Mercurial > SDL_sound_CoreAudio
annotate audio_convert.c @ 501:7558f9d03d6f
Fixed ogg.h detection in configure script.
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Tue, 06 Dec 2005 12:07:40 +0000 |
| parents | 9e761a594df1 |
| children |
| rev | line source |
|---|---|
| 141 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
| 3 Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga | |
| 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 | |
| 20 slouken@devolution.com | |
| 21 */ | |
| 22 | |
| 23 /* | |
| 24 * This file was derived from SDL's SDL_audiocvt.c and is an attempt to | |
| 25 * address the shortcomings of it. | |
| 26 * | |
| 27 * Perhaps we can adapt some good filters from SoX? | |
| 28 */ | |
| 29 | |
| 30 #if HAVE_CONFIG_H | |
| 31 # include <config.h> | |
| 32 #endif | |
| 33 | |
| 34 #include "SDL.h" | |
| 35 #include "SDL_sound.h" | |
| 36 | |
| 37 #define __SDL_SOUND_INTERNAL__ | |
| 38 #include "SDL_sound_internal.h" | |
| 39 | |
| 40 /* Functions for audio drivers to perform runtime conversion of audio format */ | |
| 41 | |
| 42 | |
| 43 /* | |
| 44 * Toggle endianness. This filter is, of course, only applied to 16-bit | |
| 45 * audio data. | |
| 46 */ | |
| 47 | |
| 342 | 48 static void Sound_ConvertEndian(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 49 { |
| 50 int i; | |
| 51 Uint8 *data, tmp; | |
| 52 | |
| 53 /* SNDDBG(("Converting audio endianness\n")); */ | |
| 54 | |
| 55 data = cvt->buf; | |
| 56 | |
| 57 for (i = cvt->len_cvt / 2; i; --i) | |
| 58 { | |
| 59 tmp = data[0]; | |
| 60 data[0] = data[1]; | |
| 61 data[1] = tmp; | |
| 62 data += 2; | |
| 63 } /* for */ | |
| 64 | |
| 65 *format = (*format ^ 0x1000); | |
| 66 } /* Sound_ConvertEndian */ | |
| 67 | |
| 68 | |
| 69 /* | |
| 70 * Toggle signed/unsigned. Apparently this is done by toggling the most | |
| 71 * significant bit of each sample. | |
| 72 */ | |
| 73 | |
| 342 | 74 static void Sound_ConvertSign(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 75 { |
| 76 int i; | |
| 77 Uint8 *data; | |
| 78 | |
| 79 /* SNDDBG(("Converting audio signedness\n")); */ | |
| 80 | |
| 81 data = cvt->buf; | |
| 82 | |
| 83 /* 16-bit sound? */ | |
| 84 if ((*format & 0xFF) == 16) | |
| 85 { | |
| 86 /* Little-endian? */ | |
| 87 if ((*format & 0x1000) != 0x1000) | |
| 88 ++data; | |
| 89 | |
| 90 for (i = cvt->len_cvt / 2; i; --i) | |
| 91 { | |
| 92 *data ^= 0x80; | |
| 93 data += 2; | |
| 94 } /* for */ | |
| 95 } /* if */ | |
| 96 else | |
| 97 { | |
| 98 for (i = cvt->len_cvt; i; --i) | |
| 99 *data++ ^= 0x80; | |
| 100 } /* else */ | |
| 101 | |
| 102 *format = (*format ^ 0x8000); | |
| 103 } /* Sound_ConvertSign */ | |
| 104 | |
| 105 | |
| 106 /* | |
| 107 * Convert 16-bit to 8-bit. This is done by taking the most significant byte | |
| 108 * of each 16-bit sample. | |
| 109 */ | |
| 110 | |
| 342 | 111 static void Sound_Convert8(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 112 { |
| 113 int i; | |
| 114 Uint8 *src, *dst; | |
| 115 | |
| 116 /* SNDDBG(("Converting to 8-bit\n")); */ | |
| 117 | |
| 118 src = cvt->buf; | |
| 119 dst = cvt->buf; | |
| 120 | |
| 121 /* Little-endian? */ | |
| 122 if ((*format & 0x1000) != 0x1000) | |
| 123 ++src; | |
| 124 | |
| 125 for (i = cvt->len_cvt / 2; i; --i) | |
| 126 { | |
| 127 *dst = *src; | |
| 128 src += 2; | |
| 129 dst += 1; | |
| 130 } /* for */ | |
| 131 | |
| 132 *format = ((*format & ~0x9010) | AUDIO_U8); | |
| 133 cvt->len_cvt /= 2; | |
| 134 } /* Sound_Convert8 */ | |
| 135 | |
| 136 | |
| 137 /* Convert 8-bit to 16-bit - LSB */ | |
| 138 | |
| 342 | 139 static void Sound_Convert16LSB(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 140 { |
| 141 int i; | |
| 142 Uint8 *src, *dst; | |
| 143 | |
| 144 /* SNDDBG(("Converting to 16-bit LSB\n")); */ | |
| 145 | |
| 146 src = cvt->buf + cvt->len_cvt; | |
| 147 dst = cvt->buf + cvt->len_cvt * 2; | |
| 148 | |
| 149 for (i = cvt->len_cvt; i; --i) | |
| 150 { | |
| 151 src -= 1; | |
| 152 dst -= 2; | |
| 153 dst[1] = *src; | |
| 154 dst[0] = 0; | |
| 155 } /* for */ | |
| 156 | |
| 157 *format = ((*format & ~0x0008) | AUDIO_U16LSB); | |
| 158 cvt->len_cvt *= 2; | |
| 159 } /* Sound_Convert16LSB */ | |
| 160 | |
| 161 | |
| 162 /* Convert 8-bit to 16-bit - MSB */ | |
| 163 | |
| 342 | 164 static void Sound_Convert16MSB(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 165 { |
| 166 int i; | |
| 167 Uint8 *src, *dst; | |
| 168 | |
| 169 /* SNDDBG(("Converting to 16-bit MSB\n")); */ | |
| 170 | |
| 171 src = cvt->buf + cvt->len_cvt; | |
| 172 dst = cvt->buf + cvt->len_cvt * 2; | |
| 173 | |
| 174 for (i = cvt->len_cvt; i; --i) | |
| 175 { | |
| 176 src -= 1; | |
| 177 dst -= 2; | |
| 178 dst[0] = *src; | |
| 179 dst[1] = 0; | |
| 180 } /* for */ | |
| 181 | |
| 182 *format = ((*format & ~0x0008) | AUDIO_U16MSB); | |
| 183 cvt->len_cvt *= 2; | |
| 184 } /* Sound_Convert16MSB */ | |
| 185 | |
| 186 | |
| 187 /* Duplicate a mono channel to both stereo channels */ | |
| 188 | |
| 342 | 189 static void Sound_ConvertStereo(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 190 { |
| 191 int i; | |
| 192 | |
| 193 /* SNDDBG(("Converting to stereo\n")); */ | |
| 194 | |
| 195 /* 16-bit sound? */ | |
| 196 if ((*format & 0xFF) == 16) | |
| 197 { | |
| 198 Uint16 *src, *dst; | |
| 199 | |
| 200 src = (Uint16 *) (cvt->buf + cvt->len_cvt); | |
| 201 dst = (Uint16 *) (cvt->buf + cvt->len_cvt * 2); | |
| 202 | |
| 203 for (i = cvt->len_cvt/2; i; --i) | |
| 204 { | |
| 205 dst -= 2; | |
| 206 src -= 1; | |
| 207 dst[0] = src[0]; | |
| 208 dst[1] = src[0]; | |
| 209 } /* for */ | |
| 210 } /* if */ | |
| 211 else | |
| 212 { | |
| 213 Uint8 *src, *dst; | |
| 214 | |
| 215 src = cvt->buf + cvt->len_cvt; | |
| 216 dst = cvt->buf + cvt->len_cvt * 2; | |
| 217 | |
| 218 for (i = cvt->len_cvt; i; --i) | |
| 219 { | |
| 220 dst -= 2; | |
| 221 src -= 1; | |
| 222 dst[0] = src[0]; | |
| 223 dst[1] = src[0]; | |
| 224 } /* for */ | |
| 225 } /* else */ | |
| 226 | |
| 227 cvt->len_cvt *= 2; | |
| 228 } /* Sound_ConvertStereo */ | |
| 229 | |
| 230 | |
| 231 /* Effectively mix right and left channels into a single channel */ | |
| 232 | |
| 342 | 233 static void Sound_ConvertMono(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 234 { |
| 235 int i; | |
| 236 Sint32 sample; | |
| 237 Uint8 *u_src, *u_dst; | |
| 238 Sint8 *s_src, *s_dst; | |
| 239 | |
| 240 /* SNDDBG(("Converting to mono\n")); */ | |
| 241 | |
| 242 switch (*format) | |
| 243 { | |
| 244 case AUDIO_U8: | |
| 245 u_src = cvt->buf; | |
| 246 u_dst = cvt->buf; | |
| 247 | |
| 248 for (i = cvt->len_cvt / 2; i; --i) | |
| 249 { | |
| 250 sample = u_src[0] + u_src[1]; | |
| 251 *u_dst = (sample > 255) ? 255 : sample; | |
| 252 u_src += 2; | |
| 253 u_dst += 1; | |
| 254 } /* for */ | |
| 255 break; | |
| 256 | |
| 257 case AUDIO_S8: | |
| 258 s_src = (Sint8 *) cvt->buf; | |
| 259 s_dst = (Sint8 *) cvt->buf; | |
| 260 | |
| 261 for (i = cvt->len_cvt / 2; i; --i) | |
| 262 { | |
| 263 sample = s_src[0] + s_src[1]; | |
| 264 if (sample > 127) | |
| 265 *s_dst = 127; | |
| 266 else if (sample < -128) | |
| 267 *s_dst = -128; | |
| 268 else | |
| 269 *s_dst = sample; | |
| 270 | |
| 271 s_src += 2; | |
| 272 s_dst += 1; | |
| 273 } /* for */ | |
| 274 break; | |
| 275 | |
| 276 case AUDIO_U16MSB: | |
| 277 u_src = cvt->buf; | |
| 278 u_dst = cvt->buf; | |
| 279 | |
| 280 for (i = cvt->len_cvt / 4; i; --i) | |
| 281 { | |
| 282 sample = (Uint16) ((u_src[0] << 8) | u_src[1]) | |
| 283 + (Uint16) ((u_src[2] << 8) | u_src[3]); | |
| 284 if (sample > 65535) | |
| 285 { | |
| 286 u_dst[0] = 0xFF; | |
| 287 u_dst[1] = 0xFF; | |
| 288 } /* if */ | |
| 289 else | |
| 290 { | |
| 291 u_dst[1] = (sample & 0xFF); | |
| 292 sample >>= 8; | |
| 293 u_dst[0] = (sample & 0xFF); | |
| 294 } /* else */ | |
| 295 u_src += 4; | |
| 296 u_dst += 2; | |
| 297 } /* for */ | |
| 298 break; | |
| 299 | |
| 300 case AUDIO_U16LSB: | |
| 301 u_src = cvt->buf; | |
| 302 u_dst = cvt->buf; | |
| 303 | |
| 304 for (i = cvt->len_cvt / 4; i; --i) | |
| 305 { | |
| 306 sample = (Uint16) ((u_src[1] << 8) | u_src[0]) | |
| 307 + (Uint16) ((u_src[3] << 8) | u_src[2]); | |
| 308 if (sample > 65535) | |
| 309 { | |
| 310 u_dst[0] = 0xFF; | |
| 311 u_dst[1] = 0xFF; | |
| 312 } /* if */ | |
| 313 else | |
| 314 { | |
| 315 u_dst[0] = (sample & 0xFF); | |
| 316 sample >>= 8; | |
| 317 u_dst[1] = (sample & 0xFF); | |
| 318 } /* else */ | |
| 319 u_src += 4; | |
| 320 u_dst += 2; | |
| 321 } /* for */ | |
| 322 break; | |
| 323 | |
| 324 case AUDIO_S16MSB: | |
| 325 u_src = cvt->buf; | |
| 326 u_dst = cvt->buf; | |
| 327 | |
| 328 for (i = cvt->len_cvt / 4; i; --i) | |
| 329 { | |
| 330 sample = (Sint16) ((u_src[0] << 8) | u_src[1]) | |
| 331 + (Sint16) ((u_src[2] << 8) | u_src[3]); | |
| 332 if (sample > 32767) | |
| 333 { | |
| 334 u_dst[0] = 0x7F; | |
| 335 u_dst[1] = 0xFF; | |
| 336 } /* if */ | |
| 337 else if (sample < -32768) | |
| 338 { | |
| 339 u_dst[0] = 0x80; | |
| 340 u_dst[1] = 0x00; | |
| 341 } /* else if */ | |
| 342 else | |
| 343 { | |
| 344 u_dst[1] = (sample & 0xFF); | |
| 345 sample >>= 8; | |
| 346 u_dst[0] = (sample & 0xFF); | |
| 347 } /* else */ | |
| 348 u_src += 4; | |
| 349 u_dst += 2; | |
| 350 } /* for */ | |
| 351 break; | |
| 352 | |
| 353 case AUDIO_S16LSB: | |
| 354 u_src = cvt->buf; | |
| 355 u_dst = cvt->buf; | |
| 356 | |
| 357 for (i = cvt->len_cvt / 4; i; --i) | |
| 358 { | |
| 359 sample = (Sint16) ((u_src[1] << 8) | u_src[0]) | |
| 360 + (Sint16) ((u_src[3] << 8) | u_src[2]); | |
| 361 if (sample > 32767) | |
| 362 { | |
| 363 u_dst[1] = 0x7F; | |
| 364 u_dst[0] = 0xFF; | |
| 365 } /* if */ | |
| 366 else if (sample < -32768) | |
| 367 { | |
| 368 u_dst[1] = 0x80; | |
| 369 u_dst[0] = 0x00; | |
| 370 } /* else if */ | |
| 371 else | |
| 372 { | |
| 373 u_dst[0] = (sample & 0xFF); | |
| 374 sample >>= 8; | |
| 375 u_dst[1] = (sample & 0xFF); | |
| 376 } /* else */ | |
| 377 u_src += 4; | |
| 378 u_dst += 2; | |
| 379 } /* for */ | |
| 380 break; | |
| 381 } /* switch */ | |
| 382 | |
| 383 cvt->len_cvt /= 2; | |
| 384 } /* Sound_ConvertMono */ | |
| 385 | |
| 386 | |
| 387 /* Convert rate up by multiple of 2 */ | |
| 388 | |
| 342 | 389 static void Sound_RateMUL2(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 390 { |
| 391 int i; | |
| 392 Uint8 *src, *dst; | |
| 393 | |
| 394 /* SNDDBG(("Converting audio rate * 2\n")); */ | |
| 395 | |
| 396 src = cvt->buf + cvt->len_cvt; | |
| 397 dst = cvt->buf + cvt->len_cvt*2; | |
| 398 | |
| 399 /* 8- or 16-bit sound? */ | |
| 400 switch (*format & 0xFF) | |
| 401 { | |
| 402 case 8: | |
| 403 for (i = cvt->len_cvt; i; --i) | |
| 404 { | |
| 405 src -= 1; | |
| 406 dst -= 2; | |
| 407 dst[0] = src[0]; | |
| 408 dst[1] = src[0]; | |
| 409 } /* for */ | |
| 410 break; | |
| 411 | |
| 412 case 16: | |
| 413 for (i = cvt->len_cvt / 2; i; --i) | |
| 414 { | |
| 415 src -= 2; | |
| 416 dst -= 4; | |
| 417 dst[0] = src[0]; | |
| 418 dst[1] = src[1]; | |
| 419 dst[2] = src[0]; | |
| 420 dst[3] = src[1]; | |
| 421 } /* for */ | |
| 422 break; | |
| 423 } /* switch */ | |
| 424 | |
| 425 cvt->len_cvt *= 2; | |
| 426 } /* Sound_RateMUL2 */ | |
| 427 | |
| 428 | |
| 429 /* Convert rate down by multiple of 2 */ | |
| 430 | |
| 342 | 431 static void Sound_RateDIV2(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 432 { |
| 433 int i; | |
| 434 Uint8 *src, *dst; | |
| 435 | |
| 436 /* SNDDBG(("Converting audio rate / 2\n")); */ | |
| 437 | |
| 438 src = cvt->buf; | |
| 439 dst = cvt->buf; | |
| 440 | |
| 441 /* 8- or 16-bit sound? */ | |
| 442 switch (*format & 0xFF) | |
| 443 { | |
| 444 case 8: | |
| 445 for (i = cvt->len_cvt / 2; i; --i) | |
| 446 { | |
| 447 dst[0] = src[0]; | |
| 448 src += 2; | |
| 449 dst += 1; | |
| 450 } /* for */ | |
| 451 break; | |
| 452 | |
| 453 case 16: | |
| 454 for (i = cvt->len_cvt / 4; i; --i) | |
| 455 { | |
| 456 dst[0] = src[0]; | |
| 457 dst[1] = src[1]; | |
| 458 src += 4; | |
| 459 dst += 2; | |
| 460 } | |
| 461 break; | |
| 462 } /* switch */ | |
| 463 | |
| 464 cvt->len_cvt /= 2; | |
| 465 } /* Sound_RateDIV2 */ | |
| 466 | |
| 467 | |
| 468 /* Very slow rate conversion routine */ | |
| 469 | |
| 342 | 470 static void Sound_RateSLOW(Sound_AudioCVT *cvt, Uint16 *format) |
| 141 | 471 { |
| 472 double ipos; | |
| 473 int i, clen; | |
| 474 Uint8 *output8; | |
| 475 Uint16 *output16; | |
| 476 | |
| 477 /* SNDDBG(("Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr)); */ | |
| 478 | |
| 479 clen = (int) ((double) cvt->len_cvt / cvt->rate_incr); | |
| 480 | |
| 481 if (cvt->rate_incr > 1.0) | |
| 482 { | |
| 483 /* 8- or 16-bit sound? */ | |
| 484 switch (*format & 0xFF) | |
| 485 { | |
| 486 case 8: | |
| 487 output8 = cvt->buf; | |
| 488 | |
| 489 ipos = 0.0; | |
| 490 for (i = clen; i; --i) | |
| 491 { | |
| 492 *output8 = cvt->buf[(int) ipos]; | |
| 493 ipos += cvt->rate_incr; | |
| 494 output8 += 1; | |
| 495 } /* for */ | |
| 496 break; | |
| 497 | |
| 498 case 16: | |
| 499 output16 = (Uint16 *) cvt->buf; | |
| 500 | |
| 501 clen &= ~1; | |
| 502 ipos = 0.0; | |
| 503 for (i = clen / 2; i; --i) | |
| 504 { | |
| 505 *output16 = ((Uint16 *) cvt->buf)[(int) ipos]; | |
| 506 ipos += cvt->rate_incr; | |
| 507 output16 += 1; | |
| 508 } /* for */ | |
| 509 break; | |
| 510 } /* switch */ | |
| 511 } /* if */ | |
| 512 else | |
| 513 { | |
| 514 /* 8- or 16-bit sound */ | |
| 515 switch (*format & 0xFF) | |
| 516 { | |
| 517 case 8: | |
| 518 output8 = cvt->buf + clen; | |
| 519 | |
| 520 ipos = (double) cvt->len_cvt; | |
| 521 for (i = clen; i; --i) | |
| 522 { | |
| 523 ipos -= cvt->rate_incr; | |
| 524 output8 -= 1; | |
| 525 *output8 = cvt->buf[(int) ipos]; | |
| 526 } /* for */ | |
| 527 break; | |
| 528 | |
| 529 case 16: | |
| 530 clen &= ~1; | |
| 531 output16 = (Uint16 *) (cvt->buf + clen); | |
| 532 ipos = (double) cvt->len_cvt / 2; | |
| 533 for (i = clen / 2; i; --i) | |
| 534 { | |
| 535 ipos -= cvt->rate_incr; | |
| 536 output16 -= 1; | |
| 537 *output16 = ((Uint16 *) cvt->buf)[(int) ipos]; | |
| 538 } /* for */ | |
| 539 break; | |
| 540 } /* switch */ | |
| 541 } /* else */ | |
| 542 | |
| 543 cvt->len_cvt = clen; | |
| 544 } /* Sound_RateSLOW */ | |
| 545 | |
| 546 | |
| 547 int Sound_ConvertAudio(Sound_AudioCVT *cvt) | |
| 548 { | |
| 549 Uint16 format; | |
| 550 | |
| 551 /* Make sure there's data to convert */ | |
| 552 if (cvt->buf == NULL) | |
| 553 { | |
|
387
fb519e6028e3
Changed all the Sound_SetError() calls to __Sound_SetError (or BAIL*_MACRO)
Ryan C. Gordon <icculus@icculus.org>
parents:
382
diff
changeset
|
554 __Sound_SetError("No buffer allocated for conversion"); |
| 141 | 555 return(-1); |
| 556 } /* if */ | |
| 557 | |
| 558 /* Return okay if no conversion is necessary */ | |
| 559 cvt->len_cvt = cvt->len; | |
| 560 if (cvt->filters[0] == NULL) | |
| 561 return(0); | |
| 562 | |
| 563 /* Set up the conversion and go! */ | |
| 564 format = cvt->src_format; | |
| 565 for (cvt->filter_index = 0; cvt->filters[cvt->filter_index]; | |
| 566 cvt->filter_index++) | |
| 567 { | |
| 568 cvt->filters[cvt->filter_index](cvt, &format); | |
| 569 } | |
| 570 return(0); | |
| 571 } /* Sound_ConvertAudio */ | |
| 572 | |
| 573 | |
| 574 /* | |
| 575 * Creates a set of audio filters to convert from one format to another. | |
| 576 * Returns -1 if the format conversion is not supported, or 1 if the | |
| 577 * audio filter is set up. | |
| 578 */ | |
| 579 | |
| 580 int Sound_BuildAudioCVT(Sound_AudioCVT *cvt, | |
| 581 Uint16 src_format, Uint8 src_channels, Uint32 src_rate, | |
|
382
ce998ee6194f
Sync'd with latest altcvt.
Ryan C. Gordon <icculus@icculus.org>
parents:
342
diff
changeset
|
582 Uint16 dst_format, Uint8 dst_channels, Uint32 dst_rate, |
|
ce998ee6194f
Sync'd with latest altcvt.
Ryan C. Gordon <icculus@icculus.org>
parents:
342
diff
changeset
|
583 Uint32 dst_size) |
| 141 | 584 { |
| 585 /* Start off with no conversion necessary */ | |
| 586 cvt->needed = 0; | |
| 587 cvt->filter_index = 0; | |
| 588 cvt->filters[0] = NULL; | |
| 589 cvt->len_mult = 1; | |
| 590 cvt->len_ratio = 1.0; | |
| 591 | |
| 592 /* First filter: Endian conversion from src to dst */ | |
| 593 if ((src_format & 0x1000) != (dst_format & 0x1000) && | |
| 594 ((src_format & 0xff) != 8)) | |
| 595 { | |
| 596 SNDDBG(("Adding filter: Sound_ConvertEndian\n")); | |
| 597 cvt->filters[cvt->filter_index++] = Sound_ConvertEndian; | |
| 598 } /* if */ | |
| 599 | |
| 600 /* Second filter: Sign conversion -- signed/unsigned */ | |
| 601 if ((src_format & 0x8000) != (dst_format & 0x8000)) | |
| 602 { | |
| 603 SNDDBG(("Adding filter: Sound_ConvertSign\n")); | |
| 604 cvt->filters[cvt->filter_index++] = Sound_ConvertSign; | |
| 605 } /* if */ | |
| 606 | |
| 607 /* Next filter: Convert 16 bit <--> 8 bit PCM. */ | |
| 608 if ((src_format & 0xFF) != (dst_format & 0xFF)) | |
| 609 { | |
| 610 switch (dst_format & 0x10FF) | |
| 611 { | |
| 612 case AUDIO_U8: | |
| 613 SNDDBG(("Adding filter: Sound_Convert8\n")); | |
| 614 cvt->filters[cvt->filter_index++] = Sound_Convert8; | |
| 615 cvt->len_ratio /= 2; | |
| 616 break; | |
| 617 | |
| 618 case AUDIO_U16LSB: | |
| 619 SNDDBG(("Adding filter: Sound_Convert16LSB\n")); | |
| 620 cvt->filters[cvt->filter_index++] = Sound_Convert16LSB; | |
| 621 cvt->len_mult *= 2; | |
| 622 cvt->len_ratio *= 2; | |
| 623 break; | |
| 624 | |
| 625 case AUDIO_U16MSB: | |
| 626 SNDDBG(("Adding filter: Sound_Convert16MSB\n")); | |
| 627 cvt->filters[cvt->filter_index++] = Sound_Convert16MSB; | |
| 628 cvt->len_mult *= 2; | |
| 629 cvt->len_ratio *= 2; | |
| 630 break; | |
| 631 } /* switch */ | |
| 632 } /* if */ | |
| 633 | |
| 634 /* Next filter: Mono/Stereo conversion */ | |
| 635 if (src_channels != dst_channels) | |
| 636 { | |
| 637 while ((src_channels * 2) <= dst_channels) | |
| 638 { | |
| 639 SNDDBG(("Adding filter: Sound_ConvertStereo\n")); | |
| 640 cvt->filters[cvt->filter_index++] = Sound_ConvertStereo; | |
| 641 cvt->len_mult *= 2; | |
| 642 src_channels *= 2; | |
| 643 cvt->len_ratio *= 2; | |
| 644 } /* while */ | |
| 645 | |
| 646 /* This assumes that 4 channel audio is in the format: | |
| 647 * Left {front/back} + Right {front/back} | |
| 648 * so converting to L/R stereo works properly. | |
| 649 */ | |
| 650 while (((src_channels % 2) == 0) && | |
| 651 ((src_channels / 2) >= dst_channels)) | |
| 652 { | |
| 653 SNDDBG(("Adding filter: Sound_ConvertMono\n")); | |
| 654 cvt->filters[cvt->filter_index++] = Sound_ConvertMono; | |
| 655 src_channels /= 2; | |
| 656 cvt->len_ratio /= 2; | |
| 657 } /* while */ | |
| 658 | |
| 659 if ( src_channels != dst_channels ) { | |
| 660 /* Uh oh.. */; | |
| 661 } /* if */ | |
| 662 } /* if */ | |
| 663 | |
| 664 /* Do rate conversion */ | |
| 665 cvt->rate_incr = 0.0; | |
| 666 if ((src_rate / 100) != (dst_rate / 100)) | |
| 667 { | |
| 668 Uint32 hi_rate, lo_rate; | |
| 669 int len_mult; | |
| 670 double len_ratio; | |
| 671 void (*rate_cvt)(Sound_AudioCVT *cvt, Uint16 *format); | |
| 672 | |
| 673 if (src_rate > dst_rate) | |
| 674 { | |
| 675 hi_rate = src_rate; | |
| 676 lo_rate = dst_rate; | |
| 677 SNDDBG(("Adding filter: Sound_RateDIV2\n")); | |
| 678 rate_cvt = Sound_RateDIV2; | |
| 679 len_mult = 1; | |
| 680 len_ratio = 0.5; | |
| 681 } /* if */ | |
| 682 else | |
| 683 { | |
| 684 hi_rate = dst_rate; | |
| 685 lo_rate = src_rate; | |
| 686 SNDDBG(("Adding filter: Sound_RateMUL2\n")); | |
| 687 rate_cvt = Sound_RateMUL2; | |
| 688 len_mult = 2; | |
| 689 len_ratio = 2.0; | |
| 690 } /* else */ | |
| 691 | |
| 692 /* If hi_rate = lo_rate*2^x then conversion is easy */ | |
| 693 while (((lo_rate * 2) / 100) <= (hi_rate / 100)) | |
| 694 { | |
| 695 cvt->filters[cvt->filter_index++] = rate_cvt; | |
| 696 cvt->len_mult *= len_mult; | |
| 697 lo_rate *= 2; | |
| 698 cvt->len_ratio *= len_ratio; | |
| 699 } /* while */ | |
| 700 | |
| 701 /* We may need a slow conversion here to finish up */ | |
| 702 if ((lo_rate / 100) != (hi_rate / 100)) | |
| 703 { | |
| 704 if (src_rate < dst_rate) | |
| 705 { | |
| 706 cvt->rate_incr = (double) lo_rate / hi_rate; | |
| 707 cvt->len_mult *= 2; | |
| 708 cvt->len_ratio /= cvt->rate_incr; | |
| 709 } /* if */ | |
| 710 else | |
| 711 { | |
| 712 cvt->rate_incr = (double) hi_rate / lo_rate; | |
| 713 cvt->len_ratio *= cvt->rate_incr; | |
| 714 } /* else */ | |
| 715 SNDDBG(("Adding filter: Sound_RateSLOW\n")); | |
| 716 cvt->filters[cvt->filter_index++] = Sound_RateSLOW; | |
| 717 } /* if */ | |
| 718 } /* if */ | |
| 719 | |
| 720 /* Set up the filter information */ | |
| 721 if (cvt->filter_index != 0) | |
| 722 { | |
| 723 cvt->needed = 1; | |
| 724 cvt->src_format = src_format; | |
| 725 cvt->dst_format = dst_format; | |
| 726 cvt->len = 0; | |
| 727 cvt->buf = NULL; | |
| 728 cvt->filters[cvt->filter_index] = NULL; | |
| 729 } /* if */ | |
| 730 | |
| 731 return(cvt->needed); | |
| 732 } /* Sound_BuildAudioCVT */ | |
|
465
636796aed4e2
Build/packaging fixes.
Ryan C. Gordon <icculus@icculus.org>
parents:
387
diff
changeset
|
733 |
|
636796aed4e2
Build/packaging fixes.
Ryan C. Gordon <icculus@icculus.org>
parents:
387
diff
changeset
|
734 /* end of audio_convert.c ... */ |
|
636796aed4e2
Build/packaging fixes.
Ryan C. Gordon <icculus@icculus.org>
parents:
387
diff
changeset
|
735 |