AudioMixer.cpp revision 3c0a0e8541846427db0587c2fffb90f60ee680b0
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 <stdint.h> 22#include <string.h> 23#include <stdlib.h> 24#include <sys/types.h> 25 26#include <utils/Errors.h> 27#include <utils/Log.h> 28 29#include <cutils/bitops.h> 30#include <cutils/compiler.h> 31#include <utils/Debug.h> 32 33#include <system/audio.h> 34 35#include <audio_utils/primitives.h> 36#include <common_time/local_clock.h> 37#include <common_time/cc_helper.h> 38 39#include <media/EffectsFactoryApi.h> 40 41#include "AudioMixer.h" 42 43namespace android { 44 45// ---------------------------------------------------------------------------- 46AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider() : AudioBufferProvider(), 47 mTrackBufferProvider(NULL), mDownmixHandle(NULL) 48{ 49} 50 51AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider() 52{ 53 ALOGV("AudioMixer deleting DownmixerBufferProvider (%p)", this); 54 EffectRelease(mDownmixHandle); 55} 56 57status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, 58 int64_t pts) { 59 //ALOGV("DownmixerBufferProvider::getNextBuffer()"); 60 if (this->mTrackBufferProvider != NULL) { 61 status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts); 62 if (res == OK) { 63 mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount; 64 mDownmixConfig.inputCfg.buffer.raw = pBuffer->raw; 65 mDownmixConfig.outputCfg.buffer.frameCount = pBuffer->frameCount; 66 mDownmixConfig.outputCfg.buffer.raw = mDownmixConfig.inputCfg.buffer.raw; 67 // in-place so overwrite the buffer contents, has been set in prepareTrackForDownmix() 68 //mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 69 70 res = (*mDownmixHandle)->process(mDownmixHandle, 71 &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer); 72 //ALOGV("getNextBuffer is downmixing"); 73 } 74 return res; 75 } else { 76 ALOGE("DownmixerBufferProvider::getNextBuffer() error: NULL track buffer provider"); 77 return NO_INIT; 78 } 79} 80 81void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) { 82 //ALOGV("DownmixerBufferProvider::releaseBuffer()"); 83 if (this->mTrackBufferProvider != NULL) { 84 mTrackBufferProvider->releaseBuffer(pBuffer); 85 } else { 86 ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider"); 87 } 88} 89 90 91// ---------------------------------------------------------------------------- 92bool AudioMixer::isMultichannelCapable = false; 93 94effect_descriptor_t AudioMixer::dwnmFxDesc; 95 96// Ensure mConfiguredNames bitmask is initialized properly on all architectures. 97// The value of 1 << x is undefined in C when x >= 32. 98 99AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) 100 : mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1), 101 mSampleRate(sampleRate) 102{ 103 // AudioMixer is not yet capable of multi-channel beyond stereo 104 COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS); 105 106 ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u", 107 maxNumTracks, MAX_NUM_TRACKS); 108 109 LocalClock lc; 110 111 mState.enabledTracks= 0; 112 mState.needsChanged = 0; 113 mState.frameCount = frameCount; 114 mState.hook = process__nop; 115 mState.outputTemp = NULL; 116 mState.resampleTemp = NULL; 117 // mState.reserved 118 119 // FIXME Most of the following initialization is probably redundant since 120 // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0 121 // and mTrackNames is initially 0. However, leave it here until that's verified. 122 track_t* t = mState.tracks; 123 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 124 // FIXME redundant per track 125 t->localTimeFreq = lc.getLocalFreq(); 126 t->resampler = NULL; 127 t->downmixerBufferProvider = NULL; 128 t++; 129 } 130 131 // find multichannel downmix effect if we have to play multichannel content 132 uint32_t numEffects = 0; 133 int ret = EffectQueryNumberEffects(&numEffects); 134 if (ret != 0) { 135 ALOGE("AudioMixer() error %d querying number of effects", ret); 136 return; 137 } 138 ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); 139 140 for (uint32_t i = 0 ; i < numEffects ; i++) { 141 if (EffectQueryEffect(i, &dwnmFxDesc) == 0) { 142 ALOGV("effect %d is called %s", i, dwnmFxDesc.name); 143 if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { 144 ALOGI("found effect \"%s\" from %s", 145 dwnmFxDesc.name, dwnmFxDesc.implementor); 146 isMultichannelCapable = true; 147 break; 148 } 149 } 150 } 151 ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect"); 152} 153 154AudioMixer::~AudioMixer() 155{ 156 track_t* t = mState.tracks; 157 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 158 delete t->resampler; 159 delete t->downmixerBufferProvider; 160 t++; 161 } 162 delete [] mState.outputTemp; 163 delete [] mState.resampleTemp; 164} 165 166int AudioMixer::getTrackName(audio_channel_mask_t channelMask) 167{ 168 uint32_t names = (~mTrackNames) & mConfiguredNames; 169 if (names != 0) { 170 int n = __builtin_ctz(names); 171 ALOGV("add track (%d)", n); 172 mTrackNames |= 1 << n; 173 // assume default parameters for the track, except where noted below 174 track_t* t = &mState.tracks[n]; 175 t->needs = 0; 176 t->volume[0] = UNITY_GAIN; 177 t->volume[1] = UNITY_GAIN; 178 // no initialization needed 179 // t->prevVolume[0] 180 // t->prevVolume[1] 181 t->volumeInc[0] = 0; 182 t->volumeInc[1] = 0; 183 t->auxLevel = 0; 184 t->auxInc = 0; 185 // no initialization needed 186 // t->prevAuxLevel 187 // t->frameCount 188 t->channelCount = 2; 189 t->enabled = false; 190 t->format = 16; 191 t->channelMask = AUDIO_CHANNEL_OUT_STEREO; 192 // setBufferProvider(name, AudioBufferProvider *) is required before enable(name) 193 t->bufferProvider = NULL; 194 t->downmixerBufferProvider = NULL; 195 t->buffer.raw = NULL; 196 // no initialization needed 197 // t->buffer.frameCount 198 t->hook = NULL; 199 t->in = NULL; 200 t->resampler = NULL; 201 t->sampleRate = mSampleRate; 202 // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name) 203 t->mainBuffer = NULL; 204 t->auxBuffer = NULL; 205 // see t->localTimeFreq in constructor above 206 207 status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask); 208 if (status == OK) { 209 return TRACK0 + n; 210 } 211 ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix", 212 channelMask); 213 } 214 return -1; 215} 216 217void AudioMixer::invalidateState(uint32_t mask) 218{ 219 if (mask) { 220 mState.needsChanged |= mask; 221 mState.hook = process__validate; 222 } 223 } 224 225status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask) 226{ 227 uint32_t channelCount = popcount(mask); 228 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 229 status_t status = OK; 230 if (channelCount > MAX_NUM_CHANNELS) { 231 pTrack->channelMask = mask; 232 pTrack->channelCount = channelCount; 233 ALOGV("initTrackDownmix(track=%d, mask=0x%x) calls prepareTrackForDownmix()", 234 trackNum, mask); 235 status = prepareTrackForDownmix(pTrack, trackNum); 236 } else { 237 unprepareTrackForDownmix(pTrack, trackNum); 238 } 239 return status; 240} 241 242void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) { 243 ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName); 244 245 if (pTrack->downmixerBufferProvider != NULL) { 246 // this track had previously been configured with a downmixer, delete it 247 ALOGV(" deleting old downmixer"); 248 pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider; 249 delete pTrack->downmixerBufferProvider; 250 pTrack->downmixerBufferProvider = NULL; 251 } else { 252 ALOGV(" nothing to do, no downmixer to delete"); 253 } 254} 255 256status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName) 257{ 258 ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask); 259 260 // discard the previous downmixer if there was one 261 unprepareTrackForDownmix(pTrack, trackName); 262 263 DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(); 264 int32_t status; 265 266 if (!isMultichannelCapable) { 267 ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content", 268 trackName); 269 goto noDownmixForActiveTrack; 270 } 271 272 if (EffectCreate(&dwnmFxDesc.uuid, 273 -2 /*sessionId*/, -2 /*ioId*/,// both not relevant here, using random value 274 &pDbp->mDownmixHandle/*pHandle*/) != 0) { 275 ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName); 276 goto noDownmixForActiveTrack; 277 } 278 279 // channel input configuration will be overridden per-track 280 pDbp->mDownmixConfig.inputCfg.channels = pTrack->channelMask; 281 pDbp->mDownmixConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; 282 pDbp->mDownmixConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 283 pDbp->mDownmixConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 284 pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate; 285 pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate; 286 pDbp->mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 287 pDbp->mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 288 // input and output buffer provider, and frame count will not be used as the downmix effect 289 // process() function is called directly (see DownmixerBufferProvider::getNextBuffer()) 290 pDbp->mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | 291 EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE; 292 pDbp->mDownmixConfig.outputCfg.mask = pDbp->mDownmixConfig.inputCfg.mask; 293 294 {// scope for local variables that are not used in goto label "noDownmixForActiveTrack" 295 int cmdStatus; 296 uint32_t replySize = sizeof(int); 297 298 // Configure and enable downmixer 299 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 300 EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/, 301 &pDbp->mDownmixConfig /*pCmdData*/, 302 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 303 if ((status != 0) || (cmdStatus != 0)) { 304 ALOGE("error %d while configuring downmixer for track %d", status, trackName); 305 goto noDownmixForActiveTrack; 306 } 307 replySize = sizeof(int); 308 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 309 EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/, 310 &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 311 if ((status != 0) || (cmdStatus != 0)) { 312 ALOGE("error %d while enabling downmixer for track %d", status, trackName); 313 goto noDownmixForActiveTrack; 314 } 315 316 // Set downmix type 317 // parameter size rounded for padding on 32bit boundary 318 const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int); 319 const int downmixParamSize = 320 sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t); 321 effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize); 322 param->psize = sizeof(downmix_params_t); 323 const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE; 324 memcpy(param->data, &downmixParam, param->psize); 325 const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD; 326 param->vsize = sizeof(downmix_type_t); 327 memcpy(param->data + psizePadded, &downmixType, param->vsize); 328 329 status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle, 330 EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize/* cmdSize */, 331 param /*pCmndData*/, &replySize /*replySize*/, &cmdStatus /*pReplyData*/); 332 333 free(param); 334 335 if ((status != 0) || (cmdStatus != 0)) { 336 ALOGE("error %d while setting downmix type for track %d", status, trackName); 337 goto noDownmixForActiveTrack; 338 } else { 339 ALOGV("downmix type set to %d for track %d", (int) downmixType, trackName); 340 } 341 }// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack" 342 343 // initialization successful: 344 // - keep track of the real buffer provider in case it was set before 345 pDbp->mTrackBufferProvider = pTrack->bufferProvider; 346 // - we'll use the downmix effect integrated inside this 347 // track's buffer provider, and we'll use it as the track's buffer provider 348 pTrack->downmixerBufferProvider = pDbp; 349 pTrack->bufferProvider = pDbp; 350 351 return NO_ERROR; 352 353noDownmixForActiveTrack: 354 delete pDbp; 355 pTrack->downmixerBufferProvider = NULL; 356 return NO_INIT; 357} 358 359void AudioMixer::deleteTrackName(int name) 360{ 361 ALOGV("AudioMixer::deleteTrackName(%d)", name); 362 name -= TRACK0; 363 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 364 ALOGV("deleteTrackName(%d)", name); 365 track_t& track(mState.tracks[ name ]); 366 if (track.enabled) { 367 track.enabled = false; 368 invalidateState(1<<name); 369 } 370 // delete the resampler 371 delete track.resampler; 372 track.resampler = NULL; 373 // delete the downmixer 374 unprepareTrackForDownmix(&mState.tracks[name], name); 375 376 mTrackNames &= ~(1<<name); 377} 378 379void AudioMixer::enable(int name) 380{ 381 name -= TRACK0; 382 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 383 track_t& track = mState.tracks[name]; 384 385 if (!track.enabled) { 386 track.enabled = true; 387 ALOGV("enable(%d)", name); 388 invalidateState(1 << name); 389 } 390} 391 392void AudioMixer::disable(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 = false; 400 ALOGV("disable(%d)", name); 401 invalidateState(1 << name); 402 } 403} 404 405void AudioMixer::setParameter(int name, int target, int param, void *value) 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 int valueInt = (int)value; 412 int32_t *valueBuf = (int32_t *)value; 413 414 switch (target) { 415 416 case TRACK: 417 switch (param) { 418 case CHANNEL_MASK: { 419 uint32_t mask = (uint32_t)value; 420 if (track.channelMask != mask) { 421 uint32_t channelCount = popcount(mask); 422 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); 423 track.channelMask = mask; 424 track.channelCount = channelCount; 425 // the mask has changed, does this track need a downmixer? 426 initTrackDownmix(&mState.tracks[name], name, mask); 427 ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask); 428 invalidateState(1 << name); 429 } 430 } break; 431 case MAIN_BUFFER: 432 if (track.mainBuffer != valueBuf) { 433 track.mainBuffer = valueBuf; 434 ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); 435 invalidateState(1 << name); 436 } 437 break; 438 case AUX_BUFFER: 439 if (track.auxBuffer != valueBuf) { 440 track.auxBuffer = valueBuf; 441 ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); 442 invalidateState(1 << name); 443 } 444 break; 445 case FORMAT: 446 ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT); 447 break; 448 // FIXME do we want to support setting the downmix type from AudioFlinger? 449 // for a specific track? or per mixer? 450 /* case DOWNMIX_TYPE: 451 break */ 452 default: 453 LOG_FATAL("bad param"); 454 } 455 break; 456 457 case RESAMPLE: 458 switch (param) { 459 case SAMPLE_RATE: 460 ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt); 461 if (track.setResampler(uint32_t(valueInt), mSampleRate)) { 462 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)", 463 uint32_t(valueInt)); 464 invalidateState(1 << name); 465 } 466 break; 467 case RESET: 468 track.resetResampler(); 469 invalidateState(1 << name); 470 break; 471 case REMOVE: 472 delete track.resampler; 473 track.resampler = NULL; 474 track.sampleRate = mSampleRate; 475 invalidateState(1 << name); 476 break; 477 default: 478 LOG_FATAL("bad param"); 479 } 480 break; 481 482 case RAMP_VOLUME: 483 case VOLUME: 484 switch (param) { 485 case VOLUME0: 486 case VOLUME1: 487 if (track.volume[param-VOLUME0] != valueInt) { 488 ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt); 489 track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16; 490 track.volume[param-VOLUME0] = valueInt; 491 if (target == VOLUME) { 492 track.prevVolume[param-VOLUME0] = valueInt << 16; 493 track.volumeInc[param-VOLUME0] = 0; 494 } else { 495 int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0]; 496 int32_t volInc = d / int32_t(mState.frameCount); 497 track.volumeInc[param-VOLUME0] = volInc; 498 if (volInc == 0) { 499 track.prevVolume[param-VOLUME0] = valueInt << 16; 500 } 501 } 502 invalidateState(1 << name); 503 } 504 break; 505 case AUXLEVEL: 506 //ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt); 507 if (track.auxLevel != valueInt) { 508 ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt); 509 track.prevAuxLevel = track.auxLevel << 16; 510 track.auxLevel = valueInt; 511 if (target == VOLUME) { 512 track.prevAuxLevel = valueInt << 16; 513 track.auxInc = 0; 514 } else { 515 int32_t d = (valueInt<<16) - track.prevAuxLevel; 516 int32_t volInc = d / int32_t(mState.frameCount); 517 track.auxInc = volInc; 518 if (volInc == 0) { 519 track.prevAuxLevel = valueInt << 16; 520 } 521 } 522 invalidateState(1 << name); 523 } 524 break; 525 default: 526 LOG_FATAL("bad param"); 527 } 528 break; 529 530 default: 531 LOG_FATAL("bad target"); 532 } 533} 534 535bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) 536{ 537 if (value != devSampleRate || resampler != NULL) { 538 if (sampleRate != value) { 539 sampleRate = value; 540 if (resampler == NULL) { 541 resampler = AudioResampler::create( 542 format, 543 // the resampler sees the number of channels after the downmixer, if any 544 downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount, 545 devSampleRate); 546 resampler->setLocalTimeFreq(localTimeFreq); 547 } 548 return true; 549 } 550 } 551 return false; 552} 553 554inline 555void AudioMixer::track_t::adjustVolumeRamp(bool aux) 556{ 557 for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) { 558 if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || 559 ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { 560 volumeInc[i] = 0; 561 prevVolume[i] = volume[i]<<16; 562 } 563 } 564 if (aux) { 565 if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) || 566 ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) { 567 auxInc = 0; 568 prevAuxLevel = auxLevel<<16; 569 } 570 } 571} 572 573size_t AudioMixer::getUnreleasedFrames(int name) const 574{ 575 name -= TRACK0; 576 if (uint32_t(name) < MAX_NUM_TRACKS) { 577 return mState.tracks[name].getUnreleasedFrames(); 578 } 579 return 0; 580} 581 582void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) 583{ 584 name -= TRACK0; 585 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 586 587 if (mState.tracks[name].downmixerBufferProvider != NULL) { 588 // update required? 589 if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) { 590 ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider); 591 // setting the buffer provider for a track that gets downmixed consists in: 592 // 1/ setting the buffer provider to the "downmix / buffer provider" wrapper 593 // so it's the one that gets called when the buffer provider is needed, 594 mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider; 595 // 2/ saving the buffer provider for the track so the wrapper can use it 596 // when it downmixes. 597 mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider; 598 } 599 } else { 600 mState.tracks[name].bufferProvider = bufferProvider; 601 } 602} 603 604 605 606void AudioMixer::process(int64_t pts) 607{ 608 mState.hook(&mState, pts); 609} 610 611 612void AudioMixer::process__validate(state_t* state, int64_t pts) 613{ 614 ALOGW_IF(!state->needsChanged, 615 "in process__validate() but nothing's invalid"); 616 617 uint32_t changed = state->needsChanged; 618 state->needsChanged = 0; // clear the validation flag 619 620 // recompute which tracks are enabled / disabled 621 uint32_t enabled = 0; 622 uint32_t disabled = 0; 623 while (changed) { 624 const int i = 31 - __builtin_clz(changed); 625 const uint32_t mask = 1<<i; 626 changed &= ~mask; 627 track_t& t = state->tracks[i]; 628 (t.enabled ? enabled : disabled) |= mask; 629 } 630 state->enabledTracks &= ~disabled; 631 state->enabledTracks |= enabled; 632 633 // compute everything we need... 634 int countActiveTracks = 0; 635 bool all16BitsStereoNoResample = true; 636 bool resampling = false; 637 bool volumeRamp = false; 638 uint32_t en = state->enabledTracks; 639 while (en) { 640 const int i = 31 - __builtin_clz(en); 641 en &= ~(1<<i); 642 643 countActiveTracks++; 644 track_t& t = state->tracks[i]; 645 uint32_t n = 0; 646 n |= NEEDS_CHANNEL_1 + t.channelCount - 1; 647 n |= NEEDS_FORMAT_16; 648 n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED; 649 if (t.auxLevel != 0 && t.auxBuffer != NULL) { 650 n |= NEEDS_AUX_ENABLED; 651 } 652 653 if (t.volumeInc[0]|t.volumeInc[1]) { 654 volumeRamp = true; 655 } else if (!t.doesResample() && t.volumeRL == 0) { 656 n |= NEEDS_MUTE_ENABLED; 657 } 658 t.needs = n; 659 660 if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) { 661 t.hook = track__nop; 662 } else { 663 if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { 664 all16BitsStereoNoResample = false; 665 } 666 if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { 667 all16BitsStereoNoResample = false; 668 resampling = true; 669 t.hook = track__genericResample; 670 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 671 "Track %d needs downmix + resample", i); 672 } else { 673 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ 674 t.hook = track__16BitsMono; 675 all16BitsStereoNoResample = false; 676 } 677 if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){ 678 t.hook = track__16BitsStereo; 679 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 680 "Track %d needs downmix", i); 681 } 682 } 683 } 684 } 685 686 // select the processing hooks 687 state->hook = process__nop; 688 if (countActiveTracks) { 689 if (resampling) { 690 if (!state->outputTemp) { 691 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 692 } 693 if (!state->resampleTemp) { 694 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 695 } 696 state->hook = process__genericResampling; 697 } else { 698 if (state->outputTemp) { 699 delete [] state->outputTemp; 700 state->outputTemp = NULL; 701 } 702 if (state->resampleTemp) { 703 delete [] state->resampleTemp; 704 state->resampleTemp = NULL; 705 } 706 state->hook = process__genericNoResampling; 707 if (all16BitsStereoNoResample && !volumeRamp) { 708 if (countActiveTracks == 1) { 709 state->hook = process__OneTrack16BitsStereoNoResampling; 710 } 711 } 712 } 713 } 714 715 ALOGV("mixer configuration change: %d activeTracks (%08x) " 716 "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d", 717 countActiveTracks, state->enabledTracks, 718 all16BitsStereoNoResample, resampling, volumeRamp); 719 720 state->hook(state, pts); 721 722 // Now that the volume ramp has been done, set optimal state and 723 // track hooks for subsequent mixer process 724 if (countActiveTracks) { 725 bool allMuted = true; 726 uint32_t en = state->enabledTracks; 727 while (en) { 728 const int i = 31 - __builtin_clz(en); 729 en &= ~(1<<i); 730 track_t& t = state->tracks[i]; 731 if (!t.doesResample() && t.volumeRL == 0) 732 { 733 t.needs |= NEEDS_MUTE_ENABLED; 734 t.hook = track__nop; 735 } else { 736 allMuted = false; 737 } 738 } 739 if (allMuted) { 740 state->hook = process__nop; 741 } else if (all16BitsStereoNoResample) { 742 if (countActiveTracks == 1) { 743 state->hook = process__OneTrack16BitsStereoNoResampling; 744 } 745 } 746 } 747} 748 749 750void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) 751{ 752 t->resampler->setSampleRate(t->sampleRate); 753 754 // ramp gain - resample to temp buffer and scale/mix in 2nd step 755 if (aux != NULL) { 756 // always resample with unity gain when sending to auxiliary buffer to be able 757 // to apply send level after resampling 758 // TODO: modify each resampler to support aux channel? 759 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 760 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 761 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 762 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 763 volumeRampStereo(t, out, outFrameCount, temp, aux); 764 } else { 765 volumeStereo(t, out, outFrameCount, temp, aux); 766 } 767 } else { 768 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 769 t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); 770 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 771 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 772 volumeRampStereo(t, out, outFrameCount, temp, aux); 773 } 774 775 // constant gain 776 else { 777 t->resampler->setVolume(t->volume[0], t->volume[1]); 778 t->resampler->resample(out, outFrameCount, t->bufferProvider); 779 } 780 } 781} 782 783void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) 784{ 785} 786 787void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 788{ 789 int32_t vl = t->prevVolume[0]; 790 int32_t vr = t->prevVolume[1]; 791 const int32_t vlInc = t->volumeInc[0]; 792 const int32_t vrInc = t->volumeInc[1]; 793 794 //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 795 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 796 // (vl + vlInc*frameCount)/65536.0f, frameCount); 797 798 // ramp volume 799 if (CC_UNLIKELY(aux != NULL)) { 800 int32_t va = t->prevAuxLevel; 801 const int32_t vaInc = t->auxInc; 802 int32_t l; 803 int32_t r; 804 805 do { 806 l = (*temp++ >> 12); 807 r = (*temp++ >> 12); 808 *out++ += (vl >> 16) * l; 809 *out++ += (vr >> 16) * r; 810 *aux++ += (va >> 17) * (l + r); 811 vl += vlInc; 812 vr += vrInc; 813 va += vaInc; 814 } while (--frameCount); 815 t->prevAuxLevel = va; 816 } else { 817 do { 818 *out++ += (vl >> 16) * (*temp++ >> 12); 819 *out++ += (vr >> 16) * (*temp++ >> 12); 820 vl += vlInc; 821 vr += vrInc; 822 } while (--frameCount); 823 } 824 t->prevVolume[0] = vl; 825 t->prevVolume[1] = vr; 826 t->adjustVolumeRamp(aux != NULL); 827} 828 829void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 830{ 831 const int16_t vl = t->volume[0]; 832 const int16_t vr = t->volume[1]; 833 834 if (CC_UNLIKELY(aux != NULL)) { 835 const int16_t va = t->auxLevel; 836 do { 837 int16_t l = (int16_t)(*temp++ >> 12); 838 int16_t r = (int16_t)(*temp++ >> 12); 839 out[0] = mulAdd(l, vl, out[0]); 840 int16_t a = (int16_t)(((int32_t)l + r) >> 1); 841 out[1] = mulAdd(r, vr, out[1]); 842 out += 2; 843 aux[0] = mulAdd(a, va, aux[0]); 844 aux++; 845 } while (--frameCount); 846 } else { 847 do { 848 int16_t l = (int16_t)(*temp++ >> 12); 849 int16_t r = (int16_t)(*temp++ >> 12); 850 out[0] = mulAdd(l, vl, out[0]); 851 out[1] = mulAdd(r, vr, out[1]); 852 out += 2; 853 } while (--frameCount); 854 } 855} 856 857void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 858{ 859 const int16_t *in = static_cast<const int16_t *>(t->in); 860 861 if (CC_UNLIKELY(aux != NULL)) { 862 int32_t l; 863 int32_t r; 864 // ramp gain 865 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 866 int32_t vl = t->prevVolume[0]; 867 int32_t vr = t->prevVolume[1]; 868 int32_t va = t->prevAuxLevel; 869 const int32_t vlInc = t->volumeInc[0]; 870 const int32_t vrInc = t->volumeInc[1]; 871 const int32_t vaInc = t->auxInc; 872 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 873 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 874 // (vl + vlInc*frameCount)/65536.0f, frameCount); 875 876 do { 877 l = (int32_t)*in++; 878 r = (int32_t)*in++; 879 *out++ += (vl >> 16) * l; 880 *out++ += (vr >> 16) * r; 881 *aux++ += (va >> 17) * (l + r); 882 vl += vlInc; 883 vr += vrInc; 884 va += vaInc; 885 } while (--frameCount); 886 887 t->prevVolume[0] = vl; 888 t->prevVolume[1] = vr; 889 t->prevAuxLevel = va; 890 t->adjustVolumeRamp(true); 891 } 892 893 // constant gain 894 else { 895 const uint32_t vrl = t->volumeRL; 896 const int16_t va = (int16_t)t->auxLevel; 897 do { 898 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 899 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1); 900 in += 2; 901 out[0] = mulAddRL(1, rl, vrl, out[0]); 902 out[1] = mulAddRL(0, rl, vrl, out[1]); 903 out += 2; 904 aux[0] = mulAdd(a, va, aux[0]); 905 aux++; 906 } while (--frameCount); 907 } 908 } else { 909 // ramp gain 910 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 911 int32_t vl = t->prevVolume[0]; 912 int32_t vr = t->prevVolume[1]; 913 const int32_t vlInc = t->volumeInc[0]; 914 const int32_t vrInc = t->volumeInc[1]; 915 916 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 917 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 918 // (vl + vlInc*frameCount)/65536.0f, frameCount); 919 920 do { 921 *out++ += (vl >> 16) * (int32_t) *in++; 922 *out++ += (vr >> 16) * (int32_t) *in++; 923 vl += vlInc; 924 vr += vrInc; 925 } while (--frameCount); 926 927 t->prevVolume[0] = vl; 928 t->prevVolume[1] = vr; 929 t->adjustVolumeRamp(false); 930 } 931 932 // constant gain 933 else { 934 const uint32_t vrl = t->volumeRL; 935 do { 936 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 937 in += 2; 938 out[0] = mulAddRL(1, rl, vrl, out[0]); 939 out[1] = mulAddRL(0, rl, vrl, out[1]); 940 out += 2; 941 } while (--frameCount); 942 } 943 } 944 t->in = in; 945} 946 947void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 948{ 949 const int16_t *in = static_cast<int16_t const *>(t->in); 950 951 if (CC_UNLIKELY(aux != NULL)) { 952 // ramp gain 953 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 954 int32_t vl = t->prevVolume[0]; 955 int32_t vr = t->prevVolume[1]; 956 int32_t va = t->prevAuxLevel; 957 const int32_t vlInc = t->volumeInc[0]; 958 const int32_t vrInc = t->volumeInc[1]; 959 const int32_t vaInc = t->auxInc; 960 961 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 962 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 963 // (vl + vlInc*frameCount)/65536.0f, frameCount); 964 965 do { 966 int32_t l = *in++; 967 *out++ += (vl >> 16) * l; 968 *out++ += (vr >> 16) * l; 969 *aux++ += (va >> 16) * l; 970 vl += vlInc; 971 vr += vrInc; 972 va += vaInc; 973 } while (--frameCount); 974 975 t->prevVolume[0] = vl; 976 t->prevVolume[1] = vr; 977 t->prevAuxLevel = va; 978 t->adjustVolumeRamp(true); 979 } 980 // constant gain 981 else { 982 const int16_t vl = t->volume[0]; 983 const int16_t vr = t->volume[1]; 984 const int16_t va = (int16_t)t->auxLevel; 985 do { 986 int16_t l = *in++; 987 out[0] = mulAdd(l, vl, out[0]); 988 out[1] = mulAdd(l, vr, out[1]); 989 out += 2; 990 aux[0] = mulAdd(l, va, aux[0]); 991 aux++; 992 } while (--frameCount); 993 } 994 } else { 995 // ramp gain 996 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 997 int32_t vl = t->prevVolume[0]; 998 int32_t vr = t->prevVolume[1]; 999 const int32_t vlInc = t->volumeInc[0]; 1000 const int32_t vrInc = t->volumeInc[1]; 1001 1002 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1003 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1004 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1005 1006 do { 1007 int32_t l = *in++; 1008 *out++ += (vl >> 16) * l; 1009 *out++ += (vr >> 16) * l; 1010 vl += vlInc; 1011 vr += vrInc; 1012 } while (--frameCount); 1013 1014 t->prevVolume[0] = vl; 1015 t->prevVolume[1] = vr; 1016 t->adjustVolumeRamp(false); 1017 } 1018 // constant gain 1019 else { 1020 const int16_t vl = t->volume[0]; 1021 const int16_t vr = t->volume[1]; 1022 do { 1023 int16_t l = *in++; 1024 out[0] = mulAdd(l, vl, out[0]); 1025 out[1] = mulAdd(l, vr, out[1]); 1026 out += 2; 1027 } while (--frameCount); 1028 } 1029 } 1030 t->in = in; 1031} 1032 1033// no-op case 1034void AudioMixer::process__nop(state_t* state, int64_t pts) 1035{ 1036 uint32_t e0 = state->enabledTracks; 1037 size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS; 1038 while (e0) { 1039 // process by group of tracks with same output buffer to 1040 // avoid multiple memset() on same buffer 1041 uint32_t e1 = e0, e2 = e0; 1042 int i = 31 - __builtin_clz(e1); 1043 track_t& t1 = state->tracks[i]; 1044 e2 &= ~(1<<i); 1045 while (e2) { 1046 i = 31 - __builtin_clz(e2); 1047 e2 &= ~(1<<i); 1048 track_t& t2 = state->tracks[i]; 1049 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1050 e1 &= ~(1<<i); 1051 } 1052 } 1053 e0 &= ~(e1); 1054 1055 memset(t1.mainBuffer, 0, bufSize); 1056 1057 while (e1) { 1058 i = 31 - __builtin_clz(e1); 1059 e1 &= ~(1<<i); 1060 t1 = state->tracks[i]; 1061 size_t outFrames = state->frameCount; 1062 while (outFrames) { 1063 t1.buffer.frameCount = outFrames; 1064 int64_t outputPTS = calculateOutputPTS( 1065 t1, pts, state->frameCount - outFrames); 1066 t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS); 1067 if (t1.buffer.raw == NULL) break; 1068 outFrames -= t1.buffer.frameCount; 1069 t1.bufferProvider->releaseBuffer(&t1.buffer); 1070 } 1071 } 1072 } 1073} 1074 1075// generic code without resampling 1076void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 1077{ 1078 int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32))); 1079 1080 // acquire each track's buffer 1081 uint32_t enabledTracks = state->enabledTracks; 1082 uint32_t e0 = enabledTracks; 1083 while (e0) { 1084 const int i = 31 - __builtin_clz(e0); 1085 e0 &= ~(1<<i); 1086 track_t& t = state->tracks[i]; 1087 t.buffer.frameCount = state->frameCount; 1088 t.bufferProvider->getNextBuffer(&t.buffer, pts); 1089 t.frameCount = t.buffer.frameCount; 1090 t.in = t.buffer.raw; 1091 // t.in == NULL can happen if the track was flushed just after having 1092 // been enabled for mixing. 1093 if (t.in == NULL) 1094 enabledTracks &= ~(1<<i); 1095 } 1096 1097 e0 = enabledTracks; 1098 while (e0) { 1099 // process by group of tracks with same output buffer to 1100 // optimize cache use 1101 uint32_t e1 = e0, e2 = e0; 1102 int j = 31 - __builtin_clz(e1); 1103 track_t& t1 = state->tracks[j]; 1104 e2 &= ~(1<<j); 1105 while (e2) { 1106 j = 31 - __builtin_clz(e2); 1107 e2 &= ~(1<<j); 1108 track_t& t2 = state->tracks[j]; 1109 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1110 e1 &= ~(1<<j); 1111 } 1112 } 1113 e0 &= ~(e1); 1114 // this assumes output 16 bits stereo, no resampling 1115 int32_t *out = t1.mainBuffer; 1116 size_t numFrames = 0; 1117 do { 1118 memset(outTemp, 0, sizeof(outTemp)); 1119 e2 = e1; 1120 while (e2) { 1121 const int i = 31 - __builtin_clz(e2); 1122 e2 &= ~(1<<i); 1123 track_t& t = state->tracks[i]; 1124 size_t outFrames = BLOCKSIZE; 1125 int32_t *aux = NULL; 1126 if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { 1127 aux = t.auxBuffer + numFrames; 1128 } 1129 while (outFrames) { 1130 size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; 1131 if (inFrames) { 1132 t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); 1133 t.frameCount -= inFrames; 1134 outFrames -= inFrames; 1135 if (CC_UNLIKELY(aux != NULL)) { 1136 aux += inFrames; 1137 } 1138 } 1139 if (t.frameCount == 0 && outFrames) { 1140 t.bufferProvider->releaseBuffer(&t.buffer); 1141 t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames); 1142 int64_t outputPTS = calculateOutputPTS( 1143 t, pts, numFrames + (BLOCKSIZE - outFrames)); 1144 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1145 t.in = t.buffer.raw; 1146 if (t.in == NULL) { 1147 enabledTracks &= ~(1<<i); 1148 e1 &= ~(1<<i); 1149 break; 1150 } 1151 t.frameCount = t.buffer.frameCount; 1152 } 1153 } 1154 } 1155 ditherAndClamp(out, outTemp, BLOCKSIZE); 1156 out += BLOCKSIZE; 1157 numFrames += BLOCKSIZE; 1158 } while (numFrames < state->frameCount); 1159 } 1160 1161 // release each track's buffer 1162 e0 = enabledTracks; 1163 while (e0) { 1164 const int i = 31 - __builtin_clz(e0); 1165 e0 &= ~(1<<i); 1166 track_t& t = state->tracks[i]; 1167 t.bufferProvider->releaseBuffer(&t.buffer); 1168 } 1169} 1170 1171 1172// generic code with resampling 1173void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 1174{ 1175 // this const just means that local variable outTemp doesn't change 1176 int32_t* const outTemp = state->outputTemp; 1177 const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount; 1178 1179 size_t numFrames = state->frameCount; 1180 1181 uint32_t e0 = state->enabledTracks; 1182 while (e0) { 1183 // process by group of tracks with same output buffer 1184 // to optimize cache use 1185 uint32_t e1 = e0, e2 = e0; 1186 int j = 31 - __builtin_clz(e1); 1187 track_t& t1 = state->tracks[j]; 1188 e2 &= ~(1<<j); 1189 while (e2) { 1190 j = 31 - __builtin_clz(e2); 1191 e2 &= ~(1<<j); 1192 track_t& t2 = state->tracks[j]; 1193 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1194 e1 &= ~(1<<j); 1195 } 1196 } 1197 e0 &= ~(e1); 1198 int32_t *out = t1.mainBuffer; 1199 memset(outTemp, 0, size); 1200 while (e1) { 1201 const int i = 31 - __builtin_clz(e1); 1202 e1 &= ~(1<<i); 1203 track_t& t = state->tracks[i]; 1204 int32_t *aux = NULL; 1205 if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { 1206 aux = t.auxBuffer; 1207 } 1208 1209 // this is a little goofy, on the resampling case we don't 1210 // acquire/release the buffers because it's done by 1211 // the resampler. 1212 if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { 1213 t.resampler->setPTS(pts); 1214 t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); 1215 } else { 1216 1217 size_t outFrames = 0; 1218 1219 while (outFrames < numFrames) { 1220 t.buffer.frameCount = numFrames - outFrames; 1221 int64_t outputPTS = calculateOutputPTS(t, pts, outFrames); 1222 t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); 1223 t.in = t.buffer.raw; 1224 // t.in == NULL can happen if the track was flushed just after having 1225 // been enabled for mixing. 1226 if (t.in == NULL) break; 1227 1228 if (CC_UNLIKELY(aux != NULL)) { 1229 aux += outFrames; 1230 } 1231 t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux); 1232 outFrames += t.buffer.frameCount; 1233 t.bufferProvider->releaseBuffer(&t.buffer); 1234 } 1235 } 1236 } 1237 ditherAndClamp(out, outTemp, numFrames); 1238 } 1239} 1240 1241// one track, 16 bits stereo without resampling is the most common case 1242void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, 1243 int64_t pts) 1244{ 1245 // This method is only called when state->enabledTracks has exactly 1246 // one bit set. The asserts below would verify this, but are commented out 1247 // since the whole point of this method is to optimize performance. 1248 //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled"); 1249 const int i = 31 - __builtin_clz(state->enabledTracks); 1250 //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled"); 1251 const track_t& t = state->tracks[i]; 1252 1253 AudioBufferProvider::Buffer& b(t.buffer); 1254 1255 int32_t* out = t.mainBuffer; 1256 size_t numFrames = state->frameCount; 1257 1258 const int16_t vl = t.volume[0]; 1259 const int16_t vr = t.volume[1]; 1260 const uint32_t vrl = t.volumeRL; 1261 while (numFrames) { 1262 b.frameCount = numFrames; 1263 int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer); 1264 t.bufferProvider->getNextBuffer(&b, outputPTS); 1265 const int16_t *in = b.i16; 1266 1267 // in == NULL can happen if the track was flushed just after having 1268 // been enabled for mixing. 1269 if (in == NULL || ((unsigned long)in & 3)) { 1270 memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); 1271 ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x", 1272 in, i, t.channelCount, t.needs); 1273 return; 1274 } 1275 size_t outFrames = b.frameCount; 1276 1277 if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) { 1278 // volume is boosted, so we might need to clamp even though 1279 // we process only one track. 1280 do { 1281 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1282 in += 2; 1283 int32_t l = mulRL(1, rl, vrl) >> 12; 1284 int32_t r = mulRL(0, rl, vrl) >> 12; 1285 // clamping... 1286 l = clamp16(l); 1287 r = clamp16(r); 1288 *out++ = (r<<16) | (l & 0xFFFF); 1289 } while (--outFrames); 1290 } else { 1291 do { 1292 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1293 in += 2; 1294 int32_t l = mulRL(1, rl, vrl) >> 12; 1295 int32_t r = mulRL(0, rl, vrl) >> 12; 1296 *out++ = (r<<16) | (l & 0xFFFF); 1297 } while (--outFrames); 1298 } 1299 numFrames -= b.frameCount; 1300 t.bufferProvider->releaseBuffer(&b); 1301 } 1302} 1303 1304#if 0 1305// 2 tracks is also a common case 1306// NEVER used in current implementation of process__validate() 1307// only use if the 2 tracks have the same output buffer 1308void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, 1309 int64_t pts) 1310{ 1311 int i; 1312 uint32_t en = state->enabledTracks; 1313 1314 i = 31 - __builtin_clz(en); 1315 const track_t& t0 = state->tracks[i]; 1316 AudioBufferProvider::Buffer& b0(t0.buffer); 1317 1318 en &= ~(1<<i); 1319 i = 31 - __builtin_clz(en); 1320 const track_t& t1 = state->tracks[i]; 1321 AudioBufferProvider::Buffer& b1(t1.buffer); 1322 1323 const int16_t *in0; 1324 const int16_t vl0 = t0.volume[0]; 1325 const int16_t vr0 = t0.volume[1]; 1326 size_t frameCount0 = 0; 1327 1328 const int16_t *in1; 1329 const int16_t vl1 = t1.volume[0]; 1330 const int16_t vr1 = t1.volume[1]; 1331 size_t frameCount1 = 0; 1332 1333 //FIXME: only works if two tracks use same buffer 1334 int32_t* out = t0.mainBuffer; 1335 size_t numFrames = state->frameCount; 1336 const int16_t *buff = NULL; 1337 1338 1339 while (numFrames) { 1340 1341 if (frameCount0 == 0) { 1342 b0.frameCount = numFrames; 1343 int64_t outputPTS = calculateOutputPTS(t0, pts, 1344 out - t0.mainBuffer); 1345 t0.bufferProvider->getNextBuffer(&b0, outputPTS); 1346 if (b0.i16 == NULL) { 1347 if (buff == NULL) { 1348 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1349 } 1350 in0 = buff; 1351 b0.frameCount = numFrames; 1352 } else { 1353 in0 = b0.i16; 1354 } 1355 frameCount0 = b0.frameCount; 1356 } 1357 if (frameCount1 == 0) { 1358 b1.frameCount = numFrames; 1359 int64_t outputPTS = calculateOutputPTS(t1, pts, 1360 out - t0.mainBuffer); 1361 t1.bufferProvider->getNextBuffer(&b1, outputPTS); 1362 if (b1.i16 == NULL) { 1363 if (buff == NULL) { 1364 buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; 1365 } 1366 in1 = buff; 1367 b1.frameCount = numFrames; 1368 } else { 1369 in1 = b1.i16; 1370 } 1371 frameCount1 = b1.frameCount; 1372 } 1373 1374 size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1; 1375 1376 numFrames -= outFrames; 1377 frameCount0 -= outFrames; 1378 frameCount1 -= outFrames; 1379 1380 do { 1381 int32_t l0 = *in0++; 1382 int32_t r0 = *in0++; 1383 l0 = mul(l0, vl0); 1384 r0 = mul(r0, vr0); 1385 int32_t l = *in1++; 1386 int32_t r = *in1++; 1387 l = mulAdd(l, vl1, l0) >> 12; 1388 r = mulAdd(r, vr1, r0) >> 12; 1389 // clamping... 1390 l = clamp16(l); 1391 r = clamp16(r); 1392 *out++ = (r<<16) | (l & 0xFFFF); 1393 } while (--outFrames); 1394 1395 if (frameCount0 == 0) { 1396 t0.bufferProvider->releaseBuffer(&b0); 1397 } 1398 if (frameCount1 == 0) { 1399 t1.bufferProvider->releaseBuffer(&b1); 1400 } 1401 } 1402 1403 delete [] buff; 1404} 1405#endif 1406 1407int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, 1408 int outputFrameIndex) 1409{ 1410 if (AudioBufferProvider::kInvalidPTS == basePTS) 1411 return AudioBufferProvider::kInvalidPTS; 1412 1413 return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate); 1414} 1415 1416// ---------------------------------------------------------------------------- 1417}; // namespace android 1418