err = snd_pcm_pause (alsa->handle, 0); | |
691 | if (err < 0) { | |
692 | alsa_logerr (err, "Failed to resume playing\n"); | |
693 | /* not fatal really */ | |
694 | } | |
695 | } | |
696 | else { | |
697 | alsa->was_enabled = 1; | |
698 | } | |
699 | } | |
700 | break; | |
701 | ||
702 | case VOICE_DISABLE: | |
703 | ldebug ("disabling voice\n"); | |
704 | if (alsa->can_pause) { | |
705 | err = snd_pcm_pause (alsa->handle, 1); | |
706 | if (err < 0) { | |
707 | alsa_logerr (err, "Failed to stop playing\n"); | |
708 | /* not fatal really */ | |
709 | } | |
710 | } | |
711 | break; | |
712 | } | |
713 | return 0; | |
714 | } | |
715 | ||
c0fe3827 | 716 | static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
1d14ffa9 FB |
717 | { |
718 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
719 | struct alsa_params_req req; | |
720 | struct alsa_params_obt obt; | |
721 | int endianness; | |
722 | int err; | |
723 | audfmt_e effective_fmt; | |
724 | snd_pcm_t *handle; | |
c0fe3827 | 725 | audsettings_t obt_as; |
1d14ffa9 | 726 | |
c0fe3827 FB |
727 | req.fmt = aud_to_alsafmt (as->fmt); |
728 | req.freq = as->freq; | |
729 | req.nchannels = as->nchannels; | |
1d14ffa9 FB |
730 | req.period_size = conf.period_size_in; |
731 | req.buffer_size = conf.buffer_size_in; | |
732 | ||
733 | if (alsa_open (1, &req, &obt, &handle)) { | |
734 | return -1; | |
735 | } | |
736 | ||
737 | err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); | |
738 | if (err) { | |
739 | alsa_anal_close (&handle); | |
740 | return -1; | |
741 | } | |
742 | ||
c0fe3827 FB |
743 | obt_as.freq = obt.freq; |
744 | obt_as.nchannels = obt.nchannels; | |
745 | obt_as.fmt = effective_fmt; | |
746 | ||
1d14ffa9 FB |
747 | audio_pcm_init_info ( |
748 | &hw->info, | |
c0fe3827 | 749 | &obt_as, |
1d14ffa9 FB |
750 | audio_need_to_swap_endian (endianness) |
751 | ); | |
752 | alsa->can_pause = obt.can_pause; | |
c0fe3827 FB |
753 | hw->samples = obt.samples; |
754 | ||
755 | alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); | |
1d14ffa9 | 756 | if (!alsa->pcm_buf) { |
4787c71d FB |
757 | dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", |
758 | hw->samples, 1 << hw->info.shift); | |
1d14ffa9 FB |
759 | alsa_anal_close (&handle); |
760 | return -1; | |
761 | } | |
762 | ||
763 | alsa->handle = handle; | |
764 | return 0; | |
765 | } | |
766 | ||
767 | static void alsa_fini_in (HWVoiceIn *hw) | |
768 | { | |
769 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
770 | ||
771 | alsa_anal_close (&alsa->handle); | |
772 | ||
773 | if (alsa->pcm_buf) { | |
774 | qemu_free (alsa->pcm_buf); | |
775 | alsa->pcm_buf = NULL; | |
776 | } | |
777 | } | |
778 | ||
779 | static int alsa_run_in (HWVoiceIn *hw) | |
780 | { | |
781 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; | |
782 | int hwshift = hw->info.shift; | |
783 | int i; | |
784 | int live = audio_pcm_hw_get_live_in (hw); | |
785 | int dead = hw->samples - live; | |
786 | struct { | |
787 | int add; | |
788 | int len; | |
789 | } bufs[2] = { | |
790 | { hw->wpos, 0 }, | |
791 | { 0, 0 } | |
792 | }; | |
793 | ||
794 | snd_pcm_uframes_t read_samples = 0; | |
795 | ||
796 | if (!dead) { | |
797 | return 0; | |
798 | } | |
799 | ||
800 | if (hw->wpos + dead > hw->samples) { | |
801 | bufs[0].len = (hw->samples - hw->wpos); | |
802 | bufs[1].len = (dead - (hw->samples - hw->wpos)); | |
803 | } | |
804 | else { | |
805 | bufs[0].len = dead; | |
806 | } | |
807 | ||
808 | ||
809 | for (i = 0; i < 2; ++i) { | |
810 | void *src; | |
811 | st_sample_t *dst; | |
812 | snd_pcm_sframes_t nread; | |
813 | snd_pcm_uframes_t len; | |
814 | ||
815 | len = bufs[i].len; | |
816 | ||
817 | src = advance (alsa->pcm_buf, bufs[i].add << hwshift); | |
818 | dst = hw->conv_buf + bufs[i].add; | |
819 | ||
820 | while (len) { | |
821 | nread = snd_pcm_readi (alsa->handle, src, len); | |
822 | ||
823 | if (nread < 0) { | |
824 | switch (nread) { | |
825 | case -EPIPE: | |
826 | if (!alsa_recover (alsa->handle)) { | |
827 | continue; | |
828 | } | |
829 | dolog ( | |
830 | "Failed to read %ld frames from %p, " | |
831 | "handle %p not prepared\n", | |
832 | len, | |
833 | src, | |
834 | alsa->handle | |
835 | ); | |
836 | goto exit; | |
837 | ||
838 | case -EAGAIN: | |
839 | continue; | |
840 | ||
841 | default: | |
842 | alsa_logerr ( | |
843 | nread, | |
844 | "Failed to read %ld frames from %p\n", | |
845 | len, | |
846 | src | |
847 | ); | |
848 | goto exit; | |
849 | } | |
850 | } | |
851 | ||
852 | hw->conv (dst, src, nread, &nominal_volume); | |
853 | ||
854 | src = advance (src, nread << hwshift); | |
855 | dst += nread; | |
856 | ||
857 | read_samples += nread; | |
858 | len -= nread; | |
859 | } | |
860 | } | |
861 | ||
862 | exit: | |
863 | hw->wpos = (hw->wpos + read_samples) % hw->samples; | |
864 | return read_samples; | |
865 | } | |
866 | ||
867 | static int alsa_read (SWVoiceIn *sw, void *buf, int size) | |
868 | { | |
869 | return audio_pcm_sw_read (sw, buf, size); | |
870 | } | |
871 | ||
872 | static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) | |
873 | { | |
874 | (void) hw; | |
875 | (void) cmd; | |
876 | return 0; | |
877 | } | |
878 | ||
879 | static void *alsa_audio_init (void) | |
880 | { | |
881 | return &conf; | |
882 | } | |
883 | ||
884 | static void alsa_audio_fini (void *opaque) | |
885 | { | |
886 | (void) opaque; | |
887 | } | |
888 | ||
889 | static struct audio_option alsa_options[] = { | |
890 | {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out, | |
891 | "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, | |
892 | {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out, | |
893 | "DAC period size", &conf.period_size_out_overriden, 0}, | |
894 | {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out, | |
895 | "DAC buffer size", &conf.buffer_size_out_overriden, 0}, | |
896 | ||
897 | {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in, | |
898 | "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, | |
899 | {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in, | |
900 | "ADC period size", &conf.period_size_in_overriden, 0}, | |
901 | {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in, | |
902 | "ADC buffer size", &conf.buffer_size_in_overriden, 0}, | |
903 | ||
904 | {"THRESHOLD", AUD_OPT_INT, &conf.threshold, | |
905 | "(undocumented)", NULL, 0}, | |
906 | ||
907 | {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out, | |
908 | "DAC device name (for instance dmix)", NULL, 0}, | |
909 | ||
910 | {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in, | |
911 | "ADC device name", NULL, 0}, | |
912 | {NULL, 0, NULL, NULL, NULL, 0} | |
913 | }; | |
914 | ||
915 | static struct audio_pcm_ops alsa_pcm_ops = { | |
916 | alsa_init_out, | |
917 | alsa_fini_out, | |
918 | alsa_run_out, | |
919 | alsa_write, | |
920 | alsa_ctl_out, | |
921 | ||
922 | alsa_init_in, | |
923 | alsa_fini_in, | |
924 | alsa_run_in, | |
925 | alsa_read, | |
926 | alsa_ctl_in | |
927 | }; | |
928 | ||
929 | struct audio_driver alsa_audio_driver = { | |
930 | INIT_FIELD (name = ) "alsa", | |
931 | INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org", | |
932 | INIT_FIELD (options = ) alsa_options, | |
933 | INIT_FIELD (init = ) alsa_audio_init, | |
934 | INIT_FIELD (fini = ) alsa_audio_fini, | |
935 | INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, | |
936 | INIT_FIELD (can_be_default = ) 1, | |
937 | INIT_FIELD (max_voices_out = ) INT_MAX, | |
938 | INIT_FIELD (max_voices_in = ) INT_MAX, | |
939 | INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut), | |
940 | INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn) | |
941 | }; |