AudioTrack.cpp revision 8ba90326d683b035d99e24db669093e4602a7149
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 int uid) 106 : mStatus(NO_INIT), 107 mIsTimed(false), 108 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 109 mPreviousSchedulingGroup(SP_DEFAULT) 110{ 111 mStatus = set(streamType, sampleRate, format, channelMask, 112 frameCount, flags, cbf, user, notificationFrames, 113 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 114 offloadInfo, uid); 115} 116 117AudioTrack::AudioTrack( 118 audio_stream_type_t streamType, 119 uint32_t sampleRate, 120 audio_format_t format, 121 audio_channel_mask_t channelMask, 122 const sp<IMemory>& sharedBuffer, 123 audio_output_flags_t flags, 124 callback_t cbf, 125 void* user, 126 int notificationFrames, 127 int sessionId, 128 transfer_type transferType, 129 const audio_offload_info_t *offloadInfo, 130 int uid) 131 : mStatus(NO_INIT), 132 mIsTimed(false), 133 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 134 mPreviousSchedulingGroup(SP_DEFAULT) 135{ 136 mStatus = set(streamType, sampleRate, format, channelMask, 137 0 /*frameCount*/, flags, cbf, user, notificationFrames, 138 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); 139} 140 141AudioTrack::~AudioTrack() 142{ 143 if (mStatus == NO_ERROR) { 144 // Make sure that callback function exits in the case where 145 // it is looping on buffer full condition in obtainBuffer(). 146 // Otherwise the callback thread will never exit. 147 stop(); 148 if (mAudioTrackThread != 0) { 149 mProxy->interrupt(); 150 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 151 mAudioTrackThread->requestExitAndWait(); 152 mAudioTrackThread.clear(); 153 } 154 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 155 mAudioTrack.clear(); 156 IPCThreadState::self()->flushCommands(); 157 AudioSystem::releaseAudioSessionId(mSessionId); 158 } 159} 160 161status_t AudioTrack::set( 162 audio_stream_type_t streamType, 163 uint32_t sampleRate, 164 audio_format_t format, 165 audio_channel_mask_t channelMask, 166 int frameCountInt, 167 audio_output_flags_t flags, 168 callback_t cbf, 169 void* user, 170 int notificationFrames, 171 const sp<IMemory>& sharedBuffer, 172 bool threadCanCallJava, 173 int sessionId, 174 transfer_type transferType, 175 const audio_offload_info_t *offloadInfo, 176 int uid) 177{ 178 switch (transferType) { 179 case TRANSFER_DEFAULT: 180 if (sharedBuffer != 0) { 181 transferType = TRANSFER_SHARED; 182 } else if (cbf == NULL || threadCanCallJava) { 183 transferType = TRANSFER_SYNC; 184 } else { 185 transferType = TRANSFER_CALLBACK; 186 } 187 break; 188 case TRANSFER_CALLBACK: 189 if (cbf == NULL || sharedBuffer != 0) { 190 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 191 return BAD_VALUE; 192 } 193 break; 194 case TRANSFER_OBTAIN: 195 case TRANSFER_SYNC: 196 if (sharedBuffer != 0) { 197 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 198 return BAD_VALUE; 199 } 200 break; 201 case TRANSFER_SHARED: 202 if (sharedBuffer == 0) { 203 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 204 return BAD_VALUE; 205 } 206 break; 207 default: 208 ALOGE("Invalid transfer type %d", transferType); 209 return BAD_VALUE; 210 } 211 mTransfer = transferType; 212 213 // FIXME "int" here is legacy and will be replaced by size_t later 214 if (frameCountInt < 0) { 215 ALOGE("Invalid frame count %d", frameCountInt); 216 return BAD_VALUE; 217 } 218 size_t frameCount = frameCountInt; 219 220 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 221 sharedBuffer->size()); 222 223 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 224 225 AutoMutex lock(mLock); 226 227 // invariant that mAudioTrack != 0 is true only after set() returns successfully 228 if (mAudioTrack != 0) { 229 ALOGE("Track already in use"); 230 return INVALID_OPERATION; 231 } 232 233 mOutput = 0; 234 235 // handle default values first. 236 if (streamType == AUDIO_STREAM_DEFAULT) { 237 streamType = AUDIO_STREAM_MUSIC; 238 } 239 240 if (sampleRate == 0) { 241 uint32_t afSampleRate; 242 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 243 return NO_INIT; 244 } 245 sampleRate = afSampleRate; 246 } 247 mSampleRate = sampleRate; 248 249 // these below should probably come from the audioFlinger too... 250 if (format == AUDIO_FORMAT_DEFAULT) { 251 format = AUDIO_FORMAT_PCM_16_BIT; 252 } 253 254 // validate parameters 255 if (!audio_is_valid_format(format)) { 256 ALOGE("Invalid format %d", format); 257 return BAD_VALUE; 258 } 259 260 if (!audio_is_output_channel(channelMask)) { 261 ALOGE("Invalid channel mask %#x", channelMask); 262 return BAD_VALUE; 263 } 264 265 // AudioFlinger does not currently support 8-bit data in shared memory 266 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 267 ALOGE("8-bit data in shared memory is not supported"); 268 return BAD_VALUE; 269 } 270 271 // force direct flag if format is not linear PCM 272 // or offload was requested 273 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 274 || !audio_is_linear_pcm(format)) { 275 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 276 ? "Offload request, forcing to Direct Output" 277 : "Not linear PCM, forcing to Direct Output"); 278 flags = (audio_output_flags_t) 279 // FIXME why can't we allow direct AND fast? 280 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 281 } 282 // only allow deep buffering for music stream type 283 if (streamType != AUDIO_STREAM_MUSIC) { 284 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 285 } 286 287 mChannelMask = channelMask; 288 uint32_t channelCount = popcount(channelMask); 289 mChannelCount = channelCount; 290 291 if (audio_is_linear_pcm(format)) { 292 mFrameSize = channelCount * audio_bytes_per_sample(format); 293 mFrameSizeAF = channelCount * sizeof(int16_t); 294 } else { 295 mFrameSize = sizeof(uint8_t); 296 mFrameSizeAF = sizeof(uint8_t); 297 } 298 299 audio_io_handle_t output = AudioSystem::getOutput( 300 streamType, 301 sampleRate, format, channelMask, 302 flags, 303 offloadInfo); 304 305 if (output == 0) { 306 ALOGE("Could not get audio output for stream type %d", streamType); 307 return BAD_VALUE; 308 } 309 310 mVolume[LEFT] = 1.0f; 311 mVolume[RIGHT] = 1.0f; 312 mSendLevel = 0.0f; 313 mFrameCount = frameCount; 314 mReqFrameCount = frameCount; 315 mNotificationFramesReq = notificationFrames; 316 mNotificationFramesAct = 0; 317 mSessionId = sessionId; 318 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 319 mClientUid = IPCThreadState::self()->getCallingUid(); 320 } else { 321 mClientUid = uid; 322 } 323 mAuxEffectId = 0; 324 mFlags = flags; 325 mCbf = cbf; 326 327 if (cbf != NULL) { 328 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 329 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 330 } 331 332 // create the IAudioTrack 333 status_t status = createTrack_l(streamType, 334 sampleRate, 335 format, 336 frameCount, 337 flags, 338 sharedBuffer, 339 output, 340 0 /*epoch*/); 341 342 if (status != NO_ERROR) { 343 if (mAudioTrackThread != 0) { 344 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 345 mAudioTrackThread->requestExitAndWait(); 346 mAudioTrackThread.clear(); 347 } 348 //Use of direct and offloaded output streams is ref counted by audio policy manager. 349 // As getOutput was called above and resulted in an output stream to be opened, 350 // we need to release it. 351 AudioSystem::releaseOutput(output); 352 return status; 353 } 354 355 mStatus = NO_ERROR; 356 mStreamType = streamType; 357 mFormat = format; 358 mSharedBuffer = sharedBuffer; 359 mState = STATE_STOPPED; 360 mUserData = user; 361 mLoopPeriod = 0; 362 mMarkerPosition = 0; 363 mMarkerReached = false; 364 mNewPosition = 0; 365 mUpdatePeriod = 0; 366 AudioSystem::acquireAudioSessionId(mSessionId); 367 mSequence = 1; 368 mObservedSequence = mSequence; 369 mInUnderrun = false; 370 mOutput = output; 371 372 return NO_ERROR; 373} 374 375// ------------------------------------------------------------------------- 376 377status_t AudioTrack::start() 378{ 379 AutoMutex lock(mLock); 380 381 if (mState == STATE_ACTIVE) { 382 return INVALID_OPERATION; 383 } 384 385 mInUnderrun = true; 386 387 State previousState = mState; 388 if (previousState == STATE_PAUSED_STOPPING) { 389 mState = STATE_STOPPING; 390 } else { 391 mState = STATE_ACTIVE; 392 } 393 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 394 // reset current position as seen by client to 0 395 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 396 // force refresh of remaining frames by processAudioBuffer() as last 397 // write before stop could be partial. 398 mRefreshRemaining = true; 399 } 400 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 401 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 402 403 sp<AudioTrackThread> t = mAudioTrackThread; 404 if (t != 0) { 405 if (previousState == STATE_STOPPING) { 406 mProxy->interrupt(); 407 } else { 408 t->resume(); 409 } 410 } else { 411 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 412 get_sched_policy(0, &mPreviousSchedulingGroup); 413 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 414 } 415 416 status_t status = NO_ERROR; 417 if (!(flags & CBLK_INVALID)) { 418 status = mAudioTrack->start(); 419 if (status == DEAD_OBJECT) { 420 flags |= CBLK_INVALID; 421 } 422 } 423 if (flags & CBLK_INVALID) { 424 status = restoreTrack_l("start"); 425 } 426 427 if (status != NO_ERROR) { 428 ALOGE("start() status %d", status); 429 mState = previousState; 430 if (t != 0) { 431 if (previousState != STATE_STOPPING) { 432 t->pause(); 433 } 434 } else { 435 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 436 set_sched_policy(0, mPreviousSchedulingGroup); 437 } 438 } 439 440 return status; 441} 442 443void AudioTrack::stop() 444{ 445 AutoMutex lock(mLock); 446 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 447 return; 448 } 449 450 if (isOffloaded()) { 451 mState = STATE_STOPPING; 452 } else { 453 mState = STATE_STOPPED; 454 } 455 456 mProxy->interrupt(); 457 mAudioTrack->stop(); 458 // the playback head position will reset to 0, so if a marker is set, we need 459 // to activate it again 460 mMarkerReached = false; 461#if 0 462 // Force flush if a shared buffer is used otherwise audioflinger 463 // will not stop before end of buffer is reached. 464 // It may be needed to make sure that we stop playback, likely in case looping is on. 465 if (mSharedBuffer != 0) { 466 flush_l(); 467 } 468#endif 469 470 sp<AudioTrackThread> t = mAudioTrackThread; 471 if (t != 0) { 472 if (!isOffloaded()) { 473 t->pause(); 474 } 475 } else { 476 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 477 set_sched_policy(0, mPreviousSchedulingGroup); 478 } 479} 480 481bool AudioTrack::stopped() const 482{ 483 AutoMutex lock(mLock); 484 return mState != STATE_ACTIVE; 485} 486 487void AudioTrack::flush() 488{ 489 if (mSharedBuffer != 0) { 490 return; 491 } 492 AutoMutex lock(mLock); 493 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 494 return; 495 } 496 flush_l(); 497} 498 499void AudioTrack::flush_l() 500{ 501 ALOG_ASSERT(mState != STATE_ACTIVE); 502 503 // clear playback marker and periodic update counter 504 mMarkerPosition = 0; 505 mMarkerReached = false; 506 mUpdatePeriod = 0; 507 mRefreshRemaining = true; 508 509 mState = STATE_FLUSHED; 510 if (isOffloaded()) { 511 mProxy->interrupt(); 512 } 513 mProxy->flush(); 514 mAudioTrack->flush(); 515} 516 517void AudioTrack::pause() 518{ 519 AutoMutex lock(mLock); 520 if (mState == STATE_ACTIVE) { 521 mState = STATE_PAUSED; 522 } else if (mState == STATE_STOPPING) { 523 mState = STATE_PAUSED_STOPPING; 524 } else { 525 return; 526 } 527 mProxy->interrupt(); 528 mAudioTrack->pause(); 529} 530 531status_t AudioTrack::setVolume(float left, float right) 532{ 533 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 534 return BAD_VALUE; 535 } 536 537 AutoMutex lock(mLock); 538 mVolume[LEFT] = left; 539 mVolume[RIGHT] = right; 540 541 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 542 543 if (isOffloaded()) { 544 mAudioTrack->signal(); 545 } 546 return NO_ERROR; 547} 548 549status_t AudioTrack::setVolume(float volume) 550{ 551 return setVolume(volume, volume); 552} 553 554status_t AudioTrack::setAuxEffectSendLevel(float level) 555{ 556 if (level < 0.0f || level > 1.0f) { 557 return BAD_VALUE; 558 } 559 560 AutoMutex lock(mLock); 561 mSendLevel = level; 562 mProxy->setSendLevel(level); 563 564 return NO_ERROR; 565} 566 567void AudioTrack::getAuxEffectSendLevel(float* level) const 568{ 569 if (level != NULL) { 570 *level = mSendLevel; 571 } 572} 573 574status_t AudioTrack::setSampleRate(uint32_t rate) 575{ 576 if (mIsTimed || isOffloaded()) { 577 return INVALID_OPERATION; 578 } 579 580 uint32_t afSamplingRate; 581 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 582 return NO_INIT; 583 } 584 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 585 if (rate == 0 || rate > afSamplingRate*2 ) { 586 return BAD_VALUE; 587 } 588 589 AutoMutex lock(mLock); 590 mSampleRate = rate; 591 mProxy->setSampleRate(rate); 592 593 return NO_ERROR; 594} 595 596uint32_t AudioTrack::getSampleRate() const 597{ 598 if (mIsTimed) { 599 return 0; 600 } 601 602 AutoMutex lock(mLock); 603 return mSampleRate; 604} 605 606status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 607{ 608 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 609 return INVALID_OPERATION; 610 } 611 612 if (loopCount == 0) { 613 ; 614 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 615 loopEnd - loopStart >= MIN_LOOP) { 616 ; 617 } else { 618 return BAD_VALUE; 619 } 620 621 AutoMutex lock(mLock); 622 // See setPosition() regarding setting parameters such as loop points or position while active 623 if (mState == STATE_ACTIVE) { 624 return INVALID_OPERATION; 625 } 626 setLoop_l(loopStart, loopEnd, loopCount); 627 return NO_ERROR; 628} 629 630void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 631{ 632 // FIXME If setting a loop also sets position to start of loop, then 633 // this is correct. Otherwise it should be removed. 634 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 635 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 636 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 637} 638 639status_t AudioTrack::setMarkerPosition(uint32_t marker) 640{ 641 // The only purpose of setting marker position is to get a callback 642 if (mCbf == NULL || isOffloaded()) { 643 return INVALID_OPERATION; 644 } 645 646 AutoMutex lock(mLock); 647 mMarkerPosition = marker; 648 mMarkerReached = false; 649 650 return NO_ERROR; 651} 652 653status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 654{ 655 if (isOffloaded()) { 656 return INVALID_OPERATION; 657 } 658 if (marker == NULL) { 659 return BAD_VALUE; 660 } 661 662 AutoMutex lock(mLock); 663 *marker = mMarkerPosition; 664 665 return NO_ERROR; 666} 667 668status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 669{ 670 // The only purpose of setting position update period is to get a callback 671 if (mCbf == NULL || isOffloaded()) { 672 return INVALID_OPERATION; 673 } 674 675 AutoMutex lock(mLock); 676 mNewPosition = mProxy->getPosition() + updatePeriod; 677 mUpdatePeriod = updatePeriod; 678 return NO_ERROR; 679} 680 681status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 682{ 683 if (isOffloaded()) { 684 return INVALID_OPERATION; 685 } 686 if (updatePeriod == NULL) { 687 return BAD_VALUE; 688 } 689 690 AutoMutex lock(mLock); 691 *updatePeriod = mUpdatePeriod; 692 693 return NO_ERROR; 694} 695 696status_t AudioTrack::setPosition(uint32_t position) 697{ 698 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 699 return INVALID_OPERATION; 700 } 701 if (position > mFrameCount) { 702 return BAD_VALUE; 703 } 704 705 AutoMutex lock(mLock); 706 // Currently we require that the player is inactive before setting parameters such as position 707 // or loop points. Otherwise, there could be a race condition: the application could read the 708 // current position, compute a new position or loop parameters, and then set that position or 709 // loop parameters but it would do the "wrong" thing since the position has continued to advance 710 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 711 // to specify how it wants to handle such scenarios. 712 if (mState == STATE_ACTIVE) { 713 return INVALID_OPERATION; 714 } 715 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 716 mLoopPeriod = 0; 717 // FIXME Check whether loops and setting position are incompatible in old code. 718 // If we use setLoop for both purposes we lose the capability to set the position while looping. 719 mStaticProxy->setLoop(position, mFrameCount, 0); 720 721 return NO_ERROR; 722} 723 724status_t AudioTrack::getPosition(uint32_t *position) const 725{ 726 if (position == NULL) { 727 return BAD_VALUE; 728 } 729 730 AutoMutex lock(mLock); 731 if (isOffloaded()) { 732 uint32_t dspFrames = 0; 733 734 if (mOutput != 0) { 735 uint32_t halFrames; 736 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 737 } 738 *position = dspFrames; 739 } else { 740 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 741 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 742 mProxy->getPosition(); 743 } 744 return NO_ERROR; 745} 746 747status_t AudioTrack::getBufferPosition(size_t *position) 748{ 749 if (mSharedBuffer == 0 || mIsTimed) { 750 return INVALID_OPERATION; 751 } 752 if (position == NULL) { 753 return BAD_VALUE; 754 } 755 756 AutoMutex lock(mLock); 757 *position = mStaticProxy->getBufferPosition(); 758 return NO_ERROR; 759} 760 761status_t AudioTrack::reload() 762{ 763 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 764 return INVALID_OPERATION; 765 } 766 767 AutoMutex lock(mLock); 768 // See setPosition() regarding setting parameters such as loop points or position while active 769 if (mState == STATE_ACTIVE) { 770 return INVALID_OPERATION; 771 } 772 mNewPosition = mUpdatePeriod; 773 mLoopPeriod = 0; 774 // FIXME The new code cannot reload while keeping a loop specified. 775 // Need to check how the old code handled this, and whether it's a significant change. 776 mStaticProxy->setLoop(0, mFrameCount, 0); 777 return NO_ERROR; 778} 779 780audio_io_handle_t AudioTrack::getOutput() 781{ 782 AutoMutex lock(mLock); 783 return mOutput; 784} 785 786// must be called with mLock held 787audio_io_handle_t AudioTrack::getOutput_l() 788{ 789 if (mOutput) { 790 return mOutput; 791 } else { 792 return AudioSystem::getOutput(mStreamType, 793 mSampleRate, mFormat, mChannelMask, mFlags); 794 } 795} 796 797status_t AudioTrack::attachAuxEffect(int effectId) 798{ 799 AutoMutex lock(mLock); 800 status_t status = mAudioTrack->attachAuxEffect(effectId); 801 if (status == NO_ERROR) { 802 mAuxEffectId = effectId; 803 } 804 return status; 805} 806 807// ------------------------------------------------------------------------- 808 809// must be called with mLock held 810status_t AudioTrack::createTrack_l( 811 audio_stream_type_t streamType, 812 uint32_t sampleRate, 813 audio_format_t format, 814 size_t frameCount, 815 audio_output_flags_t flags, 816 const sp<IMemory>& sharedBuffer, 817 audio_io_handle_t output, 818 size_t epoch) 819{ 820 status_t status; 821 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 822 if (audioFlinger == 0) { 823 ALOGE("Could not get audioflinger"); 824 return NO_INIT; 825 } 826 827 // Not all of these values are needed under all conditions, but it is easier to get them all 828 829 uint32_t afLatency; 830 status = AudioSystem::getLatency(output, streamType, &afLatency); 831 if (status != NO_ERROR) { 832 ALOGE("getLatency(%d) failed status %d", output, status); 833 return NO_INIT; 834 } 835 836 size_t afFrameCount; 837 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 838 if (status != NO_ERROR) { 839 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 840 return NO_INIT; 841 } 842 843 uint32_t afSampleRate; 844 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 845 if (status != NO_ERROR) { 846 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 847 return NO_INIT; 848 } 849 850 // Client decides whether the track is TIMED (see below), but can only express a preference 851 // for FAST. Server will perform additional tests. 852 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 853 // either of these use cases: 854 // use case 1: shared buffer 855 (sharedBuffer != 0) || 856 // use case 2: callback handler 857 (mCbf != NULL))) { 858 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 859 // once denied, do not request again if IAudioTrack is re-created 860 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 861 mFlags = flags; 862 } 863 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 864 865 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 866 // n = 1 fast track; nBuffering is ignored 867 // n = 2 normal track, no sample rate conversion 868 // n = 3 normal track, with sample rate conversion 869 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 870 // n > 3 very high latency or very small notification interval; nBuffering is ignored 871 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 872 873 mNotificationFramesAct = mNotificationFramesReq; 874 875 if (!audio_is_linear_pcm(format)) { 876 877 if (sharedBuffer != 0) { 878 // Same comment as below about ignoring frameCount parameter for set() 879 frameCount = sharedBuffer->size(); 880 } else if (frameCount == 0) { 881 frameCount = afFrameCount; 882 } 883 if (mNotificationFramesAct != frameCount) { 884 mNotificationFramesAct = frameCount; 885 } 886 } else if (sharedBuffer != 0) { 887 888 // Ensure that buffer alignment matches channel count 889 // 8-bit data in shared memory is not currently supported by AudioFlinger 890 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 891 if (mChannelCount > 1) { 892 // More than 2 channels does not require stronger alignment than stereo 893 alignment <<= 1; 894 } 895 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 896 ALOGE("Invalid buffer alignment: address %p, channel count %u", 897 sharedBuffer->pointer(), mChannelCount); 898 return BAD_VALUE; 899 } 900 901 // When initializing a shared buffer AudioTrack via constructors, 902 // there's no frameCount parameter. 903 // But when initializing a shared buffer AudioTrack via set(), 904 // there _is_ a frameCount parameter. We silently ignore it. 905 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 906 907 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 908 909 // FIXME move these calculations and associated checks to server 910 911 // Ensure that buffer depth covers at least audio hardware latency 912 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 913 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 914 afFrameCount, minBufCount, afSampleRate, afLatency); 915 if (minBufCount <= nBuffering) { 916 minBufCount = nBuffering; 917 } 918 919 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 920 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 921 ", afLatency=%d", 922 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 923 924 if (frameCount == 0) { 925 frameCount = minFrameCount; 926 } else if (frameCount < minFrameCount) { 927 // not ALOGW because it happens all the time when playing key clicks over A2DP 928 ALOGV("Minimum buffer size corrected from %d to %d", 929 frameCount, minFrameCount); 930 frameCount = minFrameCount; 931 } 932 // Make sure that application is notified with sufficient margin before underrun 933 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 934 mNotificationFramesAct = frameCount/nBuffering; 935 } 936 937 } else { 938 // For fast tracks, the frame count calculations and checks are done by server 939 } 940 941 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 942 if (mIsTimed) { 943 trackFlags |= IAudioFlinger::TRACK_TIMED; 944 } 945 946 pid_t tid = -1; 947 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 948 trackFlags |= IAudioFlinger::TRACK_FAST; 949 if (mAudioTrackThread != 0) { 950 tid = mAudioTrackThread->getTid(); 951 } 952 } 953 954 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 955 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 956 } 957 958 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 959 sampleRate, 960 // AudioFlinger only sees 16-bit PCM 961 format == AUDIO_FORMAT_PCM_8_BIT ? 962 AUDIO_FORMAT_PCM_16_BIT : format, 963 mChannelMask, 964 frameCount, 965 &trackFlags, 966 sharedBuffer, 967 output, 968 tid, 969 &mSessionId, 970 mName, 971 mClientUid, 972 &status); 973 974 if (track == 0) { 975 ALOGE("AudioFlinger could not create track, status: %d", status); 976 return status; 977 } 978 sp<IMemory> iMem = track->getCblk(); 979 if (iMem == 0) { 980 ALOGE("Could not get control block"); 981 return NO_INIT; 982 } 983 // invariant that mAudioTrack != 0 is true only after set() returns successfully 984 if (mAudioTrack != 0) { 985 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 986 mDeathNotifier.clear(); 987 } 988 mAudioTrack = track; 989 mCblkMemory = iMem; 990 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 991 mCblk = cblk; 992 size_t temp = cblk->frameCount_; 993 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 994 // In current design, AudioTrack client checks and ensures frame count validity before 995 // passing it to AudioFlinger so AudioFlinger should not return a different value except 996 // for fast track as it uses a special method of assigning frame count. 997 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 998 } 999 frameCount = temp; 1000 mAwaitBoost = false; 1001 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1002 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1003 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1004 mAwaitBoost = true; 1005 if (sharedBuffer == 0) { 1006 // double-buffering is not required for fast tracks, due to tighter scheduling 1007 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 1008 mNotificationFramesAct = frameCount; 1009 } 1010 } 1011 } else { 1012 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1013 // once denied, do not request again if IAudioTrack is re-created 1014 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1015 mFlags = flags; 1016 if (sharedBuffer == 0) { 1017 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1018 mNotificationFramesAct = frameCount/nBuffering; 1019 } 1020 } 1021 } 1022 } 1023 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1024 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1025 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1026 } else { 1027 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1028 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1029 mFlags = flags; 1030 return NO_INIT; 1031 } 1032 } 1033 1034 mRefreshRemaining = true; 1035 1036 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1037 // is the value of pointer() for the shared buffer, otherwise buffers points 1038 // immediately after the control block. This address is for the mapping within client 1039 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1040 void* buffers; 1041 if (sharedBuffer == 0) { 1042 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1043 } else { 1044 buffers = sharedBuffer->pointer(); 1045 } 1046 1047 mAudioTrack->attachAuxEffect(mAuxEffectId); 1048 // FIXME don't believe this lie 1049 mLatency = afLatency + (1000*frameCount) / sampleRate; 1050 mFrameCount = frameCount; 1051 // If IAudioTrack is re-created, don't let the requested frameCount 1052 // decrease. This can confuse clients that cache frameCount(). 1053 if (frameCount > mReqFrameCount) { 1054 mReqFrameCount = frameCount; 1055 } 1056 1057 // update proxy 1058 if (sharedBuffer == 0) { 1059 mStaticProxy.clear(); 1060 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1061 } else { 1062 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1063 mProxy = mStaticProxy; 1064 } 1065 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1066 uint16_t(mVolume[LEFT] * 0x1000)); 1067 mProxy->setSendLevel(mSendLevel); 1068 mProxy->setSampleRate(mSampleRate); 1069 mProxy->setEpoch(epoch); 1070 mProxy->setMinimum(mNotificationFramesAct); 1071 1072 mDeathNotifier = new DeathNotifier(this); 1073 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1074 1075 return NO_ERROR; 1076} 1077 1078status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1079{ 1080 if (audioBuffer == NULL) { 1081 return BAD_VALUE; 1082 } 1083 if (mTransfer != TRANSFER_OBTAIN) { 1084 audioBuffer->frameCount = 0; 1085 audioBuffer->size = 0; 1086 audioBuffer->raw = NULL; 1087 return INVALID_OPERATION; 1088 } 1089 1090 const struct timespec *requested; 1091 if (waitCount == -1) { 1092 requested = &ClientProxy::kForever; 1093 } else if (waitCount == 0) { 1094 requested = &ClientProxy::kNonBlocking; 1095 } else if (waitCount > 0) { 1096 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1097 struct timespec timeout; 1098 timeout.tv_sec = ms / 1000; 1099 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1100 requested = &timeout; 1101 } else { 1102 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1103 requested = NULL; 1104 } 1105 return obtainBuffer(audioBuffer, requested); 1106} 1107 1108status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1109 struct timespec *elapsed, size_t *nonContig) 1110{ 1111 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1112 uint32_t oldSequence = 0; 1113 uint32_t newSequence; 1114 1115 Proxy::Buffer buffer; 1116 status_t status = NO_ERROR; 1117 1118 static const int32_t kMaxTries = 5; 1119 int32_t tryCounter = kMaxTries; 1120 1121 do { 1122 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1123 // keep them from going away if another thread re-creates the track during obtainBuffer() 1124 sp<AudioTrackClientProxy> proxy; 1125 sp<IMemory> iMem; 1126 1127 { // start of lock scope 1128 AutoMutex lock(mLock); 1129 1130 newSequence = mSequence; 1131 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1132 if (status == DEAD_OBJECT) { 1133 // re-create track, unless someone else has already done so 1134 if (newSequence == oldSequence) { 1135 status = restoreTrack_l("obtainBuffer"); 1136 if (status != NO_ERROR) { 1137 buffer.mFrameCount = 0; 1138 buffer.mRaw = NULL; 1139 buffer.mNonContig = 0; 1140 break; 1141 } 1142 } 1143 } 1144 oldSequence = newSequence; 1145 1146 // Keep the extra references 1147 proxy = mProxy; 1148 iMem = mCblkMemory; 1149 1150 if (mState == STATE_STOPPING) { 1151 status = -EINTR; 1152 buffer.mFrameCount = 0; 1153 buffer.mRaw = NULL; 1154 buffer.mNonContig = 0; 1155 break; 1156 } 1157 1158 // Non-blocking if track is stopped or paused 1159 if (mState != STATE_ACTIVE) { 1160 requested = &ClientProxy::kNonBlocking; 1161 } 1162 1163 } // end of lock scope 1164 1165 buffer.mFrameCount = audioBuffer->frameCount; 1166 // FIXME starts the requested timeout and elapsed over from scratch 1167 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1168 1169 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1170 1171 audioBuffer->frameCount = buffer.mFrameCount; 1172 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1173 audioBuffer->raw = buffer.mRaw; 1174 if (nonContig != NULL) { 1175 *nonContig = buffer.mNonContig; 1176 } 1177 return status; 1178} 1179 1180void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1181{ 1182 if (mTransfer == TRANSFER_SHARED) { 1183 return; 1184 } 1185 1186 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1187 if (stepCount == 0) { 1188 return; 1189 } 1190 1191 Proxy::Buffer buffer; 1192 buffer.mFrameCount = stepCount; 1193 buffer.mRaw = audioBuffer->raw; 1194 1195 AutoMutex lock(mLock); 1196 mInUnderrun = false; 1197 mProxy->releaseBuffer(&buffer); 1198 1199 // restart track if it was disabled by audioflinger due to previous underrun 1200 if (mState == STATE_ACTIVE) { 1201 audio_track_cblk_t* cblk = mCblk; 1202 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1203 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1204 this, mName.string()); 1205 // FIXME ignoring status 1206 mAudioTrack->start(); 1207 } 1208 } 1209} 1210 1211// ------------------------------------------------------------------------- 1212 1213ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1214{ 1215 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1216 return INVALID_OPERATION; 1217 } 1218 1219 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1220 // Sanity-check: user is most-likely passing an error code, and it would 1221 // make the return value ambiguous (actualSize vs error). 1222 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1223 return BAD_VALUE; 1224 } 1225 1226 size_t written = 0; 1227 Buffer audioBuffer; 1228 1229 while (userSize >= mFrameSize) { 1230 audioBuffer.frameCount = userSize / mFrameSize; 1231 1232 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1233 if (err < 0) { 1234 if (written > 0) { 1235 break; 1236 } 1237 return ssize_t(err); 1238 } 1239 1240 size_t toWrite; 1241 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1242 // Divide capacity by 2 to take expansion into account 1243 toWrite = audioBuffer.size >> 1; 1244 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1245 } else { 1246 toWrite = audioBuffer.size; 1247 memcpy(audioBuffer.i8, buffer, toWrite); 1248 } 1249 buffer = ((const char *) buffer) + toWrite; 1250 userSize -= toWrite; 1251 written += toWrite; 1252 1253 releaseBuffer(&audioBuffer); 1254 } 1255 1256 return written; 1257} 1258 1259// ------------------------------------------------------------------------- 1260 1261TimedAudioTrack::TimedAudioTrack() { 1262 mIsTimed = true; 1263} 1264 1265status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1266{ 1267 AutoMutex lock(mLock); 1268 status_t result = UNKNOWN_ERROR; 1269 1270#if 1 1271 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1272 // while we are accessing the cblk 1273 sp<IAudioTrack> audioTrack = mAudioTrack; 1274 sp<IMemory> iMem = mCblkMemory; 1275#endif 1276 1277 // If the track is not invalid already, try to allocate a buffer. alloc 1278 // fails indicating that the server is dead, flag the track as invalid so 1279 // we can attempt to restore in just a bit. 1280 audio_track_cblk_t* cblk = mCblk; 1281 if (!(cblk->mFlags & CBLK_INVALID)) { 1282 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1283 if (result == DEAD_OBJECT) { 1284 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1285 } 1286 } 1287 1288 // If the track is invalid at this point, attempt to restore it. and try the 1289 // allocation one more time. 1290 if (cblk->mFlags & CBLK_INVALID) { 1291 result = restoreTrack_l("allocateTimedBuffer"); 1292 1293 if (result == NO_ERROR) { 1294 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1295 } 1296 } 1297 1298 return result; 1299} 1300 1301status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1302 int64_t pts) 1303{ 1304 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1305 { 1306 AutoMutex lock(mLock); 1307 audio_track_cblk_t* cblk = mCblk; 1308 // restart track if it was disabled by audioflinger due to previous underrun 1309 if (buffer->size() != 0 && status == NO_ERROR && 1310 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1311 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1312 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1313 // FIXME ignoring status 1314 mAudioTrack->start(); 1315 } 1316 } 1317 return status; 1318} 1319 1320status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1321 TargetTimeline target) 1322{ 1323 return mAudioTrack->setMediaTimeTransform(xform, target); 1324} 1325 1326// ------------------------------------------------------------------------- 1327 1328nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1329{ 1330 // Currently the AudioTrack thread is not created if there are no callbacks. 1331 // Would it ever make sense to run the thread, even without callbacks? 1332 // If so, then replace this by checks at each use for mCbf != NULL. 1333 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1334 1335 mLock.lock(); 1336 if (mAwaitBoost) { 1337 mAwaitBoost = false; 1338 mLock.unlock(); 1339 static const int32_t kMaxTries = 5; 1340 int32_t tryCounter = kMaxTries; 1341 uint32_t pollUs = 10000; 1342 do { 1343 int policy = sched_getscheduler(0); 1344 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1345 break; 1346 } 1347 usleep(pollUs); 1348 pollUs <<= 1; 1349 } while (tryCounter-- > 0); 1350 if (tryCounter < 0) { 1351 ALOGE("did not receive expected priority boost on time"); 1352 } 1353 // Run again immediately 1354 return 0; 1355 } 1356 1357 // Can only reference mCblk while locked 1358 int32_t flags = android_atomic_and( 1359 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1360 1361 // Check for track invalidation 1362 if (flags & CBLK_INVALID) { 1363 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1364 // AudioSystem cache. We should not exit here but after calling the callback so 1365 // that the upper layers can recreate the track 1366 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1367 status_t status = restoreTrack_l("processAudioBuffer"); 1368 mLock.unlock(); 1369 // Run again immediately, but with a new IAudioTrack 1370 return 0; 1371 } 1372 } 1373 1374 bool waitStreamEnd = mState == STATE_STOPPING; 1375 bool active = mState == STATE_ACTIVE; 1376 1377 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1378 bool newUnderrun = false; 1379 if (flags & CBLK_UNDERRUN) { 1380#if 0 1381 // Currently in shared buffer mode, when the server reaches the end of buffer, 1382 // the track stays active in continuous underrun state. It's up to the application 1383 // to pause or stop the track, or set the position to a new offset within buffer. 1384 // This was some experimental code to auto-pause on underrun. Keeping it here 1385 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1386 if (mTransfer == TRANSFER_SHARED) { 1387 mState = STATE_PAUSED; 1388 active = false; 1389 } 1390#endif 1391 if (!mInUnderrun) { 1392 mInUnderrun = true; 1393 newUnderrun = true; 1394 } 1395 } 1396 1397 // Get current position of server 1398 size_t position = mProxy->getPosition(); 1399 1400 // Manage marker callback 1401 bool markerReached = false; 1402 size_t markerPosition = mMarkerPosition; 1403 // FIXME fails for wraparound, need 64 bits 1404 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1405 mMarkerReached = markerReached = true; 1406 } 1407 1408 // Determine number of new position callback(s) that will be needed, while locked 1409 size_t newPosCount = 0; 1410 size_t newPosition = mNewPosition; 1411 size_t updatePeriod = mUpdatePeriod; 1412 // FIXME fails for wraparound, need 64 bits 1413 if (updatePeriod > 0 && position >= newPosition) { 1414 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1415 mNewPosition += updatePeriod * newPosCount; 1416 } 1417 1418 // Cache other fields that will be needed soon 1419 uint32_t loopPeriod = mLoopPeriod; 1420 uint32_t sampleRate = mSampleRate; 1421 size_t notificationFrames = mNotificationFramesAct; 1422 if (mRefreshRemaining) { 1423 mRefreshRemaining = false; 1424 mRemainingFrames = notificationFrames; 1425 mRetryOnPartialBuffer = false; 1426 } 1427 size_t misalignment = mProxy->getMisalignment(); 1428 uint32_t sequence = mSequence; 1429 1430 // These fields don't need to be cached, because they are assigned only by set(): 1431 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1432 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1433 1434 mLock.unlock(); 1435 1436 if (waitStreamEnd) { 1437 AutoMutex lock(mLock); 1438 1439 sp<AudioTrackClientProxy> proxy = mProxy; 1440 sp<IMemory> iMem = mCblkMemory; 1441 1442 struct timespec timeout; 1443 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1444 timeout.tv_nsec = 0; 1445 1446 mLock.unlock(); 1447 status_t status = mProxy->waitStreamEndDone(&timeout); 1448 mLock.lock(); 1449 switch (status) { 1450 case NO_ERROR: 1451 case DEAD_OBJECT: 1452 case TIMED_OUT: 1453 mLock.unlock(); 1454 mCbf(EVENT_STREAM_END, mUserData, NULL); 1455 mLock.lock(); 1456 if (mState == STATE_STOPPING) { 1457 mState = STATE_STOPPED; 1458 if (status != DEAD_OBJECT) { 1459 return NS_INACTIVE; 1460 } 1461 } 1462 return 0; 1463 default: 1464 return 0; 1465 } 1466 } 1467 1468 // perform callbacks while unlocked 1469 if (newUnderrun) { 1470 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1471 } 1472 // FIXME we will miss loops if loop cycle was signaled several times since last call 1473 // to processAudioBuffer() 1474 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1475 mCbf(EVENT_LOOP_END, mUserData, NULL); 1476 } 1477 if (flags & CBLK_BUFFER_END) { 1478 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1479 } 1480 if (markerReached) { 1481 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1482 } 1483 while (newPosCount > 0) { 1484 size_t temp = newPosition; 1485 mCbf(EVENT_NEW_POS, mUserData, &temp); 1486 newPosition += updatePeriod; 1487 newPosCount--; 1488 } 1489 1490 if (mObservedSequence != sequence) { 1491 mObservedSequence = sequence; 1492 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1493 // for offloaded tracks, just wait for the upper layers to recreate the track 1494 if (isOffloaded()) { 1495 return NS_INACTIVE; 1496 } 1497 } 1498 1499 // if inactive, then don't run me again until re-started 1500 if (!active) { 1501 return NS_INACTIVE; 1502 } 1503 1504 // Compute the estimated time until the next timed event (position, markers, loops) 1505 // FIXME only for non-compressed audio 1506 uint32_t minFrames = ~0; 1507 if (!markerReached && position < markerPosition) { 1508 minFrames = markerPosition - position; 1509 } 1510 if (loopPeriod > 0 && loopPeriod < minFrames) { 1511 minFrames = loopPeriod; 1512 } 1513 if (updatePeriod > 0 && updatePeriod < minFrames) { 1514 minFrames = updatePeriod; 1515 } 1516 1517 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1518 static const uint32_t kPoll = 0; 1519 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1520 minFrames = kPoll * notificationFrames; 1521 } 1522 1523 // Convert frame units to time units 1524 nsecs_t ns = NS_WHENEVER; 1525 if (minFrames != (uint32_t) ~0) { 1526 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1527 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1528 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1529 } 1530 1531 // If not supplying data by EVENT_MORE_DATA, then we're done 1532 if (mTransfer != TRANSFER_CALLBACK) { 1533 return ns; 1534 } 1535 1536 struct timespec timeout; 1537 const struct timespec *requested = &ClientProxy::kForever; 1538 if (ns != NS_WHENEVER) { 1539 timeout.tv_sec = ns / 1000000000LL; 1540 timeout.tv_nsec = ns % 1000000000LL; 1541 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1542 requested = &timeout; 1543 } 1544 1545 while (mRemainingFrames > 0) { 1546 1547 Buffer audioBuffer; 1548 audioBuffer.frameCount = mRemainingFrames; 1549 size_t nonContig; 1550 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1551 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1552 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1553 requested = &ClientProxy::kNonBlocking; 1554 size_t avail = audioBuffer.frameCount + nonContig; 1555 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1556 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1557 if (err != NO_ERROR) { 1558 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1559 (isOffloaded() && (err == DEAD_OBJECT))) { 1560 return 0; 1561 } 1562 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1563 return NS_NEVER; 1564 } 1565 1566 if (mRetryOnPartialBuffer && !isOffloaded()) { 1567 mRetryOnPartialBuffer = false; 1568 if (avail < mRemainingFrames) { 1569 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1570 if (ns < 0 || myns < ns) { 1571 ns = myns; 1572 } 1573 return ns; 1574 } 1575 } 1576 1577 // Divide buffer size by 2 to take into account the expansion 1578 // due to 8 to 16 bit conversion: the callback must fill only half 1579 // of the destination buffer 1580 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1581 audioBuffer.size >>= 1; 1582 } 1583 1584 size_t reqSize = audioBuffer.size; 1585 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1586 size_t writtenSize = audioBuffer.size; 1587 size_t writtenFrames = writtenSize / mFrameSize; 1588 1589 // Sanity check on returned size 1590 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1591 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1592 reqSize, (int) writtenSize); 1593 return NS_NEVER; 1594 } 1595 1596 if (writtenSize == 0) { 1597 // The callback is done filling buffers 1598 // Keep this thread going to handle timed events and 1599 // still try to get more data in intervals of WAIT_PERIOD_MS 1600 // but don't just loop and block the CPU, so wait 1601 return WAIT_PERIOD_MS * 1000000LL; 1602 } 1603 1604 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1605 // 8 to 16 bit conversion, note that source and destination are the same address 1606 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1607 audioBuffer.size <<= 1; 1608 } 1609 1610 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1611 audioBuffer.frameCount = releasedFrames; 1612 mRemainingFrames -= releasedFrames; 1613 if (misalignment >= releasedFrames) { 1614 misalignment -= releasedFrames; 1615 } else { 1616 misalignment = 0; 1617 } 1618 1619 releaseBuffer(&audioBuffer); 1620 1621 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1622 // if callback doesn't like to accept the full chunk 1623 if (writtenSize < reqSize) { 1624 continue; 1625 } 1626 1627 // There could be enough non-contiguous frames available to satisfy the remaining request 1628 if (mRemainingFrames <= nonContig) { 1629 continue; 1630 } 1631 1632#if 0 1633 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1634 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1635 // that total to a sum == notificationFrames. 1636 if (0 < misalignment && misalignment <= mRemainingFrames) { 1637 mRemainingFrames = misalignment; 1638 return (mRemainingFrames * 1100000000LL) / sampleRate; 1639 } 1640#endif 1641 1642 } 1643 mRemainingFrames = notificationFrames; 1644 mRetryOnPartialBuffer = true; 1645 1646 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1647 return 0; 1648} 1649 1650status_t AudioTrack::restoreTrack_l(const char *from) 1651{ 1652 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1653 isOffloaded() ? "Offloaded" : "PCM", from); 1654 ++mSequence; 1655 status_t result; 1656 1657 // refresh the audio configuration cache in this process to make sure we get new 1658 // output parameters in getOutput_l() and createTrack_l() 1659 AudioSystem::clearAudioConfigCache(); 1660 1661 if (isOffloaded()) { 1662 return DEAD_OBJECT; 1663 } 1664 1665 // force new output query from audio policy manager; 1666 mOutput = 0; 1667 audio_io_handle_t output = getOutput_l(); 1668 1669 // if the new IAudioTrack is created, createTrack_l() will modify the 1670 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1671 // It will also delete the strong references on previous IAudioTrack and IMemory 1672 1673 // take the frames that will be lost by track recreation into account in saved position 1674 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1675 mNewPosition = position + mUpdatePeriod; 1676 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1677 result = createTrack_l(mStreamType, 1678 mSampleRate, 1679 mFormat, 1680 mReqFrameCount, // so that frame count never goes down 1681 mFlags, 1682 mSharedBuffer, 1683 output, 1684 position /*epoch*/); 1685 1686 if (result == NO_ERROR) { 1687 // continue playback from last known position, but 1688 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1689 if (mStaticProxy != NULL) { 1690 mLoopPeriod = 0; 1691 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1692 } 1693 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1694 // track destruction have been played? This is critical for SoundPool implementation 1695 // This must be broken, and needs to be tested/debugged. 1696#if 0 1697 // restore write index and set other indexes to reflect empty buffer status 1698 if (!strcmp(from, "start")) { 1699 // Make sure that a client relying on callback events indicating underrun or 1700 // the actual amount of audio frames played (e.g SoundPool) receives them. 1701 if (mSharedBuffer == 0) { 1702 // restart playback even if buffer is not completely filled. 1703 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1704 } 1705 } 1706#endif 1707 if (mState == STATE_ACTIVE) { 1708 result = mAudioTrack->start(); 1709 } 1710 } 1711 if (result != NO_ERROR) { 1712 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1713 // As getOutput was called above and resulted in an output stream to be opened, 1714 // we need to release it. 1715 AudioSystem::releaseOutput(output); 1716 ALOGW("restoreTrack_l() failed status %d", result); 1717 mState = STATE_STOPPED; 1718 } 1719 1720 return result; 1721} 1722 1723status_t AudioTrack::setParameters(const String8& keyValuePairs) 1724{ 1725 AutoMutex lock(mLock); 1726 return mAudioTrack->setParameters(keyValuePairs); 1727} 1728 1729status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1730{ 1731 AutoMutex lock(mLock); 1732 // FIXME not implemented for fast tracks; should use proxy and SSQ 1733 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1734 return INVALID_OPERATION; 1735 } 1736 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1737 return INVALID_OPERATION; 1738 } 1739 status_t status = mAudioTrack->getTimestamp(timestamp); 1740 if (status == NO_ERROR) { 1741 timestamp.mPosition += mProxy->getEpoch(); 1742 } 1743 return status; 1744} 1745 1746String8 AudioTrack::getParameters(const String8& keys) 1747{ 1748 if (mOutput) { 1749 return AudioSystem::getParameters(mOutput, keys); 1750 } else { 1751 return String8::empty(); 1752 } 1753} 1754 1755status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1756{ 1757 1758 const size_t SIZE = 256; 1759 char buffer[SIZE]; 1760 String8 result; 1761 1762 result.append(" AudioTrack::dump\n"); 1763 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1764 mVolume[0], mVolume[1]); 1765 result.append(buffer); 1766 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1767 mChannelCount, mFrameCount); 1768 result.append(buffer); 1769 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1770 result.append(buffer); 1771 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1772 result.append(buffer); 1773 ::write(fd, result.string(), result.size()); 1774 return NO_ERROR; 1775} 1776 1777uint32_t AudioTrack::getUnderrunFrames() const 1778{ 1779 AutoMutex lock(mLock); 1780 return mProxy->getUnderrunFrames(); 1781} 1782 1783// ========================================================================= 1784 1785void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1786{ 1787 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1788 if (audioTrack != 0) { 1789 AutoMutex lock(audioTrack->mLock); 1790 audioTrack->mProxy->binderDied(); 1791 } 1792} 1793 1794// ========================================================================= 1795 1796AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1797 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1798 mIgnoreNextPausedInt(false) 1799{ 1800} 1801 1802AudioTrack::AudioTrackThread::~AudioTrackThread() 1803{ 1804} 1805 1806bool AudioTrack::AudioTrackThread::threadLoop() 1807{ 1808 { 1809 AutoMutex _l(mMyLock); 1810 if (mPaused) { 1811 mMyCond.wait(mMyLock); 1812 // caller will check for exitPending() 1813 return true; 1814 } 1815 if (mIgnoreNextPausedInt) { 1816 mIgnoreNextPausedInt = false; 1817 mPausedInt = false; 1818 } 1819 if (mPausedInt) { 1820 if (mPausedNs > 0) { 1821 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1822 } else { 1823 mMyCond.wait(mMyLock); 1824 } 1825 mPausedInt = false; 1826 return true; 1827 } 1828 } 1829 nsecs_t ns = mReceiver.processAudioBuffer(this); 1830 switch (ns) { 1831 case 0: 1832 return true; 1833 case NS_INACTIVE: 1834 pauseInternal(); 1835 return true; 1836 case NS_NEVER: 1837 return false; 1838 case NS_WHENEVER: 1839 // FIXME increase poll interval, or make event-driven 1840 ns = 1000000000LL; 1841 // fall through 1842 default: 1843 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1844 pauseInternal(ns); 1845 return true; 1846 } 1847} 1848 1849void AudioTrack::AudioTrackThread::requestExit() 1850{ 1851 // must be in this order to avoid a race condition 1852 Thread::requestExit(); 1853 resume(); 1854} 1855 1856void AudioTrack::AudioTrackThread::pause() 1857{ 1858 AutoMutex _l(mMyLock); 1859 mPaused = true; 1860} 1861 1862void AudioTrack::AudioTrackThread::resume() 1863{ 1864 AutoMutex _l(mMyLock); 1865 mIgnoreNextPausedInt = true; 1866 if (mPaused || mPausedInt) { 1867 mPaused = false; 1868 mPausedInt = false; 1869 mMyCond.signal(); 1870 } 1871} 1872 1873void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1874{ 1875 AutoMutex _l(mMyLock); 1876 mPausedInt = true; 1877 mPausedNs = ns; 1878} 1879 1880}; // namespace android 1881