Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 302:8a86bdf34f0f
Added Atari joystick support (thanks Patrice!)
author | Sam Lantinga <slouken@libsdl.org> |
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date | Thu, 07 Mar 2002 20:23:11 +0000 |
parents | f6ffac90895c |
children | 30935e76acb5 |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
297
f6ffac90895c
Updated copyright information for 2002
Sam Lantinga <slouken@libsdl.org>
parents:
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3 Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002 Sam Lantinga |
0 | 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 | |
252
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Updated the source with the correct e-mail address
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
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20 slouken@libsdl.org |
0 | 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 } |