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comparison src/audio/alsa/SDL_alsa_audio.c @ 0:74212992fb08

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Initial revision
author Sam Lantinga <slouken@lokigames.com>
date Thu, 26 Apr 2001 16:45:43 +0000
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children e8157fcb3114
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-1:000000000000 0:74212992fb08
1 /*
2 SDL - Simple DirectMedia Layer
3 Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
4
5 This library is free software; you can redistribute it and/or
6 modify it under the terms of the GNU Library General Public
7 License as published by the Free Software Foundation; either
8 version 2 of the License, or (at your option) any later version.
9
10 This library is distributed in the hope that it will be useful,
11 but WITHOUT ANY WARRANTY; without even the implied warranty of
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 Library General Public License for more details.
14
15 You should have received a copy of the GNU Library General Public
16 License along with this library; if not, write to the Free
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
18
19 Sam Lantinga
20 slouken@devolution.com
21 */
22
23
24
25 /* Allow access to a raw mixing buffer */
26
27 #include <stdlib.h>
28 #include <stdio.h>
29 #include <string.h>
30 #include <errno.h>
31 #include <unistd.h>
32 #include <fcntl.h>
33 #include <signal.h>
34 #include <sys/types.h>
35 #include <sys/time.h>
36
37 #include "SDL_audio.h"
38 #include "SDL_error.h"
39 #include "SDL_audiomem.h"
40 #include "SDL_audio_c.h"
41 #include "SDL_timer.h"
42 #include "SDL_alsa_audio.h"
43
44 /* The tag name used by ALSA audio */
45 #define DRIVER_NAME "alsa"
46
47 /* default card and device numbers as listed in dev/snd */
48 static int card_no = 0;
49 static int device_no = 0;
50
51 /* default channel communication parameters */
52 #define DEFAULT_CPARAMS_RATE 22050
53 #define DEFAULT_CPARAMS_VOICES 1
54 #define DEFAULT_CPARAMS_FRAG_SIZE 512
55 #define DEFAULT_CPARAMS_FRAGS_MIN 1
56 #define DEFAULT_CPARAMS_FRAGS_MAX -1
57
58 /* Open the audio device for playback, and don't block if busy */
59 #define OPEN_FLAGS (SND_PCM_OPEN_PLAYBACK|SND_PCM_OPEN_NONBLOCK)
60
61 /* Audio driver functions */
62 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec);
63 static void PCM_WaitAudio(_THIS);
64 static void PCM_PlayAudio(_THIS);
65 static Uint8 *PCM_GetAudioBuf(_THIS);
66 static void PCM_CloseAudio(_THIS);
67
68 /* PCM transfer channel parameters initialize function */
69 static void init_pcm_cparams(snd_pcm_channel_params_t* cparams)
70 {
71 memset(cparams,0,sizeof(snd_pcm_channel_params_t));
72
73 cparams->channel = SND_PCM_CHANNEL_PLAYBACK;
74 cparams->mode = SND_PCM_MODE_BLOCK;
75 cparams->start_mode = SND_PCM_START_DATA; //_FULL
76 cparams->stop_mode = SND_PCM_STOP_STOP;
77 cparams->format.format = SND_PCM_SFMT_S16_LE;
78 cparams->format.interleave = 1;
79 cparams->format.rate = DEFAULT_CPARAMS_RATE;
80 cparams->format.voices = DEFAULT_CPARAMS_VOICES;
81 cparams->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
82 cparams->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
83 cparams->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
84 }
85
86 /* Audio driver bootstrap functions */
87
88 static int Audio_Available(void)
89 /*
90 See if we can open a nonblocking channel.
91 Return value '1' means we can.
92 Return value '0' means we cannot.
93 */
94 {
95 int available;
96 int rval;
97 snd_pcm_t *handle;
98 snd_pcm_channel_params_t cparams;
99 #ifdef DEBUG_AUDIO
100 snd_pcm_channel_status_t cstatus;
101 #endif
102
103 available = 0;
104 handle = NULL;
105
106 init_pcm_cparams(&cparams);
107
108 rval = snd_pcm_open(&handle, card_no, device_no, OPEN_FLAGS);
109 if (rval >= 0)
110 {
111 rval = snd_pcm_plugin_params(handle, &cparams);
112
113 #ifdef DEBUG_AUDIO
114 snd_pcm_plugin_status(handle, &cstatus);
115 printf("status after snd_pcm_plugin_params call = %d\n",cstatus.status);
116 #endif
117 if (rval >= 0)
118 {
119 available = 1;
120 }
121 else
122 {
123 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
124 }
125
126 if ((rval = snd_pcm_close(handle)) < 0)
127 {
128 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
129 available = 0;
130 }
131 }
132 else
133 {
134 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
135 }
136
137 return(available);
138 }
139
140 static void Audio_DeleteDevice(SDL_AudioDevice *device)
141 {
142 free(device->hidden);
143 free(device);
144 }
145
146 static SDL_AudioDevice *Audio_CreateDevice(int devindex)
147 {
148 SDL_AudioDevice *this;
149
150 /* Initialize all variables that we clean on shutdown */
151 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
152 if ( this ) {
153 memset(this, 0, (sizeof *this));
154 this->hidden = (struct SDL_PrivateAudioData *)
155 malloc((sizeof *this->hidden));
156 }
157 if ( (this == NULL) || (this->hidden == NULL) ) {
158 SDL_OutOfMemory();
159 if ( this ) {
160 free(this);
161 }
162 return(0);
163 }
164 memset(this->hidden, 0, (sizeof *this->hidden));
165 audio_handle = NULL;
166
167 /* Set the function pointers */
168 this->OpenAudio = PCM_OpenAudio;
169 this->WaitAudio = PCM_WaitAudio;
170 this->PlayAudio = PCM_PlayAudio;
171 this->GetAudioBuf = PCM_GetAudioBuf;
172 this->CloseAudio = PCM_CloseAudio;
173
174 this->free = Audio_DeleteDevice;
175
176 return this;
177 }
178
179 AudioBootStrap ALSA_bootstrap = {
180 DRIVER_NAME, "ALSA PCM audio",
181 Audio_Available, Audio_CreateDevice
182 };
183
184 /* This function waits until it is possible to write a full sound buffer */
185 static void PCM_WaitAudio(_THIS)
186 {
187
188 /* Check to see if the thread-parent process is still alive */
189 { static int cnt = 0;
190 /* Note that this only works with thread implementations
191 that use a different process id for each thread.
192 */
193 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
194 if ( kill(parent, 0) < 0 ) {
195 this->enabled = 0;
196 }
197 }
198 }
199
200 /* See if we need to use timed audio synchronization */
201 if ( frame_ticks )
202 {
203 /* Use timer for general audio synchronization */
204 Sint32 ticks;
205
206 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
207 if ( ticks > 0 )
208 {
209 SDL_Delay(ticks);
210 }
211 }
212 else
213 {
214 /* Use select() for audio synchronization */
215 fd_set fdset;
216 struct timeval timeout;
217 FD_ZERO(&fdset);
218 FD_SET(audio_fd, &fdset);
219 timeout.tv_sec = 10;
220 timeout.tv_usec = 0;
221 #ifdef DEBUG_AUDIO
222 fprintf(stderr, "Waiting for audio to get ready\n");
223 #endif
224 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 )
225 {
226 const char *message =
227 "Audio timeout - buggy audio driver? (disabled)";
228 /* In general we should never print to the screen,
229 but in this case we have no other way of letting
230 the user know what happened.
231 */
232 fprintf(stderr, "SDL: %s\n", message);
233 this->enabled = 0;
234 /* Don't try to close - may hang */
235 audio_fd = -1;
236 #ifdef DEBUG_AUDIO
237 fprintf(stderr, "Done disabling audio\n");
238 #endif
239 }
240 #ifdef DEBUG_AUDIO
241 fprintf(stderr, "Ready!\n");
242 #endif
243 }
244 }
245
246 static snd_pcm_channel_status_t cstatus;
247
248 static void PCM_PlayAudio(_THIS)
249 {
250 int written, rval;
251
252 /* Write the audio data, checking for EAGAIN (buffer full) and underrun */
253 do {
254 written = snd_pcm_plugin_write(audio_handle, pcm_buf, pcm_len);
255 #ifdef DEBUG_AUDIO
256 fprintf(stderr, "written = %d pcm_len = %d\n",written,pcm_len);
257 #endif
258 if (written != pcm_len)
259 {
260 if (errno == EAGAIN)
261 {
262 SDL_Delay(1); /* Let a little CPU time go by and try to write again */
263 #ifdef DEBUG_AUDIO
264 fprintf(stderr, "errno == EAGAIN\n");
265 #endif
266 }
267 else
268 {
269 if( (rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0 )
270 {
271 SDL_SetError("snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
272 return;
273 }
274 if ( (cstatus.status == SND_PCM_STATUS_UNDERRUN)
275 ||(cstatus.status == SND_PCM_STATUS_READY) )
276 {
277 #ifdef DEBUG_AUDIO
278 fprintf(stderr, "buffer underrun\n");
279 #endif
280 if ( (rval = snd_pcm_plugin_prepare (audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0 )
281 {
282 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n",snd_strerror(rval) );
283 return;
284 }
285 /* if we reach here, try to write again */
286 }
287 }
288 }
289 } while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
290
291 /* Set the next write frame */
292 if ( frame_ticks ) {
293 next_frame += frame_ticks;
294 }
295
296 /* If we couldn't write, assume fatal error for now */
297 if ( written < 0 ) {
298 this->enabled = 0;
299 }
300 return;
301 }
302
303 static Uint8 *PCM_GetAudioBuf(_THIS)
304 {
305 return(pcm_buf);
306 }
307
308 static void PCM_CloseAudio(_THIS)
309 {
310 int rval;
311
312 if ( pcm_buf != NULL ) {
313 free(pcm_buf);
314 pcm_buf = NULL;
315 }
316 if ( audio_handle != NULL ) {
317 if ((rval = snd_pcm_plugin_flush(audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0)
318 {
319 SDL_SetError("snd_pcm_plugin_flush failed: %s\n",snd_strerror(rval));
320 return;
321 }
322 if ((rval = snd_pcm_close(audio_handle)) < 0)
323 {
324 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
325 return;
326 }
327 audio_handle = NULL;
328 }
329 }
330
331 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec)
332 {
333 int rval;
334 snd_pcm_channel_params_t cparams;
335 snd_pcm_channel_setup_t csetup;
336 int format;
337 Uint16 test_format;
338 int twidth;
339
340 /* initialize channel transfer parameters to default */
341 init_pcm_cparams(&cparams);
342
343 /* Reset the timer synchronization flag */
344 frame_ticks = 0.0;
345
346 /* Open the audio device */
347
348 rval = snd_pcm_open(&audio_handle, card_no, device_no, OPEN_FLAGS);
349 if ( rval < 0 ) {
350 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
351 return(-1);
352 }
353
354 #ifdef PLUGIN_DISABLE_MMAP /* This is gone in newer versions of ALSA? */
355 /* disable count status parameter */
356 if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
357 {
358 SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
359 return(-1);
360 }
361 #endif
362
363 pcm_buf = NULL;
364
365 /* Try for a closest match on audio format */
366 format = 0;
367 for ( test_format = SDL_FirstAudioFormat(spec->format);
368 ! format && test_format; )
369 {
370 #ifdef DEBUG_AUDIO
371 fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
372 #endif
373 /* if match found set format to equivalent ALSA format */
374 switch ( test_format ) {
375 case AUDIO_U8:
376 format = SND_PCM_SFMT_U8;
377 cparams.buf.block.frag_size = spec->samples * spec->channels;
378 break;
379 case AUDIO_S8:
380 format = SND_PCM_SFMT_S8;
381 cparams.buf.block.frag_size = spec->samples * spec->channels;
382 break;
383 case AUDIO_S16LSB:
384 format = SND_PCM_SFMT_S16_LE;
385 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
386 break;
387 case AUDIO_S16MSB:
388 format = SND_PCM_SFMT_S16_BE;
389 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
390 break;
391 case AUDIO_U16LSB:
392 format = SND_PCM_SFMT_U16_LE;
393 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
394 break;
395 case AUDIO_U16MSB:
396 format = SND_PCM_SFMT_U16_BE;
397 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
398 break;
399 default:
400 break;
401 }
402 if ( ! format ) {
403 test_format = SDL_NextAudioFormat();
404 }
405 }
406 if ( format == 0 ) {
407 SDL_SetError("Couldn't find any hardware audio formats");
408 return(-1);
409 }
410 spec->format = test_format;
411
412 /* Set the audio format */
413 cparams.format.format = format;
414
415 /* Set mono or stereo audio (currently only two channels supported) */
416 cparams.format.voices = spec->channels;
417
418 #ifdef DEBUG_AUDIO
419 printf("intializing channels %d\n", cparams.format.voices);
420 #endif
421
422 /* Set rate */
423 cparams.format.rate = spec->freq ;
424
425 /* Setup the transfer parameters according to cparams */
426 rval = snd_pcm_plugin_params(audio_handle, &cparams);
427 if (rval < 0) {
428 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
429 return(-1);
430 }
431
432 /* Make sure channel is setup right one last time */
433 memset( &csetup, 0, sizeof( csetup ) );
434 csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
435 if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
436 {
437 SDL_SetError("Unable to setup playback channel\n" );
438 return(-1);
439 }
440
441 #ifdef DEBUG_AUDIO
442 else
443 {
444 fprintf(stderr,"requested format: %d\n",cparams.format.format);
445 fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
446 fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
447
448 fprintf(stderr,"real format: %d\n", csetup.format.format );
449 fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
450 fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
451 }
452 #endif // DEBUG_AUDIO
453
454 /* Allocate memory to the audio buffer and initialize with silence
455 (Note that buffer size must be a multiple of fragment size, so find closest multiple)
456 */
457
458 twidth = snd_pcm_format_width(format);
459 if (twidth < 0) {
460 printf("snd_pcm_format_width failed\n");
461 twidth = 0;
462 }
463 #ifdef DEBUG_AUDIO
464 printf("format is %d bits wide\n",twidth);
465 #endif
466
467 pcm_len = csetup.buf.block.frag_size * (twidth/8) * csetup.format.voices ;
468
469 #ifdef DEBUG_AUDIO
470 printf("pcm_len set to %d\n", pcm_len);
471 #endif
472
473 if (pcm_len == 0)
474 {
475 pcm_len = csetup.buf.block.frag_size;
476 }
477
478 pcm_buf = (Uint8*)malloc(pcm_len);
479 if (pcm_buf == NULL) {
480 SDL_SetError("pcm_buf malloc failed\n");
481 return(-1);
482 }
483 memset(pcm_buf,spec->silence,pcm_len);
484
485 #ifdef DEBUG_AUDIO
486 fprintf(stderr,"pcm_buf malloced and silenced.\n");
487 #endif
488
489 /* get the file descriptor */
490 if( (audio_fd = snd_pcm_file_descriptor(audio_handle, device_no)) < 0)
491 {
492 fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
493 }
494
495 /* Trigger audio playback */
496 rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
497 if (rval < 0) {
498 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
499 return(-1);
500 }
501 rval = snd_pcm_playback_go(audio_handle);
502 if (rval < 0) {
503 SDL_SetError("snd_pcm_playback_go failed: %s\n", snd_strerror (rval));
504 return(-1);
505 }
506
507 /* Check to see if we need to use select() workaround */
508 { char *workaround;
509 workaround = getenv("SDL_DSP_NOSELECT");
510 if ( workaround ) {
511 frame_ticks = (float)(spec->samples*1000)/spec->freq;
512 next_frame = SDL_GetTicks()+frame_ticks;
513 }
514 }
515
516 /* Get the parent process id (we're the parent of the audio thread) */
517 parent = getpid();
518
519 /* We're ready to rock and roll. :-) */
520 return(0);
521 }