| 1 | /* |
| 2 | SDL - Simple DirectMedia Layer |
| 3 | Copyright (C) 1997-2009 Sam Lantinga |
| 4 | |
| 5 | This library is free software; you can redistribute it and/or |
| 6 | modify it under the terms of the GNU Lesser General Public |
| 7 | License as published by the Free Software Foundation; either |
| 8 | version 2.1 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 | Lesser General Public License for more details. |
| 14 | |
| 15 | You should have received a copy of the GNU Lesser General Public |
| 16 | License along with this library; if not, write to the Free Software |
| 17 | Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
| 18 | |
| 19 | Sam Lantinga |
| 20 | slouken@libsdl.org |
| 21 | */ |
| 22 | #include "SDL_config.h" |
| 23 | |
| 24 | /* Microsoft WAVE file loading routines */ |
| 25 | |
| 26 | #include "SDL_audio.h" |
| 27 | #include "SDL_wave.h" |
| 28 | |
| 29 | |
| 30 | static int ReadChunk(SDL_RWops *src, Chunk *chunk); |
| 31 | |
| 32 | struct MS_ADPCM_decodestate { |
| 33 | Uint8 hPredictor; |
| 34 | Uint16 iDelta; |
| 35 | Sint16 iSamp1; |
| 36 | Sint16 iSamp2; |
| 37 | }; |
| 38 | static struct MS_ADPCM_decoder { |
| 39 | WaveFMT wavefmt; |
| 40 | Uint16 wSamplesPerBlock; |
| 41 | Uint16 wNumCoef; |
| 42 | Sint16 aCoeff[7][2]; |
| 43 | /* * * */ |
| 44 | struct MS_ADPCM_decodestate state[2]; |
| 45 | } MS_ADPCM_state; |
| 46 | |
| 47 | static int InitMS_ADPCM(WaveFMT *format) |
| 48 | { |
| 49 | Uint8 *rogue_feel; |
| 50 | Uint16 extra_info; |
| 51 | int i; |
| 52 | |
| 53 | /* Set the rogue pointer to the MS_ADPCM specific data */ |
| 54 | MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); |
| 55 | MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); |
| 56 | MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); |
| 57 | MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); |
| 58 | MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); |
| 59 | MS_ADPCM_state.wavefmt.bitspersample = |
| 60 | SDL_SwapLE16(format->bitspersample); |
| 61 | rogue_feel = (Uint8 *)format+sizeof(*format); |
| 62 | if ( sizeof(*format) == 16 ) { |
| 63 | extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 64 | rogue_feel += sizeof(Uint16); |
| 65 | } |
| 66 | MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 67 | rogue_feel += sizeof(Uint16); |
| 68 | MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 69 | rogue_feel += sizeof(Uint16); |
| 70 | if ( MS_ADPCM_state.wNumCoef != 7 ) { |
| 71 | SDL_SetError("Unknown set of MS_ADPCM coefficients"); |
| 72 | return(-1); |
| 73 | } |
| 74 | for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) { |
| 75 | MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 76 | rogue_feel += sizeof(Uint16); |
| 77 | MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 78 | rogue_feel += sizeof(Uint16); |
| 79 | } |
| 80 | return(0); |
| 81 | } |
| 82 | |
| 83 | static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state, |
| 84 | Uint8 nybble, Sint16 *coeff) |
| 85 | { |
| 86 | const Sint32 max_audioval = ((1<<(16-1))-1); |
| 87 | const Sint32 min_audioval = -(1<<(16-1)); |
| 88 | const Sint32 adaptive[] = { |
| 89 | 230, 230, 230, 230, 307, 409, 512, 614, |
| 90 | 768, 614, 512, 409, 307, 230, 230, 230 |
| 91 | }; |
| 92 | Sint32 new_sample, delta; |
| 93 | |
| 94 | new_sample = ((state->iSamp1 * coeff[0]) + |
| 95 | (state->iSamp2 * coeff[1]))/256; |
| 96 | if ( nybble & 0x08 ) { |
| 97 | new_sample += state->iDelta * (nybble-0x10); |
| 98 | } else { |
| 99 | new_sample += state->iDelta * nybble; |
| 100 | } |
| 101 | if ( new_sample < min_audioval ) { |
| 102 | new_sample = min_audioval; |
| 103 | } else |
| 104 | if ( new_sample > max_audioval ) { |
| 105 | new_sample = max_audioval; |
| 106 | } |
| 107 | delta = ((Sint32)state->iDelta * adaptive[nybble])/256; |
| 108 | if ( delta < 16 ) { |
| 109 | delta = 16; |
| 110 | } |
| 111 | state->iDelta = (Uint16)delta; |
| 112 | state->iSamp2 = state->iSamp1; |
| 113 | state->iSamp1 = (Sint16)new_sample; |
| 114 | return(new_sample); |
| 115 | } |
| 116 | |
| 117 | static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) |
| 118 | { |
| 119 | struct MS_ADPCM_decodestate *state[2]; |
| 120 | Uint8 *freeable, *encoded, *decoded; |
| 121 | Sint32 encoded_len, samplesleft; |
| 122 | Sint8 nybble, stereo; |
| 123 | Sint16 *coeff[2]; |
| 124 | Sint32 new_sample; |
| 125 | |
| 126 | /* Allocate the proper sized output buffer */ |
| 127 | encoded_len = *audio_len; |
| 128 | encoded = *audio_buf; |
| 129 | freeable = *audio_buf; |
| 130 | *audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * |
| 131 | MS_ADPCM_state.wSamplesPerBlock* |
| 132 | MS_ADPCM_state.wavefmt.channels*sizeof(Sint16); |
| 133 | *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 134 | if ( *audio_buf == NULL ) { |
| 135 | SDL_Error(SDL_ENOMEM); |
| 136 | return(-1); |
| 137 | } |
| 138 | decoded = *audio_buf; |
| 139 | |
| 140 | /* Get ready... Go! */ |
| 141 | stereo = (MS_ADPCM_state.wavefmt.channels == 2); |
| 142 | state[0] = &MS_ADPCM_state.state[0]; |
| 143 | state[1] = &MS_ADPCM_state.state[stereo]; |
| 144 | while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) { |
| 145 | /* Grab the initial information for this block */ |
| 146 | state[0]->hPredictor = *encoded++; |
| 147 | if ( stereo ) { |
| 148 | state[1]->hPredictor = *encoded++; |
| 149 | } |
| 150 | state[0]->iDelta = ((encoded[1]<<8)|encoded[0]); |
| 151 | encoded += sizeof(Sint16); |
| 152 | if ( stereo ) { |
| 153 | state[1]->iDelta = ((encoded[1]<<8)|encoded[0]); |
| 154 | encoded += sizeof(Sint16); |
| 155 | } |
| 156 | state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]); |
| 157 | encoded += sizeof(Sint16); |
| 158 | if ( stereo ) { |
| 159 | state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]); |
| 160 | encoded += sizeof(Sint16); |
| 161 | } |
| 162 | state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]); |
| 163 | encoded += sizeof(Sint16); |
| 164 | if ( stereo ) { |
| 165 | state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]); |
| 166 | encoded += sizeof(Sint16); |
| 167 | } |
| 168 | coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor]; |
| 169 | coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor]; |
| 170 | |
| 171 | /* Store the two initial samples we start with */ |
| 172 | decoded[0] = state[0]->iSamp2&0xFF; |
| 173 | decoded[1] = state[0]->iSamp2>>8; |
| 174 | decoded += 2; |
| 175 | if ( stereo ) { |
| 176 | decoded[0] = state[1]->iSamp2&0xFF; |
| 177 | decoded[1] = state[1]->iSamp2>>8; |
| 178 | decoded += 2; |
| 179 | } |
| 180 | decoded[0] = state[0]->iSamp1&0xFF; |
| 181 | decoded[1] = state[0]->iSamp1>>8; |
| 182 | decoded += 2; |
| 183 | if ( stereo ) { |
| 184 | decoded[0] = state[1]->iSamp1&0xFF; |
| 185 | decoded[1] = state[1]->iSamp1>>8; |
| 186 | decoded += 2; |
| 187 | } |
| 188 | |
| 189 | /* Decode and store the other samples in this block */ |
| 190 | samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)* |
| 191 | MS_ADPCM_state.wavefmt.channels; |
| 192 | while ( samplesleft > 0 ) { |
| 193 | nybble = (*encoded)>>4; |
| 194 | new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]); |
| 195 | decoded[0] = new_sample&0xFF; |
| 196 | new_sample >>= 8; |
| 197 | decoded[1] = new_sample&0xFF; |
| 198 | decoded += 2; |
| 199 | |
| 200 | nybble = (*encoded)&0x0F; |
| 201 | new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]); |
| 202 | decoded[0] = new_sample&0xFF; |
| 203 | new_sample >>= 8; |
| 204 | decoded[1] = new_sample&0xFF; |
| 205 | decoded += 2; |
| 206 | |
| 207 | ++encoded; |
| 208 | samplesleft -= 2; |
| 209 | } |
| 210 | encoded_len -= MS_ADPCM_state.wavefmt.blockalign; |
| 211 | } |
| 212 | SDL_free(freeable); |
| 213 | return(0); |
| 214 | } |
| 215 | |
| 216 | struct IMA_ADPCM_decodestate { |
| 217 | Sint32 sample; |
| 218 | Sint8 index; |
| 219 | }; |
| 220 | static struct IMA_ADPCM_decoder { |
| 221 | WaveFMT wavefmt; |
| 222 | Uint16 wSamplesPerBlock; |
| 223 | /* * * */ |
| 224 | struct IMA_ADPCM_decodestate state[2]; |
| 225 | } IMA_ADPCM_state; |
| 226 | |
| 227 | static int InitIMA_ADPCM(WaveFMT *format) |
| 228 | { |
| 229 | Uint8 *rogue_feel; |
| 230 | Uint16 extra_info; |
| 231 | |
| 232 | /* Set the rogue pointer to the IMA_ADPCM specific data */ |
| 233 | IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); |
| 234 | IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); |
| 235 | IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); |
| 236 | IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); |
| 237 | IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); |
| 238 | IMA_ADPCM_state.wavefmt.bitspersample = |
| 239 | SDL_SwapLE16(format->bitspersample); |
| 240 | rogue_feel = (Uint8 *)format+sizeof(*format); |
| 241 | if ( sizeof(*format) == 16 ) { |
| 242 | extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 243 | rogue_feel += sizeof(Uint16); |
| 244 | } |
| 245 | IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); |
| 246 | return(0); |
| 247 | } |
| 248 | |
| 249 | static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble) |
| 250 | { |
| 251 | const Sint32 max_audioval = ((1<<(16-1))-1); |
| 252 | const Sint32 min_audioval = -(1<<(16-1)); |
| 253 | const int index_table[16] = { |
| 254 | -1, -1, -1, -1, |
| 255 | 2, 4, 6, 8, |
| 256 | -1, -1, -1, -1, |
| 257 | 2, 4, 6, 8 |
| 258 | }; |
| 259 | const Sint32 step_table[89] = { |
| 260 | 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, |
| 261 | 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, |
| 262 | 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, |
| 263 | 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, |
| 264 | 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, |
| 265 | 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, |
| 266 | 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, |
| 267 | 22385, 24623, 27086, 29794, 32767 |
| 268 | }; |
| 269 | Sint32 delta, step; |
| 270 | |
| 271 | /* Compute difference and new sample value */ |
| 272 | step = step_table[state->index]; |
| 273 | delta = step >> 3; |
| 274 | if ( nybble & 0x04 ) delta += step; |
| 275 | if ( nybble & 0x02 ) delta += (step >> 1); |
| 276 | if ( nybble & 0x01 ) delta += (step >> 2); |
| 277 | if ( nybble & 0x08 ) delta = -delta; |
| 278 | state->sample += delta; |
| 279 | |
| 280 | /* Update index value */ |
| 281 | state->index += index_table[nybble]; |
| 282 | if ( state->index > 88 ) { |
| 283 | state->index = 88; |
| 284 | } else |
| 285 | if ( state->index < 0 ) { |
| 286 | state->index = 0; |
| 287 | } |
| 288 | |
| 289 | /* Clamp output sample */ |
| 290 | if ( state->sample > max_audioval ) { |
| 291 | state->sample = max_audioval; |
| 292 | } else |
| 293 | if ( state->sample < min_audioval ) { |
| 294 | state->sample = min_audioval; |
| 295 | } |
| 296 | return(state->sample); |
| 297 | } |
| 298 | |
| 299 | /* Fill the decode buffer with a channel block of data (8 samples) */ |
| 300 | static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded, |
| 301 | int channel, int numchannels, struct IMA_ADPCM_decodestate *state) |
| 302 | { |
| 303 | int i; |
| 304 | Sint8 nybble; |
| 305 | Sint32 new_sample; |
| 306 | |
| 307 | decoded += (channel * 2); |
| 308 | for ( i=0; i<4; ++i ) { |
| 309 | nybble = (*encoded)&0x0F; |
| 310 | new_sample = IMA_ADPCM_nibble(state, nybble); |
| 311 | decoded[0] = new_sample&0xFF; |
| 312 | new_sample >>= 8; |
| 313 | decoded[1] = new_sample&0xFF; |
| 314 | decoded += 2 * numchannels; |
| 315 | |
| 316 | nybble = (*encoded)>>4; |
| 317 | new_sample = IMA_ADPCM_nibble(state, nybble); |
| 318 | decoded[0] = new_sample&0xFF; |
| 319 | new_sample >>= 8; |
| 320 | decoded[1] = new_sample&0xFF; |
| 321 | decoded += 2 * numchannels; |
| 322 | |
| 323 | ++encoded; |
| 324 | } |
| 325 | } |
| 326 | |
| 327 | static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) |
| 328 | { |
| 329 | struct IMA_ADPCM_decodestate *state; |
| 330 | Uint8 *freeable, *encoded, *decoded; |
| 331 | Sint32 encoded_len, samplesleft; |
| 332 | unsigned int c, channels; |
| 333 | |
| 334 | /* Check to make sure we have enough variables in the state array */ |
| 335 | channels = IMA_ADPCM_state.wavefmt.channels; |
| 336 | if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) { |
| 337 | SDL_SetError("IMA ADPCM decoder can only handle %d channels", |
| 338 | SDL_arraysize(IMA_ADPCM_state.state)); |
| 339 | return(-1); |
| 340 | } |
| 341 | state = IMA_ADPCM_state.state; |
| 342 | |
| 343 | /* Allocate the proper sized output buffer */ |
| 344 | encoded_len = *audio_len; |
| 345 | encoded = *audio_buf; |
| 346 | freeable = *audio_buf; |
| 347 | *audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * |
| 348 | IMA_ADPCM_state.wSamplesPerBlock* |
| 349 | IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16); |
| 350 | *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 351 | if ( *audio_buf == NULL ) { |
| 352 | SDL_Error(SDL_ENOMEM); |
| 353 | return(-1); |
| 354 | } |
| 355 | decoded = *audio_buf; |
| 356 | |
| 357 | /* Get ready... Go! */ |
| 358 | while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) { |
| 359 | /* Grab the initial information for this block */ |
| 360 | for ( c=0; c<channels; ++c ) { |
| 361 | /* Fill the state information for this block */ |
| 362 | state[c].sample = ((encoded[1]<<8)|encoded[0]); |
| 363 | encoded += 2; |
| 364 | if ( state[c].sample & 0x8000 ) { |
| 365 | state[c].sample -= 0x10000; |
| 366 | } |
| 367 | state[c].index = *encoded++; |
| 368 | /* Reserved byte in buffer header, should be 0 */ |
| 369 | if ( *encoded++ != 0 ) { |
| 370 | /* Uh oh, corrupt data? Buggy code? */; |
| 371 | } |
| 372 | |
| 373 | /* Store the initial sample we start with */ |
| 374 | decoded[0] = (Uint8)(state[c].sample&0xFF); |
| 375 | decoded[1] = (Uint8)(state[c].sample>>8); |
| 376 | decoded += 2; |
| 377 | } |
| 378 | |
| 379 | /* Decode and store the other samples in this block */ |
| 380 | samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels; |
| 381 | while ( samplesleft > 0 ) { |
| 382 | for ( c=0; c<channels; ++c ) { |
| 383 | Fill_IMA_ADPCM_block(decoded, encoded, |
| 384 | c, channels, &state[c]); |
| 385 | encoded += 4; |
| 386 | samplesleft -= 8; |
| 387 | } |
| 388 | decoded += (channels * 8 * 2); |
| 389 | } |
| 390 | encoded_len -= IMA_ADPCM_state.wavefmt.blockalign; |
| 391 | } |
| 392 | SDL_free(freeable); |
| 393 | return(0); |
| 394 | } |
| 395 | |
| 396 | SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, |
| 397 | SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) |
| 398 | { |
| 399 | int was_error; |
| 400 | Chunk chunk; |
| 401 | int lenread; |
| 402 | int MS_ADPCM_encoded, IMA_ADPCM_encoded; |
| 403 | int samplesize; |
| 404 | |
| 405 | /* WAV magic header */ |
| 406 | Uint32 RIFFchunk; |
| 407 | Uint32 wavelen = 0; |
| 408 | Uint32 WAVEmagic; |
| 409 | Uint32 headerDiff = 0; |
| 410 | |
| 411 | /* FMT chunk */ |
| 412 | WaveFMT *format = NULL; |
| 413 | |
| 414 | /* Make sure we are passed a valid data source */ |
| 415 | was_error = 0; |
| 416 | if ( src == NULL ) { |
| 417 | was_error = 1; |
| 418 | goto done; |
| 419 | } |
| 420 | |
| 421 | /* Check the magic header */ |
| 422 | RIFFchunk = SDL_ReadLE32(src); |
| 423 | wavelen = SDL_ReadLE32(src); |
| 424 | if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */ |
| 425 | WAVEmagic = wavelen; |
| 426 | wavelen = RIFFchunk; |
| 427 | RIFFchunk = RIFF; |
| 428 | } else { |
| 429 | WAVEmagic = SDL_ReadLE32(src); |
| 430 | } |
| 431 | if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) { |
| 432 | SDL_SetError("Unrecognized file type (not WAVE)"); |
| 433 | was_error = 1; |
| 434 | goto done; |
| 435 | } |
| 436 | headerDiff += sizeof(Uint32); /* for WAVE */ |
| 437 | |
| 438 | /* Read the audio data format chunk */ |
| 439 | chunk.data = NULL; |
| 440 | do { |
| 441 | if ( chunk.data != NULL ) { |
| 442 | SDL_free(chunk.data); |
| 443 | chunk.data = NULL; |
| 444 | } |
| 445 | lenread = ReadChunk(src, &chunk); |
| 446 | if ( lenread < 0 ) { |
| 447 | was_error = 1; |
| 448 | goto done; |
| 449 | } |
| 450 | /* 2 Uint32's for chunk header+len, plus the lenread */ |
| 451 | headerDiff += lenread + 2 * sizeof(Uint32); |
| 452 | } while ( (chunk.magic == FACT) || (chunk.magic == LIST) ); |
| 453 | |
| 454 | /* Decode the audio data format */ |
| 455 | format = (WaveFMT *)chunk.data; |
| 456 | if ( chunk.magic != FMT ) { |
| 457 | SDL_SetError("Complex WAVE files not supported"); |
| 458 | was_error = 1; |
| 459 | goto done; |
| 460 | } |
| 461 | MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; |
| 462 | switch (SDL_SwapLE16(format->encoding)) { |
| 463 | case PCM_CODE: |
| 464 | /* We can understand this */ |
| 465 | break; |
| 466 | case MS_ADPCM_CODE: |
| 467 | /* Try to understand this */ |
| 468 | if ( InitMS_ADPCM(format) < 0 ) { |
| 469 | was_error = 1; |
| 470 | goto done; |
| 471 | } |
| 472 | MS_ADPCM_encoded = 1; |
| 473 | break; |
| 474 | case IMA_ADPCM_CODE: |
| 475 | /* Try to understand this */ |
| 476 | if ( InitIMA_ADPCM(format) < 0 ) { |
| 477 | was_error = 1; |
| 478 | goto done; |
| 479 | } |
| 480 | IMA_ADPCM_encoded = 1; |
| 481 | break; |
| 482 | case MP3_CODE: |
| 483 | SDL_SetError("MPEG Layer 3 data not supported", |
| 484 | SDL_SwapLE16(format->encoding)); |
| 485 | was_error = 1; |
| 486 | goto done; |
| 487 | default: |
| 488 | SDL_SetError("Unknown WAVE data format: 0x%.4x", |
| 489 | SDL_SwapLE16(format->encoding)); |
| 490 | was_error = 1; |
| 491 | goto done; |
| 492 | } |
| 493 | SDL_memset(spec, 0, (sizeof *spec)); |
| 494 | spec->freq = SDL_SwapLE32(format->frequency); |
| 495 | switch (SDL_SwapLE16(format->bitspersample)) { |
| 496 | case 4: |
| 497 | if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { |
| 498 | spec->format = AUDIO_S16; |
| 499 | } else { |
| 500 | was_error = 1; |
| 501 | } |
| 502 | break; |
| 503 | case 8: |
| 504 | spec->format = AUDIO_U8; |
| 505 | break; |
| 506 | case 16: |
| 507 | spec->format = AUDIO_S16; |
| 508 | break; |
| 509 | default: |
| 510 | was_error = 1; |
| 511 | break; |
| 512 | } |
| 513 | if ( was_error ) { |
| 514 | SDL_SetError("Unknown %d-bit PCM data format", |
| 515 | SDL_SwapLE16(format->bitspersample)); |
| 516 | goto done; |
| 517 | } |
| 518 | spec->channels = (Uint8)SDL_SwapLE16(format->channels); |
| 519 | spec->samples = 4096; /* Good default buffer size */ |
| 520 | |
| 521 | /* Read the audio data chunk */ |
| 522 | *audio_buf = NULL; |
| 523 | do { |
| 524 | if ( *audio_buf != NULL ) { |
| 525 | SDL_free(*audio_buf); |
| 526 | *audio_buf = NULL; |
| 527 | } |
| 528 | lenread = ReadChunk(src, &chunk); |
| 529 | if ( lenread < 0 ) { |
| 530 | was_error = 1; |
| 531 | goto done; |
| 532 | } |
| 533 | *audio_len = lenread; |
| 534 | *audio_buf = chunk.data; |
| 535 | if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32); |
| 536 | } while ( chunk.magic != DATA ); |
| 537 | headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */ |
| 538 | |
| 539 | if ( MS_ADPCM_encoded ) { |
| 540 | if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { |
| 541 | was_error = 1; |
| 542 | goto done; |
| 543 | } |
| 544 | } |
| 545 | if ( IMA_ADPCM_encoded ) { |
| 546 | if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { |
| 547 | was_error = 1; |
| 548 | goto done; |
| 549 | } |
| 550 | } |
| 551 | |
| 552 | /* Don't return a buffer that isn't a multiple of samplesize */ |
| 553 | samplesize = ((spec->format & 0xFF)/8)*spec->channels; |
| 554 | *audio_len &= ~(samplesize-1); |
| 555 | |
| 556 | done: |
| 557 | if ( format != NULL ) { |
| 558 | SDL_free(format); |
| 559 | } |
| 560 | if ( src ) { |
| 561 | if ( freesrc ) { |
| 562 | SDL_RWclose(src); |
| 563 | } else { |
| 564 | /* seek to the end of the file (given by the RIFF chunk) */ |
| 565 | SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR); |
| 566 | } |
| 567 | } |
| 568 | if ( was_error ) { |
| 569 | spec = NULL; |
| 570 | } |
| 571 | return(spec); |
| 572 | } |
| 573 | |
| 574 | /* Since the WAV memory is allocated in the shared library, it must also |
| 575 | be freed here. (Necessary under Win32, VC++) |
| 576 | */ |
| 577 | void SDL_FreeWAV(Uint8 *audio_buf) |
| 578 | { |
| 579 | if ( audio_buf != NULL ) { |
| 580 | SDL_free(audio_buf); |
| 581 | } |
| 582 | } |
| 583 | |
| 584 | static int ReadChunk(SDL_RWops *src, Chunk *chunk) |
| 585 | { |
| 586 | chunk->magic = SDL_ReadLE32(src); |
| 587 | chunk->length = SDL_ReadLE32(src); |
| 588 | chunk->data = (Uint8 *)SDL_malloc(chunk->length); |
| 589 | if ( chunk->data == NULL ) { |
| 590 | SDL_Error(SDL_ENOMEM); |
| 591 | return(-1); |
| 592 | } |
| 593 | if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { |
| 594 | SDL_Error(SDL_EFREAD); |
| 595 | SDL_free(chunk->data); |
| 596 | chunk->data = NULL; |
| 597 | return(-1); |
| 598 | } |
| 599 | return(chunk->length); |
| 600 | } |