AudioTrack.cpp revision bce50bfc3846ab008bafa75c5d3f29fd7b5395f7
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 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31#define WAIT_STREAM_END_TIMEOUT_SEC 120 32 33 34namespace android { 35// --------------------------------------------------------------------------- 36 37// static 38status_t AudioTrack::getMinFrameCount( 39 size_t* frameCount, 40 audio_stream_type_t streamType, 41 uint32_t sampleRate) 42{ 43 if (frameCount == NULL) { 44 return BAD_VALUE; 45 } 46 47 // FIXME merge with similar code in createTrack_l(), except we're missing 48 // some information here that is available in createTrack_l(): 49 // audio_io_handle_t output 50 // audio_format_t format 51 // audio_channel_mask_t channelMask 52 // audio_output_flags_t flags 53 uint32_t afSampleRate; 54 status_t status; 55 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 56 if (status != NO_ERROR) { 57 ALOGE("Unable to query output sample rate for stream type %d; status %d", 58 streamType, status); 59 return status; 60 } 61 size_t afFrameCount; 62 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 63 if (status != NO_ERROR) { 64 ALOGE("Unable to query output frame count for stream type %d; status %d", 65 streamType, status); 66 return status; 67 } 68 uint32_t afLatency; 69 status = AudioSystem::getOutputLatency(&afLatency, streamType); 70 if (status != NO_ERROR) { 71 ALOGE("Unable to query output latency for stream type %d; status %d", 72 streamType, status); 73 return status; 74 } 75 76 // Ensure that buffer depth covers at least audio hardware latency 77 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 78 if (minBufCount < 2) { 79 minBufCount = 2; 80 } 81 82 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 83 afFrameCount * minBufCount * sampleRate / afSampleRate; 84 // The formula above should always produce a non-zero value, but return an error 85 // in the unlikely event that it does not, as that's part of the API contract. 86 if (*frameCount == 0) { 87 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 88 streamType, sampleRate); 89 return BAD_VALUE; 90 } 91 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 92 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 93 return NO_ERROR; 94} 95 96// --------------------------------------------------------------------------- 97 98AudioTrack::AudioTrack() 99 : mStatus(NO_INIT), 100 mIsTimed(false), 101 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 102 mPreviousSchedulingGroup(SP_DEFAULT) 103{ 104} 105 106AudioTrack::AudioTrack( 107 audio_stream_type_t streamType, 108 uint32_t sampleRate, 109 audio_format_t format, 110 audio_channel_mask_t channelMask, 111 size_t frameCount, 112 audio_output_flags_t flags, 113 callback_t cbf, 114 void* user, 115 uint32_t notificationFrames, 116 int sessionId, 117 transfer_type transferType, 118 const audio_offload_info_t *offloadInfo, 119 int uid, 120 pid_t pid) 121 : mStatus(NO_INIT), 122 mIsTimed(false), 123 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 124 mPreviousSchedulingGroup(SP_DEFAULT) 125{ 126 mStatus = set(streamType, sampleRate, format, channelMask, 127 frameCount, flags, cbf, user, notificationFrames, 128 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 129 offloadInfo, uid, pid); 130} 131 132AudioTrack::AudioTrack( 133 audio_stream_type_t streamType, 134 uint32_t sampleRate, 135 audio_format_t format, 136 audio_channel_mask_t channelMask, 137 const sp<IMemory>& sharedBuffer, 138 audio_output_flags_t flags, 139 callback_t cbf, 140 void* user, 141 uint32_t notificationFrames, 142 int sessionId, 143 transfer_type transferType, 144 const audio_offload_info_t *offloadInfo, 145 int uid, 146 pid_t pid) 147 : mStatus(NO_INIT), 148 mIsTimed(false), 149 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 150 mPreviousSchedulingGroup(SP_DEFAULT) 151{ 152 mStatus = set(streamType, sampleRate, format, channelMask, 153 0 /*frameCount*/, flags, cbf, user, notificationFrames, 154 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 155 uid, pid); 156} 157 158AudioTrack::~AudioTrack() 159{ 160 if (mStatus == NO_ERROR) { 161 // Make sure that callback function exits in the case where 162 // it is looping on buffer full condition in obtainBuffer(). 163 // Otherwise the callback thread will never exit. 164 stop(); 165 if (mAudioTrackThread != 0) { 166 mProxy->interrupt(); 167 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 168 mAudioTrackThread->requestExitAndWait(); 169 mAudioTrackThread.clear(); 170 } 171 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 172 mAudioTrack.clear(); 173 IPCThreadState::self()->flushCommands(); 174 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 175 IPCThreadState::self()->getCallingPid(), mClientPid); 176 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 177 } 178} 179 180status_t AudioTrack::set( 181 audio_stream_type_t streamType, 182 uint32_t sampleRate, 183 audio_format_t format, 184 audio_channel_mask_t channelMask, 185 size_t frameCount, 186 audio_output_flags_t flags, 187 callback_t cbf, 188 void* user, 189 uint32_t notificationFrames, 190 const sp<IMemory>& sharedBuffer, 191 bool threadCanCallJava, 192 int sessionId, 193 transfer_type transferType, 194 const audio_offload_info_t *offloadInfo, 195 int uid, 196 pid_t pid) 197{ 198 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 199 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 200 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 201 sessionId, transferType); 202 203 switch (transferType) { 204 case TRANSFER_DEFAULT: 205 if (sharedBuffer != 0) { 206 transferType = TRANSFER_SHARED; 207 } else if (cbf == NULL || threadCanCallJava) { 208 transferType = TRANSFER_SYNC; 209 } else { 210 transferType = TRANSFER_CALLBACK; 211 } 212 break; 213 case TRANSFER_CALLBACK: 214 if (cbf == NULL || sharedBuffer != 0) { 215 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 216 return BAD_VALUE; 217 } 218 break; 219 case TRANSFER_OBTAIN: 220 case TRANSFER_SYNC: 221 if (sharedBuffer != 0) { 222 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 223 return BAD_VALUE; 224 } 225 break; 226 case TRANSFER_SHARED: 227 if (sharedBuffer == 0) { 228 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 229 return BAD_VALUE; 230 } 231 break; 232 default: 233 ALOGE("Invalid transfer type %d", transferType); 234 return BAD_VALUE; 235 } 236 mSharedBuffer = sharedBuffer; 237 mTransfer = transferType; 238 239 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 240 sharedBuffer->size()); 241 242 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 243 244 AutoMutex lock(mLock); 245 246 // invariant that mAudioTrack != 0 is true only after set() returns successfully 247 if (mAudioTrack != 0) { 248 ALOGE("Track already in use"); 249 return INVALID_OPERATION; 250 } 251 252 // handle default values first. 253 if (streamType == AUDIO_STREAM_DEFAULT) { 254 streamType = AUDIO_STREAM_MUSIC; 255 } 256 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 257 ALOGE("Invalid stream type %d", streamType); 258 return BAD_VALUE; 259 } 260 mStreamType = streamType; 261 262 status_t status; 263 if (sampleRate == 0) { 264 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 265 if (status != NO_ERROR) { 266 ALOGE("Could not get output sample rate for stream type %d; status %d", 267 streamType, status); 268 return status; 269 } 270 } 271 mSampleRate = sampleRate; 272 273 // these below should probably come from the audioFlinger too... 274 if (format == AUDIO_FORMAT_DEFAULT) { 275 format = AUDIO_FORMAT_PCM_16_BIT; 276 } 277 278 // validate parameters 279 if (!audio_is_valid_format(format)) { 280 ALOGE("Invalid format %#x", format); 281 return BAD_VALUE; 282 } 283 mFormat = format; 284 285 if (!audio_is_output_channel(channelMask)) { 286 ALOGE("Invalid channel mask %#x", channelMask); 287 return BAD_VALUE; 288 } 289 mChannelMask = channelMask; 290 uint32_t channelCount = popcount(channelMask); 291 mChannelCount = channelCount; 292 293 // AudioFlinger does not currently support 8-bit data in shared memory 294 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 295 ALOGE("8-bit data in shared memory is not supported"); 296 return BAD_VALUE; 297 } 298 299 // force direct flag if format is not linear PCM 300 // or offload was requested 301 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 302 || !audio_is_linear_pcm(format)) { 303 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 304 ? "Offload request, forcing to Direct Output" 305 : "Not linear PCM, forcing to Direct Output"); 306 flags = (audio_output_flags_t) 307 // FIXME why can't we allow direct AND fast? 308 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 309 } 310 // only allow deep buffering for music stream type 311 if (streamType != AUDIO_STREAM_MUSIC) { 312 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 313 } 314 315 if (audio_is_linear_pcm(format)) { 316 mFrameSize = channelCount * audio_bytes_per_sample(format); 317 mFrameSizeAF = channelCount * sizeof(int16_t); 318 } else { 319 mFrameSize = sizeof(uint8_t); 320 mFrameSizeAF = sizeof(uint8_t); 321 } 322 323 // Make copy of input parameter offloadInfo so that in the future: 324 // (a) createTrack_l doesn't need it as an input parameter 325 // (b) we can support re-creation of offloaded tracks 326 if (offloadInfo != NULL) { 327 mOffloadInfoCopy = *offloadInfo; 328 mOffloadInfo = &mOffloadInfoCopy; 329 } else { 330 mOffloadInfo = NULL; 331 } 332 333 mVolume[LEFT] = 1.0f; 334 mVolume[RIGHT] = 1.0f; 335 mSendLevel = 0.0f; 336 // mFrameCount is initialized in createTrack_l 337 mReqFrameCount = frameCount; 338 mNotificationFramesReq = notificationFrames; 339 mNotificationFramesAct = 0; 340 mSessionId = sessionId; 341 int callingpid = IPCThreadState::self()->getCallingPid(); 342 int mypid = getpid(); 343 if (uid == -1 || (callingpid != mypid)) { 344 mClientUid = IPCThreadState::self()->getCallingUid(); 345 } else { 346 mClientUid = uid; 347 } 348 if (pid == -1 || (callingpid != mypid)) { 349 mClientPid = callingpid; 350 } else { 351 mClientPid = pid; 352 } 353 mAuxEffectId = 0; 354 mFlags = flags; 355 mCbf = cbf; 356 357 if (cbf != NULL) { 358 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 359 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 360 } 361 362 // create the IAudioTrack 363 status = createTrack_l(0 /*epoch*/); 364 365 if (status != NO_ERROR) { 366 if (mAudioTrackThread != 0) { 367 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 368 mAudioTrackThread->requestExitAndWait(); 369 mAudioTrackThread.clear(); 370 } 371 // Use of direct and offloaded output streams is ref counted by audio policy manager. 372#if 0 // FIXME This should no longer be needed 373 //Use of direct and offloaded output streams is ref counted by audio policy manager. 374 // As getOutput was called above and resulted in an output stream to be opened, 375 // we need to release it. 376 if (mOutput != 0) { 377 AudioSystem::releaseOutput(mOutput); 378 mOutput = 0; 379 } 380#endif 381 return status; 382 } 383 384 mStatus = NO_ERROR; 385 mState = STATE_STOPPED; 386 mUserData = user; 387 mLoopPeriod = 0; 388 mMarkerPosition = 0; 389 mMarkerReached = false; 390 mNewPosition = 0; 391 mUpdatePeriod = 0; 392 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 393 mSequence = 1; 394 mObservedSequence = mSequence; 395 mInUnderrun = false; 396 397 return NO_ERROR; 398} 399 400// ------------------------------------------------------------------------- 401 402status_t AudioTrack::start() 403{ 404 AutoMutex lock(mLock); 405 406 if (mState == STATE_ACTIVE) { 407 return INVALID_OPERATION; 408 } 409 410 mInUnderrun = true; 411 412 State previousState = mState; 413 if (previousState == STATE_PAUSED_STOPPING) { 414 mState = STATE_STOPPING; 415 } else { 416 mState = STATE_ACTIVE; 417 } 418 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 419 // reset current position as seen by client to 0 420 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 421 // force refresh of remaining frames by processAudioBuffer() as last 422 // write before stop could be partial. 423 mRefreshRemaining = true; 424 } 425 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 426 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 427 428 sp<AudioTrackThread> t = mAudioTrackThread; 429 if (t != 0) { 430 if (previousState == STATE_STOPPING) { 431 mProxy->interrupt(); 432 } else { 433 t->resume(); 434 } 435 } else { 436 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 437 get_sched_policy(0, &mPreviousSchedulingGroup); 438 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 439 } 440 441 status_t status = NO_ERROR; 442 if (!(flags & CBLK_INVALID)) { 443 status = mAudioTrack->start(); 444 if (status == DEAD_OBJECT) { 445 flags |= CBLK_INVALID; 446 } 447 } 448 if (flags & CBLK_INVALID) { 449 status = restoreTrack_l("start"); 450 } 451 452 if (status != NO_ERROR) { 453 ALOGE("start() status %d", status); 454 mState = previousState; 455 if (t != 0) { 456 if (previousState != STATE_STOPPING) { 457 t->pause(); 458 } 459 } else { 460 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 461 set_sched_policy(0, mPreviousSchedulingGroup); 462 } 463 } 464 465 return status; 466} 467 468void AudioTrack::stop() 469{ 470 AutoMutex lock(mLock); 471 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 472 return; 473 } 474 475 if (isOffloaded_l()) { 476 mState = STATE_STOPPING; 477 } else { 478 mState = STATE_STOPPED; 479 } 480 481 mProxy->interrupt(); 482 mAudioTrack->stop(); 483 // the playback head position will reset to 0, so if a marker is set, we need 484 // to activate it again 485 mMarkerReached = false; 486#if 0 487 // Force flush if a shared buffer is used otherwise audioflinger 488 // will not stop before end of buffer is reached. 489 // It may be needed to make sure that we stop playback, likely in case looping is on. 490 if (mSharedBuffer != 0) { 491 flush_l(); 492 } 493#endif 494 495 sp<AudioTrackThread> t = mAudioTrackThread; 496 if (t != 0) { 497 if (!isOffloaded_l()) { 498 t->pause(); 499 } 500 } else { 501 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 502 set_sched_policy(0, mPreviousSchedulingGroup); 503 } 504} 505 506bool AudioTrack::stopped() const 507{ 508 AutoMutex lock(mLock); 509 return mState != STATE_ACTIVE; 510} 511 512void AudioTrack::flush() 513{ 514 if (mSharedBuffer != 0) { 515 return; 516 } 517 AutoMutex lock(mLock); 518 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 519 return; 520 } 521 flush_l(); 522} 523 524void AudioTrack::flush_l() 525{ 526 ALOG_ASSERT(mState != STATE_ACTIVE); 527 528 // clear playback marker and periodic update counter 529 mMarkerPosition = 0; 530 mMarkerReached = false; 531 mUpdatePeriod = 0; 532 mRefreshRemaining = true; 533 534 mState = STATE_FLUSHED; 535 if (isOffloaded_l()) { 536 mProxy->interrupt(); 537 } 538 mProxy->flush(); 539 mAudioTrack->flush(); 540} 541 542void AudioTrack::pause() 543{ 544 AutoMutex lock(mLock); 545 if (mState == STATE_ACTIVE) { 546 mState = STATE_PAUSED; 547 } else if (mState == STATE_STOPPING) { 548 mState = STATE_PAUSED_STOPPING; 549 } else { 550 return; 551 } 552 mProxy->interrupt(); 553 mAudioTrack->pause(); 554} 555 556status_t AudioTrack::setVolume(float left, float right) 557{ 558 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 559 return BAD_VALUE; 560 } 561 562 AutoMutex lock(mLock); 563 mVolume[LEFT] = left; 564 mVolume[RIGHT] = right; 565 566 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 567 568 if (isOffloaded_l()) { 569 mAudioTrack->signal(); 570 } 571 return NO_ERROR; 572} 573 574status_t AudioTrack::setVolume(float volume) 575{ 576 return setVolume(volume, volume); 577} 578 579status_t AudioTrack::setAuxEffectSendLevel(float level) 580{ 581 if (level < 0.0f || level > 1.0f) { 582 return BAD_VALUE; 583 } 584 585 AutoMutex lock(mLock); 586 mSendLevel = level; 587 mProxy->setSendLevel(level); 588 589 return NO_ERROR; 590} 591 592void AudioTrack::getAuxEffectSendLevel(float* level) const 593{ 594 if (level != NULL) { 595 *level = mSendLevel; 596 } 597} 598 599status_t AudioTrack::setSampleRate(uint32_t rate) 600{ 601 if (mIsTimed || isOffloaded()) { 602 return INVALID_OPERATION; 603 } 604 605 uint32_t afSamplingRate; 606 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 607 return NO_INIT; 608 } 609 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 610 if (rate == 0 || rate > afSamplingRate*2 ) { 611 return BAD_VALUE; 612 } 613 614 AutoMutex lock(mLock); 615 mSampleRate = rate; 616 mProxy->setSampleRate(rate); 617 618 return NO_ERROR; 619} 620 621uint32_t AudioTrack::getSampleRate() const 622{ 623 if (mIsTimed) { 624 return 0; 625 } 626 627 AutoMutex lock(mLock); 628 629 // sample rate can be updated during playback by the offloaded decoder so we need to 630 // query the HAL and update if needed. 631// FIXME use Proxy return channel to update the rate from server and avoid polling here 632 if (isOffloaded_l()) { 633 if (mOutput != 0) { 634 uint32_t sampleRate = 0; 635 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 636 if (status == NO_ERROR) { 637 mSampleRate = sampleRate; 638 } 639 } 640 } 641 return mSampleRate; 642} 643 644status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 645{ 646 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 647 return INVALID_OPERATION; 648 } 649 650 if (loopCount == 0) { 651 ; 652 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 653 loopEnd - loopStart >= MIN_LOOP) { 654 ; 655 } else { 656 return BAD_VALUE; 657 } 658 659 AutoMutex lock(mLock); 660 // See setPosition() regarding setting parameters such as loop points or position while active 661 if (mState == STATE_ACTIVE) { 662 return INVALID_OPERATION; 663 } 664 setLoop_l(loopStart, loopEnd, loopCount); 665 return NO_ERROR; 666} 667 668void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 669{ 670 // FIXME If setting a loop also sets position to start of loop, then 671 // this is correct. Otherwise it should be removed. 672 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 673 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 674 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 675} 676 677status_t AudioTrack::setMarkerPosition(uint32_t marker) 678{ 679 // The only purpose of setting marker position is to get a callback 680 if (mCbf == NULL || isOffloaded()) { 681 return INVALID_OPERATION; 682 } 683 684 AutoMutex lock(mLock); 685 mMarkerPosition = marker; 686 mMarkerReached = false; 687 688 return NO_ERROR; 689} 690 691status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 692{ 693 if (isOffloaded()) { 694 return INVALID_OPERATION; 695 } 696 if (marker == NULL) { 697 return BAD_VALUE; 698 } 699 700 AutoMutex lock(mLock); 701 *marker = mMarkerPosition; 702 703 return NO_ERROR; 704} 705 706status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 707{ 708 // The only purpose of setting position update period is to get a callback 709 if (mCbf == NULL || isOffloaded()) { 710 return INVALID_OPERATION; 711 } 712 713 AutoMutex lock(mLock); 714 mNewPosition = mProxy->getPosition() + updatePeriod; 715 mUpdatePeriod = updatePeriod; 716 717 return NO_ERROR; 718} 719 720status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 721{ 722 if (isOffloaded()) { 723 return INVALID_OPERATION; 724 } 725 if (updatePeriod == NULL) { 726 return BAD_VALUE; 727 } 728 729 AutoMutex lock(mLock); 730 *updatePeriod = mUpdatePeriod; 731 732 return NO_ERROR; 733} 734 735status_t AudioTrack::setPosition(uint32_t position) 736{ 737 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 738 return INVALID_OPERATION; 739 } 740 if (position > mFrameCount) { 741 return BAD_VALUE; 742 } 743 744 AutoMutex lock(mLock); 745 // Currently we require that the player is inactive before setting parameters such as position 746 // or loop points. Otherwise, there could be a race condition: the application could read the 747 // current position, compute a new position or loop parameters, and then set that position or 748 // loop parameters but it would do the "wrong" thing since the position has continued to advance 749 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 750 // to specify how it wants to handle such scenarios. 751 if (mState == STATE_ACTIVE) { 752 return INVALID_OPERATION; 753 } 754 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 755 mLoopPeriod = 0; 756 // FIXME Check whether loops and setting position are incompatible in old code. 757 // If we use setLoop for both purposes we lose the capability to set the position while looping. 758 mStaticProxy->setLoop(position, mFrameCount, 0); 759 760 return NO_ERROR; 761} 762 763status_t AudioTrack::getPosition(uint32_t *position) const 764{ 765 if (position == NULL) { 766 return BAD_VALUE; 767 } 768 769 AutoMutex lock(mLock); 770 if (isOffloaded_l()) { 771 uint32_t dspFrames = 0; 772 773 if (mOutput != 0) { 774 uint32_t halFrames; 775 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 776 } 777 *position = dspFrames; 778 } else { 779 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 780 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 781 mProxy->getPosition(); 782 } 783 return NO_ERROR; 784} 785 786status_t AudioTrack::getBufferPosition(uint32_t *position) 787{ 788 if (mSharedBuffer == 0 || mIsTimed) { 789 return INVALID_OPERATION; 790 } 791 if (position == NULL) { 792 return BAD_VALUE; 793 } 794 795 AutoMutex lock(mLock); 796 *position = mStaticProxy->getBufferPosition(); 797 return NO_ERROR; 798} 799 800status_t AudioTrack::reload() 801{ 802 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 803 return INVALID_OPERATION; 804 } 805 806 AutoMutex lock(mLock); 807 // See setPosition() regarding setting parameters such as loop points or position while active 808 if (mState == STATE_ACTIVE) { 809 return INVALID_OPERATION; 810 } 811 mNewPosition = mUpdatePeriod; 812 mLoopPeriod = 0; 813 // FIXME The new code cannot reload while keeping a loop specified. 814 // Need to check how the old code handled this, and whether it's a significant change. 815 mStaticProxy->setLoop(0, mFrameCount, 0); 816 return NO_ERROR; 817} 818 819audio_io_handle_t AudioTrack::getOutput() const 820{ 821 AutoMutex lock(mLock); 822 return mOutput; 823} 824 825status_t AudioTrack::attachAuxEffect(int effectId) 826{ 827 AutoMutex lock(mLock); 828 status_t status = mAudioTrack->attachAuxEffect(effectId); 829 if (status == NO_ERROR) { 830 mAuxEffectId = effectId; 831 } 832 return status; 833} 834 835// ------------------------------------------------------------------------- 836 837// must be called with mLock held 838status_t AudioTrack::createTrack_l(size_t epoch) 839{ 840 status_t status; 841 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 842 if (audioFlinger == 0) { 843 ALOGE("Could not get audioflinger"); 844 return NO_INIT; 845 } 846 847 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 848 mChannelMask, mFlags, mOffloadInfo); 849 if (output == 0) { 850 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 851 "channel mask %#x, flags %#x", 852 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 853 return BAD_VALUE; 854 } 855 { 856 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 857 // we must release it ourselves if anything goes wrong. 858 859 // Not all of these values are needed under all conditions, but it is easier to get them all 860 861 uint32_t afLatency; 862 status = AudioSystem::getLatency(output, mStreamType, &afLatency); 863 if (status != NO_ERROR) { 864 ALOGE("getLatency(%d) failed status %d", output, status); 865 goto release; 866 } 867 868 size_t afFrameCount; 869 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 870 if (status != NO_ERROR) { 871 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 872 goto release; 873 } 874 875 uint32_t afSampleRate; 876 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 877 if (status != NO_ERROR) { 878 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 879 goto release; 880 } 881 882 // Client decides whether the track is TIMED (see below), but can only express a preference 883 // for FAST. Server will perform additional tests. 884 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 885 // either of these use cases: 886 // use case 1: shared buffer 887 (mSharedBuffer != 0) || 888 // use case 2: callback handler 889 (mCbf != NULL)) && 890 // matching sample rate 891 (mSampleRate == afSampleRate))) { 892 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 893 // once denied, do not request again if IAudioTrack is re-created 894 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 895 } 896 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 897 898 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 899 // n = 1 fast track with single buffering; nBuffering is ignored 900 // n = 2 fast track with double buffering 901 // n = 2 normal track, no sample rate conversion 902 // n = 3 normal track, with sample rate conversion 903 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 904 // n > 3 very high latency or very small notification interval; nBuffering is ignored 905 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 906 907 mNotificationFramesAct = mNotificationFramesReq; 908 909 size_t frameCount = mReqFrameCount; 910 if (!audio_is_linear_pcm(mFormat)) { 911 912 if (mSharedBuffer != 0) { 913 // Same comment as below about ignoring frameCount parameter for set() 914 frameCount = mSharedBuffer->size(); 915 } else if (frameCount == 0) { 916 frameCount = afFrameCount; 917 } 918 if (mNotificationFramesAct != frameCount) { 919 mNotificationFramesAct = frameCount; 920 } 921 } else if (mSharedBuffer != 0) { 922 923 // Ensure that buffer alignment matches channel count 924 // 8-bit data in shared memory is not currently supported by AudioFlinger 925 size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 926 if (mChannelCount > 1) { 927 // More than 2 channels does not require stronger alignment than stereo 928 alignment <<= 1; 929 } 930 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 931 ALOGE("Invalid buffer alignment: address %p, channel count %u", 932 mSharedBuffer->pointer(), mChannelCount); 933 status = BAD_VALUE; 934 goto release; 935 } 936 937 // When initializing a shared buffer AudioTrack via constructors, 938 // there's no frameCount parameter. 939 // But when initializing a shared buffer AudioTrack via set(), 940 // there _is_ a frameCount parameter. We silently ignore it. 941 frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); 942 943 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 944 945 // FIXME move these calculations and associated checks to server 946 947 // Ensure that buffer depth covers at least audio hardware latency 948 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 949 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 950 afFrameCount, minBufCount, afSampleRate, afLatency); 951 if (minBufCount <= nBuffering) { 952 minBufCount = nBuffering; 953 } 954 955 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 956 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 957 ", afLatency=%d", 958 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 959 960 if (frameCount == 0) { 961 frameCount = minFrameCount; 962 } else if (frameCount < minFrameCount) { 963 // not ALOGW because it happens all the time when playing key clicks over A2DP 964 ALOGV("Minimum buffer size corrected from %d to %d", 965 frameCount, minFrameCount); 966 frameCount = minFrameCount; 967 } 968 // Make sure that application is notified with sufficient margin before underrun 969 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 970 mNotificationFramesAct = frameCount/nBuffering; 971 } 972 973 } else { 974 // For fast tracks, the frame count calculations and checks are done by server 975 } 976 977 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 978 if (mIsTimed) { 979 trackFlags |= IAudioFlinger::TRACK_TIMED; 980 } 981 982 pid_t tid = -1; 983 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 984 trackFlags |= IAudioFlinger::TRACK_FAST; 985 if (mAudioTrackThread != 0) { 986 tid = mAudioTrackThread->getTid(); 987 } 988 } 989 990 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 991 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 992 } 993 994 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 995 // but we will still need the original value also 996 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 997 mSampleRate, 998 // AudioFlinger only sees 16-bit PCM 999 mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1000 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1001 mChannelMask, 1002 &temp, 1003 &trackFlags, 1004 mSharedBuffer, 1005 output, 1006 tid, 1007 &mSessionId, 1008 mName, 1009 mClientUid, 1010 &status); 1011 1012 if (status != NO_ERROR) { 1013 ALOGE("AudioFlinger could not create track, status: %d", status); 1014 goto release; 1015 } 1016 ALOG_ASSERT(track != 0); 1017 1018 // AudioFlinger now owns the reference to the I/O handle, 1019 // so we are no longer responsible for releasing it. 1020 1021 sp<IMemory> iMem = track->getCblk(); 1022 if (iMem == 0) { 1023 ALOGE("Could not get control block"); 1024 return NO_INIT; 1025 } 1026 void *iMemPointer = iMem->pointer(); 1027 if (iMemPointer == NULL) { 1028 ALOGE("Could not get control block pointer"); 1029 return NO_INIT; 1030 } 1031 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1032 if (mAudioTrack != 0) { 1033 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1034 mDeathNotifier.clear(); 1035 } 1036 mAudioTrack = track; 1037 1038 mCblkMemory = iMem; 1039 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1040 mCblk = cblk; 1041 // note that temp is the (possibly revised) value of frameCount 1042 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1043 // In current design, AudioTrack client checks and ensures frame count validity before 1044 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1045 // for fast track as it uses a special method of assigning frame count. 1046 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1047 } 1048 frameCount = temp; 1049 1050 mAwaitBoost = false; 1051 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1052 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1053 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1054 mAwaitBoost = true; 1055 if (mSharedBuffer == 0) { 1056 // Theoretically double-buffering is not required for fast tracks, 1057 // due to tighter scheduling. But in practice, to accommodate kernels with 1058 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1059 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1060 mNotificationFramesAct = frameCount/nBuffering; 1061 } 1062 } 1063 } else { 1064 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1065 // once denied, do not request again if IAudioTrack is re-created 1066 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1067 if (mSharedBuffer == 0) { 1068 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1069 mNotificationFramesAct = frameCount/nBuffering; 1070 } 1071 } 1072 } 1073 } 1074 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1075 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1076 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1077 } else { 1078 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1079 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1080 // FIXME This is a warning, not an error, so don't return error status 1081 //return NO_INIT; 1082 } 1083 } 1084 1085 // We retain a copy of the I/O handle, but don't own the reference 1086 mOutput = output; 1087 mRefreshRemaining = true; 1088 1089 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1090 // is the value of pointer() for the shared buffer, otherwise buffers points 1091 // immediately after the control block. This address is for the mapping within client 1092 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1093 void* buffers; 1094 if (mSharedBuffer == 0) { 1095 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1096 } else { 1097 buffers = mSharedBuffer->pointer(); 1098 } 1099 1100 mAudioTrack->attachAuxEffect(mAuxEffectId); 1101 // FIXME don't believe this lie 1102 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1103 1104 mFrameCount = frameCount; 1105 // If IAudioTrack is re-created, don't let the requested frameCount 1106 // decrease. This can confuse clients that cache frameCount(). 1107 if (frameCount > mReqFrameCount) { 1108 mReqFrameCount = frameCount; 1109 } 1110 1111 // update proxy 1112 if (mSharedBuffer == 0) { 1113 mStaticProxy.clear(); 1114 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1115 } else { 1116 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1117 mProxy = mStaticProxy; 1118 } 1119 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1120 uint16_t(mVolume[LEFT] * 0x1000)); 1121 mProxy->setSendLevel(mSendLevel); 1122 mProxy->setSampleRate(mSampleRate); 1123 mProxy->setEpoch(epoch); 1124 mProxy->setMinimum(mNotificationFramesAct); 1125 1126 mDeathNotifier = new DeathNotifier(this); 1127 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1128 1129 return NO_ERROR; 1130 } 1131 1132release: 1133 AudioSystem::releaseOutput(output); 1134 if (status == NO_ERROR) { 1135 status = NO_INIT; 1136 } 1137 return status; 1138} 1139 1140status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1141{ 1142 if (audioBuffer == NULL) { 1143 return BAD_VALUE; 1144 } 1145 if (mTransfer != TRANSFER_OBTAIN) { 1146 audioBuffer->frameCount = 0; 1147 audioBuffer->size = 0; 1148 audioBuffer->raw = NULL; 1149 return INVALID_OPERATION; 1150 } 1151 1152 const struct timespec *requested; 1153 struct timespec timeout; 1154 if (waitCount == -1) { 1155 requested = &ClientProxy::kForever; 1156 } else if (waitCount == 0) { 1157 requested = &ClientProxy::kNonBlocking; 1158 } else if (waitCount > 0) { 1159 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1160 timeout.tv_sec = ms / 1000; 1161 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1162 requested = &timeout; 1163 } else { 1164 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1165 requested = NULL; 1166 } 1167 return obtainBuffer(audioBuffer, requested); 1168} 1169 1170status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1171 struct timespec *elapsed, size_t *nonContig) 1172{ 1173 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1174 uint32_t oldSequence = 0; 1175 uint32_t newSequence; 1176 1177 Proxy::Buffer buffer; 1178 status_t status = NO_ERROR; 1179 1180 static const int32_t kMaxTries = 5; 1181 int32_t tryCounter = kMaxTries; 1182 1183 do { 1184 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1185 // keep them from going away if another thread re-creates the track during obtainBuffer() 1186 sp<AudioTrackClientProxy> proxy; 1187 sp<IMemory> iMem; 1188 1189 { // start of lock scope 1190 AutoMutex lock(mLock); 1191 1192 newSequence = mSequence; 1193 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1194 if (status == DEAD_OBJECT) { 1195 // re-create track, unless someone else has already done so 1196 if (newSequence == oldSequence) { 1197 status = restoreTrack_l("obtainBuffer"); 1198 if (status != NO_ERROR) { 1199 buffer.mFrameCount = 0; 1200 buffer.mRaw = NULL; 1201 buffer.mNonContig = 0; 1202 break; 1203 } 1204 } 1205 } 1206 oldSequence = newSequence; 1207 1208 // Keep the extra references 1209 proxy = mProxy; 1210 iMem = mCblkMemory; 1211 1212 if (mState == STATE_STOPPING) { 1213 status = -EINTR; 1214 buffer.mFrameCount = 0; 1215 buffer.mRaw = NULL; 1216 buffer.mNonContig = 0; 1217 break; 1218 } 1219 1220 // Non-blocking if track is stopped or paused 1221 if (mState != STATE_ACTIVE) { 1222 requested = &ClientProxy::kNonBlocking; 1223 } 1224 1225 } // end of lock scope 1226 1227 buffer.mFrameCount = audioBuffer->frameCount; 1228 // FIXME starts the requested timeout and elapsed over from scratch 1229 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1230 1231 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1232 1233 audioBuffer->frameCount = buffer.mFrameCount; 1234 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1235 audioBuffer->raw = buffer.mRaw; 1236 if (nonContig != NULL) { 1237 *nonContig = buffer.mNonContig; 1238 } 1239 return status; 1240} 1241 1242void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1243{ 1244 if (mTransfer == TRANSFER_SHARED) { 1245 return; 1246 } 1247 1248 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1249 if (stepCount == 0) { 1250 return; 1251 } 1252 1253 Proxy::Buffer buffer; 1254 buffer.mFrameCount = stepCount; 1255 buffer.mRaw = audioBuffer->raw; 1256 1257 AutoMutex lock(mLock); 1258 mInUnderrun = false; 1259 mProxy->releaseBuffer(&buffer); 1260 1261 // restart track if it was disabled by audioflinger due to previous underrun 1262 if (mState == STATE_ACTIVE) { 1263 audio_track_cblk_t* cblk = mCblk; 1264 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1265 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1266 this, mName.string()); 1267 // FIXME ignoring status 1268 mAudioTrack->start(); 1269 } 1270 } 1271} 1272 1273// ------------------------------------------------------------------------- 1274 1275ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1276{ 1277 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1278 return INVALID_OPERATION; 1279 } 1280 1281 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1282 // Sanity-check: user is most-likely passing an error code, and it would 1283 // make the return value ambiguous (actualSize vs error). 1284 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1285 return BAD_VALUE; 1286 } 1287 1288 size_t written = 0; 1289 Buffer audioBuffer; 1290 1291 while (userSize >= mFrameSize) { 1292 audioBuffer.frameCount = userSize / mFrameSize; 1293 1294 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1295 if (err < 0) { 1296 if (written > 0) { 1297 break; 1298 } 1299 return ssize_t(err); 1300 } 1301 1302 size_t toWrite; 1303 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1304 // Divide capacity by 2 to take expansion into account 1305 toWrite = audioBuffer.size >> 1; 1306 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1307 } else { 1308 toWrite = audioBuffer.size; 1309 memcpy(audioBuffer.i8, buffer, toWrite); 1310 } 1311 buffer = ((const char *) buffer) + toWrite; 1312 userSize -= toWrite; 1313 written += toWrite; 1314 1315 releaseBuffer(&audioBuffer); 1316 } 1317 1318 return written; 1319} 1320 1321// ------------------------------------------------------------------------- 1322 1323TimedAudioTrack::TimedAudioTrack() { 1324 mIsTimed = true; 1325} 1326 1327status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1328{ 1329 AutoMutex lock(mLock); 1330 status_t result = UNKNOWN_ERROR; 1331 1332#if 1 1333 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1334 // while we are accessing the cblk 1335 sp<IAudioTrack> audioTrack = mAudioTrack; 1336 sp<IMemory> iMem = mCblkMemory; 1337#endif 1338 1339 // If the track is not invalid already, try to allocate a buffer. alloc 1340 // fails indicating that the server is dead, flag the track as invalid so 1341 // we can attempt to restore in just a bit. 1342 audio_track_cblk_t* cblk = mCblk; 1343 if (!(cblk->mFlags & CBLK_INVALID)) { 1344 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1345 if (result == DEAD_OBJECT) { 1346 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1347 } 1348 } 1349 1350 // If the track is invalid at this point, attempt to restore it. and try the 1351 // allocation one more time. 1352 if (cblk->mFlags & CBLK_INVALID) { 1353 result = restoreTrack_l("allocateTimedBuffer"); 1354 1355 if (result == NO_ERROR) { 1356 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1357 } 1358 } 1359 1360 return result; 1361} 1362 1363status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1364 int64_t pts) 1365{ 1366 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1367 { 1368 AutoMutex lock(mLock); 1369 audio_track_cblk_t* cblk = mCblk; 1370 // restart track if it was disabled by audioflinger due to previous underrun 1371 if (buffer->size() != 0 && status == NO_ERROR && 1372 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1373 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1374 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1375 // FIXME ignoring status 1376 mAudioTrack->start(); 1377 } 1378 } 1379 return status; 1380} 1381 1382status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1383 TargetTimeline target) 1384{ 1385 return mAudioTrack->setMediaTimeTransform(xform, target); 1386} 1387 1388// ------------------------------------------------------------------------- 1389 1390nsecs_t AudioTrack::processAudioBuffer() 1391{ 1392 // Currently the AudioTrack thread is not created if there are no callbacks. 1393 // Would it ever make sense to run the thread, even without callbacks? 1394 // If so, then replace this by checks at each use for mCbf != NULL. 1395 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1396 1397 mLock.lock(); 1398 if (mAwaitBoost) { 1399 mAwaitBoost = false; 1400 mLock.unlock(); 1401 static const int32_t kMaxTries = 5; 1402 int32_t tryCounter = kMaxTries; 1403 uint32_t pollUs = 10000; 1404 do { 1405 int policy = sched_getscheduler(0); 1406 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1407 break; 1408 } 1409 usleep(pollUs); 1410 pollUs <<= 1; 1411 } while (tryCounter-- > 0); 1412 if (tryCounter < 0) { 1413 ALOGE("did not receive expected priority boost on time"); 1414 } 1415 // Run again immediately 1416 return 0; 1417 } 1418 1419 // Can only reference mCblk while locked 1420 int32_t flags = android_atomic_and( 1421 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1422 1423 // Check for track invalidation 1424 if (flags & CBLK_INVALID) { 1425 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1426 // AudioSystem cache. We should not exit here but after calling the callback so 1427 // that the upper layers can recreate the track 1428 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1429 status_t status = restoreTrack_l("processAudioBuffer"); 1430 mLock.unlock(); 1431 // Run again immediately, but with a new IAudioTrack 1432 return 0; 1433 } 1434 } 1435 1436 bool waitStreamEnd = mState == STATE_STOPPING; 1437 bool active = mState == STATE_ACTIVE; 1438 1439 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1440 bool newUnderrun = false; 1441 if (flags & CBLK_UNDERRUN) { 1442#if 0 1443 // Currently in shared buffer mode, when the server reaches the end of buffer, 1444 // the track stays active in continuous underrun state. It's up to the application 1445 // to pause or stop the track, or set the position to a new offset within buffer. 1446 // This was some experimental code to auto-pause on underrun. Keeping it here 1447 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1448 if (mTransfer == TRANSFER_SHARED) { 1449 mState = STATE_PAUSED; 1450 active = false; 1451 } 1452#endif 1453 if (!mInUnderrun) { 1454 mInUnderrun = true; 1455 newUnderrun = true; 1456 } 1457 } 1458 1459 // Get current position of server 1460 size_t position = mProxy->getPosition(); 1461 1462 // Manage marker callback 1463 bool markerReached = false; 1464 size_t markerPosition = mMarkerPosition; 1465 // FIXME fails for wraparound, need 64 bits 1466 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1467 mMarkerReached = markerReached = true; 1468 } 1469 1470 // Determine number of new position callback(s) that will be needed, while locked 1471 size_t newPosCount = 0; 1472 size_t newPosition = mNewPosition; 1473 size_t updatePeriod = mUpdatePeriod; 1474 // FIXME fails for wraparound, need 64 bits 1475 if (updatePeriod > 0 && position >= newPosition) { 1476 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1477 mNewPosition += updatePeriod * newPosCount; 1478 } 1479 1480 // Cache other fields that will be needed soon 1481 uint32_t loopPeriod = mLoopPeriod; 1482 uint32_t sampleRate = mSampleRate; 1483 uint32_t notificationFrames = mNotificationFramesAct; 1484 if (mRefreshRemaining) { 1485 mRefreshRemaining = false; 1486 mRemainingFrames = notificationFrames; 1487 mRetryOnPartialBuffer = false; 1488 } 1489 size_t misalignment = mProxy->getMisalignment(); 1490 uint32_t sequence = mSequence; 1491 1492 // These fields don't need to be cached, because they are assigned only by set(): 1493 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1494 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1495 1496 mLock.unlock(); 1497 1498 if (waitStreamEnd) { 1499 AutoMutex lock(mLock); 1500 1501 sp<AudioTrackClientProxy> proxy = mProxy; 1502 sp<IMemory> iMem = mCblkMemory; 1503 1504 struct timespec timeout; 1505 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1506 timeout.tv_nsec = 0; 1507 1508 mLock.unlock(); 1509 status_t status = mProxy->waitStreamEndDone(&timeout); 1510 mLock.lock(); 1511 switch (status) { 1512 case NO_ERROR: 1513 case DEAD_OBJECT: 1514 case TIMED_OUT: 1515 mLock.unlock(); 1516 mCbf(EVENT_STREAM_END, mUserData, NULL); 1517 mLock.lock(); 1518 if (mState == STATE_STOPPING) { 1519 mState = STATE_STOPPED; 1520 if (status != DEAD_OBJECT) { 1521 return NS_INACTIVE; 1522 } 1523 } 1524 return 0; 1525 default: 1526 return 0; 1527 } 1528 } 1529 1530 // perform callbacks while unlocked 1531 if (newUnderrun) { 1532 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1533 } 1534 // FIXME we will miss loops if loop cycle was signaled several times since last call 1535 // to processAudioBuffer() 1536 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1537 mCbf(EVENT_LOOP_END, mUserData, NULL); 1538 } 1539 if (flags & CBLK_BUFFER_END) { 1540 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1541 } 1542 if (markerReached) { 1543 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1544 } 1545 while (newPosCount > 0) { 1546 size_t temp = newPosition; 1547 mCbf(EVENT_NEW_POS, mUserData, &temp); 1548 newPosition += updatePeriod; 1549 newPosCount--; 1550 } 1551 1552 if (mObservedSequence != sequence) { 1553 mObservedSequence = sequence; 1554 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1555 // for offloaded tracks, just wait for the upper layers to recreate the track 1556 if (isOffloaded()) { 1557 return NS_INACTIVE; 1558 } 1559 } 1560 1561 // if inactive, then don't run me again until re-started 1562 if (!active) { 1563 return NS_INACTIVE; 1564 } 1565 1566 // Compute the estimated time until the next timed event (position, markers, loops) 1567 // FIXME only for non-compressed audio 1568 uint32_t minFrames = ~0; 1569 if (!markerReached && position < markerPosition) { 1570 minFrames = markerPosition - position; 1571 } 1572 if (loopPeriod > 0 && loopPeriod < minFrames) { 1573 minFrames = loopPeriod; 1574 } 1575 if (updatePeriod > 0 && updatePeriod < minFrames) { 1576 minFrames = updatePeriod; 1577 } 1578 1579 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1580 static const uint32_t kPoll = 0; 1581 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1582 minFrames = kPoll * notificationFrames; 1583 } 1584 1585 // Convert frame units to time units 1586 nsecs_t ns = NS_WHENEVER; 1587 if (minFrames != (uint32_t) ~0) { 1588 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1589 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1590 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1591 } 1592 1593 // If not supplying data by EVENT_MORE_DATA, then we're done 1594 if (mTransfer != TRANSFER_CALLBACK) { 1595 return ns; 1596 } 1597 1598 struct timespec timeout; 1599 const struct timespec *requested = &ClientProxy::kForever; 1600 if (ns != NS_WHENEVER) { 1601 timeout.tv_sec = ns / 1000000000LL; 1602 timeout.tv_nsec = ns % 1000000000LL; 1603 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1604 requested = &timeout; 1605 } 1606 1607 while (mRemainingFrames > 0) { 1608 1609 Buffer audioBuffer; 1610 audioBuffer.frameCount = mRemainingFrames; 1611 size_t nonContig; 1612 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1613 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1614 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1615 requested = &ClientProxy::kNonBlocking; 1616 size_t avail = audioBuffer.frameCount + nonContig; 1617 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1618 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1619 if (err != NO_ERROR) { 1620 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1621 (isOffloaded() && (err == DEAD_OBJECT))) { 1622 return 0; 1623 } 1624 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1625 return NS_NEVER; 1626 } 1627 1628 if (mRetryOnPartialBuffer && !isOffloaded()) { 1629 mRetryOnPartialBuffer = false; 1630 if (avail < mRemainingFrames) { 1631 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1632 if (ns < 0 || myns < ns) { 1633 ns = myns; 1634 } 1635 return ns; 1636 } 1637 } 1638 1639 // Divide buffer size by 2 to take into account the expansion 1640 // due to 8 to 16 bit conversion: the callback must fill only half 1641 // of the destination buffer 1642 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1643 audioBuffer.size >>= 1; 1644 } 1645 1646 size_t reqSize = audioBuffer.size; 1647 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1648 size_t writtenSize = audioBuffer.size; 1649 1650 // Sanity check on returned size 1651 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1652 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1653 reqSize, (int) writtenSize); 1654 return NS_NEVER; 1655 } 1656 1657 if (writtenSize == 0) { 1658 // The callback is done filling buffers 1659 // Keep this thread going to handle timed events and 1660 // still try to get more data in intervals of WAIT_PERIOD_MS 1661 // but don't just loop and block the CPU, so wait 1662 return WAIT_PERIOD_MS * 1000000LL; 1663 } 1664 1665 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1666 // 8 to 16 bit conversion, note that source and destination are the same address 1667 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1668 audioBuffer.size <<= 1; 1669 } 1670 1671 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1672 audioBuffer.frameCount = releasedFrames; 1673 mRemainingFrames -= releasedFrames; 1674 if (misalignment >= releasedFrames) { 1675 misalignment -= releasedFrames; 1676 } else { 1677 misalignment = 0; 1678 } 1679 1680 releaseBuffer(&audioBuffer); 1681 1682 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1683 // if callback doesn't like to accept the full chunk 1684 if (writtenSize < reqSize) { 1685 continue; 1686 } 1687 1688 // There could be enough non-contiguous frames available to satisfy the remaining request 1689 if (mRemainingFrames <= nonContig) { 1690 continue; 1691 } 1692 1693#if 0 1694 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1695 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1696 // that total to a sum == notificationFrames. 1697 if (0 < misalignment && misalignment <= mRemainingFrames) { 1698 mRemainingFrames = misalignment; 1699 return (mRemainingFrames * 1100000000LL) / sampleRate; 1700 } 1701#endif 1702 1703 } 1704 mRemainingFrames = notificationFrames; 1705 mRetryOnPartialBuffer = true; 1706 1707 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1708 return 0; 1709} 1710 1711status_t AudioTrack::restoreTrack_l(const char *from) 1712{ 1713 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1714 isOffloaded_l() ? "Offloaded" : "PCM", from); 1715 ++mSequence; 1716 status_t result; 1717 1718 // refresh the audio configuration cache in this process to make sure we get new 1719 // output parameters in createTrack_l() 1720 AudioSystem::clearAudioConfigCache(); 1721 1722 if (isOffloaded_l()) { 1723 // FIXME re-creation of offloaded tracks is not yet implemented 1724 return DEAD_OBJECT; 1725 } 1726 1727 // if the new IAudioTrack is created, createTrack_l() will modify the 1728 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1729 // It will also delete the strong references on previous IAudioTrack and IMemory 1730 1731 // take the frames that will be lost by track recreation into account in saved position 1732 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1733 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1734 result = createTrack_l(position /*epoch*/); 1735 1736 if (result == NO_ERROR) { 1737 // continue playback from last known position, but 1738 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1739 if (mStaticProxy != NULL) { 1740 mLoopPeriod = 0; 1741 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1742 } 1743 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1744 // track destruction have been played? This is critical for SoundPool implementation 1745 // This must be broken, and needs to be tested/debugged. 1746#if 0 1747 // restore write index and set other indexes to reflect empty buffer status 1748 if (!strcmp(from, "start")) { 1749 // Make sure that a client relying on callback events indicating underrun or 1750 // the actual amount of audio frames played (e.g SoundPool) receives them. 1751 if (mSharedBuffer == 0) { 1752 // restart playback even if buffer is not completely filled. 1753 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1754 } 1755 } 1756#endif 1757 if (mState == STATE_ACTIVE) { 1758 result = mAudioTrack->start(); 1759 } 1760 } 1761 if (result != NO_ERROR) { 1762 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1763#if 0 // FIXME This should no longer be needed 1764 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1765 // As getOutput was called above and resulted in an output stream to be opened, 1766 // we need to release it. 1767 if (mOutput != 0) { 1768 AudioSystem::releaseOutput(mOutput); 1769 mOutput = 0; 1770 } 1771#endif 1772 ALOGW("restoreTrack_l() failed status %d", result); 1773 mState = STATE_STOPPED; 1774 } 1775 1776 return result; 1777} 1778 1779status_t AudioTrack::setParameters(const String8& keyValuePairs) 1780{ 1781 AutoMutex lock(mLock); 1782 return mAudioTrack->setParameters(keyValuePairs); 1783} 1784 1785status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1786{ 1787 AutoMutex lock(mLock); 1788 // FIXME not implemented for fast tracks; should use proxy and SSQ 1789 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1790 return INVALID_OPERATION; 1791 } 1792 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1793 return INVALID_OPERATION; 1794 } 1795 status_t status = mAudioTrack->getTimestamp(timestamp); 1796 if (status == NO_ERROR) { 1797 timestamp.mPosition += mProxy->getEpoch(); 1798 } 1799 return status; 1800} 1801 1802String8 AudioTrack::getParameters(const String8& keys) 1803{ 1804 audio_io_handle_t output = getOutput(); 1805 if (output != 0) { 1806 return AudioSystem::getParameters(output, keys); 1807 } else { 1808 return String8::empty(); 1809 } 1810} 1811 1812bool AudioTrack::isOffloaded() const 1813{ 1814 AutoMutex lock(mLock); 1815 return isOffloaded_l(); 1816} 1817 1818status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1819{ 1820 1821 const size_t SIZE = 256; 1822 char buffer[SIZE]; 1823 String8 result; 1824 1825 result.append(" AudioTrack::dump\n"); 1826 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1827 mVolume[0], mVolume[1]); 1828 result.append(buffer); 1829 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1830 mChannelCount, mFrameCount); 1831 result.append(buffer); 1832 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1833 result.append(buffer); 1834 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1835 result.append(buffer); 1836 ::write(fd, result.string(), result.size()); 1837 return NO_ERROR; 1838} 1839 1840uint32_t AudioTrack::getUnderrunFrames() const 1841{ 1842 AutoMutex lock(mLock); 1843 return mProxy->getUnderrunFrames(); 1844} 1845 1846// ========================================================================= 1847 1848void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1849{ 1850 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1851 if (audioTrack != 0) { 1852 AutoMutex lock(audioTrack->mLock); 1853 audioTrack->mProxy->binderDied(); 1854 } 1855} 1856 1857// ========================================================================= 1858 1859AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1860 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1861 mIgnoreNextPausedInt(false) 1862{ 1863} 1864 1865AudioTrack::AudioTrackThread::~AudioTrackThread() 1866{ 1867} 1868 1869bool AudioTrack::AudioTrackThread::threadLoop() 1870{ 1871 { 1872 AutoMutex _l(mMyLock); 1873 if (mPaused) { 1874 mMyCond.wait(mMyLock); 1875 // caller will check for exitPending() 1876 return true; 1877 } 1878 if (mIgnoreNextPausedInt) { 1879 mIgnoreNextPausedInt = false; 1880 mPausedInt = false; 1881 } 1882 if (mPausedInt) { 1883 if (mPausedNs > 0) { 1884 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1885 } else { 1886 mMyCond.wait(mMyLock); 1887 } 1888 mPausedInt = false; 1889 return true; 1890 } 1891 } 1892 nsecs_t ns = mReceiver.processAudioBuffer(); 1893 switch (ns) { 1894 case 0: 1895 return true; 1896 case NS_INACTIVE: 1897 pauseInternal(); 1898 return true; 1899 case NS_NEVER: 1900 return false; 1901 case NS_WHENEVER: 1902 // FIXME increase poll interval, or make event-driven 1903 ns = 1000000000LL; 1904 // fall through 1905 default: 1906 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1907 pauseInternal(ns); 1908 return true; 1909 } 1910} 1911 1912void AudioTrack::AudioTrackThread::requestExit() 1913{ 1914 // must be in this order to avoid a race condition 1915 Thread::requestExit(); 1916 resume(); 1917} 1918 1919void AudioTrack::AudioTrackThread::pause() 1920{ 1921 AutoMutex _l(mMyLock); 1922 mPaused = true; 1923} 1924 1925void AudioTrack::AudioTrackThread::resume() 1926{ 1927 AutoMutex _l(mMyLock); 1928 mIgnoreNextPausedInt = true; 1929 if (mPaused || mPausedInt) { 1930 mPaused = false; 1931 mPausedInt = false; 1932 mMyCond.signal(); 1933 } 1934} 1935 1936void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1937{ 1938 AutoMutex _l(mMyLock); 1939 mPausedInt = true; 1940 mPausedNs = ns; 1941} 1942 1943}; // namespace android 1944