Mercurial > sdl-ios-xcode
annotate src/audio/paudio/SDL_paudio.c @ 4361:991e4bcfc7a8 SDL-1.2
Added an error message for when ALSA audio fails
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Mon, 19 Oct 2009 05:15:21 +0000 |
| parents | a1b03ba2fcd0 |
| children |
| rev | line source |
|---|---|
| 0 | 1 /* |
|
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Updated copyright information and removed rcs id lines (problematic in branch merges)
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2 SDL - Simple DirectMedia Layer |
| 4159 | 3 Copyright (C) 1997-2009 Sam Lantinga |
| 0 | 4 |
| 5 This library is free software; you can redistribute it and/or | |
|
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6 modify it under the terms of the GNU Lesser General Public |
| 0 | 7 License as published by the Free Software Foundation; either |
|
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8 version 2.1 of the License, or (at your option) any later version. |
| 0 | 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 | |
|
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13 Lesser General Public License for more details. |
| 0 | 14 |
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15 You should have received a copy of the GNU Lesser General Public |
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c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
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16 License along with this library; if not, write to the Free Software |
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17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
| 0 | 18 |
| 19 Carsten Griwodz | |
| 20 griff@kom.tu-darmstadt.de | |
| 21 | |
| 22 based on linux/SDL_dspaudio.c by Sam Lantinga | |
| 23 */ | |
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24 #include "SDL_config.h" |
| 0 | 25 |
| 26 /* Allow access to a raw mixing buffer */ | |
| 27 | |
| 28 #include <errno.h> | |
| 29 #include <unistd.h> | |
| 30 #include <fcntl.h> | |
| 31 #include <sys/time.h> | |
| 32 #include <sys/ioctl.h> | |
| 33 #include <sys/stat.h> | |
| 34 | |
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35 #include "SDL_timer.h" |
| 0 | 36 #include "SDL_audio.h" |
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37 #include "../SDL_audiomem.h" |
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38 #include "../SDL_audio_c.h" |
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New configure-based build system. Still work in progress, but much improved
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39 #include "../SDL_audiodev_c.h" |
| 0 | 40 #include "SDL_paudio.h" |
| 41 | |
| 42 #define DEBUG_AUDIO 1 | |
| 43 | |
| 44 /* A conflict within AIX 4.3.3 <sys/> headers and probably others as well. | |
| 45 * I guess nobody ever uses audio... Shame over AIX header files. */ | |
| 46 #include <sys/machine.h> | |
| 47 #undef BIG_ENDIAN | |
| 48 #include <sys/audio.h> | |
| 49 | |
| 50 /* The tag name used by paud audio */ | |
| 51 #define Paud_DRIVER_NAME "paud" | |
| 52 | |
| 53 /* Open the audio device for playback, and don't block if busy */ | |
| 54 /* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */ | |
| 55 #define OPEN_FLAGS O_WRONLY | |
| 56 | |
| 57 /* Audio driver functions */ | |
| 58 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec); | |
| 59 static void Paud_WaitAudio(_THIS); | |
| 60 static void Paud_PlayAudio(_THIS); | |
| 61 static Uint8 *Paud_GetAudioBuf(_THIS); | |
| 62 static void Paud_CloseAudio(_THIS); | |
| 63 | |
| 64 /* Audio driver bootstrap functions */ | |
| 65 | |
| 66 static int Audio_Available(void) | |
| 67 { | |
| 68 int fd; | |
| 69 int available; | |
| 70 | |
| 71 available = 0; | |
| 72 fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); | |
| 73 if ( fd >= 0 ) { | |
| 74 available = 1; | |
| 75 close(fd); | |
| 76 } | |
| 77 return(available); | |
| 78 } | |
| 79 | |
| 80 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
| 81 { | |
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82 SDL_free(device->hidden); |
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83 SDL_free(device); |
| 0 | 84 } |
| 85 | |
| 86 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
| 87 { | |
| 88 SDL_AudioDevice *this; | |
| 89 | |
| 90 /* Initialize all variables that we clean on shutdown */ | |
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91 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
| 0 | 92 if ( this ) { |
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93 SDL_memset(this, 0, (sizeof *this)); |
| 0 | 94 this->hidden = (struct SDL_PrivateAudioData *) |
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95 SDL_malloc((sizeof *this->hidden)); |
| 0 | 96 } |
| 97 if ( (this == NULL) || (this->hidden == NULL) ) { | |
| 98 SDL_OutOfMemory(); | |
| 99 if ( this ) { | |
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100 SDL_free(this); |
| 0 | 101 } |
| 102 return(0); | |
| 103 } | |
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104 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
| 0 | 105 audio_fd = -1; |
| 106 | |
| 107 /* Set the function pointers */ | |
| 108 this->OpenAudio = Paud_OpenAudio; | |
| 109 this->WaitAudio = Paud_WaitAudio; | |
| 110 this->PlayAudio = Paud_PlayAudio; | |
| 111 this->GetAudioBuf = Paud_GetAudioBuf; | |
| 112 this->CloseAudio = Paud_CloseAudio; | |
| 113 | |
| 114 this->free = Audio_DeleteDevice; | |
| 115 | |
| 116 return this; | |
| 117 } | |
| 118 | |
| 119 AudioBootStrap Paud_bootstrap = { | |
| 120 Paud_DRIVER_NAME, "AIX Paudio", | |
| 121 Audio_Available, Audio_CreateDevice | |
| 122 }; | |
| 123 | |
| 124 /* This function waits until it is possible to write a full sound buffer */ | |
| 125 static void Paud_WaitAudio(_THIS) | |
| 126 { | |
| 127 fd_set fdset; | |
| 128 | |
| 129 /* See if we need to use timed audio synchronization */ | |
| 130 if ( frame_ticks ) { | |
| 131 /* Use timer for general audio synchronization */ | |
| 132 Sint32 ticks; | |
| 133 | |
| 134 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; | |
| 135 if ( ticks > 0 ) { | |
| 136 SDL_Delay(ticks); | |
| 137 } | |
| 138 } else { | |
| 139 audio_buffer paud_bufinfo; | |
| 140 | |
| 141 /* Use select() for audio synchronization */ | |
| 142 struct timeval timeout; | |
| 143 FD_ZERO(&fdset); | |
| 144 FD_SET(audio_fd, &fdset); | |
| 145 | |
| 146 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
| 147 #ifdef DEBUG_AUDIO | |
| 148 fprintf(stderr, "Couldn't get audio buffer information\n"); | |
| 149 #endif | |
| 150 timeout.tv_sec = 10; | |
| 151 timeout.tv_usec = 0; | |
| 152 } else { | |
| 153 long ms_in_buf = paud_bufinfo.write_buf_time; | |
| 154 timeout.tv_sec = ms_in_buf/1000; | |
| 155 ms_in_buf = ms_in_buf - timeout.tv_sec*1000; | |
| 156 timeout.tv_usec = ms_in_buf*1000; | |
| 157 #ifdef DEBUG_AUDIO | |
| 158 fprintf( stderr, | |
| 159 "Waiting for write_buf_time=%ld,%ld\n", | |
| 160 timeout.tv_sec, | |
| 161 timeout.tv_usec ); | |
| 162 #endif | |
| 163 } | |
| 164 | |
| 165 #ifdef DEBUG_AUDIO | |
| 166 fprintf(stderr, "Waiting for audio to get ready\n"); | |
| 167 #endif | |
| 168 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) { | |
| 169 const char *message = "Audio timeout - buggy audio driver? (disabled)"; | |
| 170 /* | |
| 171 * In general we should never print to the screen, | |
| 172 * but in this case we have no other way of letting | |
| 173 * the user know what happened. | |
| 174 */ | |
| 175 fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message); | |
| 176 this->enabled = 0; | |
| 177 /* Don't try to close - may hang */ | |
| 178 audio_fd = -1; | |
| 179 #ifdef DEBUG_AUDIO | |
| 180 fprintf(stderr, "Done disabling audio\n"); | |
| 181 #endif | |
| 182 } | |
| 183 #ifdef DEBUG_AUDIO | |
| 184 fprintf(stderr, "Ready!\n"); | |
| 185 #endif | |
| 186 } | |
| 187 } | |
| 188 | |
| 189 static void Paud_PlayAudio(_THIS) | |
| 190 { | |
| 191 int written; | |
| 192 | |
| 193 /* Write the audio data, checking for EAGAIN on broken audio drivers */ | |
| 194 do { | |
| 195 written = write(audio_fd, mixbuf, mixlen); | |
| 196 if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) { | |
| 197 SDL_Delay(1); /* Let a little CPU time go by */ | |
| 198 } | |
| 199 } while ( (written < 0) && | |
| 200 ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) ); | |
| 201 | |
| 202 /* If timer synchronization is enabled, set the next write frame */ | |
| 203 if ( frame_ticks ) { | |
| 204 next_frame += frame_ticks; | |
| 205 } | |
| 206 | |
| 207 /* If we couldn't write, assume fatal error for now */ | |
| 208 if ( written < 0 ) { | |
| 209 this->enabled = 0; | |
| 210 } | |
| 211 #ifdef DEBUG_AUDIO | |
| 212 fprintf(stderr, "Wrote %d bytes of audio data\n", written); | |
| 213 #endif | |
| 214 } | |
| 215 | |
| 216 static Uint8 *Paud_GetAudioBuf(_THIS) | |
| 217 { | |
| 218 return mixbuf; | |
| 219 } | |
| 220 | |
| 221 static void Paud_CloseAudio(_THIS) | |
| 222 { | |
| 223 if ( mixbuf != NULL ) { | |
| 224 SDL_FreeAudioMem(mixbuf); | |
| 225 mixbuf = NULL; | |
| 226 } | |
| 227 if ( audio_fd >= 0 ) { | |
| 228 close(audio_fd); | |
| 229 audio_fd = -1; | |
| 230 } | |
| 231 } | |
| 232 | |
| 233 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec) | |
| 234 { | |
| 235 char audiodev[1024]; | |
| 236 int format; | |
| 237 int bytes_per_sample; | |
| 238 Uint16 test_format; | |
| 239 audio_init paud_init; | |
| 240 audio_buffer paud_bufinfo; | |
| 241 audio_status paud_status; | |
| 242 audio_control paud_control; | |
| 243 audio_change paud_change; | |
| 244 | |
| 245 /* Reset the timer synchronization flag */ | |
| 246 frame_ticks = 0.0; | |
| 247 | |
| 248 /* Open the audio device */ | |
| 249 audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); | |
| 250 if ( audio_fd < 0 ) { | |
| 251 SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); | |
| 252 return -1; | |
| 253 } | |
| 254 | |
| 255 /* | |
| 256 * We can't set the buffer size - just ask the device for the maximum | |
| 257 * that we can have. | |
| 258 */ | |
| 259 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
| 260 SDL_SetError("Couldn't get audio buffer information"); | |
| 261 return -1; | |
| 262 } | |
| 263 | |
| 264 mixbuf = NULL; | |
| 265 | |
| 266 if ( spec->channels > 1 ) | |
| 267 spec->channels = 2; | |
| 268 else | |
| 269 spec->channels = 1; | |
| 270 | |
| 271 /* | |
| 272 * Fields in the audio_init structure: | |
| 273 * | |
| 274 * Ignored by us: | |
| 275 * | |
| 276 * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only? | |
| 277 * paud.slot_number; * slot number of the adapter | |
| 278 * paud.device_id; * adapter identification number | |
| 279 * | |
| 280 * Input: | |
| 281 * | |
| 282 * paud.srate; * the sampling rate in Hz | |
| 283 * paud.bits_per_sample; * 8, 16, 32, ... | |
| 284 * paud.bsize; * block size for this rate | |
| 285 * paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX | |
| 286 * paud.channels; * 1=mono, 2=stereo | |
| 287 * paud.flags; * FIXED - fixed length data | |
| 288 * * LEFT_ALIGNED, RIGHT_ALIGNED (var len only) | |
| 289 * * TWOS_COMPLEMENT - 2's complement data | |
| 290 * * SIGNED - signed? comment seems wrong in sys/audio.h | |
| 291 * * BIG_ENDIAN | |
| 292 * paud.operation; * PLAY, RECORD | |
| 293 * | |
| 294 * Output: | |
| 295 * | |
| 296 * paud.flags; * PITCH - pitch is supported | |
| 297 * * INPUT - input is supported | |
| 298 * * OUTPUT - output is supported | |
| 299 * * MONITOR - monitor is supported | |
| 300 * * VOLUME - volume is supported | |
| 301 * * VOLUME_DELAY - volume delay is supported | |
| 302 * * BALANCE - balance is supported | |
| 303 * * BALANCE_DELAY - balance delay is supported | |
| 304 * * TREBLE - treble control is supported | |
| 305 * * BASS - bass control is supported | |
| 306 * * BESTFIT_PROVIDED - best fit returned | |
| 307 * * LOAD_CODE - DSP load needed | |
| 308 * paud.rc; * NO_PLAY - DSP code can't do play requests | |
| 309 * * NO_RECORD - DSP code can't do record requests | |
| 310 * * INVALID_REQUEST - request was invalid | |
| 311 * * CONFLICT - conflict with open's flags | |
| 312 * * OVERLOADED - out of DSP MIPS or memory | |
| 313 * paud.position_resolution; * smallest increment for position | |
| 314 */ | |
| 315 | |
| 316 paud_init.srate = spec->freq; | |
| 317 paud_init.mode = PCM; | |
| 318 paud_init.operation = PLAY; | |
| 319 paud_init.channels = spec->channels; | |
| 320 | |
| 321 /* Try for a closest match on audio format */ | |
| 322 format = 0; | |
| 323 for ( test_format = SDL_FirstAudioFormat(spec->format); | |
| 324 ! format && test_format; ) { | |
| 325 #ifdef DEBUG_AUDIO | |
| 326 fprintf(stderr, "Trying format 0x%4.4x\n", test_format); | |
| 327 #endif | |
| 328 switch ( test_format ) { | |
| 329 case AUDIO_U8: | |
| 330 bytes_per_sample = 1; | |
| 331 paud_init.bits_per_sample = 8; | |
| 332 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
| 333 format = 1; | |
| 334 break; | |
| 335 case AUDIO_S8: | |
| 336 bytes_per_sample = 1; | |
| 337 paud_init.bits_per_sample = 8; | |
| 338 paud_init.flags = SIGNED | | |
| 339 TWOS_COMPLEMENT | FIXED; | |
| 340 format = 1; | |
| 341 break; | |
| 342 case AUDIO_S16LSB: | |
| 343 bytes_per_sample = 2; | |
| 344 paud_init.bits_per_sample = 16; | |
| 345 paud_init.flags = SIGNED | | |
| 346 TWOS_COMPLEMENT | FIXED; | |
| 347 format = 1; | |
| 348 break; | |
| 349 case AUDIO_S16MSB: | |
| 350 bytes_per_sample = 2; | |
| 351 paud_init.bits_per_sample = 16; | |
| 352 paud_init.flags = BIG_ENDIAN | | |
| 353 SIGNED | | |
| 354 TWOS_COMPLEMENT | FIXED; | |
| 355 format = 1; | |
| 356 break; | |
| 357 case AUDIO_U16LSB: | |
| 358 bytes_per_sample = 2; | |
| 359 paud_init.bits_per_sample = 16; | |
| 360 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
| 361 format = 1; | |
| 362 break; | |
| 363 case AUDIO_U16MSB: | |
| 364 bytes_per_sample = 2; | |
| 365 paud_init.bits_per_sample = 16; | |
| 366 paud_init.flags = BIG_ENDIAN | | |
| 367 TWOS_COMPLEMENT | FIXED; | |
| 368 format = 1; | |
| 369 break; | |
| 370 default: | |
| 371 break; | |
| 372 } | |
| 373 if ( ! format ) { | |
| 374 test_format = SDL_NextAudioFormat(); | |
| 375 } | |
| 376 } | |
| 377 if ( format == 0 ) { | |
| 378 #ifdef DEBUG_AUDIO | |
| 379 fprintf(stderr, "Couldn't find any hardware audio formats\n"); | |
| 380 #endif | |
| 381 SDL_SetError("Couldn't find any hardware audio formats"); | |
| 382 return -1; | |
| 383 } | |
| 384 spec->format = test_format; | |
| 385 | |
| 386 /* | |
| 387 * We know the buffer size and the max number of subsequent writes | |
| 388 * that can be pending. If more than one can pend, allow the application | |
| 389 * to do something like double buffering between our write buffer and | |
| 390 * the device's own buffer that we are filling with write() anyway. | |
| 391 * | |
| 392 * We calculate spec->samples like this because SDL_CalculateAudioSpec() | |
| 393 * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2) | |
| 394 * into spec->size in return. | |
| 395 */ | |
| 396 if ( paud_bufinfo.request_buf_cap == 1 ) | |
| 397 { | |
| 398 spec->samples = paud_bufinfo.write_buf_cap | |
| 399 / bytes_per_sample | |
| 400 / spec->channels; | |
| 401 } | |
| 402 else | |
| 403 { | |
| 404 spec->samples = paud_bufinfo.write_buf_cap | |
| 405 / bytes_per_sample | |
| 406 / spec->channels | |
| 407 / 2; | |
| 408 } | |
| 409 paud_init.bsize = bytes_per_sample * spec->channels; | |
| 410 | |
| 411 SDL_CalculateAudioSpec(spec); | |
| 412 | |
| 413 /* | |
| 414 * The AIX paud device init can't modify the values of the audio_init | |
| 415 * structure that we pass to it. So we don't need any recalculation | |
| 416 * of this stuff and no reinit call as in linux dsp and dma code. | |
| 417 * | |
| 418 * /dev/paud supports all of the encoding formats, so we don't need | |
| 419 * to do anything like reopening the device, either. | |
| 420 */ | |
| 421 if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) { | |
| 422 switch ( paud_init.rc ) | |
| 423 { | |
| 424 case 1 : | |
| 425 SDL_SetError("Couldn't set audio format: DSP can't do play requests"); | |
| 426 return -1; | |
| 427 break; | |
| 428 case 2 : | |
| 429 SDL_SetError("Couldn't set audio format: DSP can't do record requests"); | |
| 430 return -1; | |
| 431 break; | |
| 432 case 4 : | |
| 433 SDL_SetError("Couldn't set audio format: request was invalid"); | |
| 434 return -1; | |
| 435 break; | |
| 436 case 5 : | |
| 437 SDL_SetError("Couldn't set audio format: conflict with open's flags"); | |
| 438 return -1; | |
| 439 break; | |
| 440 case 6 : | |
| 441 SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory"); | |
| 442 return -1; | |
| 443 break; | |
| 444 default : | |
| 445 SDL_SetError("Couldn't set audio format: not documented in sys/audio.h"); | |
| 446 return -1; | |
| 447 break; | |
| 448 } | |
| 449 } | |
| 450 | |
| 451 /* Allocate mixing buffer */ | |
| 452 mixlen = spec->size; | |
| 453 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); | |
| 454 if ( mixbuf == NULL ) { | |
| 455 return -1; | |
| 456 } | |
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457 SDL_memset(mixbuf, spec->silence, spec->size); |
| 0 | 458 |
| 459 /* | |
| 460 * Set some paramters: full volume, first speaker that we can find. | |
| 461 * Ignore the other settings for now. | |
| 462 */ | |
| 463 paud_change.input = AUDIO_IGNORE; /* the new input source */ | |
| 464 paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */ | |
| 465 paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */ | |
| 466 paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */ | |
| 467 paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */ | |
| 468 paud_change.balance = 0x3fffffff; /* the new balance */ | |
| 469 paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */ | |
| 470 paud_change.treble = AUDIO_IGNORE; /* the new treble state */ | |
| 471 paud_change.bass = AUDIO_IGNORE; /* the new bass state */ | |
| 472 paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */ | |
| 473 | |
| 474 paud_control.ioctl_request = AUDIO_CHANGE; | |
| 475 paud_control.request_info = (char*)&paud_change; | |
| 476 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
| 477 #ifdef DEBUG_AUDIO | |
| 478 fprintf(stderr, "Can't change audio display settings\n" ); | |
| 479 #endif | |
| 480 } | |
| 481 | |
| 482 /* | |
| 483 * Tell the device to expect data. Actual start will wait for | |
| 484 * the first write() call. | |
| 485 */ | |
| 486 paud_control.ioctl_request = AUDIO_START; | |
| 487 paud_control.position = 0; | |
| 488 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
| 489 #ifdef DEBUG_AUDIO | |
| 490 fprintf(stderr, "Can't start audio play\n" ); | |
| 491 #endif | |
| 492 SDL_SetError("Can't start audio play"); | |
| 493 return -1; | |
| 494 } | |
| 495 | |
| 496 /* Check to see if we need to use select() workaround */ | |
| 497 { char *workaround; | |
|
1336
3692456e7b0f
Use SDL_ prefixed versions of C library functions.
Sam Lantinga <slouken@libsdl.org>
parents:
1312
diff
changeset
|
498 workaround = SDL_getenv("SDL_DSP_NOSELECT"); |
| 0 | 499 if ( workaround ) { |
| 500 frame_ticks = (float)(spec->samples*1000)/spec->freq; | |
| 501 next_frame = SDL_GetTicks()+frame_ticks; | |
| 502 } | |
| 503 } | |
| 504 | |
| 505 /* Get the parent process id (we're the parent of the audio thread) */ | |
| 506 parent = getpid(); | |
| 507 | |
| 508 /* We're ready to rock and roll. :-) */ | |
| 509 return 0; | |
| 510 } | |
| 511 |