Mercurial > sdl-ios-xcode
annotate src/audio/SDL_audiocvt.c @ 844:cd0c77df70a4
*** empty log message ***
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Wed, 18 Feb 2004 03:57:13 +0000 |
| parents | b8d311d90021 |
| children | 41a59de7f2ed |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
769
b8d311d90021
Updated copyright information for 2004 (Happy New Year!)
Sam Lantinga <slouken@libsdl.org>
parents:
297
diff
changeset
|
3 Copyright (C) 1997-2004 Sam Lantinga |
| 0 | 4 |
| 5 This library is free software; you can redistribute it and/or | |
| 6 modify it under the terms of the GNU Library General Public | |
| 7 License as published by the Free Software Foundation; either | |
| 8 version 2 of the License, or (at your option) any later version. | |
| 9 | |
| 10 This library is distributed in the hope that it will be useful, | |
| 11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
| 13 Library General Public License for more details. | |
| 14 | |
| 15 You should have received a copy of the GNU Library General Public | |
| 16 License along with this library; if not, write to the Free | |
| 17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
| 18 | |
| 19 Sam Lantinga | |
|
252
e8157fcb3114
Updated the source with the correct e-mail address
Sam Lantinga <slouken@libsdl.org>
parents:
0
diff
changeset
|
20 slouken@libsdl.org |
| 0 | 21 */ |
| 22 | |
| 23 #ifdef SAVE_RCSID | |
| 24 static char rcsid = | |
| 25 "@(#) $Id$"; | |
| 26 #endif | |
| 27 | |
| 28 /* Functions for audio drivers to perform runtime conversion of audio format */ | |
| 29 | |
| 30 #include <stdio.h> | |
| 31 | |
| 32 #include "SDL_error.h" | |
| 33 #include "SDL_audio.h" | |
| 34 | |
| 35 | |
| 36 /* Effectively mix right and left channels into a single channel */ | |
| 37 void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format) | |
| 38 { | |
| 39 int i; | |
| 40 Sint32 sample; | |
| 41 | |
| 42 #ifdef DEBUG_CONVERT | |
| 43 fprintf(stderr, "Converting to mono\n"); | |
| 44 #endif | |
| 45 switch (format&0x8018) { | |
| 46 | |
| 47 case AUDIO_U8: { | |
| 48 Uint8 *src, *dst; | |
| 49 | |
| 50 src = cvt->buf; | |
| 51 dst = cvt->buf; | |
| 52 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 53 sample = src[0] + src[1]; | |
| 54 if ( sample > 255 ) { | |
| 55 *dst = 255; | |
| 56 } else { | |
| 57 *dst = sample; | |
| 58 } | |
| 59 src += 2; | |
| 60 dst += 1; | |
| 61 } | |
| 62 } | |
| 63 break; | |
| 64 | |
| 65 case AUDIO_S8: { | |
| 66 Sint8 *src, *dst; | |
| 67 | |
| 68 src = (Sint8 *)cvt->buf; | |
| 69 dst = (Sint8 *)cvt->buf; | |
| 70 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 71 sample = src[0] + src[1]; | |
| 72 if ( sample > 127 ) { | |
| 73 *dst = 127; | |
| 74 } else | |
| 75 if ( sample < -128 ) { | |
| 76 *dst = -128; | |
| 77 } else { | |
| 78 *dst = sample; | |
| 79 } | |
| 80 src += 2; | |
| 81 dst += 1; | |
| 82 } | |
| 83 } | |
| 84 break; | |
| 85 | |
| 86 case AUDIO_U16: { | |
| 87 Uint8 *src, *dst; | |
| 88 | |
| 89 src = cvt->buf; | |
| 90 dst = cvt->buf; | |
| 91 if ( (format & 0x1000) == 0x1000 ) { | |
| 92 for ( i=cvt->len_cvt/4; i; --i ) { | |
| 93 sample = (Uint16)((src[0]<<8)|src[1])+ | |
| 94 (Uint16)((src[2]<<8)|src[3]); | |
| 95 if ( sample > 65535 ) { | |
| 96 dst[0] = 0xFF; | |
| 97 dst[1] = 0xFF; | |
| 98 } else { | |
| 99 dst[1] = (sample&0xFF); | |
| 100 sample >>= 8; | |
| 101 dst[0] = (sample&0xFF); | |
| 102 } | |
| 103 src += 4; | |
| 104 dst += 2; | |
| 105 } | |
| 106 } else { | |
| 107 for ( i=cvt->len_cvt/4; i; --i ) { | |
| 108 sample = (Uint16)((src[1]<<8)|src[0])+ | |
| 109 (Uint16)((src[3]<<8)|src[2]); | |
| 110 if ( sample > 65535 ) { | |
| 111 dst[0] = 0xFF; | |
| 112 dst[1] = 0xFF; | |
| 113 } else { | |
| 114 dst[0] = (sample&0xFF); | |
| 115 sample >>= 8; | |
| 116 dst[1] = (sample&0xFF); | |
| 117 } | |
| 118 src += 4; | |
| 119 dst += 2; | |
| 120 } | |
| 121 } | |
| 122 } | |
| 123 break; | |
| 124 | |
| 125 case AUDIO_S16: { | |
| 126 Uint8 *src, *dst; | |
| 127 | |
| 128 src = cvt->buf; | |
| 129 dst = cvt->buf; | |
| 130 if ( (format & 0x1000) == 0x1000 ) { | |
| 131 for ( i=cvt->len_cvt/4; i; --i ) { | |
| 132 sample = (Sint16)((src[0]<<8)|src[1])+ | |
| 133 (Sint16)((src[2]<<8)|src[3]); | |
| 134 if ( sample > 32767 ) { | |
| 135 dst[0] = 0x7F; | |
| 136 dst[1] = 0xFF; | |
| 137 } else | |
| 138 if ( sample < -32768 ) { | |
| 139 dst[0] = 0x80; | |
| 140 dst[1] = 0x00; | |
| 141 } else { | |
| 142 dst[1] = (sample&0xFF); | |
| 143 sample >>= 8; | |
| 144 dst[0] = (sample&0xFF); | |
| 145 } | |
| 146 src += 4; | |
| 147 dst += 2; | |
| 148 } | |
| 149 } else { | |
| 150 for ( i=cvt->len_cvt/4; i; --i ) { | |
| 151 sample = (Sint16)((src[1]<<8)|src[0])+ | |
| 152 (Sint16)((src[3]<<8)|src[2]); | |
| 153 if ( sample > 32767 ) { | |
| 154 dst[1] = 0x7F; | |
| 155 dst[0] = 0xFF; | |
| 156 } else | |
| 157 if ( sample < -32768 ) { | |
| 158 dst[1] = 0x80; | |
| 159 dst[0] = 0x00; | |
| 160 } else { | |
| 161 dst[0] = (sample&0xFF); | |
| 162 sample >>= 8; | |
| 163 dst[1] = (sample&0xFF); | |
| 164 } | |
| 165 src += 4; | |
| 166 dst += 2; | |
| 167 } | |
| 168 } | |
| 169 } | |
| 170 break; | |
| 171 } | |
| 172 cvt->len_cvt /= 2; | |
| 173 if ( cvt->filters[++cvt->filter_index] ) { | |
| 174 cvt->filters[cvt->filter_index](cvt, format); | |
| 175 } | |
| 176 } | |
| 177 | |
| 178 | |
| 179 /* Duplicate a mono channel to both stereo channels */ | |
| 180 void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format) | |
| 181 { | |
| 182 int i; | |
| 183 | |
| 184 #ifdef DEBUG_CONVERT | |
| 185 fprintf(stderr, "Converting to stereo\n"); | |
| 186 #endif | |
| 187 if ( (format & 0xFF) == 16 ) { | |
| 188 Uint16 *src, *dst; | |
| 189 | |
| 190 src = (Uint16 *)(cvt->buf+cvt->len_cvt); | |
| 191 dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2); | |
| 192 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 193 dst -= 2; | |
| 194 src -= 1; | |
| 195 dst[0] = src[0]; | |
| 196 dst[1] = src[0]; | |
| 197 } | |
| 198 } else { | |
| 199 Uint8 *src, *dst; | |
| 200 | |
| 201 src = cvt->buf+cvt->len_cvt; | |
| 202 dst = cvt->buf+cvt->len_cvt*2; | |
| 203 for ( i=cvt->len_cvt; i; --i ) { | |
| 204 dst -= 2; | |
| 205 src -= 1; | |
| 206 dst[0] = src[0]; | |
| 207 dst[1] = src[0]; | |
| 208 } | |
| 209 } | |
| 210 cvt->len_cvt *= 2; | |
| 211 if ( cvt->filters[++cvt->filter_index] ) { | |
| 212 cvt->filters[cvt->filter_index](cvt, format); | |
| 213 } | |
| 214 } | |
| 215 | |
| 216 /* Convert 8-bit to 16-bit - LSB */ | |
| 217 void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format) | |
| 218 { | |
| 219 int i; | |
| 220 Uint8 *src, *dst; | |
| 221 | |
| 222 #ifdef DEBUG_CONVERT | |
| 223 fprintf(stderr, "Converting to 16-bit LSB\n"); | |
| 224 #endif | |
| 225 src = cvt->buf+cvt->len_cvt; | |
| 226 dst = cvt->buf+cvt->len_cvt*2; | |
| 227 for ( i=cvt->len_cvt; i; --i ) { | |
| 228 src -= 1; | |
| 229 dst -= 2; | |
| 230 dst[1] = *src; | |
| 231 dst[0] = 0; | |
| 232 } | |
| 233 format = ((format & ~0x0008) | AUDIO_U16LSB); | |
| 234 cvt->len_cvt *= 2; | |
| 235 if ( cvt->filters[++cvt->filter_index] ) { | |
| 236 cvt->filters[cvt->filter_index](cvt, format); | |
| 237 } | |
| 238 } | |
| 239 /* Convert 8-bit to 16-bit - MSB */ | |
| 240 void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format) | |
| 241 { | |
| 242 int i; | |
| 243 Uint8 *src, *dst; | |
| 244 | |
| 245 #ifdef DEBUG_CONVERT | |
| 246 fprintf(stderr, "Converting to 16-bit MSB\n"); | |
| 247 #endif | |
| 248 src = cvt->buf+cvt->len_cvt; | |
| 249 dst = cvt->buf+cvt->len_cvt*2; | |
| 250 for ( i=cvt->len_cvt; i; --i ) { | |
| 251 src -= 1; | |
| 252 dst -= 2; | |
| 253 dst[0] = *src; | |
| 254 dst[1] = 0; | |
| 255 } | |
| 256 format = ((format & ~0x0008) | AUDIO_U16MSB); | |
| 257 cvt->len_cvt *= 2; | |
| 258 if ( cvt->filters[++cvt->filter_index] ) { | |
| 259 cvt->filters[cvt->filter_index](cvt, format); | |
| 260 } | |
| 261 } | |
| 262 | |
| 263 /* Convert 16-bit to 8-bit */ | |
| 264 void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format) | |
| 265 { | |
| 266 int i; | |
| 267 Uint8 *src, *dst; | |
| 268 | |
| 269 #ifdef DEBUG_CONVERT | |
| 270 fprintf(stderr, "Converting to 8-bit\n"); | |
| 271 #endif | |
| 272 src = cvt->buf; | |
| 273 dst = cvt->buf; | |
| 274 if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ | |
| 275 ++src; | |
| 276 } | |
| 277 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 278 *dst = *src; | |
| 279 src += 2; | |
| 280 dst += 1; | |
| 281 } | |
| 282 format = ((format & ~0x9010) | AUDIO_U8); | |
| 283 cvt->len_cvt /= 2; | |
| 284 if ( cvt->filters[++cvt->filter_index] ) { | |
| 285 cvt->filters[cvt->filter_index](cvt, format); | |
| 286 } | |
| 287 } | |
| 288 | |
| 289 /* Toggle signed/unsigned */ | |
| 290 void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format) | |
| 291 { | |
| 292 int i; | |
| 293 Uint8 *data; | |
| 294 | |
| 295 #ifdef DEBUG_CONVERT | |
| 296 fprintf(stderr, "Converting audio signedness\n"); | |
| 297 #endif | |
| 298 data = cvt->buf; | |
| 299 if ( (format & 0xFF) == 16 ) { | |
| 300 if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ | |
| 301 ++data; | |
| 302 } | |
| 303 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 304 *data ^= 0x80; | |
| 305 data += 2; | |
| 306 } | |
| 307 } else { | |
| 308 for ( i=cvt->len_cvt; i; --i ) { | |
| 309 *data++ ^= 0x80; | |
| 310 } | |
| 311 } | |
| 312 format = (format ^ 0x8000); | |
| 313 if ( cvt->filters[++cvt->filter_index] ) { | |
| 314 cvt->filters[cvt->filter_index](cvt, format); | |
| 315 } | |
| 316 } | |
| 317 | |
| 318 /* Toggle endianness */ | |
| 319 void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format) | |
| 320 { | |
| 321 int i; | |
| 322 Uint8 *data, tmp; | |
| 323 | |
| 324 #ifdef DEBUG_CONVERT | |
| 325 fprintf(stderr, "Converting audio endianness\n"); | |
| 326 #endif | |
| 327 data = cvt->buf; | |
| 328 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 329 tmp = data[0]; | |
| 330 data[0] = data[1]; | |
| 331 data[1] = tmp; | |
| 332 data += 2; | |
| 333 } | |
| 334 format = (format ^ 0x1000); | |
| 335 if ( cvt->filters[++cvt->filter_index] ) { | |
| 336 cvt->filters[cvt->filter_index](cvt, format); | |
| 337 } | |
| 338 } | |
| 339 | |
| 340 /* Convert rate up by multiple of 2 */ | |
| 341 void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format) | |
| 342 { | |
| 343 int i; | |
| 344 Uint8 *src, *dst; | |
| 345 | |
| 346 #ifdef DEBUG_CONVERT | |
| 347 fprintf(stderr, "Converting audio rate * 2\n"); | |
| 348 #endif | |
| 349 src = cvt->buf+cvt->len_cvt; | |
| 350 dst = cvt->buf+cvt->len_cvt*2; | |
| 351 switch (format & 0xFF) { | |
| 352 case 8: | |
| 353 for ( i=cvt->len_cvt; i; --i ) { | |
| 354 src -= 1; | |
| 355 dst -= 2; | |
| 356 dst[0] = src[0]; | |
| 357 dst[1] = src[0]; | |
| 358 } | |
| 359 break; | |
| 360 case 16: | |
| 361 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 362 src -= 2; | |
| 363 dst -= 4; | |
| 364 dst[0] = src[0]; | |
| 365 dst[1] = src[1]; | |
| 366 dst[2] = src[0]; | |
| 367 dst[3] = src[1]; | |
| 368 } | |
| 369 break; | |
| 370 } | |
| 371 cvt->len_cvt *= 2; | |
| 372 if ( cvt->filters[++cvt->filter_index] ) { | |
| 373 cvt->filters[cvt->filter_index](cvt, format); | |
| 374 } | |
| 375 } | |
| 376 | |
| 377 /* Convert rate down by multiple of 2 */ | |
| 378 void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format) | |
| 379 { | |
| 380 int i; | |
| 381 Uint8 *src, *dst; | |
| 382 | |
| 383 #ifdef DEBUG_CONVERT | |
| 384 fprintf(stderr, "Converting audio rate / 2\n"); | |
| 385 #endif | |
| 386 src = cvt->buf; | |
| 387 dst = cvt->buf; | |
| 388 switch (format & 0xFF) { | |
| 389 case 8: | |
| 390 for ( i=cvt->len_cvt/2; i; --i ) { | |
| 391 dst[0] = src[0]; | |
| 392 src += 2; | |
| 393 dst += 1; | |
| 394 } | |
| 395 break; | |
| 396 case 16: | |
| 397 for ( i=cvt->len_cvt/4; i; --i ) { | |
| 398 dst[0] = src[0]; | |
| 399 dst[1] = src[1]; | |
| 400 src += 4; | |
| 401 dst += 2; | |
| 402 } | |
| 403 break; | |
| 404 } | |
| 405 cvt->len_cvt /= 2; | |
| 406 if ( cvt->filters[++cvt->filter_index] ) { | |
| 407 cvt->filters[cvt->filter_index](cvt, format); | |
| 408 } | |
| 409 } | |
| 410 | |
| 411 /* Very slow rate conversion routine */ | |
| 412 void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format) | |
| 413 { | |
| 414 double ipos; | |
| 415 int i, clen; | |
| 416 | |
| 417 #ifdef DEBUG_CONVERT | |
| 418 fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr); | |
| 419 #endif | |
| 420 clen = (int)((double)cvt->len_cvt / cvt->rate_incr); | |
| 421 if ( cvt->rate_incr > 1.0 ) { | |
| 422 switch (format & 0xFF) { | |
| 423 case 8: { | |
| 424 Uint8 *output; | |
| 425 | |
| 426 output = cvt->buf; | |
| 427 ipos = 0.0; | |
| 428 for ( i=clen; i; --i ) { | |
| 429 *output = cvt->buf[(int)ipos]; | |
| 430 ipos += cvt->rate_incr; | |
| 431 output += 1; | |
| 432 } | |
| 433 } | |
| 434 break; | |
| 435 | |
| 436 case 16: { | |
| 437 Uint16 *output; | |
| 438 | |
| 439 clen &= ~1; | |
| 440 output = (Uint16 *)cvt->buf; | |
| 441 ipos = 0.0; | |
| 442 for ( i=clen/2; i; --i ) { | |
| 443 *output=((Uint16 *)cvt->buf)[(int)ipos]; | |
| 444 ipos += cvt->rate_incr; | |
| 445 output += 1; | |
| 446 } | |
| 447 } | |
| 448 break; | |
| 449 } | |
| 450 } else { | |
| 451 switch (format & 0xFF) { | |
| 452 case 8: { | |
| 453 Uint8 *output; | |
| 454 | |
| 455 output = cvt->buf+clen; | |
| 456 ipos = (double)cvt->len_cvt; | |
| 457 for ( i=clen; i; --i ) { | |
| 458 ipos -= cvt->rate_incr; | |
| 459 output -= 1; | |
| 460 *output = cvt->buf[(int)ipos]; | |
| 461 } | |
| 462 } | |
| 463 break; | |
| 464 | |
| 465 case 16: { | |
| 466 Uint16 *output; | |
| 467 | |
| 468 clen &= ~1; | |
| 469 output = (Uint16 *)(cvt->buf+clen); | |
| 470 ipos = (double)cvt->len_cvt/2; | |
| 471 for ( i=clen/2; i; --i ) { | |
| 472 ipos -= cvt->rate_incr; | |
| 473 output -= 1; | |
| 474 *output=((Uint16 *)cvt->buf)[(int)ipos]; | |
| 475 } | |
| 476 } | |
| 477 break; | |
| 478 } | |
| 479 } | |
| 480 cvt->len_cvt = clen; | |
| 481 if ( cvt->filters[++cvt->filter_index] ) { | |
| 482 cvt->filters[cvt->filter_index](cvt, format); | |
| 483 } | |
| 484 } | |
| 485 | |
| 486 int SDL_ConvertAudio(SDL_AudioCVT *cvt) | |
| 487 { | |
| 488 /* Make sure there's data to convert */ | |
| 489 if ( cvt->buf == NULL ) { | |
| 490 SDL_SetError("No buffer allocated for conversion"); | |
| 491 return(-1); | |
| 492 } | |
| 493 /* Return okay if no conversion is necessary */ | |
| 494 cvt->len_cvt = cvt->len; | |
| 495 if ( cvt->filters[0] == NULL ) { | |
| 496 return(0); | |
| 497 } | |
| 498 | |
| 499 /* Set up the conversion and go! */ | |
| 500 cvt->filter_index = 0; | |
| 501 cvt->filters[0](cvt, cvt->src_format); | |
| 502 return(0); | |
| 503 } | |
| 504 | |
| 505 /* Creates a set of audio filters to convert from one format to another. | |
| 506 Returns -1 if the format conversion is not supported, or 1 if the | |
| 507 audio filter is set up. | |
| 508 */ | |
| 509 | |
| 510 int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, | |
| 511 Uint16 src_format, Uint8 src_channels, int src_rate, | |
| 512 Uint16 dst_format, Uint8 dst_channels, int dst_rate) | |
| 513 { | |
| 514 /* Start off with no conversion necessary */ | |
| 515 cvt->needed = 0; | |
| 516 cvt->filter_index = 0; | |
| 517 cvt->filters[0] = NULL; | |
| 518 cvt->len_mult = 1; | |
| 519 cvt->len_ratio = 1.0; | |
| 520 | |
| 521 /* First filter: Endian conversion from src to dst */ | |
| 522 if ( (src_format & 0x1000) != (dst_format & 0x1000) | |
| 523 && ((src_format & 0xff) != 8) ) { | |
| 524 cvt->filters[cvt->filter_index++] = SDL_ConvertEndian; | |
| 525 } | |
| 526 | |
| 527 /* Second filter: Sign conversion -- signed/unsigned */ | |
| 528 if ( (src_format & 0x8000) != (dst_format & 0x8000) ) { | |
| 529 cvt->filters[cvt->filter_index++] = SDL_ConvertSign; | |
| 530 } | |
| 531 | |
| 532 /* Next filter: Convert 16 bit <--> 8 bit PCM */ | |
| 533 if ( (src_format & 0xFF) != (dst_format & 0xFF) ) { | |
| 534 switch (dst_format&0x10FF) { | |
| 535 case AUDIO_U8: | |
| 536 cvt->filters[cvt->filter_index++] = | |
| 537 SDL_Convert8; | |
| 538 cvt->len_ratio /= 2; | |
| 539 break; | |
| 540 case AUDIO_U16LSB: | |
| 541 cvt->filters[cvt->filter_index++] = | |
| 542 SDL_Convert16LSB; | |
| 543 cvt->len_mult *= 2; | |
| 544 cvt->len_ratio *= 2; | |
| 545 break; | |
| 546 case AUDIO_U16MSB: | |
| 547 cvt->filters[cvt->filter_index++] = | |
| 548 SDL_Convert16MSB; | |
| 549 cvt->len_mult *= 2; | |
| 550 cvt->len_ratio *= 2; | |
| 551 break; | |
| 552 } | |
| 553 } | |
| 554 | |
| 555 /* Last filter: Mono/Stereo conversion */ | |
| 556 if ( src_channels != dst_channels ) { | |
| 557 while ( (src_channels*2) <= dst_channels ) { | |
| 558 cvt->filters[cvt->filter_index++] = | |
| 559 SDL_ConvertStereo; | |
| 560 cvt->len_mult *= 2; | |
| 561 src_channels *= 2; | |
| 562 cvt->len_ratio *= 2; | |
| 563 } | |
| 564 /* This assumes that 4 channel audio is in the format: | |
| 565 Left {front/back} + Right {front/back} | |
| 566 so converting to L/R stereo works properly. | |
| 567 */ | |
| 568 while ( ((src_channels%2) == 0) && | |
| 569 ((src_channels/2) >= dst_channels) ) { | |
| 570 cvt->filters[cvt->filter_index++] = | |
| 571 SDL_ConvertMono; | |
| 572 src_channels /= 2; | |
| 573 cvt->len_ratio /= 2; | |
| 574 } | |
| 575 if ( src_channels != dst_channels ) { | |
| 576 /* Uh oh.. */; | |
| 577 } | |
| 578 } | |
| 579 | |
| 580 /* Do rate conversion */ | |
| 581 cvt->rate_incr = 0.0; | |
| 582 if ( (src_rate/100) != (dst_rate/100) ) { | |
| 583 Uint32 hi_rate, lo_rate; | |
| 584 int len_mult; | |
| 585 double len_ratio; | |
| 586 void (*rate_cvt)(SDL_AudioCVT *cvt, Uint16 format); | |
| 587 | |
| 588 if ( src_rate > dst_rate ) { | |
| 589 hi_rate = src_rate; | |
| 590 lo_rate = dst_rate; | |
| 591 rate_cvt = SDL_RateDIV2; | |
| 592 len_mult = 1; | |
| 593 len_ratio = 0.5; | |
| 594 } else { | |
| 595 hi_rate = dst_rate; | |
| 596 lo_rate = src_rate; | |
| 597 rate_cvt = SDL_RateMUL2; | |
| 598 len_mult = 2; | |
| 599 len_ratio = 2.0; | |
| 600 } | |
| 601 /* If hi_rate = lo_rate*2^x then conversion is easy */ | |
| 602 while ( ((lo_rate*2)/100) <= (hi_rate/100) ) { | |
| 603 cvt->filters[cvt->filter_index++] = rate_cvt; | |
| 604 cvt->len_mult *= len_mult; | |
| 605 lo_rate *= 2; | |
| 606 cvt->len_ratio *= len_ratio; | |
| 607 } | |
| 608 /* We may need a slow conversion here to finish up */ | |
| 609 if ( (lo_rate/100) != (hi_rate/100) ) { | |
| 610 #if 1 | |
| 611 /* The problem with this is that if the input buffer is | |
| 612 say 1K, and the conversion rate is say 1.1, then the | |
| 613 output buffer is 1.1K, which may not be an acceptable | |
| 614 buffer size for the audio driver (not a power of 2) | |
| 615 */ | |
| 616 /* For now, punt and hope the rate distortion isn't great. | |
| 617 */ | |
| 618 #else | |
| 619 if ( src_rate < dst_rate ) { | |
| 620 cvt->rate_incr = (double)lo_rate/hi_rate; | |
| 621 cvt->len_mult *= 2; | |
| 622 cvt->len_ratio /= cvt->rate_incr; | |
| 623 } else { | |
| 624 cvt->rate_incr = (double)hi_rate/lo_rate; | |
| 625 cvt->len_ratio *= cvt->rate_incr; | |
| 626 } | |
| 627 cvt->filters[cvt->filter_index++] = SDL_RateSLOW; | |
| 628 #endif | |
| 629 } | |
| 630 } | |
| 631 | |
| 632 /* Set up the filter information */ | |
| 633 if ( cvt->filter_index != 0 ) { | |
| 634 cvt->needed = 1; | |
| 635 cvt->src_format = src_format; | |
| 636 cvt->dst_format = dst_format; | |
| 637 cvt->len = 0; | |
| 638 cvt->buf = NULL; | |
| 639 cvt->filters[cvt->filter_index] = NULL; | |
| 640 } | |
| 641 return(cvt->needed); | |
| 642 } |