root / audio / alsaaudio.c @ 6276c767
History | View | Annotate | Download (26.2 kB)
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 |
} ALSAVoiceOut; |
35 |
|
36 |
typedef struct ALSAVoiceIn { |
37 |
HWVoiceIn hw; |
38 |
snd_pcm_t *handle; |
39 |
void *pcm_buf;
|
40 |
} ALSAVoiceIn; |
41 |
|
42 |
static struct { |
43 |
int size_in_usec_in;
|
44 |
int size_in_usec_out;
|
45 |
const char *pcm_name_in; |
46 |
const char *pcm_name_out; |
47 |
unsigned int buffer_size_in; |
48 |
unsigned int period_size_in; |
49 |
unsigned int buffer_size_out; |
50 |
unsigned int period_size_out; |
51 |
unsigned int threshold; |
52 |
|
53 |
int buffer_size_in_overridden;
|
54 |
int period_size_in_overridden;
|
55 |
|
56 |
int buffer_size_out_overridden;
|
57 |
int period_size_out_overridden;
|
58 |
int verbose;
|
59 |
} conf = { |
60 |
#define DEFAULT_BUFFER_SIZE 1024 |
61 |
#define DEFAULT_PERIOD_SIZE 256 |
62 |
#ifdef HIGH_LATENCY
|
63 |
.size_in_usec_in = 1,
|
64 |
.size_in_usec_out = 1,
|
65 |
#endif
|
66 |
.pcm_name_out = "default",
|
67 |
.pcm_name_in = "default",
|
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 |
.buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
|
75 |
.period_size_in = DEFAULT_PERIOD_SIZE * 4,
|
76 |
.buffer_size_out = DEFAULT_BUFFER_SIZE, |
77 |
.period_size_out = DEFAULT_PERIOD_SIZE, |
78 |
.buffer_size_in_overridden = 0,
|
79 |
.buffer_size_out_overridden = 0,
|
80 |
.period_size_in_overridden = 0,
|
81 |
.period_size_out_overridden = 0,
|
82 |
#endif
|
83 |
.threshold = 0,
|
84 |
.verbose = 0
|
85 |
}; |
86 |
|
87 |
struct alsa_params_req {
|
88 |
int freq;
|
89 |
audfmt_e fmt; |
90 |
int nchannels;
|
91 |
unsigned int buffer_size; |
92 |
unsigned int period_size; |
93 |
}; |
94 |
|
95 |
struct alsa_params_obt {
|
96 |
int freq;
|
97 |
audfmt_e fmt; |
98 |
int nchannels;
|
99 |
snd_pcm_uframes_t samples; |
100 |
}; |
101 |
|
102 |
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
103 |
{ |
104 |
va_list ap; |
105 |
|
106 |
va_start (ap, fmt); |
107 |
AUD_vlog (AUDIO_CAP, fmt, ap); |
108 |
va_end (ap); |
109 |
|
110 |
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
111 |
} |
112 |
|
113 |
static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( |
114 |
int err,
|
115 |
const char *typ, |
116 |
const char *fmt, |
117 |
... |
118 |
) |
119 |
{ |
120 |
va_list ap; |
121 |
|
122 |
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
|
123 |
|
124 |
va_start (ap, fmt); |
125 |
AUD_vlog (AUDIO_CAP, fmt, ap); |
126 |
va_end (ap); |
127 |
|
128 |
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
129 |
} |
130 |
|
131 |
static void alsa_anal_close (snd_pcm_t **handlep) |
132 |
{ |
133 |
int err = snd_pcm_close (*handlep);
|
134 |
if (err) {
|
135 |
alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
|
136 |
} |
137 |
*handlep = NULL;
|
138 |
} |
139 |
|
140 |
static int alsa_write (SWVoiceOut *sw, void *buf, int len) |
141 |
{ |
142 |
return audio_pcm_sw_write (sw, buf, len);
|
143 |
} |
144 |
|
145 |
static int aud_to_alsafmt (audfmt_e fmt) |
146 |
{ |
147 |
switch (fmt) {
|
148 |
case AUD_FMT_S8:
|
149 |
return SND_PCM_FORMAT_S8;
|
150 |
|
151 |
case AUD_FMT_U8:
|
152 |
return SND_PCM_FORMAT_U8;
|
153 |
|
154 |
case AUD_FMT_S16:
|
155 |
return SND_PCM_FORMAT_S16_LE;
|
156 |
|
157 |
case AUD_FMT_U16:
|
158 |
return SND_PCM_FORMAT_U16_LE;
|
159 |
|
160 |
case AUD_FMT_S32:
|
161 |
return SND_PCM_FORMAT_S32_LE;
|
162 |
|
163 |
case AUD_FMT_U32:
|
164 |
return SND_PCM_FORMAT_U32_LE;
|
165 |
|
166 |
default:
|
167 |
dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
168 |
#ifdef DEBUG_AUDIO
|
169 |
abort (); |
170 |
#endif
|
171 |
return SND_PCM_FORMAT_U8;
|
172 |
} |
173 |
} |
174 |
|
175 |
static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) |
176 |
{ |
177 |
switch (alsafmt) {
|
178 |
case SND_PCM_FORMAT_S8:
|
179 |
*endianness = 0;
|
180 |
*fmt = AUD_FMT_S8; |
181 |
break;
|
182 |
|
183 |
case SND_PCM_FORMAT_U8:
|
184 |
*endianness = 0;
|
185 |
*fmt = AUD_FMT_U8; |
186 |
break;
|
187 |
|
188 |
case SND_PCM_FORMAT_S16_LE:
|
189 |
*endianness = 0;
|
190 |
*fmt = AUD_FMT_S16; |
191 |
break;
|
192 |
|
193 |
case SND_PCM_FORMAT_U16_LE:
|
194 |
*endianness = 0;
|
195 |
*fmt = AUD_FMT_U16; |
196 |
break;
|
197 |
|
198 |
case SND_PCM_FORMAT_S16_BE:
|
199 |
*endianness = 1;
|
200 |
*fmt = AUD_FMT_S16; |
201 |
break;
|
202 |
|
203 |
case SND_PCM_FORMAT_U16_BE:
|
204 |
*endianness = 1;
|
205 |
*fmt = AUD_FMT_U16; |
206 |
break;
|
207 |
|
208 |
case SND_PCM_FORMAT_S32_LE:
|
209 |
*endianness = 0;
|
210 |
*fmt = AUD_FMT_S32; |
211 |
break;
|
212 |
|
213 |
case SND_PCM_FORMAT_U32_LE:
|
214 |
*endianness = 0;
|
215 |
*fmt = AUD_FMT_U32; |
216 |
break;
|
217 |
|
218 |
case SND_PCM_FORMAT_S32_BE:
|
219 |
*endianness = 1;
|
220 |
*fmt = AUD_FMT_S32; |
221 |
break;
|
222 |
|
223 |
case SND_PCM_FORMAT_U32_BE:
|
224 |
*endianness = 1;
|
225 |
*fmt = AUD_FMT_U32; |
226 |
break;
|
227 |
|
228 |
default:
|
229 |
dolog ("Unrecognized audio format %d\n", alsafmt);
|
230 |
return -1; |
231 |
} |
232 |
|
233 |
return 0; |
234 |
} |
235 |
|
236 |
#if defined DEBUG_MISMATCHES || defined DEBUG
|
237 |
static void alsa_dump_info (struct alsa_params_req *req, |
238 |
struct alsa_params_obt *obt)
|
239 |
{ |
240 |
dolog ("parameter | requested value | obtained value\n");
|
241 |
dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
|
242 |
dolog ("channels | %10d | %10d\n",
|
243 |
req->nchannels, obt->nchannels); |
244 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
245 |
dolog ("============================================\n");
|
246 |
dolog ("requested: buffer size %d period size %d\n",
|
247 |
req->buffer_size, req->period_size); |
248 |
dolog ("obtained: samples %ld\n", obt->samples);
|
249 |
} |
250 |
#endif
|
251 |
|
252 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
253 |
{ |
254 |
int err;
|
255 |
snd_pcm_sw_params_t *sw_params; |
256 |
|
257 |
snd_pcm_sw_params_alloca (&sw_params); |
258 |
|
259 |
err = snd_pcm_sw_params_current (handle, sw_params); |
260 |
if (err < 0) { |
261 |
dolog ("Could not fully initialize DAC\n");
|
262 |
alsa_logerr (err, "Failed to get current software parameters\n");
|
263 |
return;
|
264 |
} |
265 |
|
266 |
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
267 |
if (err < 0) { |
268 |
dolog ("Could not fully initialize DAC\n");
|
269 |
alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
270 |
threshold); |
271 |
return;
|
272 |
} |
273 |
|
274 |
err = snd_pcm_sw_params (handle, sw_params); |
275 |
if (err < 0) { |
276 |
dolog ("Could not fully initialize DAC\n");
|
277 |
alsa_logerr (err, "Failed to set software parameters\n");
|
278 |
return;
|
279 |
} |
280 |
} |
281 |
|
282 |
static int alsa_open (int in, struct alsa_params_req *req, |
283 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
284 |
{ |
285 |
snd_pcm_t *handle; |
286 |
snd_pcm_hw_params_t *hw_params; |
287 |
int err, freq, nchannels;
|
288 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
289 |
unsigned int period_size, buffer_size; |
290 |
snd_pcm_uframes_t obt_buffer_size; |
291 |
const char *typ = in ? "ADC" : "DAC"; |
292 |
|
293 |
freq = req->freq; |
294 |
period_size = req->period_size; |
295 |
buffer_size = req->buffer_size; |
296 |
nchannels = req->nchannels; |
297 |
|
298 |
snd_pcm_hw_params_alloca (&hw_params); |
299 |
|
300 |
err = snd_pcm_open ( |
301 |
&handle, |
302 |
pcm_name, |
303 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
304 |
SND_PCM_NONBLOCK |
305 |
); |
306 |
if (err < 0) { |
307 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
308 |
return -1; |
309 |
} |
310 |
|
311 |
err = snd_pcm_hw_params_any (handle, hw_params); |
312 |
if (err < 0) { |
313 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
314 |
goto err;
|
315 |
} |
316 |
|
317 |
err = snd_pcm_hw_params_set_access ( |
318 |
handle, |
319 |
hw_params, |
320 |
SND_PCM_ACCESS_RW_INTERLEAVED |
321 |
); |
322 |
if (err < 0) { |
323 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
324 |
goto err;
|
325 |
} |
326 |
|
327 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
328 |
if (err < 0) { |
329 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
330 |
goto err;
|
331 |
} |
332 |
|
333 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
334 |
if (err < 0) { |
335 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
336 |
goto err;
|
337 |
} |
338 |
|
339 |
err = snd_pcm_hw_params_set_channels_near ( |
340 |
handle, |
341 |
hw_params, |
342 |
&nchannels |
343 |
); |
344 |
if (err < 0) { |
345 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
346 |
req->nchannels); |
347 |
goto err;
|
348 |
} |
349 |
|
350 |
if (nchannels != 1 && nchannels != 2) { |
351 |
alsa_logerr2 (err, typ, |
352 |
"Can not handle obtained number of channels %d\n",
|
353 |
nchannels); |
354 |
goto err;
|
355 |
} |
356 |
|
357 |
if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
|
358 |
if (!buffer_size) {
|
359 |
buffer_size = DEFAULT_BUFFER_SIZE; |
360 |
period_size= DEFAULT_PERIOD_SIZE; |
361 |
} |
362 |
} |
363 |
|
364 |
if (buffer_size) {
|
365 |
if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
|
366 |
if (period_size) {
|
367 |
err = snd_pcm_hw_params_set_period_time_near ( |
368 |
handle, |
369 |
hw_params, |
370 |
&period_size, |
371 |
0
|
372 |
); |
373 |
if (err < 0) { |
374 |
alsa_logerr2 (err, typ, |
375 |
"Failed to set period time %d\n",
|
376 |
req->period_size); |
377 |
goto err;
|
378 |
} |
379 |
} |
380 |
|
381 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
382 |
handle, |
383 |
hw_params, |
384 |
&buffer_size, |
385 |
0
|
386 |
); |
387 |
|
388 |
if (err < 0) { |
389 |
alsa_logerr2 (err, typ, |
390 |
"Failed to set buffer time %d\n",
|
391 |
req->buffer_size); |
392 |
goto err;
|
393 |
} |
394 |
} |
395 |
else {
|
396 |
int dir;
|
397 |
snd_pcm_uframes_t minval; |
398 |
|
399 |
if (period_size) {
|
400 |
minval = period_size; |
401 |
dir = 0;
|
402 |
|
403 |
err = snd_pcm_hw_params_get_period_size_min ( |
404 |
hw_params, |
405 |
&minval, |
406 |
&dir |
407 |
); |
408 |
if (err < 0) { |
409 |
alsa_logerr ( |
410 |
err, |
411 |
"Could not get minmal period size for %s\n",
|
412 |
typ |
413 |
); |
414 |
} |
415 |
else {
|
416 |
if (period_size < minval) {
|
417 |
if ((in && conf.period_size_in_overridden)
|
418 |
|| (!in && conf.period_size_out_overridden)) { |
419 |
dolog ("%s period size(%d) is less "
|
420 |
"than minmal period size(%ld)\n",
|
421 |
typ, |
422 |
period_size, |
423 |
minval); |
424 |
} |
425 |
period_size = minval; |
426 |
} |
427 |
} |
428 |
|
429 |
err = snd_pcm_hw_params_set_period_size ( |
430 |
handle, |
431 |
hw_params, |
432 |
period_size, |
433 |
0
|
434 |
); |
435 |
if (err < 0) { |
436 |
alsa_logerr2 (err, typ, "Failed to set period size %d\n",
|
437 |
req->period_size); |
438 |
goto err;
|
439 |
} |
440 |
} |
441 |
|
442 |
minval = buffer_size; |
443 |
err = snd_pcm_hw_params_get_buffer_size_min ( |
444 |
hw_params, |
445 |
&minval |
446 |
); |
447 |
if (err < 0) { |
448 |
alsa_logerr (err, "Could not get minmal buffer size for %s\n",
|
449 |
typ); |
450 |
} |
451 |
else {
|
452 |
if (buffer_size < minval) {
|
453 |
if ((in && conf.buffer_size_in_overridden)
|
454 |
|| (!in && conf.buffer_size_out_overridden)) { |
455 |
dolog ( |
456 |
"%s buffer size(%d) is less "
|
457 |
"than minimal buffer size(%ld)\n",
|
458 |
typ, |
459 |
buffer_size, |
460 |
minval |
461 |
); |
462 |
} |
463 |
buffer_size = minval; |
464 |
} |
465 |
} |
466 |
|
467 |
err = snd_pcm_hw_params_set_buffer_size ( |
468 |
handle, |
469 |
hw_params, |
470 |
buffer_size |
471 |
); |
472 |
if (err < 0) { |
473 |
alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
|
474 |
req->buffer_size); |
475 |
goto err;
|
476 |
} |
477 |
} |
478 |
} |
479 |
else {
|
480 |
dolog ("warning: Buffer size is not set\n");
|
481 |
} |
482 |
|
483 |
err = snd_pcm_hw_params (handle, hw_params); |
484 |
if (err < 0) { |
485 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
486 |
goto err;
|
487 |
} |
488 |
|
489 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
490 |
if (err < 0) { |
491 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
492 |
goto err;
|
493 |
} |
494 |
|
495 |
err = snd_pcm_prepare (handle); |
496 |
if (err < 0) { |
497 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
498 |
goto err;
|
499 |
} |
500 |
|
501 |
if (!in && conf.threshold) {
|
502 |
snd_pcm_uframes_t threshold; |
503 |
int bytes_per_sec;
|
504 |
|
505 |
bytes_per_sec = freq |
506 |
<< (nchannels == 2)
|
507 |
<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); |
508 |
|
509 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
510 |
alsa_set_threshold (handle, threshold); |
511 |
} |
512 |
|
513 |
obt->fmt = req->fmt; |
514 |
obt->nchannels = nchannels; |
515 |
obt->freq = freq; |
516 |
obt->samples = obt_buffer_size; |
517 |
*handlep = handle; |
518 |
|
519 |
#if defined DEBUG_MISMATCHES || defined DEBUG
|
520 |
if (obt->fmt != req->fmt ||
|
521 |
obt->nchannels != req->nchannels || |
522 |
obt->freq != req->freq) { |
523 |
dolog ("Audio paramters mismatch for %s\n", typ);
|
524 |
alsa_dump_info (req, obt); |
525 |
} |
526 |
#endif
|
527 |
|
528 |
#ifdef DEBUG
|
529 |
alsa_dump_info (req, obt); |
530 |
#endif
|
531 |
return 0; |
532 |
|
533 |
err:
|
534 |
alsa_anal_close (&handle); |
535 |
return -1; |
536 |
} |
537 |
|
538 |
static int alsa_recover (snd_pcm_t *handle) |
539 |
{ |
540 |
int err = snd_pcm_prepare (handle);
|
541 |
if (err < 0) { |
542 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
543 |
return -1; |
544 |
} |
545 |
return 0; |
546 |
} |
547 |
|
548 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
549 |
{ |
550 |
snd_pcm_sframes_t avail; |
551 |
|
552 |
avail = snd_pcm_avail_update (handle); |
553 |
if (avail < 0) { |
554 |
if (avail == -EPIPE) {
|
555 |
if (!alsa_recover (handle)) {
|
556 |
avail = snd_pcm_avail_update (handle); |
557 |
} |
558 |
} |
559 |
|
560 |
if (avail < 0) { |
561 |
alsa_logerr (avail, |
562 |
"Could not obtain number of available frames\n");
|
563 |
return -1; |
564 |
} |
565 |
} |
566 |
|
567 |
return avail;
|
568 |
} |
569 |
|
570 |
static int alsa_run_out (HWVoiceOut *hw) |
571 |
{ |
572 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
573 |
int rpos, live, decr;
|
574 |
int samples;
|
575 |
uint8_t *dst; |
576 |
st_sample_t *src; |
577 |
snd_pcm_sframes_t avail; |
578 |
|
579 |
live = audio_pcm_hw_get_live_out (hw); |
580 |
if (!live) {
|
581 |
return 0; |
582 |
} |
583 |
|
584 |
avail = alsa_get_avail (alsa->handle); |
585 |
if (avail < 0) { |
586 |
dolog ("Could not get number of available playback frames\n");
|
587 |
return 0; |
588 |
} |
589 |
|
590 |
decr = audio_MIN (live, avail); |
591 |
samples = decr; |
592 |
rpos = hw->rpos; |
593 |
while (samples) {
|
594 |
int left_till_end_samples = hw->samples - rpos;
|
595 |
int len = audio_MIN (samples, left_till_end_samples);
|
596 |
snd_pcm_sframes_t written; |
597 |
|
598 |
src = hw->mix_buf + rpos; |
599 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
600 |
|
601 |
hw->clip (dst, src, len); |
602 |
|
603 |
while (len) {
|
604 |
written = snd_pcm_writei (alsa->handle, dst, len); |
605 |
|
606 |
if (written <= 0) { |
607 |
switch (written) {
|
608 |
case 0: |
609 |
if (conf.verbose) {
|
610 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
611 |
} |
612 |
goto exit;
|
613 |
|
614 |
case -EPIPE:
|
615 |
if (alsa_recover (alsa->handle)) {
|
616 |
alsa_logerr (written, "Failed to write %d frames\n",
|
617 |
len); |
618 |
goto exit;
|
619 |
} |
620 |
if (conf.verbose) {
|
621 |
dolog ("Recovering from playback xrun\n");
|
622 |
} |
623 |
continue;
|
624 |
|
625 |
case -EAGAIN:
|
626 |
goto exit;
|
627 |
|
628 |
default:
|
629 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
630 |
len, dst); |
631 |
goto exit;
|
632 |
} |
633 |
} |
634 |
|
635 |
rpos = (rpos + written) % hw->samples; |
636 |
samples -= written; |
637 |
len -= written; |
638 |
dst = advance (dst, written << hw->info.shift); |
639 |
src += written; |
640 |
} |
641 |
} |
642 |
|
643 |
exit:
|
644 |
hw->rpos = rpos; |
645 |
return decr;
|
646 |
} |
647 |
|
648 |
static void alsa_fini_out (HWVoiceOut *hw) |
649 |
{ |
650 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
651 |
|
652 |
ldebug ("alsa_fini\n");
|
653 |
alsa_anal_close (&alsa->handle); |
654 |
|
655 |
if (alsa->pcm_buf) {
|
656 |
qemu_free (alsa->pcm_buf); |
657 |
alsa->pcm_buf = NULL;
|
658 |
} |
659 |
} |
660 |
|
661 |
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
662 |
{ |
663 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
664 |
struct alsa_params_req req;
|
665 |
struct alsa_params_obt obt;
|
666 |
audfmt_e effective_fmt; |
667 |
int endianness;
|
668 |
int err;
|
669 |
snd_pcm_t *handle; |
670 |
audsettings_t obt_as; |
671 |
|
672 |
req.fmt = aud_to_alsafmt (as->fmt); |
673 |
req.freq = as->freq; |
674 |
req.nchannels = as->nchannels; |
675 |
req.period_size = conf.period_size_out; |
676 |
req.buffer_size = conf.buffer_size_out; |
677 |
|
678 |
if (alsa_open (0, &req, &obt, &handle)) { |
679 |
return -1; |
680 |
} |
681 |
|
682 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
683 |
if (err) {
|
684 |
alsa_anal_close (&handle); |
685 |
return -1; |
686 |
} |
687 |
|
688 |
obt_as.freq = obt.freq; |
689 |
obt_as.nchannels = obt.nchannels; |
690 |
obt_as.fmt = effective_fmt; |
691 |
obt_as.endianness = endianness; |
692 |
|
693 |
audio_pcm_init_info (&hw->info, &obt_as); |
694 |
hw->samples = obt.samples; |
695 |
|
696 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
697 |
if (!alsa->pcm_buf) {
|
698 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
699 |
hw->samples, 1 << hw->info.shift);
|
700 |
alsa_anal_close (&handle); |
701 |
return -1; |
702 |
} |
703 |
|
704 |
alsa->handle = handle; |
705 |
return 0; |
706 |
} |
707 |
|
708 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
709 |
{ |
710 |
int err;
|
711 |
|
712 |
if (pause) {
|
713 |
err = snd_pcm_drop (handle); |
714 |
if (err < 0) { |
715 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
716 |
return -1; |
717 |
} |
718 |
} |
719 |
else {
|
720 |
err = snd_pcm_prepare (handle); |
721 |
if (err < 0) { |
722 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
723 |
return -1; |
724 |
} |
725 |
} |
726 |
|
727 |
return 0; |
728 |
} |
729 |
|
730 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
731 |
{ |
732 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
733 |
|
734 |
switch (cmd) {
|
735 |
case VOICE_ENABLE:
|
736 |
ldebug ("enabling voice\n");
|
737 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
738 |
|
739 |
case VOICE_DISABLE:
|
740 |
ldebug ("disabling voice\n");
|
741 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
742 |
} |
743 |
|
744 |
return -1; |
745 |
} |
746 |
|
747 |
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
748 |
{ |
749 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
750 |
struct alsa_params_req req;
|
751 |
struct alsa_params_obt obt;
|
752 |
int endianness;
|
753 |
int err;
|
754 |
audfmt_e effective_fmt; |
755 |
snd_pcm_t *handle; |
756 |
audsettings_t obt_as; |
757 |
|
758 |
req.fmt = aud_to_alsafmt (as->fmt); |
759 |
req.freq = as->freq; |
760 |
req.nchannels = as->nchannels; |
761 |
req.period_size = conf.period_size_in; |
762 |
req.buffer_size = conf.buffer_size_in; |
763 |
|
764 |
if (alsa_open (1, &req, &obt, &handle)) { |
765 |
return -1; |
766 |
} |
767 |
|
768 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
769 |
if (err) {
|
770 |
alsa_anal_close (&handle); |
771 |
return -1; |
772 |
} |
773 |
|
774 |
obt_as.freq = obt.freq; |
775 |
obt_as.nchannels = obt.nchannels; |
776 |
obt_as.fmt = effective_fmt; |
777 |
obt_as.endianness = endianness; |
778 |
|
779 |
audio_pcm_init_info (&hw->info, &obt_as); |
780 |
hw->samples = obt.samples; |
781 |
|
782 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
783 |
if (!alsa->pcm_buf) {
|
784 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
785 |
hw->samples, 1 << hw->info.shift);
|
786 |
alsa_anal_close (&handle); |
787 |
return -1; |
788 |
} |
789 |
|
790 |
alsa->handle = handle; |
791 |
return 0; |
792 |
} |
793 |
|
794 |
static void alsa_fini_in (HWVoiceIn *hw) |
795 |
{ |
796 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
797 |
|
798 |
alsa_anal_close (&alsa->handle); |
799 |
|
800 |
if (alsa->pcm_buf) {
|
801 |
qemu_free (alsa->pcm_buf); |
802 |
alsa->pcm_buf = NULL;
|
803 |
} |
804 |
} |
805 |
|
806 |
static int alsa_run_in (HWVoiceIn *hw) |
807 |
{ |
808 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
809 |
int hwshift = hw->info.shift;
|
810 |
int i;
|
811 |
int live = audio_pcm_hw_get_live_in (hw);
|
812 |
int dead = hw->samples - live;
|
813 |
int decr;
|
814 |
struct {
|
815 |
int add;
|
816 |
int len;
|
817 |
} bufs[2] = {
|
818 |
{ hw->wpos, 0 },
|
819 |
{ 0, 0 } |
820 |
}; |
821 |
snd_pcm_sframes_t avail; |
822 |
snd_pcm_uframes_t read_samples = 0;
|
823 |
|
824 |
if (!dead) {
|
825 |
return 0; |
826 |
} |
827 |
|
828 |
avail = alsa_get_avail (alsa->handle); |
829 |
if (avail < 0) { |
830 |
dolog ("Could not get number of captured frames\n");
|
831 |
return 0; |
832 |
} |
833 |
|
834 |
if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
|
835 |
avail = hw->samples; |
836 |
} |
837 |
|
838 |
decr = audio_MIN (dead, avail); |
839 |
if (!decr) {
|
840 |
return 0; |
841 |
} |
842 |
|
843 |
if (hw->wpos + decr > hw->samples) {
|
844 |
bufs[0].len = (hw->samples - hw->wpos);
|
845 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
846 |
} |
847 |
else {
|
848 |
bufs[0].len = decr;
|
849 |
} |
850 |
|
851 |
for (i = 0; i < 2; ++i) { |
852 |
void *src;
|
853 |
st_sample_t *dst; |
854 |
snd_pcm_sframes_t nread; |
855 |
snd_pcm_uframes_t len; |
856 |
|
857 |
len = bufs[i].len; |
858 |
|
859 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
860 |
dst = hw->conv_buf + bufs[i].add; |
861 |
|
862 |
while (len) {
|
863 |
nread = snd_pcm_readi (alsa->handle, src, len); |
864 |
|
865 |
if (nread <= 0) { |
866 |
switch (nread) {
|
867 |
case 0: |
868 |
if (conf.verbose) {
|
869 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
870 |
} |
871 |
goto exit;
|
872 |
|
873 |
case -EPIPE:
|
874 |
if (alsa_recover (alsa->handle)) {
|
875 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
876 |
goto exit;
|
877 |
} |
878 |
if (conf.verbose) {
|
879 |
dolog ("Recovering from capture xrun\n");
|
880 |
} |
881 |
continue;
|
882 |
|
883 |
case -EAGAIN:
|
884 |
goto exit;
|
885 |
|
886 |
default:
|
887 |
alsa_logerr ( |
888 |
nread, |
889 |
"Failed to read %ld frames from %p\n",
|
890 |
len, |
891 |
src |
892 |
); |
893 |
goto exit;
|
894 |
} |
895 |
} |
896 |
|
897 |
hw->conv (dst, src, nread, &nominal_volume); |
898 |
|
899 |
src = advance (src, nread << hwshift); |
900 |
dst += nread; |
901 |
|
902 |
read_samples += nread; |
903 |
len -= nread; |
904 |
} |
905 |
} |
906 |
|
907 |
exit:
|
908 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
909 |
return read_samples;
|
910 |
} |
911 |
|
912 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
913 |
{ |
914 |
return audio_pcm_sw_read (sw, buf, size);
|
915 |
} |
916 |
|
917 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
918 |
{ |
919 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
920 |
|
921 |
switch (cmd) {
|
922 |
case VOICE_ENABLE:
|
923 |
ldebug ("enabling voice\n");
|
924 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
925 |
|
926 |
case VOICE_DISABLE:
|
927 |
ldebug ("disabling voice\n");
|
928 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
929 |
} |
930 |
|
931 |
return -1; |
932 |
} |
933 |
|
934 |
static void *alsa_audio_init (void) |
935 |
{ |
936 |
return &conf;
|
937 |
} |
938 |
|
939 |
static void alsa_audio_fini (void *opaque) |
940 |
{ |
941 |
(void) opaque;
|
942 |
} |
943 |
|
944 |
static struct audio_option alsa_options[] = { |
945 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
946 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
947 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
948 |
"DAC period size", &conf.period_size_out_overridden, 0}, |
949 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
950 |
"DAC buffer size", &conf.buffer_size_out_overridden, 0}, |
951 |
|
952 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
953 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
954 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
955 |
"ADC period size", &conf.period_size_in_overridden, 0}, |
956 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
957 |
"ADC buffer size", &conf.buffer_size_in_overridden, 0}, |
958 |
|
959 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
960 |
"(undocumented)", NULL, 0}, |
961 |
|
962 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
963 |
"DAC device name (for instance dmix)", NULL, 0}, |
964 |
|
965 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
966 |
"ADC device name", NULL, 0}, |
967 |
|
968 |
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
|
969 |
"Behave in a more verbose way", NULL, 0}, |
970 |
|
971 |
{NULL, 0, NULL, NULL, NULL, 0} |
972 |
}; |
973 |
|
974 |
static struct audio_pcm_ops alsa_pcm_ops = { |
975 |
alsa_init_out, |
976 |
alsa_fini_out, |
977 |
alsa_run_out, |
978 |
alsa_write, |
979 |
alsa_ctl_out, |
980 |
|
981 |
alsa_init_in, |
982 |
alsa_fini_in, |
983 |
alsa_run_in, |
984 |
alsa_read, |
985 |
alsa_ctl_in |
986 |
}; |
987 |
|
988 |
struct audio_driver alsa_audio_driver = {
|
989 |
INIT_FIELD (name = ) "alsa",
|
990 |
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
|
991 |
INIT_FIELD (options = ) alsa_options, |
992 |
INIT_FIELD (init = ) alsa_audio_init, |
993 |
INIT_FIELD (fini = ) alsa_audio_fini, |
994 |
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, |
995 |
INIT_FIELD (can_be_default = ) 1,
|
996 |
INIT_FIELD (max_voices_out = ) INT_MAX, |
997 |
INIT_FIELD (max_voices_in = ) INT_MAX, |
998 |
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
|
999 |
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
|
1000 |
}; |