AudioTrack.cpp revision d054c32443a493513ab63529b0c8b1aca290278c
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 if (mAudioTrack != 0) { 152 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 153 mAudioTrack.clear(); 154 } 155 IPCThreadState::self()->flushCommands(); 156 AudioSystem::releaseAudioSessionId(mSessionId); 157 } 158} 159 160status_t AudioTrack::set( 161 audio_stream_type_t streamType, 162 uint32_t sampleRate, 163 audio_format_t format, 164 audio_channel_mask_t channelMask, 165 int frameCountInt, 166 audio_output_flags_t flags, 167 callback_t cbf, 168 void* user, 169 int notificationFrames, 170 const sp<IMemory>& sharedBuffer, 171 bool threadCanCallJava, 172 int sessionId, 173 transfer_type transferType, 174 const audio_offload_info_t *offloadInfo) 175{ 176 switch (transferType) { 177 case TRANSFER_DEFAULT: 178 if (sharedBuffer != 0) { 179 transferType = TRANSFER_SHARED; 180 } else if (cbf == NULL || threadCanCallJava) { 181 transferType = TRANSFER_SYNC; 182 } else { 183 transferType = TRANSFER_CALLBACK; 184 } 185 break; 186 case TRANSFER_CALLBACK: 187 if (cbf == NULL || sharedBuffer != 0) { 188 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 189 return BAD_VALUE; 190 } 191 break; 192 case TRANSFER_OBTAIN: 193 case TRANSFER_SYNC: 194 if (sharedBuffer != 0) { 195 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 196 return BAD_VALUE; 197 } 198 break; 199 case TRANSFER_SHARED: 200 if (sharedBuffer == 0) { 201 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 202 return BAD_VALUE; 203 } 204 break; 205 default: 206 ALOGE("Invalid transfer type %d", transferType); 207 return BAD_VALUE; 208 } 209 mTransfer = transferType; 210 211 // FIXME "int" here is legacy and will be replaced by size_t later 212 if (frameCountInt < 0) { 213 ALOGE("Invalid frame count %d", frameCountInt); 214 return BAD_VALUE; 215 } 216 size_t frameCount = frameCountInt; 217 218 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 219 sharedBuffer->size()); 220 221 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 222 223 AutoMutex lock(mLock); 224 225 if (mAudioTrack != 0) { 226 ALOGE("Track already in use"); 227 return INVALID_OPERATION; 228 } 229 230 mOutput = 0; 231 232 // handle default values first. 233 if (streamType == AUDIO_STREAM_DEFAULT) { 234 streamType = AUDIO_STREAM_MUSIC; 235 } 236 237 if (sampleRate == 0) { 238 uint32_t afSampleRate; 239 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 240 return NO_INIT; 241 } 242 sampleRate = afSampleRate; 243 } 244 mSampleRate = sampleRate; 245 246 // these below should probably come from the audioFlinger too... 247 if (format == AUDIO_FORMAT_DEFAULT) { 248 format = AUDIO_FORMAT_PCM_16_BIT; 249 } 250 if (channelMask == 0) { 251 channelMask = AUDIO_CHANNEL_OUT_STEREO; 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 // AudioFlinger does not currently support 8-bit data in shared memory 261 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 262 ALOGE("8-bit data in shared memory is not supported"); 263 return BAD_VALUE; 264 } 265 266 // force direct flag if format is not linear PCM 267 // or offload was requested 268 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 269 || !audio_is_linear_pcm(format)) { 270 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 271 ? "Offload request, forcing to Direct Output" 272 : "Not linear PCM, forcing to Direct Output"); 273 flags = (audio_output_flags_t) 274 // FIXME why can't we allow direct AND fast? 275 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 276 } 277 // only allow deep buffering for music stream type 278 if (streamType != AUDIO_STREAM_MUSIC) { 279 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 280 } 281 282 if (!audio_is_output_channel(channelMask)) { 283 ALOGE("Invalid channel mask %#x", channelMask); 284 return BAD_VALUE; 285 } 286 mChannelMask = channelMask; 287 uint32_t channelCount = popcount(channelMask); 288 mChannelCount = channelCount; 289 290 if (audio_is_linear_pcm(format)) { 291 mFrameSize = channelCount * audio_bytes_per_sample(format); 292 mFrameSizeAF = channelCount * sizeof(int16_t); 293 } else { 294 mFrameSize = sizeof(uint8_t); 295 mFrameSizeAF = sizeof(uint8_t); 296 } 297 298 audio_io_handle_t output = AudioSystem::getOutput( 299 streamType, 300 sampleRate, format, channelMask, 301 flags, 302 offloadInfo); 303 304 if (output == 0) { 305 ALOGE("Could not get audio output for stream type %d", streamType); 306 return BAD_VALUE; 307 } 308 309 mVolume[LEFT] = 1.0f; 310 mVolume[RIGHT] = 1.0f; 311 mSendLevel = 0.0f; 312 mFrameCount = frameCount; 313 mReqFrameCount = frameCount; 314 mNotificationFramesReq = notificationFrames; 315 mNotificationFramesAct = 0; 316 mSessionId = sessionId; 317 mAuxEffectId = 0; 318 mFlags = flags; 319 mCbf = cbf; 320 321 if (cbf != NULL) { 322 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 323 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 324 } 325 326 // create the IAudioTrack 327 status_t status = createTrack_l(streamType, 328 sampleRate, 329 format, 330 frameCount, 331 flags, 332 sharedBuffer, 333 output, 334 0 /*epoch*/); 335 336 if (status != NO_ERROR) { 337 if (mAudioTrackThread != 0) { 338 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 339 mAudioTrackThread->requestExitAndWait(); 340 mAudioTrackThread.clear(); 341 } 342 //Use of direct and offloaded output streams is ref counted by audio policy manager. 343 // As getOutput was called above and resulted in an output stream to be opened, 344 // we need to release it. 345 AudioSystem::releaseOutput(output); 346 return status; 347 } 348 349 mStatus = NO_ERROR; 350 mStreamType = streamType; 351 mFormat = format; 352 mSharedBuffer = sharedBuffer; 353 mState = STATE_STOPPED; 354 mUserData = user; 355 mLoopPeriod = 0; 356 mMarkerPosition = 0; 357 mMarkerReached = false; 358 mNewPosition = 0; 359 mUpdatePeriod = 0; 360 AudioSystem::acquireAudioSessionId(mSessionId); 361 mSequence = 1; 362 mObservedSequence = mSequence; 363 mInUnderrun = false; 364 mOutput = output; 365 366 return NO_ERROR; 367} 368 369// ------------------------------------------------------------------------- 370 371status_t AudioTrack::start() 372{ 373 AutoMutex lock(mLock); 374 375 if (mState == STATE_ACTIVE) { 376 return INVALID_OPERATION; 377 } 378 379 mInUnderrun = true; 380 381 State previousState = mState; 382 if (previousState == STATE_PAUSED_STOPPING) { 383 mState = STATE_STOPPING; 384 } else { 385 mState = STATE_ACTIVE; 386 } 387 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 388 // reset current position as seen by client to 0 389 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 390 } 391 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 392 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 393 394 sp<AudioTrackThread> t = mAudioTrackThread; 395 if (t != 0) { 396 if (previousState == STATE_STOPPING) { 397 mProxy->interrupt(); 398 } else { 399 t->resume(); 400 } 401 } else { 402 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 403 get_sched_policy(0, &mPreviousSchedulingGroup); 404 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 405 } 406 407 status_t status = NO_ERROR; 408 if (!(flags & CBLK_INVALID)) { 409 status = mAudioTrack->start(); 410 if (status == DEAD_OBJECT) { 411 flags |= CBLK_INVALID; 412 } 413 } 414 if (flags & CBLK_INVALID) { 415 status = restoreTrack_l("start"); 416 } 417 418 if (status != NO_ERROR) { 419 ALOGE("start() status %d", status); 420 mState = previousState; 421 if (t != 0) { 422 if (previousState != STATE_STOPPING) { 423 t->pause(); 424 } 425 } else { 426 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 427 set_sched_policy(0, mPreviousSchedulingGroup); 428 } 429 } 430 431 return status; 432} 433 434void AudioTrack::stop() 435{ 436 AutoMutex lock(mLock); 437 // FIXME pause then stop should not be a nop 438 if (mState != STATE_ACTIVE) { 439 return; 440 } 441 442 if (isOffloaded()) { 443 mState = STATE_STOPPING; 444 } else { 445 mState = STATE_STOPPED; 446 } 447 448 mProxy->interrupt(); 449 mAudioTrack->stop(); 450 // the playback head position will reset to 0, so if a marker is set, we need 451 // to activate it again 452 mMarkerReached = false; 453#if 0 454 // Force flush if a shared buffer is used otherwise audioflinger 455 // will not stop before end of buffer is reached. 456 // It may be needed to make sure that we stop playback, likely in case looping is on. 457 if (mSharedBuffer != 0) { 458 flush_l(); 459 } 460#endif 461 462 sp<AudioTrackThread> t = mAudioTrackThread; 463 if (t != 0) { 464 if (!isOffloaded()) { 465 t->pause(); 466 } 467 } else { 468 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 469 set_sched_policy(0, mPreviousSchedulingGroup); 470 } 471} 472 473bool AudioTrack::stopped() const 474{ 475 AutoMutex lock(mLock); 476 return mState != STATE_ACTIVE; 477} 478 479void AudioTrack::flush() 480{ 481 if (mSharedBuffer != 0) { 482 return; 483 } 484 AutoMutex lock(mLock); 485 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 486 return; 487 } 488 flush_l(); 489} 490 491void AudioTrack::flush_l() 492{ 493 ALOG_ASSERT(mState != STATE_ACTIVE); 494 495 // clear playback marker and periodic update counter 496 mMarkerPosition = 0; 497 mMarkerReached = false; 498 mUpdatePeriod = 0; 499 mRefreshRemaining = true; 500 501 mState = STATE_FLUSHED; 502 if (isOffloaded()) { 503 mProxy->interrupt(); 504 } 505 mProxy->flush(); 506 mAudioTrack->flush(); 507} 508 509void AudioTrack::pause() 510{ 511 AutoMutex lock(mLock); 512 if (mState == STATE_ACTIVE) { 513 mState = STATE_PAUSED; 514 } else if (mState == STATE_STOPPING) { 515 mState = STATE_PAUSED_STOPPING; 516 } else { 517 return; 518 } 519 mProxy->interrupt(); 520 mAudioTrack->pause(); 521} 522 523status_t AudioTrack::setVolume(float left, float right) 524{ 525 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 526 return BAD_VALUE; 527 } 528 529 AutoMutex lock(mLock); 530 mVolume[LEFT] = left; 531 mVolume[RIGHT] = right; 532 533 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 534 535 return NO_ERROR; 536} 537 538status_t AudioTrack::setVolume(float volume) 539{ 540 return setVolume(volume, volume); 541} 542 543status_t AudioTrack::setAuxEffectSendLevel(float level) 544{ 545 if (level < 0.0f || level > 1.0f) { 546 return BAD_VALUE; 547 } 548 549 AutoMutex lock(mLock); 550 mSendLevel = level; 551 mProxy->setSendLevel(level); 552 553 return NO_ERROR; 554} 555 556void AudioTrack::getAuxEffectSendLevel(float* level) const 557{ 558 if (level != NULL) { 559 *level = mSendLevel; 560 } 561} 562 563status_t AudioTrack::setSampleRate(uint32_t rate) 564{ 565 if (mIsTimed || isOffloaded()) { 566 return INVALID_OPERATION; 567 } 568 569 uint32_t afSamplingRate; 570 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 571 return NO_INIT; 572 } 573 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 574 if (rate == 0 || rate > afSamplingRate*2 ) { 575 return BAD_VALUE; 576 } 577 578 AutoMutex lock(mLock); 579 mSampleRate = rate; 580 mProxy->setSampleRate(rate); 581 582 return NO_ERROR; 583} 584 585uint32_t AudioTrack::getSampleRate() const 586{ 587 if (mIsTimed) { 588 return 0; 589 } 590 591 AutoMutex lock(mLock); 592 return mSampleRate; 593} 594 595status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 596{ 597 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 598 return INVALID_OPERATION; 599 } 600 601 if (loopCount == 0) { 602 ; 603 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 604 loopEnd - loopStart >= MIN_LOOP) { 605 ; 606 } else { 607 return BAD_VALUE; 608 } 609 610 AutoMutex lock(mLock); 611 // See setPosition() regarding setting parameters such as loop points or position while active 612 if (mState == STATE_ACTIVE) { 613 return INVALID_OPERATION; 614 } 615 setLoop_l(loopStart, loopEnd, loopCount); 616 return NO_ERROR; 617} 618 619void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 620{ 621 // FIXME If setting a loop also sets position to start of loop, then 622 // this is correct. Otherwise it should be removed. 623 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 624 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 625 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 626} 627 628status_t AudioTrack::setMarkerPosition(uint32_t marker) 629{ 630 // The only purpose of setting marker position is to get a callback 631 if (mCbf == NULL || isOffloaded()) { 632 return INVALID_OPERATION; 633 } 634 635 AutoMutex lock(mLock); 636 mMarkerPosition = marker; 637 mMarkerReached = false; 638 639 return NO_ERROR; 640} 641 642status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 643{ 644 if (isOffloaded()) { 645 return INVALID_OPERATION; 646 } 647 if (marker == NULL) { 648 return BAD_VALUE; 649 } 650 651 AutoMutex lock(mLock); 652 *marker = mMarkerPosition; 653 654 return NO_ERROR; 655} 656 657status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 658{ 659 // The only purpose of setting position update period is to get a callback 660 if (mCbf == NULL || isOffloaded()) { 661 return INVALID_OPERATION; 662 } 663 664 AutoMutex lock(mLock); 665 mNewPosition = mProxy->getPosition() + updatePeriod; 666 mUpdatePeriod = updatePeriod; 667 return NO_ERROR; 668} 669 670status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 671{ 672 if (isOffloaded()) { 673 return INVALID_OPERATION; 674 } 675 if (updatePeriod == NULL) { 676 return BAD_VALUE; 677 } 678 679 AutoMutex lock(mLock); 680 *updatePeriod = mUpdatePeriod; 681 682 return NO_ERROR; 683} 684 685status_t AudioTrack::setPosition(uint32_t position) 686{ 687 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 688 return INVALID_OPERATION; 689 } 690 if (position > mFrameCount) { 691 return BAD_VALUE; 692 } 693 694 AutoMutex lock(mLock); 695 // Currently we require that the player is inactive before setting parameters such as position 696 // or loop points. Otherwise, there could be a race condition: the application could read the 697 // current position, compute a new position or loop parameters, and then set that position or 698 // loop parameters but it would do the "wrong" thing since the position has continued to advance 699 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 700 // to specify how it wants to handle such scenarios. 701 if (mState == STATE_ACTIVE) { 702 return INVALID_OPERATION; 703 } 704 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 705 mLoopPeriod = 0; 706 // FIXME Check whether loops and setting position are incompatible in old code. 707 // If we use setLoop for both purposes we lose the capability to set the position while looping. 708 mStaticProxy->setLoop(position, mFrameCount, 0); 709 710 return NO_ERROR; 711} 712 713status_t AudioTrack::getPosition(uint32_t *position) const 714{ 715 if (position == NULL) { 716 return BAD_VALUE; 717 } 718 719 AutoMutex lock(mLock); 720 if (isOffloaded()) { 721 uint32_t dspFrames = 0; 722 723 if (mOutput != 0) { 724 uint32_t halFrames; 725 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 726 } 727 *position = dspFrames; 728 } else { 729 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 730 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 731 mProxy->getPosition(); 732 } 733 return NO_ERROR; 734} 735 736status_t AudioTrack::getBufferPosition(size_t *position) 737{ 738 if (mSharedBuffer == 0 || mIsTimed) { 739 return INVALID_OPERATION; 740 } 741 if (position == NULL) { 742 return BAD_VALUE; 743 } 744 745 AutoMutex lock(mLock); 746 *position = mStaticProxy->getBufferPosition(); 747 return NO_ERROR; 748} 749 750status_t AudioTrack::reload() 751{ 752 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 753 return INVALID_OPERATION; 754 } 755 756 AutoMutex lock(mLock); 757 // See setPosition() regarding setting parameters such as loop points or position while active 758 if (mState == STATE_ACTIVE) { 759 return INVALID_OPERATION; 760 } 761 mNewPosition = mUpdatePeriod; 762 mLoopPeriod = 0; 763 // FIXME The new code cannot reload while keeping a loop specified. 764 // Need to check how the old code handled this, and whether it's a significant change. 765 mStaticProxy->setLoop(0, mFrameCount, 0); 766 return NO_ERROR; 767} 768 769audio_io_handle_t AudioTrack::getOutput() 770{ 771 AutoMutex lock(mLock); 772 return mOutput; 773} 774 775// must be called with mLock held 776audio_io_handle_t AudioTrack::getOutput_l() 777{ 778 if (mOutput) { 779 return mOutput; 780 } else { 781 return AudioSystem::getOutput(mStreamType, 782 mSampleRate, mFormat, mChannelMask, mFlags); 783 } 784} 785 786status_t AudioTrack::attachAuxEffect(int effectId) 787{ 788 AutoMutex lock(mLock); 789 status_t status = mAudioTrack->attachAuxEffect(effectId); 790 if (status == NO_ERROR) { 791 mAuxEffectId = effectId; 792 } 793 return status; 794} 795 796// ------------------------------------------------------------------------- 797 798// must be called with mLock held 799status_t AudioTrack::createTrack_l( 800 audio_stream_type_t streamType, 801 uint32_t sampleRate, 802 audio_format_t format, 803 size_t frameCount, 804 audio_output_flags_t flags, 805 const sp<IMemory>& sharedBuffer, 806 audio_io_handle_t output, 807 size_t epoch) 808{ 809 status_t status; 810 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 811 if (audioFlinger == 0) { 812 ALOGE("Could not get audioflinger"); 813 return NO_INIT; 814 } 815 816 uint32_t afLatency; 817 if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { 818 ALOGE("getLatency(%d) failed status %d", output, status); 819 return NO_INIT; 820 } 821 822 // Client decides whether the track is TIMED (see below), but can only express a preference 823 // for FAST. Server will perform additional tests. 824 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 825 // either of these use cases: 826 // use case 1: shared buffer 827 (sharedBuffer != 0) || 828 // use case 2: callback handler 829 (mCbf != NULL))) { 830 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 831 // once denied, do not request again if IAudioTrack is re-created 832 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 833 mFlags = flags; 834 } 835 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 836 837 mNotificationFramesAct = mNotificationFramesReq; 838 839 if (!audio_is_linear_pcm(format)) { 840 841 if (sharedBuffer != 0) { 842 // Same comment as below about ignoring frameCount parameter for set() 843 frameCount = sharedBuffer->size(); 844 } else if (frameCount == 0) { 845 size_t afFrameCount; 846 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 847 if (status != NO_ERROR) { 848 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, 849 status); 850 return NO_INIT; 851 } 852 frameCount = afFrameCount; 853 } 854 if (mNotificationFramesAct != frameCount) { 855 mNotificationFramesAct = frameCount; 856 } 857 } else if (sharedBuffer != 0) { 858 859 // Ensure that buffer alignment matches channel count 860 // 8-bit data in shared memory is not currently supported by AudioFlinger 861 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 862 if (mChannelCount > 1) { 863 // More than 2 channels does not require stronger alignment than stereo 864 alignment <<= 1; 865 } 866 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 867 ALOGE("Invalid buffer alignment: address %p, channel count %u", 868 sharedBuffer->pointer(), mChannelCount); 869 return BAD_VALUE; 870 } 871 872 // When initializing a shared buffer AudioTrack via constructors, 873 // there's no frameCount parameter. 874 // But when initializing a shared buffer AudioTrack via set(), 875 // there _is_ a frameCount parameter. We silently ignore it. 876 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 877 878 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 879 880 // FIXME move these calculations and associated checks to server 881 uint32_t afSampleRate; 882 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 883 if (status != NO_ERROR) { 884 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, 885 status); 886 return NO_INIT; 887 } 888 size_t afFrameCount; 889 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 890 if (status != NO_ERROR) { 891 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 892 return NO_INIT; 893 } 894 895 // Ensure that buffer depth covers at least audio hardware latency 896 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 897 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 898 afFrameCount, minBufCount, afSampleRate, afLatency); 899 if (minBufCount <= 2) { 900 minBufCount = sampleRate == afSampleRate ? 2 : 3; 901 } 902 903 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 904 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 905 ", afLatency=%d", 906 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 907 908 if (frameCount == 0) { 909 frameCount = minFrameCount; 910 } 911 // Make sure that application is notified with sufficient margin 912 // before underrun 913 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 914 mNotificationFramesAct = frameCount/2; 915 } 916 if (frameCount < minFrameCount) { 917 // not ALOGW because it happens all the time when playing key clicks over A2DP 918 ALOGV("Minimum buffer size corrected from %d to %d", 919 frameCount, minFrameCount); 920 frameCount = minFrameCount; 921 } 922 923 } else { 924 // For fast tracks, the frame count calculations and checks are done by server 925 } 926 927 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 928 if (mIsTimed) { 929 trackFlags |= IAudioFlinger::TRACK_TIMED; 930 } 931 932 pid_t tid = -1; 933 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 934 trackFlags |= IAudioFlinger::TRACK_FAST; 935 if (mAudioTrackThread != 0) { 936 tid = mAudioTrackThread->getTid(); 937 } 938 } 939 940 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 941 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 942 } 943 944 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 945 sampleRate, 946 // AudioFlinger only sees 16-bit PCM 947 format == AUDIO_FORMAT_PCM_8_BIT ? 948 AUDIO_FORMAT_PCM_16_BIT : format, 949 mChannelMask, 950 frameCount, 951 &trackFlags, 952 sharedBuffer, 953 output, 954 tid, 955 &mSessionId, 956 mName, 957 &status); 958 959 if (track == 0) { 960 ALOGE("AudioFlinger could not create track, status: %d", status); 961 return status; 962 } 963 sp<IMemory> iMem = track->getCblk(); 964 if (iMem == 0) { 965 ALOGE("Could not get control block"); 966 return NO_INIT; 967 } 968 if (mAudioTrack != 0) { 969 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 970 mDeathNotifier.clear(); 971 } 972 mAudioTrack = track; 973 mCblkMemory = iMem; 974 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 975 mCblk = cblk; 976 size_t temp = cblk->frameCount_; 977 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 978 // In current design, AudioTrack client checks and ensures frame count validity before 979 // passing it to AudioFlinger so AudioFlinger should not return a different value except 980 // for fast track as it uses a special method of assigning frame count. 981 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 982 } 983 frameCount = temp; 984 mAwaitBoost = false; 985 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 986 if (trackFlags & IAudioFlinger::TRACK_FAST) { 987 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 988 mAwaitBoost = true; 989 if (sharedBuffer == 0) { 990 // double-buffering is not required for fast tracks, due to tighter scheduling 991 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 992 mNotificationFramesAct = frameCount; 993 } 994 } 995 } else { 996 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 997 // once denied, do not request again if IAudioTrack is re-created 998 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 999 mFlags = flags; 1000 if (sharedBuffer == 0) { 1001 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 1002 mNotificationFramesAct = frameCount/2; 1003 } 1004 } 1005 } 1006 } 1007 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1008 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1009 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1010 } else { 1011 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1012 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1013 mFlags = flags; 1014 return NO_INIT; 1015 } 1016 } 1017 1018 mRefreshRemaining = true; 1019 1020 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1021 // is the value of pointer() for the shared buffer, otherwise buffers points 1022 // immediately after the control block. This address is for the mapping within client 1023 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1024 void* buffers; 1025 if (sharedBuffer == 0) { 1026 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1027 } else { 1028 buffers = sharedBuffer->pointer(); 1029 } 1030 1031 mAudioTrack->attachAuxEffect(mAuxEffectId); 1032 // FIXME don't believe this lie 1033 mLatency = afLatency + (1000*frameCount) / sampleRate; 1034 mFrameCount = frameCount; 1035 // If IAudioTrack is re-created, don't let the requested frameCount 1036 // decrease. This can confuse clients that cache frameCount(). 1037 if (frameCount > mReqFrameCount) { 1038 mReqFrameCount = frameCount; 1039 } 1040 1041 // update proxy 1042 if (sharedBuffer == 0) { 1043 mStaticProxy.clear(); 1044 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1045 } else { 1046 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1047 mProxy = mStaticProxy; 1048 } 1049 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1050 uint16_t(mVolume[LEFT] * 0x1000)); 1051 mProxy->setSendLevel(mSendLevel); 1052 mProxy->setSampleRate(mSampleRate); 1053 mProxy->setEpoch(epoch); 1054 mProxy->setMinimum(mNotificationFramesAct); 1055 1056 mDeathNotifier = new DeathNotifier(this); 1057 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1058 1059 return NO_ERROR; 1060} 1061 1062status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1063{ 1064 if (audioBuffer == NULL) { 1065 return BAD_VALUE; 1066 } 1067 if (mTransfer != TRANSFER_OBTAIN) { 1068 audioBuffer->frameCount = 0; 1069 audioBuffer->size = 0; 1070 audioBuffer->raw = NULL; 1071 return INVALID_OPERATION; 1072 } 1073 1074 const struct timespec *requested; 1075 if (waitCount == -1) { 1076 requested = &ClientProxy::kForever; 1077 } else if (waitCount == 0) { 1078 requested = &ClientProxy::kNonBlocking; 1079 } else if (waitCount > 0) { 1080 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1081 struct timespec timeout; 1082 timeout.tv_sec = ms / 1000; 1083 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1084 requested = &timeout; 1085 } else { 1086 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1087 requested = NULL; 1088 } 1089 return obtainBuffer(audioBuffer, requested); 1090} 1091 1092status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1093 struct timespec *elapsed, size_t *nonContig) 1094{ 1095 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1096 uint32_t oldSequence = 0; 1097 uint32_t newSequence; 1098 1099 Proxy::Buffer buffer; 1100 status_t status = NO_ERROR; 1101 1102 static const int32_t kMaxTries = 5; 1103 int32_t tryCounter = kMaxTries; 1104 1105 do { 1106 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1107 // keep them from going away if another thread re-creates the track during obtainBuffer() 1108 sp<AudioTrackClientProxy> proxy; 1109 sp<IMemory> iMem; 1110 1111 { // start of lock scope 1112 AutoMutex lock(mLock); 1113 1114 newSequence = mSequence; 1115 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1116 if (status == DEAD_OBJECT) { 1117 // re-create track, unless someone else has already done so 1118 if (newSequence == oldSequence) { 1119 status = restoreTrack_l("obtainBuffer"); 1120 if (status != NO_ERROR) { 1121 buffer.mFrameCount = 0; 1122 buffer.mRaw = NULL; 1123 buffer.mNonContig = 0; 1124 break; 1125 } 1126 } 1127 } 1128 oldSequence = newSequence; 1129 1130 // Keep the extra references 1131 proxy = mProxy; 1132 iMem = mCblkMemory; 1133 1134 if (mState == STATE_STOPPING) { 1135 status = -EINTR; 1136 buffer.mFrameCount = 0; 1137 buffer.mRaw = NULL; 1138 buffer.mNonContig = 0; 1139 break; 1140 } 1141 1142 // Non-blocking if track is stopped or paused 1143 if (mState != STATE_ACTIVE) { 1144 requested = &ClientProxy::kNonBlocking; 1145 } 1146 1147 } // end of lock scope 1148 1149 buffer.mFrameCount = audioBuffer->frameCount; 1150 // FIXME starts the requested timeout and elapsed over from scratch 1151 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1152 1153 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1154 1155 audioBuffer->frameCount = buffer.mFrameCount; 1156 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1157 audioBuffer->raw = buffer.mRaw; 1158 if (nonContig != NULL) { 1159 *nonContig = buffer.mNonContig; 1160 } 1161 return status; 1162} 1163 1164void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1165{ 1166 if (mTransfer == TRANSFER_SHARED) { 1167 return; 1168 } 1169 1170 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1171 if (stepCount == 0) { 1172 return; 1173 } 1174 1175 Proxy::Buffer buffer; 1176 buffer.mFrameCount = stepCount; 1177 buffer.mRaw = audioBuffer->raw; 1178 1179 AutoMutex lock(mLock); 1180 mInUnderrun = false; 1181 mProxy->releaseBuffer(&buffer); 1182 1183 // restart track if it was disabled by audioflinger due to previous underrun 1184 if (mState == STATE_ACTIVE) { 1185 audio_track_cblk_t* cblk = mCblk; 1186 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1187 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1188 this, mName.string()); 1189 // FIXME ignoring status 1190 mAudioTrack->start(); 1191 } 1192 } 1193} 1194 1195// ------------------------------------------------------------------------- 1196 1197ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1198{ 1199 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1200 return INVALID_OPERATION; 1201 } 1202 1203 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1204 // Sanity-check: user is most-likely passing an error code, and it would 1205 // make the return value ambiguous (actualSize vs error). 1206 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1207 return BAD_VALUE; 1208 } 1209 1210 size_t written = 0; 1211 Buffer audioBuffer; 1212 1213 while (userSize >= mFrameSize) { 1214 audioBuffer.frameCount = userSize / mFrameSize; 1215 1216 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1217 if (err < 0) { 1218 if (written > 0) { 1219 break; 1220 } 1221 return ssize_t(err); 1222 } 1223 1224 size_t toWrite; 1225 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1226 // Divide capacity by 2 to take expansion into account 1227 toWrite = audioBuffer.size >> 1; 1228 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1229 } else { 1230 toWrite = audioBuffer.size; 1231 memcpy(audioBuffer.i8, buffer, toWrite); 1232 } 1233 buffer = ((const char *) buffer) + toWrite; 1234 userSize -= toWrite; 1235 written += toWrite; 1236 1237 releaseBuffer(&audioBuffer); 1238 } 1239 1240 return written; 1241} 1242 1243// ------------------------------------------------------------------------- 1244 1245TimedAudioTrack::TimedAudioTrack() { 1246 mIsTimed = true; 1247} 1248 1249status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1250{ 1251 AutoMutex lock(mLock); 1252 status_t result = UNKNOWN_ERROR; 1253 1254#if 1 1255 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1256 // while we are accessing the cblk 1257 sp<IAudioTrack> audioTrack = mAudioTrack; 1258 sp<IMemory> iMem = mCblkMemory; 1259#endif 1260 1261 // If the track is not invalid already, try to allocate a buffer. alloc 1262 // fails indicating that the server is dead, flag the track as invalid so 1263 // we can attempt to restore in just a bit. 1264 audio_track_cblk_t* cblk = mCblk; 1265 if (!(cblk->mFlags & CBLK_INVALID)) { 1266 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1267 if (result == DEAD_OBJECT) { 1268 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1269 } 1270 } 1271 1272 // If the track is invalid at this point, attempt to restore it. and try the 1273 // allocation one more time. 1274 if (cblk->mFlags & CBLK_INVALID) { 1275 result = restoreTrack_l("allocateTimedBuffer"); 1276 1277 if (result == NO_ERROR) { 1278 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1279 } 1280 } 1281 1282 return result; 1283} 1284 1285status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1286 int64_t pts) 1287{ 1288 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1289 { 1290 AutoMutex lock(mLock); 1291 audio_track_cblk_t* cblk = mCblk; 1292 // restart track if it was disabled by audioflinger due to previous underrun 1293 if (buffer->size() != 0 && status == NO_ERROR && 1294 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1295 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1296 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1297 // FIXME ignoring status 1298 mAudioTrack->start(); 1299 } 1300 } 1301 return status; 1302} 1303 1304status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1305 TargetTimeline target) 1306{ 1307 return mAudioTrack->setMediaTimeTransform(xform, target); 1308} 1309 1310// ------------------------------------------------------------------------- 1311 1312nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1313{ 1314 // Currently the AudioTrack thread is not created if there are no callbacks. 1315 // Would it ever make sense to run the thread, even without callbacks? 1316 // If so, then replace this by checks at each use for mCbf != NULL. 1317 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1318 1319 mLock.lock(); 1320 if (mAwaitBoost) { 1321 mAwaitBoost = false; 1322 mLock.unlock(); 1323 static const int32_t kMaxTries = 5; 1324 int32_t tryCounter = kMaxTries; 1325 uint32_t pollUs = 10000; 1326 do { 1327 int policy = sched_getscheduler(0); 1328 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1329 break; 1330 } 1331 usleep(pollUs); 1332 pollUs <<= 1; 1333 } while (tryCounter-- > 0); 1334 if (tryCounter < 0) { 1335 ALOGE("did not receive expected priority boost on time"); 1336 } 1337 // Run again immediately 1338 return 0; 1339 } 1340 1341 // Can only reference mCblk while locked 1342 int32_t flags = android_atomic_and( 1343 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1344 1345 // Check for track invalidation 1346 if (flags & CBLK_INVALID) { 1347 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1348 // AudioSystem cache. We should not exit here but after calling the callback so 1349 // that the upper layers can recreate the track 1350 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1351 status_t status = restoreTrack_l("processAudioBuffer"); 1352 mLock.unlock(); 1353 // Run again immediately, but with a new IAudioTrack 1354 return 0; 1355 } 1356 } 1357 1358 bool waitStreamEnd = mState == STATE_STOPPING; 1359 bool active = mState == STATE_ACTIVE; 1360 1361 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1362 bool newUnderrun = false; 1363 if (flags & CBLK_UNDERRUN) { 1364#if 0 1365 // Currently in shared buffer mode, when the server reaches the end of buffer, 1366 // the track stays active in continuous underrun state. It's up to the application 1367 // to pause or stop the track, or set the position to a new offset within buffer. 1368 // This was some experimental code to auto-pause on underrun. Keeping it here 1369 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1370 if (mTransfer == TRANSFER_SHARED) { 1371 mState = STATE_PAUSED; 1372 active = false; 1373 } 1374#endif 1375 if (!mInUnderrun) { 1376 mInUnderrun = true; 1377 newUnderrun = true; 1378 } 1379 } 1380 1381 // Get current position of server 1382 size_t position = mProxy->getPosition(); 1383 1384 // Manage marker callback 1385 bool markerReached = false; 1386 size_t markerPosition = mMarkerPosition; 1387 // FIXME fails for wraparound, need 64 bits 1388 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1389 mMarkerReached = markerReached = true; 1390 } 1391 1392 // Determine number of new position callback(s) that will be needed, while locked 1393 size_t newPosCount = 0; 1394 size_t newPosition = mNewPosition; 1395 size_t updatePeriod = mUpdatePeriod; 1396 // FIXME fails for wraparound, need 64 bits 1397 if (updatePeriod > 0 && position >= newPosition) { 1398 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1399 mNewPosition += updatePeriod * newPosCount; 1400 } 1401 1402 // Cache other fields that will be needed soon 1403 uint32_t loopPeriod = mLoopPeriod; 1404 uint32_t sampleRate = mSampleRate; 1405 size_t notificationFrames = mNotificationFramesAct; 1406 if (mRefreshRemaining) { 1407 mRefreshRemaining = false; 1408 mRemainingFrames = notificationFrames; 1409 mRetryOnPartialBuffer = false; 1410 } 1411 size_t misalignment = mProxy->getMisalignment(); 1412 uint32_t sequence = mSequence; 1413 1414 // These fields don't need to be cached, because they are assigned only by set(): 1415 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1416 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1417 1418 mLock.unlock(); 1419 1420 if (waitStreamEnd) { 1421 AutoMutex lock(mLock); 1422 1423 sp<AudioTrackClientProxy> proxy = mProxy; 1424 sp<IMemory> iMem = mCblkMemory; 1425 1426 struct timespec timeout; 1427 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1428 timeout.tv_nsec = 0; 1429 1430 mLock.unlock(); 1431 status_t status = mProxy->waitStreamEndDone(&timeout); 1432 mLock.lock(); 1433 switch (status) { 1434 case NO_ERROR: 1435 case DEAD_OBJECT: 1436 case TIMED_OUT: 1437 mLock.unlock(); 1438 mCbf(EVENT_STREAM_END, mUserData, NULL); 1439 mLock.lock(); 1440 if (mState == STATE_STOPPING) { 1441 mState = STATE_STOPPED; 1442 if (status != DEAD_OBJECT) { 1443 return NS_INACTIVE; 1444 } 1445 } 1446 return 0; 1447 default: 1448 return 0; 1449 } 1450 } 1451 1452 // perform callbacks while unlocked 1453 if (newUnderrun) { 1454 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1455 } 1456 // FIXME we will miss loops if loop cycle was signaled several times since last call 1457 // to processAudioBuffer() 1458 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1459 mCbf(EVENT_LOOP_END, mUserData, NULL); 1460 } 1461 if (flags & CBLK_BUFFER_END) { 1462 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1463 } 1464 if (markerReached) { 1465 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1466 } 1467 while (newPosCount > 0) { 1468 size_t temp = newPosition; 1469 mCbf(EVENT_NEW_POS, mUserData, &temp); 1470 newPosition += updatePeriod; 1471 newPosCount--; 1472 } 1473 1474 if (mObservedSequence != sequence) { 1475 mObservedSequence = sequence; 1476 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1477 // for offloaded tracks, just wait for the upper layers to recreate the track 1478 if (isOffloaded()) { 1479 return NS_INACTIVE; 1480 } 1481 } 1482 1483 // if inactive, then don't run me again until re-started 1484 if (!active) { 1485 return NS_INACTIVE; 1486 } 1487 1488 // Compute the estimated time until the next timed event (position, markers, loops) 1489 // FIXME only for non-compressed audio 1490 uint32_t minFrames = ~0; 1491 if (!markerReached && position < markerPosition) { 1492 minFrames = markerPosition - position; 1493 } 1494 if (loopPeriod > 0 && loopPeriod < minFrames) { 1495 minFrames = loopPeriod; 1496 } 1497 if (updatePeriod > 0 && updatePeriod < minFrames) { 1498 minFrames = updatePeriod; 1499 } 1500 1501 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1502 static const uint32_t kPoll = 0; 1503 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1504 minFrames = kPoll * notificationFrames; 1505 } 1506 1507 // Convert frame units to time units 1508 nsecs_t ns = NS_WHENEVER; 1509 if (minFrames != (uint32_t) ~0) { 1510 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1511 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1512 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1513 } 1514 1515 // If not supplying data by EVENT_MORE_DATA, then we're done 1516 if (mTransfer != TRANSFER_CALLBACK) { 1517 return ns; 1518 } 1519 1520 struct timespec timeout; 1521 const struct timespec *requested = &ClientProxy::kForever; 1522 if (ns != NS_WHENEVER) { 1523 timeout.tv_sec = ns / 1000000000LL; 1524 timeout.tv_nsec = ns % 1000000000LL; 1525 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1526 requested = &timeout; 1527 } 1528 1529 while (mRemainingFrames > 0) { 1530 1531 Buffer audioBuffer; 1532 audioBuffer.frameCount = mRemainingFrames; 1533 size_t nonContig; 1534 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1535 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1536 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1537 requested = &ClientProxy::kNonBlocking; 1538 size_t avail = audioBuffer.frameCount + nonContig; 1539 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1540 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1541 if (err != NO_ERROR) { 1542 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1543 (isOffloaded() && (err == DEAD_OBJECT))) { 1544 return 0; 1545 } 1546 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1547 return NS_NEVER; 1548 } 1549 1550 if (mRetryOnPartialBuffer) { 1551 mRetryOnPartialBuffer = false; 1552 if (avail < mRemainingFrames) { 1553 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1554 if (ns < 0 || myns < ns) { 1555 ns = myns; 1556 } 1557 return ns; 1558 } 1559 } 1560 1561 // Divide buffer size by 2 to take into account the expansion 1562 // due to 8 to 16 bit conversion: the callback must fill only half 1563 // of the destination buffer 1564 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1565 audioBuffer.size >>= 1; 1566 } 1567 1568 size_t reqSize = audioBuffer.size; 1569 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1570 size_t writtenSize = audioBuffer.size; 1571 size_t writtenFrames = writtenSize / mFrameSize; 1572 1573 // Sanity check on returned size 1574 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1575 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1576 reqSize, (int) writtenSize); 1577 return NS_NEVER; 1578 } 1579 1580 if (writtenSize == 0) { 1581 // The callback is done filling buffers 1582 // Keep this thread going to handle timed events and 1583 // still try to get more data in intervals of WAIT_PERIOD_MS 1584 // but don't just loop and block the CPU, so wait 1585 return WAIT_PERIOD_MS * 1000000LL; 1586 } 1587 1588 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1589 // 8 to 16 bit conversion, note that source and destination are the same address 1590 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1591 audioBuffer.size <<= 1; 1592 } 1593 1594 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1595 audioBuffer.frameCount = releasedFrames; 1596 mRemainingFrames -= releasedFrames; 1597 if (misalignment >= releasedFrames) { 1598 misalignment -= releasedFrames; 1599 } else { 1600 misalignment = 0; 1601 } 1602 1603 releaseBuffer(&audioBuffer); 1604 1605 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1606 // if callback doesn't like to accept the full chunk 1607 if (writtenSize < reqSize) { 1608 continue; 1609 } 1610 1611 // There could be enough non-contiguous frames available to satisfy the remaining request 1612 if (mRemainingFrames <= nonContig) { 1613 continue; 1614 } 1615 1616#if 0 1617 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1618 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1619 // that total to a sum == notificationFrames. 1620 if (0 < misalignment && misalignment <= mRemainingFrames) { 1621 mRemainingFrames = misalignment; 1622 return (mRemainingFrames * 1100000000LL) / sampleRate; 1623 } 1624#endif 1625 1626 } 1627 mRemainingFrames = notificationFrames; 1628 mRetryOnPartialBuffer = true; 1629 1630 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1631 return 0; 1632} 1633 1634status_t AudioTrack::restoreTrack_l(const char *from) 1635{ 1636 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1637 isOffloaded() ? "Offloaded" : "PCM", from); 1638 ++mSequence; 1639 status_t result; 1640 1641 // refresh the audio configuration cache in this process to make sure we get new 1642 // output parameters in getOutput_l() and createTrack_l() 1643 AudioSystem::clearAudioConfigCache(); 1644 1645 if (isOffloaded()) { 1646 return DEAD_OBJECT; 1647 } 1648 1649 // force new output query from audio policy manager; 1650 mOutput = 0; 1651 audio_io_handle_t output = getOutput_l(); 1652 1653 // if the new IAudioTrack is created, createTrack_l() will modify the 1654 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1655 // It will also delete the strong references on previous IAudioTrack and IMemory 1656 size_t position = mProxy->getPosition(); 1657 mNewPosition = position + mUpdatePeriod; 1658 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1659 result = createTrack_l(mStreamType, 1660 mSampleRate, 1661 mFormat, 1662 mReqFrameCount, // so that frame count never goes down 1663 mFlags, 1664 mSharedBuffer, 1665 output, 1666 position /*epoch*/); 1667 1668 if (result == NO_ERROR) { 1669 // continue playback from last known position, but 1670 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1671 if (mStaticProxy != NULL) { 1672 mLoopPeriod = 0; 1673 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1674 } 1675 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1676 // track destruction have been played? This is critical for SoundPool implementation 1677 // This must be broken, and needs to be tested/debugged. 1678#if 0 1679 // restore write index and set other indexes to reflect empty buffer status 1680 if (!strcmp(from, "start")) { 1681 // Make sure that a client relying on callback events indicating underrun or 1682 // the actual amount of audio frames played (e.g SoundPool) receives them. 1683 if (mSharedBuffer == 0) { 1684 // restart playback even if buffer is not completely filled. 1685 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1686 } 1687 } 1688#endif 1689 if (mState == STATE_ACTIVE) { 1690 result = mAudioTrack->start(); 1691 } 1692 } 1693 if (result != NO_ERROR) { 1694 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1695 // As getOutput was called above and resulted in an output stream to be opened, 1696 // we need to release it. 1697 AudioSystem::releaseOutput(output); 1698 ALOGW("restoreTrack_l() failed status %d", result); 1699 mState = STATE_STOPPED; 1700 } 1701 1702 return result; 1703} 1704 1705status_t AudioTrack::setParameters(const String8& keyValuePairs) 1706{ 1707 AutoMutex lock(mLock); 1708 if (mAudioTrack != 0) { 1709 return mAudioTrack->setParameters(keyValuePairs); 1710 } else { 1711 return NO_INIT; 1712 } 1713} 1714 1715String8 AudioTrack::getParameters(const String8& keys) 1716{ 1717 if (mOutput) { 1718 return AudioSystem::getParameters(mOutput, keys); 1719 } else { 1720 return String8::empty(); 1721 } 1722} 1723 1724status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1725{ 1726 1727 const size_t SIZE = 256; 1728 char buffer[SIZE]; 1729 String8 result; 1730 1731 result.append(" AudioTrack::dump\n"); 1732 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1733 mVolume[0], mVolume[1]); 1734 result.append(buffer); 1735 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1736 mChannelCount, mFrameCount); 1737 result.append(buffer); 1738 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1739 result.append(buffer); 1740 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1741 result.append(buffer); 1742 ::write(fd, result.string(), result.size()); 1743 return NO_ERROR; 1744} 1745 1746uint32_t AudioTrack::getUnderrunFrames() const 1747{ 1748 AutoMutex lock(mLock); 1749 return mProxy->getUnderrunFrames(); 1750} 1751 1752// ========================================================================= 1753 1754void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1755{ 1756 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1757 if (audioTrack != 0) { 1758 AutoMutex lock(audioTrack->mLock); 1759 audioTrack->mProxy->binderDied(); 1760 } 1761} 1762 1763// ========================================================================= 1764 1765AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1766 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 1767{ 1768} 1769 1770AudioTrack::AudioTrackThread::~AudioTrackThread() 1771{ 1772} 1773 1774bool AudioTrack::AudioTrackThread::threadLoop() 1775{ 1776 { 1777 AutoMutex _l(mMyLock); 1778 if (mPaused) { 1779 mMyCond.wait(mMyLock); 1780 // caller will check for exitPending() 1781 return true; 1782 } 1783 } 1784 nsecs_t ns = mReceiver.processAudioBuffer(this); 1785 switch (ns) { 1786 case 0: 1787 return true; 1788 case NS_WHENEVER: 1789 sleep(1); 1790 return true; 1791 case NS_INACTIVE: 1792 pauseConditional(); 1793 return true; 1794 case NS_NEVER: 1795 return false; 1796 default: 1797 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1798 struct timespec req; 1799 req.tv_sec = ns / 1000000000LL; 1800 req.tv_nsec = ns % 1000000000LL; 1801 nanosleep(&req, NULL /*rem*/); 1802 return true; 1803 } 1804} 1805 1806void AudioTrack::AudioTrackThread::requestExit() 1807{ 1808 // must be in this order to avoid a race condition 1809 Thread::requestExit(); 1810 resume(); 1811} 1812 1813void AudioTrack::AudioTrackThread::pause() 1814{ 1815 AutoMutex _l(mMyLock); 1816 mPaused = true; 1817 mResumeLatch = false; 1818} 1819 1820void AudioTrack::AudioTrackThread::pauseConditional() 1821{ 1822 AutoMutex _l(mMyLock); 1823 if (mResumeLatch) { 1824 mResumeLatch = false; 1825 } else { 1826 mPaused = true; 1827 } 1828} 1829 1830void AudioTrack::AudioTrackThread::resume() 1831{ 1832 AutoMutex _l(mMyLock); 1833 if (mPaused) { 1834 mPaused = false; 1835 mResumeLatch = false; 1836 mMyCond.signal(); 1837 } else { 1838 mResumeLatch = true; 1839 } 1840} 1841 1842}; // namespace android 1843