AudioTrack.cpp revision b5fed68bcdd6f44424c9e4d12bfe9a3ff51bd62e
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 // default to 0 in case of error 48 *frameCount = 0; 49 50 // FIXME merge with similar code in createTrack_l(), except we're missing 51 // some information here that is available in createTrack_l(): 52 // audio_io_handle_t output 53 // audio_format_t format 54 // audio_channel_mask_t channelMask 55 // audio_output_flags_t flags 56 uint32_t afSampleRate; 57 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 58 return NO_INIT; 59 } 60 size_t afFrameCount; 61 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 62 return NO_INIT; 63 } 64 uint32_t afLatency; 65 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 66 return NO_INIT; 67 } 68 69 // Ensure that buffer depth covers at least audio hardware latency 70 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 71 if (minBufCount < 2) { 72 minBufCount = 2; 73 } 74 75 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 76 afFrameCount * minBufCount * sampleRate / afSampleRate; 77 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 78 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 79 return NO_ERROR; 80} 81 82// --------------------------------------------------------------------------- 83 84AudioTrack::AudioTrack() 85 : mStatus(NO_INIT), 86 mIsTimed(false), 87 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 88 mPreviousSchedulingGroup(SP_DEFAULT) 89{ 90} 91 92AudioTrack::AudioTrack( 93 audio_stream_type_t streamType, 94 uint32_t sampleRate, 95 audio_format_t format, 96 audio_channel_mask_t channelMask, 97 int frameCount, 98 audio_output_flags_t flags, 99 callback_t cbf, 100 void* user, 101 int notificationFrames, 102 int sessionId, 103 transfer_type transferType, 104 const audio_offload_info_t *offloadInfo) 105 : mStatus(NO_INIT), 106 mIsTimed(false), 107 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 108 mPreviousSchedulingGroup(SP_DEFAULT) 109{ 110 mStatus = set(streamType, sampleRate, format, channelMask, 111 frameCount, flags, cbf, user, notificationFrames, 112 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 113} 114 115AudioTrack::AudioTrack( 116 audio_stream_type_t streamType, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 const sp<IMemory>& sharedBuffer, 121 audio_output_flags_t flags, 122 callback_t cbf, 123 void* user, 124 int notificationFrames, 125 int sessionId, 126 transfer_type transferType, 127 const audio_offload_info_t *offloadInfo) 128 : mStatus(NO_INIT), 129 mIsTimed(false), 130 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 131 mPreviousSchedulingGroup(SP_DEFAULT) 132{ 133 mStatus = set(streamType, sampleRate, format, channelMask, 134 0 /*frameCount*/, flags, cbf, user, notificationFrames, 135 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 136} 137 138AudioTrack::~AudioTrack() 139{ 140 if (mStatus == NO_ERROR) { 141 // Make sure that callback function exits in the case where 142 // it is looping on buffer full condition in obtainBuffer(). 143 // Otherwise the callback thread will never exit. 144 stop(); 145 if (mAudioTrackThread != 0) { 146 mProxy->interrupt(); 147 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 148 mAudioTrackThread->requestExitAndWait(); 149 mAudioTrackThread.clear(); 150 } 151 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 152 mAudioTrack.clear(); 153 IPCThreadState::self()->flushCommands(); 154 AudioSystem::releaseAudioSessionId(mSessionId); 155 } 156} 157 158status_t AudioTrack::set( 159 audio_stream_type_t streamType, 160 uint32_t sampleRate, 161 audio_format_t format, 162 audio_channel_mask_t channelMask, 163 int frameCountInt, 164 audio_output_flags_t flags, 165 callback_t cbf, 166 void* user, 167 int notificationFrames, 168 const sp<IMemory>& sharedBuffer, 169 bool threadCanCallJava, 170 int sessionId, 171 transfer_type transferType, 172 const audio_offload_info_t *offloadInfo) 173{ 174 switch (transferType) { 175 case TRANSFER_DEFAULT: 176 if (sharedBuffer != 0) { 177 transferType = TRANSFER_SHARED; 178 } else if (cbf == NULL || threadCanCallJava) { 179 transferType = TRANSFER_SYNC; 180 } else { 181 transferType = TRANSFER_CALLBACK; 182 } 183 break; 184 case TRANSFER_CALLBACK: 185 if (cbf == NULL || sharedBuffer != 0) { 186 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 187 return BAD_VALUE; 188 } 189 break; 190 case TRANSFER_OBTAIN: 191 case TRANSFER_SYNC: 192 if (sharedBuffer != 0) { 193 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 194 return BAD_VALUE; 195 } 196 break; 197 case TRANSFER_SHARED: 198 if (sharedBuffer == 0) { 199 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 200 return BAD_VALUE; 201 } 202 break; 203 default: 204 ALOGE("Invalid transfer type %d", transferType); 205 return BAD_VALUE; 206 } 207 mTransfer = transferType; 208 209 // FIXME "int" here is legacy and will be replaced by size_t later 210 if (frameCountInt < 0) { 211 ALOGE("Invalid frame count %d", frameCountInt); 212 return BAD_VALUE; 213 } 214 size_t frameCount = frameCountInt; 215 216 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 217 sharedBuffer->size()); 218 219 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 220 221 AutoMutex lock(mLock); 222 223 // invariant that mAudioTrack != 0 is true only after set() returns successfully 224 if (mAudioTrack != 0) { 225 ALOGE("Track already in use"); 226 return INVALID_OPERATION; 227 } 228 229 mOutput = 0; 230 231 // handle default values first. 232 if (streamType == AUDIO_STREAM_DEFAULT) { 233 streamType = AUDIO_STREAM_MUSIC; 234 } 235 236 if (sampleRate == 0) { 237 uint32_t afSampleRate; 238 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 239 return NO_INIT; 240 } 241 sampleRate = afSampleRate; 242 } 243 mSampleRate = sampleRate; 244 245 // these below should probably come from the audioFlinger too... 246 if (format == AUDIO_FORMAT_DEFAULT) { 247 format = AUDIO_FORMAT_PCM_16_BIT; 248 } 249 if (channelMask == 0) { 250 channelMask = AUDIO_CHANNEL_OUT_STEREO; 251 } 252 253 // validate parameters 254 if (!audio_is_valid_format(format)) { 255 ALOGE("Invalid format %d", format); 256 return BAD_VALUE; 257 } 258 259 // AudioFlinger does not currently support 8-bit data in shared memory 260 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 261 ALOGE("8-bit data in shared memory is not supported"); 262 return BAD_VALUE; 263 } 264 265 // force direct flag if format is not linear PCM 266 // or offload was requested 267 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 268 || !audio_is_linear_pcm(format)) { 269 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 270 ? "Offload request, forcing to Direct Output" 271 : "Not linear PCM, forcing to Direct Output"); 272 flags = (audio_output_flags_t) 273 // FIXME why can't we allow direct AND fast? 274 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 275 } 276 // only allow deep buffering for music stream type 277 if (streamType != AUDIO_STREAM_MUSIC) { 278 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 279 } 280 281 if (!audio_is_output_channel(channelMask)) { 282 ALOGE("Invalid channel mask %#x", channelMask); 283 return BAD_VALUE; 284 } 285 mChannelMask = channelMask; 286 uint32_t channelCount = popcount(channelMask); 287 mChannelCount = channelCount; 288 289 if (audio_is_linear_pcm(format)) { 290 mFrameSize = channelCount * audio_bytes_per_sample(format); 291 mFrameSizeAF = channelCount * sizeof(int16_t); 292 } else { 293 mFrameSize = sizeof(uint8_t); 294 mFrameSizeAF = sizeof(uint8_t); 295 } 296 297 audio_io_handle_t output = AudioSystem::getOutput( 298 streamType, 299 sampleRate, format, channelMask, 300 flags, 301 offloadInfo); 302 303 if (output == 0) { 304 ALOGE("Could not get audio output for stream type %d", streamType); 305 return BAD_VALUE; 306 } 307 308 mVolume[LEFT] = 1.0f; 309 mVolume[RIGHT] = 1.0f; 310 mSendLevel = 0.0f; 311 mFrameCount = frameCount; 312 mReqFrameCount = frameCount; 313 mNotificationFramesReq = notificationFrames; 314 mNotificationFramesAct = 0; 315 mSessionId = sessionId; 316 mAuxEffectId = 0; 317 mFlags = flags; 318 mCbf = cbf; 319 320 if (cbf != NULL) { 321 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 322 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 323 } 324 325 // create the IAudioTrack 326 status_t status = createTrack_l(streamType, 327 sampleRate, 328 format, 329 frameCount, 330 flags, 331 sharedBuffer, 332 output, 333 0 /*epoch*/); 334 335 if (status != NO_ERROR) { 336 if (mAudioTrackThread != 0) { 337 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 338 mAudioTrackThread->requestExitAndWait(); 339 mAudioTrackThread.clear(); 340 } 341 //Use of direct and offloaded output streams is ref counted by audio policy manager. 342 // As getOutput was called above and resulted in an output stream to be opened, 343 // we need to release it. 344 AudioSystem::releaseOutput(output); 345 return status; 346 } 347 348 mStatus = NO_ERROR; 349 mStreamType = streamType; 350 mFormat = format; 351 mSharedBuffer = sharedBuffer; 352 mState = STATE_STOPPED; 353 mUserData = user; 354 mLoopPeriod = 0; 355 mMarkerPosition = 0; 356 mMarkerReached = false; 357 mNewPosition = 0; 358 mUpdatePeriod = 0; 359 AudioSystem::acquireAudioSessionId(mSessionId); 360 mSequence = 1; 361 mObservedSequence = mSequence; 362 mInUnderrun = false; 363 mOutput = output; 364 365 return NO_ERROR; 366} 367 368// ------------------------------------------------------------------------- 369 370status_t AudioTrack::start() 371{ 372 AutoMutex lock(mLock); 373 374 if (mState == STATE_ACTIVE) { 375 return INVALID_OPERATION; 376 } 377 378 mInUnderrun = true; 379 380 State previousState = mState; 381 if (previousState == STATE_PAUSED_STOPPING) { 382 mState = STATE_STOPPING; 383 } else { 384 mState = STATE_ACTIVE; 385 } 386 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 387 // reset current position as seen by client to 0 388 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 389 // force refresh of remaining frames by processAudioBuffer() as last 390 // write before stop could be partial. 391 mRefreshRemaining = true; 392 } 393 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 394 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 395 396 sp<AudioTrackThread> t = mAudioTrackThread; 397 if (t != 0) { 398 if (previousState == STATE_STOPPING) { 399 mProxy->interrupt(); 400 } else { 401 t->resume(); 402 } 403 } else { 404 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 405 get_sched_policy(0, &mPreviousSchedulingGroup); 406 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 407 } 408 409 status_t status = NO_ERROR; 410 if (!(flags & CBLK_INVALID)) { 411 status = mAudioTrack->start(); 412 if (status == DEAD_OBJECT) { 413 flags |= CBLK_INVALID; 414 } 415 } 416 if (flags & CBLK_INVALID) { 417 status = restoreTrack_l("start"); 418 } 419 420 if (status != NO_ERROR) { 421 ALOGE("start() status %d", status); 422 mState = previousState; 423 if (t != 0) { 424 if (previousState != STATE_STOPPING) { 425 t->pause(); 426 } 427 } else { 428 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 429 set_sched_policy(0, mPreviousSchedulingGroup); 430 } 431 } 432 433 return status; 434} 435 436void AudioTrack::stop() 437{ 438 AutoMutex lock(mLock); 439 // FIXME pause then stop should not be a nop 440 if (mState != STATE_ACTIVE) { 441 return; 442 } 443 444 if (isOffloaded()) { 445 mState = STATE_STOPPING; 446 } else { 447 mState = STATE_STOPPED; 448 } 449 450 mProxy->interrupt(); 451 mAudioTrack->stop(); 452 // the playback head position will reset to 0, so if a marker is set, we need 453 // to activate it again 454 mMarkerReached = false; 455#if 0 456 // Force flush if a shared buffer is used otherwise audioflinger 457 // will not stop before end of buffer is reached. 458 // It may be needed to make sure that we stop playback, likely in case looping is on. 459 if (mSharedBuffer != 0) { 460 flush_l(); 461 } 462#endif 463 464 sp<AudioTrackThread> t = mAudioTrackThread; 465 if (t != 0) { 466 if (!isOffloaded()) { 467 t->pause(); 468 } 469 } else { 470 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 471 set_sched_policy(0, mPreviousSchedulingGroup); 472 } 473} 474 475bool AudioTrack::stopped() const 476{ 477 AutoMutex lock(mLock); 478 return mState != STATE_ACTIVE; 479} 480 481void AudioTrack::flush() 482{ 483 if (mSharedBuffer != 0) { 484 return; 485 } 486 AutoMutex lock(mLock); 487 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 488 return; 489 } 490 flush_l(); 491} 492 493void AudioTrack::flush_l() 494{ 495 ALOG_ASSERT(mState != STATE_ACTIVE); 496 497 // clear playback marker and periodic update counter 498 mMarkerPosition = 0; 499 mMarkerReached = false; 500 mUpdatePeriod = 0; 501 mRefreshRemaining = true; 502 503 mState = STATE_FLUSHED; 504 if (isOffloaded()) { 505 mProxy->interrupt(); 506 } 507 mProxy->flush(); 508 mAudioTrack->flush(); 509} 510 511void AudioTrack::pause() 512{ 513 AutoMutex lock(mLock); 514 if (mState == STATE_ACTIVE) { 515 mState = STATE_PAUSED; 516 } else if (mState == STATE_STOPPING) { 517 mState = STATE_PAUSED_STOPPING; 518 } else { 519 return; 520 } 521 mProxy->interrupt(); 522 mAudioTrack->pause(); 523} 524 525status_t AudioTrack::setVolume(float left, float right) 526{ 527 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 528 return BAD_VALUE; 529 } 530 531 AutoMutex lock(mLock); 532 mVolume[LEFT] = left; 533 mVolume[RIGHT] = right; 534 535 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 536 537 if (isOffloaded()) { 538 mAudioTrack->signal(); 539 } 540 return NO_ERROR; 541} 542 543status_t AudioTrack::setVolume(float volume) 544{ 545 return setVolume(volume, volume); 546} 547 548status_t AudioTrack::setAuxEffectSendLevel(float level) 549{ 550 if (level < 0.0f || level > 1.0f) { 551 return BAD_VALUE; 552 } 553 554 AutoMutex lock(mLock); 555 mSendLevel = level; 556 mProxy->setSendLevel(level); 557 558 return NO_ERROR; 559} 560 561void AudioTrack::getAuxEffectSendLevel(float* level) const 562{ 563 if (level != NULL) { 564 *level = mSendLevel; 565 } 566} 567 568status_t AudioTrack::setSampleRate(uint32_t rate) 569{ 570 if (mIsTimed || isOffloaded()) { 571 return INVALID_OPERATION; 572 } 573 574 uint32_t afSamplingRate; 575 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 576 return NO_INIT; 577 } 578 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 579 if (rate == 0 || rate > afSamplingRate*2 ) { 580 return BAD_VALUE; 581 } 582 583 AutoMutex lock(mLock); 584 mSampleRate = rate; 585 mProxy->setSampleRate(rate); 586 587 return NO_ERROR; 588} 589 590uint32_t AudioTrack::getSampleRate() const 591{ 592 if (mIsTimed) { 593 return 0; 594 } 595 596 AutoMutex lock(mLock); 597 return mSampleRate; 598} 599 600status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 601{ 602 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 603 return INVALID_OPERATION; 604 } 605 606 if (loopCount == 0) { 607 ; 608 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 609 loopEnd - loopStart >= MIN_LOOP) { 610 ; 611 } else { 612 return BAD_VALUE; 613 } 614 615 AutoMutex lock(mLock); 616 // See setPosition() regarding setting parameters such as loop points or position while active 617 if (mState == STATE_ACTIVE) { 618 return INVALID_OPERATION; 619 } 620 setLoop_l(loopStart, loopEnd, loopCount); 621 return NO_ERROR; 622} 623 624void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 625{ 626 // FIXME If setting a loop also sets position to start of loop, then 627 // this is correct. Otherwise it should be removed. 628 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 629 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 630 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 631} 632 633status_t AudioTrack::setMarkerPosition(uint32_t marker) 634{ 635 // The only purpose of setting marker position is to get a callback 636 if (mCbf == NULL || isOffloaded()) { 637 return INVALID_OPERATION; 638 } 639 640 AutoMutex lock(mLock); 641 mMarkerPosition = marker; 642 mMarkerReached = false; 643 644 return NO_ERROR; 645} 646 647status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 648{ 649 if (isOffloaded()) { 650 return INVALID_OPERATION; 651 } 652 if (marker == NULL) { 653 return BAD_VALUE; 654 } 655 656 AutoMutex lock(mLock); 657 *marker = mMarkerPosition; 658 659 return NO_ERROR; 660} 661 662status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 663{ 664 // The only purpose of setting position update period is to get a callback 665 if (mCbf == NULL || isOffloaded()) { 666 return INVALID_OPERATION; 667 } 668 669 AutoMutex lock(mLock); 670 mNewPosition = mProxy->getPosition() + updatePeriod; 671 mUpdatePeriod = updatePeriod; 672 return NO_ERROR; 673} 674 675status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 676{ 677 if (isOffloaded()) { 678 return INVALID_OPERATION; 679 } 680 if (updatePeriod == NULL) { 681 return BAD_VALUE; 682 } 683 684 AutoMutex lock(mLock); 685 *updatePeriod = mUpdatePeriod; 686 687 return NO_ERROR; 688} 689 690status_t AudioTrack::setPosition(uint32_t position) 691{ 692 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 693 return INVALID_OPERATION; 694 } 695 if (position > mFrameCount) { 696 return BAD_VALUE; 697 } 698 699 AutoMutex lock(mLock); 700 // Currently we require that the player is inactive before setting parameters such as position 701 // or loop points. Otherwise, there could be a race condition: the application could read the 702 // current position, compute a new position or loop parameters, and then set that position or 703 // loop parameters but it would do the "wrong" thing since the position has continued to advance 704 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 705 // to specify how it wants to handle such scenarios. 706 if (mState == STATE_ACTIVE) { 707 return INVALID_OPERATION; 708 } 709 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 710 mLoopPeriod = 0; 711 // FIXME Check whether loops and setting position are incompatible in old code. 712 // If we use setLoop for both purposes we lose the capability to set the position while looping. 713 mStaticProxy->setLoop(position, mFrameCount, 0); 714 715 return NO_ERROR; 716} 717 718status_t AudioTrack::getPosition(uint32_t *position) const 719{ 720 if (position == NULL) { 721 return BAD_VALUE; 722 } 723 724 AutoMutex lock(mLock); 725 if (isOffloaded()) { 726 uint32_t dspFrames = 0; 727 728 if (mOutput != 0) { 729 uint32_t halFrames; 730 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 731 } 732 *position = dspFrames; 733 } else { 734 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 735 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 736 mProxy->getPosition(); 737 } 738 return NO_ERROR; 739} 740 741status_t AudioTrack::getBufferPosition(size_t *position) 742{ 743 if (mSharedBuffer == 0 || mIsTimed) { 744 return INVALID_OPERATION; 745 } 746 if (position == NULL) { 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 *position = mStaticProxy->getBufferPosition(); 752 return NO_ERROR; 753} 754 755status_t AudioTrack::reload() 756{ 757 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 758 return INVALID_OPERATION; 759 } 760 761 AutoMutex lock(mLock); 762 // See setPosition() regarding setting parameters such as loop points or position while active 763 if (mState == STATE_ACTIVE) { 764 return INVALID_OPERATION; 765 } 766 mNewPosition = mUpdatePeriod; 767 mLoopPeriod = 0; 768 // FIXME The new code cannot reload while keeping a loop specified. 769 // Need to check how the old code handled this, and whether it's a significant change. 770 mStaticProxy->setLoop(0, mFrameCount, 0); 771 return NO_ERROR; 772} 773 774audio_io_handle_t AudioTrack::getOutput() 775{ 776 AutoMutex lock(mLock); 777 return mOutput; 778} 779 780// must be called with mLock held 781audio_io_handle_t AudioTrack::getOutput_l() 782{ 783 if (mOutput) { 784 return mOutput; 785 } else { 786 return AudioSystem::getOutput(mStreamType, 787 mSampleRate, mFormat, mChannelMask, mFlags); 788 } 789} 790 791status_t AudioTrack::attachAuxEffect(int effectId) 792{ 793 AutoMutex lock(mLock); 794 status_t status = mAudioTrack->attachAuxEffect(effectId); 795 if (status == NO_ERROR) { 796 mAuxEffectId = effectId; 797 } 798 return status; 799} 800 801// ------------------------------------------------------------------------- 802 803// must be called with mLock held 804status_t AudioTrack::createTrack_l( 805 audio_stream_type_t streamType, 806 uint32_t sampleRate, 807 audio_format_t format, 808 size_t frameCount, 809 audio_output_flags_t flags, 810 const sp<IMemory>& sharedBuffer, 811 audio_io_handle_t output, 812 size_t epoch) 813{ 814 status_t status; 815 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 816 if (audioFlinger == 0) { 817 ALOGE("Could not get audioflinger"); 818 return NO_INIT; 819 } 820 821 // Not all of these values are needed under all conditions, but it is easier to get them all 822 823 uint32_t afLatency; 824 status = AudioSystem::getLatency(output, streamType, &afLatency); 825 if (status != NO_ERROR) { 826 ALOGE("getLatency(%d) failed status %d", output, status); 827 return NO_INIT; 828 } 829 830 size_t afFrameCount; 831 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 832 if (status != NO_ERROR) { 833 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 834 return NO_INIT; 835 } 836 837 uint32_t afSampleRate; 838 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 839 if (status != NO_ERROR) { 840 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 841 return NO_INIT; 842 } 843 844 // Client decides whether the track is TIMED (see below), but can only express a preference 845 // for FAST. Server will perform additional tests. 846 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 847 // either of these use cases: 848 // use case 1: shared buffer 849 (sharedBuffer != 0) || 850 // use case 2: callback handler 851 (mCbf != NULL))) { 852 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 853 // once denied, do not request again if IAudioTrack is re-created 854 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 855 mFlags = flags; 856 } 857 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 858 859 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 860 // n = 1 fast track with single buffering; nBuffering is ignored 861 // n = 2 fast track with double buffering 862 // n = 2 normal track, no sample rate conversion 863 // n = 3 normal track, with sample rate conversion 864 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 865 // n > 3 very high latency or very small notification interval; nBuffering is ignored 866 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 867 868 mNotificationFramesAct = mNotificationFramesReq; 869 870 if (!audio_is_linear_pcm(format)) { 871 872 if (sharedBuffer != 0) { 873 // Same comment as below about ignoring frameCount parameter for set() 874 frameCount = sharedBuffer->size(); 875 } else if (frameCount == 0) { 876 frameCount = afFrameCount; 877 } 878 if (mNotificationFramesAct != frameCount) { 879 mNotificationFramesAct = frameCount; 880 } 881 } else if (sharedBuffer != 0) { 882 883 // Ensure that buffer alignment matches channel count 884 // 8-bit data in shared memory is not currently supported by AudioFlinger 885 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 886 if (mChannelCount > 1) { 887 // More than 2 channels does not require stronger alignment than stereo 888 alignment <<= 1; 889 } 890 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 891 ALOGE("Invalid buffer alignment: address %p, channel count %u", 892 sharedBuffer->pointer(), mChannelCount); 893 return BAD_VALUE; 894 } 895 896 // When initializing a shared buffer AudioTrack via constructors, 897 // there's no frameCount parameter. 898 // But when initializing a shared buffer AudioTrack via set(), 899 // there _is_ a frameCount parameter. We silently ignore it. 900 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 901 902 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 903 904 // FIXME move these calculations and associated checks to server 905 906 // Ensure that buffer depth covers at least audio hardware latency 907 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 908 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 909 afFrameCount, minBufCount, afSampleRate, afLatency); 910 if (minBufCount <= nBuffering) { 911 minBufCount = nBuffering; 912 } 913 914 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 915 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 916 ", afLatency=%d", 917 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 918 919 if (frameCount == 0) { 920 frameCount = minFrameCount; 921 } else if (frameCount < minFrameCount) { 922 // not ALOGW because it happens all the time when playing key clicks over A2DP 923 ALOGV("Minimum buffer size corrected from %d to %d", 924 frameCount, minFrameCount); 925 frameCount = minFrameCount; 926 } 927 // Make sure that application is notified with sufficient margin before underrun 928 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 929 mNotificationFramesAct = frameCount/nBuffering; 930 } 931 932 } else { 933 // For fast tracks, the frame count calculations and checks are done by server 934 } 935 936 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 937 if (mIsTimed) { 938 trackFlags |= IAudioFlinger::TRACK_TIMED; 939 } 940 941 pid_t tid = -1; 942 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 943 trackFlags |= IAudioFlinger::TRACK_FAST; 944 if (mAudioTrackThread != 0) { 945 tid = mAudioTrackThread->getTid(); 946 } 947 } 948 949 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 950 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 951 } 952 953 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 954 sampleRate, 955 // AudioFlinger only sees 16-bit PCM 956 format == AUDIO_FORMAT_PCM_8_BIT ? 957 AUDIO_FORMAT_PCM_16_BIT : format, 958 mChannelMask, 959 frameCount, 960 &trackFlags, 961 sharedBuffer, 962 output, 963 tid, 964 &mSessionId, 965 mName, 966 &status); 967 968 if (track == 0) { 969 ALOGE("AudioFlinger could not create track, status: %d", status); 970 return status; 971 } 972 sp<IMemory> iMem = track->getCblk(); 973 if (iMem == 0) { 974 ALOGE("Could not get control block"); 975 return NO_INIT; 976 } 977 // invariant that mAudioTrack != 0 is true only after set() returns successfully 978 if (mAudioTrack != 0) { 979 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 980 mDeathNotifier.clear(); 981 } 982 mAudioTrack = track; 983 mCblkMemory = iMem; 984 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 985 mCblk = cblk; 986 size_t temp = cblk->frameCount_; 987 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 988 // In current design, AudioTrack client checks and ensures frame count validity before 989 // passing it to AudioFlinger so AudioFlinger should not return a different value except 990 // for fast track as it uses a special method of assigning frame count. 991 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 992 } 993 frameCount = temp; 994 mAwaitBoost = false; 995 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 996 if (trackFlags & IAudioFlinger::TRACK_FAST) { 997 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 998 mAwaitBoost = true; 999 if (sharedBuffer == 0) { 1000 // Theoretically double-buffering is not required for fast tracks, 1001 // due to tighter scheduling. But in practice, to accommodate kernels with 1002 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1003 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1004 mNotificationFramesAct = frameCount/nBuffering; 1005 } 1006 } 1007 } else { 1008 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1009 // once denied, do not request again if IAudioTrack is re-created 1010 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1011 mFlags = flags; 1012 if (sharedBuffer == 0) { 1013 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1014 mNotificationFramesAct = frameCount/nBuffering; 1015 } 1016 } 1017 } 1018 } 1019 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1020 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1021 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1022 } else { 1023 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1024 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1025 mFlags = flags; 1026 return NO_INIT; 1027 } 1028 } 1029 1030 mRefreshRemaining = true; 1031 1032 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1033 // is the value of pointer() for the shared buffer, otherwise buffers points 1034 // immediately after the control block. This address is for the mapping within client 1035 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1036 void* buffers; 1037 if (sharedBuffer == 0) { 1038 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1039 } else { 1040 buffers = sharedBuffer->pointer(); 1041 } 1042 1043 mAudioTrack->attachAuxEffect(mAuxEffectId); 1044 // FIXME don't believe this lie 1045 mLatency = afLatency + (1000*frameCount) / sampleRate; 1046 mFrameCount = frameCount; 1047 // If IAudioTrack is re-created, don't let the requested frameCount 1048 // decrease. This can confuse clients that cache frameCount(). 1049 if (frameCount > mReqFrameCount) { 1050 mReqFrameCount = frameCount; 1051 } 1052 1053 // update proxy 1054 if (sharedBuffer == 0) { 1055 mStaticProxy.clear(); 1056 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1057 } else { 1058 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1059 mProxy = mStaticProxy; 1060 } 1061 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1062 uint16_t(mVolume[LEFT] * 0x1000)); 1063 mProxy->setSendLevel(mSendLevel); 1064 mProxy->setSampleRate(mSampleRate); 1065 mProxy->setEpoch(epoch); 1066 mProxy->setMinimum(mNotificationFramesAct); 1067 1068 mDeathNotifier = new DeathNotifier(this); 1069 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1070 1071 return NO_ERROR; 1072} 1073 1074status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1075{ 1076 if (audioBuffer == NULL) { 1077 return BAD_VALUE; 1078 } 1079 if (mTransfer != TRANSFER_OBTAIN) { 1080 audioBuffer->frameCount = 0; 1081 audioBuffer->size = 0; 1082 audioBuffer->raw = NULL; 1083 return INVALID_OPERATION; 1084 } 1085 1086 const struct timespec *requested; 1087 if (waitCount == -1) { 1088 requested = &ClientProxy::kForever; 1089 } else if (waitCount == 0) { 1090 requested = &ClientProxy::kNonBlocking; 1091 } else if (waitCount > 0) { 1092 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1093 struct timespec timeout; 1094 timeout.tv_sec = ms / 1000; 1095 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1096 requested = &timeout; 1097 } else { 1098 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1099 requested = NULL; 1100 } 1101 return obtainBuffer(audioBuffer, requested); 1102} 1103 1104status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1105 struct timespec *elapsed, size_t *nonContig) 1106{ 1107 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1108 uint32_t oldSequence = 0; 1109 uint32_t newSequence; 1110 1111 Proxy::Buffer buffer; 1112 status_t status = NO_ERROR; 1113 1114 static const int32_t kMaxTries = 5; 1115 int32_t tryCounter = kMaxTries; 1116 1117 do { 1118 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1119 // keep them from going away if another thread re-creates the track during obtainBuffer() 1120 sp<AudioTrackClientProxy> proxy; 1121 sp<IMemory> iMem; 1122 1123 { // start of lock scope 1124 AutoMutex lock(mLock); 1125 1126 newSequence = mSequence; 1127 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1128 if (status == DEAD_OBJECT) { 1129 // re-create track, unless someone else has already done so 1130 if (newSequence == oldSequence) { 1131 status = restoreTrack_l("obtainBuffer"); 1132 if (status != NO_ERROR) { 1133 buffer.mFrameCount = 0; 1134 buffer.mRaw = NULL; 1135 buffer.mNonContig = 0; 1136 break; 1137 } 1138 } 1139 } 1140 oldSequence = newSequence; 1141 1142 // Keep the extra references 1143 proxy = mProxy; 1144 iMem = mCblkMemory; 1145 1146 if (mState == STATE_STOPPING) { 1147 status = -EINTR; 1148 buffer.mFrameCount = 0; 1149 buffer.mRaw = NULL; 1150 buffer.mNonContig = 0; 1151 break; 1152 } 1153 1154 // Non-blocking if track is stopped or paused 1155 if (mState != STATE_ACTIVE) { 1156 requested = &ClientProxy::kNonBlocking; 1157 } 1158 1159 } // end of lock scope 1160 1161 buffer.mFrameCount = audioBuffer->frameCount; 1162 // FIXME starts the requested timeout and elapsed over from scratch 1163 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1164 1165 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1166 1167 audioBuffer->frameCount = buffer.mFrameCount; 1168 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1169 audioBuffer->raw = buffer.mRaw; 1170 if (nonContig != NULL) { 1171 *nonContig = buffer.mNonContig; 1172 } 1173 return status; 1174} 1175 1176void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1177{ 1178 if (mTransfer == TRANSFER_SHARED) { 1179 return; 1180 } 1181 1182 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1183 if (stepCount == 0) { 1184 return; 1185 } 1186 1187 Proxy::Buffer buffer; 1188 buffer.mFrameCount = stepCount; 1189 buffer.mRaw = audioBuffer->raw; 1190 1191 AutoMutex lock(mLock); 1192 mInUnderrun = false; 1193 mProxy->releaseBuffer(&buffer); 1194 1195 // restart track if it was disabled by audioflinger due to previous underrun 1196 if (mState == STATE_ACTIVE) { 1197 audio_track_cblk_t* cblk = mCblk; 1198 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1199 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1200 this, mName.string()); 1201 // FIXME ignoring status 1202 mAudioTrack->start(); 1203 } 1204 } 1205} 1206 1207// ------------------------------------------------------------------------- 1208 1209ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1210{ 1211 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1212 return INVALID_OPERATION; 1213 } 1214 1215 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1216 // Sanity-check: user is most-likely passing an error code, and it would 1217 // make the return value ambiguous (actualSize vs error). 1218 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1219 return BAD_VALUE; 1220 } 1221 1222 size_t written = 0; 1223 Buffer audioBuffer; 1224 1225 while (userSize >= mFrameSize) { 1226 audioBuffer.frameCount = userSize / mFrameSize; 1227 1228 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1229 if (err < 0) { 1230 if (written > 0) { 1231 break; 1232 } 1233 return ssize_t(err); 1234 } 1235 1236 size_t toWrite; 1237 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1238 // Divide capacity by 2 to take expansion into account 1239 toWrite = audioBuffer.size >> 1; 1240 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1241 } else { 1242 toWrite = audioBuffer.size; 1243 memcpy(audioBuffer.i8, buffer, toWrite); 1244 } 1245 buffer = ((const char *) buffer) + toWrite; 1246 userSize -= toWrite; 1247 written += toWrite; 1248 1249 releaseBuffer(&audioBuffer); 1250 } 1251 1252 return written; 1253} 1254 1255// ------------------------------------------------------------------------- 1256 1257TimedAudioTrack::TimedAudioTrack() { 1258 mIsTimed = true; 1259} 1260 1261status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1262{ 1263 AutoMutex lock(mLock); 1264 status_t result = UNKNOWN_ERROR; 1265 1266#if 1 1267 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1268 // while we are accessing the cblk 1269 sp<IAudioTrack> audioTrack = mAudioTrack; 1270 sp<IMemory> iMem = mCblkMemory; 1271#endif 1272 1273 // If the track is not invalid already, try to allocate a buffer. alloc 1274 // fails indicating that the server is dead, flag the track as invalid so 1275 // we can attempt to restore in just a bit. 1276 audio_track_cblk_t* cblk = mCblk; 1277 if (!(cblk->mFlags & CBLK_INVALID)) { 1278 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1279 if (result == DEAD_OBJECT) { 1280 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1281 } 1282 } 1283 1284 // If the track is invalid at this point, attempt to restore it. and try the 1285 // allocation one more time. 1286 if (cblk->mFlags & CBLK_INVALID) { 1287 result = restoreTrack_l("allocateTimedBuffer"); 1288 1289 if (result == NO_ERROR) { 1290 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1291 } 1292 } 1293 1294 return result; 1295} 1296 1297status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1298 int64_t pts) 1299{ 1300 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1301 { 1302 AutoMutex lock(mLock); 1303 audio_track_cblk_t* cblk = mCblk; 1304 // restart track if it was disabled by audioflinger due to previous underrun 1305 if (buffer->size() != 0 && status == NO_ERROR && 1306 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1307 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1308 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1309 // FIXME ignoring status 1310 mAudioTrack->start(); 1311 } 1312 } 1313 return status; 1314} 1315 1316status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1317 TargetTimeline target) 1318{ 1319 return mAudioTrack->setMediaTimeTransform(xform, target); 1320} 1321 1322// ------------------------------------------------------------------------- 1323 1324nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1325{ 1326 // Currently the AudioTrack thread is not created if there are no callbacks. 1327 // Would it ever make sense to run the thread, even without callbacks? 1328 // If so, then replace this by checks at each use for mCbf != NULL. 1329 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1330 1331 mLock.lock(); 1332 if (mAwaitBoost) { 1333 mAwaitBoost = false; 1334 mLock.unlock(); 1335 static const int32_t kMaxTries = 5; 1336 int32_t tryCounter = kMaxTries; 1337 uint32_t pollUs = 10000; 1338 do { 1339 int policy = sched_getscheduler(0); 1340 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1341 break; 1342 } 1343 usleep(pollUs); 1344 pollUs <<= 1; 1345 } while (tryCounter-- > 0); 1346 if (tryCounter < 0) { 1347 ALOGE("did not receive expected priority boost on time"); 1348 } 1349 // Run again immediately 1350 return 0; 1351 } 1352 1353 // Can only reference mCblk while locked 1354 int32_t flags = android_atomic_and( 1355 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1356 1357 // Check for track invalidation 1358 if (flags & CBLK_INVALID) { 1359 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1360 // AudioSystem cache. We should not exit here but after calling the callback so 1361 // that the upper layers can recreate the track 1362 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1363 status_t status = restoreTrack_l("processAudioBuffer"); 1364 mLock.unlock(); 1365 // Run again immediately, but with a new IAudioTrack 1366 return 0; 1367 } 1368 } 1369 1370 bool waitStreamEnd = mState == STATE_STOPPING; 1371 bool active = mState == STATE_ACTIVE; 1372 1373 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1374 bool newUnderrun = false; 1375 if (flags & CBLK_UNDERRUN) { 1376#if 0 1377 // Currently in shared buffer mode, when the server reaches the end of buffer, 1378 // the track stays active in continuous underrun state. It's up to the application 1379 // to pause or stop the track, or set the position to a new offset within buffer. 1380 // This was some experimental code to auto-pause on underrun. Keeping it here 1381 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1382 if (mTransfer == TRANSFER_SHARED) { 1383 mState = STATE_PAUSED; 1384 active = false; 1385 } 1386#endif 1387 if (!mInUnderrun) { 1388 mInUnderrun = true; 1389 newUnderrun = true; 1390 } 1391 } 1392 1393 // Get current position of server 1394 size_t position = mProxy->getPosition(); 1395 1396 // Manage marker callback 1397 bool markerReached = false; 1398 size_t markerPosition = mMarkerPosition; 1399 // FIXME fails for wraparound, need 64 bits 1400 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1401 mMarkerReached = markerReached = true; 1402 } 1403 1404 // Determine number of new position callback(s) that will be needed, while locked 1405 size_t newPosCount = 0; 1406 size_t newPosition = mNewPosition; 1407 size_t updatePeriod = mUpdatePeriod; 1408 // FIXME fails for wraparound, need 64 bits 1409 if (updatePeriod > 0 && position >= newPosition) { 1410 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1411 mNewPosition += updatePeriod * newPosCount; 1412 } 1413 1414 // Cache other fields that will be needed soon 1415 uint32_t loopPeriod = mLoopPeriod; 1416 uint32_t sampleRate = mSampleRate; 1417 size_t notificationFrames = mNotificationFramesAct; 1418 if (mRefreshRemaining) { 1419 mRefreshRemaining = false; 1420 mRemainingFrames = notificationFrames; 1421 mRetryOnPartialBuffer = false; 1422 } 1423 size_t misalignment = mProxy->getMisalignment(); 1424 uint32_t sequence = mSequence; 1425 1426 // These fields don't need to be cached, because they are assigned only by set(): 1427 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1428 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1429 1430 mLock.unlock(); 1431 1432 if (waitStreamEnd) { 1433 AutoMutex lock(mLock); 1434 1435 sp<AudioTrackClientProxy> proxy = mProxy; 1436 sp<IMemory> iMem = mCblkMemory; 1437 1438 struct timespec timeout; 1439 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1440 timeout.tv_nsec = 0; 1441 1442 mLock.unlock(); 1443 status_t status = mProxy->waitStreamEndDone(&timeout); 1444 mLock.lock(); 1445 switch (status) { 1446 case NO_ERROR: 1447 case DEAD_OBJECT: 1448 case TIMED_OUT: 1449 mLock.unlock(); 1450 mCbf(EVENT_STREAM_END, mUserData, NULL); 1451 mLock.lock(); 1452 if (mState == STATE_STOPPING) { 1453 mState = STATE_STOPPED; 1454 if (status != DEAD_OBJECT) { 1455 return NS_INACTIVE; 1456 } 1457 } 1458 return 0; 1459 default: 1460 return 0; 1461 } 1462 } 1463 1464 // perform callbacks while unlocked 1465 if (newUnderrun) { 1466 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1467 } 1468 // FIXME we will miss loops if loop cycle was signaled several times since last call 1469 // to processAudioBuffer() 1470 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1471 mCbf(EVENT_LOOP_END, mUserData, NULL); 1472 } 1473 if (flags & CBLK_BUFFER_END) { 1474 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1475 } 1476 if (markerReached) { 1477 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1478 } 1479 while (newPosCount > 0) { 1480 size_t temp = newPosition; 1481 mCbf(EVENT_NEW_POS, mUserData, &temp); 1482 newPosition += updatePeriod; 1483 newPosCount--; 1484 } 1485 1486 if (mObservedSequence != sequence) { 1487 mObservedSequence = sequence; 1488 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1489 // for offloaded tracks, just wait for the upper layers to recreate the track 1490 if (isOffloaded()) { 1491 return NS_INACTIVE; 1492 } 1493 } 1494 1495 // if inactive, then don't run me again until re-started 1496 if (!active) { 1497 return NS_INACTIVE; 1498 } 1499 1500 // Compute the estimated time until the next timed event (position, markers, loops) 1501 // FIXME only for non-compressed audio 1502 uint32_t minFrames = ~0; 1503 if (!markerReached && position < markerPosition) { 1504 minFrames = markerPosition - position; 1505 } 1506 if (loopPeriod > 0 && loopPeriod < minFrames) { 1507 minFrames = loopPeriod; 1508 } 1509 if (updatePeriod > 0 && updatePeriod < minFrames) { 1510 minFrames = updatePeriod; 1511 } 1512 1513 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1514 static const uint32_t kPoll = 0; 1515 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1516 minFrames = kPoll * notificationFrames; 1517 } 1518 1519 // Convert frame units to time units 1520 nsecs_t ns = NS_WHENEVER; 1521 if (minFrames != (uint32_t) ~0) { 1522 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1523 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1524 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1525 } 1526 1527 // If not supplying data by EVENT_MORE_DATA, then we're done 1528 if (mTransfer != TRANSFER_CALLBACK) { 1529 return ns; 1530 } 1531 1532 struct timespec timeout; 1533 const struct timespec *requested = &ClientProxy::kForever; 1534 if (ns != NS_WHENEVER) { 1535 timeout.tv_sec = ns / 1000000000LL; 1536 timeout.tv_nsec = ns % 1000000000LL; 1537 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1538 requested = &timeout; 1539 } 1540 1541 while (mRemainingFrames > 0) { 1542 1543 Buffer audioBuffer; 1544 audioBuffer.frameCount = mRemainingFrames; 1545 size_t nonContig; 1546 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1547 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1548 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1549 requested = &ClientProxy::kNonBlocking; 1550 size_t avail = audioBuffer.frameCount + nonContig; 1551 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1552 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1553 if (err != NO_ERROR) { 1554 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1555 (isOffloaded() && (err == DEAD_OBJECT))) { 1556 return 0; 1557 } 1558 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1559 return NS_NEVER; 1560 } 1561 1562 if (mRetryOnPartialBuffer && !isOffloaded()) { 1563 mRetryOnPartialBuffer = false; 1564 if (avail < mRemainingFrames) { 1565 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1566 if (ns < 0 || myns < ns) { 1567 ns = myns; 1568 } 1569 return ns; 1570 } 1571 } 1572 1573 // Divide buffer size by 2 to take into account the expansion 1574 // due to 8 to 16 bit conversion: the callback must fill only half 1575 // of the destination buffer 1576 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1577 audioBuffer.size >>= 1; 1578 } 1579 1580 size_t reqSize = audioBuffer.size; 1581 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1582 size_t writtenSize = audioBuffer.size; 1583 size_t writtenFrames = writtenSize / mFrameSize; 1584 1585 // Sanity check on returned size 1586 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1587 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1588 reqSize, (int) writtenSize); 1589 return NS_NEVER; 1590 } 1591 1592 if (writtenSize == 0) { 1593 // The callback is done filling buffers 1594 // Keep this thread going to handle timed events and 1595 // still try to get more data in intervals of WAIT_PERIOD_MS 1596 // but don't just loop and block the CPU, so wait 1597 return WAIT_PERIOD_MS * 1000000LL; 1598 } 1599 1600 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1601 // 8 to 16 bit conversion, note that source and destination are the same address 1602 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1603 audioBuffer.size <<= 1; 1604 } 1605 1606 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1607 audioBuffer.frameCount = releasedFrames; 1608 mRemainingFrames -= releasedFrames; 1609 if (misalignment >= releasedFrames) { 1610 misalignment -= releasedFrames; 1611 } else { 1612 misalignment = 0; 1613 } 1614 1615 releaseBuffer(&audioBuffer); 1616 1617 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1618 // if callback doesn't like to accept the full chunk 1619 if (writtenSize < reqSize) { 1620 continue; 1621 } 1622 1623 // There could be enough non-contiguous frames available to satisfy the remaining request 1624 if (mRemainingFrames <= nonContig) { 1625 continue; 1626 } 1627 1628#if 0 1629 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1630 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1631 // that total to a sum == notificationFrames. 1632 if (0 < misalignment && misalignment <= mRemainingFrames) { 1633 mRemainingFrames = misalignment; 1634 return (mRemainingFrames * 1100000000LL) / sampleRate; 1635 } 1636#endif 1637 1638 } 1639 mRemainingFrames = notificationFrames; 1640 mRetryOnPartialBuffer = true; 1641 1642 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1643 return 0; 1644} 1645 1646status_t AudioTrack::restoreTrack_l(const char *from) 1647{ 1648 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1649 isOffloaded() ? "Offloaded" : "PCM", from); 1650 ++mSequence; 1651 status_t result; 1652 1653 // refresh the audio configuration cache in this process to make sure we get new 1654 // output parameters in getOutput_l() and createTrack_l() 1655 AudioSystem::clearAudioConfigCache(); 1656 1657 if (isOffloaded()) { 1658 return DEAD_OBJECT; 1659 } 1660 1661 // force new output query from audio policy manager; 1662 mOutput = 0; 1663 audio_io_handle_t output = getOutput_l(); 1664 1665 // if the new IAudioTrack is created, createTrack_l() will modify the 1666 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1667 // It will also delete the strong references on previous IAudioTrack and IMemory 1668 1669 // take the frames that will be lost by track recreation into account in saved position 1670 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1671 mNewPosition = position + mUpdatePeriod; 1672 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1673 result = createTrack_l(mStreamType, 1674 mSampleRate, 1675 mFormat, 1676 mReqFrameCount, // so that frame count never goes down 1677 mFlags, 1678 mSharedBuffer, 1679 output, 1680 position /*epoch*/); 1681 1682 if (result == NO_ERROR) { 1683 // continue playback from last known position, but 1684 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1685 if (mStaticProxy != NULL) { 1686 mLoopPeriod = 0; 1687 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1688 } 1689 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1690 // track destruction have been played? This is critical for SoundPool implementation 1691 // This must be broken, and needs to be tested/debugged. 1692#if 0 1693 // restore write index and set other indexes to reflect empty buffer status 1694 if (!strcmp(from, "start")) { 1695 // Make sure that a client relying on callback events indicating underrun or 1696 // the actual amount of audio frames played (e.g SoundPool) receives them. 1697 if (mSharedBuffer == 0) { 1698 // restart playback even if buffer is not completely filled. 1699 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1700 } 1701 } 1702#endif 1703 if (mState == STATE_ACTIVE) { 1704 result = mAudioTrack->start(); 1705 } 1706 } 1707 if (result != NO_ERROR) { 1708 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1709 // As getOutput was called above and resulted in an output stream to be opened, 1710 // we need to release it. 1711 AudioSystem::releaseOutput(output); 1712 ALOGW("restoreTrack_l() failed status %d", result); 1713 mState = STATE_STOPPED; 1714 } 1715 1716 return result; 1717} 1718 1719status_t AudioTrack::setParameters(const String8& keyValuePairs) 1720{ 1721 AutoMutex lock(mLock); 1722 return mAudioTrack->setParameters(keyValuePairs); 1723} 1724 1725status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1726{ 1727 AutoMutex lock(mLock); 1728 // FIXME not implemented for fast tracks; should use proxy and SSQ 1729 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1730 return INVALID_OPERATION; 1731 } 1732 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1733 return INVALID_OPERATION; 1734 } 1735 status_t status = mAudioTrack->getTimestamp(timestamp); 1736 if (status == NO_ERROR) { 1737 timestamp.mPosition += mProxy->getEpoch(); 1738 } 1739 return status; 1740} 1741 1742String8 AudioTrack::getParameters(const String8& keys) 1743{ 1744 if (mOutput) { 1745 return AudioSystem::getParameters(mOutput, keys); 1746 } else { 1747 return String8::empty(); 1748 } 1749} 1750 1751status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1752{ 1753 1754 const size_t SIZE = 256; 1755 char buffer[SIZE]; 1756 String8 result; 1757 1758 result.append(" AudioTrack::dump\n"); 1759 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1760 mVolume[0], mVolume[1]); 1761 result.append(buffer); 1762 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1763 mChannelCount, mFrameCount); 1764 result.append(buffer); 1765 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1766 result.append(buffer); 1767 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1768 result.append(buffer); 1769 ::write(fd, result.string(), result.size()); 1770 return NO_ERROR; 1771} 1772 1773uint32_t AudioTrack::getUnderrunFrames() const 1774{ 1775 AutoMutex lock(mLock); 1776 return mProxy->getUnderrunFrames(); 1777} 1778 1779// ========================================================================= 1780 1781void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1782{ 1783 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1784 if (audioTrack != 0) { 1785 AutoMutex lock(audioTrack->mLock); 1786 audioTrack->mProxy->binderDied(); 1787 } 1788} 1789 1790// ========================================================================= 1791 1792AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1793 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1794 mIgnoreNextPausedInt(false) 1795{ 1796} 1797 1798AudioTrack::AudioTrackThread::~AudioTrackThread() 1799{ 1800} 1801 1802bool AudioTrack::AudioTrackThread::threadLoop() 1803{ 1804 { 1805 AutoMutex _l(mMyLock); 1806 if (mPaused) { 1807 mMyCond.wait(mMyLock); 1808 // caller will check for exitPending() 1809 return true; 1810 } 1811 if (mIgnoreNextPausedInt) { 1812 mIgnoreNextPausedInt = false; 1813 mPausedInt = false; 1814 } 1815 if (mPausedInt) { 1816 if (mPausedNs > 0) { 1817 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1818 } else { 1819 mMyCond.wait(mMyLock); 1820 } 1821 mPausedInt = false; 1822 return true; 1823 } 1824 } 1825 nsecs_t ns = mReceiver.processAudioBuffer(this); 1826 switch (ns) { 1827 case 0: 1828 return true; 1829 case NS_INACTIVE: 1830 pauseInternal(); 1831 return true; 1832 case NS_NEVER: 1833 return false; 1834 case NS_WHENEVER: 1835 // FIXME increase poll interval, or make event-driven 1836 ns = 1000000000LL; 1837 // fall through 1838 default: 1839 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1840 pauseInternal(ns); 1841 return true; 1842 } 1843} 1844 1845void AudioTrack::AudioTrackThread::requestExit() 1846{ 1847 // must be in this order to avoid a race condition 1848 Thread::requestExit(); 1849 resume(); 1850} 1851 1852void AudioTrack::AudioTrackThread::pause() 1853{ 1854 AutoMutex _l(mMyLock); 1855 mPaused = true; 1856} 1857 1858void AudioTrack::AudioTrackThread::resume() 1859{ 1860 AutoMutex _l(mMyLock); 1861 mIgnoreNextPausedInt = true; 1862 if (mPaused || mPausedInt) { 1863 mPaused = false; 1864 mPausedInt = false; 1865 mMyCond.signal(); 1866 } 1867} 1868 1869void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1870{ 1871 AutoMutex _l(mMyLock); 1872 mPausedInt = true; 1873 mPausedNs = ns; 1874} 1875 1876}; // namespace android 1877