Mercurial > sdl-ios-xcode
annotate src/audio/SDL_wave.c @ 1352:c643342f3f33
Disable freaking autoheader. :)
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Thu, 09 Feb 2006 05:46:55 +0000 |
| parents | 604d73db6802 |
| children | c71e05b4dc2e |
| 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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Sam Lantinga <slouken@libsdl.org>
parents:
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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 */ |
| 22 | |
| 23 #ifndef DISABLE_FILE | |
| 24 | |
| 25 /* Microsoft WAVE file loading routines */ | |
| 26 | |
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27 #include "SDL_stdlib.h" |
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28 #include "SDL_string.h" |
| 0 | 29 #include "SDL_error.h" |
| 30 #include "SDL_audio.h" | |
| 31 #include "SDL_wave.h" | |
| 32 #include "SDL_endian.h" | |
| 33 | |
| 34 | |
| 35 static int ReadChunk(SDL_RWops *src, Chunk *chunk); | |
| 36 | |
| 37 struct MS_ADPCM_decodestate { | |
| 38 Uint8 hPredictor; | |
| 39 Uint16 iDelta; | |
| 40 Sint16 iSamp1; | |
| 41 Sint16 iSamp2; | |
| 42 }; | |
| 43 static struct MS_ADPCM_decoder { | |
| 44 WaveFMT wavefmt; | |
| 45 Uint16 wSamplesPerBlock; | |
| 46 Uint16 wNumCoef; | |
| 47 Sint16 aCoeff[7][2]; | |
| 48 /* * * */ | |
| 49 struct MS_ADPCM_decodestate state[2]; | |
| 50 } MS_ADPCM_state; | |
| 51 | |
| 52 static int InitMS_ADPCM(WaveFMT *format) | |
| 53 { | |
| 54 Uint8 *rogue_feel; | |
| 55 Uint16 extra_info; | |
| 56 int i; | |
| 57 | |
| 58 /* Set the rogue pointer to the MS_ADPCM specific data */ | |
| 59 MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 60 MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 61 MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 62 MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 63 MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 64 MS_ADPCM_state.wavefmt.bitspersample = | |
| 65 SDL_SwapLE16(format->bitspersample); | |
| 66 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 67 if ( sizeof(*format) == 16 ) { | |
| 68 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 69 rogue_feel += sizeof(Uint16); | |
| 70 } | |
| 71 MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 72 rogue_feel += sizeof(Uint16); | |
| 73 MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 74 rogue_feel += sizeof(Uint16); | |
| 75 if ( MS_ADPCM_state.wNumCoef != 7 ) { | |
| 76 SDL_SetError("Unknown set of MS_ADPCM coefficients"); | |
| 77 return(-1); | |
| 78 } | |
| 79 for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) { | |
| 80 MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 81 rogue_feel += sizeof(Uint16); | |
| 82 MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 83 rogue_feel += sizeof(Uint16); | |
| 84 } | |
| 85 return(0); | |
| 86 } | |
| 87 | |
| 88 static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state, | |
| 89 Uint8 nybble, Sint16 *coeff) | |
| 90 { | |
| 91 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 92 const Sint32 min_audioval = -(1<<(16-1)); | |
| 93 const Sint32 adaptive[] = { | |
| 94 230, 230, 230, 230, 307, 409, 512, 614, | |
| 95 768, 614, 512, 409, 307, 230, 230, 230 | |
| 96 }; | |
| 97 Sint32 new_sample, delta; | |
| 98 | |
| 99 new_sample = ((state->iSamp1 * coeff[0]) + | |
| 100 (state->iSamp2 * coeff[1]))/256; | |
| 101 if ( nybble & 0x08 ) { | |
| 102 new_sample += state->iDelta * (nybble-0x10); | |
| 103 } else { | |
| 104 new_sample += state->iDelta * nybble; | |
| 105 } | |
| 106 if ( new_sample < min_audioval ) { | |
| 107 new_sample = min_audioval; | |
| 108 } else | |
| 109 if ( new_sample > max_audioval ) { | |
| 110 new_sample = max_audioval; | |
| 111 } | |
| 112 delta = ((Sint32)state->iDelta * adaptive[nybble])/256; | |
| 113 if ( delta < 16 ) { | |
| 114 delta = 16; | |
| 115 } | |
| 116 state->iDelta = delta; | |
| 117 state->iSamp2 = state->iSamp1; | |
| 118 state->iSamp1 = new_sample; | |
| 119 return(new_sample); | |
| 120 } | |
| 121 | |
| 122 static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 123 { | |
| 124 struct MS_ADPCM_decodestate *state[2]; | |
| 125 Uint8 *freeable, *encoded, *decoded; | |
| 126 Sint32 encoded_len, samplesleft; | |
| 127 Sint8 nybble, stereo; | |
| 128 Sint16 *coeff[2]; | |
| 129 Sint32 new_sample; | |
| 130 | |
| 131 /* Allocate the proper sized output buffer */ | |
| 132 encoded_len = *audio_len; | |
| 133 encoded = *audio_buf; | |
| 134 freeable = *audio_buf; | |
| 135 *audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * | |
| 136 MS_ADPCM_state.wSamplesPerBlock* | |
| 137 MS_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
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138 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 139 if ( *audio_buf == NULL ) { |
| 140 SDL_Error(SDL_ENOMEM); | |
| 141 return(-1); | |
| 142 } | |
| 143 decoded = *audio_buf; | |
| 144 | |
| 145 /* Get ready... Go! */ | |
| 146 stereo = (MS_ADPCM_state.wavefmt.channels == 2); | |
| 147 state[0] = &MS_ADPCM_state.state[0]; | |
| 148 state[1] = &MS_ADPCM_state.state[stereo]; | |
| 149 while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) { | |
| 150 /* Grab the initial information for this block */ | |
| 151 state[0]->hPredictor = *encoded++; | |
| 152 if ( stereo ) { | |
| 153 state[1]->hPredictor = *encoded++; | |
| 154 } | |
| 155 state[0]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 156 encoded += sizeof(Sint16); | |
| 157 if ( stereo ) { | |
| 158 state[1]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 159 encoded += sizeof(Sint16); | |
| 160 } | |
| 161 state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 162 encoded += sizeof(Sint16); | |
| 163 if ( stereo ) { | |
| 164 state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 165 encoded += sizeof(Sint16); | |
| 166 } | |
| 167 state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 168 encoded += sizeof(Sint16); | |
| 169 if ( stereo ) { | |
| 170 state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 171 encoded += sizeof(Sint16); | |
| 172 } | |
| 173 coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor]; | |
| 174 coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor]; | |
| 175 | |
| 176 /* Store the two initial samples we start with */ | |
| 177 decoded[0] = state[0]->iSamp2&0xFF; | |
| 178 decoded[1] = state[0]->iSamp2>>8; | |
| 179 decoded += 2; | |
| 180 if ( stereo ) { | |
| 181 decoded[0] = state[1]->iSamp2&0xFF; | |
| 182 decoded[1] = state[1]->iSamp2>>8; | |
| 183 decoded += 2; | |
| 184 } | |
| 185 decoded[0] = state[0]->iSamp1&0xFF; | |
| 186 decoded[1] = state[0]->iSamp1>>8; | |
| 187 decoded += 2; | |
| 188 if ( stereo ) { | |
| 189 decoded[0] = state[1]->iSamp1&0xFF; | |
| 190 decoded[1] = state[1]->iSamp1>>8; | |
| 191 decoded += 2; | |
| 192 } | |
| 193 | |
| 194 /* Decode and store the other samples in this block */ | |
| 195 samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)* | |
| 196 MS_ADPCM_state.wavefmt.channels; | |
| 197 while ( samplesleft > 0 ) { | |
| 198 nybble = (*encoded)>>4; | |
| 199 new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]); | |
| 200 decoded[0] = new_sample&0xFF; | |
| 201 new_sample >>= 8; | |
| 202 decoded[1] = new_sample&0xFF; | |
| 203 decoded += 2; | |
| 204 | |
| 205 nybble = (*encoded)&0x0F; | |
| 206 new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]); | |
| 207 decoded[0] = new_sample&0xFF; | |
| 208 new_sample >>= 8; | |
| 209 decoded[1] = new_sample&0xFF; | |
| 210 decoded += 2; | |
| 211 | |
| 212 ++encoded; | |
| 213 samplesleft -= 2; | |
| 214 } | |
| 215 encoded_len -= MS_ADPCM_state.wavefmt.blockalign; | |
| 216 } | |
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217 SDL_free(freeable); |
| 0 | 218 return(0); |
| 219 } | |
| 220 | |
| 221 struct IMA_ADPCM_decodestate { | |
| 222 Sint32 sample; | |
| 223 Sint8 index; | |
| 224 }; | |
| 225 static struct IMA_ADPCM_decoder { | |
| 226 WaveFMT wavefmt; | |
| 227 Uint16 wSamplesPerBlock; | |
| 228 /* * * */ | |
| 229 struct IMA_ADPCM_decodestate state[2]; | |
| 230 } IMA_ADPCM_state; | |
| 231 | |
| 232 static int InitIMA_ADPCM(WaveFMT *format) | |
| 233 { | |
| 234 Uint8 *rogue_feel; | |
| 235 Uint16 extra_info; | |
| 236 | |
| 237 /* Set the rogue pointer to the IMA_ADPCM specific data */ | |
| 238 IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 239 IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 240 IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 241 IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 242 IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 243 IMA_ADPCM_state.wavefmt.bitspersample = | |
| 244 SDL_SwapLE16(format->bitspersample); | |
| 245 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 246 if ( sizeof(*format) == 16 ) { | |
| 247 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 248 rogue_feel += sizeof(Uint16); | |
| 249 } | |
| 250 IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 251 return(0); | |
| 252 } | |
| 253 | |
| 254 static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble) | |
| 255 { | |
| 256 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 257 const Sint32 min_audioval = -(1<<(16-1)); | |
| 258 const int index_table[16] = { | |
| 259 -1, -1, -1, -1, | |
| 260 2, 4, 6, 8, | |
| 261 -1, -1, -1, -1, | |
| 262 2, 4, 6, 8 | |
| 263 }; | |
| 264 const Sint32 step_table[89] = { | |
| 265 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, | |
| 266 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, | |
| 267 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, | |
| 268 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, | |
| 269 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, | |
| 270 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, | |
| 271 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, | |
| 272 22385, 24623, 27086, 29794, 32767 | |
| 273 }; | |
| 274 Sint32 delta, step; | |
| 275 | |
| 276 /* Compute difference and new sample value */ | |
| 277 step = step_table[state->index]; | |
| 278 delta = step >> 3; | |
| 279 if ( nybble & 0x04 ) delta += step; | |
| 280 if ( nybble & 0x02 ) delta += (step >> 1); | |
| 281 if ( nybble & 0x01 ) delta += (step >> 2); | |
| 282 if ( nybble & 0x08 ) delta = -delta; | |
| 283 state->sample += delta; | |
| 284 | |
| 285 /* Update index value */ | |
| 286 state->index += index_table[nybble]; | |
| 287 if ( state->index > 88 ) { | |
| 288 state->index = 88; | |
| 289 } else | |
| 290 if ( state->index < 0 ) { | |
| 291 state->index = 0; | |
| 292 } | |
| 293 | |
| 294 /* Clamp output sample */ | |
| 295 if ( state->sample > max_audioval ) { | |
| 296 state->sample = max_audioval; | |
| 297 } else | |
| 298 if ( state->sample < min_audioval ) { | |
| 299 state->sample = min_audioval; | |
| 300 } | |
| 301 return(state->sample); | |
| 302 } | |
| 303 | |
| 304 /* Fill the decode buffer with a channel block of data (8 samples) */ | |
| 305 static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded, | |
| 306 int channel, int numchannels, struct IMA_ADPCM_decodestate *state) | |
| 307 { | |
| 308 int i; | |
| 309 Sint8 nybble; | |
| 310 Sint32 new_sample; | |
| 311 | |
| 312 decoded += (channel * 2); | |
| 313 for ( i=0; i<4; ++i ) { | |
| 314 nybble = (*encoded)&0x0F; | |
| 315 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 316 decoded[0] = new_sample&0xFF; | |
| 317 new_sample >>= 8; | |
| 318 decoded[1] = new_sample&0xFF; | |
| 319 decoded += 2 * numchannels; | |
| 320 | |
| 321 nybble = (*encoded)>>4; | |
| 322 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 323 decoded[0] = new_sample&0xFF; | |
| 324 new_sample >>= 8; | |
| 325 decoded[1] = new_sample&0xFF; | |
| 326 decoded += 2 * numchannels; | |
| 327 | |
| 328 ++encoded; | |
| 329 } | |
| 330 } | |
| 331 | |
| 332 static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 333 { | |
| 334 struct IMA_ADPCM_decodestate *state; | |
| 335 Uint8 *freeable, *encoded, *decoded; | |
| 336 Sint32 encoded_len, samplesleft; | |
| 337 int c, channels; | |
| 338 | |
| 339 /* Check to make sure we have enough variables in the state array */ | |
| 340 channels = IMA_ADPCM_state.wavefmt.channels; | |
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341 if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) { |
| 0 | 342 SDL_SetError("IMA ADPCM decoder can only handle %d channels", |
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343 SDL_arraysize(IMA_ADPCM_state.state)); |
| 0 | 344 return(-1); |
| 345 } | |
| 346 state = IMA_ADPCM_state.state; | |
| 347 | |
| 348 /* Allocate the proper sized output buffer */ | |
| 349 encoded_len = *audio_len; | |
| 350 encoded = *audio_buf; | |
| 351 freeable = *audio_buf; | |
| 352 *audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * | |
| 353 IMA_ADPCM_state.wSamplesPerBlock* | |
| 354 IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
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355 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 356 if ( *audio_buf == NULL ) { |
| 357 SDL_Error(SDL_ENOMEM); | |
| 358 return(-1); | |
| 359 } | |
| 360 decoded = *audio_buf; | |
| 361 | |
| 362 /* Get ready... Go! */ | |
| 363 while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) { | |
| 364 /* Grab the initial information for this block */ | |
| 365 for ( c=0; c<channels; ++c ) { | |
| 366 /* Fill the state information for this block */ | |
| 367 state[c].sample = ((encoded[1]<<8)|encoded[0]); | |
| 368 encoded += 2; | |
| 369 if ( state[c].sample & 0x8000 ) { | |
| 370 state[c].sample -= 0x10000; | |
| 371 } | |
| 372 state[c].index = *encoded++; | |
| 373 /* Reserved byte in buffer header, should be 0 */ | |
| 374 if ( *encoded++ != 0 ) { | |
| 375 /* Uh oh, corrupt data? Buggy code? */; | |
| 376 } | |
| 377 | |
| 378 /* Store the initial sample we start with */ | |
| 379 decoded[0] = state[c].sample&0xFF; | |
| 380 decoded[1] = state[c].sample>>8; | |
| 381 decoded += 2; | |
| 382 } | |
| 383 | |
| 384 /* Decode and store the other samples in this block */ | |
| 385 samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels; | |
| 386 while ( samplesleft > 0 ) { | |
| 387 for ( c=0; c<channels; ++c ) { | |
| 388 Fill_IMA_ADPCM_block(decoded, encoded, | |
| 389 c, channels, &state[c]); | |
| 390 encoded += 4; | |
| 391 samplesleft -= 8; | |
| 392 } | |
| 393 decoded += (channels * 8 * 2); | |
| 394 } | |
| 395 encoded_len -= IMA_ADPCM_state.wavefmt.blockalign; | |
| 396 } | |
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397 SDL_free(freeable); |
| 0 | 398 return(0); |
| 399 } | |
| 400 | |
| 401 SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, | |
| 402 SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) | |
| 403 { | |
| 404 int was_error; | |
| 405 Chunk chunk; | |
| 406 int lenread; | |
| 407 int MS_ADPCM_encoded, IMA_ADPCM_encoded; | |
| 408 int samplesize; | |
| 409 | |
| 410 /* WAV magic header */ | |
| 411 Uint32 RIFFchunk; | |
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412 Uint32 wavelen = 0; |
| 0 | 413 Uint32 WAVEmagic; |
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414 Uint32 headerDiff = 0; |
| 0 | 415 |
| 416 /* FMT chunk */ | |
| 417 WaveFMT *format = NULL; | |
| 418 | |
| 419 /* Make sure we are passed a valid data source */ | |
| 420 was_error = 0; | |
| 421 if ( src == NULL ) { | |
| 422 was_error = 1; | |
| 423 goto done; | |
| 424 } | |
| 425 | |
| 426 /* Check the magic header */ | |
| 427 RIFFchunk = SDL_ReadLE32(src); | |
| 428 wavelen = SDL_ReadLE32(src); | |
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429 if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */ |
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430 WAVEmagic = wavelen; |
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Handle the case where the WAVE magic number was already read in a non-seekable
Sam Lantinga <slouken@libsdl.org>
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431 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>
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432 RIFFchunk = RIFF; |
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Handle the case where the WAVE magic number was already read in a non-seekable
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433 } else { |
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434 WAVEmagic = SDL_ReadLE32(src); |
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435 } |
| 0 | 436 if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) { |
| 437 SDL_SetError("Unrecognized file type (not WAVE)"); | |
| 438 was_error = 1; | |
| 439 goto done; | |
| 440 } | |
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Date: 10 Jun 2003 15:30:59 -0400
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441 headerDiff += sizeof(Uint32); // for WAVE |
| 0 | 442 |
| 443 /* Read the audio data format chunk */ | |
| 444 chunk.data = NULL; | |
| 445 do { | |
| 446 if ( chunk.data != NULL ) { | |
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Use SDL_ prefixed versions of C library functions.
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447 SDL_free(chunk.data); |
| 0 | 448 } |
| 449 lenread = ReadChunk(src, &chunk); | |
| 450 if ( lenread < 0 ) { | |
| 451 was_error = 1; | |
| 452 goto done; | |
| 453 } | |
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Date: 10 Jun 2003 15:30:59 -0400
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454 // 2 Uint32's for chunk header+len, plus the lenread |
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80f8c94b5199
Date: 10 Jun 2003 15:30:59 -0400
Sam Lantinga <slouken@libsdl.org>
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455 headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 456 } while ( (chunk.magic == FACT) || (chunk.magic == LIST) ); |
| 457 | |
| 458 /* Decode the audio data format */ | |
| 459 format = (WaveFMT *)chunk.data; | |
| 460 if ( chunk.magic != FMT ) { | |
| 461 SDL_SetError("Complex WAVE files not supported"); | |
| 462 was_error = 1; | |
| 463 goto done; | |
| 464 } | |
| 465 MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; | |
| 466 switch (SDL_SwapLE16(format->encoding)) { | |
| 467 case PCM_CODE: | |
| 468 /* We can understand this */ | |
| 469 break; | |
| 470 case MS_ADPCM_CODE: | |
| 471 /* Try to understand this */ | |
| 472 if ( InitMS_ADPCM(format) < 0 ) { | |
| 473 was_error = 1; | |
| 474 goto done; | |
| 475 } | |
| 476 MS_ADPCM_encoded = 1; | |
| 477 break; | |
| 478 case IMA_ADPCM_CODE: | |
| 479 /* Try to understand this */ | |
| 480 if ( InitIMA_ADPCM(format) < 0 ) { | |
| 481 was_error = 1; | |
| 482 goto done; | |
| 483 } | |
| 484 IMA_ADPCM_encoded = 1; | |
| 485 break; | |
| 486 default: | |
| 487 SDL_SetError("Unknown WAVE data format: 0x%.4x", | |
| 488 SDL_SwapLE16(format->encoding)); | |
| 489 was_error = 1; | |
| 490 goto done; | |
| 491 } | |
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492 SDL_memset(spec, 0, (sizeof *spec)); |
| 0 | 493 spec->freq = SDL_SwapLE32(format->frequency); |
| 494 switch (SDL_SwapLE16(format->bitspersample)) { | |
| 495 case 4: | |
| 496 if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { | |
| 497 spec->format = AUDIO_S16; | |
| 498 } else { | |
| 499 was_error = 1; | |
| 500 } | |
| 501 break; | |
| 502 case 8: | |
| 503 spec->format = AUDIO_U8; | |
| 504 break; | |
| 505 case 16: | |
| 506 spec->format = AUDIO_S16; | |
| 507 break; | |
| 508 default: | |
| 509 was_error = 1; | |
| 510 break; | |
| 511 } | |
| 512 if ( was_error ) { | |
| 513 SDL_SetError("Unknown %d-bit PCM data format", | |
| 514 SDL_SwapLE16(format->bitspersample)); | |
| 515 goto done; | |
| 516 } | |
| 517 spec->channels = (Uint8)SDL_SwapLE16(format->channels); | |
| 518 spec->samples = 4096; /* Good default buffer size */ | |
| 519 | |
| 520 /* Read the audio data chunk */ | |
| 521 *audio_buf = NULL; | |
| 522 do { | |
| 523 if ( *audio_buf != NULL ) { | |
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524 SDL_free(*audio_buf); |
| 0 | 525 } |
| 526 lenread = ReadChunk(src, &chunk); | |
| 527 if ( lenread < 0 ) { | |
| 528 was_error = 1; | |
| 529 goto done; | |
| 530 } | |
| 531 *audio_len = lenread; | |
| 532 *audio_buf = chunk.data; | |
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533 if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 534 } while ( chunk.magic != DATA ); |
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535 headerDiff += 2 * sizeof(Uint32); // for the data chunk and len |
| 0 | 536 |
| 537 if ( MS_ADPCM_encoded ) { | |
| 538 if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 539 was_error = 1; | |
| 540 goto done; | |
| 541 } | |
| 542 } | |
| 543 if ( IMA_ADPCM_encoded ) { | |
| 544 if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 545 was_error = 1; | |
| 546 goto done; | |
| 547 } | |
| 548 } | |
| 549 | |
| 550 /* Don't return a buffer that isn't a multiple of samplesize */ | |
| 551 samplesize = ((spec->format & 0xFF)/8)*spec->channels; | |
| 552 *audio_len &= ~(samplesize-1); | |
| 553 | |
| 554 done: | |
| 555 if ( format != NULL ) { | |
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Use SDL_ prefixed versions of C library functions.
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556 SDL_free(format); |
| 0 | 557 } |
| 558 if ( freesrc && src ) { | |
| 559 SDL_RWclose(src); | |
| 560 } | |
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561 else { |
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Date: 10 Jun 2003 15:30:59 -0400
Sam Lantinga <slouken@libsdl.org>
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562 // seek to the end of the file (given by the RIFF chunk) |
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It's now possible to build SDL without any C runtime at all on Windows,
Sam Lantinga <slouken@libsdl.org>
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563 SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR); |
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564 } |
| 0 | 565 if ( was_error ) { |
| 566 spec = NULL; | |
| 567 } | |
| 568 return(spec); | |
| 569 } | |
| 570 | |
| 571 /* Since the WAV memory is allocated in the shared library, it must also | |
| 572 be freed here. (Necessary under Win32, VC++) | |
| 573 */ | |
| 574 void SDL_FreeWAV(Uint8 *audio_buf) | |
| 575 { | |
| 576 if ( audio_buf != NULL ) { | |
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Use SDL_ prefixed versions of C library functions.
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diff
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577 SDL_free(audio_buf); |
| 0 | 578 } |
| 579 } | |
| 580 | |
| 581 static int ReadChunk(SDL_RWops *src, Chunk *chunk) | |
| 582 { | |
| 583 chunk->magic = SDL_ReadLE32(src); | |
| 584 chunk->length = SDL_ReadLE32(src); | |
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585 chunk->data = (Uint8 *)SDL_malloc(chunk->length); |
| 0 | 586 if ( chunk->data == NULL ) { |
| 587 SDL_Error(SDL_ENOMEM); | |
| 588 return(-1); | |
| 589 } | |
| 590 if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { | |
| 591 SDL_Error(SDL_EFREAD); | |
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Sam Lantinga <slouken@libsdl.org>
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592 SDL_free(chunk->data); |
| 0 | 593 return(-1); |
| 594 } | |
| 595 return(chunk->length); | |
| 596 } | |
| 597 | |
| 598 #endif /* ENABLE_FILE */ |