1 /*
2 * QEMU ALSA audio driver
3 *
4 * Copyright (c) 2008 The Android Open Source Project
5 * Copyright (c) 2005 Vassili Karpov (malc)
6 *
7 * Permission is hereby granted, free of charge, to any person obtaining a copy
8 * of this software and associated documentation files (the "Software"), to deal
9 * in the Software without restriction, including without limitation the rights
10 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
11 * copies of the Software, and to permit persons to whom the Software is
12 * furnished to do so, subject to the following conditions:
13 *
14 * The above copyright notice and this permission notice shall be included in
15 * all copies or substantial portions of the Software.
16 *
17 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
18 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
19 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
20 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
21 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
22 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
23 * THE SOFTWARE.
24 */
25 #include <alsa/asoundlib.h>
26 #include "qemu-common.h"
27 #include "audio.h"
28
29 #define AUDIO_CAP "alsa"
30 #include "audio_int.h"
31 #include <dlfcn.h>
32 #include <pthread.h>
33 #include "qemu_debug.h"
34
35 #define DEBUG 1
36
37 #if DEBUG
38 # include <stdio.h>
39 # define D(...) VERBOSE_PRINT(audio,__VA_ARGS__)
40 # define D_ACTIVE VERBOSE_CHECK(audio)
41 # define O(...) VERBOSE_PRINT(audioout,__VA_ARGS__)
42 # define I(...) VERBOSE_PRINT(audioin,__VA_ARGS__)
43 #else
44 # define D(...) ((void)0)
45 # define D_ACTIVE 0
46 # define O(...) ((void)0)
47 # define I(...) ((void)0)
48 #endif
49
50
51 #define STRINGIFY_(x) #x
52 #define STRINGIFY(x) STRINGIFY_(x)
53
54 #define DYNLINK_FUNCTIONS \
55 DYNLINK_FUNC(size_t,snd_pcm_sw_params_sizeof,(void)) \
56 DYNLINK_FUNC(int,snd_pcm_hw_params_current,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \
57 DYNLINK_FUNC(int,snd_pcm_sw_params_set_start_threshold,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val)) \
58 DYNLINK_FUNC(int,snd_pcm_sw_params,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params)) \
59 DYNLINK_FUNC(int,snd_pcm_sw_params_current,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params)) \
60 DYNLINK_FUNC(size_t,snd_pcm_hw_params_sizeof,(void)) \
61 DYNLINK_FUNC(int,snd_pcm_hw_params_any,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \
62 DYNLINK_FUNC(int,snd_pcm_hw_params_set_access,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access)) \
63 DYNLINK_FUNC(int,snd_pcm_hw_params_get_format,(const snd_pcm_hw_params_t *params, snd_pcm_format_t *val)) \
64 DYNLINK_FUNC(int,snd_pcm_hw_params_set_format,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val)) \
65 DYNLINK_FUNC(int,snd_pcm_hw_params_set_rate_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \
66 DYNLINK_FUNC(int,snd_pcm_hw_params_set_channels_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val)) \
67 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_time_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \
68 DYNLINK_FUNC(int,snd_pcm_hw_params,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \
69 DYNLINK_FUNC(int,snd_pcm_hw_params_get_buffer_size,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \
70 DYNLINK_FUNC(int,snd_pcm_prepare,(snd_pcm_t *pcm)) \
71 DYNLINK_FUNC(int,snd_pcm_hw_params_get_period_size,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir)) \
72 DYNLINK_FUNC(int,snd_pcm_hw_params_get_period_size_min,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir)) \
73 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_size,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int dir)) \
74 DYNLINK_FUNC(int,snd_pcm_hw_params_get_buffer_size_min,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \
75 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_size,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val)) \
76 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_time_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \
77 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_avail_update,(snd_pcm_t *pcm)) \
78 DYNLINK_FUNC(int,snd_pcm_drop,(snd_pcm_t *pcm)) \
79 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_writei,(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size)) \
80 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_readi,(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size)) \
81 DYNLINK_FUNC(snd_pcm_state_t,snd_pcm_state,(snd_pcm_t *pcm)) \
82 DYNLINK_FUNC(const char*,snd_strerror,(int errnum)) \
83 DYNLINK_FUNC(int,snd_pcm_open,(snd_pcm_t **pcm, const char *name,snd_pcm_stream_t stream, int mode)) \
84 DYNLINK_FUNC(int,snd_pcm_close,(snd_pcm_t *pcm)) \
85 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_size_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \
86 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_size_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir)) \
87 DYNLINK_FUNC(int,snd_pcm_hw_params_get_format,(const snd_pcm_hw_params_t *params, snd_pcm_format_t *val)) \
88
89 #define DYNLINK_FUNCTIONS_INIT \
90 alsa_dynlink_init
91
92 #include "dynlink.h"
93
94 /* these are inlined functions in the original headers */
95 #define FF_snd_pcm_hw_params_alloca(ptr) \
96 do { *ptr = (snd_pcm_hw_params_t *) alloca(FF(snd_pcm_hw_params_sizeof)()); memset(*ptr, 0, FF(snd_pcm_hw_params_sizeof)()); } while (0)
97
98 #define FF_snd_pcm_sw_params_alloca(ptr) \
99 do { *ptr = (snd_pcm_sw_params_t *) alloca(FF(snd_pcm_sw_params_sizeof)()); memset(*ptr, 0, FF(snd_pcm_sw_params_sizeof)()); } while (0)
100
101 static void* alsa_lib;
102
103 typedef struct ALSAVoiceOut {
104 HWVoiceOut hw;
105 void *pcm_buf;
106 snd_pcm_t *handle;
107 } ALSAVoiceOut;
108
109 typedef struct ALSAVoiceIn {
110 HWVoiceIn hw;
111 snd_pcm_t *handle;
112 void *pcm_buf;
113 } ALSAVoiceIn;
114
115 static struct {
116 int size_in_usec_in;
117 int size_in_usec_out;
118 const char *pcm_name_in;
119 const char *pcm_name_out;
120 unsigned int buffer_size_in;
121 unsigned int period_size_in;
122 unsigned int buffer_size_out;
123 unsigned int period_size_out;
124 unsigned int threshold;
125
126 int buffer_size_in_overridden;
127 int period_size_in_overridden;
128
129 int buffer_size_out_overridden;
130 int period_size_out_overridden;
131 int verbose;
132 } conf = {
133 .buffer_size_out = 1024,
134 .pcm_name_out = "default",
135 .pcm_name_in = "default",
136 };
137
138 struct alsa_params_req {
139 int freq;
140 snd_pcm_format_t fmt;
141 int nchannels;
142 int size_in_usec;
143 int override_mask;
144 unsigned int buffer_size;
145 unsigned int period_size;
146 };
147
148 struct alsa_params_obt {
149 int freq;
150 audfmt_e fmt;
151 int endianness;
152 int nchannels;
153 snd_pcm_uframes_t samples;
154 };
155
alsa_logerr(int err,const char * fmt,...)156 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
157 {
158 va_list ap;
159
160 va_start (ap, fmt);
161 AUD_vlog (AUDIO_CAP, fmt, ap);
162 va_end (ap);
163
164 AUD_log (AUDIO_CAP, "Reason: %s\n", FF(snd_strerror) (err));
165 }
166
alsa_logerr2(int err,const char * typ,const char * fmt,...)167 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
168 int err,
169 const char *typ,
170 const char *fmt,
171 ...
172 )
173 {
174 va_list ap;
175
176 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
177
178 va_start (ap, fmt);
179 AUD_vlog (AUDIO_CAP, fmt, ap);
180 va_end (ap);
181
182 AUD_log (AUDIO_CAP, "Reason: %s\n", FF(snd_strerror) (err));
183 }
184
alsa_anal_close(snd_pcm_t ** handlep)185 static void alsa_anal_close (snd_pcm_t **handlep)
186 {
187 int err = FF(snd_pcm_close) (*handlep);
188 if (err) {
189 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
190 }
191 *handlep = NULL;
192 }
193
alsa_write(SWVoiceOut * sw,void * buf,int len)194 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
195 {
196 return audio_pcm_sw_write (sw, buf, len);
197 }
198
aud_to_alsafmt(audfmt_e fmt)199 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
200 {
201 switch (fmt) {
202 case AUD_FMT_S8:
203 return SND_PCM_FORMAT_S8;
204
205 case AUD_FMT_U8:
206 return SND_PCM_FORMAT_U8;
207
208 case AUD_FMT_S16:
209 return SND_PCM_FORMAT_S16_LE;
210
211 case AUD_FMT_U16:
212 return SND_PCM_FORMAT_U16_LE;
213
214 case AUD_FMT_S32:
215 return SND_PCM_FORMAT_S32_LE;
216
217 case AUD_FMT_U32:
218 return SND_PCM_FORMAT_U32_LE;
219
220 default:
221 dolog ("Internal logic error: Bad audio format %d\n", fmt);
222 #ifdef DEBUG_AUDIO
223 abort ();
224 #endif
225 return SND_PCM_FORMAT_U8;
226 }
227 }
228
alsa_to_audfmt(snd_pcm_format_t alsafmt,audfmt_e * fmt,int * endianness)229 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
230 int *endianness)
231 {
232 switch (alsafmt) {
233 case SND_PCM_FORMAT_S8:
234 *endianness = 0;
235 *fmt = AUD_FMT_S8;
236 break;
237
238 case SND_PCM_FORMAT_U8:
239 *endianness = 0;
240 *fmt = AUD_FMT_U8;
241 break;
242
243 case SND_PCM_FORMAT_S16_LE:
244 *endianness = 0;
245 *fmt = AUD_FMT_S16;
246 break;
247
248 case SND_PCM_FORMAT_U16_LE:
249 *endianness = 0;
250 *fmt = AUD_FMT_U16;
251 break;
252
253 case SND_PCM_FORMAT_S16_BE:
254 *endianness = 1;
255 *fmt = AUD_FMT_S16;
256 break;
257
258 case SND_PCM_FORMAT_U16_BE:
259 *endianness = 1;
260 *fmt = AUD_FMT_U16;
261 break;
262
263 case SND_PCM_FORMAT_S32_LE:
264 *endianness = 0;
265 *fmt = AUD_FMT_S32;
266 break;
267
268 case SND_PCM_FORMAT_U32_LE:
269 *endianness = 0;
270 *fmt = AUD_FMT_U32;
271 break;
272
273 case SND_PCM_FORMAT_S32_BE:
274 *endianness = 1;
275 *fmt = AUD_FMT_S32;
276 break;
277
278 case SND_PCM_FORMAT_U32_BE:
279 *endianness = 1;
280 *fmt = AUD_FMT_U32;
281 break;
282
283 default:
284 dolog ("Unrecognized audio format %d\n", alsafmt);
285 return -1;
286 }
287
288 return 0;
289 }
290
alsa_dump_info(struct alsa_params_req * req,struct alsa_params_obt * obt)291 static void alsa_dump_info (struct alsa_params_req *req,
292 struct alsa_params_obt *obt)
293 {
294 dolog ("parameter | requested value | obtained value\n");
295 dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
296 dolog ("channels | %10d | %10d\n",
297 req->nchannels, obt->nchannels);
298 dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
299 dolog ("============================================\n");
300 dolog ("requested: buffer size %d period size %d\n",
301 req->buffer_size, req->period_size);
302 dolog ("obtained: samples %ld\n", obt->samples);
303 }
304
alsa_set_threshold(snd_pcm_t * handle,snd_pcm_uframes_t threshold)305 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
306 {
307 int err;
308 snd_pcm_sw_params_t *sw_params;
309
310 FF_snd_pcm_sw_params_alloca (&sw_params);
311
312 err = FF(snd_pcm_sw_params_current) (handle, sw_params);
313 if (err < 0) {
314 dolog ("Could not fully initialize DAC\n");
315 alsa_logerr (err, "Failed to get current software parameters\n");
316 return;
317 }
318
319 err = FF(snd_pcm_sw_params_set_start_threshold) (handle, sw_params, threshold);
320 if (err < 0) {
321 dolog ("Could not fully initialize DAC\n");
322 alsa_logerr (err, "Failed to set software threshold to %ld\n",
323 threshold);
324 return;
325 }
326
327 err = FF(snd_pcm_sw_params) (handle, sw_params);
328 if (err < 0) {
329 dolog ("Could not fully initialize DAC\n");
330 alsa_logerr (err, "Failed to set software parameters\n");
331 return;
332 }
333 }
334
alsa_open(int in,struct alsa_params_req * req,struct alsa_params_obt * obt,snd_pcm_t ** handlep)335 static int alsa_open (int in, struct alsa_params_req *req,
336 struct alsa_params_obt *obt, snd_pcm_t **handlep)
337 {
338 snd_pcm_t *handle;
339 snd_pcm_hw_params_t *hw_params;
340 int err;
341 int size_in_usec;
342 unsigned int freq, nchannels;
343 const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
344 snd_pcm_uframes_t obt_buffer_size;
345 const char *typ = in ? "ADC" : "DAC";
346 snd_pcm_format_t obtfmt;
347
348 freq = req->freq;
349 nchannels = req->nchannels;
350 size_in_usec = req->size_in_usec;
351
352 FF_snd_pcm_hw_params_alloca (&hw_params);
353
354 err = FF(snd_pcm_open) (
355 &handle,
356 pcm_name,
357 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
358 SND_PCM_NONBLOCK
359 );
360 if (err < 0) {
361 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
362 return -1;
363 }
364
365 err = FF(snd_pcm_hw_params_any) (handle, hw_params);
366 if (err < 0) {
367 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
368 goto err;
369 }
370
371 err = FF(snd_pcm_hw_params_set_access) (
372 handle,
373 hw_params,
374 SND_PCM_ACCESS_RW_INTERLEAVED
375 );
376 if (err < 0) {
377 alsa_logerr2 (err, typ, "Failed to set access type\n");
378 goto err;
379 }
380
381 err = FF(snd_pcm_hw_params_set_format) (handle, hw_params, req->fmt);
382 if (err < 0 && conf.verbose) {
383 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
384 goto err;
385 }
386
387 err = FF(snd_pcm_hw_params_set_rate_near) (handle, hw_params, &freq, 0);
388 if (err < 0) {
389 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
390 goto err;
391 }
392
393 err = FF(snd_pcm_hw_params_set_channels_near) (
394 handle,
395 hw_params,
396 &nchannels
397 );
398 if (err < 0) {
399 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
400 req->nchannels);
401 goto err;
402 }
403
404 if (nchannels != 1 && nchannels != 2) {
405 alsa_logerr2 (err, typ,
406 "Can not handle obtained number of channels %d\n",
407 nchannels);
408 goto err;
409 }
410
411 if (req->buffer_size) {
412 unsigned long obt;
413
414 if (size_in_usec) {
415 int dir = 0;
416 unsigned int btime = req->buffer_size;
417
418 err = FF(snd_pcm_hw_params_set_buffer_time_near) (
419 handle,
420 hw_params,
421 &btime,
422 &dir
423 );
424 obt = btime;
425 }
426 else {
427 snd_pcm_uframes_t bsize = req->buffer_size;
428
429 err = FF(snd_pcm_hw_params_set_buffer_size_near) (
430 handle,
431 hw_params,
432 &bsize
433 );
434 obt = bsize;
435 }
436 if (err < 0) {
437 alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
438 size_in_usec ? "time" : "size", req->buffer_size);
439 goto err;
440 }
441
442 if ((req->override_mask & 2) && (obt - req->buffer_size))
443 dolog ("Requested buffer %s %u was rejected, using %lu\n",
444 size_in_usec ? "time" : "size", req->buffer_size, obt);
445 }
446
447 if (req->period_size) {
448 unsigned long obt;
449
450 if (size_in_usec) {
451 int dir = 0;
452 unsigned int ptime = req->period_size;
453
454 err = FF(snd_pcm_hw_params_set_period_time_near) (
455 handle,
456 hw_params,
457 &ptime,
458 &dir
459 );
460 obt = ptime;
461 }
462 else {
463 int dir = 0;
464 snd_pcm_uframes_t psize = req->period_size;
465
466 err = FF(snd_pcm_hw_params_set_period_size_near) (
467 handle,
468 hw_params,
469 &psize,
470 &dir
471 );
472 obt = psize;
473 }
474
475 if (err < 0) {
476 alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
477 size_in_usec ? "time" : "size", req->period_size);
478 goto err;
479 }
480
481 if ((req->override_mask & 1) && (obt - req->period_size))
482 dolog ("Requested period %s %u was rejected, using %lu\n",
483 size_in_usec ? "time" : "size", req->period_size, obt);
484 }
485
486 err = FF(snd_pcm_hw_params) (handle, hw_params);
487 if (err < 0) {
488 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
489 goto err;
490 }
491
492 err = FF(snd_pcm_hw_params_get_buffer_size) (hw_params, &obt_buffer_size);
493 if (err < 0) {
494 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
495 goto err;
496 }
497
498 err = FF(snd_pcm_hw_params_get_format)(hw_params, &obtfmt);
499 err = FF(snd_pcm_hw_params_get_format) (hw_params, &obtfmt);
500 if (err < 0) {
501 alsa_logerr2 (err, typ, "Failed to get format\n");
502 goto err;
503 }
504
505 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
506 dolog ("Invalid format was returned %d\n", obtfmt);
507 goto err;
508 }
509
510 err = FF(snd_pcm_prepare) (handle);
511 if (err < 0) {
512 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
513 goto err;
514 }
515
516 if (!in && conf.threshold) {
517 snd_pcm_uframes_t threshold;
518 int bytes_per_sec;
519
520 bytes_per_sec = freq << (nchannels == 2);
521
522 switch (obt->fmt) {
523 case AUD_FMT_S8:
524 case AUD_FMT_U8:
525 break;
526
527 case AUD_FMT_S16:
528 case AUD_FMT_U16:
529 bytes_per_sec <<= 1;
530 break;
531
532 case AUD_FMT_S32:
533 case AUD_FMT_U32:
534 bytes_per_sec <<= 2;
535 break;
536 }
537
538 threshold = (conf.threshold * bytes_per_sec) / 1000;
539 alsa_set_threshold (handle, threshold);
540 }
541
542 obt->nchannels = nchannels;
543 obt->freq = freq;
544 obt->samples = obt_buffer_size;
545
546 *handlep = handle;
547
548 if (conf.verbose &&
549 (obt->fmt != req->fmt ||
550 obt->nchannels != req->nchannels ||
551 obt->freq != req->freq)) {
552 dolog ("Audio paramters for %s\n", typ);
553 alsa_dump_info (req, obt);
554 }
555
556 #ifdef DEBUG
557 alsa_dump_info (req, obt);
558 #endif
559 return 0;
560
561 err:
562 alsa_anal_close (&handle);
563 return -1;
564 }
565
alsa_recover(snd_pcm_t * handle)566 static int alsa_recover (snd_pcm_t *handle)
567 {
568 int err = FF(snd_pcm_prepare) (handle);
569 if (err < 0) {
570 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
571 return -1;
572 }
573 return 0;
574 }
575
alsa_get_avail(snd_pcm_t * handle)576 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
577 {
578 snd_pcm_sframes_t avail;
579
580 avail = FF(snd_pcm_avail_update) (handle);
581 if (avail < 0) {
582 if (avail == -EPIPE) {
583 if (!alsa_recover (handle)) {
584 avail = FF(snd_pcm_avail_update) (handle);
585 }
586 }
587
588 if (avail < 0) {
589 alsa_logerr (avail,
590 "Could not obtain number of available frames\n");
591 return -1;
592 }
593 }
594
595 return avail;
596 }
597
alsa_run_out(HWVoiceOut * hw)598 static int alsa_run_out (HWVoiceOut *hw)
599 {
600 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
601 int rpos, live, decr;
602 int samples;
603 uint8_t *dst;
604 struct st_sample *src;
605 snd_pcm_sframes_t avail;
606
607 live = audio_pcm_hw_get_live_out (hw);
608 if (!live) {
609 return 0;
610 }
611
612 avail = alsa_get_avail (alsa->handle);
613 if (avail < 0) {
614 dolog ("Could not get number of available playback frames\n");
615 return 0;
616 }
617
618 decr = audio_MIN (live, avail);
619 samples = decr;
620 rpos = hw->rpos;
621 while (samples) {
622 int left_till_end_samples = hw->samples - rpos;
623 int len = audio_MIN (samples, left_till_end_samples);
624 snd_pcm_sframes_t written;
625
626 src = hw->mix_buf + rpos;
627 dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
628
629 hw->clip (dst, src, len);
630
631 while (len) {
632 written = FF(snd_pcm_writei) (alsa->handle, dst, len);
633
634 if (written <= 0) {
635 switch (written) {
636 case 0:
637 if (conf.verbose) {
638 dolog ("Failed to write %d frames (wrote zero)\n", len);
639 }
640 goto exit;
641
642 case -EPIPE:
643 if (alsa_recover (alsa->handle)) {
644 alsa_logerr (written, "Failed to write %d frames\n",
645 len);
646 goto exit;
647 }
648 if (conf.verbose) {
649 dolog ("Recovering from playback xrun\n");
650 }
651 continue;
652
653 case -EAGAIN:
654 goto exit;
655
656 default:
657 alsa_logerr (written, "Failed to write %d frames to %p\n",
658 len, dst);
659 goto exit;
660 }
661 }
662
663 rpos = (rpos + written) % hw->samples;
664 samples -= written;
665 len -= written;
666 dst = advance (dst, written << hw->info.shift);
667 src += written;
668 }
669 }
670
671 exit:
672 hw->rpos = rpos;
673 return decr;
674 }
675
alsa_fini_out(HWVoiceOut * hw)676 static void alsa_fini_out (HWVoiceOut *hw)
677 {
678 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
679
680 ldebug ("alsa_fini\n");
681 alsa_anal_close (&alsa->handle);
682
683 if (alsa->pcm_buf) {
684 qemu_free (alsa->pcm_buf);
685 alsa->pcm_buf = NULL;
686 }
687 }
688
alsa_init_out(HWVoiceOut * hw,struct audsettings * as)689 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
690 {
691 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
692 struct alsa_params_req req;
693 struct alsa_params_obt obt;
694 snd_pcm_t *handle;
695 struct audsettings obt_as;
696 int result = -1;
697
698 /* shut alsa debug spew */
699 if (!D_ACTIVE)
700 stdio_disable();
701
702 req.fmt = aud_to_alsafmt (as->fmt);
703 req.freq = as->freq;
704 req.nchannels = as->nchannels;
705 req.period_size = conf.period_size_out;
706 req.buffer_size = conf.buffer_size_out;
707 req.size_in_usec = conf.size_in_usec_out;
708 req.override_mask = !!conf.period_size_out_overridden
709 | (!!conf.buffer_size_out_overridden << 1);
710
711 if (alsa_open (0, &req, &obt, &handle)) {
712 goto Exit;
713 }
714
715 obt_as.freq = obt.freq;
716 obt_as.nchannels = obt.nchannels;
717 obt_as.fmt = obt.fmt;
718 obt_as.endianness = obt.endianness;
719
720 audio_pcm_init_info (&hw->info, &obt_as);
721 hw->samples = obt.samples;
722
723 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
724 if (!alsa->pcm_buf) {
725 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
726 hw->samples, 1 << hw->info.shift);
727 alsa_anal_close (&handle);
728 goto Exit;
729 }
730
731 alsa->handle = handle;
732 result = 0; /* success */
733
734 Exit:
735 if (!D_ACTIVE)
736 stdio_enable();
737
738 return result;
739 }
740
alsa_voice_ctl(snd_pcm_t * handle,const char * typ,int pause)741 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
742 {
743 int err;
744
745 if (pause) {
746 err = FF(snd_pcm_drop) (handle);
747 if (err < 0) {
748 alsa_logerr (err, "Could not stop %s\n", typ);
749 return -1;
750 }
751 }
752 else {
753 err = FF(snd_pcm_prepare) (handle);
754 if (err < 0) {
755 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
756 return -1;
757 }
758 }
759
760 return 0;
761 }
762
alsa_ctl_out(HWVoiceOut * hw,int cmd,...)763 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
764 {
765 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
766
767 switch (cmd) {
768 case VOICE_ENABLE:
769 ldebug ("enabling voice\n");
770 return alsa_voice_ctl (alsa->handle, "playback", 0);
771
772 case VOICE_DISABLE:
773 ldebug ("disabling voice\n");
774 return alsa_voice_ctl (alsa->handle, "playback", 1);
775 }
776
777 return -1;
778 }
779
alsa_init_in(HWVoiceIn * hw,struct audsettings * as)780 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
781 {
782 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
783 struct alsa_params_req req;
784 struct alsa_params_obt obt;
785 snd_pcm_t *handle;
786 struct audsettings obt_as;
787 int result = -1;
788
789 /* shut alsa debug spew */
790 if (!D_ACTIVE)
791 stdio_disable();
792
793 req.fmt = aud_to_alsafmt (as->fmt);
794 req.freq = as->freq;
795 req.nchannels = as->nchannels;
796 req.period_size = conf.period_size_in;
797 req.buffer_size = conf.buffer_size_in;
798 req.size_in_usec = conf.size_in_usec_in;
799 req.override_mask = !!conf.period_size_in_overridden
800 | (!!conf.buffer_size_in_overridden << 1);
801
802 if (alsa_open (1, &req, &obt, &handle)) {
803 goto Exit;
804 }
805
806 obt_as.freq = obt.freq;
807 obt_as.nchannels = obt.nchannels;
808 obt_as.fmt = obt.fmt;
809 obt_as.endianness = obt.endianness;
810
811 audio_pcm_init_info (&hw->info, &obt_as);
812 hw->samples = obt.samples;
813
814 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
815 if (!alsa->pcm_buf) {
816 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
817 hw->samples, 1 << hw->info.shift);
818 alsa_anal_close (&handle);
819 goto Exit;
820 }
821
822 alsa->handle = handle;
823 result = 0; /* success */
824
825 Exit:
826 if (!D_ACTIVE)
827 stdio_enable();
828
829 return result;
830 }
831
alsa_fini_in(HWVoiceIn * hw)832 static void alsa_fini_in (HWVoiceIn *hw)
833 {
834 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
835
836 alsa_anal_close (&alsa->handle);
837
838 if (alsa->pcm_buf) {
839 qemu_free (alsa->pcm_buf);
840 alsa->pcm_buf = NULL;
841 }
842 }
843
alsa_run_in(HWVoiceIn * hw)844 static int alsa_run_in (HWVoiceIn *hw)
845 {
846 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
847 int hwshift = hw->info.shift;
848 int i;
849 int live = audio_pcm_hw_get_live_in (hw);
850 int dead = hw->samples - live;
851 int decr;
852 struct {
853 int add;
854 int len;
855 } bufs[2] = {
856 { hw->wpos, 0 },
857 { 0, 0 }
858 };
859 snd_pcm_sframes_t avail;
860 snd_pcm_uframes_t read_samples = 0;
861
862 if (!dead) {
863 return 0;
864 }
865
866 avail = alsa_get_avail (alsa->handle);
867 if (avail < 0) {
868 dolog ("Could not get number of captured frames\n");
869 return 0;
870 }
871
872 if (!avail && (FF(snd_pcm_state) (alsa->handle) == SND_PCM_STATE_PREPARED)) {
873 avail = hw->samples;
874 }
875
876 decr = audio_MIN (dead, avail);
877 if (!decr) {
878 return 0;
879 }
880
881 if (hw->wpos + decr > hw->samples) {
882 bufs[0].len = (hw->samples - hw->wpos);
883 bufs[1].len = (decr - (hw->samples - hw->wpos));
884 }
885 else {
886 bufs[0].len = decr;
887 }
888
889 for (i = 0; i < 2; ++i) {
890 void *src;
891 struct st_sample *dst;
892 snd_pcm_sframes_t nread;
893 snd_pcm_uframes_t len;
894
895 len = bufs[i].len;
896
897 src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
898 dst = hw->conv_buf + bufs[i].add;
899
900 while (len) {
901 nread = FF(snd_pcm_readi) (alsa->handle, src, len);
902
903 if (nread <= 0) {
904 switch (nread) {
905 case 0:
906 if (conf.verbose) {
907 dolog ("Failed to read %ld frames (read zero)\n", len);
908 }
909 goto exit;
910
911 case -EPIPE:
912 if (alsa_recover (alsa->handle)) {
913 alsa_logerr (nread, "Failed to read %ld frames\n", len);
914 goto exit;
915 }
916 if (conf.verbose) {
917 dolog ("Recovering from capture xrun\n");
918 }
919 continue;
920
921 case -EAGAIN:
922 goto exit;
923
924 default:
925 alsa_logerr (
926 nread,
927 "Failed to read %ld frames from %p\n",
928 len,
929 src
930 );
931 goto exit;
932 }
933 }
934
935 hw->conv (dst, src, nread, &nominal_volume);
936
937 src = advance (src, nread << hwshift);
938 dst += nread;
939
940 read_samples += nread;
941 len -= nread;
942 }
943 }
944
945 exit:
946 hw->wpos = (hw->wpos + read_samples) % hw->samples;
947 return read_samples;
948 }
949
alsa_read(SWVoiceIn * sw,void * buf,int size)950 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
951 {
952 return audio_pcm_sw_read (sw, buf, size);
953 }
954
alsa_ctl_in(HWVoiceIn * hw,int cmd,...)955 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
956 {
957 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
958
959 switch (cmd) {
960 case VOICE_ENABLE:
961 ldebug ("enabling voice\n");
962 return alsa_voice_ctl (alsa->handle, "capture", 0);
963
964 case VOICE_DISABLE:
965 ldebug ("disabling voice\n");
966 return alsa_voice_ctl (alsa->handle, "capture", 1);
967 }
968
969 return -1;
970 }
971
alsa_audio_init(void)972 static void *alsa_audio_init (void)
973 {
974 void* result = NULL;
975
976 alsa_lib = dlopen( "libasound.so", RTLD_NOW );
977 if (alsa_lib == NULL)
978 alsa_lib = dlopen( "libasound.so.2", RTLD_NOW );
979
980 if (alsa_lib == NULL) {
981 ldebug("could not find libasound on this system\n");
982 goto Exit;
983 }
984
985 if (alsa_dynlink_init(alsa_lib) < 0)
986 goto Fail;
987
988 result = &conf;
989 goto Exit;
990
991 Fail:
992 ldebug("%s: failed to open library\n", __FUNCTION__);
993 dlclose(alsa_lib);
994
995 Exit:
996 return result;
997 }
998
alsa_audio_fini(void * opaque)999 static void alsa_audio_fini (void *opaque)
1000 {
1001 if (alsa_lib != NULL) {
1002 dlclose(alsa_lib);
1003 alsa_lib = NULL;
1004 }
1005 (void) opaque;
1006 }
1007
1008 static struct audio_option alsa_options[] = {
1009 {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
1010 "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1011 {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
1012 "DAC period size (0 to go with system default)",
1013 &conf.period_size_out_overridden, 0},
1014 {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
1015 "DAC buffer size (0 to go with system default)",
1016 &conf.buffer_size_out_overridden, 0},
1017
1018 {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
1019 "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1020 {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
1021 "ADC period size (0 to go with system default)",
1022 &conf.period_size_in_overridden, 0},
1023 {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
1024 "ADC buffer size (0 to go with system default)",
1025 &conf.buffer_size_in_overridden, 0},
1026
1027 {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
1028 "(undocumented)", NULL, 0},
1029
1030 {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
1031 "DAC device name (for instance dmix)", NULL, 0},
1032
1033 {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
1034 "ADC device name", NULL, 0},
1035
1036 {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
1037 "Behave in a more verbose way", NULL, 0},
1038
1039 {NULL, 0, NULL, NULL, NULL, 0}
1040 };
1041
1042 static struct audio_pcm_ops alsa_pcm_ops = {
1043 alsa_init_out,
1044 alsa_fini_out,
1045 alsa_run_out,
1046 alsa_write,
1047 alsa_ctl_out,
1048
1049 alsa_init_in,
1050 alsa_fini_in,
1051 alsa_run_in,
1052 alsa_read,
1053 alsa_ctl_in
1054 };
1055
1056 struct audio_driver alsa_audio_driver = {
1057 INIT_FIELD (name = ) "alsa",
1058 INIT_FIELD (descr = ) "ALSA audio (www.alsa-project.org)",
1059 INIT_FIELD (options = ) alsa_options,
1060 INIT_FIELD (init = ) alsa_audio_init,
1061 INIT_FIELD (fini = ) alsa_audio_fini,
1062 INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
1063 INIT_FIELD (can_be_default = ) 1,
1064 INIT_FIELD (max_voices_out = ) INT_MAX,
1065 INIT_FIELD (max_voices_in = ) INT_MAX,
1066 INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1067 INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
1068 };
1069