AudioTrack.cpp revision e3247bf8dd4f8fa8dfa3a108260241ae4a967569
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 int frameCount, 112 audio_output_flags_t flags, 113 callback_t cbf, 114 void* user, 115 int 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 int 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 int frameCountInt, 186 audio_output_flags_t flags, 187 callback_t cbf, 188 void* user, 189 int 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 %d, " 199 "flags #%x, notificationFrames %d, sessionId %d, transferType %d", 200 streamType, sampleRate, format, channelMask, frameCountInt, 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 // FIXME "int" here is legacy and will be replaced by size_t later 240 if (frameCountInt < 0) { 241 ALOGE("Invalid frame count %d", frameCountInt); 242 return BAD_VALUE; 243 } 244 size_t frameCount = frameCountInt; 245 246 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 247 sharedBuffer->size()); 248 249 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 250 251 AutoMutex lock(mLock); 252 253 // invariant that mAudioTrack != 0 is true only after set() returns successfully 254 if (mAudioTrack != 0) { 255 ALOGE("Track already in use"); 256 return INVALID_OPERATION; 257 } 258 259 // handle default values first. 260 if (streamType == AUDIO_STREAM_DEFAULT) { 261 streamType = AUDIO_STREAM_MUSIC; 262 } 263 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 264 ALOGE("Invalid stream type %d", streamType); 265 return BAD_VALUE; 266 } 267 mStreamType = streamType; 268 269 status_t status; 270 if (sampleRate == 0) { 271 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 272 if (status != NO_ERROR) { 273 ALOGE("Could not get output sample rate for stream type %d; status %d", 274 streamType, status); 275 return status; 276 } 277 } 278 mSampleRate = sampleRate; 279 280 // these below should probably come from the audioFlinger too... 281 if (format == AUDIO_FORMAT_DEFAULT) { 282 format = AUDIO_FORMAT_PCM_16_BIT; 283 } 284 285 // validate parameters 286 if (!audio_is_valid_format(format)) { 287 ALOGE("Invalid format %#x", format); 288 return BAD_VALUE; 289 } 290 mFormat = format; 291 292 if (!audio_is_output_channel(channelMask)) { 293 ALOGE("Invalid channel mask %#x", channelMask); 294 return BAD_VALUE; 295 } 296 mChannelMask = channelMask; 297 uint32_t channelCount = popcount(channelMask); 298 mChannelCount = channelCount; 299 300 // AudioFlinger does not currently support 8-bit data in shared memory 301 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 302 ALOGE("8-bit data in shared memory is not supported"); 303 return BAD_VALUE; 304 } 305 306 // force direct flag if format is not linear PCM 307 // or offload was requested 308 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 309 || !audio_is_linear_pcm(format)) { 310 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 311 ? "Offload request, forcing to Direct Output" 312 : "Not linear PCM, forcing to Direct Output"); 313 flags = (audio_output_flags_t) 314 // FIXME why can't we allow direct AND fast? 315 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 316 } 317 // only allow deep buffering for music stream type 318 if (streamType != AUDIO_STREAM_MUSIC) { 319 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 320 } 321 322 if (audio_is_linear_pcm(format)) { 323 mFrameSize = channelCount * audio_bytes_per_sample(format); 324 mFrameSizeAF = channelCount * sizeof(int16_t); 325 } else { 326 mFrameSize = sizeof(uint8_t); 327 mFrameSizeAF = sizeof(uint8_t); 328 } 329 330 // Make copy of input parameter offloadInfo so that in the future: 331 // (a) createTrack_l doesn't need it as an input parameter 332 // (b) we can support re-creation of offloaded tracks 333 if (offloadInfo != NULL) { 334 mOffloadInfoCopy = *offloadInfo; 335 mOffloadInfo = &mOffloadInfoCopy; 336 } else { 337 mOffloadInfo = NULL; 338 } 339 340 mVolume[LEFT] = 1.0f; 341 mVolume[RIGHT] = 1.0f; 342 mSendLevel = 0.0f; 343 // mFrameCount is initialized in createTrack_l 344 mReqFrameCount = frameCount; 345 mNotificationFramesReq = notificationFrames; 346 mNotificationFramesAct = 0; 347 mSessionId = sessionId; 348 int callingpid = IPCThreadState::self()->getCallingPid(); 349 int mypid = getpid(); 350 if (uid == -1 || (callingpid != mypid)) { 351 mClientUid = IPCThreadState::self()->getCallingUid(); 352 } else { 353 mClientUid = uid; 354 } 355 if (pid == -1 || (callingpid != mypid)) { 356 mClientPid = callingpid; 357 } else { 358 mClientPid = pid; 359 } 360 mAuxEffectId = 0; 361 mFlags = flags; 362 mCbf = cbf; 363 364 if (cbf != NULL) { 365 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 366 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 367 } 368 369 // create the IAudioTrack 370 status = createTrack_l(0 /*epoch*/); 371 372 if (status != NO_ERROR) { 373 if (mAudioTrackThread != 0) { 374 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 375 mAudioTrackThread->requestExitAndWait(); 376 mAudioTrackThread.clear(); 377 } 378 // Use of direct and offloaded output streams is ref counted by audio policy manager. 379#if 0 // FIXME This should no longer be needed 380 //Use of direct and offloaded output streams is ref counted by audio policy manager. 381 // As getOutput was called above and resulted in an output stream to be opened, 382 // we need to release it. 383 if (mOutput != 0) { 384 AudioSystem::releaseOutput(mOutput); 385 mOutput = 0; 386 } 387#endif 388 return status; 389 } 390 391 mStatus = NO_ERROR; 392 mState = STATE_STOPPED; 393 mUserData = user; 394 mLoopPeriod = 0; 395 mMarkerPosition = 0; 396 mMarkerReached = false; 397 mNewPosition = 0; 398 mUpdatePeriod = 0; 399 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 400 mSequence = 1; 401 mObservedSequence = mSequence; 402 mInUnderrun = false; 403 404 return NO_ERROR; 405} 406 407// ------------------------------------------------------------------------- 408 409status_t AudioTrack::start() 410{ 411 AutoMutex lock(mLock); 412 413 if (mState == STATE_ACTIVE) { 414 return INVALID_OPERATION; 415 } 416 417 mInUnderrun = true; 418 419 State previousState = mState; 420 if (previousState == STATE_PAUSED_STOPPING) { 421 mState = STATE_STOPPING; 422 } else { 423 mState = STATE_ACTIVE; 424 } 425 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 426 // reset current position as seen by client to 0 427 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 428 // force refresh of remaining frames by processAudioBuffer() as last 429 // write before stop could be partial. 430 mRefreshRemaining = true; 431 } 432 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 433 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 434 435 sp<AudioTrackThread> t = mAudioTrackThread; 436 if (t != 0) { 437 if (previousState == STATE_STOPPING) { 438 mProxy->interrupt(); 439 } else { 440 t->resume(); 441 } 442 } else { 443 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 444 get_sched_policy(0, &mPreviousSchedulingGroup); 445 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 446 } 447 448 status_t status = NO_ERROR; 449 if (!(flags & CBLK_INVALID)) { 450 status = mAudioTrack->start(); 451 if (status == DEAD_OBJECT) { 452 flags |= CBLK_INVALID; 453 } 454 } 455 if (flags & CBLK_INVALID) { 456 status = restoreTrack_l("start"); 457 } 458 459 if (status != NO_ERROR) { 460 ALOGE("start() status %d", status); 461 mState = previousState; 462 if (t != 0) { 463 if (previousState != STATE_STOPPING) { 464 t->pause(); 465 } 466 } else { 467 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 468 set_sched_policy(0, mPreviousSchedulingGroup); 469 } 470 } 471 472 return status; 473} 474 475void AudioTrack::stop() 476{ 477 AutoMutex lock(mLock); 478 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 479 return; 480 } 481 482 if (isOffloaded_l()) { 483 mState = STATE_STOPPING; 484 } else { 485 mState = STATE_STOPPED; 486 } 487 488 mProxy->interrupt(); 489 mAudioTrack->stop(); 490 // the playback head position will reset to 0, so if a marker is set, we need 491 // to activate it again 492 mMarkerReached = false; 493#if 0 494 // Force flush if a shared buffer is used otherwise audioflinger 495 // will not stop before end of buffer is reached. 496 // It may be needed to make sure that we stop playback, likely in case looping is on. 497 if (mSharedBuffer != 0) { 498 flush_l(); 499 } 500#endif 501 502 sp<AudioTrackThread> t = mAudioTrackThread; 503 if (t != 0) { 504 if (!isOffloaded_l()) { 505 t->pause(); 506 } 507 } else { 508 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 509 set_sched_policy(0, mPreviousSchedulingGroup); 510 } 511} 512 513bool AudioTrack::stopped() const 514{ 515 AutoMutex lock(mLock); 516 return mState != STATE_ACTIVE; 517} 518 519void AudioTrack::flush() 520{ 521 if (mSharedBuffer != 0) { 522 return; 523 } 524 AutoMutex lock(mLock); 525 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 526 return; 527 } 528 flush_l(); 529} 530 531void AudioTrack::flush_l() 532{ 533 ALOG_ASSERT(mState != STATE_ACTIVE); 534 535 // clear playback marker and periodic update counter 536 mMarkerPosition = 0; 537 mMarkerReached = false; 538 mUpdatePeriod = 0; 539 mRefreshRemaining = true; 540 541 mState = STATE_FLUSHED; 542 if (isOffloaded_l()) { 543 mProxy->interrupt(); 544 } 545 mProxy->flush(); 546 mAudioTrack->flush(); 547} 548 549void AudioTrack::pause() 550{ 551 AutoMutex lock(mLock); 552 if (mState == STATE_ACTIVE) { 553 mState = STATE_PAUSED; 554 } else if (mState == STATE_STOPPING) { 555 mState = STATE_PAUSED_STOPPING; 556 } else { 557 return; 558 } 559 mProxy->interrupt(); 560 mAudioTrack->pause(); 561} 562 563status_t AudioTrack::setVolume(float left, float right) 564{ 565 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 566 return BAD_VALUE; 567 } 568 569 AutoMutex lock(mLock); 570 mVolume[LEFT] = left; 571 mVolume[RIGHT] = right; 572 573 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 574 575 if (isOffloaded_l()) { 576 mAudioTrack->signal(); 577 } 578 return NO_ERROR; 579} 580 581status_t AudioTrack::setVolume(float volume) 582{ 583 return setVolume(volume, volume); 584} 585 586status_t AudioTrack::setAuxEffectSendLevel(float level) 587{ 588 if (level < 0.0f || level > 1.0f) { 589 return BAD_VALUE; 590 } 591 592 AutoMutex lock(mLock); 593 mSendLevel = level; 594 mProxy->setSendLevel(level); 595 596 return NO_ERROR; 597} 598 599void AudioTrack::getAuxEffectSendLevel(float* level) const 600{ 601 if (level != NULL) { 602 *level = mSendLevel; 603 } 604} 605 606status_t AudioTrack::setSampleRate(uint32_t rate) 607{ 608 if (mIsTimed || isOffloaded()) { 609 return INVALID_OPERATION; 610 } 611 612 uint32_t afSamplingRate; 613 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 614 return NO_INIT; 615 } 616 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 617 if (rate == 0 || rate > afSamplingRate*2 ) { 618 return BAD_VALUE; 619 } 620 621 AutoMutex lock(mLock); 622 mSampleRate = rate; 623 mProxy->setSampleRate(rate); 624 625 return NO_ERROR; 626} 627 628uint32_t AudioTrack::getSampleRate() const 629{ 630 if (mIsTimed) { 631 return 0; 632 } 633 634 AutoMutex lock(mLock); 635 636 // sample rate can be updated during playback by the offloaded decoder so we need to 637 // query the HAL and update if needed. 638// FIXME use Proxy return channel to update the rate from server and avoid polling here 639 if (isOffloaded_l()) { 640 if (mOutput != 0) { 641 uint32_t sampleRate = 0; 642 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 643 if (status == NO_ERROR) { 644 mSampleRate = sampleRate; 645 } 646 } 647 } 648 return mSampleRate; 649} 650 651status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 652{ 653 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 654 return INVALID_OPERATION; 655 } 656 657 if (loopCount == 0) { 658 ; 659 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 660 loopEnd - loopStart >= MIN_LOOP) { 661 ; 662 } else { 663 return BAD_VALUE; 664 } 665 666 AutoMutex lock(mLock); 667 // See setPosition() regarding setting parameters such as loop points or position while active 668 if (mState == STATE_ACTIVE) { 669 return INVALID_OPERATION; 670 } 671 setLoop_l(loopStart, loopEnd, loopCount); 672 return NO_ERROR; 673} 674 675void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 676{ 677 // FIXME If setting a loop also sets position to start of loop, then 678 // this is correct. Otherwise it should be removed. 679 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 680 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 681 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 682} 683 684status_t AudioTrack::setMarkerPosition(uint32_t marker) 685{ 686 // The only purpose of setting marker position is to get a callback 687 if (mCbf == NULL || isOffloaded()) { 688 return INVALID_OPERATION; 689 } 690 691 AutoMutex lock(mLock); 692 mMarkerPosition = marker; 693 mMarkerReached = false; 694 695 return NO_ERROR; 696} 697 698status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 699{ 700 if (isOffloaded()) { 701 return INVALID_OPERATION; 702 } 703 if (marker == NULL) { 704 return BAD_VALUE; 705 } 706 707 AutoMutex lock(mLock); 708 *marker = mMarkerPosition; 709 710 return NO_ERROR; 711} 712 713status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 714{ 715 // The only purpose of setting position update period is to get a callback 716 if (mCbf == NULL || isOffloaded()) { 717 return INVALID_OPERATION; 718 } 719 720 AutoMutex lock(mLock); 721 mNewPosition = mProxy->getPosition() + updatePeriod; 722 mUpdatePeriod = updatePeriod; 723 724 return NO_ERROR; 725} 726 727status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 728{ 729 if (isOffloaded()) { 730 return INVALID_OPERATION; 731 } 732 if (updatePeriod == NULL) { 733 return BAD_VALUE; 734 } 735 736 AutoMutex lock(mLock); 737 *updatePeriod = mUpdatePeriod; 738 739 return NO_ERROR; 740} 741 742status_t AudioTrack::setPosition(uint32_t position) 743{ 744 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 745 return INVALID_OPERATION; 746 } 747 if (position > mFrameCount) { 748 return BAD_VALUE; 749 } 750 751 AutoMutex lock(mLock); 752 // Currently we require that the player is inactive before setting parameters such as position 753 // or loop points. Otherwise, there could be a race condition: the application could read the 754 // current position, compute a new position or loop parameters, and then set that position or 755 // loop parameters but it would do the "wrong" thing since the position has continued to advance 756 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 757 // to specify how it wants to handle such scenarios. 758 if (mState == STATE_ACTIVE) { 759 return INVALID_OPERATION; 760 } 761 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 762 mLoopPeriod = 0; 763 // FIXME Check whether loops and setting position are incompatible in old code. 764 // If we use setLoop for both purposes we lose the capability to set the position while looping. 765 mStaticProxy->setLoop(position, mFrameCount, 0); 766 767 return NO_ERROR; 768} 769 770status_t AudioTrack::getPosition(uint32_t *position) const 771{ 772 if (position == NULL) { 773 return BAD_VALUE; 774 } 775 776 AutoMutex lock(mLock); 777 if (isOffloaded_l()) { 778 uint32_t dspFrames = 0; 779 780 if (mOutput != 0) { 781 uint32_t halFrames; 782 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 783 } 784 *position = dspFrames; 785 } else { 786 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 787 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 788 mProxy->getPosition(); 789 } 790 return NO_ERROR; 791} 792 793status_t AudioTrack::getBufferPosition(uint32_t *position) 794{ 795 if (mSharedBuffer == 0 || mIsTimed) { 796 return INVALID_OPERATION; 797 } 798 if (position == NULL) { 799 return BAD_VALUE; 800 } 801 802 AutoMutex lock(mLock); 803 *position = mStaticProxy->getBufferPosition(); 804 return NO_ERROR; 805} 806 807status_t AudioTrack::reload() 808{ 809 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 810 return INVALID_OPERATION; 811 } 812 813 AutoMutex lock(mLock); 814 // See setPosition() regarding setting parameters such as loop points or position while active 815 if (mState == STATE_ACTIVE) { 816 return INVALID_OPERATION; 817 } 818 mNewPosition = mUpdatePeriod; 819 mLoopPeriod = 0; 820 // FIXME The new code cannot reload while keeping a loop specified. 821 // Need to check how the old code handled this, and whether it's a significant change. 822 mStaticProxy->setLoop(0, mFrameCount, 0); 823 return NO_ERROR; 824} 825 826audio_io_handle_t AudioTrack::getOutput() const 827{ 828 AutoMutex lock(mLock); 829 return mOutput; 830} 831 832status_t AudioTrack::attachAuxEffect(int effectId) 833{ 834 AutoMutex lock(mLock); 835 status_t status = mAudioTrack->attachAuxEffect(effectId); 836 if (status == NO_ERROR) { 837 mAuxEffectId = effectId; 838 } 839 return status; 840} 841 842// ------------------------------------------------------------------------- 843 844// must be called with mLock held 845status_t AudioTrack::createTrack_l(size_t epoch) 846{ 847 status_t status; 848 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 849 if (audioFlinger == 0) { 850 ALOGE("Could not get audioflinger"); 851 return NO_INIT; 852 } 853 854 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 855 mChannelMask, mFlags, mOffloadInfo); 856 if (output == 0) { 857 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 858 "channel mask %#x, flags %#x", 859 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 860 return BAD_VALUE; 861 } 862 { 863 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 864 // we must release it ourselves if anything goes wrong. 865 866 // Not all of these values are needed under all conditions, but it is easier to get them all 867 868 uint32_t afLatency; 869 status = AudioSystem::getLatency(output, mStreamType, &afLatency); 870 if (status != NO_ERROR) { 871 ALOGE("getLatency(%d) failed status %d", output, status); 872 goto release; 873 } 874 875 size_t afFrameCount; 876 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 877 if (status != NO_ERROR) { 878 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 879 goto release; 880 } 881 882 uint32_t afSampleRate; 883 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 884 if (status != NO_ERROR) { 885 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 886 goto release; 887 } 888 889 // Client decides whether the track is TIMED (see below), but can only express a preference 890 // for FAST. Server will perform additional tests. 891 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 892 // either of these use cases: 893 // use case 1: shared buffer 894 (mSharedBuffer != 0) || 895 // use case 2: callback handler 896 (mCbf != NULL)) && 897 // matching sample rate 898 (mSampleRate == afSampleRate))) { 899 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 900 // once denied, do not request again if IAudioTrack is re-created 901 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 902 } 903 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 904 905 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 906 // n = 1 fast track with single buffering; nBuffering is ignored 907 // n = 2 fast track with double buffering 908 // n = 2 normal track, no sample rate conversion 909 // n = 3 normal track, with sample rate conversion 910 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 911 // n > 3 very high latency or very small notification interval; nBuffering is ignored 912 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 913 914 mNotificationFramesAct = mNotificationFramesReq; 915 916 size_t frameCount = mReqFrameCount; 917 if (!audio_is_linear_pcm(mFormat)) { 918 919 if (mSharedBuffer != 0) { 920 // Same comment as below about ignoring frameCount parameter for set() 921 frameCount = mSharedBuffer->size(); 922 } else if (frameCount == 0) { 923 frameCount = afFrameCount; 924 } 925 if (mNotificationFramesAct != frameCount) { 926 mNotificationFramesAct = frameCount; 927 } 928 } else if (mSharedBuffer != 0) { 929 930 // Ensure that buffer alignment matches channel count 931 // 8-bit data in shared memory is not currently supported by AudioFlinger 932 size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 933 if (mChannelCount > 1) { 934 // More than 2 channels does not require stronger alignment than stereo 935 alignment <<= 1; 936 } 937 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 938 ALOGE("Invalid buffer alignment: address %p, channel count %u", 939 mSharedBuffer->pointer(), mChannelCount); 940 status = BAD_VALUE; 941 goto release; 942 } 943 944 // When initializing a shared buffer AudioTrack via constructors, 945 // there's no frameCount parameter. 946 // But when initializing a shared buffer AudioTrack via set(), 947 // there _is_ a frameCount parameter. We silently ignore it. 948 frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); 949 950 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 951 952 // FIXME move these calculations and associated checks to server 953 954 // Ensure that buffer depth covers at least audio hardware latency 955 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 956 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 957 afFrameCount, minBufCount, afSampleRate, afLatency); 958 if (minBufCount <= nBuffering) { 959 minBufCount = nBuffering; 960 } 961 962 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 963 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 964 ", afLatency=%d", 965 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 966 967 if (frameCount == 0) { 968 frameCount = minFrameCount; 969 } else if (frameCount < minFrameCount) { 970 // not ALOGW because it happens all the time when playing key clicks over A2DP 971 ALOGV("Minimum buffer size corrected from %d to %d", 972 frameCount, minFrameCount); 973 frameCount = minFrameCount; 974 } 975 // Make sure that application is notified with sufficient margin before underrun 976 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 977 mNotificationFramesAct = frameCount/nBuffering; 978 } 979 980 } else { 981 // For fast tracks, the frame count calculations and checks are done by server 982 } 983 984 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 985 if (mIsTimed) { 986 trackFlags |= IAudioFlinger::TRACK_TIMED; 987 } 988 989 pid_t tid = -1; 990 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 991 trackFlags |= IAudioFlinger::TRACK_FAST; 992 if (mAudioTrackThread != 0) { 993 tid = mAudioTrackThread->getTid(); 994 } 995 } 996 997 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 998 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 999 } 1000 1001 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1002 // but we will still need the original value also 1003 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1004 mSampleRate, 1005 // AudioFlinger only sees 16-bit PCM 1006 mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1007 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1008 mChannelMask, 1009 &temp, 1010 &trackFlags, 1011 mSharedBuffer, 1012 output, 1013 tid, 1014 &mSessionId, 1015 mName, 1016 mClientUid, 1017 &status); 1018 1019 if (track == 0) { 1020 ALOGE("AudioFlinger could not create track, status: %d", status); 1021 goto release; 1022 } 1023 // AudioFlinger now owns the reference to the I/O handle, 1024 // so we are no longer responsible for releasing it. 1025 1026 sp<IMemory> iMem = track->getCblk(); 1027 if (iMem == 0) { 1028 ALOGE("Could not get control block"); 1029 return NO_INIT; 1030 } 1031 void *iMemPointer = iMem->pointer(); 1032 if (iMemPointer == NULL) { 1033 ALOGE("Could not get control block pointer"); 1034 return NO_INIT; 1035 } 1036 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1037 if (mAudioTrack != 0) { 1038 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1039 mDeathNotifier.clear(); 1040 } 1041 mAudioTrack = track; 1042 mCblkMemory = iMem; 1043 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1044 mCblk = cblk; 1045 // note that temp is the (possibly revised) value of frameCount 1046 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1047 // In current design, AudioTrack client checks and ensures frame count validity before 1048 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1049 // for fast track as it uses a special method of assigning frame count. 1050 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1051 } 1052 frameCount = temp; 1053 mAwaitBoost = false; 1054 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1055 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1056 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1057 mAwaitBoost = true; 1058 if (mSharedBuffer == 0) { 1059 // Theoretically double-buffering is not required for fast tracks, 1060 // due to tighter scheduling. But in practice, to accommodate kernels with 1061 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1062 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1063 mNotificationFramesAct = frameCount/nBuffering; 1064 } 1065 } 1066 } else { 1067 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1068 // once denied, do not request again if IAudioTrack is re-created 1069 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1070 if (mSharedBuffer == 0) { 1071 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1072 mNotificationFramesAct = frameCount/nBuffering; 1073 } 1074 } 1075 } 1076 } 1077 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1078 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1079 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1080 } else { 1081 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1082 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1083 // FIXME This is a warning, not an error, so don't return error status 1084 //return NO_INIT; 1085 } 1086 } 1087 1088 // We retain a copy of the I/O handle, but don't own the reference 1089 mOutput = output; 1090 mRefreshRemaining = true; 1091 1092 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1093 // is the value of pointer() for the shared buffer, otherwise buffers points 1094 // immediately after the control block. This address is for the mapping within client 1095 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1096 void* buffers; 1097 if (mSharedBuffer == 0) { 1098 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1099 } else { 1100 buffers = mSharedBuffer->pointer(); 1101 } 1102 1103 mAudioTrack->attachAuxEffect(mAuxEffectId); 1104 // FIXME don't believe this lie 1105 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1106 mFrameCount = frameCount; 1107 // If IAudioTrack is re-created, don't let the requested frameCount 1108 // decrease. This can confuse clients that cache frameCount(). 1109 if (frameCount > mReqFrameCount) { 1110 mReqFrameCount = frameCount; 1111 } 1112 1113 // update proxy 1114 if (mSharedBuffer == 0) { 1115 mStaticProxy.clear(); 1116 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1117 } else { 1118 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1119 mProxy = mStaticProxy; 1120 } 1121 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1122 uint16_t(mVolume[LEFT] * 0x1000)); 1123 mProxy->setSendLevel(mSendLevel); 1124 mProxy->setSampleRate(mSampleRate); 1125 mProxy->setEpoch(epoch); 1126 mProxy->setMinimum(mNotificationFramesAct); 1127 1128 mDeathNotifier = new DeathNotifier(this); 1129 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1130 1131 return NO_ERROR; 1132 } 1133 1134release: 1135 AudioSystem::releaseOutput(output); 1136 if (status == NO_ERROR) { 1137 status = NO_INIT; 1138 } 1139 return status; 1140} 1141 1142status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1143{ 1144 if (audioBuffer == NULL) { 1145 return BAD_VALUE; 1146 } 1147 if (mTransfer != TRANSFER_OBTAIN) { 1148 audioBuffer->frameCount = 0; 1149 audioBuffer->size = 0; 1150 audioBuffer->raw = NULL; 1151 return INVALID_OPERATION; 1152 } 1153 1154 const struct timespec *requested; 1155 struct timespec timeout; 1156 if (waitCount == -1) { 1157 requested = &ClientProxy::kForever; 1158 } else if (waitCount == 0) { 1159 requested = &ClientProxy::kNonBlocking; 1160 } else if (waitCount > 0) { 1161 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1162 timeout.tv_sec = ms / 1000; 1163 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1164 requested = &timeout; 1165 } else { 1166 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1167 requested = NULL; 1168 } 1169 return obtainBuffer(audioBuffer, requested); 1170} 1171 1172status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1173 struct timespec *elapsed, size_t *nonContig) 1174{ 1175 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1176 uint32_t oldSequence = 0; 1177 uint32_t newSequence; 1178 1179 Proxy::Buffer buffer; 1180 status_t status = NO_ERROR; 1181 1182 static const int32_t kMaxTries = 5; 1183 int32_t tryCounter = kMaxTries; 1184 1185 do { 1186 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1187 // keep them from going away if another thread re-creates the track during obtainBuffer() 1188 sp<AudioTrackClientProxy> proxy; 1189 sp<IMemory> iMem; 1190 1191 { // start of lock scope 1192 AutoMutex lock(mLock); 1193 1194 newSequence = mSequence; 1195 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1196 if (status == DEAD_OBJECT) { 1197 // re-create track, unless someone else has already done so 1198 if (newSequence == oldSequence) { 1199 status = restoreTrack_l("obtainBuffer"); 1200 if (status != NO_ERROR) { 1201 buffer.mFrameCount = 0; 1202 buffer.mRaw = NULL; 1203 buffer.mNonContig = 0; 1204 break; 1205 } 1206 } 1207 } 1208 oldSequence = newSequence; 1209 1210 // Keep the extra references 1211 proxy = mProxy; 1212 iMem = mCblkMemory; 1213 1214 if (mState == STATE_STOPPING) { 1215 status = -EINTR; 1216 buffer.mFrameCount = 0; 1217 buffer.mRaw = NULL; 1218 buffer.mNonContig = 0; 1219 break; 1220 } 1221 1222 // Non-blocking if track is stopped or paused 1223 if (mState != STATE_ACTIVE) { 1224 requested = &ClientProxy::kNonBlocking; 1225 } 1226 1227 } // end of lock scope 1228 1229 buffer.mFrameCount = audioBuffer->frameCount; 1230 // FIXME starts the requested timeout and elapsed over from scratch 1231 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1232 1233 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1234 1235 audioBuffer->frameCount = buffer.mFrameCount; 1236 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1237 audioBuffer->raw = buffer.mRaw; 1238 if (nonContig != NULL) { 1239 *nonContig = buffer.mNonContig; 1240 } 1241 return status; 1242} 1243 1244void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1245{ 1246 if (mTransfer == TRANSFER_SHARED) { 1247 return; 1248 } 1249 1250 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1251 if (stepCount == 0) { 1252 return; 1253 } 1254 1255 Proxy::Buffer buffer; 1256 buffer.mFrameCount = stepCount; 1257 buffer.mRaw = audioBuffer->raw; 1258 1259 AutoMutex lock(mLock); 1260 mInUnderrun = false; 1261 mProxy->releaseBuffer(&buffer); 1262 1263 // restart track if it was disabled by audioflinger due to previous underrun 1264 if (mState == STATE_ACTIVE) { 1265 audio_track_cblk_t* cblk = mCblk; 1266 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1267 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1268 this, mName.string()); 1269 // FIXME ignoring status 1270 mAudioTrack->start(); 1271 } 1272 } 1273} 1274 1275// ------------------------------------------------------------------------- 1276 1277ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1278{ 1279 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1280 return INVALID_OPERATION; 1281 } 1282 1283 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1284 // Sanity-check: user is most-likely passing an error code, and it would 1285 // make the return value ambiguous (actualSize vs error). 1286 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1287 return BAD_VALUE; 1288 } 1289 1290 size_t written = 0; 1291 Buffer audioBuffer; 1292 1293 while (userSize >= mFrameSize) { 1294 audioBuffer.frameCount = userSize / mFrameSize; 1295 1296 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1297 if (err < 0) { 1298 if (written > 0) { 1299 break; 1300 } 1301 return ssize_t(err); 1302 } 1303 1304 size_t toWrite; 1305 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1306 // Divide capacity by 2 to take expansion into account 1307 toWrite = audioBuffer.size >> 1; 1308 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1309 } else { 1310 toWrite = audioBuffer.size; 1311 memcpy(audioBuffer.i8, buffer, toWrite); 1312 } 1313 buffer = ((const char *) buffer) + toWrite; 1314 userSize -= toWrite; 1315 written += toWrite; 1316 1317 releaseBuffer(&audioBuffer); 1318 } 1319 1320 return written; 1321} 1322 1323// ------------------------------------------------------------------------- 1324 1325TimedAudioTrack::TimedAudioTrack() { 1326 mIsTimed = true; 1327} 1328 1329status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1330{ 1331 AutoMutex lock(mLock); 1332 status_t result = UNKNOWN_ERROR; 1333 1334#if 1 1335 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1336 // while we are accessing the cblk 1337 sp<IAudioTrack> audioTrack = mAudioTrack; 1338 sp<IMemory> iMem = mCblkMemory; 1339#endif 1340 1341 // If the track is not invalid already, try to allocate a buffer. alloc 1342 // fails indicating that the server is dead, flag the track as invalid so 1343 // we can attempt to restore in just a bit. 1344 audio_track_cblk_t* cblk = mCblk; 1345 if (!(cblk->mFlags & CBLK_INVALID)) { 1346 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1347 if (result == DEAD_OBJECT) { 1348 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1349 } 1350 } 1351 1352 // If the track is invalid at this point, attempt to restore it. and try the 1353 // allocation one more time. 1354 if (cblk->mFlags & CBLK_INVALID) { 1355 result = restoreTrack_l("allocateTimedBuffer"); 1356 1357 if (result == NO_ERROR) { 1358 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1359 } 1360 } 1361 1362 return result; 1363} 1364 1365status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1366 int64_t pts) 1367{ 1368 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1369 { 1370 AutoMutex lock(mLock); 1371 audio_track_cblk_t* cblk = mCblk; 1372 // restart track if it was disabled by audioflinger due to previous underrun 1373 if (buffer->size() != 0 && status == NO_ERROR && 1374 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1375 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1376 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1377 // FIXME ignoring status 1378 mAudioTrack->start(); 1379 } 1380 } 1381 return status; 1382} 1383 1384status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1385 TargetTimeline target) 1386{ 1387 return mAudioTrack->setMediaTimeTransform(xform, target); 1388} 1389 1390// ------------------------------------------------------------------------- 1391 1392nsecs_t AudioTrack::processAudioBuffer() 1393{ 1394 // Currently the AudioTrack thread is not created if there are no callbacks. 1395 // Would it ever make sense to run the thread, even without callbacks? 1396 // If so, then replace this by checks at each use for mCbf != NULL. 1397 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1398 1399 mLock.lock(); 1400 if (mAwaitBoost) { 1401 mAwaitBoost = false; 1402 mLock.unlock(); 1403 static const int32_t kMaxTries = 5; 1404 int32_t tryCounter = kMaxTries; 1405 uint32_t pollUs = 10000; 1406 do { 1407 int policy = sched_getscheduler(0); 1408 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1409 break; 1410 } 1411 usleep(pollUs); 1412 pollUs <<= 1; 1413 } while (tryCounter-- > 0); 1414 if (tryCounter < 0) { 1415 ALOGE("did not receive expected priority boost on time"); 1416 } 1417 // Run again immediately 1418 return 0; 1419 } 1420 1421 // Can only reference mCblk while locked 1422 int32_t flags = android_atomic_and( 1423 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1424 1425 // Check for track invalidation 1426 if (flags & CBLK_INVALID) { 1427 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1428 // AudioSystem cache. We should not exit here but after calling the callback so 1429 // that the upper layers can recreate the track 1430 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1431 status_t status = restoreTrack_l("processAudioBuffer"); 1432 mLock.unlock(); 1433 // Run again immediately, but with a new IAudioTrack 1434 return 0; 1435 } 1436 } 1437 1438 bool waitStreamEnd = mState == STATE_STOPPING; 1439 bool active = mState == STATE_ACTIVE; 1440 1441 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1442 bool newUnderrun = false; 1443 if (flags & CBLK_UNDERRUN) { 1444#if 0 1445 // Currently in shared buffer mode, when the server reaches the end of buffer, 1446 // the track stays active in continuous underrun state. It's up to the application 1447 // to pause or stop the track, or set the position to a new offset within buffer. 1448 // This was some experimental code to auto-pause on underrun. Keeping it here 1449 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1450 if (mTransfer == TRANSFER_SHARED) { 1451 mState = STATE_PAUSED; 1452 active = false; 1453 } 1454#endif 1455 if (!mInUnderrun) { 1456 mInUnderrun = true; 1457 newUnderrun = true; 1458 } 1459 } 1460 1461 // Get current position of server 1462 size_t position = mProxy->getPosition(); 1463 1464 // Manage marker callback 1465 bool markerReached = false; 1466 size_t markerPosition = mMarkerPosition; 1467 // FIXME fails for wraparound, need 64 bits 1468 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1469 mMarkerReached = markerReached = true; 1470 } 1471 1472 // Determine number of new position callback(s) that will be needed, while locked 1473 size_t newPosCount = 0; 1474 size_t newPosition = mNewPosition; 1475 size_t updatePeriod = mUpdatePeriod; 1476 // FIXME fails for wraparound, need 64 bits 1477 if (updatePeriod > 0 && position >= newPosition) { 1478 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1479 mNewPosition += updatePeriod * newPosCount; 1480 } 1481 1482 // Cache other fields that will be needed soon 1483 uint32_t loopPeriod = mLoopPeriod; 1484 uint32_t sampleRate = mSampleRate; 1485 size_t notificationFrames = mNotificationFramesAct; 1486 if (mRefreshRemaining) { 1487 mRefreshRemaining = false; 1488 mRemainingFrames = notificationFrames; 1489 mRetryOnPartialBuffer = false; 1490 } 1491 size_t misalignment = mProxy->getMisalignment(); 1492 uint32_t sequence = mSequence; 1493 1494 // These fields don't need to be cached, because they are assigned only by set(): 1495 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1496 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1497 1498 mLock.unlock(); 1499 1500 if (waitStreamEnd) { 1501 AutoMutex lock(mLock); 1502 1503 sp<AudioTrackClientProxy> proxy = mProxy; 1504 sp<IMemory> iMem = mCblkMemory; 1505 1506 struct timespec timeout; 1507 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1508 timeout.tv_nsec = 0; 1509 1510 mLock.unlock(); 1511 status_t status = mProxy->waitStreamEndDone(&timeout); 1512 mLock.lock(); 1513 switch (status) { 1514 case NO_ERROR: 1515 case DEAD_OBJECT: 1516 case TIMED_OUT: 1517 mLock.unlock(); 1518 mCbf(EVENT_STREAM_END, mUserData, NULL); 1519 mLock.lock(); 1520 if (mState == STATE_STOPPING) { 1521 mState = STATE_STOPPED; 1522 if (status != DEAD_OBJECT) { 1523 return NS_INACTIVE; 1524 } 1525 } 1526 return 0; 1527 default: 1528 return 0; 1529 } 1530 } 1531 1532 // perform callbacks while unlocked 1533 if (newUnderrun) { 1534 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1535 } 1536 // FIXME we will miss loops if loop cycle was signaled several times since last call 1537 // to processAudioBuffer() 1538 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1539 mCbf(EVENT_LOOP_END, mUserData, NULL); 1540 } 1541 if (flags & CBLK_BUFFER_END) { 1542 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1543 } 1544 if (markerReached) { 1545 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1546 } 1547 while (newPosCount > 0) { 1548 size_t temp = newPosition; 1549 mCbf(EVENT_NEW_POS, mUserData, &temp); 1550 newPosition += updatePeriod; 1551 newPosCount--; 1552 } 1553 1554 if (mObservedSequence != sequence) { 1555 mObservedSequence = sequence; 1556 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1557 // for offloaded tracks, just wait for the upper layers to recreate the track 1558 if (isOffloaded()) { 1559 return NS_INACTIVE; 1560 } 1561 } 1562 1563 // if inactive, then don't run me again until re-started 1564 if (!active) { 1565 return NS_INACTIVE; 1566 } 1567 1568 // Compute the estimated time until the next timed event (position, markers, loops) 1569 // FIXME only for non-compressed audio 1570 uint32_t minFrames = ~0; 1571 if (!markerReached && position < markerPosition) { 1572 minFrames = markerPosition - position; 1573 } 1574 if (loopPeriod > 0 && loopPeriod < minFrames) { 1575 minFrames = loopPeriod; 1576 } 1577 if (updatePeriod > 0 && updatePeriod < minFrames) { 1578 minFrames = updatePeriod; 1579 } 1580 1581 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1582 static const uint32_t kPoll = 0; 1583 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1584 minFrames = kPoll * notificationFrames; 1585 } 1586 1587 // Convert frame units to time units 1588 nsecs_t ns = NS_WHENEVER; 1589 if (minFrames != (uint32_t) ~0) { 1590 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1591 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1592 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1593 } 1594 1595 // If not supplying data by EVENT_MORE_DATA, then we're done 1596 if (mTransfer != TRANSFER_CALLBACK) { 1597 return ns; 1598 } 1599 1600 struct timespec timeout; 1601 const struct timespec *requested = &ClientProxy::kForever; 1602 if (ns != NS_WHENEVER) { 1603 timeout.tv_sec = ns / 1000000000LL; 1604 timeout.tv_nsec = ns % 1000000000LL; 1605 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1606 requested = &timeout; 1607 } 1608 1609 while (mRemainingFrames > 0) { 1610 1611 Buffer audioBuffer; 1612 audioBuffer.frameCount = mRemainingFrames; 1613 size_t nonContig; 1614 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1615 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1616 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1617 requested = &ClientProxy::kNonBlocking; 1618 size_t avail = audioBuffer.frameCount + nonContig; 1619 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1620 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1621 if (err != NO_ERROR) { 1622 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1623 (isOffloaded() && (err == DEAD_OBJECT))) { 1624 return 0; 1625 } 1626 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1627 return NS_NEVER; 1628 } 1629 1630 if (mRetryOnPartialBuffer && !isOffloaded()) { 1631 mRetryOnPartialBuffer = false; 1632 if (avail < mRemainingFrames) { 1633 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1634 if (ns < 0 || myns < ns) { 1635 ns = myns; 1636 } 1637 return ns; 1638 } 1639 } 1640 1641 // Divide buffer size by 2 to take into account the expansion 1642 // due to 8 to 16 bit conversion: the callback must fill only half 1643 // of the destination buffer 1644 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1645 audioBuffer.size >>= 1; 1646 } 1647 1648 size_t reqSize = audioBuffer.size; 1649 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1650 size_t writtenSize = audioBuffer.size; 1651 1652 // Sanity check on returned size 1653 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1654 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1655 reqSize, (int) writtenSize); 1656 return NS_NEVER; 1657 } 1658 1659 if (writtenSize == 0) { 1660 // The callback is done filling buffers 1661 // Keep this thread going to handle timed events and 1662 // still try to get more data in intervals of WAIT_PERIOD_MS 1663 // but don't just loop and block the CPU, so wait 1664 return WAIT_PERIOD_MS * 1000000LL; 1665 } 1666 1667 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1668 // 8 to 16 bit conversion, note that source and destination are the same address 1669 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1670 audioBuffer.size <<= 1; 1671 } 1672 1673 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1674 audioBuffer.frameCount = releasedFrames; 1675 mRemainingFrames -= releasedFrames; 1676 if (misalignment >= releasedFrames) { 1677 misalignment -= releasedFrames; 1678 } else { 1679 misalignment = 0; 1680 } 1681 1682 releaseBuffer(&audioBuffer); 1683 1684 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1685 // if callback doesn't like to accept the full chunk 1686 if (writtenSize < reqSize) { 1687 continue; 1688 } 1689 1690 // There could be enough non-contiguous frames available to satisfy the remaining request 1691 if (mRemainingFrames <= nonContig) { 1692 continue; 1693 } 1694 1695#if 0 1696 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1697 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1698 // that total to a sum == notificationFrames. 1699 if (0 < misalignment && misalignment <= mRemainingFrames) { 1700 mRemainingFrames = misalignment; 1701 return (mRemainingFrames * 1100000000LL) / sampleRate; 1702 } 1703#endif 1704 1705 } 1706 mRemainingFrames = notificationFrames; 1707 mRetryOnPartialBuffer = true; 1708 1709 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1710 return 0; 1711} 1712 1713status_t AudioTrack::restoreTrack_l(const char *from) 1714{ 1715 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1716 isOffloaded_l() ? "Offloaded" : "PCM", from); 1717 ++mSequence; 1718 status_t result; 1719 1720 // refresh the audio configuration cache in this process to make sure we get new 1721 // output parameters in createTrack_l() 1722 AudioSystem::clearAudioConfigCache(); 1723 1724 if (isOffloaded_l()) { 1725 // FIXME re-creation of offloaded tracks is not yet implemented 1726 return DEAD_OBJECT; 1727 } 1728 1729 // if the new IAudioTrack is created, createTrack_l() will modify the 1730 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1731 // It will also delete the strong references on previous IAudioTrack and IMemory 1732 1733 // take the frames that will be lost by track recreation into account in saved position 1734 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1735 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1736 result = createTrack_l(position /*epoch*/); 1737 1738 if (result == NO_ERROR) { 1739 // continue playback from last known position, but 1740 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1741 if (mStaticProxy != NULL) { 1742 mLoopPeriod = 0; 1743 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1744 } 1745 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1746 // track destruction have been played? This is critical for SoundPool implementation 1747 // This must be broken, and needs to be tested/debugged. 1748#if 0 1749 // restore write index and set other indexes to reflect empty buffer status 1750 if (!strcmp(from, "start")) { 1751 // Make sure that a client relying on callback events indicating underrun or 1752 // the actual amount of audio frames played (e.g SoundPool) receives them. 1753 if (mSharedBuffer == 0) { 1754 // restart playback even if buffer is not completely filled. 1755 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1756 } 1757 } 1758#endif 1759 if (mState == STATE_ACTIVE) { 1760 result = mAudioTrack->start(); 1761 } 1762 } 1763 if (result != NO_ERROR) { 1764 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1765#if 0 // FIXME This should no longer be needed 1766 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1767 // As getOutput was called above and resulted in an output stream to be opened, 1768 // we need to release it. 1769 if (mOutput != 0) { 1770 AudioSystem::releaseOutput(mOutput); 1771 mOutput = 0; 1772 } 1773#endif 1774 ALOGW("restoreTrack_l() failed status %d", result); 1775 mState = STATE_STOPPED; 1776 } 1777 1778 return result; 1779} 1780 1781status_t AudioTrack::setParameters(const String8& keyValuePairs) 1782{ 1783 AutoMutex lock(mLock); 1784 return mAudioTrack->setParameters(keyValuePairs); 1785} 1786 1787status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1788{ 1789 AutoMutex lock(mLock); 1790 // FIXME not implemented for fast tracks; should use proxy and SSQ 1791 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1792 return INVALID_OPERATION; 1793 } 1794 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1795 return INVALID_OPERATION; 1796 } 1797 status_t status = mAudioTrack->getTimestamp(timestamp); 1798 if (status == NO_ERROR) { 1799 timestamp.mPosition += mProxy->getEpoch(); 1800 } 1801 return status; 1802} 1803 1804String8 AudioTrack::getParameters(const String8& keys) 1805{ 1806 audio_io_handle_t output = getOutput(); 1807 if (output != 0) { 1808 return AudioSystem::getParameters(output, keys); 1809 } else { 1810 return String8::empty(); 1811 } 1812} 1813 1814bool AudioTrack::isOffloaded() const 1815{ 1816 AutoMutex lock(mLock); 1817 return isOffloaded_l(); 1818} 1819 1820status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1821{ 1822 1823 const size_t SIZE = 256; 1824 char buffer[SIZE]; 1825 String8 result; 1826 1827 result.append(" AudioTrack::dump\n"); 1828 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1829 mVolume[0], mVolume[1]); 1830 result.append(buffer); 1831 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1832 mChannelCount, mFrameCount); 1833 result.append(buffer); 1834 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1835 result.append(buffer); 1836 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1837 result.append(buffer); 1838 ::write(fd, result.string(), result.size()); 1839 return NO_ERROR; 1840} 1841 1842uint32_t AudioTrack::getUnderrunFrames() const 1843{ 1844 AutoMutex lock(mLock); 1845 return mProxy->getUnderrunFrames(); 1846} 1847 1848// ========================================================================= 1849 1850void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1851{ 1852 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1853 if (audioTrack != 0) { 1854 AutoMutex lock(audioTrack->mLock); 1855 audioTrack->mProxy->binderDied(); 1856 } 1857} 1858 1859// ========================================================================= 1860 1861AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1862 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1863 mIgnoreNextPausedInt(false) 1864{ 1865} 1866 1867AudioTrack::AudioTrackThread::~AudioTrackThread() 1868{ 1869} 1870 1871bool AudioTrack::AudioTrackThread::threadLoop() 1872{ 1873 { 1874 AutoMutex _l(mMyLock); 1875 if (mPaused) { 1876 mMyCond.wait(mMyLock); 1877 // caller will check for exitPending() 1878 return true; 1879 } 1880 if (mIgnoreNextPausedInt) { 1881 mIgnoreNextPausedInt = false; 1882 mPausedInt = false; 1883 } 1884 if (mPausedInt) { 1885 if (mPausedNs > 0) { 1886 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1887 } else { 1888 mMyCond.wait(mMyLock); 1889 } 1890 mPausedInt = false; 1891 return true; 1892 } 1893 } 1894 nsecs_t ns = mReceiver.processAudioBuffer(); 1895 switch (ns) { 1896 case 0: 1897 return true; 1898 case NS_INACTIVE: 1899 pauseInternal(); 1900 return true; 1901 case NS_NEVER: 1902 return false; 1903 case NS_WHENEVER: 1904 // FIXME increase poll interval, or make event-driven 1905 ns = 1000000000LL; 1906 // fall through 1907 default: 1908 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1909 pauseInternal(ns); 1910 return true; 1911 } 1912} 1913 1914void AudioTrack::AudioTrackThread::requestExit() 1915{ 1916 // must be in this order to avoid a race condition 1917 Thread::requestExit(); 1918 resume(); 1919} 1920 1921void AudioTrack::AudioTrackThread::pause() 1922{ 1923 AutoMutex _l(mMyLock); 1924 mPaused = true; 1925} 1926 1927void AudioTrack::AudioTrackThread::resume() 1928{ 1929 AutoMutex _l(mMyLock); 1930 mIgnoreNextPausedInt = true; 1931 if (mPaused || mPausedInt) { 1932 mPaused = false; 1933 mPausedInt = false; 1934 mMyCond.signal(); 1935 } 1936} 1937 1938void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1939{ 1940 AutoMutex _l(mMyLock); 1941 mPausedInt = true; 1942 mPausedNs = ns; 1943} 1944 1945}; // namespace android 1946