Mercurial > sdl-ios-xcode
annotate src/audio/paudio/SDL_paudio.c @ 1336:3692456e7b0f
Use SDL_ prefixed versions of C library functions.
FIXME:
Change #include <stdlib.h> to #include "SDL_stdlib.h"
Change #include <string.h> to #include "SDL_string.h"
Make sure nothing else broke because of this...
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Tue, 07 Feb 2006 06:59:48 +0000 |
| parents | c9b51268668f |
| children | 604d73db6802 |
| rev | line source |
|---|---|
| 0 | 1 /* |
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2 SDL - Simple DirectMedia Layer |
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3 Copyright (C) 1997-2006 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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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 */ | |
| 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 <sys/time.h> | |
| 34 #include <sys/ioctl.h> | |
| 35 #include <sys/stat.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_audiodev_c.h" | |
| 43 #include "SDL_paudio.h" | |
| 44 | |
| 45 #define DEBUG_AUDIO 1 | |
| 46 | |
| 47 /* A conflict within AIX 4.3.3 <sys/> headers and probably others as well. | |
| 48 * I guess nobody ever uses audio... Shame over AIX header files. */ | |
| 49 #include <sys/machine.h> | |
| 50 #undef BIG_ENDIAN | |
| 51 #include <sys/audio.h> | |
| 52 | |
| 53 /* The tag name used by paud audio */ | |
| 54 #define Paud_DRIVER_NAME "paud" | |
| 55 | |
| 56 /* Open the audio device for playback, and don't block if busy */ | |
| 57 /* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */ | |
| 58 #define OPEN_FLAGS O_WRONLY | |
| 59 | |
| 60 /* Audio driver functions */ | |
| 61 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec); | |
| 62 static void Paud_WaitAudio(_THIS); | |
| 63 static void Paud_PlayAudio(_THIS); | |
| 64 static Uint8 *Paud_GetAudioBuf(_THIS); | |
| 65 static void Paud_CloseAudio(_THIS); | |
| 66 | |
| 67 /* Audio driver bootstrap functions */ | |
| 68 | |
| 69 static int Audio_Available(void) | |
| 70 { | |
| 71 int fd; | |
| 72 int available; | |
| 73 | |
| 74 available = 0; | |
| 75 fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); | |
| 76 if ( fd >= 0 ) { | |
| 77 available = 1; | |
| 78 close(fd); | |
| 79 } | |
| 80 return(available); | |
| 81 } | |
| 82 | |
| 83 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
| 84 { | |
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85 SDL_free(device->hidden); |
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86 SDL_free(device); |
| 0 | 87 } |
| 88 | |
| 89 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
| 90 { | |
| 91 SDL_AudioDevice *this; | |
| 92 | |
| 93 /* Initialize all variables that we clean on shutdown */ | |
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94 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
| 0 | 95 if ( this ) { |
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96 SDL_memset(this, 0, (sizeof *this)); |
| 0 | 97 this->hidden = (struct SDL_PrivateAudioData *) |
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98 SDL_malloc((sizeof *this->hidden)); |
| 0 | 99 } |
| 100 if ( (this == NULL) || (this->hidden == NULL) ) { | |
| 101 SDL_OutOfMemory(); | |
| 102 if ( this ) { | |
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103 SDL_free(this); |
| 0 | 104 } |
| 105 return(0); | |
| 106 } | |
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107 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
| 0 | 108 audio_fd = -1; |
| 109 | |
| 110 /* Set the function pointers */ | |
| 111 this->OpenAudio = Paud_OpenAudio; | |
| 112 this->WaitAudio = Paud_WaitAudio; | |
| 113 this->PlayAudio = Paud_PlayAudio; | |
| 114 this->GetAudioBuf = Paud_GetAudioBuf; | |
| 115 this->CloseAudio = Paud_CloseAudio; | |
| 116 | |
| 117 this->free = Audio_DeleteDevice; | |
| 118 | |
| 119 return this; | |
| 120 } | |
| 121 | |
| 122 AudioBootStrap Paud_bootstrap = { | |
| 123 Paud_DRIVER_NAME, "AIX Paudio", | |
| 124 Audio_Available, Audio_CreateDevice | |
| 125 }; | |
| 126 | |
| 127 /* This function waits until it is possible to write a full sound buffer */ | |
| 128 static void Paud_WaitAudio(_THIS) | |
| 129 { | |
| 130 fd_set fdset; | |
| 131 | |
| 132 /* See if we need to use timed audio synchronization */ | |
| 133 if ( frame_ticks ) { | |
| 134 /* Use timer for general audio synchronization */ | |
| 135 Sint32 ticks; | |
| 136 | |
| 137 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; | |
| 138 if ( ticks > 0 ) { | |
| 139 SDL_Delay(ticks); | |
| 140 } | |
| 141 } else { | |
| 142 audio_buffer paud_bufinfo; | |
| 143 | |
| 144 /* Use select() for audio synchronization */ | |
| 145 struct timeval timeout; | |
| 146 FD_ZERO(&fdset); | |
| 147 FD_SET(audio_fd, &fdset); | |
| 148 | |
| 149 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
| 150 #ifdef DEBUG_AUDIO | |
| 151 fprintf(stderr, "Couldn't get audio buffer information\n"); | |
| 152 #endif | |
| 153 timeout.tv_sec = 10; | |
| 154 timeout.tv_usec = 0; | |
| 155 } else { | |
| 156 long ms_in_buf = paud_bufinfo.write_buf_time; | |
| 157 timeout.tv_sec = ms_in_buf/1000; | |
| 158 ms_in_buf = ms_in_buf - timeout.tv_sec*1000; | |
| 159 timeout.tv_usec = ms_in_buf*1000; | |
| 160 #ifdef DEBUG_AUDIO | |
| 161 fprintf( stderr, | |
| 162 "Waiting for write_buf_time=%ld,%ld\n", | |
| 163 timeout.tv_sec, | |
| 164 timeout.tv_usec ); | |
| 165 #endif | |
| 166 } | |
| 167 | |
| 168 #ifdef DEBUG_AUDIO | |
| 169 fprintf(stderr, "Waiting for audio to get ready\n"); | |
| 170 #endif | |
| 171 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) { | |
| 172 const char *message = "Audio timeout - buggy audio driver? (disabled)"; | |
| 173 /* | |
| 174 * In general we should never print to the screen, | |
| 175 * but in this case we have no other way of letting | |
| 176 * the user know what happened. | |
| 177 */ | |
| 178 fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message); | |
| 179 this->enabled = 0; | |
| 180 /* Don't try to close - may hang */ | |
| 181 audio_fd = -1; | |
| 182 #ifdef DEBUG_AUDIO | |
| 183 fprintf(stderr, "Done disabling audio\n"); | |
| 184 #endif | |
| 185 } | |
| 186 #ifdef DEBUG_AUDIO | |
| 187 fprintf(stderr, "Ready!\n"); | |
| 188 #endif | |
| 189 } | |
| 190 } | |
| 191 | |
| 192 static void Paud_PlayAudio(_THIS) | |
| 193 { | |
| 194 int written; | |
| 195 | |
| 196 /* Write the audio data, checking for EAGAIN on broken audio drivers */ | |
| 197 do { | |
| 198 written = write(audio_fd, mixbuf, mixlen); | |
| 199 if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) { | |
| 200 SDL_Delay(1); /* Let a little CPU time go by */ | |
| 201 } | |
| 202 } while ( (written < 0) && | |
| 203 ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) ); | |
| 204 | |
| 205 /* If timer synchronization is enabled, set the next write frame */ | |
| 206 if ( frame_ticks ) { | |
| 207 next_frame += frame_ticks; | |
| 208 } | |
| 209 | |
| 210 /* If we couldn't write, assume fatal error for now */ | |
| 211 if ( written < 0 ) { | |
| 212 this->enabled = 0; | |
| 213 } | |
| 214 #ifdef DEBUG_AUDIO | |
| 215 fprintf(stderr, "Wrote %d bytes of audio data\n", written); | |
| 216 #endif | |
| 217 } | |
| 218 | |
| 219 static Uint8 *Paud_GetAudioBuf(_THIS) | |
| 220 { | |
| 221 return mixbuf; | |
| 222 } | |
| 223 | |
| 224 static void Paud_CloseAudio(_THIS) | |
| 225 { | |
| 226 if ( mixbuf != NULL ) { | |
| 227 SDL_FreeAudioMem(mixbuf); | |
| 228 mixbuf = NULL; | |
| 229 } | |
| 230 if ( audio_fd >= 0 ) { | |
| 231 close(audio_fd); | |
| 232 audio_fd = -1; | |
| 233 } | |
| 234 } | |
| 235 | |
| 236 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec) | |
| 237 { | |
| 238 char audiodev[1024]; | |
| 239 int format; | |
| 240 int bytes_per_sample; | |
| 241 Uint16 test_format; | |
| 242 audio_init paud_init; | |
| 243 audio_buffer paud_bufinfo; | |
| 244 audio_status paud_status; | |
| 245 audio_control paud_control; | |
| 246 audio_change paud_change; | |
| 247 | |
| 248 /* Reset the timer synchronization flag */ | |
| 249 frame_ticks = 0.0; | |
| 250 | |
| 251 /* Open the audio device */ | |
| 252 audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); | |
| 253 if ( audio_fd < 0 ) { | |
| 254 SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); | |
| 255 return -1; | |
| 256 } | |
| 257 | |
| 258 /* | |
| 259 * We can't set the buffer size - just ask the device for the maximum | |
| 260 * that we can have. | |
| 261 */ | |
| 262 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
| 263 SDL_SetError("Couldn't get audio buffer information"); | |
| 264 return -1; | |
| 265 } | |
| 266 | |
| 267 mixbuf = NULL; | |
| 268 | |
| 269 if ( spec->channels > 1 ) | |
| 270 spec->channels = 2; | |
| 271 else | |
| 272 spec->channels = 1; | |
| 273 | |
| 274 /* | |
| 275 * Fields in the audio_init structure: | |
| 276 * | |
| 277 * Ignored by us: | |
| 278 * | |
| 279 * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only? | |
| 280 * paud.slot_number; * slot number of the adapter | |
| 281 * paud.device_id; * adapter identification number | |
| 282 * | |
| 283 * Input: | |
| 284 * | |
| 285 * paud.srate; * the sampling rate in Hz | |
| 286 * paud.bits_per_sample; * 8, 16, 32, ... | |
| 287 * paud.bsize; * block size for this rate | |
| 288 * paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX | |
| 289 * paud.channels; * 1=mono, 2=stereo | |
| 290 * paud.flags; * FIXED - fixed length data | |
| 291 * * LEFT_ALIGNED, RIGHT_ALIGNED (var len only) | |
| 292 * * TWOS_COMPLEMENT - 2's complement data | |
| 293 * * SIGNED - signed? comment seems wrong in sys/audio.h | |
| 294 * * BIG_ENDIAN | |
| 295 * paud.operation; * PLAY, RECORD | |
| 296 * | |
| 297 * Output: | |
| 298 * | |
| 299 * paud.flags; * PITCH - pitch is supported | |
| 300 * * INPUT - input is supported | |
| 301 * * OUTPUT - output is supported | |
| 302 * * MONITOR - monitor is supported | |
| 303 * * VOLUME - volume is supported | |
| 304 * * VOLUME_DELAY - volume delay is supported | |
| 305 * * BALANCE - balance is supported | |
| 306 * * BALANCE_DELAY - balance delay is supported | |
| 307 * * TREBLE - treble control is supported | |
| 308 * * BASS - bass control is supported | |
| 309 * * BESTFIT_PROVIDED - best fit returned | |
| 310 * * LOAD_CODE - DSP load needed | |
| 311 * paud.rc; * NO_PLAY - DSP code can't do play requests | |
| 312 * * NO_RECORD - DSP code can't do record requests | |
| 313 * * INVALID_REQUEST - request was invalid | |
| 314 * * CONFLICT - conflict with open's flags | |
| 315 * * OVERLOADED - out of DSP MIPS or memory | |
| 316 * paud.position_resolution; * smallest increment for position | |
| 317 */ | |
| 318 | |
| 319 paud_init.srate = spec->freq; | |
| 320 paud_init.mode = PCM; | |
| 321 paud_init.operation = PLAY; | |
| 322 paud_init.channels = spec->channels; | |
| 323 | |
| 324 /* Try for a closest match on audio format */ | |
| 325 format = 0; | |
| 326 for ( test_format = SDL_FirstAudioFormat(spec->format); | |
| 327 ! format && test_format; ) { | |
| 328 #ifdef DEBUG_AUDIO | |
| 329 fprintf(stderr, "Trying format 0x%4.4x\n", test_format); | |
| 330 #endif | |
| 331 switch ( test_format ) { | |
| 332 case AUDIO_U8: | |
| 333 bytes_per_sample = 1; | |
| 334 paud_init.bits_per_sample = 8; | |
| 335 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
| 336 format = 1; | |
| 337 break; | |
| 338 case AUDIO_S8: | |
| 339 bytes_per_sample = 1; | |
| 340 paud_init.bits_per_sample = 8; | |
| 341 paud_init.flags = SIGNED | | |
| 342 TWOS_COMPLEMENT | FIXED; | |
| 343 format = 1; | |
| 344 break; | |
| 345 case AUDIO_S16LSB: | |
| 346 bytes_per_sample = 2; | |
| 347 paud_init.bits_per_sample = 16; | |
| 348 paud_init.flags = SIGNED | | |
| 349 TWOS_COMPLEMENT | FIXED; | |
| 350 format = 1; | |
| 351 break; | |
| 352 case AUDIO_S16MSB: | |
| 353 bytes_per_sample = 2; | |
| 354 paud_init.bits_per_sample = 16; | |
| 355 paud_init.flags = BIG_ENDIAN | | |
| 356 SIGNED | | |
| 357 TWOS_COMPLEMENT | FIXED; | |
| 358 format = 1; | |
| 359 break; | |
| 360 case AUDIO_U16LSB: | |
| 361 bytes_per_sample = 2; | |
| 362 paud_init.bits_per_sample = 16; | |
| 363 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
| 364 format = 1; | |
| 365 break; | |
| 366 case AUDIO_U16MSB: | |
| 367 bytes_per_sample = 2; | |
| 368 paud_init.bits_per_sample = 16; | |
| 369 paud_init.flags = BIG_ENDIAN | | |
| 370 TWOS_COMPLEMENT | FIXED; | |
| 371 format = 1; | |
| 372 break; | |
| 373 default: | |
| 374 break; | |
| 375 } | |
| 376 if ( ! format ) { | |
| 377 test_format = SDL_NextAudioFormat(); | |
| 378 } | |
| 379 } | |
| 380 if ( format == 0 ) { | |
| 381 #ifdef DEBUG_AUDIO | |
| 382 fprintf(stderr, "Couldn't find any hardware audio formats\n"); | |
| 383 #endif | |
| 384 SDL_SetError("Couldn't find any hardware audio formats"); | |
| 385 return -1; | |
| 386 } | |
| 387 spec->format = test_format; | |
| 388 | |
| 389 /* | |
| 390 * We know the buffer size and the max number of subsequent writes | |
| 391 * that can be pending. If more than one can pend, allow the application | |
| 392 * to do something like double buffering between our write buffer and | |
| 393 * the device's own buffer that we are filling with write() anyway. | |
| 394 * | |
| 395 * We calculate spec->samples like this because SDL_CalculateAudioSpec() | |
| 396 * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2) | |
| 397 * into spec->size in return. | |
| 398 */ | |
| 399 if ( paud_bufinfo.request_buf_cap == 1 ) | |
| 400 { | |
| 401 spec->samples = paud_bufinfo.write_buf_cap | |
| 402 / bytes_per_sample | |
| 403 / spec->channels; | |
| 404 } | |
| 405 else | |
| 406 { | |
| 407 spec->samples = paud_bufinfo.write_buf_cap | |
| 408 / bytes_per_sample | |
| 409 / spec->channels | |
| 410 / 2; | |
| 411 } | |
| 412 paud_init.bsize = bytes_per_sample * spec->channels; | |
| 413 | |
| 414 SDL_CalculateAudioSpec(spec); | |
| 415 | |
| 416 /* | |
| 417 * The AIX paud device init can't modify the values of the audio_init | |
| 418 * structure that we pass to it. So we don't need any recalculation | |
| 419 * of this stuff and no reinit call as in linux dsp and dma code. | |
| 420 * | |
| 421 * /dev/paud supports all of the encoding formats, so we don't need | |
| 422 * to do anything like reopening the device, either. | |
| 423 */ | |
| 424 if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) { | |
| 425 switch ( paud_init.rc ) | |
| 426 { | |
| 427 case 1 : | |
| 428 SDL_SetError("Couldn't set audio format: DSP can't do play requests"); | |
| 429 return -1; | |
| 430 break; | |
| 431 case 2 : | |
| 432 SDL_SetError("Couldn't set audio format: DSP can't do record requests"); | |
| 433 return -1; | |
| 434 break; | |
| 435 case 4 : | |
| 436 SDL_SetError("Couldn't set audio format: request was invalid"); | |
| 437 return -1; | |
| 438 break; | |
| 439 case 5 : | |
| 440 SDL_SetError("Couldn't set audio format: conflict with open's flags"); | |
| 441 return -1; | |
| 442 break; | |
| 443 case 6 : | |
| 444 SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory"); | |
| 445 return -1; | |
| 446 break; | |
| 447 default : | |
| 448 SDL_SetError("Couldn't set audio format: not documented in sys/audio.h"); | |
| 449 return -1; | |
| 450 break; | |
| 451 } | |
| 452 } | |
| 453 | |
| 454 /* Allocate mixing buffer */ | |
| 455 mixlen = spec->size; | |
| 456 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); | |
| 457 if ( mixbuf == NULL ) { | |
| 458 return -1; | |
| 459 } | |
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460 SDL_memset(mixbuf, spec->silence, spec->size); |
| 0 | 461 |
| 462 /* | |
| 463 * Set some paramters: full volume, first speaker that we can find. | |
| 464 * Ignore the other settings for now. | |
| 465 */ | |
| 466 paud_change.input = AUDIO_IGNORE; /* the new input source */ | |
| 467 paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */ | |
| 468 paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */ | |
| 469 paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */ | |
| 470 paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */ | |
| 471 paud_change.balance = 0x3fffffff; /* the new balance */ | |
| 472 paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */ | |
| 473 paud_change.treble = AUDIO_IGNORE; /* the new treble state */ | |
| 474 paud_change.bass = AUDIO_IGNORE; /* the new bass state */ | |
| 475 paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */ | |
| 476 | |
| 477 paud_control.ioctl_request = AUDIO_CHANGE; | |
| 478 paud_control.request_info = (char*)&paud_change; | |
| 479 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
| 480 #ifdef DEBUG_AUDIO | |
| 481 fprintf(stderr, "Can't change audio display settings\n" ); | |
| 482 #endif | |
| 483 } | |
| 484 | |
| 485 /* | |
| 486 * Tell the device to expect data. Actual start will wait for | |
| 487 * the first write() call. | |
| 488 */ | |
| 489 paud_control.ioctl_request = AUDIO_START; | |
| 490 paud_control.position = 0; | |
| 491 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
| 492 #ifdef DEBUG_AUDIO | |
| 493 fprintf(stderr, "Can't start audio play\n" ); | |
| 494 #endif | |
| 495 SDL_SetError("Can't start audio play"); | |
| 496 return -1; | |
| 497 } | |
| 498 | |
| 499 /* Check to see if we need to use select() workaround */ | |
| 500 { char *workaround; | |
|
1336
3692456e7b0f
Use SDL_ prefixed versions of C library functions.
Sam Lantinga <slouken@libsdl.org>
parents:
1312
diff
changeset
|
501 workaround = SDL_getenv("SDL_DSP_NOSELECT"); |
| 0 | 502 if ( workaround ) { |
| 503 frame_ticks = (float)(spec->samples*1000)/spec->freq; | |
| 504 next_frame = SDL_GetTicks()+frame_ticks; | |
| 505 } | |
| 506 } | |
| 507 | |
| 508 /* Get the parent process id (we're the parent of the audio thread) */ | |
| 509 parent = getpid(); | |
| 510 | |
| 511 /* We're ready to rock and roll. :-) */ | |
| 512 return 0; | |
| 513 } | |
| 514 |