Mercurial > sdl-ios-xcode
annotate src/audio/sun/SDL_sunaudio.c @ 4121:917ca146d901 SDL-1.2
J. Snell fixed bug #482
Fixed arguments used with sdl-config in sdl.m4
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Sat, 29 Dec 2007 20:08:51 +0000 |
| parents | af30090c0330 |
| children | 782fd950bd46 c121d94672cb a1b03ba2fcd0 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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Updated copyright information and removed rcs id lines (problematic in branch merges)
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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 |
|
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
Sam Lantinga <slouken@libsdl.org>
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16 License along with this library; if not, write to the Free Software |
|
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
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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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More header massaging... works great on Windows. ;-)
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40 #include "SDL_timer.h" |
| 0 | 41 #include "SDL_audio.h" |
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New configure-based build system. Still work in progress, but much improved
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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 } |