AudioTrack.cpp revision ce8828a016b082f730152af2204b8ea3610dc1ec
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 return NO_ERROR; 538} 539 540status_t AudioTrack::setVolume(float volume) 541{ 542 return setVolume(volume, volume); 543} 544 545status_t AudioTrack::setAuxEffectSendLevel(float level) 546{ 547 if (level < 0.0f || level > 1.0f) { 548 return BAD_VALUE; 549 } 550 551 AutoMutex lock(mLock); 552 mSendLevel = level; 553 mProxy->setSendLevel(level); 554 555 return NO_ERROR; 556} 557 558void AudioTrack::getAuxEffectSendLevel(float* level) const 559{ 560 if (level != NULL) { 561 *level = mSendLevel; 562 } 563} 564 565status_t AudioTrack::setSampleRate(uint32_t rate) 566{ 567 if (mIsTimed || isOffloaded()) { 568 return INVALID_OPERATION; 569 } 570 571 uint32_t afSamplingRate; 572 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 573 return NO_INIT; 574 } 575 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 576 if (rate == 0 || rate > afSamplingRate*2 ) { 577 return BAD_VALUE; 578 } 579 580 AutoMutex lock(mLock); 581 mSampleRate = rate; 582 mProxy->setSampleRate(rate); 583 584 return NO_ERROR; 585} 586 587uint32_t AudioTrack::getSampleRate() const 588{ 589 if (mIsTimed) { 590 return 0; 591 } 592 593 AutoMutex lock(mLock); 594 return mSampleRate; 595} 596 597status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 598{ 599 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 600 return INVALID_OPERATION; 601 } 602 603 if (loopCount == 0) { 604 ; 605 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 606 loopEnd - loopStart >= MIN_LOOP) { 607 ; 608 } else { 609 return BAD_VALUE; 610 } 611 612 AutoMutex lock(mLock); 613 // See setPosition() regarding setting parameters such as loop points or position while active 614 if (mState == STATE_ACTIVE) { 615 return INVALID_OPERATION; 616 } 617 setLoop_l(loopStart, loopEnd, loopCount); 618 return NO_ERROR; 619} 620 621void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 622{ 623 // FIXME If setting a loop also sets position to start of loop, then 624 // this is correct. Otherwise it should be removed. 625 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 626 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 627 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 628} 629 630status_t AudioTrack::setMarkerPosition(uint32_t marker) 631{ 632 // The only purpose of setting marker position is to get a callback 633 if (mCbf == NULL || isOffloaded()) { 634 return INVALID_OPERATION; 635 } 636 637 AutoMutex lock(mLock); 638 mMarkerPosition = marker; 639 mMarkerReached = false; 640 641 return NO_ERROR; 642} 643 644status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 645{ 646 if (isOffloaded()) { 647 return INVALID_OPERATION; 648 } 649 if (marker == NULL) { 650 return BAD_VALUE; 651 } 652 653 AutoMutex lock(mLock); 654 *marker = mMarkerPosition; 655 656 return NO_ERROR; 657} 658 659status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 660{ 661 // The only purpose of setting position update period is to get a callback 662 if (mCbf == NULL || isOffloaded()) { 663 return INVALID_OPERATION; 664 } 665 666 AutoMutex lock(mLock); 667 mNewPosition = mProxy->getPosition() + updatePeriod; 668 mUpdatePeriod = updatePeriod; 669 return NO_ERROR; 670} 671 672status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 673{ 674 if (isOffloaded()) { 675 return INVALID_OPERATION; 676 } 677 if (updatePeriod == NULL) { 678 return BAD_VALUE; 679 } 680 681 AutoMutex lock(mLock); 682 *updatePeriod = mUpdatePeriod; 683 684 return NO_ERROR; 685} 686 687status_t AudioTrack::setPosition(uint32_t position) 688{ 689 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 690 return INVALID_OPERATION; 691 } 692 if (position > mFrameCount) { 693 return BAD_VALUE; 694 } 695 696 AutoMutex lock(mLock); 697 // Currently we require that the player is inactive before setting parameters such as position 698 // or loop points. Otherwise, there could be a race condition: the application could read the 699 // current position, compute a new position or loop parameters, and then set that position or 700 // loop parameters but it would do the "wrong" thing since the position has continued to advance 701 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 702 // to specify how it wants to handle such scenarios. 703 if (mState == STATE_ACTIVE) { 704 return INVALID_OPERATION; 705 } 706 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 707 mLoopPeriod = 0; 708 // FIXME Check whether loops and setting position are incompatible in old code. 709 // If we use setLoop for both purposes we lose the capability to set the position while looping. 710 mStaticProxy->setLoop(position, mFrameCount, 0); 711 712 return NO_ERROR; 713} 714 715status_t AudioTrack::getPosition(uint32_t *position) const 716{ 717 if (position == NULL) { 718 return BAD_VALUE; 719 } 720 721 AutoMutex lock(mLock); 722 if (isOffloaded()) { 723 uint32_t dspFrames = 0; 724 725 if (mOutput != 0) { 726 uint32_t halFrames; 727 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 728 } 729 *position = dspFrames; 730 } else { 731 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 732 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 733 mProxy->getPosition(); 734 } 735 return NO_ERROR; 736} 737 738status_t AudioTrack::getBufferPosition(size_t *position) 739{ 740 if (mSharedBuffer == 0 || mIsTimed) { 741 return INVALID_OPERATION; 742 } 743 if (position == NULL) { 744 return BAD_VALUE; 745 } 746 747 AutoMutex lock(mLock); 748 *position = mStaticProxy->getBufferPosition(); 749 return NO_ERROR; 750} 751 752status_t AudioTrack::reload() 753{ 754 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 755 return INVALID_OPERATION; 756 } 757 758 AutoMutex lock(mLock); 759 // See setPosition() regarding setting parameters such as loop points or position while active 760 if (mState == STATE_ACTIVE) { 761 return INVALID_OPERATION; 762 } 763 mNewPosition = mUpdatePeriod; 764 mLoopPeriod = 0; 765 // FIXME The new code cannot reload while keeping a loop specified. 766 // Need to check how the old code handled this, and whether it's a significant change. 767 mStaticProxy->setLoop(0, mFrameCount, 0); 768 return NO_ERROR; 769} 770 771audio_io_handle_t AudioTrack::getOutput() 772{ 773 AutoMutex lock(mLock); 774 return mOutput; 775} 776 777// must be called with mLock held 778audio_io_handle_t AudioTrack::getOutput_l() 779{ 780 if (mOutput) { 781 return mOutput; 782 } else { 783 return AudioSystem::getOutput(mStreamType, 784 mSampleRate, mFormat, mChannelMask, mFlags); 785 } 786} 787 788status_t AudioTrack::attachAuxEffect(int effectId) 789{ 790 AutoMutex lock(mLock); 791 status_t status = mAudioTrack->attachAuxEffect(effectId); 792 if (status == NO_ERROR) { 793 mAuxEffectId = effectId; 794 } 795 return status; 796} 797 798// ------------------------------------------------------------------------- 799 800// must be called with mLock held 801status_t AudioTrack::createTrack_l( 802 audio_stream_type_t streamType, 803 uint32_t sampleRate, 804 audio_format_t format, 805 size_t frameCount, 806 audio_output_flags_t flags, 807 const sp<IMemory>& sharedBuffer, 808 audio_io_handle_t output, 809 size_t epoch) 810{ 811 status_t status; 812 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 813 if (audioFlinger == 0) { 814 ALOGE("Could not get audioflinger"); 815 return NO_INIT; 816 } 817 818 // Not all of these values are needed under all conditions, but it is easier to get them all 819 820 uint32_t afLatency; 821 status = AudioSystem::getLatency(output, streamType, &afLatency); 822 if (status != NO_ERROR) { 823 ALOGE("getLatency(%d) failed status %d", output, status); 824 return NO_INIT; 825 } 826 827 size_t afFrameCount; 828 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 829 if (status != NO_ERROR) { 830 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 831 return NO_INIT; 832 } 833 834 uint32_t afSampleRate; 835 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 836 if (status != NO_ERROR) { 837 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 838 return NO_INIT; 839 } 840 841 // Client decides whether the track is TIMED (see below), but can only express a preference 842 // for FAST. Server will perform additional tests. 843 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 844 // either of these use cases: 845 // use case 1: shared buffer 846 (sharedBuffer != 0) || 847 // use case 2: callback handler 848 (mCbf != NULL))) { 849 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 850 // once denied, do not request again if IAudioTrack is re-created 851 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 852 mFlags = flags; 853 } 854 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 855 856 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 857 // n = 1 fast track; nBuffering is ignored 858 // n = 2 normal track, no sample rate conversion 859 // n = 3 normal track, with sample rate conversion 860 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 861 // n > 3 very high latency or very small notification interval; nBuffering is ignored 862 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 863 864 mNotificationFramesAct = mNotificationFramesReq; 865 866 if (!audio_is_linear_pcm(format)) { 867 868 if (sharedBuffer != 0) { 869 // Same comment as below about ignoring frameCount parameter for set() 870 frameCount = sharedBuffer->size(); 871 } else if (frameCount == 0) { 872 frameCount = afFrameCount; 873 } 874 if (mNotificationFramesAct != frameCount) { 875 mNotificationFramesAct = frameCount; 876 } 877 } else if (sharedBuffer != 0) { 878 879 // Ensure that buffer alignment matches channel count 880 // 8-bit data in shared memory is not currently supported by AudioFlinger 881 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 882 if (mChannelCount > 1) { 883 // More than 2 channels does not require stronger alignment than stereo 884 alignment <<= 1; 885 } 886 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 887 ALOGE("Invalid buffer alignment: address %p, channel count %u", 888 sharedBuffer->pointer(), mChannelCount); 889 return BAD_VALUE; 890 } 891 892 // When initializing a shared buffer AudioTrack via constructors, 893 // there's no frameCount parameter. 894 // But when initializing a shared buffer AudioTrack via set(), 895 // there _is_ a frameCount parameter. We silently ignore it. 896 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 897 898 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 899 900 // FIXME move these calculations and associated checks to server 901 902 // Ensure that buffer depth covers at least audio hardware latency 903 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 904 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 905 afFrameCount, minBufCount, afSampleRate, afLatency); 906 if (minBufCount <= nBuffering) { 907 minBufCount = nBuffering; 908 } 909 910 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 911 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 912 ", afLatency=%d", 913 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 914 915 if (frameCount == 0) { 916 frameCount = minFrameCount; 917 } else if (frameCount < minFrameCount) { 918 // not ALOGW because it happens all the time when playing key clicks over A2DP 919 ALOGV("Minimum buffer size corrected from %d to %d", 920 frameCount, minFrameCount); 921 frameCount = minFrameCount; 922 } 923 // Make sure that application is notified with sufficient margin before underrun 924 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 925 mNotificationFramesAct = frameCount/nBuffering; 926 } 927 928 } else { 929 // For fast tracks, the frame count calculations and checks are done by server 930 } 931 932 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 933 if (mIsTimed) { 934 trackFlags |= IAudioFlinger::TRACK_TIMED; 935 } 936 937 pid_t tid = -1; 938 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 939 trackFlags |= IAudioFlinger::TRACK_FAST; 940 if (mAudioTrackThread != 0) { 941 tid = mAudioTrackThread->getTid(); 942 } 943 } 944 945 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 946 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 947 } 948 949 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 950 sampleRate, 951 // AudioFlinger only sees 16-bit PCM 952 format == AUDIO_FORMAT_PCM_8_BIT ? 953 AUDIO_FORMAT_PCM_16_BIT : format, 954 mChannelMask, 955 frameCount, 956 &trackFlags, 957 sharedBuffer, 958 output, 959 tid, 960 &mSessionId, 961 mName, 962 &status); 963 964 if (track == 0) { 965 ALOGE("AudioFlinger could not create track, status: %d", status); 966 return status; 967 } 968 sp<IMemory> iMem = track->getCblk(); 969 if (iMem == 0) { 970 ALOGE("Could not get control block"); 971 return NO_INIT; 972 } 973 // invariant that mAudioTrack != 0 is true only after set() returns successfully 974 if (mAudioTrack != 0) { 975 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 976 mDeathNotifier.clear(); 977 } 978 mAudioTrack = track; 979 mCblkMemory = iMem; 980 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 981 mCblk = cblk; 982 size_t temp = cblk->frameCount_; 983 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 984 // In current design, AudioTrack client checks and ensures frame count validity before 985 // passing it to AudioFlinger so AudioFlinger should not return a different value except 986 // for fast track as it uses a special method of assigning frame count. 987 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 988 } 989 frameCount = temp; 990 mAwaitBoost = false; 991 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 992 if (trackFlags & IAudioFlinger::TRACK_FAST) { 993 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 994 mAwaitBoost = true; 995 if (sharedBuffer == 0) { 996 // double-buffering is not required for fast tracks, due to tighter scheduling 997 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 998 mNotificationFramesAct = frameCount; 999 } 1000 } 1001 } else { 1002 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1003 // once denied, do not request again if IAudioTrack is re-created 1004 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1005 mFlags = flags; 1006 if (sharedBuffer == 0) { 1007 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1008 mNotificationFramesAct = frameCount/nBuffering; 1009 } 1010 } 1011 } 1012 } 1013 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1014 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1015 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1016 } else { 1017 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1018 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1019 mFlags = flags; 1020 return NO_INIT; 1021 } 1022 } 1023 1024 mRefreshRemaining = true; 1025 1026 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1027 // is the value of pointer() for the shared buffer, otherwise buffers points 1028 // immediately after the control block. This address is for the mapping within client 1029 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1030 void* buffers; 1031 if (sharedBuffer == 0) { 1032 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1033 } else { 1034 buffers = sharedBuffer->pointer(); 1035 } 1036 1037 mAudioTrack->attachAuxEffect(mAuxEffectId); 1038 // FIXME don't believe this lie 1039 mLatency = afLatency + (1000*frameCount) / sampleRate; 1040 mFrameCount = frameCount; 1041 // If IAudioTrack is re-created, don't let the requested frameCount 1042 // decrease. This can confuse clients that cache frameCount(). 1043 if (frameCount > mReqFrameCount) { 1044 mReqFrameCount = frameCount; 1045 } 1046 1047 // update proxy 1048 if (sharedBuffer == 0) { 1049 mStaticProxy.clear(); 1050 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1051 } else { 1052 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1053 mProxy = mStaticProxy; 1054 } 1055 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1056 uint16_t(mVolume[LEFT] * 0x1000)); 1057 mProxy->setSendLevel(mSendLevel); 1058 mProxy->setSampleRate(mSampleRate); 1059 mProxy->setEpoch(epoch); 1060 mProxy->setMinimum(mNotificationFramesAct); 1061 1062 mDeathNotifier = new DeathNotifier(this); 1063 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1064 1065 return NO_ERROR; 1066} 1067 1068status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1069{ 1070 if (audioBuffer == NULL) { 1071 return BAD_VALUE; 1072 } 1073 if (mTransfer != TRANSFER_OBTAIN) { 1074 audioBuffer->frameCount = 0; 1075 audioBuffer->size = 0; 1076 audioBuffer->raw = NULL; 1077 return INVALID_OPERATION; 1078 } 1079 1080 const struct timespec *requested; 1081 if (waitCount == -1) { 1082 requested = &ClientProxy::kForever; 1083 } else if (waitCount == 0) { 1084 requested = &ClientProxy::kNonBlocking; 1085 } else if (waitCount > 0) { 1086 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1087 struct timespec timeout; 1088 timeout.tv_sec = ms / 1000; 1089 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1090 requested = &timeout; 1091 } else { 1092 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1093 requested = NULL; 1094 } 1095 return obtainBuffer(audioBuffer, requested); 1096} 1097 1098status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1099 struct timespec *elapsed, size_t *nonContig) 1100{ 1101 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1102 uint32_t oldSequence = 0; 1103 uint32_t newSequence; 1104 1105 Proxy::Buffer buffer; 1106 status_t status = NO_ERROR; 1107 1108 static const int32_t kMaxTries = 5; 1109 int32_t tryCounter = kMaxTries; 1110 1111 do { 1112 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1113 // keep them from going away if another thread re-creates the track during obtainBuffer() 1114 sp<AudioTrackClientProxy> proxy; 1115 sp<IMemory> iMem; 1116 1117 { // start of lock scope 1118 AutoMutex lock(mLock); 1119 1120 newSequence = mSequence; 1121 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1122 if (status == DEAD_OBJECT) { 1123 // re-create track, unless someone else has already done so 1124 if (newSequence == oldSequence) { 1125 status = restoreTrack_l("obtainBuffer"); 1126 if (status != NO_ERROR) { 1127 buffer.mFrameCount = 0; 1128 buffer.mRaw = NULL; 1129 buffer.mNonContig = 0; 1130 break; 1131 } 1132 } 1133 } 1134 oldSequence = newSequence; 1135 1136 // Keep the extra references 1137 proxy = mProxy; 1138 iMem = mCblkMemory; 1139 1140 if (mState == STATE_STOPPING) { 1141 status = -EINTR; 1142 buffer.mFrameCount = 0; 1143 buffer.mRaw = NULL; 1144 buffer.mNonContig = 0; 1145 break; 1146 } 1147 1148 // Non-blocking if track is stopped or paused 1149 if (mState != STATE_ACTIVE) { 1150 requested = &ClientProxy::kNonBlocking; 1151 } 1152 1153 } // end of lock scope 1154 1155 buffer.mFrameCount = audioBuffer->frameCount; 1156 // FIXME starts the requested timeout and elapsed over from scratch 1157 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1158 1159 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1160 1161 audioBuffer->frameCount = buffer.mFrameCount; 1162 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1163 audioBuffer->raw = buffer.mRaw; 1164 if (nonContig != NULL) { 1165 *nonContig = buffer.mNonContig; 1166 } 1167 return status; 1168} 1169 1170void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1171{ 1172 if (mTransfer == TRANSFER_SHARED) { 1173 return; 1174 } 1175 1176 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1177 if (stepCount == 0) { 1178 return; 1179 } 1180 1181 Proxy::Buffer buffer; 1182 buffer.mFrameCount = stepCount; 1183 buffer.mRaw = audioBuffer->raw; 1184 1185 AutoMutex lock(mLock); 1186 mInUnderrun = false; 1187 mProxy->releaseBuffer(&buffer); 1188 1189 // restart track if it was disabled by audioflinger due to previous underrun 1190 if (mState == STATE_ACTIVE) { 1191 audio_track_cblk_t* cblk = mCblk; 1192 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1193 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1194 this, mName.string()); 1195 // FIXME ignoring status 1196 mAudioTrack->start(); 1197 } 1198 } 1199} 1200 1201// ------------------------------------------------------------------------- 1202 1203ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1204{ 1205 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1206 return INVALID_OPERATION; 1207 } 1208 1209 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1210 // Sanity-check: user is most-likely passing an error code, and it would 1211 // make the return value ambiguous (actualSize vs error). 1212 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1213 return BAD_VALUE; 1214 } 1215 1216 size_t written = 0; 1217 Buffer audioBuffer; 1218 1219 while (userSize >= mFrameSize) { 1220 audioBuffer.frameCount = userSize / mFrameSize; 1221 1222 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1223 if (err < 0) { 1224 if (written > 0) { 1225 break; 1226 } 1227 return ssize_t(err); 1228 } 1229 1230 size_t toWrite; 1231 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1232 // Divide capacity by 2 to take expansion into account 1233 toWrite = audioBuffer.size >> 1; 1234 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1235 } else { 1236 toWrite = audioBuffer.size; 1237 memcpy(audioBuffer.i8, buffer, toWrite); 1238 } 1239 buffer = ((const char *) buffer) + toWrite; 1240 userSize -= toWrite; 1241 written += toWrite; 1242 1243 releaseBuffer(&audioBuffer); 1244 } 1245 1246 return written; 1247} 1248 1249// ------------------------------------------------------------------------- 1250 1251TimedAudioTrack::TimedAudioTrack() { 1252 mIsTimed = true; 1253} 1254 1255status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1256{ 1257 AutoMutex lock(mLock); 1258 status_t result = UNKNOWN_ERROR; 1259 1260#if 1 1261 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1262 // while we are accessing the cblk 1263 sp<IAudioTrack> audioTrack = mAudioTrack; 1264 sp<IMemory> iMem = mCblkMemory; 1265#endif 1266 1267 // If the track is not invalid already, try to allocate a buffer. alloc 1268 // fails indicating that the server is dead, flag the track as invalid so 1269 // we can attempt to restore in just a bit. 1270 audio_track_cblk_t* cblk = mCblk; 1271 if (!(cblk->mFlags & CBLK_INVALID)) { 1272 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1273 if (result == DEAD_OBJECT) { 1274 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1275 } 1276 } 1277 1278 // If the track is invalid at this point, attempt to restore it. and try the 1279 // allocation one more time. 1280 if (cblk->mFlags & CBLK_INVALID) { 1281 result = restoreTrack_l("allocateTimedBuffer"); 1282 1283 if (result == NO_ERROR) { 1284 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1285 } 1286 } 1287 1288 return result; 1289} 1290 1291status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1292 int64_t pts) 1293{ 1294 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1295 { 1296 AutoMutex lock(mLock); 1297 audio_track_cblk_t* cblk = mCblk; 1298 // restart track if it was disabled by audioflinger due to previous underrun 1299 if (buffer->size() != 0 && status == NO_ERROR && 1300 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1301 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1302 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1303 // FIXME ignoring status 1304 mAudioTrack->start(); 1305 } 1306 } 1307 return status; 1308} 1309 1310status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1311 TargetTimeline target) 1312{ 1313 return mAudioTrack->setMediaTimeTransform(xform, target); 1314} 1315 1316// ------------------------------------------------------------------------- 1317 1318nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1319{ 1320 // Currently the AudioTrack thread is not created if there are no callbacks. 1321 // Would it ever make sense to run the thread, even without callbacks? 1322 // If so, then replace this by checks at each use for mCbf != NULL. 1323 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1324 1325 mLock.lock(); 1326 if (mAwaitBoost) { 1327 mAwaitBoost = false; 1328 mLock.unlock(); 1329 static const int32_t kMaxTries = 5; 1330 int32_t tryCounter = kMaxTries; 1331 uint32_t pollUs = 10000; 1332 do { 1333 int policy = sched_getscheduler(0); 1334 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1335 break; 1336 } 1337 usleep(pollUs); 1338 pollUs <<= 1; 1339 } while (tryCounter-- > 0); 1340 if (tryCounter < 0) { 1341 ALOGE("did not receive expected priority boost on time"); 1342 } 1343 // Run again immediately 1344 return 0; 1345 } 1346 1347 // Can only reference mCblk while locked 1348 int32_t flags = android_atomic_and( 1349 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1350 1351 // Check for track invalidation 1352 if (flags & CBLK_INVALID) { 1353 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1354 // AudioSystem cache. We should not exit here but after calling the callback so 1355 // that the upper layers can recreate the track 1356 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1357 status_t status = restoreTrack_l("processAudioBuffer"); 1358 mLock.unlock(); 1359 // Run again immediately, but with a new IAudioTrack 1360 return 0; 1361 } 1362 } 1363 1364 bool waitStreamEnd = mState == STATE_STOPPING; 1365 bool active = mState == STATE_ACTIVE; 1366 1367 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1368 bool newUnderrun = false; 1369 if (flags & CBLK_UNDERRUN) { 1370#if 0 1371 // Currently in shared buffer mode, when the server reaches the end of buffer, 1372 // the track stays active in continuous underrun state. It's up to the application 1373 // to pause or stop the track, or set the position to a new offset within buffer. 1374 // This was some experimental code to auto-pause on underrun. Keeping it here 1375 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1376 if (mTransfer == TRANSFER_SHARED) { 1377 mState = STATE_PAUSED; 1378 active = false; 1379 } 1380#endif 1381 if (!mInUnderrun) { 1382 mInUnderrun = true; 1383 newUnderrun = true; 1384 } 1385 } 1386 1387 // Get current position of server 1388 size_t position = mProxy->getPosition(); 1389 1390 // Manage marker callback 1391 bool markerReached = false; 1392 size_t markerPosition = mMarkerPosition; 1393 // FIXME fails for wraparound, need 64 bits 1394 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1395 mMarkerReached = markerReached = true; 1396 } 1397 1398 // Determine number of new position callback(s) that will be needed, while locked 1399 size_t newPosCount = 0; 1400 size_t newPosition = mNewPosition; 1401 size_t updatePeriod = mUpdatePeriod; 1402 // FIXME fails for wraparound, need 64 bits 1403 if (updatePeriod > 0 && position >= newPosition) { 1404 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1405 mNewPosition += updatePeriod * newPosCount; 1406 } 1407 1408 // Cache other fields that will be needed soon 1409 uint32_t loopPeriod = mLoopPeriod; 1410 uint32_t sampleRate = mSampleRate; 1411 size_t notificationFrames = mNotificationFramesAct; 1412 if (mRefreshRemaining) { 1413 mRefreshRemaining = false; 1414 mRemainingFrames = notificationFrames; 1415 mRetryOnPartialBuffer = false; 1416 } 1417 size_t misalignment = mProxy->getMisalignment(); 1418 uint32_t sequence = mSequence; 1419 1420 // These fields don't need to be cached, because they are assigned only by set(): 1421 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1422 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1423 1424 mLock.unlock(); 1425 1426 if (waitStreamEnd) { 1427 AutoMutex lock(mLock); 1428 1429 sp<AudioTrackClientProxy> proxy = mProxy; 1430 sp<IMemory> iMem = mCblkMemory; 1431 1432 struct timespec timeout; 1433 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1434 timeout.tv_nsec = 0; 1435 1436 mLock.unlock(); 1437 status_t status = mProxy->waitStreamEndDone(&timeout); 1438 mLock.lock(); 1439 switch (status) { 1440 case NO_ERROR: 1441 case DEAD_OBJECT: 1442 case TIMED_OUT: 1443 mLock.unlock(); 1444 mCbf(EVENT_STREAM_END, mUserData, NULL); 1445 mLock.lock(); 1446 if (mState == STATE_STOPPING) { 1447 mState = STATE_STOPPED; 1448 if (status != DEAD_OBJECT) { 1449 return NS_INACTIVE; 1450 } 1451 } 1452 return 0; 1453 default: 1454 return 0; 1455 } 1456 } 1457 1458 // perform callbacks while unlocked 1459 if (newUnderrun) { 1460 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1461 } 1462 // FIXME we will miss loops if loop cycle was signaled several times since last call 1463 // to processAudioBuffer() 1464 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1465 mCbf(EVENT_LOOP_END, mUserData, NULL); 1466 } 1467 if (flags & CBLK_BUFFER_END) { 1468 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1469 } 1470 if (markerReached) { 1471 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1472 } 1473 while (newPosCount > 0) { 1474 size_t temp = newPosition; 1475 mCbf(EVENT_NEW_POS, mUserData, &temp); 1476 newPosition += updatePeriod; 1477 newPosCount--; 1478 } 1479 1480 if (mObservedSequence != sequence) { 1481 mObservedSequence = sequence; 1482 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1483 // for offloaded tracks, just wait for the upper layers to recreate the track 1484 if (isOffloaded()) { 1485 return NS_INACTIVE; 1486 } 1487 } 1488 1489 // if inactive, then don't run me again until re-started 1490 if (!active) { 1491 return NS_INACTIVE; 1492 } 1493 1494 // Compute the estimated time until the next timed event (position, markers, loops) 1495 // FIXME only for non-compressed audio 1496 uint32_t minFrames = ~0; 1497 if (!markerReached && position < markerPosition) { 1498 minFrames = markerPosition - position; 1499 } 1500 if (loopPeriod > 0 && loopPeriod < minFrames) { 1501 minFrames = loopPeriod; 1502 } 1503 if (updatePeriod > 0 && updatePeriod < minFrames) { 1504 minFrames = updatePeriod; 1505 } 1506 1507 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1508 static const uint32_t kPoll = 0; 1509 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1510 minFrames = kPoll * notificationFrames; 1511 } 1512 1513 // Convert frame units to time units 1514 nsecs_t ns = NS_WHENEVER; 1515 if (minFrames != (uint32_t) ~0) { 1516 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1517 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1518 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1519 } 1520 1521 // If not supplying data by EVENT_MORE_DATA, then we're done 1522 if (mTransfer != TRANSFER_CALLBACK) { 1523 return ns; 1524 } 1525 1526 struct timespec timeout; 1527 const struct timespec *requested = &ClientProxy::kForever; 1528 if (ns != NS_WHENEVER) { 1529 timeout.tv_sec = ns / 1000000000LL; 1530 timeout.tv_nsec = ns % 1000000000LL; 1531 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1532 requested = &timeout; 1533 } 1534 1535 while (mRemainingFrames > 0) { 1536 1537 Buffer audioBuffer; 1538 audioBuffer.frameCount = mRemainingFrames; 1539 size_t nonContig; 1540 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1541 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1542 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1543 requested = &ClientProxy::kNonBlocking; 1544 size_t avail = audioBuffer.frameCount + nonContig; 1545 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1546 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1547 if (err != NO_ERROR) { 1548 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1549 (isOffloaded() && (err == DEAD_OBJECT))) { 1550 return 0; 1551 } 1552 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1553 return NS_NEVER; 1554 } 1555 1556 if (mRetryOnPartialBuffer && !isOffloaded()) { 1557 mRetryOnPartialBuffer = false; 1558 if (avail < mRemainingFrames) { 1559 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1560 if (ns < 0 || myns < ns) { 1561 ns = myns; 1562 } 1563 return ns; 1564 } 1565 } 1566 1567 // Divide buffer size by 2 to take into account the expansion 1568 // due to 8 to 16 bit conversion: the callback must fill only half 1569 // of the destination buffer 1570 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1571 audioBuffer.size >>= 1; 1572 } 1573 1574 size_t reqSize = audioBuffer.size; 1575 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1576 size_t writtenSize = audioBuffer.size; 1577 size_t writtenFrames = writtenSize / mFrameSize; 1578 1579 // Sanity check on returned size 1580 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1581 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1582 reqSize, (int) writtenSize); 1583 return NS_NEVER; 1584 } 1585 1586 if (writtenSize == 0) { 1587 // The callback is done filling buffers 1588 // Keep this thread going to handle timed events and 1589 // still try to get more data in intervals of WAIT_PERIOD_MS 1590 // but don't just loop and block the CPU, so wait 1591 return WAIT_PERIOD_MS * 1000000LL; 1592 } 1593 1594 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1595 // 8 to 16 bit conversion, note that source and destination are the same address 1596 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1597 audioBuffer.size <<= 1; 1598 } 1599 1600 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1601 audioBuffer.frameCount = releasedFrames; 1602 mRemainingFrames -= releasedFrames; 1603 if (misalignment >= releasedFrames) { 1604 misalignment -= releasedFrames; 1605 } else { 1606 misalignment = 0; 1607 } 1608 1609 releaseBuffer(&audioBuffer); 1610 1611 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1612 // if callback doesn't like to accept the full chunk 1613 if (writtenSize < reqSize) { 1614 continue; 1615 } 1616 1617 // There could be enough non-contiguous frames available to satisfy the remaining request 1618 if (mRemainingFrames <= nonContig) { 1619 continue; 1620 } 1621 1622#if 0 1623 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1624 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1625 // that total to a sum == notificationFrames. 1626 if (0 < misalignment && misalignment <= mRemainingFrames) { 1627 mRemainingFrames = misalignment; 1628 return (mRemainingFrames * 1100000000LL) / sampleRate; 1629 } 1630#endif 1631 1632 } 1633 mRemainingFrames = notificationFrames; 1634 mRetryOnPartialBuffer = true; 1635 1636 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1637 return 0; 1638} 1639 1640status_t AudioTrack::restoreTrack_l(const char *from) 1641{ 1642 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1643 isOffloaded() ? "Offloaded" : "PCM", from); 1644 ++mSequence; 1645 status_t result; 1646 1647 // refresh the audio configuration cache in this process to make sure we get new 1648 // output parameters in getOutput_l() and createTrack_l() 1649 AudioSystem::clearAudioConfigCache(); 1650 1651 if (isOffloaded()) { 1652 return DEAD_OBJECT; 1653 } 1654 1655 // force new output query from audio policy manager; 1656 mOutput = 0; 1657 audio_io_handle_t output = getOutput_l(); 1658 1659 // if the new IAudioTrack is created, createTrack_l() will modify the 1660 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1661 // It will also delete the strong references on previous IAudioTrack and IMemory 1662 size_t position = mProxy->getPosition(); 1663 mNewPosition = position + mUpdatePeriod; 1664 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1665 result = createTrack_l(mStreamType, 1666 mSampleRate, 1667 mFormat, 1668 mReqFrameCount, // so that frame count never goes down 1669 mFlags, 1670 mSharedBuffer, 1671 output, 1672 position /*epoch*/); 1673 1674 if (result == NO_ERROR) { 1675 // continue playback from last known position, but 1676 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1677 if (mStaticProxy != NULL) { 1678 mLoopPeriod = 0; 1679 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1680 } 1681 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1682 // track destruction have been played? This is critical for SoundPool implementation 1683 // This must be broken, and needs to be tested/debugged. 1684#if 0 1685 // restore write index and set other indexes to reflect empty buffer status 1686 if (!strcmp(from, "start")) { 1687 // Make sure that a client relying on callback events indicating underrun or 1688 // the actual amount of audio frames played (e.g SoundPool) receives them. 1689 if (mSharedBuffer == 0) { 1690 // restart playback even if buffer is not completely filled. 1691 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1692 } 1693 } 1694#endif 1695 if (mState == STATE_ACTIVE) { 1696 result = mAudioTrack->start(); 1697 } 1698 } 1699 if (result != NO_ERROR) { 1700 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1701 // As getOutput was called above and resulted in an output stream to be opened, 1702 // we need to release it. 1703 AudioSystem::releaseOutput(output); 1704 ALOGW("restoreTrack_l() failed status %d", result); 1705 mState = STATE_STOPPED; 1706 } 1707 1708 return result; 1709} 1710 1711status_t AudioTrack::setParameters(const String8& keyValuePairs) 1712{ 1713 AutoMutex lock(mLock); 1714 return mAudioTrack->setParameters(keyValuePairs); 1715} 1716 1717status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1718{ 1719 AutoMutex lock(mLock); 1720 // FIXME not implemented for fast tracks; should use proxy and SSQ 1721 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1722 return INVALID_OPERATION; 1723 } 1724 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1725 return INVALID_OPERATION; 1726 } 1727 status_t status = mAudioTrack->getTimestamp(timestamp); 1728 if (status == NO_ERROR) { 1729 timestamp.mPosition += mProxy->getEpoch(); 1730 } 1731 return status; 1732} 1733 1734String8 AudioTrack::getParameters(const String8& keys) 1735{ 1736 if (mOutput) { 1737 return AudioSystem::getParameters(mOutput, keys); 1738 } else { 1739 return String8::empty(); 1740 } 1741} 1742 1743status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1744{ 1745 1746 const size_t SIZE = 256; 1747 char buffer[SIZE]; 1748 String8 result; 1749 1750 result.append(" AudioTrack::dump\n"); 1751 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1752 mVolume[0], mVolume[1]); 1753 result.append(buffer); 1754 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1755 mChannelCount, mFrameCount); 1756 result.append(buffer); 1757 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1758 result.append(buffer); 1759 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1760 result.append(buffer); 1761 ::write(fd, result.string(), result.size()); 1762 return NO_ERROR; 1763} 1764 1765uint32_t AudioTrack::getUnderrunFrames() const 1766{ 1767 AutoMutex lock(mLock); 1768 return mProxy->getUnderrunFrames(); 1769} 1770 1771// ========================================================================= 1772 1773void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1774{ 1775 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1776 if (audioTrack != 0) { 1777 AutoMutex lock(audioTrack->mLock); 1778 audioTrack->mProxy->binderDied(); 1779 } 1780} 1781 1782// ========================================================================= 1783 1784AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1785 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 1786{ 1787} 1788 1789AudioTrack::AudioTrackThread::~AudioTrackThread() 1790{ 1791} 1792 1793bool AudioTrack::AudioTrackThread::threadLoop() 1794{ 1795 { 1796 AutoMutex _l(mMyLock); 1797 if (mPaused) { 1798 mMyCond.wait(mMyLock); 1799 // caller will check for exitPending() 1800 return true; 1801 } 1802 } 1803 nsecs_t ns = mReceiver.processAudioBuffer(this); 1804 switch (ns) { 1805 case 0: 1806 return true; 1807 case NS_WHENEVER: 1808 sleep(1); 1809 return true; 1810 case NS_INACTIVE: 1811 pauseConditional(); 1812 return true; 1813 case NS_NEVER: 1814 return false; 1815 default: 1816 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1817 struct timespec req; 1818 req.tv_sec = ns / 1000000000LL; 1819 req.tv_nsec = ns % 1000000000LL; 1820 nanosleep(&req, NULL /*rem*/); 1821 return true; 1822 } 1823} 1824 1825void AudioTrack::AudioTrackThread::requestExit() 1826{ 1827 // must be in this order to avoid a race condition 1828 Thread::requestExit(); 1829 resume(); 1830} 1831 1832void AudioTrack::AudioTrackThread::pause() 1833{ 1834 AutoMutex _l(mMyLock); 1835 mPaused = true; 1836 mResumeLatch = false; 1837} 1838 1839void AudioTrack::AudioTrackThread::pauseConditional() 1840{ 1841 AutoMutex _l(mMyLock); 1842 if (mResumeLatch) { 1843 mResumeLatch = false; 1844 } else { 1845 mPaused = true; 1846 } 1847} 1848 1849void AudioTrack::AudioTrackThread::resume() 1850{ 1851 AutoMutex _l(mMyLock); 1852 if (mPaused) { 1853 mPaused = false; 1854 mResumeLatch = false; 1855 mMyCond.signal(); 1856 } else { 1857 mResumeLatch = true; 1858 } 1859} 1860 1861}; // namespace android 1862