Mercurial > sdl-ios-xcode
annotate src/audio/SDL_wave.c @ 1643:51038e80ae59
More general fix for bug #189
The clipping is done at a higher level, and the low level functions are
passed clipped rectangles. Drivers which don't support source clipping
have not been changed, so the image will be squished instead of clipped,
but at least they will no longer crash when the destination rect was out
of bounds.
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Mon, 17 Apr 2006 06:47:23 +0000 |
| parents | 97d0966f4bf7 |
| children | 14717b52abc0 |
| 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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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 */ |
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22 #include "SDL_config.h" |
| 0 | 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 } | |
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111 state->iDelta = (Uint16)delta; |
| 0 | 112 state->iSamp2 = state->iSamp1; |
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113 state->iSamp1 = (Sint16)new_sample; |
| 0 | 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); | |
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133 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 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 } | |
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212 SDL_free(freeable); |
| 0 | 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; | |
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332 unsigned int c, channels; |
| 0 | 333 |
| 334 /* Check to make sure we have enough variables in the state array */ | |
| 335 channels = IMA_ADPCM_state.wavefmt.channels; | |
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336 if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) { |
| 0 | 337 SDL_SetError("IMA ADPCM decoder can only handle %d channels", |
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338 SDL_arraysize(IMA_ADPCM_state.state)); |
| 0 | 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); | |
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350 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 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 */ | |
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374 decoded[0] = (Uint8)(state[c].sample&0xFF); |
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375 decoded[1] = (Uint8)(state[c].sample>>8); |
| 0 | 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 } | |
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392 SDL_free(freeable); |
| 0 | 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; | |
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407 Uint32 wavelen = 0; |
| 0 | 408 Uint32 WAVEmagic; |
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409 Uint32 headerDiff = 0; |
| 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); | |
|
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424 if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */ |
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425 WAVEmagic = wavelen; |
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426 wavelen = RIFFchunk; |
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427 RIFFchunk = RIFF; |
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428 } else { |
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429 WAVEmagic = SDL_ReadLE32(src); |
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430 } |
| 0 | 431 if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) { |
| 432 SDL_SetError("Unrecognized file type (not WAVE)"); | |
| 433 was_error = 1; | |
| 434 goto done; | |
| 435 } | |
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436 headerDiff += sizeof(Uint32); /* for WAVE */ |
| 0 | 437 |
| 438 /* Read the audio data format chunk */ | |
| 439 chunk.data = NULL; | |
| 440 do { | |
| 441 if ( chunk.data != NULL ) { | |
|
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442 SDL_free(chunk.data); |
| 0 | 443 } |
| 444 lenread = ReadChunk(src, &chunk); | |
| 445 if ( lenread < 0 ) { | |
| 446 was_error = 1; | |
| 447 goto done; | |
| 448 } | |
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449 /* 2 Uint32's for chunk header+len, plus the lenread */ |
|
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450 headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 451 } while ( (chunk.magic == FACT) || (chunk.magic == LIST) ); |
| 452 | |
| 453 /* Decode the audio data format */ | |
| 454 format = (WaveFMT *)chunk.data; | |
| 455 if ( chunk.magic != FMT ) { | |
| 456 SDL_SetError("Complex WAVE files not supported"); | |
| 457 was_error = 1; | |
| 458 goto done; | |
| 459 } | |
| 460 MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; | |
| 461 switch (SDL_SwapLE16(format->encoding)) { | |
| 462 case PCM_CODE: | |
| 463 /* We can understand this */ | |
| 464 break; | |
| 465 case MS_ADPCM_CODE: | |
| 466 /* Try to understand this */ | |
| 467 if ( InitMS_ADPCM(format) < 0 ) { | |
| 468 was_error = 1; | |
| 469 goto done; | |
| 470 } | |
| 471 MS_ADPCM_encoded = 1; | |
| 472 break; | |
| 473 case IMA_ADPCM_CODE: | |
| 474 /* Try to understand this */ | |
| 475 if ( InitIMA_ADPCM(format) < 0 ) { | |
| 476 was_error = 1; | |
| 477 goto done; | |
| 478 } | |
| 479 IMA_ADPCM_encoded = 1; | |
| 480 break; | |
| 481 default: | |
| 482 SDL_SetError("Unknown WAVE data format: 0x%.4x", | |
| 483 SDL_SwapLE16(format->encoding)); | |
| 484 was_error = 1; | |
| 485 goto done; | |
| 486 } | |
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487 SDL_memset(spec, 0, (sizeof *spec)); |
| 0 | 488 spec->freq = SDL_SwapLE32(format->frequency); |
| 489 switch (SDL_SwapLE16(format->bitspersample)) { | |
| 490 case 4: | |
| 491 if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { | |
| 492 spec->format = AUDIO_S16; | |
| 493 } else { | |
| 494 was_error = 1; | |
| 495 } | |
| 496 break; | |
| 497 case 8: | |
| 498 spec->format = AUDIO_U8; | |
| 499 break; | |
| 500 case 16: | |
| 501 spec->format = AUDIO_S16; | |
| 502 break; | |
| 503 default: | |
| 504 was_error = 1; | |
| 505 break; | |
| 506 } | |
| 507 if ( was_error ) { | |
| 508 SDL_SetError("Unknown %d-bit PCM data format", | |
| 509 SDL_SwapLE16(format->bitspersample)); | |
| 510 goto done; | |
| 511 } | |
| 512 spec->channels = (Uint8)SDL_SwapLE16(format->channels); | |
| 513 spec->samples = 4096; /* Good default buffer size */ | |
| 514 | |
| 515 /* Read the audio data chunk */ | |
| 516 *audio_buf = NULL; | |
| 517 do { | |
| 518 if ( *audio_buf != NULL ) { | |
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519 SDL_free(*audio_buf); |
| 0 | 520 } |
| 521 lenread = ReadChunk(src, &chunk); | |
| 522 if ( lenread < 0 ) { | |
| 523 was_error = 1; | |
| 524 goto done; | |
| 525 } | |
| 526 *audio_len = lenread; | |
| 527 *audio_buf = chunk.data; | |
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528 if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 529 } while ( chunk.magic != DATA ); |
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530 headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */ |
| 0 | 531 |
| 532 if ( MS_ADPCM_encoded ) { | |
| 533 if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 534 was_error = 1; | |
| 535 goto done; | |
| 536 } | |
| 537 } | |
| 538 if ( IMA_ADPCM_encoded ) { | |
| 539 if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 540 was_error = 1; | |
| 541 goto done; | |
| 542 } | |
| 543 } | |
| 544 | |
| 545 /* Don't return a buffer that isn't a multiple of samplesize */ | |
| 546 samplesize = ((spec->format & 0xFF)/8)*spec->channels; | |
| 547 *audio_len &= ~(samplesize-1); | |
| 548 | |
| 549 done: | |
| 550 if ( format != NULL ) { | |
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551 SDL_free(format); |
| 0 | 552 } |
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553 if ( src ) { |
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554 if ( freesrc ) { |
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555 SDL_RWclose(src); |
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556 } else { |
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557 /* seek to the end of the file (given by the RIFF chunk) */ |
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558 SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR); |
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559 } |
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560 } |
| 0 | 561 if ( was_error ) { |
| 562 spec = NULL; | |
| 563 } | |
| 564 return(spec); | |
| 565 } | |
| 566 | |
| 567 /* Since the WAV memory is allocated in the shared library, it must also | |
| 568 be freed here. (Necessary under Win32, VC++) | |
| 569 */ | |
| 570 void SDL_FreeWAV(Uint8 *audio_buf) | |
| 571 { | |
| 572 if ( audio_buf != NULL ) { | |
|
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573 SDL_free(audio_buf); |
| 0 | 574 } |
| 575 } | |
| 576 | |
| 577 static int ReadChunk(SDL_RWops *src, Chunk *chunk) | |
| 578 { | |
| 579 chunk->magic = SDL_ReadLE32(src); | |
| 580 chunk->length = SDL_ReadLE32(src); | |
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581 chunk->data = (Uint8 *)SDL_malloc(chunk->length); |
| 0 | 582 if ( chunk->data == NULL ) { |
| 583 SDL_Error(SDL_ENOMEM); | |
| 584 return(-1); | |
| 585 } | |
| 586 if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { | |
| 587 SDL_Error(SDL_EFREAD); | |
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588 SDL_free(chunk->data); |
| 0 | 589 return(-1); |
| 590 } | |
| 591 return(chunk->length); | |
| 592 } |