| 1 | /* Copyright (C) 2010-2020 The RetroArch team |
| 2 | * |
| 3 | * --------------------------------------------------------------------------------------- |
| 4 | * The following license statement only applies to this file (audio_mixer.c). |
| 5 | * --------------------------------------------------------------------------------------- |
| 6 | * |
| 7 | * Permission is hereby granted, free of charge, |
| 8 | * to any person obtaining a copy of this software and associated documentation files (the "Software"), |
| 9 | * to deal in the Software without restriction, including without limitation the rights to |
| 10 | * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, |
| 11 | * and to permit persons to whom the Software is furnished to do so, subject to the following conditions: |
| 12 | * |
| 13 | * The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. |
| 14 | * |
| 15 | * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, |
| 16 | * INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
| 17 | * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. |
| 18 | * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, |
| 19 | * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
| 20 | * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. |
| 21 | */ |
| 22 | |
| 23 | #ifdef HAVE_CONFIG_H |
| 24 | #include "../../config.h" |
| 25 | #endif |
| 26 | |
| 27 | #include <audio/audio_mixer.h> |
| 28 | #include <audio/audio_resampler.h> |
| 29 | |
| 30 | #ifdef HAVE_RWAV |
| 31 | #include <formats/rwav.h> |
| 32 | #endif |
| 33 | #include <memalign.h> |
| 34 | |
| 35 | #include <stdio.h> |
| 36 | #include <stdlib.h> |
| 37 | #include <string.h> |
| 38 | #include <math.h> |
| 39 | |
| 40 | #ifdef HAVE_STB_VORBIS |
| 41 | #define STB_VORBIS_NO_PUSHDATA_API |
| 42 | #define STB_VORBIS_NO_STDIO |
| 43 | #define STB_VORBIS_NO_CRT |
| 44 | |
| 45 | #include <stb/stb_vorbis.h> |
| 46 | #endif |
| 47 | |
| 48 | #ifdef HAVE_DR_FLAC |
| 49 | #define DR_FLAC_IMPLEMENTATION |
| 50 | #include <dr/dr_flac.h> |
| 51 | #endif |
| 52 | |
| 53 | #ifdef HAVE_DR_MP3 |
| 54 | #define DR_MP3_IMPLEMENTATION |
| 55 | #include <retro_assert.h> |
| 56 | #define DRMP3_ASSERT(expression) retro_assert(expression) |
| 57 | #include <dr/dr_mp3.h> |
| 58 | #endif |
| 59 | |
| 60 | #ifdef HAVE_IBXM |
| 61 | #include <ibxm/ibxm.h> |
| 62 | #endif |
| 63 | |
| 64 | #ifdef HAVE_THREADS |
| 65 | #include <rthreads/rthreads.h> |
| 66 | #define AUDIO_MIXER_LOCK(voice) slock_lock(voice->lock) |
| 67 | #define AUDIO_MIXER_UNLOCK(voice) slock_unlock(voice->lock) |
| 68 | #else |
| 69 | #define AUDIO_MIXER_LOCK(voice) do {} while(0) |
| 70 | #define AUDIO_MIXER_UNLOCK(voice) do {} while(0) |
| 71 | #endif |
| 72 | |
| 73 | #define AUDIO_MIXER_MAX_VOICES 8 |
| 74 | #define AUDIO_MIXER_TEMP_BUFFER 8192 |
| 75 | |
| 76 | struct audio_mixer_sound |
| 77 | { |
| 78 | enum audio_mixer_type type; |
| 79 | |
| 80 | union |
| 81 | { |
| 82 | struct |
| 83 | { |
| 84 | /* wav */ |
| 85 | const float* pcm; |
| 86 | unsigned frames; |
| 87 | } wav; |
| 88 | |
| 89 | #ifdef HAVE_STB_VORBIS |
| 90 | struct |
| 91 | { |
| 92 | /* ogg */ |
| 93 | const void* data; |
| 94 | unsigned size; |
| 95 | } ogg; |
| 96 | #endif |
| 97 | |
| 98 | #ifdef HAVE_DR_FLAC |
| 99 | struct |
| 100 | { |
| 101 | /* flac */ |
| 102 | const void* data; |
| 103 | unsigned size; |
| 104 | } flac; |
| 105 | #endif |
| 106 | |
| 107 | #ifdef HAVE_DR_MP3 |
| 108 | struct |
| 109 | { |
| 110 | /* mp */ |
| 111 | const void* data; |
| 112 | unsigned size; |
| 113 | } mp3; |
| 114 | #endif |
| 115 | |
| 116 | #ifdef HAVE_IBXM |
| 117 | struct |
| 118 | { |
| 119 | /* mod/s3m/xm */ |
| 120 | const void* data; |
| 121 | unsigned size; |
| 122 | } mod; |
| 123 | #endif |
| 124 | } types; |
| 125 | }; |
| 126 | |
| 127 | struct audio_mixer_voice |
| 128 | { |
| 129 | struct |
| 130 | { |
| 131 | struct |
| 132 | { |
| 133 | unsigned position; |
| 134 | } wav; |
| 135 | |
| 136 | #ifdef HAVE_STB_VORBIS |
| 137 | struct |
| 138 | { |
| 139 | stb_vorbis *stream; |
| 140 | void *resampler_data; |
| 141 | const retro_resampler_t *resampler; |
| 142 | float *buffer; |
| 143 | unsigned position; |
| 144 | unsigned samples; |
| 145 | unsigned buf_samples; |
| 146 | float ratio; |
| 147 | } ogg; |
| 148 | #endif |
| 149 | |
| 150 | #ifdef HAVE_DR_FLAC |
| 151 | struct |
| 152 | { |
| 153 | float* buffer; |
| 154 | drflac *stream; |
| 155 | void *resampler_data; |
| 156 | const retro_resampler_t *resampler; |
| 157 | unsigned position; |
| 158 | unsigned samples; |
| 159 | unsigned buf_samples; |
| 160 | float ratio; |
| 161 | } flac; |
| 162 | #endif |
| 163 | |
| 164 | #ifdef HAVE_DR_MP3 |
| 165 | struct |
| 166 | { |
| 167 | drmp3 stream; |
| 168 | void *resampler_data; |
| 169 | const retro_resampler_t *resampler; |
| 170 | float* buffer; |
| 171 | unsigned position; |
| 172 | unsigned samples; |
| 173 | unsigned buf_samples; |
| 174 | float ratio; |
| 175 | } mp3; |
| 176 | #endif |
| 177 | |
| 178 | #ifdef HAVE_IBXM |
| 179 | struct |
| 180 | { |
| 181 | int* buffer; |
| 182 | struct replay* stream; |
| 183 | struct module* module; |
| 184 | unsigned position; |
| 185 | unsigned samples; |
| 186 | unsigned buf_samples; |
| 187 | } mod; |
| 188 | #endif |
| 189 | } types; |
| 190 | audio_mixer_sound_t *sound; |
| 191 | audio_mixer_stop_cb_t stop_cb; |
| 192 | unsigned type; |
| 193 | float volume; |
| 194 | bool repeat; |
| 195 | #ifdef HAVE_THREADS |
| 196 | slock_t *lock; |
| 197 | #endif |
| 198 | }; |
| 199 | |
| 200 | /* TODO/FIXME - static globals */ |
| 201 | static struct audio_mixer_voice s_voices[AUDIO_MIXER_MAX_VOICES] = {0}; |
| 202 | static unsigned s_rate = 0; |
| 203 | |
| 204 | static void audio_mixer_release(audio_mixer_voice_t* voice); |
| 205 | |
| 206 | #ifdef HAVE_RWAV |
| 207 | static bool wav_to_float(const rwav_t* wav, float** pcm, size_t samples_out) |
| 208 | { |
| 209 | size_t i; |
| 210 | /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */ |
| 211 | float *f = (float*)memalign_alloc(16, |
| 212 | ((samples_out + 15) & ~15) * sizeof(float)); |
| 213 | |
| 214 | if (!f) |
| 215 | return false; |
| 216 | |
| 217 | *pcm = f; |
| 218 | |
| 219 | if (wav->bitspersample == 8) |
| 220 | { |
| 221 | float sample = 0.0f; |
| 222 | const uint8_t *u8 = (const uint8_t*)wav->samples; |
| 223 | |
| 224 | if (wav->numchannels == 1) |
| 225 | { |
| 226 | for (i = wav->numsamples; i != 0; i--) |
| 227 | { |
| 228 | sample = (float)*u8++ / 255.0f; |
| 229 | sample = sample * 2.0f - 1.0f; |
| 230 | *f++ = sample; |
| 231 | *f++ = sample; |
| 232 | } |
| 233 | } |
| 234 | else if (wav->numchannels == 2) |
| 235 | { |
| 236 | for (i = wav->numsamples; i != 0; i--) |
| 237 | { |
| 238 | sample = (float)*u8++ / 255.0f; |
| 239 | sample = sample * 2.0f - 1.0f; |
| 240 | *f++ = sample; |
| 241 | sample = (float)*u8++ / 255.0f; |
| 242 | sample = sample * 2.0f - 1.0f; |
| 243 | *f++ = sample; |
| 244 | } |
| 245 | } |
| 246 | } |
| 247 | else |
| 248 | { |
| 249 | /* TODO/FIXME note to leiradel - can we use audio/conversion/s16_to_float |
| 250 | * functions here? */ |
| 251 | |
| 252 | float sample = 0.0f; |
| 253 | const int16_t *s16 = (const int16_t*)wav->samples; |
| 254 | |
| 255 | if (wav->numchannels == 1) |
| 256 | { |
| 257 | for (i = wav->numsamples; i != 0; i--) |
| 258 | { |
| 259 | sample = (float)((int)*s16++ + 32768) / 65535.0f; |
| 260 | sample = sample * 2.0f - 1.0f; |
| 261 | *f++ = sample; |
| 262 | *f++ = sample; |
| 263 | } |
| 264 | } |
| 265 | else if (wav->numchannels == 2) |
| 266 | { |
| 267 | for (i = wav->numsamples; i != 0; i--) |
| 268 | { |
| 269 | sample = (float)((int)*s16++ + 32768) / 65535.0f; |
| 270 | sample = sample * 2.0f - 1.0f; |
| 271 | *f++ = sample; |
| 272 | sample = (float)((int)*s16++ + 32768) / 65535.0f; |
| 273 | sample = sample * 2.0f - 1.0f; |
| 274 | *f++ = sample; |
| 275 | } |
| 276 | } |
| 277 | } |
| 278 | |
| 279 | return true; |
| 280 | } |
| 281 | |
| 282 | static bool one_shot_resample(const float* in, size_t samples_in, |
| 283 | unsigned rate, const char *resampler_ident, enum resampler_quality quality, |
| 284 | float** out, size_t* samples_out) |
| 285 | { |
| 286 | struct resampler_data info; |
| 287 | void* data = NULL; |
| 288 | const retro_resampler_t* resampler = NULL; |
| 289 | float ratio = (double)s_rate / (double)rate; |
| 290 | |
| 291 | if (!retro_resampler_realloc(&data, &resampler, |
| 292 | resampler_ident, quality, ratio)) |
| 293 | return false; |
| 294 | |
| 295 | /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We |
| 296 | * add 16 more samples in the formula below just as safeguard, because |
| 297 | * resampler->process sometimes reports more output samples than the |
| 298 | * formula below calculates. Ideally, audio resamplers should have a |
| 299 | * function to return the number of samples they will output given a |
| 300 | * count of input samples. */ |
| 301 | *samples_out = (size_t)(samples_in * ratio); |
| 302 | *out = (float*)memalign_alloc(16, |
| 303 | (((*samples_out + 16) + 15) & ~15) * sizeof(float)); |
| 304 | |
| 305 | if (*out == NULL) |
| 306 | return false; |
| 307 | |
| 308 | info.data_in = in; |
| 309 | info.data_out = *out; |
| 310 | info.input_frames = samples_in / 2; |
| 311 | info.output_frames = 0; |
| 312 | info.ratio = ratio; |
| 313 | |
| 314 | resampler->process(data, &info); |
| 315 | resampler->free(data); |
| 316 | return true; |
| 317 | } |
| 318 | #endif |
| 319 | |
| 320 | void audio_mixer_init(unsigned rate) |
| 321 | { |
| 322 | unsigned i; |
| 323 | |
| 324 | s_rate = rate; |
| 325 | |
| 326 | for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++) |
| 327 | { |
| 328 | audio_mixer_voice_t *voice = &s_voices[i]; |
| 329 | |
| 330 | voice->type = AUDIO_MIXER_TYPE_NONE; |
| 331 | #ifdef HAVE_THREADS |
| 332 | if (!voice->lock) |
| 333 | voice->lock = slock_new(); |
| 334 | #endif |
| 335 | } |
| 336 | } |
| 337 | |
| 338 | void audio_mixer_done(void) |
| 339 | { |
| 340 | unsigned i; |
| 341 | |
| 342 | for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++) |
| 343 | { |
| 344 | audio_mixer_voice_t *voice = &s_voices[i]; |
| 345 | |
| 346 | AUDIO_MIXER_LOCK(voice); |
| 347 | audio_mixer_release(voice); |
| 348 | AUDIO_MIXER_UNLOCK(voice); |
| 349 | #ifdef HAVE_THREADS |
| 350 | slock_free(voice->lock); |
| 351 | voice->lock = NULL; |
| 352 | #endif |
| 353 | } |
| 354 | } |
| 355 | |
| 356 | audio_mixer_sound_t* audio_mixer_load_wav(void *buffer, int32_t size, |
| 357 | const char *resampler_ident, enum resampler_quality quality) |
| 358 | { |
| 359 | #ifdef HAVE_RWAV |
| 360 | /* WAV data */ |
| 361 | rwav_t wav; |
| 362 | /* WAV samples converted to float */ |
| 363 | float* pcm = NULL; |
| 364 | size_t samples = 0; |
| 365 | /* Result */ |
| 366 | audio_mixer_sound_t* sound = NULL; |
| 367 | |
| 368 | wav.bitspersample = 0; |
| 369 | wav.numchannels = 0; |
| 370 | wav.samplerate = 0; |
| 371 | wav.numsamples = 0; |
| 372 | wav.subchunk2size = 0; |
| 373 | wav.samples = NULL; |
| 374 | |
| 375 | if ((rwav_load(&wav, buffer, size)) != RWAV_ITERATE_DONE) |
| 376 | return NULL; |
| 377 | |
| 378 | samples = wav.numsamples * 2; |
| 379 | |
| 380 | if (!wav_to_float(&wav, &pcm, samples)) |
| 381 | return NULL; |
| 382 | |
| 383 | if (wav.samplerate != s_rate) |
| 384 | { |
| 385 | float* resampled = NULL; |
| 386 | |
| 387 | if (!one_shot_resample(pcm, samples, wav.samplerate, |
| 388 | resampler_ident, quality, |
| 389 | &resampled, &samples)) |
| 390 | return NULL; |
| 391 | |
| 392 | memalign_free((void*)pcm); |
| 393 | pcm = resampled; |
| 394 | } |
| 395 | |
| 396 | sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound)); |
| 397 | |
| 398 | if (!sound) |
| 399 | { |
| 400 | memalign_free((void*)pcm); |
| 401 | return NULL; |
| 402 | } |
| 403 | |
| 404 | sound->type = AUDIO_MIXER_TYPE_WAV; |
| 405 | sound->types.wav.frames = (unsigned)(samples / 2); |
| 406 | sound->types.wav.pcm = pcm; |
| 407 | |
| 408 | rwav_free(&wav); |
| 409 | |
| 410 | return sound; |
| 411 | #else |
| 412 | return NULL; |
| 413 | #endif |
| 414 | } |
| 415 | |
| 416 | audio_mixer_sound_t* audio_mixer_load_ogg(void *buffer, int32_t size) |
| 417 | { |
| 418 | #ifdef HAVE_STB_VORBIS |
| 419 | audio_mixer_sound_t* sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound)); |
| 420 | |
| 421 | if (!sound) |
| 422 | return NULL; |
| 423 | |
| 424 | sound->type = AUDIO_MIXER_TYPE_OGG; |
| 425 | sound->types.ogg.size = size; |
| 426 | sound->types.ogg.data = buffer; |
| 427 | |
| 428 | return sound; |
| 429 | #else |
| 430 | return NULL; |
| 431 | #endif |
| 432 | } |
| 433 | |
| 434 | audio_mixer_sound_t* audio_mixer_load_flac(void *buffer, int32_t size) |
| 435 | { |
| 436 | #ifdef HAVE_DR_FLAC |
| 437 | audio_mixer_sound_t* sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound)); |
| 438 | |
| 439 | if (!sound) |
| 440 | return NULL; |
| 441 | |
| 442 | sound->type = AUDIO_MIXER_TYPE_FLAC; |
| 443 | sound->types.flac.size = size; |
| 444 | sound->types.flac.data = buffer; |
| 445 | |
| 446 | return sound; |
| 447 | #else |
| 448 | return NULL; |
| 449 | #endif |
| 450 | } |
| 451 | |
| 452 | audio_mixer_sound_t* audio_mixer_load_mp3(void *buffer, int32_t size) |
| 453 | { |
| 454 | #ifdef HAVE_DR_MP3 |
| 455 | audio_mixer_sound_t* sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound)); |
| 456 | |
| 457 | if (!sound) |
| 458 | return NULL; |
| 459 | |
| 460 | sound->type = AUDIO_MIXER_TYPE_MP3; |
| 461 | sound->types.mp3.size = size; |
| 462 | sound->types.mp3.data = buffer; |
| 463 | |
| 464 | return sound; |
| 465 | #else |
| 466 | return NULL; |
| 467 | #endif |
| 468 | } |
| 469 | |
| 470 | audio_mixer_sound_t* audio_mixer_load_mod(void *buffer, int32_t size) |
| 471 | { |
| 472 | #ifdef HAVE_IBXM |
| 473 | audio_mixer_sound_t* sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound)); |
| 474 | |
| 475 | if (!sound) |
| 476 | return NULL; |
| 477 | |
| 478 | sound->type = AUDIO_MIXER_TYPE_MOD; |
| 479 | sound->types.mod.size = size; |
| 480 | sound->types.mod.data = buffer; |
| 481 | |
| 482 | return sound; |
| 483 | #else |
| 484 | return NULL; |
| 485 | #endif |
| 486 | } |
| 487 | |
| 488 | void audio_mixer_destroy(audio_mixer_sound_t* sound) |
| 489 | { |
| 490 | void *handle = NULL; |
| 491 | if (!sound) |
| 492 | return; |
| 493 | |
| 494 | switch (sound->type) |
| 495 | { |
| 496 | case AUDIO_MIXER_TYPE_WAV: |
| 497 | handle = (void*)sound->types.wav.pcm; |
| 498 | if (handle) |
| 499 | memalign_free(handle); |
| 500 | break; |
| 501 | case AUDIO_MIXER_TYPE_OGG: |
| 502 | #ifdef HAVE_STB_VORBIS |
| 503 | handle = (void*)sound->types.ogg.data; |
| 504 | if (handle) |
| 505 | free(handle); |
| 506 | #endif |
| 507 | break; |
| 508 | case AUDIO_MIXER_TYPE_MOD: |
| 509 | #ifdef HAVE_IBXM |
| 510 | handle = (void*)sound->types.mod.data; |
| 511 | if (handle) |
| 512 | free(handle); |
| 513 | #endif |
| 514 | break; |
| 515 | case AUDIO_MIXER_TYPE_FLAC: |
| 516 | #ifdef HAVE_DR_FLAC |
| 517 | handle = (void*)sound->types.flac.data; |
| 518 | if (handle) |
| 519 | free(handle); |
| 520 | #endif |
| 521 | break; |
| 522 | case AUDIO_MIXER_TYPE_MP3: |
| 523 | #ifdef HAVE_DR_MP3 |
| 524 | handle = (void*)sound->types.mp3.data; |
| 525 | if (handle) |
| 526 | free(handle); |
| 527 | #endif |
| 528 | break; |
| 529 | case AUDIO_MIXER_TYPE_NONE: |
| 530 | break; |
| 531 | } |
| 532 | |
| 533 | free(sound); |
| 534 | } |
| 535 | |
| 536 | static bool audio_mixer_play_wav(audio_mixer_sound_t* sound, |
| 537 | audio_mixer_voice_t* voice, bool repeat, float volume, |
| 538 | audio_mixer_stop_cb_t stop_cb) |
| 539 | { |
| 540 | voice->types.wav.position = 0; |
| 541 | return true; |
| 542 | } |
| 543 | |
| 544 | #ifdef HAVE_STB_VORBIS |
| 545 | static bool audio_mixer_play_ogg( |
| 546 | audio_mixer_sound_t* sound, |
| 547 | audio_mixer_voice_t* voice, |
| 548 | bool repeat, float volume, |
| 549 | const char *resampler_ident, |
| 550 | enum resampler_quality quality, |
| 551 | audio_mixer_stop_cb_t stop_cb) |
| 552 | { |
| 553 | stb_vorbis_info info; |
| 554 | int res = 0; |
| 555 | float ratio = 1.0f; |
| 556 | unsigned samples = 0; |
| 557 | void *ogg_buffer = NULL; |
| 558 | void *resampler_data = NULL; |
| 559 | const retro_resampler_t* resamp = NULL; |
| 560 | stb_vorbis *stb_vorbis = stb_vorbis_open_memory( |
| 561 | (const unsigned char*)sound->types.ogg.data, |
| 562 | sound->types.ogg.size, &res, NULL); |
| 563 | |
| 564 | if (!stb_vorbis) |
| 565 | return false; |
| 566 | |
| 567 | info = stb_vorbis_get_info(stb_vorbis); |
| 568 | |
| 569 | if (info.sample_rate != s_rate) |
| 570 | { |
| 571 | ratio = (double)s_rate / (double)info.sample_rate; |
| 572 | |
| 573 | if (!retro_resampler_realloc(&resampler_data, |
| 574 | &resamp, resampler_ident, quality, |
| 575 | ratio)) |
| 576 | goto error; |
| 577 | } |
| 578 | |
| 579 | /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We |
| 580 | * add 16 more samples in the formula below just as safeguard, because |
| 581 | * resampler->process sometimes reports more output samples than the |
| 582 | * formula below calculates. Ideally, audio resamplers should have a |
| 583 | * function to return the number of samples they will output given a |
| 584 | * count of input samples. */ |
| 585 | samples = (unsigned)(AUDIO_MIXER_TEMP_BUFFER * ratio); |
| 586 | ogg_buffer = (float*)memalign_alloc(16, |
| 587 | (((samples + 16) + 15) & ~15) * sizeof(float)); |
| 588 | |
| 589 | if (!ogg_buffer) |
| 590 | { |
| 591 | if (resamp && resampler_data) |
| 592 | resamp->free(resampler_data); |
| 593 | goto error; |
| 594 | } |
| 595 | |
| 596 | voice->types.ogg.resampler = resamp; |
| 597 | voice->types.ogg.resampler_data = resampler_data; |
| 598 | voice->types.ogg.buffer = (float*)ogg_buffer; |
| 599 | voice->types.ogg.buf_samples = samples; |
| 600 | voice->types.ogg.ratio = ratio; |
| 601 | voice->types.ogg.stream = stb_vorbis; |
| 602 | voice->types.ogg.position = 0; |
| 603 | voice->types.ogg.samples = 0; |
| 604 | |
| 605 | return true; |
| 606 | |
| 607 | error: |
| 608 | stb_vorbis_close(stb_vorbis); |
| 609 | return false; |
| 610 | } |
| 611 | |
| 612 | static void audio_mixer_release_ogg(audio_mixer_voice_t* voice) |
| 613 | { |
| 614 | if (voice->types.ogg.stream) |
| 615 | stb_vorbis_close(voice->types.ogg.stream); |
| 616 | if (voice->types.ogg.resampler && voice->types.ogg.resampler_data) |
| 617 | voice->types.ogg.resampler->free(voice->types.ogg.resampler_data); |
| 618 | if (voice->types.ogg.buffer) |
| 619 | memalign_free(voice->types.ogg.buffer); |
| 620 | } |
| 621 | |
| 622 | #endif |
| 623 | |
| 624 | #ifdef HAVE_IBXM |
| 625 | static bool audio_mixer_play_mod( |
| 626 | audio_mixer_sound_t* sound, |
| 627 | audio_mixer_voice_t* voice, |
| 628 | bool repeat, float volume, |
| 629 | audio_mixer_stop_cb_t stop_cb) |
| 630 | { |
| 631 | struct data data; |
| 632 | char message[64]; |
| 633 | int buf_samples = 0; |
| 634 | int samples = 0; |
| 635 | void *mod_buffer = NULL; |
| 636 | struct module* module = NULL; |
| 637 | struct replay* replay = NULL; |
| 638 | |
| 639 | data.buffer = (char*)sound->types.mod.data; |
| 640 | data.length = sound->types.mod.size; |
| 641 | module = module_load(&data, message); |
| 642 | |
| 643 | if (!module) |
| 644 | { |
| 645 | printf("audio_mixer_play_mod module_load() failed with error: %s\n", message); |
| 646 | goto error; |
| 647 | } |
| 648 | |
| 649 | if (voice->types.mod.module) |
| 650 | dispose_module(voice->types.mod.module); |
| 651 | |
| 652 | voice->types.mod.module = module; |
| 653 | |
| 654 | replay = new_replay(module, s_rate, 1); |
| 655 | |
| 656 | if (!replay) |
| 657 | { |
| 658 | printf("audio_mixer_play_mod new_replay() failed\n"); |
| 659 | goto error; |
| 660 | } |
| 661 | |
| 662 | buf_samples = calculate_mix_buf_len(s_rate); |
| 663 | mod_buffer = memalign_alloc(16, ((buf_samples + 15) & ~15) * sizeof(int)); |
| 664 | |
| 665 | if (!mod_buffer) |
| 666 | { |
| 667 | printf("audio_mixer_play_mod cannot allocate mod_buffer !\n"); |
| 668 | goto error; |
| 669 | } |
| 670 | |
| 671 | samples = replay_calculate_duration(replay); |
| 672 | |
| 673 | if (!samples) |
| 674 | { |
| 675 | printf("audio_mixer_play_mod cannot retrieve duration !\n"); |
| 676 | goto error; |
| 677 | } |
| 678 | |
| 679 | voice->types.mod.buffer = (int*)mod_buffer; |
| 680 | voice->types.mod.buf_samples = buf_samples; |
| 681 | voice->types.mod.stream = replay; |
| 682 | voice->types.mod.position = 0; |
| 683 | voice->types.mod.samples = 0; /* samples; */ |
| 684 | |
| 685 | return true; |
| 686 | |
| 687 | error: |
| 688 | if (mod_buffer) |
| 689 | memalign_free(mod_buffer); |
| 690 | if (module) |
| 691 | dispose_module(module); |
| 692 | return false; |
| 693 | |
| 694 | } |
| 695 | |
| 696 | static void audio_mixer_release_mod(audio_mixer_voice_t* voice) |
| 697 | { |
| 698 | if (voice->types.mod.stream) |
| 699 | dispose_replay(voice->types.mod.stream); |
| 700 | if (voice->types.mod.buffer) |
| 701 | memalign_free(voice->types.mod.buffer); |
| 702 | } |
| 703 | #endif |
| 704 | |
| 705 | #ifdef HAVE_DR_FLAC |
| 706 | static bool audio_mixer_play_flac( |
| 707 | audio_mixer_sound_t* sound, |
| 708 | audio_mixer_voice_t* voice, |
| 709 | bool repeat, float volume, |
| 710 | const char *resampler_ident, |
| 711 | enum resampler_quality quality, |
| 712 | audio_mixer_stop_cb_t stop_cb) |
| 713 | { |
| 714 | float ratio = 1.0f; |
| 715 | unsigned samples = 0; |
| 716 | void *flac_buffer = NULL; |
| 717 | void *resampler_data = NULL; |
| 718 | const retro_resampler_t* resamp = NULL; |
| 719 | drflac *dr_flac = drflac_open_memory((const unsigned char*)sound->types.flac.data,sound->types.flac.size); |
| 720 | |
| 721 | if (!dr_flac) |
| 722 | return false; |
| 723 | if (dr_flac->sampleRate != s_rate) |
| 724 | { |
| 725 | ratio = (double)s_rate / (double)(dr_flac->sampleRate); |
| 726 | |
| 727 | if (!retro_resampler_realloc(&resampler_data, |
| 728 | &resamp, resampler_ident, quality, |
| 729 | ratio)) |
| 730 | goto error; |
| 731 | } |
| 732 | |
| 733 | /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We |
| 734 | * add 16 more samples in the formula below just as safeguard, because |
| 735 | * resampler->process sometimes reports more output samples than the |
| 736 | * formula below calculates. Ideally, audio resamplers should have a |
| 737 | * function to return the number of samples they will output given a |
| 738 | * count of input samples. */ |
| 739 | samples = (unsigned)(AUDIO_MIXER_TEMP_BUFFER * ratio); |
| 740 | flac_buffer = (float*)memalign_alloc(16, |
| 741 | (((samples + 16) + 15) & ~15) * sizeof(float)); |
| 742 | |
| 743 | if (!flac_buffer) |
| 744 | { |
| 745 | if (resamp && resamp->free) |
| 746 | resamp->free(resampler_data); |
| 747 | goto error; |
| 748 | } |
| 749 | |
| 750 | voice->types.flac.resampler = resamp; |
| 751 | voice->types.flac.resampler_data = resampler_data; |
| 752 | voice->types.flac.buffer = (float*)flac_buffer; |
| 753 | voice->types.flac.buf_samples = samples; |
| 754 | voice->types.flac.ratio = ratio; |
| 755 | voice->types.flac.stream = dr_flac; |
| 756 | voice->types.flac.position = 0; |
| 757 | voice->types.flac.samples = 0; |
| 758 | |
| 759 | return true; |
| 760 | |
| 761 | error: |
| 762 | drflac_close(dr_flac); |
| 763 | return false; |
| 764 | } |
| 765 | |
| 766 | static void audio_mixer_release_flac(audio_mixer_voice_t* voice) |
| 767 | { |
| 768 | if (voice->types.flac.stream) |
| 769 | drflac_close(voice->types.flac.stream); |
| 770 | if (voice->types.flac.resampler && voice->types.flac.resampler_data) |
| 771 | voice->types.flac.resampler->free(voice->types.flac.resampler_data); |
| 772 | if (voice->types.flac.buffer) |
| 773 | memalign_free(voice->types.flac.buffer); |
| 774 | } |
| 775 | #endif |
| 776 | |
| 777 | #ifdef HAVE_DR_MP3 |
| 778 | static bool audio_mixer_play_mp3( |
| 779 | audio_mixer_sound_t* sound, |
| 780 | audio_mixer_voice_t* voice, |
| 781 | bool repeat, float volume, |
| 782 | const char *resampler_ident, |
| 783 | enum resampler_quality quality, |
| 784 | audio_mixer_stop_cb_t stop_cb) |
| 785 | { |
| 786 | float ratio = 1.0f; |
| 787 | unsigned samples = 0; |
| 788 | void *mp3_buffer = NULL; |
| 789 | void *resampler_data = NULL; |
| 790 | const retro_resampler_t* resamp = NULL; |
| 791 | bool res; |
| 792 | |
| 793 | res = drmp3_init_memory(&voice->types.mp3.stream, (const unsigned char*)sound->types.mp3.data, sound->types.mp3.size, NULL); |
| 794 | |
| 795 | if (!res) |
| 796 | return false; |
| 797 | |
| 798 | if (voice->types.mp3.stream.sampleRate != s_rate) |
| 799 | { |
| 800 | ratio = (double)s_rate / (double)(voice->types.mp3.stream.sampleRate); |
| 801 | |
| 802 | if (!retro_resampler_realloc(&resampler_data, |
| 803 | &resamp, resampler_ident, quality, |
| 804 | ratio)) |
| 805 | goto error; |
| 806 | } |
| 807 | |
| 808 | /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We |
| 809 | * add 16 more samples in the formula below just as safeguard, because |
| 810 | * resampler->process sometimes reports more output samples than the |
| 811 | * formula below calculates. Ideally, audio resamplers should have a |
| 812 | * function to return the number of samples they will output given a |
| 813 | * count of input samples. */ |
| 814 | samples = (unsigned)(AUDIO_MIXER_TEMP_BUFFER * ratio); |
| 815 | mp3_buffer = (float*)memalign_alloc(16, |
| 816 | (((samples + 16) + 15) & ~15) * sizeof(float)); |
| 817 | |
| 818 | if (!mp3_buffer) |
| 819 | { |
| 820 | if (resamp && resampler_data) |
| 821 | resamp->free(resampler_data); |
| 822 | goto error; |
| 823 | } |
| 824 | |
| 825 | voice->types.mp3.resampler = resamp; |
| 826 | voice->types.mp3.resampler_data = resampler_data; |
| 827 | voice->types.mp3.buffer = (float*)mp3_buffer; |
| 828 | voice->types.mp3.buf_samples = samples; |
| 829 | voice->types.mp3.ratio = ratio; |
| 830 | voice->types.mp3.position = 0; |
| 831 | voice->types.mp3.samples = 0; |
| 832 | |
| 833 | return true; |
| 834 | |
| 835 | error: |
| 836 | drmp3_uninit(&voice->types.mp3.stream); |
| 837 | return false; |
| 838 | } |
| 839 | |
| 840 | static void audio_mixer_release_mp3(audio_mixer_voice_t* voice) |
| 841 | { |
| 842 | if (voice->types.mp3.resampler && voice->types.mp3.resampler_data) |
| 843 | voice->types.mp3.resampler->free(voice->types.mp3.resampler_data); |
| 844 | if (voice->types.mp3.buffer) |
| 845 | memalign_free(voice->types.mp3.buffer); |
| 846 | if (voice->types.mp3.stream.pData) |
| 847 | drmp3_uninit(&voice->types.mp3.stream); |
| 848 | } |
| 849 | |
| 850 | #endif |
| 851 | |
| 852 | audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, |
| 853 | bool repeat, float volume, |
| 854 | const char *resampler_ident, |
| 855 | enum resampler_quality quality, |
| 856 | audio_mixer_stop_cb_t stop_cb) |
| 857 | { |
| 858 | unsigned i; |
| 859 | bool res = false; |
| 860 | audio_mixer_voice_t* voice = s_voices; |
| 861 | |
| 862 | if (!sound) |
| 863 | return NULL; |
| 864 | |
| 865 | for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++, voice++) |
| 866 | { |
| 867 | if (voice->type != AUDIO_MIXER_TYPE_NONE) |
| 868 | continue; |
| 869 | |
| 870 | AUDIO_MIXER_LOCK(voice); |
| 871 | |
| 872 | if (voice->type != AUDIO_MIXER_TYPE_NONE) |
| 873 | { |
| 874 | AUDIO_MIXER_UNLOCK(voice); |
| 875 | continue; |
| 876 | } |
| 877 | |
| 878 | /* claim the voice, also helps with cleanup on error */ |
| 879 | voice->type = sound->type; |
| 880 | |
| 881 | switch (sound->type) |
| 882 | { |
| 883 | case AUDIO_MIXER_TYPE_WAV: |
| 884 | res = audio_mixer_play_wav(sound, voice, repeat, volume, stop_cb); |
| 885 | break; |
| 886 | case AUDIO_MIXER_TYPE_OGG: |
| 887 | #ifdef HAVE_STB_VORBIS |
| 888 | res = audio_mixer_play_ogg(sound, voice, repeat, volume, |
| 889 | resampler_ident, quality, stop_cb); |
| 890 | #endif |
| 891 | break; |
| 892 | case AUDIO_MIXER_TYPE_MOD: |
| 893 | #ifdef HAVE_IBXM |
| 894 | res = audio_mixer_play_mod(sound, voice, repeat, volume, stop_cb); |
| 895 | #endif |
| 896 | break; |
| 897 | case AUDIO_MIXER_TYPE_FLAC: |
| 898 | #ifdef HAVE_DR_FLAC |
| 899 | res = audio_mixer_play_flac(sound, voice, repeat, volume, |
| 900 | resampler_ident, quality, stop_cb); |
| 901 | #endif |
| 902 | break; |
| 903 | case AUDIO_MIXER_TYPE_MP3: |
| 904 | #ifdef HAVE_DR_MP3 |
| 905 | res = audio_mixer_play_mp3(sound, voice, repeat, volume, |
| 906 | resampler_ident, quality, stop_cb); |
| 907 | #endif |
| 908 | break; |
| 909 | case AUDIO_MIXER_TYPE_NONE: |
| 910 | break; |
| 911 | } |
| 912 | |
| 913 | break; |
| 914 | } |
| 915 | |
| 916 | if (res) |
| 917 | { |
| 918 | voice->repeat = repeat; |
| 919 | voice->volume = volume; |
| 920 | voice->sound = sound; |
| 921 | voice->stop_cb = stop_cb; |
| 922 | AUDIO_MIXER_UNLOCK(voice); |
| 923 | } |
| 924 | else |
| 925 | { |
| 926 | if (i < AUDIO_MIXER_MAX_VOICES) |
| 927 | { |
| 928 | audio_mixer_release(voice); |
| 929 | AUDIO_MIXER_UNLOCK(voice); |
| 930 | } |
| 931 | voice = NULL; |
| 932 | } |
| 933 | |
| 934 | return voice; |
| 935 | } |
| 936 | |
| 937 | /* Need to hold lock for voice. */ |
| 938 | static void audio_mixer_release(audio_mixer_voice_t* voice) |
| 939 | { |
| 940 | if (!voice) |
| 941 | return; |
| 942 | |
| 943 | switch (voice->type) |
| 944 | { |
| 945 | #ifdef HAVE_STB_VORBIS |
| 946 | case AUDIO_MIXER_TYPE_OGG: |
| 947 | audio_mixer_release_ogg(voice); |
| 948 | break; |
| 949 | #endif |
| 950 | #ifdef HAVE_IBXM |
| 951 | case AUDIO_MIXER_TYPE_MOD: |
| 952 | audio_mixer_release_mod(voice); |
| 953 | break; |
| 954 | #endif |
| 955 | #ifdef HAVE_DR_FLAC |
| 956 | case AUDIO_MIXER_TYPE_FLAC: |
| 957 | audio_mixer_release_flac(voice); |
| 958 | break; |
| 959 | #endif |
| 960 | #ifdef HAVE_DR_MP3 |
| 961 | case AUDIO_MIXER_TYPE_MP3: |
| 962 | audio_mixer_release_mp3(voice); |
| 963 | break; |
| 964 | #endif |
| 965 | default: |
| 966 | break; |
| 967 | } |
| 968 | |
| 969 | memset(&voice->types, 0, sizeof(voice->types)); |
| 970 | voice->type = AUDIO_MIXER_TYPE_NONE; |
| 971 | } |
| 972 | |
| 973 | void audio_mixer_stop(audio_mixer_voice_t* voice) |
| 974 | { |
| 975 | audio_mixer_stop_cb_t stop_cb = NULL; |
| 976 | audio_mixer_sound_t* sound = NULL; |
| 977 | |
| 978 | if (voice) |
| 979 | { |
| 980 | AUDIO_MIXER_LOCK(voice); |
| 981 | stop_cb = voice->stop_cb; |
| 982 | sound = voice->sound; |
| 983 | |
| 984 | audio_mixer_release(voice); |
| 985 | |
| 986 | AUDIO_MIXER_UNLOCK(voice); |
| 987 | |
| 988 | if (stop_cb) |
| 989 | stop_cb(sound, AUDIO_MIXER_SOUND_STOPPED); |
| 990 | } |
| 991 | } |
| 992 | |
| 993 | static void audio_mixer_mix_wav(float* buffer, size_t num_frames, |
| 994 | audio_mixer_voice_t* voice, |
| 995 | float volume) |
| 996 | { |
| 997 | int i; |
| 998 | unsigned buf_free = (unsigned)(num_frames * 2); |
| 999 | const audio_mixer_sound_t* sound = voice->sound; |
| 1000 | unsigned pcm_available = sound->types.wav.frames |
| 1001 | * 2 - voice->types.wav.position; |
| 1002 | const float* pcm = sound->types.wav.pcm + |
| 1003 | voice->types.wav.position; |
| 1004 | |
| 1005 | again: |
| 1006 | if (pcm_available < buf_free) |
| 1007 | { |
| 1008 | for (i = pcm_available; i != 0; i--) |
| 1009 | *buffer++ += *pcm++ * volume; |
| 1010 | |
| 1011 | if (voice->repeat) |
| 1012 | { |
| 1013 | if (voice->stop_cb) |
| 1014 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED); |
| 1015 | |
| 1016 | buf_free -= pcm_available; |
| 1017 | pcm_available = sound->types.wav.frames * 2; |
| 1018 | pcm = sound->types.wav.pcm; |
| 1019 | voice->types.wav.position = 0; |
| 1020 | goto again; |
| 1021 | } |
| 1022 | |
| 1023 | if (voice->stop_cb) |
| 1024 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED); |
| 1025 | |
| 1026 | audio_mixer_release(voice); |
| 1027 | } |
| 1028 | else |
| 1029 | { |
| 1030 | for (i = buf_free; i != 0; i--) |
| 1031 | *buffer++ += *pcm++ * volume; |
| 1032 | |
| 1033 | voice->types.wav.position += buf_free; |
| 1034 | } |
| 1035 | } |
| 1036 | |
| 1037 | #ifdef HAVE_STB_VORBIS |
| 1038 | static void audio_mixer_mix_ogg(float* buffer, size_t num_frames, |
| 1039 | audio_mixer_voice_t* voice, |
| 1040 | float volume) |
| 1041 | { |
| 1042 | int i; |
| 1043 | float* temp_buffer = NULL; |
| 1044 | unsigned buf_free = (unsigned)(num_frames * 2); |
| 1045 | unsigned temp_samples = 0; |
| 1046 | float* pcm = NULL; |
| 1047 | |
| 1048 | if (!voice->types.ogg.stream) |
| 1049 | return; |
| 1050 | |
| 1051 | if (voice->types.ogg.position == voice->types.ogg.samples) |
| 1052 | { |
| 1053 | again: |
| 1054 | if (temp_buffer == NULL) |
| 1055 | temp_buffer = (float*)malloc(AUDIO_MIXER_TEMP_BUFFER * sizeof(float)); |
| 1056 | |
| 1057 | temp_samples = stb_vorbis_get_samples_float_interleaved( |
| 1058 | voice->types.ogg.stream, 2, temp_buffer, |
| 1059 | AUDIO_MIXER_TEMP_BUFFER) * 2; |
| 1060 | |
| 1061 | if (temp_samples == 0) |
| 1062 | { |
| 1063 | if (voice->repeat) |
| 1064 | { |
| 1065 | if (voice->stop_cb) |
| 1066 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED); |
| 1067 | |
| 1068 | stb_vorbis_seek_start(voice->types.ogg.stream); |
| 1069 | goto again; |
| 1070 | } |
| 1071 | |
| 1072 | if (voice->stop_cb) |
| 1073 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED); |
| 1074 | |
| 1075 | audio_mixer_release(voice); |
| 1076 | goto cleanup; |
| 1077 | } |
| 1078 | |
| 1079 | if (voice->types.ogg.resampler) |
| 1080 | { |
| 1081 | struct resampler_data info; |
| 1082 | info.data_in = temp_buffer; |
| 1083 | info.data_out = voice->types.ogg.buffer; |
| 1084 | info.input_frames = temp_samples / 2; |
| 1085 | info.output_frames = 0; |
| 1086 | info.ratio = voice->types.ogg.ratio; |
| 1087 | |
| 1088 | voice->types.ogg.resampler->process( |
| 1089 | voice->types.ogg.resampler_data, &info); |
| 1090 | } |
| 1091 | else |
| 1092 | memcpy(voice->types.ogg.buffer, temp_buffer, |
| 1093 | temp_samples * sizeof(float)); |
| 1094 | |
| 1095 | voice->types.ogg.position = 0; |
| 1096 | voice->types.ogg.samples = voice->types.ogg.buf_samples; |
| 1097 | } |
| 1098 | |
| 1099 | pcm = voice->types.ogg.buffer + voice->types.ogg.position; |
| 1100 | |
| 1101 | if (voice->types.ogg.samples < buf_free) |
| 1102 | { |
| 1103 | for (i = voice->types.ogg.samples; i != 0; i--) |
| 1104 | *buffer++ += *pcm++ * volume; |
| 1105 | |
| 1106 | buf_free -= voice->types.ogg.samples; |
| 1107 | goto again; |
| 1108 | } |
| 1109 | |
| 1110 | for (i = buf_free; i != 0; --i ) |
| 1111 | *buffer++ += *pcm++ * volume; |
| 1112 | |
| 1113 | voice->types.ogg.position += buf_free; |
| 1114 | voice->types.ogg.samples -= buf_free; |
| 1115 | |
| 1116 | cleanup: |
| 1117 | if (temp_buffer != NULL) |
| 1118 | free(temp_buffer); |
| 1119 | } |
| 1120 | #endif |
| 1121 | |
| 1122 | #ifdef HAVE_IBXM |
| 1123 | static void audio_mixer_mix_mod(float* buffer, size_t num_frames, |
| 1124 | audio_mixer_voice_t* voice, |
| 1125 | float volume) |
| 1126 | { |
| 1127 | int i; |
| 1128 | float samplef = 0.0f; |
| 1129 | unsigned temp_samples = 0; |
| 1130 | unsigned buf_free = (unsigned)(num_frames * 2); |
| 1131 | int* pcm = NULL; |
| 1132 | |
| 1133 | if (voice->types.mod.samples == 0) |
| 1134 | { |
| 1135 | again: |
| 1136 | temp_samples = replay_get_audio( |
| 1137 | voice->types.mod.stream, voice->types.mod.buffer, 0 ) * 2; |
| 1138 | |
| 1139 | if (temp_samples == 0) |
| 1140 | { |
| 1141 | if (voice->repeat) |
| 1142 | { |
| 1143 | if (voice->stop_cb) |
| 1144 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED); |
| 1145 | |
| 1146 | replay_seek( voice->types.mod.stream, 0); |
| 1147 | goto again; |
| 1148 | } |
| 1149 | |
| 1150 | if (voice->stop_cb) |
| 1151 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED); |
| 1152 | |
| 1153 | audio_mixer_release(voice); |
| 1154 | return; |
| 1155 | } |
| 1156 | |
| 1157 | voice->types.mod.position = 0; |
| 1158 | voice->types.mod.samples = temp_samples; |
| 1159 | } |
| 1160 | pcm = voice->types.mod.buffer + voice->types.mod.position; |
| 1161 | |
| 1162 | if (voice->types.mod.samples < buf_free) |
| 1163 | { |
| 1164 | for (i = voice->types.mod.samples; i != 0; i--) |
| 1165 | { |
| 1166 | samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f; |
| 1167 | samplef = samplef * 2.0f - 1.0f; |
| 1168 | *buffer++ += samplef * volume; |
| 1169 | } |
| 1170 | |
| 1171 | buf_free -= voice->types.mod.samples; |
| 1172 | goto again; |
| 1173 | } |
| 1174 | |
| 1175 | for (i = buf_free; i != 0; --i ) |
| 1176 | { |
| 1177 | samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f; |
| 1178 | samplef = samplef * 2.0f - 1.0f; |
| 1179 | *buffer++ += samplef * volume; |
| 1180 | } |
| 1181 | |
| 1182 | voice->types.mod.position += buf_free; |
| 1183 | voice->types.mod.samples -= buf_free; |
| 1184 | } |
| 1185 | #endif |
| 1186 | |
| 1187 | #ifdef HAVE_DR_FLAC |
| 1188 | static void audio_mixer_mix_flac(float* buffer, size_t num_frames, |
| 1189 | audio_mixer_voice_t* voice, |
| 1190 | float volume) |
| 1191 | { |
| 1192 | int i; |
| 1193 | struct resampler_data info; |
| 1194 | float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 }; |
| 1195 | unsigned buf_free = (unsigned)(num_frames * 2); |
| 1196 | unsigned temp_samples = 0; |
| 1197 | float *pcm = NULL; |
| 1198 | |
| 1199 | if (voice->types.flac.position == voice->types.flac.samples) |
| 1200 | { |
| 1201 | again: |
| 1202 | temp_samples = (unsigned)drflac_read_f32( voice->types.flac.stream, AUDIO_MIXER_TEMP_BUFFER, temp_buffer); |
| 1203 | if (temp_samples == 0) |
| 1204 | { |
| 1205 | if (voice->repeat) |
| 1206 | { |
| 1207 | if (voice->stop_cb) |
| 1208 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED); |
| 1209 | |
| 1210 | drflac_seek_to_sample(voice->types.flac.stream,0); |
| 1211 | goto again; |
| 1212 | } |
| 1213 | |
| 1214 | if (voice->stop_cb) |
| 1215 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED); |
| 1216 | |
| 1217 | audio_mixer_release(voice); |
| 1218 | return; |
| 1219 | } |
| 1220 | |
| 1221 | info.data_in = temp_buffer; |
| 1222 | info.data_out = voice->types.flac.buffer; |
| 1223 | info.input_frames = temp_samples / 2; |
| 1224 | info.output_frames = 0; |
| 1225 | info.ratio = voice->types.flac.ratio; |
| 1226 | |
| 1227 | if (voice->types.flac.resampler) |
| 1228 | voice->types.flac.resampler->process( |
| 1229 | voice->types.flac.resampler_data, &info); |
| 1230 | else |
| 1231 | memcpy(voice->types.flac.buffer, temp_buffer, temp_samples * sizeof(float)); |
| 1232 | voice->types.flac.position = 0; |
| 1233 | voice->types.flac.samples = voice->types.flac.buf_samples; |
| 1234 | } |
| 1235 | |
| 1236 | pcm = voice->types.flac.buffer + voice->types.flac.position; |
| 1237 | |
| 1238 | if (voice->types.flac.samples < buf_free) |
| 1239 | { |
| 1240 | for (i = voice->types.flac.samples; i != 0; i--) |
| 1241 | *buffer++ += *pcm++ * volume; |
| 1242 | |
| 1243 | buf_free -= voice->types.flac.samples; |
| 1244 | goto again; |
| 1245 | } |
| 1246 | |
| 1247 | for (i = buf_free; i != 0; --i ) |
| 1248 | *buffer++ += *pcm++ * volume; |
| 1249 | |
| 1250 | voice->types.flac.position += buf_free; |
| 1251 | voice->types.flac.samples -= buf_free; |
| 1252 | } |
| 1253 | #endif |
| 1254 | |
| 1255 | #ifdef HAVE_DR_MP3 |
| 1256 | static void audio_mixer_mix_mp3(float* buffer, size_t num_frames, |
| 1257 | audio_mixer_voice_t* voice, |
| 1258 | float volume) |
| 1259 | { |
| 1260 | int i; |
| 1261 | struct resampler_data info; |
| 1262 | float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 }; |
| 1263 | unsigned buf_free = (unsigned)(num_frames * 2); |
| 1264 | unsigned temp_samples = 0; |
| 1265 | float* pcm = NULL; |
| 1266 | |
| 1267 | if (voice->types.mp3.position == voice->types.mp3.samples) |
| 1268 | { |
| 1269 | again: |
| 1270 | temp_samples = (unsigned)drmp3_read_f32( |
| 1271 | &voice->types.mp3.stream, |
| 1272 | AUDIO_MIXER_TEMP_BUFFER / 2, temp_buffer) * 2; |
| 1273 | |
| 1274 | if (temp_samples == 0) |
| 1275 | { |
| 1276 | if (voice->repeat) |
| 1277 | { |
| 1278 | if (voice->stop_cb) |
| 1279 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED); |
| 1280 | |
| 1281 | drmp3_seek_to_frame(&voice->types.mp3.stream,0); |
| 1282 | goto again; |
| 1283 | } |
| 1284 | |
| 1285 | if (voice->stop_cb) |
| 1286 | voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED); |
| 1287 | |
| 1288 | audio_mixer_release(voice); |
| 1289 | return; |
| 1290 | } |
| 1291 | |
| 1292 | info.data_in = temp_buffer; |
| 1293 | info.data_out = voice->types.mp3.buffer; |
| 1294 | info.input_frames = temp_samples / 2; |
| 1295 | info.output_frames = 0; |
| 1296 | info.ratio = voice->types.mp3.ratio; |
| 1297 | |
| 1298 | if (voice->types.mp3.resampler) |
| 1299 | voice->types.mp3.resampler->process( |
| 1300 | voice->types.mp3.resampler_data, &info); |
| 1301 | else |
| 1302 | memcpy(voice->types.mp3.buffer, temp_buffer, |
| 1303 | temp_samples * sizeof(float)); |
| 1304 | voice->types.mp3.position = 0; |
| 1305 | voice->types.mp3.samples = voice->types.mp3.buf_samples; |
| 1306 | } |
| 1307 | |
| 1308 | pcm = voice->types.mp3.buffer + voice->types.mp3.position; |
| 1309 | |
| 1310 | if (voice->types.mp3.samples < buf_free) |
| 1311 | { |
| 1312 | for (i = voice->types.mp3.samples; i != 0; i--) |
| 1313 | *buffer++ += *pcm++ * volume; |
| 1314 | |
| 1315 | buf_free -= voice->types.mp3.samples; |
| 1316 | goto again; |
| 1317 | } |
| 1318 | |
| 1319 | for (i = buf_free; i != 0; --i ) |
| 1320 | *buffer++ += *pcm++ * volume; |
| 1321 | |
| 1322 | voice->types.mp3.position += buf_free; |
| 1323 | voice->types.mp3.samples -= buf_free; |
| 1324 | } |
| 1325 | #endif |
| 1326 | |
| 1327 | void audio_mixer_mix(float* buffer, size_t num_frames, |
| 1328 | float volume_override, bool override) |
| 1329 | { |
| 1330 | unsigned i; |
| 1331 | size_t j = 0; |
| 1332 | float* sample = NULL; |
| 1333 | audio_mixer_voice_t* voice = s_voices; |
| 1334 | |
| 1335 | for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++, voice++) |
| 1336 | { |
| 1337 | float volume; |
| 1338 | |
| 1339 | AUDIO_MIXER_LOCK(voice); |
| 1340 | |
| 1341 | volume = (override) ? volume_override : voice->volume; |
| 1342 | |
| 1343 | switch (voice->type) |
| 1344 | { |
| 1345 | case AUDIO_MIXER_TYPE_WAV: |
| 1346 | audio_mixer_mix_wav(buffer, num_frames, voice, volume); |
| 1347 | break; |
| 1348 | case AUDIO_MIXER_TYPE_OGG: |
| 1349 | #ifdef HAVE_STB_VORBIS |
| 1350 | audio_mixer_mix_ogg(buffer, num_frames, voice, volume); |
| 1351 | #endif |
| 1352 | break; |
| 1353 | case AUDIO_MIXER_TYPE_MOD: |
| 1354 | #ifdef HAVE_IBXM |
| 1355 | audio_mixer_mix_mod(buffer, num_frames, voice, volume); |
| 1356 | #endif |
| 1357 | break; |
| 1358 | case AUDIO_MIXER_TYPE_FLAC: |
| 1359 | #ifdef HAVE_DR_FLAC |
| 1360 | audio_mixer_mix_flac(buffer, num_frames, voice, volume); |
| 1361 | #endif |
| 1362 | break; |
| 1363 | case AUDIO_MIXER_TYPE_MP3: |
| 1364 | #ifdef HAVE_DR_MP3 |
| 1365 | audio_mixer_mix_mp3(buffer, num_frames, voice, volume); |
| 1366 | #endif |
| 1367 | break; |
| 1368 | case AUDIO_MIXER_TYPE_NONE: |
| 1369 | break; |
| 1370 | } |
| 1371 | |
| 1372 | AUDIO_MIXER_UNLOCK(voice); |
| 1373 | } |
| 1374 | |
| 1375 | for (j = 0, sample = buffer; j < num_frames * 2; j++, sample++) |
| 1376 | { |
| 1377 | if (*sample < -1.0f) |
| 1378 | *sample = -1.0f; |
| 1379 | else if (*sample > 1.0f) |
| 1380 | *sample = 1.0f; |
| 1381 | } |
| 1382 | } |
| 1383 | |
| 1384 | float audio_mixer_voice_get_volume(audio_mixer_voice_t *voice) |
| 1385 | { |
| 1386 | if (!voice) |
| 1387 | return 0.0f; |
| 1388 | |
| 1389 | return voice->volume; |
| 1390 | } |
| 1391 | |
| 1392 | void audio_mixer_voice_set_volume(audio_mixer_voice_t *voice, float val) |
| 1393 | { |
| 1394 | if (!voice) |
| 1395 | return; |
| 1396 | |
| 1397 | AUDIO_MIXER_LOCK(voice); |
| 1398 | voice->volume = val; |
| 1399 | AUDIO_MIXER_UNLOCK(voice); |
| 1400 | } |