Mercurial > sdl-ios-xcode
annotate src/audio/SDL_wave.c @ 1367:e440d5c488c1
Fixes for BeOS and Solaris builds
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Fri, 17 Feb 2006 08:43:23 +0000 |
| parents | 19418e4422cb |
| children | d910939febfa |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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Updated copyright information and removed rcs id lines (problematic in branch merges)
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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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Sam Lantinga <slouken@libsdl.org>
parents:
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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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Sam Lantinga <slouken@libsdl.org>
parents:
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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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Sam Lantinga <slouken@libsdl.org>
parents:
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changeset
|
13 Lesser General Public License for more details. |
| 0 | 14 |
|
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Sam Lantinga <slouken@libsdl.org>
parents:
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diff
changeset
|
15 You should have received a copy of the GNU Lesser General Public |
|
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
Sam Lantinga <slouken@libsdl.org>
parents:
1260
diff
changeset
|
16 License along with this library; if not, write to the Free Software |
|
c9b51268668f
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Sam Lantinga <slouken@libsdl.org>
parents:
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diff
changeset
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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 /* Microsoft WAVE file loading routines */ | |
| 24 | |
| 25 #include "SDL_audio.h" | |
| 26 #include "SDL_wave.h" | |
| 27 | |
| 28 | |
| 29 static int ReadChunk(SDL_RWops *src, Chunk *chunk); | |
| 30 | |
| 31 struct MS_ADPCM_decodestate { | |
| 32 Uint8 hPredictor; | |
| 33 Uint16 iDelta; | |
| 34 Sint16 iSamp1; | |
| 35 Sint16 iSamp2; | |
| 36 }; | |
| 37 static struct MS_ADPCM_decoder { | |
| 38 WaveFMT wavefmt; | |
| 39 Uint16 wSamplesPerBlock; | |
| 40 Uint16 wNumCoef; | |
| 41 Sint16 aCoeff[7][2]; | |
| 42 /* * * */ | |
| 43 struct MS_ADPCM_decodestate state[2]; | |
| 44 } MS_ADPCM_state; | |
| 45 | |
| 46 static int InitMS_ADPCM(WaveFMT *format) | |
| 47 { | |
| 48 Uint8 *rogue_feel; | |
| 49 Uint16 extra_info; | |
| 50 int i; | |
| 51 | |
| 52 /* Set the rogue pointer to the MS_ADPCM specific data */ | |
| 53 MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 54 MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 55 MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 56 MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 57 MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 58 MS_ADPCM_state.wavefmt.bitspersample = | |
| 59 SDL_SwapLE16(format->bitspersample); | |
| 60 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 61 if ( sizeof(*format) == 16 ) { | |
| 62 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 63 rogue_feel += sizeof(Uint16); | |
| 64 } | |
| 65 MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 66 rogue_feel += sizeof(Uint16); | |
| 67 MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 68 rogue_feel += sizeof(Uint16); | |
| 69 if ( MS_ADPCM_state.wNumCoef != 7 ) { | |
| 70 SDL_SetError("Unknown set of MS_ADPCM coefficients"); | |
| 71 return(-1); | |
| 72 } | |
| 73 for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) { | |
| 74 MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 75 rogue_feel += sizeof(Uint16); | |
| 76 MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 77 rogue_feel += sizeof(Uint16); | |
| 78 } | |
| 79 return(0); | |
| 80 } | |
| 81 | |
| 82 static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state, | |
| 83 Uint8 nybble, Sint16 *coeff) | |
| 84 { | |
| 85 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 86 const Sint32 min_audioval = -(1<<(16-1)); | |
| 87 const Sint32 adaptive[] = { | |
| 88 230, 230, 230, 230, 307, 409, 512, 614, | |
| 89 768, 614, 512, 409, 307, 230, 230, 230 | |
| 90 }; | |
| 91 Sint32 new_sample, delta; | |
| 92 | |
| 93 new_sample = ((state->iSamp1 * coeff[0]) + | |
| 94 (state->iSamp2 * coeff[1]))/256; | |
| 95 if ( nybble & 0x08 ) { | |
| 96 new_sample += state->iDelta * (nybble-0x10); | |
| 97 } else { | |
| 98 new_sample += state->iDelta * nybble; | |
| 99 } | |
| 100 if ( new_sample < min_audioval ) { | |
| 101 new_sample = min_audioval; | |
| 102 } else | |
| 103 if ( new_sample > max_audioval ) { | |
| 104 new_sample = max_audioval; | |
| 105 } | |
| 106 delta = ((Sint32)state->iDelta * adaptive[nybble])/256; | |
| 107 if ( delta < 16 ) { | |
| 108 delta = 16; | |
| 109 } | |
| 110 state->iDelta = delta; | |
| 111 state->iSamp2 = state->iSamp1; | |
| 112 state->iSamp1 = new_sample; | |
| 113 return(new_sample); | |
| 114 } | |
| 115 | |
| 116 static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 117 { | |
| 118 struct MS_ADPCM_decodestate *state[2]; | |
| 119 Uint8 *freeable, *encoded, *decoded; | |
| 120 Sint32 encoded_len, samplesleft; | |
| 121 Sint8 nybble, stereo; | |
| 122 Sint16 *coeff[2]; | |
| 123 Sint32 new_sample; | |
| 124 | |
| 125 /* Allocate the proper sized output buffer */ | |
| 126 encoded_len = *audio_len; | |
| 127 encoded = *audio_buf; | |
| 128 freeable = *audio_buf; | |
| 129 *audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * | |
| 130 MS_ADPCM_state.wSamplesPerBlock* | |
| 131 MS_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
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132 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 133 if ( *audio_buf == NULL ) { |
| 134 SDL_Error(SDL_ENOMEM); | |
| 135 return(-1); | |
| 136 } | |
| 137 decoded = *audio_buf; | |
| 138 | |
| 139 /* Get ready... Go! */ | |
| 140 stereo = (MS_ADPCM_state.wavefmt.channels == 2); | |
| 141 state[0] = &MS_ADPCM_state.state[0]; | |
| 142 state[1] = &MS_ADPCM_state.state[stereo]; | |
| 143 while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) { | |
| 144 /* Grab the initial information for this block */ | |
| 145 state[0]->hPredictor = *encoded++; | |
| 146 if ( stereo ) { | |
| 147 state[1]->hPredictor = *encoded++; | |
| 148 } | |
| 149 state[0]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 150 encoded += sizeof(Sint16); | |
| 151 if ( stereo ) { | |
| 152 state[1]->iDelta = ((encoded[1]<<8)|encoded[0]); | |
| 153 encoded += sizeof(Sint16); | |
| 154 } | |
| 155 state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 156 encoded += sizeof(Sint16); | |
| 157 if ( stereo ) { | |
| 158 state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]); | |
| 159 encoded += sizeof(Sint16); | |
| 160 } | |
| 161 state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 162 encoded += sizeof(Sint16); | |
| 163 if ( stereo ) { | |
| 164 state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]); | |
| 165 encoded += sizeof(Sint16); | |
| 166 } | |
| 167 coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor]; | |
| 168 coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor]; | |
| 169 | |
| 170 /* Store the two initial samples we start with */ | |
| 171 decoded[0] = state[0]->iSamp2&0xFF; | |
| 172 decoded[1] = state[0]->iSamp2>>8; | |
| 173 decoded += 2; | |
| 174 if ( stereo ) { | |
| 175 decoded[0] = state[1]->iSamp2&0xFF; | |
| 176 decoded[1] = state[1]->iSamp2>>8; | |
| 177 decoded += 2; | |
| 178 } | |
| 179 decoded[0] = state[0]->iSamp1&0xFF; | |
| 180 decoded[1] = state[0]->iSamp1>>8; | |
| 181 decoded += 2; | |
| 182 if ( stereo ) { | |
| 183 decoded[0] = state[1]->iSamp1&0xFF; | |
| 184 decoded[1] = state[1]->iSamp1>>8; | |
| 185 decoded += 2; | |
| 186 } | |
| 187 | |
| 188 /* Decode and store the other samples in this block */ | |
| 189 samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)* | |
| 190 MS_ADPCM_state.wavefmt.channels; | |
| 191 while ( samplesleft > 0 ) { | |
| 192 nybble = (*encoded)>>4; | |
| 193 new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]); | |
| 194 decoded[0] = new_sample&0xFF; | |
| 195 new_sample >>= 8; | |
| 196 decoded[1] = new_sample&0xFF; | |
| 197 decoded += 2; | |
| 198 | |
| 199 nybble = (*encoded)&0x0F; | |
| 200 new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]); | |
| 201 decoded[0] = new_sample&0xFF; | |
| 202 new_sample >>= 8; | |
| 203 decoded[1] = new_sample&0xFF; | |
| 204 decoded += 2; | |
| 205 | |
| 206 ++encoded; | |
| 207 samplesleft -= 2; | |
| 208 } | |
| 209 encoded_len -= MS_ADPCM_state.wavefmt.blockalign; | |
| 210 } | |
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211 SDL_free(freeable); |
| 0 | 212 return(0); |
| 213 } | |
| 214 | |
| 215 struct IMA_ADPCM_decodestate { | |
| 216 Sint32 sample; | |
| 217 Sint8 index; | |
| 218 }; | |
| 219 static struct IMA_ADPCM_decoder { | |
| 220 WaveFMT wavefmt; | |
| 221 Uint16 wSamplesPerBlock; | |
| 222 /* * * */ | |
| 223 struct IMA_ADPCM_decodestate state[2]; | |
| 224 } IMA_ADPCM_state; | |
| 225 | |
| 226 static int InitIMA_ADPCM(WaveFMT *format) | |
| 227 { | |
| 228 Uint8 *rogue_feel; | |
| 229 Uint16 extra_info; | |
| 230 | |
| 231 /* Set the rogue pointer to the IMA_ADPCM specific data */ | |
| 232 IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); | |
| 233 IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); | |
| 234 IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); | |
| 235 IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); | |
| 236 IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); | |
| 237 IMA_ADPCM_state.wavefmt.bitspersample = | |
| 238 SDL_SwapLE16(format->bitspersample); | |
| 239 rogue_feel = (Uint8 *)format+sizeof(*format); | |
| 240 if ( sizeof(*format) == 16 ) { | |
| 241 extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 242 rogue_feel += sizeof(Uint16); | |
| 243 } | |
| 244 IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); | |
| 245 return(0); | |
| 246 } | |
| 247 | |
| 248 static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble) | |
| 249 { | |
| 250 const Sint32 max_audioval = ((1<<(16-1))-1); | |
| 251 const Sint32 min_audioval = -(1<<(16-1)); | |
| 252 const int index_table[16] = { | |
| 253 -1, -1, -1, -1, | |
| 254 2, 4, 6, 8, | |
| 255 -1, -1, -1, -1, | |
| 256 2, 4, 6, 8 | |
| 257 }; | |
| 258 const Sint32 step_table[89] = { | |
| 259 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, | |
| 260 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, | |
| 261 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, | |
| 262 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, | |
| 263 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, | |
| 264 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, | |
| 265 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, | |
| 266 22385, 24623, 27086, 29794, 32767 | |
| 267 }; | |
| 268 Sint32 delta, step; | |
| 269 | |
| 270 /* Compute difference and new sample value */ | |
| 271 step = step_table[state->index]; | |
| 272 delta = step >> 3; | |
| 273 if ( nybble & 0x04 ) delta += step; | |
| 274 if ( nybble & 0x02 ) delta += (step >> 1); | |
| 275 if ( nybble & 0x01 ) delta += (step >> 2); | |
| 276 if ( nybble & 0x08 ) delta = -delta; | |
| 277 state->sample += delta; | |
| 278 | |
| 279 /* Update index value */ | |
| 280 state->index += index_table[nybble]; | |
| 281 if ( state->index > 88 ) { | |
| 282 state->index = 88; | |
| 283 } else | |
| 284 if ( state->index < 0 ) { | |
| 285 state->index = 0; | |
| 286 } | |
| 287 | |
| 288 /* Clamp output sample */ | |
| 289 if ( state->sample > max_audioval ) { | |
| 290 state->sample = max_audioval; | |
| 291 } else | |
| 292 if ( state->sample < min_audioval ) { | |
| 293 state->sample = min_audioval; | |
| 294 } | |
| 295 return(state->sample); | |
| 296 } | |
| 297 | |
| 298 /* Fill the decode buffer with a channel block of data (8 samples) */ | |
| 299 static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded, | |
| 300 int channel, int numchannels, struct IMA_ADPCM_decodestate *state) | |
| 301 { | |
| 302 int i; | |
| 303 Sint8 nybble; | |
| 304 Sint32 new_sample; | |
| 305 | |
| 306 decoded += (channel * 2); | |
| 307 for ( i=0; i<4; ++i ) { | |
| 308 nybble = (*encoded)&0x0F; | |
| 309 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 310 decoded[0] = new_sample&0xFF; | |
| 311 new_sample >>= 8; | |
| 312 decoded[1] = new_sample&0xFF; | |
| 313 decoded += 2 * numchannels; | |
| 314 | |
| 315 nybble = (*encoded)>>4; | |
| 316 new_sample = IMA_ADPCM_nibble(state, nybble); | |
| 317 decoded[0] = new_sample&0xFF; | |
| 318 new_sample >>= 8; | |
| 319 decoded[1] = new_sample&0xFF; | |
| 320 decoded += 2 * numchannels; | |
| 321 | |
| 322 ++encoded; | |
| 323 } | |
| 324 } | |
| 325 | |
| 326 static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) | |
| 327 { | |
| 328 struct IMA_ADPCM_decodestate *state; | |
| 329 Uint8 *freeable, *encoded, *decoded; | |
| 330 Sint32 encoded_len, samplesleft; | |
| 331 int c, channels; | |
| 332 | |
| 333 /* Check to make sure we have enough variables in the state array */ | |
| 334 channels = IMA_ADPCM_state.wavefmt.channels; | |
|
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335 if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) { |
| 0 | 336 SDL_SetError("IMA ADPCM decoder can only handle %d channels", |
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337 SDL_arraysize(IMA_ADPCM_state.state)); |
| 0 | 338 return(-1); |
| 339 } | |
| 340 state = IMA_ADPCM_state.state; | |
| 341 | |
| 342 /* Allocate the proper sized output buffer */ | |
| 343 encoded_len = *audio_len; | |
| 344 encoded = *audio_buf; | |
| 345 freeable = *audio_buf; | |
| 346 *audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * | |
| 347 IMA_ADPCM_state.wSamplesPerBlock* | |
| 348 IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16); | |
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parents:
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diff
changeset
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349 *audio_buf = (Uint8 *)SDL_malloc(*audio_len); |
| 0 | 350 if ( *audio_buf == NULL ) { |
| 351 SDL_Error(SDL_ENOMEM); | |
| 352 return(-1); | |
| 353 } | |
| 354 decoded = *audio_buf; | |
| 355 | |
| 356 /* Get ready... Go! */ | |
| 357 while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) { | |
| 358 /* Grab the initial information for this block */ | |
| 359 for ( c=0; c<channels; ++c ) { | |
| 360 /* Fill the state information for this block */ | |
| 361 state[c].sample = ((encoded[1]<<8)|encoded[0]); | |
| 362 encoded += 2; | |
| 363 if ( state[c].sample & 0x8000 ) { | |
| 364 state[c].sample -= 0x10000; | |
| 365 } | |
| 366 state[c].index = *encoded++; | |
| 367 /* Reserved byte in buffer header, should be 0 */ | |
| 368 if ( *encoded++ != 0 ) { | |
| 369 /* Uh oh, corrupt data? Buggy code? */; | |
| 370 } | |
| 371 | |
| 372 /* Store the initial sample we start with */ | |
| 373 decoded[0] = state[c].sample&0xFF; | |
| 374 decoded[1] = state[c].sample>>8; | |
| 375 decoded += 2; | |
| 376 } | |
| 377 | |
| 378 /* Decode and store the other samples in this block */ | |
| 379 samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels; | |
| 380 while ( samplesleft > 0 ) { | |
| 381 for ( c=0; c<channels; ++c ) { | |
| 382 Fill_IMA_ADPCM_block(decoded, encoded, | |
| 383 c, channels, &state[c]); | |
| 384 encoded += 4; | |
| 385 samplesleft -= 8; | |
| 386 } | |
| 387 decoded += (channels * 8 * 2); | |
| 388 } | |
| 389 encoded_len -= IMA_ADPCM_state.wavefmt.blockalign; | |
| 390 } | |
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|
391 SDL_free(freeable); |
| 0 | 392 return(0); |
| 393 } | |
| 394 | |
| 395 SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, | |
| 396 SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) | |
| 397 { | |
| 398 int was_error; | |
| 399 Chunk chunk; | |
| 400 int lenread; | |
| 401 int MS_ADPCM_encoded, IMA_ADPCM_encoded; | |
| 402 int samplesize; | |
| 403 | |
| 404 /* WAV magic header */ | |
| 405 Uint32 RIFFchunk; | |
|
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406 Uint32 wavelen = 0; |
| 0 | 407 Uint32 WAVEmagic; |
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408 Uint32 headerDiff = 0; |
| 0 | 409 |
| 410 /* FMT chunk */ | |
| 411 WaveFMT *format = NULL; | |
| 412 | |
| 413 /* Make sure we are passed a valid data source */ | |
| 414 was_error = 0; | |
| 415 if ( src == NULL ) { | |
| 416 was_error = 1; | |
| 417 goto done; | |
| 418 } | |
| 419 | |
| 420 /* Check the magic header */ | |
| 421 RIFFchunk = SDL_ReadLE32(src); | |
| 422 wavelen = SDL_ReadLE32(src); | |
|
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423 if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */ |
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|
424 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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changeset
|
425 wavelen = RIFFchunk; |
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Handle the case where the WAVE magic number was already read in a non-seekable
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|
426 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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427 } else { |
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428 WAVEmagic = SDL_ReadLE32(src); |
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429 } |
| 0 | 430 if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) { |
| 431 SDL_SetError("Unrecognized file type (not WAVE)"); | |
| 432 was_error = 1; | |
| 433 goto done; | |
| 434 } | |
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435 headerDiff += sizeof(Uint32); // for WAVE |
| 0 | 436 |
| 437 /* Read the audio data format chunk */ | |
| 438 chunk.data = NULL; | |
| 439 do { | |
| 440 if ( chunk.data != NULL ) { | |
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441 SDL_free(chunk.data); |
| 0 | 442 } |
| 443 lenread = ReadChunk(src, &chunk); | |
| 444 if ( lenread < 0 ) { | |
| 445 was_error = 1; | |
| 446 goto done; | |
| 447 } | |
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448 // 2 Uint32's for chunk header+len, plus the lenread |
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449 headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 450 } while ( (chunk.magic == FACT) || (chunk.magic == LIST) ); |
| 451 | |
| 452 /* Decode the audio data format */ | |
| 453 format = (WaveFMT *)chunk.data; | |
| 454 if ( chunk.magic != FMT ) { | |
| 455 SDL_SetError("Complex WAVE files not supported"); | |
| 456 was_error = 1; | |
| 457 goto done; | |
| 458 } | |
| 459 MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; | |
| 460 switch (SDL_SwapLE16(format->encoding)) { | |
| 461 case PCM_CODE: | |
| 462 /* We can understand this */ | |
| 463 break; | |
| 464 case MS_ADPCM_CODE: | |
| 465 /* Try to understand this */ | |
| 466 if ( InitMS_ADPCM(format) < 0 ) { | |
| 467 was_error = 1; | |
| 468 goto done; | |
| 469 } | |
| 470 MS_ADPCM_encoded = 1; | |
| 471 break; | |
| 472 case IMA_ADPCM_CODE: | |
| 473 /* Try to understand this */ | |
| 474 if ( InitIMA_ADPCM(format) < 0 ) { | |
| 475 was_error = 1; | |
| 476 goto done; | |
| 477 } | |
| 478 IMA_ADPCM_encoded = 1; | |
| 479 break; | |
| 480 default: | |
| 481 SDL_SetError("Unknown WAVE data format: 0x%.4x", | |
| 482 SDL_SwapLE16(format->encoding)); | |
| 483 was_error = 1; | |
| 484 goto done; | |
| 485 } | |
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486 SDL_memset(spec, 0, (sizeof *spec)); |
| 0 | 487 spec->freq = SDL_SwapLE32(format->frequency); |
| 488 switch (SDL_SwapLE16(format->bitspersample)) { | |
| 489 case 4: | |
| 490 if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { | |
| 491 spec->format = AUDIO_S16; | |
| 492 } else { | |
| 493 was_error = 1; | |
| 494 } | |
| 495 break; | |
| 496 case 8: | |
| 497 spec->format = AUDIO_U8; | |
| 498 break; | |
| 499 case 16: | |
| 500 spec->format = AUDIO_S16; | |
| 501 break; | |
| 502 default: | |
| 503 was_error = 1; | |
| 504 break; | |
| 505 } | |
| 506 if ( was_error ) { | |
| 507 SDL_SetError("Unknown %d-bit PCM data format", | |
| 508 SDL_SwapLE16(format->bitspersample)); | |
| 509 goto done; | |
| 510 } | |
| 511 spec->channels = (Uint8)SDL_SwapLE16(format->channels); | |
| 512 spec->samples = 4096; /* Good default buffer size */ | |
| 513 | |
| 514 /* Read the audio data chunk */ | |
| 515 *audio_buf = NULL; | |
| 516 do { | |
| 517 if ( *audio_buf != NULL ) { | |
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518 SDL_free(*audio_buf); |
| 0 | 519 } |
| 520 lenread = ReadChunk(src, &chunk); | |
| 521 if ( lenread < 0 ) { | |
| 522 was_error = 1; | |
| 523 goto done; | |
| 524 } | |
| 525 *audio_len = lenread; | |
| 526 *audio_buf = chunk.data; | |
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527 if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32); |
| 0 | 528 } while ( chunk.magic != DATA ); |
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529 headerDiff += 2 * sizeof(Uint32); // for the data chunk and len |
| 0 | 530 |
| 531 if ( MS_ADPCM_encoded ) { | |
| 532 if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 533 was_error = 1; | |
| 534 goto done; | |
| 535 } | |
| 536 } | |
| 537 if ( IMA_ADPCM_encoded ) { | |
| 538 if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { | |
| 539 was_error = 1; | |
| 540 goto done; | |
| 541 } | |
| 542 } | |
| 543 | |
| 544 /* Don't return a buffer that isn't a multiple of samplesize */ | |
| 545 samplesize = ((spec->format & 0xFF)/8)*spec->channels; | |
| 546 *audio_len &= ~(samplesize-1); | |
| 547 | |
| 548 done: | |
| 549 if ( format != NULL ) { | |
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550 SDL_free(format); |
| 0 | 551 } |
| 552 if ( freesrc && src ) { | |
| 553 SDL_RWclose(src); | |
| 554 } | |
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555 else { |
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Date: 10 Jun 2003 15:30:59 -0400
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556 // 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,
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557 SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR); |
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558 } |
| 0 | 559 if ( was_error ) { |
| 560 spec = NULL; | |
| 561 } | |
| 562 return(spec); | |
| 563 } | |
| 564 | |
| 565 /* Since the WAV memory is allocated in the shared library, it must also | |
| 566 be freed here. (Necessary under Win32, VC++) | |
| 567 */ | |
| 568 void SDL_FreeWAV(Uint8 *audio_buf) | |
| 569 { | |
| 570 if ( audio_buf != NULL ) { | |
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|
571 SDL_free(audio_buf); |
| 0 | 572 } |
| 573 } | |
| 574 | |
| 575 static int ReadChunk(SDL_RWops *src, Chunk *chunk) | |
| 576 { | |
| 577 chunk->magic = SDL_ReadLE32(src); | |
| 578 chunk->length = SDL_ReadLE32(src); | |
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579 chunk->data = (Uint8 *)SDL_malloc(chunk->length); |
| 0 | 580 if ( chunk->data == NULL ) { |
| 581 SDL_Error(SDL_ENOMEM); | |
| 582 return(-1); | |
| 583 } | |
| 584 if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { | |
| 585 SDL_Error(SDL_EFREAD); | |
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586 SDL_free(chunk->data); |
| 0 | 587 return(-1); |
| 588 } | |
| 589 return(chunk->length); | |
| 590 } |