Mercurial > sdl-ios-xcode
annotate src/audio/sun/SDL_sunaudio.c @ 4001:6831b8723a85 SDL-1.2
Don't initialize the audio buffer passed to the application's audio callback,
since they are expected to entirely fill it with data or silence.
For legacy apps that might expect the buffer to already have silence and thus
may not fill the buffer in the callback, there's an environment variable to
expose the old behaviour.
Fixes Bugzilla #416.
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Thu, 05 Jul 2007 02:24:36 +0000 |
| parents | af30090c0330 |
| children | 782fd950bd46 c121d94672cb a1b03ba2fcd0 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 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 Sam Lantinga | |
|
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20 slouken@libsdl.org |
| 0 | 21 */ |
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22 #include "SDL_config.h" |
| 0 | 23 |
| 24 /* Allow access to a raw mixing buffer */ | |
| 25 | |
| 26 #include <fcntl.h> | |
| 27 #include <errno.h> | |
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28 #ifdef __NETBSD__ |
| 0 | 29 #include <sys/ioctl.h> |
| 30 #include <sys/audioio.h> | |
| 31 #endif | |
| 32 #ifdef __SVR4 | |
| 33 #include <sys/audioio.h> | |
| 34 #else | |
| 35 #include <sys/time.h> | |
| 36 #include <sys/types.h> | |
| 37 #endif | |
| 38 #include <unistd.h> | |
| 39 | |
|
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40 #include "SDL_timer.h" |
| 0 | 41 #include "SDL_audio.h" |
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42 #include "../SDL_audiomem.h" |
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43 #include "../SDL_audio_c.h" |
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44 #include "../SDL_audiodev_c.h" |
| 0 | 45 #include "SDL_sunaudio.h" |
| 46 | |
| 47 /* Open the audio device for playback, and don't block if busy */ | |
| 48 #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) | |
| 49 | |
| 50 /* Audio driver functions */ | |
| 51 static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec); | |
| 52 static void DSP_WaitAudio(_THIS); | |
| 53 static void DSP_PlayAudio(_THIS); | |
| 54 static Uint8 *DSP_GetAudioBuf(_THIS); | |
| 55 static void DSP_CloseAudio(_THIS); | |
| 56 | |
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57 static Uint8 snd2au(int sample); |
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58 |
| 0 | 59 /* Audio driver bootstrap functions */ |
| 60 | |
| 61 static int Audio_Available(void) | |
| 62 { | |
| 63 int fd; | |
| 64 int available; | |
| 65 | |
| 66 available = 0; | |
| 67 fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 1); | |
| 68 if ( fd >= 0 ) { | |
| 69 available = 1; | |
| 70 close(fd); | |
| 71 } | |
| 72 return(available); | |
| 73 } | |
| 74 | |
| 75 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
| 76 { | |
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77 SDL_free(device->hidden); |
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78 SDL_free(device); |
| 0 | 79 } |
| 80 | |
| 81 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
| 82 { | |
| 83 SDL_AudioDevice *this; | |
| 84 | |
| 85 /* Initialize all variables that we clean on shutdown */ | |
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86 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
| 0 | 87 if ( this ) { |
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88 SDL_memset(this, 0, (sizeof *this)); |
| 0 | 89 this->hidden = (struct SDL_PrivateAudioData *) |
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90 SDL_malloc((sizeof *this->hidden)); |
| 0 | 91 } |
| 92 if ( (this == NULL) || (this->hidden == NULL) ) { | |
| 93 SDL_OutOfMemory(); | |
| 94 if ( this ) { | |
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95 SDL_free(this); |
| 0 | 96 } |
| 97 return(0); | |
| 98 } | |
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99 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
| 0 | 100 audio_fd = -1; |
| 101 | |
| 102 /* Set the function pointers */ | |
| 103 this->OpenAudio = DSP_OpenAudio; | |
| 104 this->WaitAudio = DSP_WaitAudio; | |
| 105 this->PlayAudio = DSP_PlayAudio; | |
| 106 this->GetAudioBuf = DSP_GetAudioBuf; | |
| 107 this->CloseAudio = DSP_CloseAudio; | |
| 108 | |
| 109 this->free = Audio_DeleteDevice; | |
| 110 | |
| 111 return this; | |
| 112 } | |
| 113 | |
|
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114 AudioBootStrap SUNAUDIO_bootstrap = { |
| 0 | 115 "audio", "UNIX /dev/audio interface", |
| 116 Audio_Available, Audio_CreateDevice | |
| 117 }; | |
| 118 | |
| 119 #ifdef DEBUG_AUDIO | |
| 120 void CheckUnderflow(_THIS) | |
| 121 { | |
| 122 #ifdef AUDIO_GETINFO | |
| 123 audio_info_t info; | |
| 124 int left; | |
| 125 | |
| 126 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
| 127 left = (written - info.play.samples); | |
| 128 if ( written && (left == 0) ) { | |
| 129 fprintf(stderr, "audio underflow!\n"); | |
| 130 } | |
| 131 #endif | |
| 132 } | |
| 133 #endif | |
| 134 | |
| 135 void DSP_WaitAudio(_THIS) | |
| 136 { | |
| 137 #ifdef AUDIO_GETINFO | |
| 138 #define SLEEP_FUDGE 10 /* 10 ms scheduling fudge factor */ | |
| 139 audio_info_t info; | |
| 140 Sint32 left; | |
| 141 | |
| 142 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
| 143 left = (written - info.play.samples); | |
| 144 if ( left > fragsize ) { | |
| 145 Sint32 sleepy; | |
| 146 | |
| 147 sleepy = ((left - fragsize)/frequency); | |
| 148 sleepy -= SLEEP_FUDGE; | |
| 149 if ( sleepy > 0 ) { | |
| 150 SDL_Delay(sleepy); | |
| 151 } | |
| 152 } | |
| 153 #else | |
| 154 fd_set fdset; | |
| 155 | |
| 156 FD_ZERO(&fdset); | |
| 157 FD_SET(audio_fd, &fdset); | |
| 158 select(audio_fd+1, NULL, &fdset, NULL, NULL); | |
| 159 #endif | |
| 160 } | |
| 161 | |
| 162 void DSP_PlayAudio(_THIS) | |
| 163 { | |
| 164 /* Write the audio data */ | |
| 165 if ( ulaw_only ) { | |
| 166 /* Assuming that this->spec.freq >= 8000 Hz */ | |
| 167 int accum, incr, pos; | |
| 168 Uint8 *aubuf; | |
| 169 | |
| 170 accum = 0; | |
| 171 incr = this->spec.freq/8; | |
| 172 aubuf = ulaw_buf; | |
| 173 switch (audio_fmt & 0xFF) { | |
| 174 case 8: { | |
| 175 Uint8 *sndbuf; | |
| 176 | |
| 177 sndbuf = mixbuf; | |
| 178 for ( pos=0; pos < fragsize; ++pos ) { | |
| 179 *aubuf = snd2au((0x80-*sndbuf)*64); | |
| 180 accum += incr; | |
| 181 while ( accum > 0 ) { | |
| 182 accum -= 1000; | |
| 183 sndbuf += 1; | |
| 184 } | |
| 185 aubuf += 1; | |
| 186 } | |
| 187 } | |
| 188 break; | |
| 189 case 16: { | |
| 190 Sint16 *sndbuf; | |
| 191 | |
| 192 sndbuf = (Sint16 *)mixbuf; | |
| 193 for ( pos=0; pos < fragsize; ++pos ) { | |
| 194 *aubuf = snd2au(*sndbuf/4); | |
| 195 accum += incr; | |
| 196 while ( accum > 0 ) { | |
| 197 accum -= 1000; | |
| 198 sndbuf += 1; | |
| 199 } | |
| 200 aubuf += 1; | |
| 201 } | |
| 202 } | |
| 203 break; | |
| 204 } | |
| 205 #ifdef DEBUG_AUDIO | |
| 206 CheckUnderflow(this); | |
| 207 #endif | |
| 208 if ( write(audio_fd, ulaw_buf, fragsize) < 0 ) { | |
| 209 /* Assume fatal error, for now */ | |
| 210 this->enabled = 0; | |
| 211 } | |
| 212 written += fragsize; | |
| 213 } else { | |
| 214 #ifdef DEBUG_AUDIO | |
| 215 CheckUnderflow(this); | |
| 216 #endif | |
| 217 if ( write(audio_fd, mixbuf, this->spec.size) < 0 ) { | |
| 218 /* Assume fatal error, for now */ | |
| 219 this->enabled = 0; | |
| 220 } | |
| 221 written += fragsize; | |
| 222 } | |
| 223 } | |
| 224 | |
| 225 Uint8 *DSP_GetAudioBuf(_THIS) | |
| 226 { | |
| 227 return(mixbuf); | |
| 228 } | |
| 229 | |
| 230 void DSP_CloseAudio(_THIS) | |
| 231 { | |
| 232 if ( mixbuf != NULL ) { | |
| 233 SDL_FreeAudioMem(mixbuf); | |
| 234 mixbuf = NULL; | |
| 235 } | |
| 236 if ( ulaw_buf != NULL ) { | |
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237 SDL_free(ulaw_buf); |
| 0 | 238 ulaw_buf = NULL; |
| 239 } | |
| 240 close(audio_fd); | |
| 241 } | |
| 242 | |
| 243 int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec) | |
| 244 { | |
| 245 char audiodev[1024]; | |
| 246 #ifdef AUDIO_SETINFO | |
| 247 int enc; | |
| 248 #endif | |
| 249 int desired_freq = spec->freq; | |
| 250 | |
| 251 /* Initialize our freeable variables, in case we fail*/ | |
| 252 audio_fd = -1; | |
| 253 mixbuf = NULL; | |
| 254 ulaw_buf = NULL; | |
| 255 | |
| 256 /* Determine the audio parameters from the AudioSpec */ | |
| 257 switch ( spec->format & 0xFF ) { | |
| 258 | |
| 259 case 8: { /* Unsigned 8 bit audio data */ | |
| 260 spec->format = AUDIO_U8; | |
| 261 #ifdef AUDIO_SETINFO | |
| 262 enc = AUDIO_ENCODING_LINEAR8; | |
| 263 #endif | |
| 264 } | |
| 265 break; | |
| 266 | |
| 267 case 16: { /* Signed 16 bit audio data */ | |
| 268 spec->format = AUDIO_S16SYS; | |
| 269 #ifdef AUDIO_SETINFO | |
| 270 enc = AUDIO_ENCODING_LINEAR; | |
| 271 #endif | |
| 272 } | |
| 273 break; | |
| 274 | |
| 275 default: { | |
| 276 SDL_SetError("Unsupported audio format"); | |
| 277 return(-1); | |
| 278 } | |
| 279 } | |
| 280 audio_fmt = spec->format; | |
| 281 | |
| 282 /* Open the audio device */ | |
| 283 audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1); | |
| 284 if ( audio_fd < 0 ) { | |
| 285 SDL_SetError("Couldn't open %s: %s", audiodev, | |
| 286 strerror(errno)); | |
| 287 return(-1); | |
| 288 } | |
| 289 | |
| 290 ulaw_only = 0; /* modern Suns do support linear audio */ | |
| 291 #ifdef AUDIO_SETINFO | |
| 292 for(;;) { | |
| 293 audio_info_t info; | |
| 294 AUDIO_INITINFO(&info); /* init all fields to "no change" */ | |
| 295 | |
| 296 /* Try to set the requested settings */ | |
| 297 info.play.sample_rate = spec->freq; | |
| 298 info.play.channels = spec->channels; | |
| 299 info.play.precision = (enc == AUDIO_ENCODING_ULAW) | |
| 300 ? 8 : spec->format & 0xff; | |
| 301 info.play.encoding = enc; | |
| 302 if( ioctl(audio_fd, AUDIO_SETINFO, &info) == 0 ) { | |
| 303 | |
| 304 /* Check to be sure we got what we wanted */ | |
| 305 if(ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) { | |
| 306 SDL_SetError("Error getting audio parameters: %s", | |
| 307 strerror(errno)); | |
| 308 return -1; | |
| 309 } | |
| 310 if(info.play.encoding == enc | |
| 311 && info.play.precision == (spec->format & 0xff) | |
| 312 && info.play.channels == spec->channels) { | |
| 313 /* Yow! All seems to be well! */ | |
| 314 spec->freq = info.play.sample_rate; | |
| 315 break; | |
| 316 } | |
| 317 } | |
| 318 | |
| 319 switch(enc) { | |
| 320 case AUDIO_ENCODING_LINEAR8: | |
| 321 /* unsigned 8bit apparently not supported here */ | |
| 322 enc = AUDIO_ENCODING_LINEAR; | |
| 323 spec->format = AUDIO_S16SYS; | |
| 324 break; /* try again */ | |
| 325 | |
| 326 case AUDIO_ENCODING_LINEAR: | |
| 327 /* linear 16bit didn't work either, resort to µ-law */ | |
| 328 enc = AUDIO_ENCODING_ULAW; | |
| 329 spec->channels = 1; | |
| 330 spec->freq = 8000; | |
| 331 spec->format = AUDIO_U8; | |
| 332 ulaw_only = 1; | |
| 333 break; | |
| 334 | |
| 335 default: | |
| 336 /* oh well... */ | |
| 337 SDL_SetError("Error setting audio parameters: %s", | |
| 338 strerror(errno)); | |
| 339 return -1; | |
| 340 } | |
| 341 } | |
| 342 #endif /* AUDIO_SETINFO */ | |
| 343 written = 0; | |
| 344 | |
| 345 /* We can actually convert on-the-fly to U-Law */ | |
| 346 if ( ulaw_only ) { | |
| 347 spec->freq = desired_freq; | |
| 348 fragsize = (spec->samples*1000)/(spec->freq/8); | |
| 349 frequency = 8; | |
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350 ulaw_buf = (Uint8 *)SDL_malloc(fragsize); |
| 0 | 351 if ( ulaw_buf == NULL ) { |
| 352 SDL_OutOfMemory(); | |
| 353 return(-1); | |
| 354 } | |
| 355 spec->channels = 1; | |
| 356 } else { | |
| 357 fragsize = spec->samples; | |
| 358 frequency = spec->freq/1000; | |
| 359 } | |
| 360 #ifdef DEBUG_AUDIO | |
| 361 fprintf(stderr, "Audio device %s U-Law only\n", | |
| 362 ulaw_only ? "is" : "is not"); | |
| 363 fprintf(stderr, "format=0x%x chan=%d freq=%d\n", | |
| 364 spec->format, spec->channels, spec->freq); | |
| 365 #endif | |
| 366 | |
| 367 /* Update the fragment size as size in bytes */ | |
| 368 SDL_CalculateAudioSpec(spec); | |
| 369 | |
| 370 /* Allocate mixing buffer */ | |
| 371 mixbuf = (Uint8 *)SDL_AllocAudioMem(spec->size); | |
| 372 if ( mixbuf == NULL ) { | |
| 373 SDL_OutOfMemory(); | |
| 374 return(-1); | |
| 375 } | |
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376 SDL_memset(mixbuf, spec->silence, spec->size); |
| 0 | 377 |
| 378 /* We're ready to rock and roll. :-) */ | |
| 379 return(0); | |
| 380 } | |
| 381 | |
| 382 /************************************************************************/ | |
| 383 /* This function (snd2au()) copyrighted: */ | |
| 384 /************************************************************************/ | |
| 385 /* Copyright 1989 by Rich Gopstein and Harris Corporation */ | |
| 386 /* */ | |
| 387 /* Permission to use, copy, modify, and distribute this software */ | |
| 388 /* and its documentation for any purpose and without fee is */ | |
| 389 /* hereby granted, provided that the above copyright notice */ | |
| 390 /* appears in all copies and that both that copyright notice and */ | |
| 391 /* this permission notice appear in supporting documentation, and */ | |
| 392 /* that the name of Rich Gopstein and Harris Corporation not be */ | |
| 393 /* used in advertising or publicity pertaining to distribution */ | |
| 394 /* of the software without specific, written prior permission. */ | |
| 395 /* Rich Gopstein and Harris Corporation make no representations */ | |
| 396 /* about the suitability of this software for any purpose. It */ | |
| 397 /* provided "as is" without express or implied warranty. */ | |
| 398 /************************************************************************/ | |
| 399 | |
| 400 static Uint8 snd2au(int sample) | |
| 401 { | |
| 402 | |
| 403 int mask; | |
| 404 | |
| 405 if (sample < 0) { | |
| 406 sample = -sample; | |
| 407 mask = 0x7f; | |
| 408 } else { | |
| 409 mask = 0xff; | |
| 410 } | |
| 411 | |
| 412 if (sample < 32) { | |
| 413 sample = 0xF0 | (15 - sample / 2); | |
| 414 } else if (sample < 96) { | |
| 415 sample = 0xE0 | (15 - (sample - 32) / 4); | |
| 416 } else if (sample < 224) { | |
| 417 sample = 0xD0 | (15 - (sample - 96) / 8); | |
| 418 } else if (sample < 480) { | |
| 419 sample = 0xC0 | (15 - (sample - 224) / 16); | |
| 420 } else if (sample < 992) { | |
| 421 sample = 0xB0 | (15 - (sample - 480) / 32); | |
| 422 } else if (sample < 2016) { | |
| 423 sample = 0xA0 | (15 - (sample - 992) / 64); | |
| 424 } else if (sample < 4064) { | |
| 425 sample = 0x90 | (15 - (sample - 2016) / 128); | |
| 426 } else if (sample < 8160) { | |
| 427 sample = 0x80 | (15 - (sample - 4064) / 256); | |
| 428 } else { | |
| 429 sample = 0x80; | |
| 430 } | |
| 431 return (mask & sample); | |
| 432 } |