1 /* 2 ** 3 ** Copyright 2007, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 #define LOG_TAG "AudioMixer" 19 //#define LOG_NDEBUG 0 20 21 #include "Configuration.h" 22 #include <stdint.h> 23 #include <string.h> 24 #include <stdlib.h> 25 #include <sys/types.h> 26 27 #include <utils/Errors.h> 28 #include <utils/Log.h> 29 30 #include <cutils/bitops.h> 31 #include <cutils/compiler.h> 32 #include <utils/Debug.h> 33 34 #include <system/audio.h> 35 36 #include <audio_utils/primitives.h> 37 #include <common_time/local_clock.h> 38 #include <common_time/cc_helper.h> 39 40 #include <media/EffectsFactoryApi.h> 41 42 #include "AudioMixer.h" 43 44 namespace android { 45 46 // ---------------------------------------------------------------------------- 47 AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider() : AudioBufferProvider(), 48 mTrackBufferProvider(NULL), mDownmixHandle(NULL) 49 { 50 } 51 52 AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider() 53 { 54 ALOGV("AudioMixer deleting DownmixerBufferProvider (%p)", this); 55 EffectRelease(mDownmixHandle); 56 } 57 58 status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, 59 int64_t pts) { 60 //ALOGV("DownmixerBufferProvider::getNextBuffer()"); 61 if (this->mTrackBufferProvider != NULL) { 62 status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts); 63 if (res == OK) { 64 mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount; 65 mDownmixConfig.inputCfg.buffer.raw = pBuffer->raw; 66 mDownmixConfig.outputCfg.buffer.frameCount = pBuffer->frameCount; 67 mDownmixConfig.outputCfg.buffer.raw = mDownmixConfig.inputCfg.buffer.raw; 68 // in-place so overwrite the buffer contents, has been set in prepareTrackForDownmix() 69 //mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 70 71 res = (*mDownmixHandle)->process(mDownmixHandle, 72 &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer); 73 //ALOGV("getNextBuffer is downmixing"); 74 } 75 return res; 76 } else { 77 ALOGE("DownmixerBufferProvider::getNextBuffer() error: NULL track buffer provider"); 78 return NO_INIT; 79 } 80 } 81 82 void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) { 83 //ALOGV("DownmixerBufferProvider::releaseBuffer()"); 84 if (this->mTrackBufferProvider != NULL) { 85 mTrackBufferProvider->releaseBuffer(pBuffer); 86 } else { 87 ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider"); 88 } 89 } 90 91 92 // ---------------------------------------------------------------------------- 93 bool AudioMixer::isMultichannelCapable = false; 94 95 effect_descriptor_t AudioMixer::dwnmFxDesc; 96 97 // Ensure mConfiguredNames bitmask is initialized properly on all architectures. 98 // The value of 1 << x is undefined in C when x >= 32. 99 100 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) 101 : mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1), 102 mSampleRate(sampleRate) 103 { 104 // AudioMixer is not yet capable of multi-channel beyond stereo 105 COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS); 106 107 ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u", 108 maxNumTracks, MAX_NUM_TRACKS); 109 110 // AudioMixer is not yet capable of more than 32 active track inputs 111 ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS); 112 113 // AudioMixer is not yet capable of multi-channel output beyond stereo 114 ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS); 115 116 LocalClock lc; 117 118 pthread_once(&sOnceControl, &sInitRoutine); 119 120 mState.enabledTracks= 0; 121 mState.needsChanged = 0; 122 mState.frameCount = frameCount; 123 mState.hook = process__nop; 124 mState.outputTemp = NULL; 125 mState.resampleTemp = NULL; 126 mState.mLog = &mDummyLog; 127 // mState.reserved 128 129 // FIXME Most of the following initialization is probably redundant since 130 // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0 131 // and mTrackNames is initially 0. However, leave it here until that's verified. 132 track_t* t = mState.tracks; 133 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 134 t->resampler = NULL; 135 t->downmixerBufferProvider = NULL; 136 t++; 137 } 138 139 // find multichannel downmix effect if we have to play multichannel content 140 uint32_t numEffects = 0; 141 int ret = EffectQueryNumberEffects(&numEffects); 142 if (ret != 0) { 143 ALOGE("AudioMixer() error %d querying number of effects", ret); 144 return; 145 } 146 ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); 147 148 for (uint32_t i = 0 ; i < numEffects ; i++) { 149 if (EffectQueryEffect(i, &dwnmFxDesc) == 0) { 150 ALOGV("effect %d is called %s", i, dwnmFxDesc.name); 151 if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { 152 ALOGI("found effect \"%s\" from %s", 153 dwnmFxDesc.name, dwnmFxDesc.implementor); 154 isMultichannelCapable = true; 155 break; 156 } 157 } 158 } 159 ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect"); 160 } 161 162 AudioMixer::~AudioMixer() 163 { 164 track_t* t = mState.tracks; 165 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 166 delete t->resampler; 167 delete t->downmixerBufferProvider; 168 t++; 169 } 170 delete [] mState.outputTemp; 171 delete [] mState.resampleTemp; 172 } 173 174 void AudioMixer::setLog(NBLog::Writer *log) 175 { 176 mState.mLog = log; 177 } 178 179 int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) 180 { 181 uint32_t names = (~mTrackNames) & mConfiguredNames; 182 if (names != 0) { 183 int n = __builtin_ctz(names); 184 ALOGV("add track (%d)", n); 185 mTrackNames |= 1 << n; 186 // assume default parameters for the track, except where noted below 187 track_t* t = &mState.tracks[n]; 188 t->needs = 0; 189 t->volume[0] = UNITY_GAIN; 190 t->volume[1] = UNITY_GAIN; 191 // no initialization needed 192 // t->prevVolume[0] 193 // t->prevVolume[1] 194 t->volumeInc[0] = 0; 195 t->volumeInc[1] = 0; 196 t->auxLevel = 0; 197 t->auxInc = 0; 198 // no initialization needed 199 // t->prevAuxLevel 200 // t->frameCount 201 t->channelCount = 2; 202 t->enabled = false; 203 t->format = 16; 204 t->channelMask = AUDIO_CHANNEL_OUT_STEREO; 205 t->sessionId = sessionId; 206 // setBufferProvider(name, AudioBufferProvider *) is required before enable(name) 207 t->bufferProvider = NULL; 208 t->buffer.raw = NULL; 209 // no initialization needed 210 // t->buffer.frameCount 211 t->hook = NULL; 212 t->in = NULL; 213 t->resampler = NULL; 214 t->sampleRate = mSampleRate; 215 // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name) 216 t->mainBuffer = NULL; 217 t->auxBuffer = NULL; 218 t->downmixerBufferProvider = NULL; 219 220 status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask); 221 if (status == OK) { 222 return TRACK0 + n; 223 } 224 ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix", 225 channelMask); 226 } 227 return -1; 228 } 229 230 void AudioMixer::invalidateState(uint32_t mask) 231 { 232 if (mask) { 233 mState.needsChanged |= mask; 234 mState.hook = process__validate; 235 } 236 } 237 238 status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask) 239 { 240 uint32_t channelCount = popcount(mask); 241 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 242 status_t status = OK; 243 if (channelCount > MAX_NUM_CHANNELS) { 244 pTrack->channelMask = mask; 245 pTrack->channelCount = channelCount; 246 ALOGV("initTrackDownmix(track=%d, mask=0x%x) calls prepareTrackForDownmix()", 247 trackNum, mask); 248 status = prepareTrackForDownmix(pTrack, trackNum); 249 } else { 250 unprepareTrackForDownmix(pTrack, trackNum); 251 } 252 return status; 253 } 254 255 void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) { 256 ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName); 257 258 if (pTrack->downmixerBufferProvider != NULL) { 259 // this track had previously been configured with a downmixer, delete it 260 ALOGV(" deleting old downmixer"); 261 pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider; 262 delete pTrack->downmixerBufferProvider; 263 pTrack->downmixerBufferProvider = NULL; 264 } else { 265 ALOGV(" nothing to do, no downmixer to delete"); 266 } 267 } 268 269 status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName) 270 { 271 ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask); 272 273 // discard the previous downmixer if there was one 274 unprepareTrackForDownmix(pTrack, trackName); 275 276 DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(); 277 int32_t status; 278 279 if (!isMultichannelCapable) { 280 ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content", 281 trackName); 282 goto noDownmixForActiveTrack; 283 } 284 285 if (EffectCreate(&dwnmFxDesc.uuid, 286 pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/, 287 &pDbp->mDownmixHandle/*pHandle*/) != 0) { 288 ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName); 289 goto noDownmixForActiveTrack; 290 } 291 292 // channel input configuration will be overridden per-track 293 pDbp->mDownmixConfig.inputCfg.channels = pTrack->channelMask; 294 pDbp->mDownmixConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; 295 pDbp->mDownmixConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 296 pDbp->mDownmixConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 297 pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate; 298 pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate; 299 pDbp->mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 300 pDbp->mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 301 // input and output buffer provider, and frame count will not be used as the downmix effect 302 // process() function is called directly (see DownmixerBufferProvider::getNextBuffer()) 303 pDbp->mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | 304 EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE; 305 pDbp->mDownmixConfig.outputCfg.mask = pDbp->mDownmixConfig.inputCfg.mask; 306 307 {// scope for local variables that are not used in goto label "noDownmixForActiveTrack" 308 int cmdStatus; 309 uint32_t replySize = sizeof(int); 310 311 // Configure and enable downmixer 312 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 313 EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/, 314 &pDbp->mDownmixConfig /*pCmdData*/, 315 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 316 if ((status != 0) || (cmdStatus != 0)) { 317 ALOGE("error %d while configuring downmixer for track %d", status, trackName); 318 goto noDownmixForActiveTrack; 319 } 320 replySize = sizeof(int); 321 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 322 EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/, 323 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 324 if ((status != 0) || (cmdStatus != 0)) { 325 ALOGE("error %d while enabling downmixer for track %d", status, trackName); 326 goto noDownmixForActiveTrack; 327 } 328 329 // Set downmix type 330 // parameter size rounded for padding on 32bit boundary 331 const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int); 332 const int downmixParamSize = 333 sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t); 334 effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize); 335 param->psize = sizeof(downmix_params_t); 336 const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE; 337 memcpy(param->data, &downmixParam, param->psize); 338 const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD; 339 param->vsize = sizeof(downmix_type_t); 340 memcpy(param->data + psizePadded, &downmixType, param->vsize); 341 342 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 343 EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize/* cmdSize */, 344 param /*pCmndData*/, &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 345 346 free(param); 347 348 if ((status != 0) || (cmdStatus != 0)) { 349 ALOGE("error %d while setting downmix type for track %d", status, trackName); 350 goto noDownmixForActiveTrack; 351 } else { 352 ALOGV("downmix type set to %d for track %d", (int) downmixType, trackName); 353 } 354 }// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack" 355 356 // initialization successful: 357 // - keep track of the real buffer provider in case it was set before 358 pDbp->mTrackBufferProvider = pTrack->bufferProvider; 359 // - we'll use the downmix effect integrated inside this 360 // track's buffer provider, and we'll use it as the track's buffer provider 361 pTrack->downmixerBufferProvider = pDbp; 362 pTrack->bufferProvider = pDbp; 363 364 return NO_ERROR; 365 366 noDownmixForActiveTrack: 367 delete pDbp; 368 pTrack->downmixerBufferProvider = NULL; 369 return NO_INIT; 370 } 371 372 void AudioMixer::deleteTrackName(int name) 373 { 374 ALOGV("AudioMixer::deleteTrackName(%d)", name); 375 name -= TRACK0; 376 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 377 ALOGV("deleteTrackName(%d)", name); 378 track_t& track(mState.tracks[ name ]); 379 if (track.enabled) { 380 track.enabled = false; 381 invalidateState(1<<name); 382 } 383 // delete the resampler 384 delete track.resampler; 385 track.resampler = NULL; 386 // delete the downmixer 387 unprepareTrackForDownmix(&mState.tracks[name], name); 388 389 mTrackNames &= ~(1<<name); 390 } 391 392 void AudioMixer::enable(int name) 393 { 394 name -= TRACK0; 395 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 396 track_t& track = mState.tracks[name]; 397 398 if (!track.enabled) { 399 track.enabled = true; 400 ALOGV("enable(%d)", name); 401 invalidateState(1 << name); 402 } 403 } 404 405 void AudioMixer::disable(int name) 406 { 407 name -= TRACK0; 408 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 409 track_t& track = mState.tracks[name]; 410 411 if (track.enabled) { 412 track.enabled = false; 413 ALOGV("disable(%d)", name); 414 invalidateState(1 << name); 415 } 416 } 417 418 void AudioMixer::setParameter(int name, int target, int param, void *value) 419 { 420 name -= TRACK0; 421 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 422 track_t& track = mState.tracks[name]; 423 424 int valueInt = (int)value; 425 int32_t *valueBuf = (int32_t *)value; 426 427 switch (target) { 428 429 case TRACK: 430 switch (param) { 431 case CHANNEL_MASK: { 432 audio_channel_mask_t mask = (audio_channel_mask_t) value; 433 if (track.channelMask != mask) { 434 uint32_t channelCount = popcount(mask); 435 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 436 track.channelMask = mask; 437 track.channelCount = channelCount; 438 // the mask has changed, does this track need a downmixer? 439 initTrackDownmix(&mState.tracks[name], name, mask); 440 ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask); 441 invalidateState(1 << name); 442 } 443 } break; 444 case MAIN_BUFFER: 445 if (track.mainBuffer != valueBuf) { 446 track.mainBuffer = valueBuf; 447 ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); 448 invalidateState(1 << name); 449 } 450 break; 451 case AUX_BUFFER: 452 if (track.auxBuffer != valueBuf) { 453 track.auxBuffer = valueBuf; 454 ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); 455 invalidateState(1 << name); 456 } 457 break; 458 case FORMAT: 459 ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT); 460 break; 461 // FIXME do we want to support setting the downmix type from AudioFlinger? 462 // for a specific track? or per mixer? 463 /* case DOWNMIX_TYPE: 464 break */ 465 default: 466 LOG_FATAL("bad param"); 467 } 468 break; 469 470 case RESAMPLE: 471 switch (param) { 472 case SAMPLE_RATE: 473 ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt); 474 if (track.setResampler(uint32_t(valueInt), mSampleRate)) { 475 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)", 476 uint32_t(valueInt)); 477 invalidateState(1 << name); 478 } 479 break; 480 case RESET: 481 track.resetResampler(); 482 invalidateState(1 << name); 483 break; 484 case REMOVE: 485 delete track.resampler; 486 track.resampler = NULL; 487 track.sampleRate = mSampleRate; 488 invalidateState(1 << name); 489 break; 490 default: 491 LOG_FATAL("bad param"); 492 } 493 break; 494 495 case RAMP_VOLUME: 496 case VOLUME: 497 switch (param) { 498 case VOLUME0: 499 case VOLUME1: 500 if (track.volume[param-VOLUME0] != valueInt) { 501 ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt); 502 track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16; 503 track.volume[param-VOLUME0] = valueInt; 504 if (target == VOLUME) { 505 track.prevVolume[param-VOLUME0] = valueInt << 16; 506 track.volumeInc[param-VOLUME0] = 0; 507 } else { 508 int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0]; 509 int32_t volInc = d / int32_t(mState.frameCount); 510 track.volumeInc[param-VOLUME0] = volInc; 511 if (volInc == 0) { 512 track.prevVolume[param-VOLUME0] = valueInt << 16; 513 } 514 } 515 invalidateState(1 << name); 516 } 517 break; 518 case AUXLEVEL: 519 //ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt); 520 if (track.auxLevel != valueInt) { 521 ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt); 522 track.prevAuxLevel = track.auxLevel << 16; 523 track.auxLevel = valueInt; 524 if (target == VOLUME) { 525 track.prevAuxLevel = valueInt << 16; 526 track.auxInc = 0; 527 } else { 528 int32_t d = (valueInt<<16) - track.prevAuxLevel; 529 int32_t volInc = d / int32_t(mState.frameCount); 530 track.auxInc = volInc; 531 if (volInc == 0) { 532 track.prevAuxLevel = valueInt << 16; 533 } 534 } 535 invalidateState(1 << name); 536 } 537 break; 538 default: 539 LOG_FATAL("bad param"); 540 } 541 break; 542 543 default: 544 LOG_FATAL("bad target"); 545 } 546 } 547 548 bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) 549 { 550 if (value != devSampleRate || resampler != NULL) { 551 if (sampleRate != value) { 552 sampleRate = value; 553 if (resampler == NULL) { 554 ALOGV("creating resampler from track %d Hz to device %d Hz", value, devSampleRate); 555 AudioResampler::src_quality quality; 556 // force lowest quality level resampler if use case isn't music or video 557 // FIXME this is flawed for dynamic sample rates, as we choose the resampler 558 // quality level based on the initial ratio, but that could change later. 559 // Should have a way to distinguish tracks with static ratios vs. dynamic ratios. 560 if (!((value == 44100 && devSampleRate == 48000) || 561 (value == 48000 && devSampleRate == 44100))) { 562 quality = AudioResampler::LOW_QUALITY; 563 } else { 564 quality = AudioResampler::DEFAULT_QUALITY; 565 } 566 resampler = AudioResampler::create( 567 format, 568 // the resampler sees the number of channels after the downmixer, if any 569 downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount, 570 devSampleRate, quality); 571 resampler->setLocalTimeFreq(sLocalTimeFreq); 572 } 573 return true; 574 } 575 } 576 return false; 577 } 578 579 inline 580 void AudioMixer::track_t::adjustVolumeRamp(bool aux) 581 { 582 for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) { 583 if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || 584 ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { 585 volumeInc[i] = 0; 586 prevVolume[i] = volume[i]<<16; 587 } 588 } 589 if (aux) { 590 if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) || 591 ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) { 592 auxInc = 0; 593 prevAuxLevel = auxLevel<<16; 594 } 595 } 596 } 597 598 size_t AudioMixer::getUnreleasedFrames(int name) const 599 { 600 name -= TRACK0; 601 if (uint32_t(name) < MAX_NUM_TRACKS) { 602 return mState.tracks[name].getUnreleasedFrames(); 603 } 604 return 0; 605 } 606 607 void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) 608 { 609 name -= TRACK0; 610 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 611 612 if (mState.tracks[name].downmixerBufferProvider != NULL) { 613 // update required? 614 if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) { 615 ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider); 616 // setting the buffer provider for a track that gets downmixed consists in: 617 // 1/ setting the buffer provider to the "downmix / buffer provider" wrapper 618 // so it's the one that gets called when the buffer provider is needed, 619 mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider; 620 // 2/ saving the buffer provider for the track so the wrapper can use it 621 // when it downmixes. 622 mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider; 623 } 624 } else { 625 mState.tracks[name].bufferProvider = bufferProvider; 626 } 627 } 628 629 630 void AudioMixer::process(int64_t pts) 631 { 632 mState.hook(&mState, pts); 633 } 634 635 636 void AudioMixer::process__validate(state_t* state, int64_t pts) 637 { 638 ALOGW_IF(!state->needsChanged, 639 "in process__validate() but nothing's invalid"); 640 641 uint32_t changed = state->needsChanged; 642 state->needsChanged = 0; // clear the validation flag 643 644 // recompute which tracks are enabled / disabled 645 uint32_t enabled = 0; 646 uint32_t disabled = 0; 647 while (changed) { 648 const int i = 31 - __builtin_clz(changed); 649 const uint32_t mask = 1<<i; 650 changed &= ~mask; 651 track_t& t = state->tracks[i]; 652 (t.enabled ? enabled : disabled) |= mask; 653 } 654 state->enabledTracks &= ~disabled; 655 state->enabledTracks |= enabled; 656 657 // compute everything we need... 658 int countActiveTracks = 0; 659 bool all16BitsStereoNoResample = true; 660 bool resampling = false; 661 bool volumeRamp = false; 662 uint32_t en = state->enabledTracks; 663 while (en) { 664 const int i = 31 - __builtin_clz(en); 665 en &= ~(1<<i); 666 667 countActiveTracks++; 668 track_t& t = state->tracks[i]; 669 uint32_t n = 0; 670 n |= NEEDS_CHANNEL_1 + t.channelCount - 1; 671 n |= NEEDS_FORMAT_16; 672 n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED; 673 if (t.auxLevel != 0 && t.auxBuffer != NULL) { 674 n |= NEEDS_AUX_ENABLED; 675 } 676 677 if (t.volumeInc[0]|t.volumeInc[1]) { 678 volumeRamp = true; 679 } else if (!t.doesResample() && t.volumeRL == 0) { 680 n |= NEEDS_MUTE_ENABLED; 681 } 682 t.needs = n; 683 684 if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) { 685 t.hook = track__nop; 686 } else { 687 if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { 688 all16BitsStereoNoResample = false; 689 } 690 if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { 691 all16BitsStereoNoResample = false; 692 resampling = true; 693 t.hook = track__genericResample; 694 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 695 "Track %d needs downmix + resample", i); 696 } else { 697 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ 698 t.hook = track__16BitsMono; 699 all16BitsStereoNoResample = false; 700 } 701 if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){ 702 t.hook = track__16BitsStereo; 703 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 704 "Track %d needs downmix", i); 705 } 706 } 707 } 708 } 709 710 // select the processing hooks 711 state->hook = process__nop; 712 if (countActiveTracks) { 713 if (resampling) { 714 if (!state->outputTemp) { 715 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 716 } 717 if (!state->resampleTemp) { 718 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 719 } 720 state->hook = process__genericResampling; 721 } else { 722 if (state->outputTemp) { 723 delete [] state->outputTemp; 724 state->outputTemp = NULL; 725 } 726 if (state->resampleTemp) { 727 delete [] state->resampleTemp; 728 state->resampleTemp = NULL; 729 } 730 state->hook = process__genericNoResampling; 731 if (all16BitsStereoNoResample && !volumeRamp) { 732 if (countActiveTracks == 1) { 733 state->hook = process__OneTrack16BitsStereoNoResampling; 734 } 735 } 736 } 737 } 738 739 ALOGV("mixer configuration change: %d activeTracks (%08x) " 740 "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d", 741 countActiveTracks, state->enabledTracks, 742 all16BitsStereoNoResample, resampling, volumeRamp); 743 744 state->hook(state, pts); 745 746 // Now that the volume ramp has been done, set optimal state and 747 // track hooks for subsequent mixer process 748 if (countActiveTracks) { 749 bool allMuted = true; 750 uint32_t en = state->enabledTracks; 751 while (en) { 752 const int i = 31 - __builtin_clz(en); 753 en &= ~(1<<i); 754 track_t& t = state->tracks[i]; 755 if (!t.doesResample() && t.volumeRL == 0) 756 { 757 t.needs |= NEEDS_MUTE_ENABLED; 758 t.hook = track__nop; 759 } else { 760 allMuted = false; 761 } 762 } 763 if (allMuted) { 764 state->hook = process__nop; 765 } else if (all16BitsStereoNoResample) { 766 if (countActiveTracks == 1) { 767 state->hook = process__OneTrack16BitsStereoNoResampling; 768 } 769 } 770 } 771 } 772 773 774 void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, 775 int32_t* temp, int32_t* aux) 776 { 777 t->resampler->setSampleRate(t->sampleRate); 778 779 // ramp gain - resample to temp buffer and scale/mix in 2nd step 780 if (aux != NULL) { 781 // always resample with unity gain when sending to auxiliary buffer to be able 782 // to apply send level after resampling 783 // TODO: modify each resampler to support aux channel? 784 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 785 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 786 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 787 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 788 volumeRampStereo(t, out, outFrameCount, temp, aux); 789 } else { 790 volumeStereo(t, out, outFrameCount, temp, aux); 791 } 792 } else { 793 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 794 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 795 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 796 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 797 volumeRampStereo(t, out, outFrameCount, temp, aux); 798 } 799 800 // constant gain 801 else { 802 t->resampler->setVolume(t->volume[0], t->volume[1]); 803 t->resampler->resample(out, outFrameCount, t->bufferProvider); 804 } 805 } 806 } 807 808 void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, 809 int32_t* aux) 810 { 811 } 812 813 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 814 int32_t* aux) 815 { 816 int32_t vl = t->prevVolume[0]; 817 int32_t vr = t->prevVolume[1]; 818 const int32_t vlInc = t->volumeInc[0]; 819 const int32_t vrInc = t->volumeInc[1]; 820 821 //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 822 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 823 // (vl + vlInc*frameCount)/65536.0f, frameCount); 824 825 // ramp volume 826 if (CC_UNLIKELY(aux != NULL)) { 827 int32_t va = t->prevAuxLevel; 828 const int32_t vaInc = t->auxInc; 829 int32_t l; 830 int32_t r; 831 832 do { 833 l = (*temp++ >> 12); 834 r = (*temp++ >> 12); 835 *out++ += (vl >> 16) * l; 836 *out++ += (vr >> 16) * r; 837 *aux++ += (va >> 17) * (l + r); 838 vl += vlInc; 839 vr += vrInc; 840 va += vaInc; 841 } while (--frameCount); 842 t->prevAuxLevel = va; 843 } else { 844 do { 845 *out++ += (vl >> 16) * (*temp++ >> 12); 846 *out++ += (vr >> 16) * (*temp++ >> 12); 847 vl += vlInc; 848 vr += vrInc; 849 } while (--frameCount); 850 } 851 t->prevVolume[0] = vl; 852 t->prevVolume[1] = vr; 853 t->adjustVolumeRamp(aux != NULL); 854 } 855 856 void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 857 int32_t* aux) 858 { 859 const int16_t vl = t->volume[0]; 860 const int16_t vr = t->volume[1]; 861 862 if (CC_UNLIKELY(aux != NULL)) { 863 const int16_t va = t->auxLevel; 864 do { 865 int16_t l = (int16_t)(*temp++ >> 12); 866 int16_t r = (int16_t)(*temp++ >> 12); 867 out[0] = mulAdd(l, vl, out[0]); 868 int16_t a = (int16_t)(((int32_t)l + r) >> 1); 869 out[1] = mulAdd(r, vr, out[1]); 870 out += 2; 871 aux[0] = mulAdd(a, va, aux[0]); 872 aux++; 873 } while (--frameCount); 874 } else { 875 do { 876 int16_t l = (int16_t)(*temp++ >> 12); 877 int16_t r = (int16_t)(*temp++ >> 12); 878 out[0] = mulAdd(l, vl, out[0]); 879 out[1] = mulAdd(r, vr, out[1]); 880 out += 2; 881 } while (--frameCount); 882 } 883 } 884 885 void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 886 int32_t* aux) 887 { 888 const int16_t *in = static_cast<const int16_t *>(t->in); 889 890 if (CC_UNLIKELY(aux != NULL)) { 891 int32_t l; 892 int32_t r; 893 // ramp gain 894 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 895 int32_t vl = t->prevVolume[0]; 896 int32_t vr = t->prevVolume[1]; 897 int32_t va = t->prevAuxLevel; 898 const int32_t vlInc = t->volumeInc[0]; 899 const int32_t vrInc = t->volumeInc[1]; 900 const int32_t vaInc = t->auxInc; 901 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 902 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 903 // (vl + vlInc*frameCount)/65536.0f, frameCount); 904 905 do { 906 l = (int32_t)*in++; 907 r = (int32_t)*in++; 908 *out++ += (vl >> 16) * l; 909 *out++ += (vr >> 16) * r; 910 *aux++ += (va >> 17) * (l + r); 911 vl += vlInc; 912 vr += vrInc; 913 va += vaInc; 914 } while (--frameCount); 915 916 t->prevVolume[0] = vl; 917 t->prevVolume[1] = vr; 918 t->prevAuxLevel = va; 919 t->adjustVolumeRamp(true); 920 } 921 922 // constant gain 923 else { 924 const uint32_t vrl = t->volumeRL; 925 const int16_t va = (int16_t)t->auxLevel; 926 do { 927 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 928 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1); 929 in += 2; 930 out[0] = mulAddRL(1, rl, vrl, out[0]); 931 out[1] = mulAddRL(0, rl, vrl, out[1]); 932 out += 2; 933 aux[0] = mulAdd(a, va, aux[0]); 934 aux++; 935 } while (--frameCount); 936 } 937 } else { 938 // ramp gain 939 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 940 int32_t vl = t->prevVolume[0]; 941 int32_t vr = t->prevVolume[1]; 942 const int32_t vlInc = t->volumeInc[0]; 943 const int32_t vrInc = t->volumeInc[1]; 944 945 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 946 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 947 // (vl + vlInc*frameCount)/65536.0f, frameCount); 948 949 do { 950 *out++ += (vl >> 16) * (int32_t) *in++; 951 *out++ += (vr >> 16) * (int32_t) *in++; 952 vl += vlInc; 953 vr += vrInc; 954 } while (--frameCount); 955 956 t->prevVolume[0] = vl; 957 t->prevVolume[1] = vr; 958 t->adjustVolumeRamp(false); 959 } 960 961 // constant gain 962 else { 963 const uint32_t vrl = t->volumeRL; 964 do { 965 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 966 in += 2; 967 out[0] = mulAddRL(1, rl, vrl, out[0]); 968 out[1] = mulAddRL(0, rl, vrl, out[1]); 969 out += 2; 970 } while (--frameCount); 971 } 972 } 973 t->in = in; 974 } 975 976 void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 977 int32_t* aux) 978 { 979 const int16_t *in = static_cast<int16_t const *>(t->in); 980 981 if (CC_UNLIKELY(aux != NULL)) { 982 // ramp gain 983 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 984 int32_t vl = t->prevVolume[0]; 985 int32_t vr = t->prevVolume[1]; 986 int32_t va = t->prevAuxLevel; 987 const int32_t vlInc = t->volumeInc[0]; 988 const int32_t vrInc = t->volumeInc[1]; 989 const int32_t vaInc = t->auxInc; 990 991 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 992 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 993 // (vl + vlInc*frameCount)/65536.0f, frameCount); 994 995 do { 996 int32_t l = *in++; 997 *out++ += (vl >> 16) * l; 998 *out++ += (vr >> 16) * l; 999 *aux++ += (va >> 16) * l; 1000 vl += vlInc; 1001 vr += vrInc; 1002 va += vaInc; 1003 } while (--frameCount); 1004 1005 t->prevVolume[0] = vl; 1006 t->prevVolume[1] = vr; 1007 t->prevAuxLevel = va; 1008 t->adjustVolumeRamp(true); 1009 } 1010 // constant gain 1011 else { 1012 const int16_t vl = t->volume[0]; 1013 const int16_t vr = t->volume[1]; 1014 const int16_t va = (int16_t)t->auxLevel; 1015 do { 1016 int16_t l = *in++; 1017 out[0] = mulAdd(l, vl, out[0]); 1018 out[1] = mulAdd(l, vr, out[1]); 1019 out += 2; 1020 aux[0] = mulAdd(l, va, aux[0]); 1021 aux++; 1022 } while (--frameCount); 1023 } 1024 } else { 1025 // ramp gain 1026 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 1027 int32_t vl = t->prevVolume[0]; 1028 int32_t vr = t->prevVolume[1]; 1029 const int32_t vlInc = t->volumeInc[0]; 1030 const int32_t vrInc = t->volumeInc[1]; 1031 1032 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1033 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1034 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1035 1036 do { 1037 int32_t l = *in++; 1038 *out++ += (vl >> 16) * l; 1039 *out++ += (vr >> 16) * l; 1040 vl += vlInc; 1041 vr += vrInc; 1042 } while (--frameCount); 1043 1044 t->prevVolume[0] = vl; 1045 t->prevVolume[1] = vr; 1046 t->adjustVolumeRamp(false); 1047 } 1048 // constant gain 1049 else { 1050 const int16_t vl = t->volume[0]; 1051 const int16_t vr = t->volume[1]; 1052 do { 1053 int16_t l = *in++; 1054 out[0] = mulAdd(l, vl, out[0]); 1055 out[1] = mulAdd(l, vr, out[1]); 1056 out += 2; 1057 } while (--frameCount); 1058 } 1059 } 1060 t->in = in; 1061 } 1062 1063 // no-op case 1064 void AudioMixer::process__nop(state_t* state, int64_t pts) 1065 { 1066 uint32_t e0 = state->enabledTracks; 1067 size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS; 1068 while (e0) { 1069 // process by group of tracks with same output buffer to 1070 // avoid multiple memset() on same buffer 1071 uint32_t e1 = e0, e2 = e0; 1072 int i = 31 - __builtin_clz(e1); 1073 { 1074 track_t& t1 = state->tracks[i]; 1075 e2 &= ~(1<<i); 1076 while (e2) { 1077 i = 31 - __builtin_clz(e2); 1078 e2 &= ~(1<<i); 1079 track_t& t2 = state->tracks[i]; 1080 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1081 e1 &= ~(1<<i); 1082 } 1083 } 1084 e0 &= ~(e1); 1085 1086 memset(t1.mainBuffer, 0, bufSize); 1087 } 1088 1089 while (e1) { 1090 i = 31 - __builtin_clz(e1); 1091 e1 &= ~(1<<i); 1092 { 1093 track_t& t3 = state->tracks[i]; 1094 size_t outFrames = state->frameCount; 1095 while (outFrames) { 1096 t3.buffer.frameCount = outFrames; 1097 int64_t outputPTS = calculateOutputPTS( 1098 t3, pts, state->frameCount - outFrames); 1099 t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS); 1100 if (t3.buffer.raw == NULL) break; 1101 outFrames -= t3.buffer.frameCount; 1102 t3.bufferProvider->releaseBuffer(&t3.buffer); 1103 } 1104 } 1105 } 1106 } 1107 } 1108 1109 // generic code without resampling 1110 void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 1111 { 1112 int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32))); 1113 1114 // acquire each track's buffer 1115 uint32_t enabledTracks = state->enabledTracks; 1116 uint32_t e0 = enabledTracks; 1117 while (e0) { 1118 const int i = 31 - __builtin_clz(e0); 1119 e0 &= ~(1<<i); 1120 track_t& t = state->tracks[i]; 1121 t.buffer.frameCount = state->frameCount; 1122 t.bufferProvider->getNextBuffer(&t.buffer, pts); 1123 t.frameCount = t.buffer.frameCount; 1124 t.in = t.buffer.raw; 1125 // t.in == NULL can happen if the track was flushed just after having 1126 // been enabled for mixing. 1127 if (t.in == NULL) 1128 enabledTracks &= ~(1<<i); 1129 } 1130 1131 e0 = enabledTracks; 1132 while (e0) { 1133 // process by group of tracks with same output buffer to 1134 // optimize cache use 1135 uint32_t e1 = e0, e2 = e0; 1136 int j = 31 - __builtin_clz(e1); 1137 track_t& t1 = state->tracks[j]; 1138 e2 &= ~(1<<j); 1139 while (e2) { 1140 j = 31 - __builtin_clz(e2); 1141 e2 &= ~(1<<j); 1142 track_t& t2 = state->tracks[j]; 1143 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1144 e1 &= ~(1<<j); 1145 } 1146 } 1147 e0 &= ~(e1); 1148 // this assumes output 16 bits stereo, no resampling 1149 int32_t *out = t1.mainBuffer; 1150 size_t numFrames = 0; 1151 do { 1152 memset(outTemp, 0, sizeof(outTemp)); 1153 e2 = e1; 1154 while (e2) { 1155 const int i = 31 - __builtin_clz(e2); 1156 e2 &= ~(1<<i); 1157 track_t& t = state->tracks[i]; 1158 size_t outFrames = BLOCKSIZE; 1159 int32_t *aux = NULL; 1160 if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { 1161 aux = t.auxBuffer + numFrames; 1162 } 1163 while (outFrames) { 1164 size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; 1165 if (inFrames) { 1166 t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, 1167 state->resampleTemp, aux); 1168 t.frameCount -= inFrames; 1169 outFrames -= inFrames; 1170 if (CC_UNLIKELY(aux != NULL)) { 1171 aux += inFrames; 1172 } 1173 } 1174 if (t.frameCount == 0 && outFrames) { 1175 t.bufferProvider->releaseBuffer(&t.buffer); 1176 t.buffer.frameCount = (state->frameCount - numFrames) - 1177 (BLOCKSIZE - outFrames); 1178 int64_t outputPTS = calculateOutputPTS( 1179 t, pts, numFrames + (BLOCKSIZE - outFrames)); 1180 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1181 t.in = t.buffer.raw; 1182 if (t.in == NULL) { 1183 enabledTracks &= ~(1<<i); 1184 e1 &= ~(1<<i); 1185 break; 1186 } 1187 t.frameCount = t.buffer.frameCount; 1188 } 1189 } 1190 } 1191 ditherAndClamp(out, outTemp, BLOCKSIZE); 1192 out += BLOCKSIZE; 1193 numFrames += BLOCKSIZE; 1194 } while (numFrames < state->frameCount); 1195 } 1196 1197 // release each track's buffer 1198 e0 = enabledTracks; 1199 while (e0) { 1200 const int i = 31 - __builtin_clz(e0); 1201 e0 &= ~(1<<i); 1202 track_t& t = state->tracks[i]; 1203 t.bufferProvider->releaseBuffer(&t.buffer); 1204 } 1205 } 1206 1207 1208 // generic code with resampling 1209 void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 1210 { 1211 // this const just means that local variable outTemp doesn't change 1212 int32_t* const outTemp = state->outputTemp; 1213 const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount; 1214 1215 size_t numFrames = state->frameCount; 1216 1217 uint32_t e0 = state->enabledTracks; 1218 while (e0) { 1219 // process by group of tracks with same output buffer 1220 // to optimize cache use 1221 uint32_t e1 = e0, e2 = e0; 1222 int j = 31 - __builtin_clz(e1); 1223 track_t& t1 = state->tracks[j]; 1224 e2 &= ~(1<<j); 1225 while (e2) { 1226 j = 31 - __builtin_clz(e2); 1227 e2 &= ~(1<<j); 1228 track_t& t2 = state->tracks[j]; 1229 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1230 e1 &= ~(1<<j); 1231 } 1232 } 1233 e0 &= ~(e1); 1234 int32_t *out = t1.mainBuffer; 1235 memset(outTemp, 0, size); 1236 while (e1) { 1237 const int i = 31 - __builtin_clz(e1); 1238 e1 &= ~(1<<i); 1239 track_t& t = state->tracks[i]; 1240 int32_t *aux = NULL; 1241 if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { 1242 aux = t.auxBuffer; 1243 } 1244 1245 // this is a little goofy, on the resampling case we don't 1246 // acquire/release the buffers because it's done by 1247 // the resampler. 1248 if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { 1249 t.resampler->setPTS(pts); 1250 t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); 1251 } else { 1252 1253 size_t outFrames = 0; 1254 1255 while (outFrames < numFrames) { 1256 t.buffer.frameCount = numFrames - outFrames; 1257 int64_t outputPTS = calculateOutputPTS(t, pts, outFrames); 1258 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1259 t.in = t.buffer.raw; 1260 // t.in == NULL can happen if the track was flushed just after having 1261 // been enabled for mixing. 1262 if (t.in == NULL) break; 1263 1264 if (CC_UNLIKELY(aux != NULL)) { 1265 aux += outFrames; 1266 } 1267 t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, 1268 state->resampleTemp, aux); 1269 outFrames += t.buffer.frameCount; 1270 t.bufferProvider->releaseBuffer(&t.buffer); 1271 } 1272 } 1273 } 1274 ditherAndClamp(out, outTemp, numFrames); 1275 } 1276 } 1277 1278 // one track, 16 bits stereo without resampling is the most common case 1279 void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, 1280 int64_t pts) 1281 { 1282 // This method is only called when state->enabledTracks has exactly 1283 // one bit set. The asserts below would verify this, but are commented out 1284 // since the whole point of this method is to optimize performance. 1285 //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled"); 1286 const int i = 31 - __builtin_clz(state->enabledTracks); 1287 //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled"); 1288 const track_t& t = state->tracks[i]; 1289 1290 AudioBufferProvider::Buffer& b(t.buffer); 1291 1292 int32_t* out = t.mainBuffer; 1293 size_t numFrames = state->frameCount; 1294 1295 const int16_t vl = t.volume[0]; 1296 const int16_t vr = t.volume[1]; 1297 const uint32_t vrl = t.volumeRL; 1298 while (numFrames) { 1299 b.frameCount = numFrames; 1300 int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer); 1301 t.bufferProvider->getNextBuffer(&b, outputPTS); 1302 const int16_t *in = b.i16; 1303 1304 // in == NULL can happen if the track was flushed just after having 1305 // been enabled for mixing. 1306 if (in == NULL || ((unsigned long)in & 3)) { 1307 memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); 1308 ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " 1309 "buffer %p track %d, channels %d, needs %08x", 1310 in, i, t.channelCount, t.needs); 1311 return; 1312 } 1313 size_t outFrames = b.frameCount; 1314 1315 if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) { 1316 // volume is boosted, so we might need to clamp even though 1317 // we process only one track. 1318 do { 1319 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1320 in += 2; 1321 int32_t l = mulRL(1, rl, vrl) >> 12; 1322 int32_t r = mulRL(0, rl, vrl) >> 12; 1323 // clamping... 1324 l = clamp16(l); 1325 r = clamp16(r); 1326 *out++ = (r<<16) | (l & 0xFFFF); 1327 } while (--outFrames); 1328 } else { 1329 do { 1330 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1331 in += 2; 1332 int32_t l = mulRL(1, rl, vrl) >> 12; 1333 int32_t r = mulRL(0, rl, vrl) >> 12; 1334 *out++ = (r<<16) | (l & 0xFFFF); 1335 } while (--outFrames); 1336 } 1337 numFrames -= b.frameCount; 1338 t.bufferProvider->releaseBuffer(&b); 1339 } 1340 } 1341 1342 #if 0 1343 // 2 tracks is also a common case 1344 // NEVER used in current implementation of process__validate() 1345 // only use if the 2 tracks have the same output buffer 1346 void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, 1347 int64_t pts) 1348 { 1349 int i; 1350 uint32_t en = state->enabledTracks; 1351 1352 i = 31 - __builtin_clz(en); 1353 const track_t& t0 = state->tracks[i]; 1354 AudioBufferProvider::Buffer& b0(t0.buffer); 1355 1356 en &= ~(1<<i); 1357 i = 31 - __builtin_clz(en); 1358 const track_t& t1 = state->tracks[i]; 1359 AudioBufferProvider::Buffer& b1(t1.buffer); 1360 1361 const int16_t *in0; 1362 const int16_t vl0 = t0.volume[0]; 1363 const int16_t vr0 = t0.volume[1]; 1364 size_t frameCount0 = 0; 1365 1366 const int16_t *in1; 1367 const int16_t vl1 = t1.volume[0]; 1368 const int16_t vr1 = t1.volume[1]; 1369 size_t frameCount1 = 0; 1370 1371 //FIXME: only works if two tracks use same buffer 1372 int32_t* out = t0.mainBuffer; 1373 size_t numFrames = state->frameCount; 1374 const int16_t *buff = NULL; 1375 1376 1377 while (numFrames) { 1378 1379 if (frameCount0 == 0) { 1380 b0.frameCount = numFrames; 1381 int64_t outputPTS = calculateOutputPTS(t0, pts, 1382 out - t0.mainBuffer); 1383 t0.bufferProvider->getNextBuffer(&b0, outputPTS); 1384 if (b0.i16 == NULL) { 1385 if (buff == NULL) { 1386 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1387 } 1388 in0 = buff; 1389 b0.frameCount = numFrames; 1390 } else { 1391 in0 = b0.i16; 1392 } 1393 frameCount0 = b0.frameCount; 1394 } 1395 if (frameCount1 == 0) { 1396 b1.frameCount = numFrames; 1397 int64_t outputPTS = calculateOutputPTS(t1, pts, 1398 out - t0.mainBuffer); 1399 t1.bufferProvider->getNextBuffer(&b1, outputPTS); 1400 if (b1.i16 == NULL) { 1401 if (buff == NULL) { 1402 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1403 } 1404 in1 = buff; 1405 b1.frameCount = numFrames; 1406 } else { 1407 in1 = b1.i16; 1408 } 1409 frameCount1 = b1.frameCount; 1410 } 1411 1412 size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1; 1413 1414 numFrames -= outFrames; 1415 frameCount0 -= outFrames; 1416 frameCount1 -= outFrames; 1417 1418 do { 1419 int32_t l0 = *in0++; 1420 int32_t r0 = *in0++; 1421 l0 = mul(l0, vl0); 1422 r0 = mul(r0, vr0); 1423 int32_t l = *in1++; 1424 int32_t r = *in1++; 1425 l = mulAdd(l, vl1, l0) >> 12; 1426 r = mulAdd(r, vr1, r0) >> 12; 1427 // clamping... 1428 l = clamp16(l); 1429 r = clamp16(r); 1430 *out++ = (r<<16) | (l & 0xFFFF); 1431 } while (--outFrames); 1432 1433 if (frameCount0 == 0) { 1434 t0.bufferProvider->releaseBuffer(&b0); 1435 } 1436 if (frameCount1 == 0) { 1437 t1.bufferProvider->releaseBuffer(&b1); 1438 } 1439 } 1440 1441 delete [] buff; 1442 } 1443 #endif 1444 1445 int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, 1446 int outputFrameIndex) 1447 { 1448 if (AudioBufferProvider::kInvalidPTS == basePTS) 1449 return AudioBufferProvider::kInvalidPTS; 1450 1451 return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate); 1452 } 1453 1454 /*static*/ uint64_t AudioMixer::sLocalTimeFreq; 1455 /*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT; 1456 1457 /*static*/ void AudioMixer::sInitRoutine() 1458 { 1459 LocalClock lc; 1460 sLocalTimeFreq = lc.getLocalFreq(); 1461 } 1462 1463 // ---------------------------------------------------------------------------- 1464 }; // namespace android 1465
