/
SDL_alsa_audio.c
603 lines (521 loc) · 19.1 KB
1
2
/*
SDL - Simple DirectMedia Layer
3
Copyright (C) 1997-2009 Sam Lantinga
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
20
slouken@libsdl.org
21
*/
22
#include "SDL_config.h"
23
24
25
26
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
27
#include <signal.h> /* For kill() */
28
29
#include "SDL_timer.h"
30
31
32
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
33
34
#include "SDL_alsa_audio.h"
35
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
36
37
38
39
40
41
42
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
43
44
45
46
/* The tag name used by ALSA audio */
#define DRIVER_NAME "alsa"
/* Audio driver functions */
47
48
49
50
51
52
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
static void ALSA_WaitAudio(_THIS);
static void ALSA_PlayAudio(_THIS);
static Uint8 *ALSA_GetAudioBuf(_THIS);
static void ALSA_CloseAudio(_THIS);
53
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
54
55
static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
56
57
58
59
60
61
static void *alsa_handle = NULL;
static int alsa_loaded = 0;
static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
62
static int (*SDL_NAME(snd_pcm_recover))(snd_pcm_t *pcm, int err, int silent);
63
64
65
66
static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
static const char *(*SDL_NAME(snd_strerror))(int errnum);
static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
67
static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void);
68
static void (*SDL_NAME(snd_pcm_hw_params_copy))(snd_pcm_hw_params_t *dst, const snd_pcm_hw_params_t *src);
69
70
71
72
static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
73
static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val);
74
75
static int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
static int (*SDL_NAME(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);
76
static int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir);
77
static int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
78
static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
79
static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
80
static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
81
static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
82
83
84
85
86
/*
*/
static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams);
static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
87
static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
88
#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
89
#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof)
90
91
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
92
93
94
95
static struct {
const char *name;
void **func;
} alsa_functions[] = {
96
97
98
{ "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) },
{ "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) },
{ "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) },
99
{ "snd_pcm_recover", (void**)(char*)&SDL_NAME(snd_pcm_recover) },
100
101
102
103
{ "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) },
{ "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) },
{ "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) },
{ "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) },
104
{ "snd_pcm_sw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof) },
105
{ "snd_pcm_hw_params_copy", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_copy) },
106
107
108
109
110
111
112
{ "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) },
{ "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) },
{ "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) },
{ "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) },
{ "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) },
{ "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) },
{ "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) },
113
{ "snd_pcm_hw_params_get_period_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size) },
114
{ "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) },
115
{ "snd_pcm_hw_params_get_periods", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods) },
116
{ "snd_pcm_hw_params_set_buffer_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) },
117
{ "snd_pcm_hw_params_get_buffer_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) },
118
{ "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) },
119
120
121
{ "snd_pcm_sw_params_current", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_current) },
{ "snd_pcm_sw_params_set_start_threshold", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold) },
{ "snd_pcm_sw_params", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) },
122
{ "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) },
123
124
125
126
};
static void UnloadALSALibrary(void) {
if (alsa_loaded) {
127
SDL_UnloadObject(alsa_handle);
128
129
130
131
132
133
134
135
alsa_handle = NULL;
alsa_loaded = 0;
}
}
static int LoadALSALibrary(void) {
int i, retval = -1;
136
alsa_handle = SDL_LoadObject(alsa_library);
137
138
139
if (alsa_handle) {
alsa_loaded = 1;
retval = 0;
140
for (i = 0; i < SDL_arraysize(alsa_functions); i++) {
141
*alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
if (!*alsa_functions[i].func) {
retval = -1;
UnloadALSALibrary();
break;
}
}
}
return retval;
}
#else
static void UnloadALSALibrary(void) {
return;
}
static int LoadALSALibrary(void) {
return 0;
}
162
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
163
164
static const char *get_audio_device(int channels)
165
{
166
167
const char *device;
168
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
169
if ( device == NULL ) {
170
171
172
173
174
175
176
177
178
179
180
switch (channels) {
case 6:
device = "plug:surround51";
break;
case 4:
device = "plug:surround40";
break;
default:
device = "default";
break;
}
181
182
}
return device;
183
184
185
186
187
188
189
}
/* Audio driver bootstrap functions */
static int Audio_Available(void)
{
int available;
190
int status;
191
192
193
snd_pcm_t *handle;
available = 0;
194
195
196
if (LoadALSALibrary() < 0) {
return available;
}
197
status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
198
199
if ( status >= 0 ) {
available = 1;
200
SDL_NAME(snd_pcm_close)(handle);
201
}
202
UnloadALSALibrary();
203
204
205
206
207
return(available);
}
static void Audio_DeleteDevice(SDL_AudioDevice *device)
{
208
209
SDL_free(device->hidden);
SDL_free(device);
210
UnloadALSALibrary();
211
212
213
214
215
216
217
}
static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
218
LoadALSALibrary();
219
this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
220
if ( this ) {
221
SDL_memset(this, 0, (sizeof *this));
222
this->hidden = (struct SDL_PrivateAudioData *)
223
SDL_malloc((sizeof *this->hidden));
224
225
226
227
}
if ( (this == NULL) || (this->hidden == NULL) ) {
SDL_OutOfMemory();
if ( this ) {
228
SDL_free(this);
229
230
231
}
return(0);
}
232
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
233
234
/* Set the function pointers */
235
236
237
238
239
this->OpenAudio = ALSA_OpenAudio;
this->WaitAudio = ALSA_WaitAudio;
this->PlayAudio = ALSA_PlayAudio;
this->GetAudioBuf = ALSA_GetAudioBuf;
this->CloseAudio = ALSA_CloseAudio;
240
241
242
243
244
245
246
this->free = Audio_DeleteDevice;
return this;
}
AudioBootStrap ALSA_bootstrap = {
247
DRIVER_NAME, "ALSA PCM audio",
248
249
250
251
Audio_Available, Audio_CreateDevice
};
/* This function waits until it is possible to write a full sound buffer */
252
static void ALSA_WaitAudio(_THIS)
253
{
254
/* We're in blocking mode, so there's nothing to do here */
255
256
}
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
/*
* http://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) mixbuf; \
const Uint32 count = (this->spec.samples / 6); \
Uint32 i; \
for (i = 0; i < count; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); }
static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); }
static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); }
static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); }
#undef SWIZ6
/*
* Called right before feeding this->mixbuf to the hardware. Swizzle channels
* from Windows/Mac order to the format alsalib will want.
*/
static __inline__ void swizzle_alsa_channels(_THIS)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
303
static void ALSA_PlayAudio(_THIS)
304
{
305
int status;
306
snd_pcm_uframes_t frames_left;
307
const Uint8 *sample_buf = (const Uint8 *) mixbuf;
308
const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * this->spec.channels;
309
310
311
swizzle_alsa_channels(this);
312
frames_left = ((snd_pcm_uframes_t) this->spec.samples);
313
314
while ( frames_left > 0 && this->enabled ) {
315
status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left);
316
if ( status < 0 ) {
317
318
status = SDL_NAME(snd_pcm_recover)(pcm_handle, status, 0);
if ( status < 0 ) {
319
/* Hmm, not much we can do - abort */
320
fprintf(stderr, "ALSA write failed (unrecoverable): %s", SDL_NAME(snd_strerror)(status));
321
322
this->enabled = 0;
return;
323
}
324
continue;
325
}
326
sample_buf += status * frame_size;
327
frames_left -= status;
328
329
330
}
}
331
static Uint8 *ALSA_GetAudioBuf(_THIS)
332
{
333
return(mixbuf);
334
335
}
336
static void ALSA_CloseAudio(_THIS)
337
{
338
339
340
if ( mixbuf != NULL ) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
341
}
342
if ( pcm_handle ) {
343
344
SDL_NAME(snd_pcm_drain)(pcm_handle);
SDL_NAME(snd_pcm_close)(pcm_handle);
345
pcm_handle = NULL;
346
347
348
}
}
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
static int ALSA_finalize_hardware(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *hwparams, int override)
{
int status;
snd_pcm_uframes_t bufsize;
/* "set" the hardware with the desired parameters */
status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);
if ( status < 0 ) {
return(-1);
}
/* Get samples for the actual buffer size */
status = SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize);
if ( status < 0 ) {
return(-1);
}
if ( !override && bufsize != spec->samples * 2 ) {
return(-1);
}
/* FIXME: Is this safe to do? */
spec->samples = bufsize / 2;
/* This is useful for debugging */
if ( getenv("SDL_AUDIO_ALSA_DEBUG") ) {
snd_pcm_sframes_t persize = 0;
unsigned int periods = 0;
SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, NULL);
SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, NULL);
fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize);
}
return(0);
}
385
static int ALSA_set_period_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override)
386
387
388
389
390
391
392
393
394
395
396
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
unsigned int periods;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);
397
398
399
400
401
402
403
if ( !override ) {
env = getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
}
}
frames = spec->samples;
status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
if ( status < 0 ) {
return(-1);
}
periods = 2;
status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, spec, hwparams, override);
}
422
static int ALSA_set_buffer_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override)
423
424
425
426
427
428
429
430
431
432
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);
433
434
435
436
437
438
439
if ( !override ) {
env = getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
440
441
442
443
444
445
446
447
448
449
450
451
}
}
frames = spec->samples * 2;
status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, spec, hwparams, override);
}
452
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
453
{
454
int status;
455
456
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
457
snd_pcm_format_t format;
458
unsigned int rate;
459
unsigned int channels;
460
snd_pcm_uframes_t bufsize;
461
Uint16 test_format;
462
463
/* Open the audio device */
464
465
466
/* Name of device should depend on # channels in spec */
status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
467
if ( status < 0 ) {
468
SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
469
470
471
return(-1);
}
472
/* Figure out what the hardware is capable of */
473
474
snd_pcm_hw_params_alloca(&hwparams);
status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);
475
if ( status < 0 ) {
476
SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
477
478
479
ALSA_CloseAudio(this);
return(-1);
}
480
481
/* SDL only uses interleaved sample output */
482
status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
483
if ( status < 0 ) {
484
SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
485
486
487
ALSA_CloseAudio(this);
return(-1);
}
488
489
/* Try for a closest match on audio format */
490
status = -1;
491
for ( test_format = SDL_FirstAudioFormat(spec->format);
492
493
test_format && (status < 0); ) {
switch ( test_format ) {
494
case AUDIO_U8:
495
format = SND_PCM_FORMAT_U8;
496
497
break;
case AUDIO_S8:
498
format = SND_PCM_FORMAT_S8;
499
500
break;
case AUDIO_S16LSB:
501
format = SND_PCM_FORMAT_S16_LE;
502
503
break;
case AUDIO_S16MSB:
504
format = SND_PCM_FORMAT_S16_BE;
505
506
break;
case AUDIO_U16LSB:
507
format = SND_PCM_FORMAT_U16_LE;
508
509
break;
case AUDIO_U16MSB:
510
format = SND_PCM_FORMAT_U16_BE;
511
512
break;
default:
513
format = 0;
514
515
break;
}
516
if ( format != 0 ) {
517
status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);
518
519
}
if ( status < 0 ) {
520
521
522
test_format = SDL_NextAudioFormat();
}
}
523
if ( status < 0 ) {
524
SDL_SetError("Couldn't find any hardware audio formats");
525
ALSA_CloseAudio(this);
526
527
528
529
return(-1);
}
spec->format = test_format;
530
/* Set the number of channels */
531
status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);
532
channels = spec->channels;
533
if ( status < 0 ) {
534
535
status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels);
if ( status < 0 ) {
536
537
538
539
SDL_SetError("Couldn't set audio channels");
ALSA_CloseAudio(this);
return(-1);
}
540
spec->channels = channels;
541
}
542
543
/* Set the audio rate */
544
545
546
rate = spec->freq;
status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
547
if ( status < 0 ) {
548
SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
549
550
551
ALSA_CloseAudio(this);
return(-1);
}
552
spec->freq = rate;
553
554
/* Set the buffer size, in samples */
555
556
557
558
if ( ALSA_set_period_size(this, spec, hwparams, 0) < 0 &&
ALSA_set_buffer_size(this, spec, hwparams, 0) < 0 ) {
/* Failed to set desired buffer size, do the best you can... */
if ( ALSA_set_period_size(this, spec, hwparams, 1) < 0 ) {
559
SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));
560
561
562
ALSA_CloseAudio(this);
return(-1);
}
563
}
564
565
566
567
568
569
570
571
572
/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);
if ( status < 0 ) {
SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
573
status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1);
574
575
576
577
578
579
if ( status < 0 ) {
SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));
ALSA_CloseAudio(this);
return(-1);
}
status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);
580
if ( status < 0 ) {
581
SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));
582
ALSA_CloseAudio(this);
583
584
585
return(-1);
}
586
587
588
589
590
591
592
593
594
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
ALSA_CloseAudio(this);
return(-1);
595
}
596
SDL_memset(mixbuf, spec->silence, spec->size);
597
598
/* Switch to blocking mode for playback */
599
SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
600
601
602
603
/* We're ready to rock and roll. :-) */
return(0);
}