Mercurial > sdl-ios-xcode
annotate src/audio/SDL_wave.c @ 956:4263beff9e38
Date: Mon, 30 Aug 2004 18:20:25 +0200
From: Joost Baas
Subject: why call arts artsc?
I, and a few other people at the mplayer-docs-mailinglist were wondering why
you decided to call arts artsc. I understand usually users have nothing to do
with libsdl, just developers, but because you can choose the audio-driver
being used by mplayer, one of which is sdl, and you can also choose the sdl
subdriver, it is necessary to have a well-known or logical name. artsc is not
the logical choice, and it's very hard to look up the right name if you don't
know what you're looking for.
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Fri, 17 Sep 2004 13:25:06 +0000 |
| parents | b8d311d90021 |
| children | 80f8c94b5199 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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Updated copyright information for 2004 (Happy New Year!)
Sam Lantinga <slouken@libsdl.org>
parents:
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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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|
20 slouken@libsdl.org |
| 0 | 21 */ |
| 22 | |
| 23 #ifdef SAVE_RCSID | |
| 24 static char rcsid = | |
| 25 "@(#) $Id$"; | |
| 26 #endif | |
| 27 | |
| 28 #ifndef DISABLE_FILE | |
| 29 | |
| 30 /* Microsoft WAVE file loading routines */ | |
| 31 | |
| 32 #include <stdlib.h> | |
| 33 #include <string.h> | |
| 34 | |
| 35 #include "SDL_error.h" | |
| 36 #include "SDL_audio.h" | |
| 37 #include "SDL_wave.h" | |
| 38 #include "SDL_endian.h" | |
| 39 | |
| 40 #ifndef NELEMS | |
| 41 #define NELEMS(array) ((sizeof array)/(sizeof array[0])) | |
| 42 #endif | |
| 43 | |
| 44 static int ReadChunk(SDL_RWops *src, Chunk *chunk); | |
| 45 | |
| 46 struct MS_ADPCM_decodestate { | |
| 47 Uint8 hPredictor; | |
| 48 Uint16 iDelta; | |
| 49 Sint16 iSamp1; | |
| 50 Sint16 iSamp2; | |
| 51 }; | |
| 52 static struct MS_ADPCM_decoder { | |
| 53 WaveFMT wavefmt; | |
| 54 Uint16 wSamplesPerBlock; | |
| 55 Uint16 wNumCoef; | |
| 56 Sint16 aCoeff[7][2]; | |
| 57 /* * * */ | |
| 58 struct MS_ADPCM_decodestate state[2]; | |
| 59 } MS_ADPCM_state; | |
| 60 | |
| 61 static int InitMS_ADPCM(WaveFMT *format) | |
| 62 { | |
| 63 Uint8 *rogue_feel; | |
| 64 Uint16 extra_info; | |
| 65 int i; | |
| 66 | |
| 67 /* Set the rogue pointer to the MS_ADPCM specific data */ | |
| 68 MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 69 MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 70 MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 71 MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 72 MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 73 MS_ADPCM_state.wavefmt.bitspersample = | |
| 74 SDL_SwapLE16(format->bitspersample); | |
| 75 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 76 if ( sizeof(*format) == 16 ) { | |
| 77 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 78 rogue_feel += sizeof(Uint16); | |
| 79 } | |
| 80 MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 81 rogue_feel += sizeof(Uint16); | |
| 82 MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 83 rogue_feel += sizeof(Uint16); | |
| 84 if ( MS_ADPCM_state.wNumCoef != 7 ) { | |
| 85 SDL_SetError("Unknown set of MS_ADPCM coefficients"); | |
| 86 return(-1); | |
| 87 } | |
| 88 for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) { | |
| 89 MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 90 rogue_feel += sizeof(Uint16); | |
| 91 MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 92 rogue_feel += sizeof(Uint16); | |
| 93 } | |
| 94 return(0); | |
| 95 } | |
| 96 | |
| 97 static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state, | |
| 98 Uint8 nybble, Sint16 *coeff) | |
| 99 { | |
| 100 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 101 const Sint32 min_audioval = -(1<<(16-1)); | |
| 102 const Sint32 adaptive[] = { | |
| 103 230, 230, 230, 230, 307, 409, 512, 614, | |
| 104 768, 614, 512, 409, 307, 230, 230, 230 | |
| 105 }; | |
| 106 Sint32 new_sample, delta; | |
| 107 | |
| 108 new_sample = ((state->iSamp1 * coeff[0]) + | |
| 109 (state->iSamp2 * coeff[1]))/256; | |
| 110 if ( nybble & 0x08 ) { | |
| 111 new_sample += state->iDelta * (nybble-0x10); | |
| 112 } else { | |
| 113 new_sample += state->iDelta * nybble; | |
| 114 } | |
| 115 if ( new_sample < min_audioval ) { | |
| 116 new_sample = min_audioval; | |
| 117 } else | |
| 118 if ( new_sample > max_audioval ) { | |
| 119 new_sample = max_audioval; | |
| 120 } | |
| 121 delta = ((Sint32)state->iDelta * adaptive[nybble])/256; | |
| 122 if ( delta < 16 ) { | |
| 123 delta = 16; | |
| 124 } | |
| 125 state->iDelta = delta; | |
| 126 state->iSamp2 = state->iSamp1; | |
| 127 state->iSamp1 = new_sample; | |
| 128 return(new_sample); | |
| 129 } | |
| 130 | |
| 131 static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 132 { | |
| 133 struct MS_ADPCM_decodestate *state[2]; | |
| 134 Uint8 *freeable, *encoded, *decoded; | |
| 135 Sint32 encoded_len, samplesleft; | |
| 136 Sint8 nybble, stereo; | |
| 137 Sint16 *coeff[2]; | |
| 138 Sint32 new_sample; | |
| 139 | |
| 140 /* Allocate the proper sized output buffer */ | |
| 141 encoded_len = *audio_len; | |
| 142 encoded = *audio_buf; | |
| 143 freeable = *audio_buf; | |
| 144 *audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * | |
| 145 MS_ADPCM_state.wSamplesPerBlock* | |
| 146 MS_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
| 147 *audio_buf = (Uint8 *)malloc(*audio_len); | |
| 148 if ( *audio_buf == NULL ) { | |
| 149 SDL_Error(SDL_ENOMEM); | |
| 150 return(-1); | |
| 151 } | |
| 152 decoded = *audio_buf; | |
| 153 | |
| 154 /* Get ready... Go! */ | |
| 155 stereo = (MS_ADPCM_state.wavefmt.channels == 2); | |
| 156 state[0] = &MS_ADPCM_state.state[0]; | |
| 157 state[1] = &MS_ADPCM_state.state[stereo]; | |
| 158 while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) { | |
| 159 /* Grab the initial information for this block */ | |
| 160 state[0]->hPredictor = *encoded++; | |
| 161 if ( stereo ) { | |
| 162 state[1]->hPredictor = *encoded++; | |
| 163 } | |
| 164 state[0]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 165 encoded += sizeof(Sint16); | |
| 166 if ( stereo ) { | |
| 167 state[1]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 168 encoded += sizeof(Sint16); | |
| 169 } | |
| 170 state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 171 encoded += sizeof(Sint16); | |
| 172 if ( stereo ) { | |
| 173 state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 174 encoded += sizeof(Sint16); | |
| 175 } | |
| 176 state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 177 encoded += sizeof(Sint16); | |
| 178 if ( stereo ) { | |
| 179 state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 180 encoded += sizeof(Sint16); | |
| 181 } | |
| 182 coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor]; | |
| 183 coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor]; | |
| 184 | |
| 185 /* Store the two initial samples we start with */ | |
| 186 decoded[0] = state[0]->iSamp2&0xFF; | |
| 187 decoded[1] = state[0]->iSamp2>>8; | |
| 188 decoded += 2; | |
| 189 if ( stereo ) { | |
| 190 decoded[0] = state[1]->iSamp2&0xFF; | |
| 191 decoded[1] = state[1]->iSamp2>>8; | |
| 192 decoded += 2; | |
| 193 } | |
| 194 decoded[0] = state[0]->iSamp1&0xFF; | |
| 195 decoded[1] = state[0]->iSamp1>>8; | |
| 196 decoded += 2; | |
| 197 if ( stereo ) { | |
| 198 decoded[0] = state[1]->iSamp1&0xFF; | |
| 199 decoded[1] = state[1]->iSamp1>>8; | |
| 200 decoded += 2; | |
| 201 } | |
| 202 | |
| 203 /* Decode and store the other samples in this block */ | |
| 204 samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)* | |
| 205 MS_ADPCM_state.wavefmt.channels; | |
| 206 while ( samplesleft > 0 ) { | |
| 207 nybble = (*encoded)>>4; | |
| 208 new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]); | |
| 209 decoded[0] = new_sample&0xFF; | |
| 210 new_sample >>= 8; | |
| 211 decoded[1] = new_sample&0xFF; | |
| 212 decoded += 2; | |
| 213 | |
| 214 nybble = (*encoded)&0x0F; | |
| 215 new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]); | |
| 216 decoded[0] = new_sample&0xFF; | |
| 217 new_sample >>= 8; | |
| 218 decoded[1] = new_sample&0xFF; | |
| 219 decoded += 2; | |
| 220 | |
| 221 ++encoded; | |
| 222 samplesleft -= 2; | |
| 223 } | |
| 224 encoded_len -= MS_ADPCM_state.wavefmt.blockalign; | |
| 225 } | |
| 226 free(freeable); | |
| 227 return(0); | |
| 228 } | |
| 229 | |
| 230 struct IMA_ADPCM_decodestate { | |
| 231 Sint32 sample; | |
| 232 Sint8 index; | |
| 233 }; | |
| 234 static struct IMA_ADPCM_decoder { | |
| 235 WaveFMT wavefmt; | |
| 236 Uint16 wSamplesPerBlock; | |
| 237 /* * * */ | |
| 238 struct IMA_ADPCM_decodestate state[2]; | |
| 239 } IMA_ADPCM_state; | |
| 240 | |
| 241 static int InitIMA_ADPCM(WaveFMT *format) | |
| 242 { | |
| 243 Uint8 *rogue_feel; | |
| 244 Uint16 extra_info; | |
| 245 | |
| 246 /* Set the rogue pointer to the IMA_ADPCM specific data */ | |
| 247 IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 248 IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 249 IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 250 IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 251 IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 252 IMA_ADPCM_state.wavefmt.bitspersample = | |
| 253 SDL_SwapLE16(format->bitspersample); | |
| 254 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 255 if ( sizeof(*format) == 16 ) { | |
| 256 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 257 rogue_feel += sizeof(Uint16); | |
| 258 } | |
| 259 IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 260 return(0); | |
| 261 } | |
| 262 | |
| 263 static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble) | |
| 264 { | |
| 265 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 266 const Sint32 min_audioval = -(1<<(16-1)); | |
| 267 const int index_table[16] = { | |
| 268 -1, -1, -1, -1, | |
| 269 2, 4, 6, 8, | |
| 270 -1, -1, -1, -1, | |
| 271 2, 4, 6, 8 | |
| 272 }; | |
| 273 const Sint32 step_table[89] = { | |
| 274 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, | |
| 275 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, | |
| 276 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, | |
| 277 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, | |
| 278 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, | |
| 279 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, | |
| 280 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, | |
| 281 22385, 24623, 27086, 29794, 32767 | |
| 282 }; | |
| 283 Sint32 delta, step; | |
| 284 | |
| 285 /* Compute difference and new sample value */ | |
| 286 step = step_table[state->index]; | |
| 287 delta = step >> 3; | |
| 288 if ( nybble & 0x04 ) delta += step; | |
| 289 if ( nybble & 0x02 ) delta += (step >> 1); | |
| 290 if ( nybble & 0x01 ) delta += (step >> 2); | |
| 291 if ( nybble & 0x08 ) delta = -delta; | |
| 292 state->sample += delta; | |
| 293 | |
| 294 /* Update index value */ | |
| 295 state->index += index_table[nybble]; | |
| 296 if ( state->index > 88 ) { | |
| 297 state->index = 88; | |
| 298 } else | |
| 299 if ( state->index < 0 ) { | |
| 300 state->index = 0; | |
| 301 } | |
| 302 | |
| 303 /* Clamp output sample */ | |
| 304 if ( state->sample > max_audioval ) { | |
| 305 state->sample = max_audioval; | |
| 306 } else | |
| 307 if ( state->sample < min_audioval ) { | |
| 308 state->sample = min_audioval; | |
| 309 } | |
| 310 return(state->sample); | |
| 311 } | |
| 312 | |
| 313 /* Fill the decode buffer with a channel block of data (8 samples) */ | |
| 314 static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded, | |
| 315 int channel, int numchannels, struct IMA_ADPCM_decodestate *state) | |
| 316 { | |
| 317 int i; | |
| 318 Sint8 nybble; | |
| 319 Sint32 new_sample; | |
| 320 | |
| 321 decoded += (channel * 2); | |
| 322 for ( i=0; i<4; ++i ) { | |
| 323 nybble = (*encoded)&0x0F; | |
| 324 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 325 decoded[0] = new_sample&0xFF; | |
| 326 new_sample >>= 8; | |
| 327 decoded[1] = new_sample&0xFF; | |
| 328 decoded += 2 * numchannels; | |
| 329 | |
| 330 nybble = (*encoded)>>4; | |
| 331 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 332 decoded[0] = new_sample&0xFF; | |
| 333 new_sample >>= 8; | |
| 334 decoded[1] = new_sample&0xFF; | |
| 335 decoded += 2 * numchannels; | |
| 336 | |
| 337 ++encoded; | |
| 338 } | |
| 339 } | |
| 340 | |
| 341 static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 342 { | |
| 343 struct IMA_ADPCM_decodestate *state; | |
| 344 Uint8 *freeable, *encoded, *decoded; | |
| 345 Sint32 encoded_len, samplesleft; | |
| 346 int c, channels; | |
| 347 | |
| 348 /* Check to make sure we have enough variables in the state array */ | |
| 349 channels = IMA_ADPCM_state.wavefmt.channels; | |
| 350 if ( channels > NELEMS(IMA_ADPCM_state.state) ) { | |
| 351 SDL_SetError("IMA ADPCM decoder can only handle %d channels", | |
| 352 NELEMS(IMA_ADPCM_state.state)); | |
| 353 return(-1); | |
| 354 } | |
| 355 state = IMA_ADPCM_state.state; | |
| 356 | |
| 357 /* Allocate the proper sized output buffer */ | |
| 358 encoded_len = *audio_len; | |
| 359 encoded = *audio_buf; | |
| 360 freeable = *audio_buf; | |
| 361 *audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * | |
| 362 IMA_ADPCM_state.wSamplesPerBlock* | |
| 363 IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
| 364 *audio_buf = (Uint8 *)malloc(*audio_len); | |
| 365 if ( *audio_buf == NULL ) { | |
| 366 SDL_Error(SDL_ENOMEM); | |
| 367 return(-1); | |
| 368 } | |
| 369 decoded = *audio_buf; | |
| 370 | |
| 371 /* Get ready... Go! */ | |
| 372 while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) { | |
| 373 /* Grab the initial information for this block */ | |
| 374 for ( c=0; c<channels; ++c ) { | |
| 375 /* Fill the state information for this block */ | |
| 376 state[c].sample = ((encoded[1]<<8)|encoded[0]); | |
| 377 encoded += 2; | |
| 378 if ( state[c].sample & 0x8000 ) { | |
| 379 state[c].sample -= 0x10000; | |
| 380 } | |
| 381 state[c].index = *encoded++; | |
| 382 /* Reserved byte in buffer header, should be 0 */ | |
| 383 if ( *encoded++ != 0 ) { | |
| 384 /* Uh oh, corrupt data? Buggy code? */; | |
| 385 } | |
| 386 | |
| 387 /* Store the initial sample we start with */ | |
| 388 decoded[0] = state[c].sample&0xFF; | |
| 389 decoded[1] = state[c].sample>>8; | |
| 390 decoded += 2; | |
| 391 } | |
| 392 | |
| 393 /* Decode and store the other samples in this block */ | |
| 394 samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels; | |
| 395 while ( samplesleft > 0 ) { | |
| 396 for ( c=0; c<channels; ++c ) { | |
| 397 Fill_IMA_ADPCM_block(decoded, encoded, | |
| 398 c, channels, &state[c]); | |
| 399 encoded += 4; | |
| 400 samplesleft -= 8; | |
| 401 } | |
| 402 decoded += (channels * 8 * 2); | |
| 403 } | |
| 404 encoded_len -= IMA_ADPCM_state.wavefmt.blockalign; | |
| 405 } | |
| 406 free(freeable); | |
| 407 return(0); | |
| 408 } | |
| 409 | |
| 410 SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, | |
| 411 SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) | |
| 412 { | |
| 413 int was_error; | |
| 414 Chunk chunk; | |
| 415 int lenread; | |
| 416 int MS_ADPCM_encoded, IMA_ADPCM_encoded; | |
| 417 int samplesize; | |
| 418 | |
| 419 /* WAV magic header */ | |
| 420 Uint32 RIFFchunk; | |
| 421 Uint32 wavelen; | |
| 422 Uint32 WAVEmagic; | |
| 423 | |
| 424 /* FMT chunk */ | |
| 425 WaveFMT *format = NULL; | |
| 426 | |
| 427 /* Make sure we are passed a valid data source */ | |
| 428 was_error = 0; | |
| 429 if ( src == NULL ) { | |
| 430 was_error = 1; | |
| 431 goto done; | |
| 432 } | |
| 433 | |
| 434 /* Check the magic header */ | |
| 435 RIFFchunk = SDL_ReadLE32(src); | |
| 436 wavelen = SDL_ReadLE32(src); | |
|
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437 if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */ |
|
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438 WAVEmagic = wavelen; |
|
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Handle the case where the WAVE magic number was already read in a non-seekable
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parents:
0
diff
changeset
|
439 wavelen = RIFFchunk; |
|
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Handle the case where the WAVE magic number was already read in a non-seekable
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
440 RIFFchunk = RIFF; |
|
02e27b705645
Handle the case where the WAVE magic number was already read in a non-seekable
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
441 } else { |
|
02e27b705645
Handle the case where the WAVE magic number was already read in a non-seekable
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
442 WAVEmagic = SDL_ReadLE32(src); |
|
02e27b705645
Handle the case where the WAVE magic number was already read in a non-seekable
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
443 } |
| 0 | 444 if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) { |
| 445 SDL_SetError("Unrecognized file type (not WAVE)"); | |
| 446 was_error = 1; | |
| 447 goto done; | |
| 448 } | |
| 449 | |
| 450 /* Read the audio data format chunk */ | |
| 451 chunk.data = NULL; | |
| 452 do { | |
| 453 if ( chunk.data != NULL ) { | |
| 454 free(chunk.data); | |
| 455 } | |
| 456 lenread = ReadChunk(src, &chunk); | |
| 457 if ( lenread < 0 ) { | |
| 458 was_error = 1; | |
| 459 goto done; | |
| 460 } | |
| 461 } while ( (chunk.magic == FACT) || (chunk.magic == LIST) ); | |
| 462 | |
| 463 /* Decode the audio data format */ | |
| 464 format = (WaveFMT *)chunk.data; | |
| 465 if ( chunk.magic != FMT ) { | |
| 466 SDL_SetError("Complex WAVE files not supported"); | |
| 467 was_error = 1; | |
| 468 goto done; | |
| 469 } | |
| 470 MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; | |
| 471 switch (SDL_SwapLE16(format->encoding)) { | |
| 472 case PCM_CODE: | |
| 473 /* We can understand this */ | |
| 474 break; | |
| 475 case MS_ADPCM_CODE: | |
| 476 /* Try to understand this */ | |
| 477 if ( InitMS_ADPCM(format) < 0 ) { | |
| 478 was_error = 1; | |
| 479 goto done; | |
| 480 } | |
| 481 MS_ADPCM_encoded = 1; | |
| 482 break; | |
| 483 case IMA_ADPCM_CODE: | |
| 484 /* Try to understand this */ | |
| 485 if ( InitIMA_ADPCM(format) < 0 ) { | |
| 486 was_error = 1; | |
| 487 goto done; | |
| 488 } | |
| 489 IMA_ADPCM_encoded = 1; | |
| 490 break; | |
| 491 default: | |
| 492 SDL_SetError("Unknown WAVE data format: 0x%.4x", | |
| 493 SDL_SwapLE16(format->encoding)); | |
| 494 was_error = 1; | |
| 495 goto done; | |
| 496 } | |
| 497 memset(spec, 0, (sizeof *spec)); | |
| 498 spec->freq = SDL_SwapLE32(format->frequency); | |
| 499 switch (SDL_SwapLE16(format->bitspersample)) { | |
| 500 case 4: | |
| 501 if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { | |
| 502 spec->format = AUDIO_S16; | |
| 503 } else { | |
| 504 was_error = 1; | |
| 505 } | |
| 506 break; | |
| 507 case 8: | |
| 508 spec->format = AUDIO_U8; | |
| 509 break; | |
| 510 case 16: | |
| 511 spec->format = AUDIO_S16; | |
| 512 break; | |
| 513 default: | |
| 514 was_error = 1; | |
| 515 break; | |
| 516 } | |
| 517 if ( was_error ) { | |
| 518 SDL_SetError("Unknown %d-bit PCM data format", | |
| 519 SDL_SwapLE16(format->bitspersample)); | |
| 520 goto done; | |
| 521 } | |
| 522 spec->channels = (Uint8)SDL_SwapLE16(format->channels); | |
| 523 spec->samples = 4096; /* Good default buffer size */ | |
| 524 | |
| 525 /* Read the audio data chunk */ | |
| 526 *audio_buf = NULL; | |
| 527 do { | |
| 528 if ( *audio_buf != NULL ) { | |
| 529 free(*audio_buf); | |
| 530 } | |
| 531 lenread = ReadChunk(src, &chunk); | |
| 532 if ( lenread < 0 ) { | |
| 533 was_error = 1; | |
| 534 goto done; | |
| 535 } | |
| 536 *audio_len = lenread; | |
| 537 *audio_buf = chunk.data; | |
| 538 } while ( chunk.magic != DATA ); | |
| 539 | |
| 540 if ( MS_ADPCM_encoded ) { | |
| 541 if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 542 was_error = 1; | |
| 543 goto done; | |
| 544 } | |
| 545 } | |
| 546 if ( IMA_ADPCM_encoded ) { | |
| 547 if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 548 was_error = 1; | |
| 549 goto done; | |
| 550 } | |
| 551 } | |
| 552 | |
| 553 /* Don't return a buffer that isn't a multiple of samplesize */ | |
| 554 samplesize = ((spec->format & 0xFF)/8)*spec->channels; | |
| 555 *audio_len &= ~(samplesize-1); | |
| 556 | |
| 557 done: | |
| 558 if ( format != NULL ) { | |
| 559 free(format); | |
| 560 } | |
| 561 if ( freesrc && src ) { | |
| 562 SDL_RWclose(src); | |
| 563 } | |
| 564 if ( was_error ) { | |
| 565 spec = NULL; | |
| 566 } | |
| 567 return(spec); | |
| 568 } | |
| 569 | |
| 570 /* Since the WAV memory is allocated in the shared library, it must also | |
| 571 be freed here. (Necessary under Win32, VC++) | |
| 572 */ | |
| 573 void SDL_FreeWAV(Uint8 *audio_buf) | |
| 574 { | |
| 575 if ( audio_buf != NULL ) { | |
| 576 free(audio_buf); | |
| 577 } | |
| 578 } | |
| 579 | |
| 580 static int ReadChunk(SDL_RWops *src, Chunk *chunk) | |
| 581 { | |
| 582 chunk->magic = SDL_ReadLE32(src); | |
| 583 chunk->length = SDL_ReadLE32(src); | |
| 584 chunk->data = (Uint8 *)malloc(chunk->length); | |
| 585 if ( chunk->data == NULL ) { | |
| 586 SDL_Error(SDL_ENOMEM); | |
| 587 return(-1); | |
| 588 } | |
| 589 if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { | |
| 590 SDL_Error(SDL_EFREAD); | |
| 591 free(chunk->data); | |
| 592 return(-1); | |
| 593 } | |
| 594 return(chunk->length); | |
| 595 } | |
| 596 | |
| 597 #endif /* ENABLE_FILE */ |