AudioMixer.cpp revision 9e0308c03d4e76d3146cbb6e30aeb3ac03f05cf5
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 44namespace android { 45 46// ---------------------------------------------------------------------------- 47AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider() : AudioBufferProvider(), 48 mTrackBufferProvider(NULL), mDownmixHandle(NULL) 49{ 50} 51 52AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider() 53{ 54 ALOGV("AudioMixer deleting DownmixerBufferProvider (%p)", this); 55 EffectRelease(mDownmixHandle); 56} 57 58status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, 59 int64_t pts) { 60 //ALOGV("DownmixerBufferProvider::getNextBuffer()"); 61 if (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 82void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) { 83 //ALOGV("DownmixerBufferProvider::releaseBuffer()"); 84 if (mTrackBufferProvider != NULL) { 85 mTrackBufferProvider->releaseBuffer(pBuffer); 86 } else { 87 ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider"); 88 } 89} 90 91 92// ---------------------------------------------------------------------------- 93bool AudioMixer::sIsMultichannelCapable = false; 94 95effect_descriptor_t AudioMixer::sDwnmFxDesc; 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 100AudioMixer::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 pthread_once(&sOnceControl, &sInitRoutine); 117 118 mState.enabledTracks= 0; 119 mState.needsChanged = 0; 120 mState.frameCount = frameCount; 121 mState.hook = process__nop; 122 mState.outputTemp = NULL; 123 mState.resampleTemp = NULL; 124 mState.mLog = &mDummyLog; 125 // mState.reserved 126 127 // FIXME Most of the following initialization is probably redundant since 128 // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0 129 // and mTrackNames is initially 0. However, leave it here until that's verified. 130 track_t* t = mState.tracks; 131 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 132 t->resampler = NULL; 133 t->downmixerBufferProvider = NULL; 134 t++; 135 } 136 137} 138 139AudioMixer::~AudioMixer() 140{ 141 track_t* t = mState.tracks; 142 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 143 delete t->resampler; 144 delete t->downmixerBufferProvider; 145 t++; 146 } 147 delete [] mState.outputTemp; 148 delete [] mState.resampleTemp; 149} 150 151void AudioMixer::setLog(NBLog::Writer *log) 152{ 153 mState.mLog = log; 154} 155 156int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) 157{ 158 uint32_t names = (~mTrackNames) & mConfiguredNames; 159 if (names != 0) { 160 int n = __builtin_ctz(names); 161 ALOGV("add track (%d)", n); 162 mTrackNames |= 1 << n; 163 // assume default parameters for the track, except where noted below 164 track_t* t = &mState.tracks[n]; 165 t->needs = 0; 166 t->volume[0] = UNITY_GAIN; 167 t->volume[1] = UNITY_GAIN; 168 // no initialization needed 169 // t->prevVolume[0] 170 // t->prevVolume[1] 171 t->volumeInc[0] = 0; 172 t->volumeInc[1] = 0; 173 t->auxLevel = 0; 174 t->auxInc = 0; 175 // no initialization needed 176 // t->prevAuxLevel 177 // t->frameCount 178 t->channelCount = 2; 179 t->enabled = false; 180 t->format = 16; 181 t->channelMask = AUDIO_CHANNEL_OUT_STEREO; 182 t->sessionId = sessionId; 183 // setBufferProvider(name, AudioBufferProvider *) is required before enable(name) 184 t->bufferProvider = NULL; 185 t->buffer.raw = NULL; 186 // no initialization needed 187 // t->buffer.frameCount 188 t->hook = NULL; 189 t->in = NULL; 190 t->resampler = NULL; 191 t->sampleRate = mSampleRate; 192 // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name) 193 t->mainBuffer = NULL; 194 t->auxBuffer = NULL; 195 t->downmixerBufferProvider = NULL; 196 197 status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask); 198 if (status == OK) { 199 return TRACK0 + n; 200 } 201 ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix", 202 channelMask); 203 } 204 return -1; 205} 206 207void AudioMixer::invalidateState(uint32_t mask) 208{ 209 if (mask != 0) { 210 mState.needsChanged |= mask; 211 mState.hook = process__validate; 212 } 213 } 214 215status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask) 216{ 217 uint32_t channelCount = popcount(mask); 218 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 219 status_t status = OK; 220 if (channelCount > MAX_NUM_CHANNELS) { 221 pTrack->channelMask = mask; 222 pTrack->channelCount = channelCount; 223 ALOGV("initTrackDownmix(track=%d, mask=0x%x) calls prepareTrackForDownmix()", 224 trackNum, mask); 225 status = prepareTrackForDownmix(pTrack, trackNum); 226 } else { 227 unprepareTrackForDownmix(pTrack, trackNum); 228 } 229 return status; 230} 231 232void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) { 233 ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName); 234 235 if (pTrack->downmixerBufferProvider != NULL) { 236 // this track had previously been configured with a downmixer, delete it 237 ALOGV(" deleting old downmixer"); 238 pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider; 239 delete pTrack->downmixerBufferProvider; 240 pTrack->downmixerBufferProvider = NULL; 241 } else { 242 ALOGV(" nothing to do, no downmixer to delete"); 243 } 244} 245 246status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName) 247{ 248 ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask); 249 250 // discard the previous downmixer if there was one 251 unprepareTrackForDownmix(pTrack, trackName); 252 253 DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(); 254 int32_t status; 255 256 if (!sIsMultichannelCapable) { 257 ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content", 258 trackName); 259 goto noDownmixForActiveTrack; 260 } 261 262 if (EffectCreate(&sDwnmFxDesc.uuid, 263 pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/, 264 &pDbp->mDownmixHandle/*pHandle*/) != 0) { 265 ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName); 266 goto noDownmixForActiveTrack; 267 } 268 269 // channel input configuration will be overridden per-track 270 pDbp->mDownmixConfig.inputCfg.channels = pTrack->channelMask; 271 pDbp->mDownmixConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; 272 pDbp->mDownmixConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 273 pDbp->mDownmixConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 274 pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate; 275 pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate; 276 pDbp->mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 277 pDbp->mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 278 // input and output buffer provider, and frame count will not be used as the downmix effect 279 // process() function is called directly (see DownmixerBufferProvider::getNextBuffer()) 280 pDbp->mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | 281 EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE; 282 pDbp->mDownmixConfig.outputCfg.mask = pDbp->mDownmixConfig.inputCfg.mask; 283 284 {// scope for local variables that are not used in goto label "noDownmixForActiveTrack" 285 int cmdStatus; 286 uint32_t replySize = sizeof(int); 287 288 // Configure and enable downmixer 289 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 290 EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/, 291 &pDbp->mDownmixConfig /*pCmdData*/, 292 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 293 if ((status != 0) || (cmdStatus != 0)) { 294 ALOGE("error %d while configuring downmixer for track %d", status, trackName); 295 goto noDownmixForActiveTrack; 296 } 297 replySize = sizeof(int); 298 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 299 EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/, 300 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 301 if ((status != 0) || (cmdStatus != 0)) { 302 ALOGE("error %d while enabling downmixer for track %d", status, trackName); 303 goto noDownmixForActiveTrack; 304 } 305 306 // Set downmix type 307 // parameter size rounded for padding on 32bit boundary 308 const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int); 309 const int downmixParamSize = 310 sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t); 311 effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize); 312 param->psize = sizeof(downmix_params_t); 313 const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE; 314 memcpy(param->data, &downmixParam, param->psize); 315 const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD; 316 param->vsize = sizeof(downmix_type_t); 317 memcpy(param->data + psizePadded, &downmixType, param->vsize); 318 319 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 320 EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize/* cmdSize */, 321 param /*pCmndData*/, &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 322 323 free(param); 324 325 if ((status != 0) || (cmdStatus != 0)) { 326 ALOGE("error %d while setting downmix type for track %d", status, trackName); 327 goto noDownmixForActiveTrack; 328 } else { 329 ALOGV("downmix type set to %d for track %d", (int) downmixType, trackName); 330 } 331 }// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack" 332 333 // initialization successful: 334 // - keep track of the real buffer provider in case it was set before 335 pDbp->mTrackBufferProvider = pTrack->bufferProvider; 336 // - we'll use the downmix effect integrated inside this 337 // track's buffer provider, and we'll use it as the track's buffer provider 338 pTrack->downmixerBufferProvider = pDbp; 339 pTrack->bufferProvider = pDbp; 340 341 return NO_ERROR; 342 343noDownmixForActiveTrack: 344 delete pDbp; 345 pTrack->downmixerBufferProvider = NULL; 346 return NO_INIT; 347} 348 349void AudioMixer::deleteTrackName(int name) 350{ 351 ALOGV("AudioMixer::deleteTrackName(%d)", name); 352 name -= TRACK0; 353 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 354 ALOGV("deleteTrackName(%d)", name); 355 track_t& track(mState.tracks[ name ]); 356 if (track.enabled) { 357 track.enabled = false; 358 invalidateState(1<<name); 359 } 360 // delete the resampler 361 delete track.resampler; 362 track.resampler = NULL; 363 // delete the downmixer 364 unprepareTrackForDownmix(&mState.tracks[name], name); 365 366 mTrackNames &= ~(1<<name); 367} 368 369void AudioMixer::enable(int name) 370{ 371 name -= TRACK0; 372 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 373 track_t& track = mState.tracks[name]; 374 375 if (!track.enabled) { 376 track.enabled = true; 377 ALOGV("enable(%d)", name); 378 invalidateState(1 << name); 379 } 380} 381 382void AudioMixer::disable(int name) 383{ 384 name -= TRACK0; 385 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 386 track_t& track = mState.tracks[name]; 387 388 if (track.enabled) { 389 track.enabled = false; 390 ALOGV("disable(%d)", name); 391 invalidateState(1 << name); 392 } 393} 394 395void AudioMixer::setParameter(int name, int target, int param, void *value) 396{ 397 name -= TRACK0; 398 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 399 track_t& track = mState.tracks[name]; 400 401 int valueInt = (int)value; 402 int32_t *valueBuf = (int32_t *)value; 403 404 switch (target) { 405 406 case TRACK: 407 switch (param) { 408 case CHANNEL_MASK: { 409 audio_channel_mask_t mask = (audio_channel_mask_t) value; 410 if (track.channelMask != mask) { 411 uint32_t channelCount = popcount(mask); 412 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 413 track.channelMask = mask; 414 track.channelCount = channelCount; 415 // the mask has changed, does this track need a downmixer? 416 initTrackDownmix(&mState.tracks[name], name, mask); 417 ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask); 418 invalidateState(1 << name); 419 } 420 } break; 421 case MAIN_BUFFER: 422 if (track.mainBuffer != valueBuf) { 423 track.mainBuffer = valueBuf; 424 ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); 425 invalidateState(1 << name); 426 } 427 break; 428 case AUX_BUFFER: 429 if (track.auxBuffer != valueBuf) { 430 track.auxBuffer = valueBuf; 431 ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); 432 invalidateState(1 << name); 433 } 434 break; 435 case FORMAT: 436 ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT); 437 break; 438 // FIXME do we want to support setting the downmix type from AudioFlinger? 439 // for a specific track? or per mixer? 440 /* case DOWNMIX_TYPE: 441 break */ 442 default: 443 LOG_FATAL("bad param"); 444 } 445 break; 446 447 case RESAMPLE: 448 switch (param) { 449 case SAMPLE_RATE: 450 ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt); 451 if (track.setResampler(uint32_t(valueInt), mSampleRate)) { 452 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)", 453 uint32_t(valueInt)); 454 invalidateState(1 << name); 455 } 456 break; 457 case RESET: 458 track.resetResampler(); 459 invalidateState(1 << name); 460 break; 461 case REMOVE: 462 delete track.resampler; 463 track.resampler = NULL; 464 track.sampleRate = mSampleRate; 465 invalidateState(1 << name); 466 break; 467 default: 468 LOG_FATAL("bad param"); 469 } 470 break; 471 472 case RAMP_VOLUME: 473 case VOLUME: 474 switch (param) { 475 case VOLUME0: 476 case VOLUME1: 477 if (track.volume[param-VOLUME0] != valueInt) { 478 ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt); 479 track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16; 480 track.volume[param-VOLUME0] = valueInt; 481 if (target == VOLUME) { 482 track.prevVolume[param-VOLUME0] = valueInt << 16; 483 track.volumeInc[param-VOLUME0] = 0; 484 } else { 485 int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0]; 486 int32_t volInc = d / int32_t(mState.frameCount); 487 track.volumeInc[param-VOLUME0] = volInc; 488 if (volInc == 0) { 489 track.prevVolume[param-VOLUME0] = valueInt << 16; 490 } 491 } 492 invalidateState(1 << name); 493 } 494 break; 495 case AUXLEVEL: 496 //ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt); 497 if (track.auxLevel != valueInt) { 498 ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt); 499 track.prevAuxLevel = track.auxLevel << 16; 500 track.auxLevel = valueInt; 501 if (target == VOLUME) { 502 track.prevAuxLevel = valueInt << 16; 503 track.auxInc = 0; 504 } else { 505 int32_t d = (valueInt<<16) - track.prevAuxLevel; 506 int32_t volInc = d / int32_t(mState.frameCount); 507 track.auxInc = volInc; 508 if (volInc == 0) { 509 track.prevAuxLevel = valueInt << 16; 510 } 511 } 512 invalidateState(1 << name); 513 } 514 break; 515 default: 516 LOG_FATAL("bad param"); 517 } 518 break; 519 520 default: 521 LOG_FATAL("bad target"); 522 } 523} 524 525bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) 526{ 527 if (value != devSampleRate || resampler != NULL) { 528 if (sampleRate != value) { 529 sampleRate = value; 530 if (resampler == NULL) { 531 ALOGV("creating resampler from track %d Hz to device %d Hz", value, devSampleRate); 532 AudioResampler::src_quality quality; 533 // force lowest quality level resampler if use case isn't music or video 534 // FIXME this is flawed for dynamic sample rates, as we choose the resampler 535 // quality level based on the initial ratio, but that could change later. 536 // Should have a way to distinguish tracks with static ratios vs. dynamic ratios. 537 if (!((value == 44100 && devSampleRate == 48000) || 538 (value == 48000 && devSampleRate == 44100))) { 539 quality = AudioResampler::DYN_LOW_QUALITY; 540 } else { 541 quality = AudioResampler::DEFAULT_QUALITY; 542 } 543 resampler = AudioResampler::create( 544 format, 545 // the resampler sees the number of channels after the downmixer, if any 546 (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount), 547 devSampleRate, quality); 548 resampler->setLocalTimeFreq(sLocalTimeFreq); 549 } 550 return true; 551 } 552 } 553 return false; 554} 555 556inline 557void AudioMixer::track_t::adjustVolumeRamp(bool aux) 558{ 559 for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) { 560 if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || 561 ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { 562 volumeInc[i] = 0; 563 prevVolume[i] = volume[i]<<16; 564 } 565 } 566 if (aux) { 567 if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) || 568 ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) { 569 auxInc = 0; 570 prevAuxLevel = auxLevel<<16; 571 } 572 } 573} 574 575size_t AudioMixer::getUnreleasedFrames(int name) const 576{ 577 name -= TRACK0; 578 if (uint32_t(name) < MAX_NUM_TRACKS) { 579 return mState.tracks[name].getUnreleasedFrames(); 580 } 581 return 0; 582} 583 584void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) 585{ 586 name -= TRACK0; 587 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 588 589 if (mState.tracks[name].downmixerBufferProvider != NULL) { 590 // update required? 591 if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) { 592 ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider); 593 // setting the buffer provider for a track that gets downmixed consists in: 594 // 1/ setting the buffer provider to the "downmix / buffer provider" wrapper 595 // so it's the one that gets called when the buffer provider is needed, 596 mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider; 597 // 2/ saving the buffer provider for the track so the wrapper can use it 598 // when it downmixes. 599 mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider; 600 } 601 } else { 602 mState.tracks[name].bufferProvider = bufferProvider; 603 } 604} 605 606 607void AudioMixer::process(int64_t pts) 608{ 609 mState.hook(&mState, pts); 610} 611 612 613void AudioMixer::process__validate(state_t* state, int64_t pts) 614{ 615 ALOGW_IF(!state->needsChanged, 616 "in process__validate() but nothing's invalid"); 617 618 uint32_t changed = state->needsChanged; 619 state->needsChanged = 0; // clear the validation flag 620 621 // recompute which tracks are enabled / disabled 622 uint32_t enabled = 0; 623 uint32_t disabled = 0; 624 while (changed) { 625 const int i = 31 - __builtin_clz(changed); 626 const uint32_t mask = 1<<i; 627 changed &= ~mask; 628 track_t& t = state->tracks[i]; 629 (t.enabled ? enabled : disabled) |= mask; 630 } 631 state->enabledTracks &= ~disabled; 632 state->enabledTracks |= enabled; 633 634 // compute everything we need... 635 int countActiveTracks = 0; 636 bool all16BitsStereoNoResample = true; 637 bool resampling = false; 638 bool volumeRamp = false; 639 uint32_t en = state->enabledTracks; 640 while (en) { 641 const int i = 31 - __builtin_clz(en); 642 en &= ~(1<<i); 643 644 countActiveTracks++; 645 track_t& t = state->tracks[i]; 646 uint32_t n = 0; 647 // FIXME can overflow (mask is only 3 bits) 648 n |= NEEDS_CHANNEL_1 + t.channelCount - 1; 649 if (t.doesResample()) { 650 n |= NEEDS_RESAMPLE; 651 } 652 if (t.auxLevel != 0 && t.auxBuffer != NULL) { 653 n |= NEEDS_AUX; 654 } 655 656 if (t.volumeInc[0]|t.volumeInc[1]) { 657 volumeRamp = true; 658 } else if (!t.doesResample() && t.volumeRL == 0) { 659 n |= NEEDS_MUTE; 660 } 661 t.needs = n; 662 663 if (n & NEEDS_MUTE) { 664 t.hook = track__nop; 665 } else { 666 if (n & NEEDS_AUX) { 667 all16BitsStereoNoResample = false; 668 } 669 if (n & NEEDS_RESAMPLE) { 670 all16BitsStereoNoResample = false; 671 resampling = true; 672 t.hook = track__genericResample; 673 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 674 "Track %d needs downmix + resample", i); 675 } else { 676 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ 677 t.hook = track__16BitsMono; 678 all16BitsStereoNoResample = false; 679 } 680 if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){ 681 t.hook = track__16BitsStereo; 682 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 683 "Track %d needs downmix", i); 684 } 685 } 686 } 687 } 688 689 // select the processing hooks 690 state->hook = process__nop; 691 if (countActiveTracks > 0) { 692 if (resampling) { 693 if (!state->outputTemp) { 694 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 695 } 696 if (!state->resampleTemp) { 697 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 698 } 699 state->hook = process__genericResampling; 700 } else { 701 if (state->outputTemp) { 702 delete [] state->outputTemp; 703 state->outputTemp = NULL; 704 } 705 if (state->resampleTemp) { 706 delete [] state->resampleTemp; 707 state->resampleTemp = NULL; 708 } 709 state->hook = process__genericNoResampling; 710 if (all16BitsStereoNoResample && !volumeRamp) { 711 if (countActiveTracks == 1) { 712 state->hook = process__OneTrack16BitsStereoNoResampling; 713 } 714 } 715 } 716 } 717 718 ALOGV("mixer configuration change: %d activeTracks (%08x) " 719 "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d", 720 countActiveTracks, state->enabledTracks, 721 all16BitsStereoNoResample, resampling, volumeRamp); 722 723 state->hook(state, pts); 724 725 // Now that the volume ramp has been done, set optimal state and 726 // track hooks for subsequent mixer process 727 if (countActiveTracks > 0) { 728 bool allMuted = true; 729 uint32_t en = state->enabledTracks; 730 while (en) { 731 const int i = 31 - __builtin_clz(en); 732 en &= ~(1<<i); 733 track_t& t = state->tracks[i]; 734 if (!t.doesResample() && t.volumeRL == 0) { 735 t.needs |= NEEDS_MUTE; 736 t.hook = track__nop; 737 } else { 738 allMuted = false; 739 } 740 } 741 if (allMuted) { 742 state->hook = process__nop; 743 } else if (all16BitsStereoNoResample) { 744 if (countActiveTracks == 1) { 745 state->hook = process__OneTrack16BitsStereoNoResampling; 746 } 747 } 748 } 749} 750 751 752void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, 753 int32_t* temp, int32_t* aux) 754{ 755 t->resampler->setSampleRate(t->sampleRate); 756 757 // ramp gain - resample to temp buffer and scale/mix in 2nd step 758 if (aux != NULL) { 759 // always resample with unity gain when sending to auxiliary buffer to be able 760 // to apply send level after resampling 761 // TODO: modify each resampler to support aux channel? 762 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 763 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 764 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 765 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 766 volumeRampStereo(t, out, outFrameCount, temp, aux); 767 } else { 768 volumeStereo(t, out, outFrameCount, temp, aux); 769 } 770 } else { 771 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 772 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 773 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 774 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 775 volumeRampStereo(t, out, outFrameCount, temp, aux); 776 } 777 778 // constant gain 779 else { 780 t->resampler->setVolume(t->volume[0], t->volume[1]); 781 t->resampler->resample(out, outFrameCount, t->bufferProvider); 782 } 783 } 784} 785 786void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused, 787 size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused) 788{ 789} 790 791void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 792 int32_t* aux) 793{ 794 int32_t vl = t->prevVolume[0]; 795 int32_t vr = t->prevVolume[1]; 796 const int32_t vlInc = t->volumeInc[0]; 797 const int32_t vrInc = t->volumeInc[1]; 798 799 //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 800 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 801 // (vl + vlInc*frameCount)/65536.0f, frameCount); 802 803 // ramp volume 804 if (CC_UNLIKELY(aux != NULL)) { 805 int32_t va = t->prevAuxLevel; 806 const int32_t vaInc = t->auxInc; 807 int32_t l; 808 int32_t r; 809 810 do { 811 l = (*temp++ >> 12); 812 r = (*temp++ >> 12); 813 *out++ += (vl >> 16) * l; 814 *out++ += (vr >> 16) * r; 815 *aux++ += (va >> 17) * (l + r); 816 vl += vlInc; 817 vr += vrInc; 818 va += vaInc; 819 } while (--frameCount); 820 t->prevAuxLevel = va; 821 } else { 822 do { 823 *out++ += (vl >> 16) * (*temp++ >> 12); 824 *out++ += (vr >> 16) * (*temp++ >> 12); 825 vl += vlInc; 826 vr += vrInc; 827 } while (--frameCount); 828 } 829 t->prevVolume[0] = vl; 830 t->prevVolume[1] = vr; 831 t->adjustVolumeRamp(aux != NULL); 832} 833 834void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 835 int32_t* aux) 836{ 837 const int16_t vl = t->volume[0]; 838 const int16_t vr = t->volume[1]; 839 840 if (CC_UNLIKELY(aux != NULL)) { 841 const int16_t va = t->auxLevel; 842 do { 843 int16_t l = (int16_t)(*temp++ >> 12); 844 int16_t r = (int16_t)(*temp++ >> 12); 845 out[0] = mulAdd(l, vl, out[0]); 846 int16_t a = (int16_t)(((int32_t)l + r) >> 1); 847 out[1] = mulAdd(r, vr, out[1]); 848 out += 2; 849 aux[0] = mulAdd(a, va, aux[0]); 850 aux++; 851 } while (--frameCount); 852 } else { 853 do { 854 int16_t l = (int16_t)(*temp++ >> 12); 855 int16_t r = (int16_t)(*temp++ >> 12); 856 out[0] = mulAdd(l, vl, out[0]); 857 out[1] = mulAdd(r, vr, out[1]); 858 out += 2; 859 } while (--frameCount); 860 } 861} 862 863void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, 864 int32_t* temp __unused, int32_t* aux) 865{ 866 const int16_t *in = static_cast<const int16_t *>(t->in); 867 868 if (CC_UNLIKELY(aux != NULL)) { 869 int32_t l; 870 int32_t r; 871 // ramp gain 872 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 873 int32_t vl = t->prevVolume[0]; 874 int32_t vr = t->prevVolume[1]; 875 int32_t va = t->prevAuxLevel; 876 const int32_t vlInc = t->volumeInc[0]; 877 const int32_t vrInc = t->volumeInc[1]; 878 const int32_t vaInc = t->auxInc; 879 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 880 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 881 // (vl + vlInc*frameCount)/65536.0f, frameCount); 882 883 do { 884 l = (int32_t)*in++; 885 r = (int32_t)*in++; 886 *out++ += (vl >> 16) * l; 887 *out++ += (vr >> 16) * r; 888 *aux++ += (va >> 17) * (l + r); 889 vl += vlInc; 890 vr += vrInc; 891 va += vaInc; 892 } while (--frameCount); 893 894 t->prevVolume[0] = vl; 895 t->prevVolume[1] = vr; 896 t->prevAuxLevel = va; 897 t->adjustVolumeRamp(true); 898 } 899 900 // constant gain 901 else { 902 const uint32_t vrl = t->volumeRL; 903 const int16_t va = (int16_t)t->auxLevel; 904 do { 905 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 906 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1); 907 in += 2; 908 out[0] = mulAddRL(1, rl, vrl, out[0]); 909 out[1] = mulAddRL(0, rl, vrl, out[1]); 910 out += 2; 911 aux[0] = mulAdd(a, va, aux[0]); 912 aux++; 913 } while (--frameCount); 914 } 915 } else { 916 // ramp gain 917 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 918 int32_t vl = t->prevVolume[0]; 919 int32_t vr = t->prevVolume[1]; 920 const int32_t vlInc = t->volumeInc[0]; 921 const int32_t vrInc = t->volumeInc[1]; 922 923 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 924 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 925 // (vl + vlInc*frameCount)/65536.0f, frameCount); 926 927 do { 928 *out++ += (vl >> 16) * (int32_t) *in++; 929 *out++ += (vr >> 16) * (int32_t) *in++; 930 vl += vlInc; 931 vr += vrInc; 932 } while (--frameCount); 933 934 t->prevVolume[0] = vl; 935 t->prevVolume[1] = vr; 936 t->adjustVolumeRamp(false); 937 } 938 939 // constant gain 940 else { 941 const uint32_t vrl = t->volumeRL; 942 do { 943 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 944 in += 2; 945 out[0] = mulAddRL(1, rl, vrl, out[0]); 946 out[1] = mulAddRL(0, rl, vrl, out[1]); 947 out += 2; 948 } while (--frameCount); 949 } 950 } 951 t->in = in; 952} 953 954void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, 955 int32_t* temp __unused, int32_t* aux) 956{ 957 const int16_t *in = static_cast<int16_t const *>(t->in); 958 959 if (CC_UNLIKELY(aux != NULL)) { 960 // ramp gain 961 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 962 int32_t vl = t->prevVolume[0]; 963 int32_t vr = t->prevVolume[1]; 964 int32_t va = t->prevAuxLevel; 965 const int32_t vlInc = t->volumeInc[0]; 966 const int32_t vrInc = t->volumeInc[1]; 967 const int32_t vaInc = t->auxInc; 968 969 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 970 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 971 // (vl + vlInc*frameCount)/65536.0f, frameCount); 972 973 do { 974 int32_t l = *in++; 975 *out++ += (vl >> 16) * l; 976 *out++ += (vr >> 16) * l; 977 *aux++ += (va >> 16) * l; 978 vl += vlInc; 979 vr += vrInc; 980 va += vaInc; 981 } while (--frameCount); 982 983 t->prevVolume[0] = vl; 984 t->prevVolume[1] = vr; 985 t->prevAuxLevel = va; 986 t->adjustVolumeRamp(true); 987 } 988 // constant gain 989 else { 990 const int16_t vl = t->volume[0]; 991 const int16_t vr = t->volume[1]; 992 const int16_t va = (int16_t)t->auxLevel; 993 do { 994 int16_t l = *in++; 995 out[0] = mulAdd(l, vl, out[0]); 996 out[1] = mulAdd(l, vr, out[1]); 997 out += 2; 998 aux[0] = mulAdd(l, va, aux[0]); 999 aux++; 1000 } while (--frameCount); 1001 } 1002 } else { 1003 // ramp gain 1004 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 1005 int32_t vl = t->prevVolume[0]; 1006 int32_t vr = t->prevVolume[1]; 1007 const int32_t vlInc = t->volumeInc[0]; 1008 const int32_t vrInc = t->volumeInc[1]; 1009 1010 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1011 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1012 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1013 1014 do { 1015 int32_t l = *in++; 1016 *out++ += (vl >> 16) * l; 1017 *out++ += (vr >> 16) * l; 1018 vl += vlInc; 1019 vr += vrInc; 1020 } while (--frameCount); 1021 1022 t->prevVolume[0] = vl; 1023 t->prevVolume[1] = vr; 1024 t->adjustVolumeRamp(false); 1025 } 1026 // constant gain 1027 else { 1028 const int16_t vl = t->volume[0]; 1029 const int16_t vr = t->volume[1]; 1030 do { 1031 int16_t l = *in++; 1032 out[0] = mulAdd(l, vl, out[0]); 1033 out[1] = mulAdd(l, vr, out[1]); 1034 out += 2; 1035 } while (--frameCount); 1036 } 1037 } 1038 t->in = in; 1039} 1040 1041// no-op case 1042void AudioMixer::process__nop(state_t* state, int64_t pts) 1043{ 1044 uint32_t e0 = state->enabledTracks; 1045 size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS; 1046 while (e0) { 1047 // process by group of tracks with same output buffer to 1048 // avoid multiple memset() on same buffer 1049 uint32_t e1 = e0, e2 = e0; 1050 int i = 31 - __builtin_clz(e1); 1051 { 1052 track_t& t1 = state->tracks[i]; 1053 e2 &= ~(1<<i); 1054 while (e2) { 1055 i = 31 - __builtin_clz(e2); 1056 e2 &= ~(1<<i); 1057 track_t& t2 = state->tracks[i]; 1058 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1059 e1 &= ~(1<<i); 1060 } 1061 } 1062 e0 &= ~(e1); 1063 1064 memset(t1.mainBuffer, 0, bufSize); 1065 } 1066 1067 while (e1) { 1068 i = 31 - __builtin_clz(e1); 1069 e1 &= ~(1<<i); 1070 { 1071 track_t& t3 = state->tracks[i]; 1072 size_t outFrames = state->frameCount; 1073 while (outFrames) { 1074 t3.buffer.frameCount = outFrames; 1075 int64_t outputPTS = calculateOutputPTS( 1076 t3, pts, state->frameCount - outFrames); 1077 t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS); 1078 if (t3.buffer.raw == NULL) break; 1079 outFrames -= t3.buffer.frameCount; 1080 t3.bufferProvider->releaseBuffer(&t3.buffer); 1081 } 1082 } 1083 } 1084 } 1085} 1086 1087// generic code without resampling 1088void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 1089{ 1090 int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32))); 1091 1092 // acquire each track's buffer 1093 uint32_t enabledTracks = state->enabledTracks; 1094 uint32_t e0 = enabledTracks; 1095 while (e0) { 1096 const int i = 31 - __builtin_clz(e0); 1097 e0 &= ~(1<<i); 1098 track_t& t = state->tracks[i]; 1099 t.buffer.frameCount = state->frameCount; 1100 t.bufferProvider->getNextBuffer(&t.buffer, pts); 1101 t.frameCount = t.buffer.frameCount; 1102 t.in = t.buffer.raw; 1103 } 1104 1105 e0 = enabledTracks; 1106 while (e0) { 1107 // process by group of tracks with same output buffer to 1108 // optimize cache use 1109 uint32_t e1 = e0, e2 = e0; 1110 int j = 31 - __builtin_clz(e1); 1111 track_t& t1 = state->tracks[j]; 1112 e2 &= ~(1<<j); 1113 while (e2) { 1114 j = 31 - __builtin_clz(e2); 1115 e2 &= ~(1<<j); 1116 track_t& t2 = state->tracks[j]; 1117 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1118 e1 &= ~(1<<j); 1119 } 1120 } 1121 e0 &= ~(e1); 1122 // this assumes output 16 bits stereo, no resampling 1123 int32_t *out = t1.mainBuffer; 1124 size_t numFrames = 0; 1125 do { 1126 memset(outTemp, 0, sizeof(outTemp)); 1127 e2 = e1; 1128 while (e2) { 1129 const int i = 31 - __builtin_clz(e2); 1130 e2 &= ~(1<<i); 1131 track_t& t = state->tracks[i]; 1132 size_t outFrames = BLOCKSIZE; 1133 int32_t *aux = NULL; 1134 if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { 1135 aux = t.auxBuffer + numFrames; 1136 } 1137 while (outFrames) { 1138 // t.in == NULL can happen if the track was flushed just after having 1139 // been enabled for mixing. 1140 if (t.in == NULL) { 1141 enabledTracks &= ~(1<<i); 1142 e1 &= ~(1<<i); 1143 break; 1144 } 1145 size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; 1146 if (inFrames > 0) { 1147 t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, 1148 state->resampleTemp, aux); 1149 t.frameCount -= inFrames; 1150 outFrames -= inFrames; 1151 if (CC_UNLIKELY(aux != NULL)) { 1152 aux += inFrames; 1153 } 1154 } 1155 if (t.frameCount == 0 && outFrames) { 1156 t.bufferProvider->releaseBuffer(&t.buffer); 1157 t.buffer.frameCount = (state->frameCount - numFrames) - 1158 (BLOCKSIZE - outFrames); 1159 int64_t outputPTS = calculateOutputPTS( 1160 t, pts, numFrames + (BLOCKSIZE - outFrames)); 1161 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1162 t.in = t.buffer.raw; 1163 if (t.in == NULL) { 1164 enabledTracks &= ~(1<<i); 1165 e1 &= ~(1<<i); 1166 break; 1167 } 1168 t.frameCount = t.buffer.frameCount; 1169 } 1170 } 1171 } 1172 ditherAndClamp(out, outTemp, BLOCKSIZE); 1173 out += BLOCKSIZE; 1174 numFrames += BLOCKSIZE; 1175 } while (numFrames < state->frameCount); 1176 } 1177 1178 // release each track's buffer 1179 e0 = enabledTracks; 1180 while (e0) { 1181 const int i = 31 - __builtin_clz(e0); 1182 e0 &= ~(1<<i); 1183 track_t& t = state->tracks[i]; 1184 t.bufferProvider->releaseBuffer(&t.buffer); 1185 } 1186} 1187 1188 1189// generic code with resampling 1190void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 1191{ 1192 // this const just means that local variable outTemp doesn't change 1193 int32_t* const outTemp = state->outputTemp; 1194 const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount; 1195 1196 size_t numFrames = state->frameCount; 1197 1198 uint32_t e0 = state->enabledTracks; 1199 while (e0) { 1200 // process by group of tracks with same output buffer 1201 // to optimize cache use 1202 uint32_t e1 = e0, e2 = e0; 1203 int j = 31 - __builtin_clz(e1); 1204 track_t& t1 = state->tracks[j]; 1205 e2 &= ~(1<<j); 1206 while (e2) { 1207 j = 31 - __builtin_clz(e2); 1208 e2 &= ~(1<<j); 1209 track_t& t2 = state->tracks[j]; 1210 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1211 e1 &= ~(1<<j); 1212 } 1213 } 1214 e0 &= ~(e1); 1215 int32_t *out = t1.mainBuffer; 1216 memset(outTemp, 0, size); 1217 while (e1) { 1218 const int i = 31 - __builtin_clz(e1); 1219 e1 &= ~(1<<i); 1220 track_t& t = state->tracks[i]; 1221 int32_t *aux = NULL; 1222 if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { 1223 aux = t.auxBuffer; 1224 } 1225 1226 // this is a little goofy, on the resampling case we don't 1227 // acquire/release the buffers because it's done by 1228 // the resampler. 1229 if (t.needs & NEEDS_RESAMPLE) { 1230 t.resampler->setPTS(pts); 1231 t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); 1232 } else { 1233 1234 size_t outFrames = 0; 1235 1236 while (outFrames < numFrames) { 1237 t.buffer.frameCount = numFrames - outFrames; 1238 int64_t outputPTS = calculateOutputPTS(t, pts, outFrames); 1239 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1240 t.in = t.buffer.raw; 1241 // t.in == NULL can happen if the track was flushed just after having 1242 // been enabled for mixing. 1243 if (t.in == NULL) break; 1244 1245 if (CC_UNLIKELY(aux != NULL)) { 1246 aux += outFrames; 1247 } 1248 t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, 1249 state->resampleTemp, aux); 1250 outFrames += t.buffer.frameCount; 1251 t.bufferProvider->releaseBuffer(&t.buffer); 1252 } 1253 } 1254 } 1255 ditherAndClamp(out, outTemp, numFrames); 1256 } 1257} 1258 1259// one track, 16 bits stereo without resampling is the most common case 1260void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, 1261 int64_t pts) 1262{ 1263 // This method is only called when state->enabledTracks has exactly 1264 // one bit set. The asserts below would verify this, but are commented out 1265 // since the whole point of this method is to optimize performance. 1266 //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled"); 1267 const int i = 31 - __builtin_clz(state->enabledTracks); 1268 //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled"); 1269 const track_t& t = state->tracks[i]; 1270 1271 AudioBufferProvider::Buffer& b(t.buffer); 1272 1273 int32_t* out = t.mainBuffer; 1274 size_t numFrames = state->frameCount; 1275 1276 const int16_t vl = t.volume[0]; 1277 const int16_t vr = t.volume[1]; 1278 const uint32_t vrl = t.volumeRL; 1279 while (numFrames) { 1280 b.frameCount = numFrames; 1281 int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer); 1282 t.bufferProvider->getNextBuffer(&b, outputPTS); 1283 const int16_t *in = b.i16; 1284 1285 // in == NULL can happen if the track was flushed just after having 1286 // been enabled for mixing. 1287 if (in == NULL || ((unsigned long)in & 3)) { 1288 memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); 1289 ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " 1290 "buffer %p track %d, channels %d, needs %08x", 1291 in, i, t.channelCount, t.needs); 1292 return; 1293 } 1294 size_t outFrames = b.frameCount; 1295 1296 if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) { 1297 // volume is boosted, so we might need to clamp even though 1298 // we process only one track. 1299 do { 1300 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1301 in += 2; 1302 int32_t l = mulRL(1, rl, vrl) >> 12; 1303 int32_t r = mulRL(0, rl, vrl) >> 12; 1304 // clamping... 1305 l = clamp16(l); 1306 r = clamp16(r); 1307 *out++ = (r<<16) | (l & 0xFFFF); 1308 } while (--outFrames); 1309 } else { 1310 do { 1311 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1312 in += 2; 1313 int32_t l = mulRL(1, rl, vrl) >> 12; 1314 int32_t r = mulRL(0, rl, vrl) >> 12; 1315 *out++ = (r<<16) | (l & 0xFFFF); 1316 } while (--outFrames); 1317 } 1318 numFrames -= b.frameCount; 1319 t.bufferProvider->releaseBuffer(&b); 1320 } 1321} 1322 1323#if 0 1324// 2 tracks is also a common case 1325// NEVER used in current implementation of process__validate() 1326// only use if the 2 tracks have the same output buffer 1327void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, 1328 int64_t pts) 1329{ 1330 int i; 1331 uint32_t en = state->enabledTracks; 1332 1333 i = 31 - __builtin_clz(en); 1334 const track_t& t0 = state->tracks[i]; 1335 AudioBufferProvider::Buffer& b0(t0.buffer); 1336 1337 en &= ~(1<<i); 1338 i = 31 - __builtin_clz(en); 1339 const track_t& t1 = state->tracks[i]; 1340 AudioBufferProvider::Buffer& b1(t1.buffer); 1341 1342 const int16_t *in0; 1343 const int16_t vl0 = t0.volume[0]; 1344 const int16_t vr0 = t0.volume[1]; 1345 size_t frameCount0 = 0; 1346 1347 const int16_t *in1; 1348 const int16_t vl1 = t1.volume[0]; 1349 const int16_t vr1 = t1.volume[1]; 1350 size_t frameCount1 = 0; 1351 1352 //FIXME: only works if two tracks use same buffer 1353 int32_t* out = t0.mainBuffer; 1354 size_t numFrames = state->frameCount; 1355 const int16_t *buff = NULL; 1356 1357 1358 while (numFrames) { 1359 1360 if (frameCount0 == 0) { 1361 b0.frameCount = numFrames; 1362 int64_t outputPTS = calculateOutputPTS(t0, pts, 1363 out - t0.mainBuffer); 1364 t0.bufferProvider->getNextBuffer(&b0, outputPTS); 1365 if (b0.i16 == NULL) { 1366 if (buff == NULL) { 1367 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1368 } 1369 in0 = buff; 1370 b0.frameCount = numFrames; 1371 } else { 1372 in0 = b0.i16; 1373 } 1374 frameCount0 = b0.frameCount; 1375 } 1376 if (frameCount1 == 0) { 1377 b1.frameCount = numFrames; 1378 int64_t outputPTS = calculateOutputPTS(t1, pts, 1379 out - t0.mainBuffer); 1380 t1.bufferProvider->getNextBuffer(&b1, outputPTS); 1381 if (b1.i16 == NULL) { 1382 if (buff == NULL) { 1383 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1384 } 1385 in1 = buff; 1386 b1.frameCount = numFrames; 1387 } else { 1388 in1 = b1.i16; 1389 } 1390 frameCount1 = b1.frameCount; 1391 } 1392 1393 size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1; 1394 1395 numFrames -= outFrames; 1396 frameCount0 -= outFrames; 1397 frameCount1 -= outFrames; 1398 1399 do { 1400 int32_t l0 = *in0++; 1401 int32_t r0 = *in0++; 1402 l0 = mul(l0, vl0); 1403 r0 = mul(r0, vr0); 1404 int32_t l = *in1++; 1405 int32_t r = *in1++; 1406 l = mulAdd(l, vl1, l0) >> 12; 1407 r = mulAdd(r, vr1, r0) >> 12; 1408 // clamping... 1409 l = clamp16(l); 1410 r = clamp16(r); 1411 *out++ = (r<<16) | (l & 0xFFFF); 1412 } while (--outFrames); 1413 1414 if (frameCount0 == 0) { 1415 t0.bufferProvider->releaseBuffer(&b0); 1416 } 1417 if (frameCount1 == 0) { 1418 t1.bufferProvider->releaseBuffer(&b1); 1419 } 1420 } 1421 1422 delete [] buff; 1423} 1424#endif 1425 1426int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, 1427 int outputFrameIndex) 1428{ 1429 if (AudioBufferProvider::kInvalidPTS == basePTS) { 1430 return AudioBufferProvider::kInvalidPTS; 1431 } 1432 1433 return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate); 1434} 1435 1436/*static*/ uint64_t AudioMixer::sLocalTimeFreq; 1437/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT; 1438 1439/*static*/ void AudioMixer::sInitRoutine() 1440{ 1441 LocalClock lc; 1442 sLocalTimeFreq = lc.getLocalFreq(); 1443 1444 // find multichannel downmix effect if we have to play multichannel content 1445 uint32_t numEffects = 0; 1446 int ret = EffectQueryNumberEffects(&numEffects); 1447 if (ret != 0) { 1448 ALOGE("AudioMixer() error %d querying number of effects", ret); 1449 return; 1450 } 1451 ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); 1452 1453 for (uint32_t i = 0 ; i < numEffects ; i++) { 1454 if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) { 1455 ALOGV("effect %d is called %s", i, sDwnmFxDesc.name); 1456 if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { 1457 ALOGI("found effect \"%s\" from %s", 1458 sDwnmFxDesc.name, sDwnmFxDesc.implementor); 1459 sIsMultichannelCapable = true; 1460 break; 1461 } 1462 } 1463 } 1464 ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect"); 1465} 1466 1467// ---------------------------------------------------------------------------- 1468}; // namespace android 1469