Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 252:e8157fcb3114
Updated the source with the correct e-mail address
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Fri, 14 Dec 2001 12:38:15 +0000 |
| parents | 74212992fb08 |
| children | f6ffac90895c |
| rev | line source |
|---|---|
| 0 | 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 | |
|
252
e8157fcb3114
Updated the source with the correct e-mail address
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
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 } |