Mercurial > sdl-ios-xcode
annotate src/audio/sun/SDL_sunaudio.c @ 1105:1c9988e47824
Disable Centscreen screensaver
| author | Patrice Mandin <patmandin@gmail.com> |
|---|---|
| date | Fri, 29 Jul 2005 10:59:02 +0000 |
| parents | b8d311d90021 |
| children | a9542c38dcdb |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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b8d311d90021
Updated copyright information for 2004 (Happy New Year!)
Sam Lantinga <slouken@libsdl.org>
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|
3 Copyright (C) 1997-2004 Sam Lantinga |
| 0 | 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 | |
|
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Updated the source with the correct e-mail address
Sam Lantinga <slouken@libsdl.org>
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20 slouken@libsdl.org |
| 0 | 21 */ |
| 22 | |
| 23 #ifdef SAVE_RCSID | |
| 24 static char rcsid = | |
| 25 "@(#) $Id$"; | |
| 26 #endif | |
| 27 | |
| 28 /* Allow access to a raw mixing buffer */ | |
| 29 | |
| 30 #include <stdlib.h> | |
| 31 #include <stdio.h> | |
| 32 #include <fcntl.h> | |
| 33 #include <errno.h> | |
| 34 #include <string.h> | |
| 35 #ifdef __NetBSD__ | |
| 36 #include <sys/ioctl.h> | |
| 37 #include <sys/audioio.h> | |
| 38 #endif | |
| 39 #ifdef __SVR4 | |
| 40 #include <sys/audioio.h> | |
| 41 #else | |
| 42 #include <sys/time.h> | |
| 43 #include <sys/types.h> | |
| 44 #endif | |
| 45 #include <unistd.h> | |
| 46 | |
| 47 #include "SDL_endian.h" | |
| 48 #include "SDL_audio.h" | |
| 49 #include "SDL_audiomem.h" | |
| 50 #include "SDL_audiodev_c.h" | |
| 51 #include "SDL_sunaudio.h" | |
| 52 #include "SDL_audio_c.h" | |
| 53 #include "SDL_timer.h" | |
| 54 | |
| 55 /* Open the audio device for playback, and don't block if busy */ | |
| 56 #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) | |
| 57 | |
| 58 /* Audio driver functions */ | |
| 59 static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec); | |
| 60 static void DSP_WaitAudio(_THIS); | |
| 61 static void DSP_PlayAudio(_THIS); | |
| 62 static Uint8 *DSP_GetAudioBuf(_THIS); | |
| 63 static void DSP_CloseAudio(_THIS); | |
| 64 | |
| 65 /* Audio driver bootstrap functions */ | |
| 66 | |
| 67 static int Audio_Available(void) | |
| 68 { | |
| 69 int fd; | |
| 70 int available; | |
| 71 | |
| 72 available = 0; | |
| 73 fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 1); | |
| 74 if ( fd >= 0 ) { | |
| 75 available = 1; | |
| 76 close(fd); | |
| 77 } | |
| 78 return(available); | |
| 79 } | |
| 80 | |
| 81 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
| 82 { | |
| 83 free(device->hidden); | |
| 84 free(device); | |
| 85 } | |
| 86 | |
| 87 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
| 88 { | |
| 89 SDL_AudioDevice *this; | |
| 90 | |
| 91 /* Initialize all variables that we clean on shutdown */ | |
| 92 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); | |
| 93 if ( this ) { | |
| 94 memset(this, 0, (sizeof *this)); | |
| 95 this->hidden = (struct SDL_PrivateAudioData *) | |
| 96 malloc((sizeof *this->hidden)); | |
| 97 } | |
| 98 if ( (this == NULL) || (this->hidden == NULL) ) { | |
| 99 SDL_OutOfMemory(); | |
| 100 if ( this ) { | |
| 101 free(this); | |
| 102 } | |
| 103 return(0); | |
| 104 } | |
| 105 memset(this->hidden, 0, (sizeof *this->hidden)); | |
| 106 audio_fd = -1; | |
| 107 | |
| 108 /* Set the function pointers */ | |
| 109 this->OpenAudio = DSP_OpenAudio; | |
| 110 this->WaitAudio = DSP_WaitAudio; | |
| 111 this->PlayAudio = DSP_PlayAudio; | |
| 112 this->GetAudioBuf = DSP_GetAudioBuf; | |
| 113 this->CloseAudio = DSP_CloseAudio; | |
| 114 | |
| 115 this->free = Audio_DeleteDevice; | |
| 116 | |
| 117 return this; | |
| 118 } | |
| 119 | |
|
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120 AudioBootStrap SUNAUDIO_bootstrap = { |
| 0 | 121 "audio", "UNIX /dev/audio interface", |
| 122 Audio_Available, Audio_CreateDevice | |
| 123 }; | |
| 124 | |
| 125 #ifdef DEBUG_AUDIO | |
| 126 void CheckUnderflow(_THIS) | |
| 127 { | |
| 128 #ifdef AUDIO_GETINFO | |
| 129 audio_info_t info; | |
| 130 int left; | |
| 131 | |
| 132 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
| 133 left = (written - info.play.samples); | |
| 134 if ( written && (left == 0) ) { | |
| 135 fprintf(stderr, "audio underflow!\n"); | |
| 136 } | |
| 137 #endif | |
| 138 } | |
| 139 #endif | |
| 140 | |
| 141 void DSP_WaitAudio(_THIS) | |
| 142 { | |
| 143 #ifdef AUDIO_GETINFO | |
| 144 #define SLEEP_FUDGE 10 /* 10 ms scheduling fudge factor */ | |
| 145 audio_info_t info; | |
| 146 Sint32 left; | |
| 147 | |
| 148 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
| 149 left = (written - info.play.samples); | |
| 150 if ( left > fragsize ) { | |
| 151 Sint32 sleepy; | |
| 152 | |
| 153 sleepy = ((left - fragsize)/frequency); | |
| 154 sleepy -= SLEEP_FUDGE; | |
| 155 if ( sleepy > 0 ) { | |
| 156 SDL_Delay(sleepy); | |
| 157 } | |
| 158 } | |
| 159 #else | |
| 160 fd_set fdset; | |
| 161 | |
| 162 FD_ZERO(&fdset); | |
| 163 FD_SET(audio_fd, &fdset); | |
| 164 select(audio_fd+1, NULL, &fdset, NULL, NULL); | |
| 165 #endif | |
| 166 } | |
| 167 | |
| 168 void DSP_PlayAudio(_THIS) | |
| 169 { | |
| 170 static Uint8 snd2au(int sample); | |
| 171 /* Write the audio data */ | |
| 172 if ( ulaw_only ) { | |
| 173 /* Assuming that this->spec.freq >= 8000 Hz */ | |
| 174 int accum, incr, pos; | |
| 175 Uint8 *aubuf; | |
| 176 | |
| 177 accum = 0; | |
| 178 incr = this->spec.freq/8; | |
| 179 aubuf = ulaw_buf; | |
| 180 switch (audio_fmt & 0xFF) { | |
| 181 case 8: { | |
| 182 Uint8 *sndbuf; | |
| 183 | |
| 184 sndbuf = mixbuf; | |
| 185 for ( pos=0; pos < fragsize; ++pos ) { | |
| 186 *aubuf = snd2au((0x80-*sndbuf)*64); | |
| 187 accum += incr; | |
| 188 while ( accum > 0 ) { | |
| 189 accum -= 1000; | |
| 190 sndbuf += 1; | |
| 191 } | |
| 192 aubuf += 1; | |
| 193 } | |
| 194 } | |
| 195 break; | |
| 196 case 16: { | |
| 197 Sint16 *sndbuf; | |
| 198 | |
| 199 sndbuf = (Sint16 *)mixbuf; | |
| 200 for ( pos=0; pos < fragsize; ++pos ) { | |
| 201 *aubuf = snd2au(*sndbuf/4); | |
| 202 accum += incr; | |
| 203 while ( accum > 0 ) { | |
| 204 accum -= 1000; | |
| 205 sndbuf += 1; | |
| 206 } | |
| 207 aubuf += 1; | |
| 208 } | |
| 209 } | |
| 210 break; | |
| 211 } | |
| 212 #ifdef DEBUG_AUDIO | |
| 213 CheckUnderflow(this); | |
| 214 #endif | |
| 215 if ( write(audio_fd, ulaw_buf, fragsize) < 0 ) { | |
| 216 /* Assume fatal error, for now */ | |
| 217 this->enabled = 0; | |
| 218 } | |
| 219 written += fragsize; | |
| 220 } else { | |
| 221 #ifdef DEBUG_AUDIO | |
| 222 CheckUnderflow(this); | |
| 223 #endif | |
| 224 if ( write(audio_fd, mixbuf, this->spec.size) < 0 ) { | |
| 225 /* Assume fatal error, for now */ | |
| 226 this->enabled = 0; | |
| 227 } | |
| 228 written += fragsize; | |
| 229 } | |
| 230 } | |
| 231 | |
| 232 Uint8 *DSP_GetAudioBuf(_THIS) | |
| 233 { | |
| 234 return(mixbuf); | |
| 235 } | |
| 236 | |
| 237 void DSP_CloseAudio(_THIS) | |
| 238 { | |
| 239 if ( mixbuf != NULL ) { | |
| 240 SDL_FreeAudioMem(mixbuf); | |
| 241 mixbuf = NULL; | |
| 242 } | |
| 243 if ( ulaw_buf != NULL ) { | |
| 244 free(ulaw_buf); | |
| 245 ulaw_buf = NULL; | |
| 246 } | |
| 247 close(audio_fd); | |
| 248 } | |
| 249 | |
| 250 int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec) | |
| 251 { | |
| 252 char audiodev[1024]; | |
| 253 #ifdef AUDIO_SETINFO | |
| 254 int enc; | |
| 255 #endif | |
| 256 int desired_freq = spec->freq; | |
| 257 | |
| 258 /* Initialize our freeable variables, in case we fail*/ | |
| 259 audio_fd = -1; | |
| 260 mixbuf = NULL; | |
| 261 ulaw_buf = NULL; | |
| 262 | |
| 263 /* Determine the audio parameters from the AudioSpec */ | |
| 264 switch ( spec->format & 0xFF ) { | |
| 265 | |
| 266 case 8: { /* Unsigned 8 bit audio data */ | |
| 267 spec->format = AUDIO_U8; | |
| 268 #ifdef AUDIO_SETINFO | |
| 269 enc = AUDIO_ENCODING_LINEAR8; | |
| 270 #endif | |
| 271 } | |
| 272 break; | |
| 273 | |
| 274 case 16: { /* Signed 16 bit audio data */ | |
| 275 spec->format = AUDIO_S16SYS; | |
| 276 #ifdef AUDIO_SETINFO | |
| 277 enc = AUDIO_ENCODING_LINEAR; | |
| 278 #endif | |
| 279 } | |
| 280 break; | |
| 281 | |
| 282 default: { | |
| 283 SDL_SetError("Unsupported audio format"); | |
| 284 return(-1); | |
| 285 } | |
| 286 } | |
| 287 audio_fmt = spec->format; | |
| 288 | |
| 289 /* Open the audio device */ | |
| 290 audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1); | |
| 291 if ( audio_fd < 0 ) { | |
| 292 SDL_SetError("Couldn't open %s: %s", audiodev, | |
| 293 strerror(errno)); | |
| 294 return(-1); | |
| 295 } | |
| 296 | |
| 297 ulaw_only = 0; /* modern Suns do support linear audio */ | |
| 298 #ifdef AUDIO_SETINFO | |
| 299 for(;;) { | |
| 300 audio_info_t info; | |
| 301 AUDIO_INITINFO(&info); /* init all fields to "no change" */ | |
| 302 | |
| 303 /* Try to set the requested settings */ | |
| 304 info.play.sample_rate = spec->freq; | |
| 305 info.play.channels = spec->channels; | |
| 306 info.play.precision = (enc == AUDIO_ENCODING_ULAW) | |
| 307 ? 8 : spec->format & 0xff; | |
| 308 info.play.encoding = enc; | |
| 309 if( ioctl(audio_fd, AUDIO_SETINFO, &info) == 0 ) { | |
| 310 | |
| 311 /* Check to be sure we got what we wanted */ | |
| 312 if(ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) { | |
| 313 SDL_SetError("Error getting audio parameters: %s", | |
| 314 strerror(errno)); | |
| 315 return -1; | |
| 316 } | |
| 317 if(info.play.encoding == enc | |
| 318 && info.play.precision == (spec->format & 0xff) | |
| 319 && info.play.channels == spec->channels) { | |
| 320 /* Yow! All seems to be well! */ | |
| 321 spec->freq = info.play.sample_rate; | |
| 322 break; | |
| 323 } | |
| 324 } | |
| 325 | |
| 326 switch(enc) { | |
| 327 case AUDIO_ENCODING_LINEAR8: | |
| 328 /* unsigned 8bit apparently not supported here */ | |
| 329 enc = AUDIO_ENCODING_LINEAR; | |
| 330 spec->format = AUDIO_S16SYS; | |
| 331 break; /* try again */ | |
| 332 | |
| 333 case AUDIO_ENCODING_LINEAR: | |
| 334 /* linear 16bit didn't work either, resort to µ-law */ | |
| 335 enc = AUDIO_ENCODING_ULAW; | |
| 336 spec->channels = 1; | |
| 337 spec->freq = 8000; | |
| 338 spec->format = AUDIO_U8; | |
| 339 ulaw_only = 1; | |
| 340 break; | |
| 341 | |
| 342 default: | |
| 343 /* oh well... */ | |
| 344 SDL_SetError("Error setting audio parameters: %s", | |
| 345 strerror(errno)); | |
| 346 return -1; | |
| 347 } | |
| 348 } | |
| 349 #endif /* AUDIO_SETINFO */ | |
| 350 written = 0; | |
| 351 | |
| 352 /* We can actually convert on-the-fly to U-Law */ | |
| 353 if ( ulaw_only ) { | |
| 354 spec->freq = desired_freq; | |
| 355 fragsize = (spec->samples*1000)/(spec->freq/8); | |
| 356 frequency = 8; | |
| 357 ulaw_buf = (Uint8 *)malloc(fragsize); | |
| 358 if ( ulaw_buf == NULL ) { | |
| 359 SDL_OutOfMemory(); | |
| 360 return(-1); | |
| 361 } | |
| 362 spec->channels = 1; | |
| 363 } else { | |
| 364 fragsize = spec->samples; | |
| 365 frequency = spec->freq/1000; | |
| 366 } | |
| 367 #ifdef DEBUG_AUDIO | |
| 368 fprintf(stderr, "Audio device %s U-Law only\n", | |
| 369 ulaw_only ? "is" : "is not"); | |
| 370 fprintf(stderr, "format=0x%x chan=%d freq=%d\n", | |
| 371 spec->format, spec->channels, spec->freq); | |
| 372 #endif | |
| 373 | |
| 374 /* Update the fragment size as size in bytes */ | |
| 375 SDL_CalculateAudioSpec(spec); | |
| 376 | |
| 377 /* Allocate mixing buffer */ | |
| 378 mixbuf = (Uint8 *)SDL_AllocAudioMem(spec->size); | |
| 379 if ( mixbuf == NULL ) { | |
| 380 SDL_OutOfMemory(); | |
| 381 return(-1); | |
| 382 } | |
| 383 memset(mixbuf, spec->silence, spec->size); | |
| 384 | |
| 385 /* We're ready to rock and roll. :-) */ | |
| 386 return(0); | |
| 387 } | |
| 388 | |
| 389 /************************************************************************/ | |
| 390 /* This function (snd2au()) copyrighted: */ | |
| 391 /************************************************************************/ | |
| 392 /* Copyright 1989 by Rich Gopstein and Harris Corporation */ | |
| 393 /* */ | |
| 394 /* Permission to use, copy, modify, and distribute this software */ | |
| 395 /* and its documentation for any purpose and without fee is */ | |
| 396 /* hereby granted, provided that the above copyright notice */ | |
| 397 /* appears in all copies and that both that copyright notice and */ | |
| 398 /* this permission notice appear in supporting documentation, and */ | |
| 399 /* that the name of Rich Gopstein and Harris Corporation not be */ | |
| 400 /* used in advertising or publicity pertaining to distribution */ | |
| 401 /* of the software without specific, written prior permission. */ | |
| 402 /* Rich Gopstein and Harris Corporation make no representations */ | |
| 403 /* about the suitability of this software for any purpose. It */ | |
| 404 /* provided "as is" without express or implied warranty. */ | |
| 405 /************************************************************************/ | |
| 406 | |
| 407 static Uint8 snd2au(int sample) | |
| 408 { | |
| 409 | |
| 410 int mask; | |
| 411 | |
| 412 if (sample < 0) { | |
| 413 sample = -sample; | |
| 414 mask = 0x7f; | |
| 415 } else { | |
| 416 mask = 0xff; | |
| 417 } | |
| 418 | |
| 419 if (sample < 32) { | |
| 420 sample = 0xF0 | (15 - sample / 2); | |
| 421 } else if (sample < 96) { | |
| 422 sample = 0xE0 | (15 - (sample - 32) / 4); | |
| 423 } else if (sample < 224) { | |
| 424 sample = 0xD0 | (15 - (sample - 96) / 8); | |
| 425 } else if (sample < 480) { | |
| 426 sample = 0xC0 | (15 - (sample - 224) / 16); | |
| 427 } else if (sample < 992) { | |
| 428 sample = 0xB0 | (15 - (sample - 480) / 32); | |
| 429 } else if (sample < 2016) { | |
| 430 sample = 0xA0 | (15 - (sample - 992) / 64); | |
| 431 } else if (sample < 4064) { | |
| 432 sample = 0x90 | (15 - (sample - 2016) / 128); | |
| 433 } else if (sample < 8160) { | |
| 434 sample = 0x80 | (15 - (sample - 4064) / 256); | |
| 435 } else { | |
| 436 sample = 0x80; | |
| 437 } | |
| 438 return (mask & sample); | |
| 439 } |