AudioTrack.cpp revision 396fabdb6efcdac5aea3d9f559d1beedf6a4cedc
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31#define WAIT_STREAM_END_TIMEOUT_SEC 120 32 33 34namespace android { 35// --------------------------------------------------------------------------- 36 37// static 38status_t AudioTrack::getMinFrameCount( 39 size_t* frameCount, 40 audio_stream_type_t streamType, 41 uint32_t sampleRate) 42{ 43 if (frameCount == NULL) { 44 return BAD_VALUE; 45 } 46 47 // FIXME merge with similar code in createTrack_l(), except we're missing 48 // some information here that is available in createTrack_l(): 49 // audio_io_handle_t output 50 // audio_format_t format 51 // audio_channel_mask_t channelMask 52 // audio_output_flags_t flags 53 uint32_t afSampleRate; 54 status_t status; 55 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 56 if (status != NO_ERROR) { 57 return status; 58 } 59 size_t afFrameCount; 60 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 61 if (status != NO_ERROR) { 62 return status; 63 } 64 uint32_t afLatency; 65 status = AudioSystem::getOutputLatency(&afLatency, streamType); 66 if (status != NO_ERROR) { 67 return status; 68 } 69 70 // Ensure that buffer depth covers at least audio hardware latency 71 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 72 if (minBufCount < 2) { 73 minBufCount = 2; 74 } 75 76 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 77 afFrameCount * minBufCount * sampleRate / afSampleRate; 78 // The formula above should always produce a non-zero value, but return an error 79 // in the unlikely event that it does not, as that's part of the API contract. 80 if (*frameCount == 0) { 81 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 82 streamType, sampleRate); 83 return BAD_VALUE; 84 } 85 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 86 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 87 return NO_ERROR; 88} 89 90// --------------------------------------------------------------------------- 91 92AudioTrack::AudioTrack() 93 : mStatus(NO_INIT), 94 mIsTimed(false), 95 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 96 mPreviousSchedulingGroup(SP_DEFAULT) 97{ 98} 99 100AudioTrack::AudioTrack( 101 audio_stream_type_t streamType, 102 uint32_t sampleRate, 103 audio_format_t format, 104 audio_channel_mask_t channelMask, 105 int frameCount, 106 audio_output_flags_t flags, 107 callback_t cbf, 108 void* user, 109 int notificationFrames, 110 int sessionId, 111 transfer_type transferType, 112 const audio_offload_info_t *offloadInfo, 113 int uid) 114 : mStatus(NO_INIT), 115 mIsTimed(false), 116 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 117 mPreviousSchedulingGroup(SP_DEFAULT) 118{ 119 mStatus = set(streamType, sampleRate, format, channelMask, 120 frameCount, flags, cbf, user, notificationFrames, 121 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 122 offloadInfo, uid); 123} 124 125AudioTrack::AudioTrack( 126 audio_stream_type_t streamType, 127 uint32_t sampleRate, 128 audio_format_t format, 129 audio_channel_mask_t channelMask, 130 const sp<IMemory>& sharedBuffer, 131 audio_output_flags_t flags, 132 callback_t cbf, 133 void* user, 134 int notificationFrames, 135 int sessionId, 136 transfer_type transferType, 137 const audio_offload_info_t *offloadInfo, 138 int uid) 139 : mStatus(NO_INIT), 140 mIsTimed(false), 141 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 142 mPreviousSchedulingGroup(SP_DEFAULT) 143{ 144 mStatus = set(streamType, sampleRate, format, channelMask, 145 0 /*frameCount*/, flags, cbf, user, notificationFrames, 146 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); 147} 148 149AudioTrack::~AudioTrack() 150{ 151 if (mStatus == NO_ERROR) { 152 // Make sure that callback function exits in the case where 153 // it is looping on buffer full condition in obtainBuffer(). 154 // Otherwise the callback thread will never exit. 155 stop(); 156 if (mAudioTrackThread != 0) { 157 mProxy->interrupt(); 158 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 159 mAudioTrackThread->requestExitAndWait(); 160 mAudioTrackThread.clear(); 161 } 162 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 163 mAudioTrack.clear(); 164 IPCThreadState::self()->flushCommands(); 165 AudioSystem::releaseAudioSessionId(mSessionId); 166 } 167} 168 169status_t AudioTrack::set( 170 audio_stream_type_t streamType, 171 uint32_t sampleRate, 172 audio_format_t format, 173 audio_channel_mask_t channelMask, 174 int frameCountInt, 175 audio_output_flags_t flags, 176 callback_t cbf, 177 void* user, 178 int notificationFrames, 179 const sp<IMemory>& sharedBuffer, 180 bool threadCanCallJava, 181 int sessionId, 182 transfer_type transferType, 183 const audio_offload_info_t *offloadInfo, 184 int uid) 185{ 186 switch (transferType) { 187 case TRANSFER_DEFAULT: 188 if (sharedBuffer != 0) { 189 transferType = TRANSFER_SHARED; 190 } else if (cbf == NULL || threadCanCallJava) { 191 transferType = TRANSFER_SYNC; 192 } else { 193 transferType = TRANSFER_CALLBACK; 194 } 195 break; 196 case TRANSFER_CALLBACK: 197 if (cbf == NULL || sharedBuffer != 0) { 198 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 199 return BAD_VALUE; 200 } 201 break; 202 case TRANSFER_OBTAIN: 203 case TRANSFER_SYNC: 204 if (sharedBuffer != 0) { 205 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 206 return BAD_VALUE; 207 } 208 break; 209 case TRANSFER_SHARED: 210 if (sharedBuffer == 0) { 211 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 212 return BAD_VALUE; 213 } 214 break; 215 default: 216 ALOGE("Invalid transfer type %d", transferType); 217 return BAD_VALUE; 218 } 219 mTransfer = transferType; 220 221 // FIXME "int" here is legacy and will be replaced by size_t later 222 if (frameCountInt < 0) { 223 ALOGE("Invalid frame count %d", frameCountInt); 224 return BAD_VALUE; 225 } 226 size_t frameCount = frameCountInt; 227 228 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 229 sharedBuffer->size()); 230 231 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 232 233 AutoMutex lock(mLock); 234 235 // invariant that mAudioTrack != 0 is true only after set() returns successfully 236 if (mAudioTrack != 0) { 237 ALOGE("Track already in use"); 238 return INVALID_OPERATION; 239 } 240 241 mOutput = 0; 242 243 // handle default values first. 244 if (streamType == AUDIO_STREAM_DEFAULT) { 245 streamType = AUDIO_STREAM_MUSIC; 246 } 247 248 if (sampleRate == 0) { 249 uint32_t afSampleRate; 250 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 251 return NO_INIT; 252 } 253 sampleRate = afSampleRate; 254 } 255 mSampleRate = sampleRate; 256 257 // these below should probably come from the audioFlinger too... 258 if (format == AUDIO_FORMAT_DEFAULT) { 259 format = AUDIO_FORMAT_PCM_16_BIT; 260 } 261 262 // validate parameters 263 if (!audio_is_valid_format(format)) { 264 ALOGE("Invalid format %d", format); 265 return BAD_VALUE; 266 } 267 268 if (!audio_is_output_channel(channelMask)) { 269 ALOGE("Invalid channel mask %#x", channelMask); 270 return BAD_VALUE; 271 } 272 273 // AudioFlinger does not currently support 8-bit data in shared memory 274 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 275 ALOGE("8-bit data in shared memory is not supported"); 276 return BAD_VALUE; 277 } 278 279 // force direct flag if format is not linear PCM 280 // or offload was requested 281 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 282 || !audio_is_linear_pcm(format)) { 283 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 284 ? "Offload request, forcing to Direct Output" 285 : "Not linear PCM, forcing to Direct Output"); 286 flags = (audio_output_flags_t) 287 // FIXME why can't we allow direct AND fast? 288 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 289 } 290 // only allow deep buffering for music stream type 291 if (streamType != AUDIO_STREAM_MUSIC) { 292 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 293 } 294 295 mChannelMask = channelMask; 296 uint32_t channelCount = popcount(channelMask); 297 mChannelCount = channelCount; 298 299 if (audio_is_linear_pcm(format)) { 300 mFrameSize = channelCount * audio_bytes_per_sample(format); 301 mFrameSizeAF = channelCount * sizeof(int16_t); 302 } else { 303 mFrameSize = sizeof(uint8_t); 304 mFrameSizeAF = sizeof(uint8_t); 305 } 306 307 audio_io_handle_t output = AudioSystem::getOutput( 308 streamType, 309 sampleRate, format, channelMask, 310 flags, 311 offloadInfo); 312 313 if (output == 0) { 314 ALOGE("Could not get audio output for stream type %d", streamType); 315 return BAD_VALUE; 316 } 317 318 mVolume[LEFT] = 1.0f; 319 mVolume[RIGHT] = 1.0f; 320 mSendLevel = 0.0f; 321 // mFrameCount is initialized in createTrack_l 322 mReqFrameCount = frameCount; 323 mNotificationFramesReq = notificationFrames; 324 mNotificationFramesAct = 0; 325 mSessionId = sessionId; 326 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 327 mClientUid = IPCThreadState::self()->getCallingUid(); 328 } else { 329 mClientUid = uid; 330 } 331 mAuxEffectId = 0; 332 mFlags = flags; 333 mCbf = cbf; 334 335 if (cbf != NULL) { 336 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 337 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 338 } 339 340 // create the IAudioTrack 341 status_t status = createTrack_l(streamType, 342 sampleRate, 343 format, 344 frameCount, 345 flags, 346 sharedBuffer, 347 output, 348 0 /*epoch*/); 349 350 if (status != NO_ERROR) { 351 if (mAudioTrackThread != 0) { 352 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 353 mAudioTrackThread->requestExitAndWait(); 354 mAudioTrackThread.clear(); 355 } 356 //Use of direct and offloaded output streams is ref counted by audio policy manager. 357 // As getOutput was called above and resulted in an output stream to be opened, 358 // we need to release it. 359 AudioSystem::releaseOutput(output); 360 return status; 361 } 362 363 mStatus = NO_ERROR; 364 mStreamType = streamType; 365 mFormat = format; 366 mSharedBuffer = sharedBuffer; 367 mState = STATE_STOPPED; 368 mUserData = user; 369 mLoopPeriod = 0; 370 mMarkerPosition = 0; 371 mMarkerReached = false; 372 mNewPosition = 0; 373 mUpdatePeriod = 0; 374 AudioSystem::acquireAudioSessionId(mSessionId); 375 mSequence = 1; 376 mObservedSequence = mSequence; 377 mInUnderrun = false; 378 mOutput = output; 379 380 return NO_ERROR; 381} 382 383// ------------------------------------------------------------------------- 384 385status_t AudioTrack::start() 386{ 387 AutoMutex lock(mLock); 388 389 if (mState == STATE_ACTIVE) { 390 return INVALID_OPERATION; 391 } 392 393 mInUnderrun = true; 394 395 State previousState = mState; 396 if (previousState == STATE_PAUSED_STOPPING) { 397 mState = STATE_STOPPING; 398 } else { 399 mState = STATE_ACTIVE; 400 } 401 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 402 // reset current position as seen by client to 0 403 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 404 // force refresh of remaining frames by processAudioBuffer() as last 405 // write before stop could be partial. 406 mRefreshRemaining = true; 407 } 408 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 409 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 410 411 sp<AudioTrackThread> t = mAudioTrackThread; 412 if (t != 0) { 413 if (previousState == STATE_STOPPING) { 414 mProxy->interrupt(); 415 } else { 416 t->resume(); 417 } 418 } else { 419 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 420 get_sched_policy(0, &mPreviousSchedulingGroup); 421 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 422 } 423 424 status_t status = NO_ERROR; 425 if (!(flags & CBLK_INVALID)) { 426 status = mAudioTrack->start(); 427 if (status == DEAD_OBJECT) { 428 flags |= CBLK_INVALID; 429 } 430 } 431 if (flags & CBLK_INVALID) { 432 status = restoreTrack_l("start"); 433 } 434 435 if (status != NO_ERROR) { 436 ALOGE("start() status %d", status); 437 mState = previousState; 438 if (t != 0) { 439 if (previousState != STATE_STOPPING) { 440 t->pause(); 441 } 442 } else { 443 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 444 set_sched_policy(0, mPreviousSchedulingGroup); 445 } 446 } 447 448 return status; 449} 450 451void AudioTrack::stop() 452{ 453 AutoMutex lock(mLock); 454 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 455 return; 456 } 457 458 if (isOffloaded()) { 459 mState = STATE_STOPPING; 460 } else { 461 mState = STATE_STOPPED; 462 } 463 464 mProxy->interrupt(); 465 mAudioTrack->stop(); 466 // the playback head position will reset to 0, so if a marker is set, we need 467 // to activate it again 468 mMarkerReached = false; 469#if 0 470 // Force flush if a shared buffer is used otherwise audioflinger 471 // will not stop before end of buffer is reached. 472 // It may be needed to make sure that we stop playback, likely in case looping is on. 473 if (mSharedBuffer != 0) { 474 flush_l(); 475 } 476#endif 477 478 sp<AudioTrackThread> t = mAudioTrackThread; 479 if (t != 0) { 480 if (!isOffloaded()) { 481 t->pause(); 482 } 483 } else { 484 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 485 set_sched_policy(0, mPreviousSchedulingGroup); 486 } 487} 488 489bool AudioTrack::stopped() const 490{ 491 AutoMutex lock(mLock); 492 return mState != STATE_ACTIVE; 493} 494 495void AudioTrack::flush() 496{ 497 if (mSharedBuffer != 0) { 498 return; 499 } 500 AutoMutex lock(mLock); 501 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 502 return; 503 } 504 flush_l(); 505} 506 507void AudioTrack::flush_l() 508{ 509 ALOG_ASSERT(mState != STATE_ACTIVE); 510 511 // clear playback marker and periodic update counter 512 mMarkerPosition = 0; 513 mMarkerReached = false; 514 mUpdatePeriod = 0; 515 mRefreshRemaining = true; 516 517 mState = STATE_FLUSHED; 518 if (isOffloaded()) { 519 mProxy->interrupt(); 520 } 521 mProxy->flush(); 522 mAudioTrack->flush(); 523} 524 525void AudioTrack::pause() 526{ 527 AutoMutex lock(mLock); 528 if (mState == STATE_ACTIVE) { 529 mState = STATE_PAUSED; 530 } else if (mState == STATE_STOPPING) { 531 mState = STATE_PAUSED_STOPPING; 532 } else { 533 return; 534 } 535 mProxy->interrupt(); 536 mAudioTrack->pause(); 537} 538 539status_t AudioTrack::setVolume(float left, float right) 540{ 541 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 542 return BAD_VALUE; 543 } 544 545 AutoMutex lock(mLock); 546 mVolume[LEFT] = left; 547 mVolume[RIGHT] = right; 548 549 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 550 551 if (isOffloaded()) { 552 mAudioTrack->signal(); 553 } 554 return NO_ERROR; 555} 556 557status_t AudioTrack::setVolume(float volume) 558{ 559 return setVolume(volume, volume); 560} 561 562status_t AudioTrack::setAuxEffectSendLevel(float level) 563{ 564 if (level < 0.0f || level > 1.0f) { 565 return BAD_VALUE; 566 } 567 568 AutoMutex lock(mLock); 569 mSendLevel = level; 570 mProxy->setSendLevel(level); 571 572 return NO_ERROR; 573} 574 575void AudioTrack::getAuxEffectSendLevel(float* level) const 576{ 577 if (level != NULL) { 578 *level = mSendLevel; 579 } 580} 581 582status_t AudioTrack::setSampleRate(uint32_t rate) 583{ 584 if (mIsTimed || isOffloaded()) { 585 return INVALID_OPERATION; 586 } 587 588 uint32_t afSamplingRate; 589 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 590 return NO_INIT; 591 } 592 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 593 if (rate == 0 || rate > afSamplingRate*2 ) { 594 return BAD_VALUE; 595 } 596 597 AutoMutex lock(mLock); 598 mSampleRate = rate; 599 mProxy->setSampleRate(rate); 600 601 return NO_ERROR; 602} 603 604uint32_t AudioTrack::getSampleRate() const 605{ 606 if (mIsTimed) { 607 return 0; 608 } 609 610 AutoMutex lock(mLock); 611 612 // sample rate can be updated during playback by the offloaded decoder so we need to 613 // query the HAL and update if needed. 614// FIXME use Proxy return channel to update the rate from server and avoid polling here 615 if (isOffloaded()) { 616 if (mOutput != 0) { 617 uint32_t sampleRate = 0; 618 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 619 if (status == NO_ERROR) { 620 mSampleRate = sampleRate; 621 } 622 } 623 } 624 return mSampleRate; 625} 626 627status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 628{ 629 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 630 return INVALID_OPERATION; 631 } 632 633 if (loopCount == 0) { 634 ; 635 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 636 loopEnd - loopStart >= MIN_LOOP) { 637 ; 638 } else { 639 return BAD_VALUE; 640 } 641 642 AutoMutex lock(mLock); 643 // See setPosition() regarding setting parameters such as loop points or position while active 644 if (mState == STATE_ACTIVE) { 645 return INVALID_OPERATION; 646 } 647 setLoop_l(loopStart, loopEnd, loopCount); 648 return NO_ERROR; 649} 650 651void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 652{ 653 // FIXME If setting a loop also sets position to start of loop, then 654 // this is correct. Otherwise it should be removed. 655 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 656 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 657 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 658} 659 660status_t AudioTrack::setMarkerPosition(uint32_t marker) 661{ 662 // The only purpose of setting marker position is to get a callback 663 if (mCbf == NULL || isOffloaded()) { 664 return INVALID_OPERATION; 665 } 666 667 AutoMutex lock(mLock); 668 mMarkerPosition = marker; 669 mMarkerReached = false; 670 671 return NO_ERROR; 672} 673 674status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 675{ 676 if (isOffloaded()) { 677 return INVALID_OPERATION; 678 } 679 if (marker == NULL) { 680 return BAD_VALUE; 681 } 682 683 AutoMutex lock(mLock); 684 *marker = mMarkerPosition; 685 686 return NO_ERROR; 687} 688 689status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 690{ 691 // The only purpose of setting position update period is to get a callback 692 if (mCbf == NULL || isOffloaded()) { 693 return INVALID_OPERATION; 694 } 695 696 AutoMutex lock(mLock); 697 mNewPosition = mProxy->getPosition() + updatePeriod; 698 mUpdatePeriod = updatePeriod; 699 return NO_ERROR; 700} 701 702status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 703{ 704 if (isOffloaded()) { 705 return INVALID_OPERATION; 706 } 707 if (updatePeriod == NULL) { 708 return BAD_VALUE; 709 } 710 711 AutoMutex lock(mLock); 712 *updatePeriod = mUpdatePeriod; 713 714 return NO_ERROR; 715} 716 717status_t AudioTrack::setPosition(uint32_t position) 718{ 719 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 720 return INVALID_OPERATION; 721 } 722 if (position > mFrameCount) { 723 return BAD_VALUE; 724 } 725 726 AutoMutex lock(mLock); 727 // Currently we require that the player is inactive before setting parameters such as position 728 // or loop points. Otherwise, there could be a race condition: the application could read the 729 // current position, compute a new position or loop parameters, and then set that position or 730 // loop parameters but it would do the "wrong" thing since the position has continued to advance 731 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 732 // to specify how it wants to handle such scenarios. 733 if (mState == STATE_ACTIVE) { 734 return INVALID_OPERATION; 735 } 736 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 737 mLoopPeriod = 0; 738 // FIXME Check whether loops and setting position are incompatible in old code. 739 // If we use setLoop for both purposes we lose the capability to set the position while looping. 740 mStaticProxy->setLoop(position, mFrameCount, 0); 741 742 return NO_ERROR; 743} 744 745status_t AudioTrack::getPosition(uint32_t *position) const 746{ 747 if (position == NULL) { 748 return BAD_VALUE; 749 } 750 751 AutoMutex lock(mLock); 752 if (isOffloaded()) { 753 uint32_t dspFrames = 0; 754 755 if (mOutput != 0) { 756 uint32_t halFrames; 757 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 758 } 759 *position = dspFrames; 760 } else { 761 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 762 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 763 mProxy->getPosition(); 764 } 765 return NO_ERROR; 766} 767 768status_t AudioTrack::getBufferPosition(size_t *position) 769{ 770 if (mSharedBuffer == 0 || mIsTimed) { 771 return INVALID_OPERATION; 772 } 773 if (position == NULL) { 774 return BAD_VALUE; 775 } 776 777 AutoMutex lock(mLock); 778 *position = mStaticProxy->getBufferPosition(); 779 return NO_ERROR; 780} 781 782status_t AudioTrack::reload() 783{ 784 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 785 return INVALID_OPERATION; 786 } 787 788 AutoMutex lock(mLock); 789 // See setPosition() regarding setting parameters such as loop points or position while active 790 if (mState == STATE_ACTIVE) { 791 return INVALID_OPERATION; 792 } 793 mNewPosition = mUpdatePeriod; 794 mLoopPeriod = 0; 795 // FIXME The new code cannot reload while keeping a loop specified. 796 // Need to check how the old code handled this, and whether it's a significant change. 797 mStaticProxy->setLoop(0, mFrameCount, 0); 798 return NO_ERROR; 799} 800 801audio_io_handle_t AudioTrack::getOutput() 802{ 803 AutoMutex lock(mLock); 804 return mOutput; 805} 806 807// must be called with mLock held 808audio_io_handle_t AudioTrack::getOutput_l() 809{ 810 if (mOutput) { 811 return mOutput; 812 } else { 813 return AudioSystem::getOutput(mStreamType, 814 mSampleRate, mFormat, mChannelMask, mFlags); 815 } 816} 817 818status_t AudioTrack::attachAuxEffect(int effectId) 819{ 820 AutoMutex lock(mLock); 821 status_t status = mAudioTrack->attachAuxEffect(effectId); 822 if (status == NO_ERROR) { 823 mAuxEffectId = effectId; 824 } 825 return status; 826} 827 828// ------------------------------------------------------------------------- 829 830// must be called with mLock held 831status_t AudioTrack::createTrack_l( 832 audio_stream_type_t streamType, 833 uint32_t sampleRate, 834 audio_format_t format, 835 size_t frameCount, 836 audio_output_flags_t flags, 837 const sp<IMemory>& sharedBuffer, 838 audio_io_handle_t output, 839 size_t epoch) 840{ 841 status_t status; 842 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 843 if (audioFlinger == 0) { 844 ALOGE("Could not get audioflinger"); 845 return NO_INIT; 846 } 847 848 // Not all of these values are needed under all conditions, but it is easier to get them all 849 850 uint32_t afLatency; 851 status = AudioSystem::getLatency(output, streamType, &afLatency); 852 if (status != NO_ERROR) { 853 ALOGE("getLatency(%d) failed status %d", output, status); 854 return NO_INIT; 855 } 856 857 size_t afFrameCount; 858 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 859 if (status != NO_ERROR) { 860 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 861 return NO_INIT; 862 } 863 864 uint32_t afSampleRate; 865 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 866 if (status != NO_ERROR) { 867 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 868 return NO_INIT; 869 } 870 871 // Client decides whether the track is TIMED (see below), but can only express a preference 872 // for FAST. Server will perform additional tests. 873 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 874 // either of these use cases: 875 // use case 1: shared buffer 876 (sharedBuffer != 0) || 877 // use case 2: callback handler 878 (mCbf != NULL))) { 879 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 880 // once denied, do not request again if IAudioTrack is re-created 881 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 882 mFlags = flags; 883 } 884 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 885 886 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 887 // n = 1 fast track with single buffering; nBuffering is ignored 888 // n = 2 fast track with double buffering 889 // n = 2 normal track, no sample rate conversion 890 // n = 3 normal track, with sample rate conversion 891 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 892 // n > 3 very high latency or very small notification interval; nBuffering is ignored 893 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 894 895 mNotificationFramesAct = mNotificationFramesReq; 896 897 if (!audio_is_linear_pcm(format)) { 898 899 if (sharedBuffer != 0) { 900 // Same comment as below about ignoring frameCount parameter for set() 901 frameCount = sharedBuffer->size(); 902 } else if (frameCount == 0) { 903 frameCount = afFrameCount; 904 } 905 if (mNotificationFramesAct != frameCount) { 906 mNotificationFramesAct = frameCount; 907 } 908 } else if (sharedBuffer != 0) { 909 910 // Ensure that buffer alignment matches channel count 911 // 8-bit data in shared memory is not currently supported by AudioFlinger 912 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 913 if (mChannelCount > 1) { 914 // More than 2 channels does not require stronger alignment than stereo 915 alignment <<= 1; 916 } 917 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 918 ALOGE("Invalid buffer alignment: address %p, channel count %u", 919 sharedBuffer->pointer(), mChannelCount); 920 return BAD_VALUE; 921 } 922 923 // When initializing a shared buffer AudioTrack via constructors, 924 // there's no frameCount parameter. 925 // But when initializing a shared buffer AudioTrack via set(), 926 // there _is_ a frameCount parameter. We silently ignore it. 927 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 928 929 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 930 931 // FIXME move these calculations and associated checks to server 932 933 // Ensure that buffer depth covers at least audio hardware latency 934 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 935 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 936 afFrameCount, minBufCount, afSampleRate, afLatency); 937 if (minBufCount <= nBuffering) { 938 minBufCount = nBuffering; 939 } 940 941 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 942 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 943 ", afLatency=%d", 944 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 945 946 if (frameCount == 0) { 947 frameCount = minFrameCount; 948 } else if (frameCount < minFrameCount) { 949 // not ALOGW because it happens all the time when playing key clicks over A2DP 950 ALOGV("Minimum buffer size corrected from %d to %d", 951 frameCount, minFrameCount); 952 frameCount = minFrameCount; 953 } 954 // Make sure that application is notified with sufficient margin before underrun 955 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 956 mNotificationFramesAct = frameCount/nBuffering; 957 } 958 959 } else { 960 // For fast tracks, the frame count calculations and checks are done by server 961 } 962 963 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 964 if (mIsTimed) { 965 trackFlags |= IAudioFlinger::TRACK_TIMED; 966 } 967 968 pid_t tid = -1; 969 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 970 trackFlags |= IAudioFlinger::TRACK_FAST; 971 if (mAudioTrackThread != 0) { 972 tid = mAudioTrackThread->getTid(); 973 } 974 } 975 976 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 977 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 978 } 979 980 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 981 sampleRate, 982 // AudioFlinger only sees 16-bit PCM 983 format == AUDIO_FORMAT_PCM_8_BIT ? 984 AUDIO_FORMAT_PCM_16_BIT : format, 985 mChannelMask, 986 frameCount, 987 &trackFlags, 988 sharedBuffer, 989 output, 990 tid, 991 &mSessionId, 992 mName, 993 mClientUid, 994 &status); 995 996 if (track == 0) { 997 ALOGE("AudioFlinger could not create track, status: %d", status); 998 return status; 999 } 1000 sp<IMemory> iMem = track->getCblk(); 1001 if (iMem == 0) { 1002 ALOGE("Could not get control block"); 1003 return NO_INIT; 1004 } 1005 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1006 if (mAudioTrack != 0) { 1007 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1008 mDeathNotifier.clear(); 1009 } 1010 mAudioTrack = track; 1011 mCblkMemory = iMem; 1012 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1013 mCblk = cblk; 1014 size_t temp = cblk->frameCount_; 1015 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1016 // In current design, AudioTrack client checks and ensures frame count validity before 1017 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1018 // for fast track as it uses a special method of assigning frame count. 1019 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1020 } 1021 frameCount = temp; 1022 mAwaitBoost = false; 1023 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1024 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1025 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1026 mAwaitBoost = true; 1027 if (sharedBuffer == 0) { 1028 // Theoretically double-buffering is not required for fast tracks, 1029 // due to tighter scheduling. But in practice, to accommodate kernels with 1030 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1031 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1032 mNotificationFramesAct = frameCount/nBuffering; 1033 } 1034 } 1035 } else { 1036 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1037 // once denied, do not request again if IAudioTrack is re-created 1038 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1039 mFlags = flags; 1040 if (sharedBuffer == 0) { 1041 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1042 mNotificationFramesAct = frameCount/nBuffering; 1043 } 1044 } 1045 } 1046 } 1047 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1048 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1049 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1050 } else { 1051 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1052 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1053 mFlags = flags; 1054 return NO_INIT; 1055 } 1056 } 1057 1058 mRefreshRemaining = true; 1059 1060 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1061 // is the value of pointer() for the shared buffer, otherwise buffers points 1062 // immediately after the control block. This address is for the mapping within client 1063 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1064 void* buffers; 1065 if (sharedBuffer == 0) { 1066 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1067 } else { 1068 buffers = sharedBuffer->pointer(); 1069 } 1070 1071 mAudioTrack->attachAuxEffect(mAuxEffectId); 1072 // FIXME don't believe this lie 1073 mLatency = afLatency + (1000*frameCount) / sampleRate; 1074 mFrameCount = frameCount; 1075 // If IAudioTrack is re-created, don't let the requested frameCount 1076 // decrease. This can confuse clients that cache frameCount(). 1077 if (frameCount > mReqFrameCount) { 1078 mReqFrameCount = frameCount; 1079 } 1080 1081 // update proxy 1082 if (sharedBuffer == 0) { 1083 mStaticProxy.clear(); 1084 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1085 } else { 1086 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1087 mProxy = mStaticProxy; 1088 } 1089 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1090 uint16_t(mVolume[LEFT] * 0x1000)); 1091 mProxy->setSendLevel(mSendLevel); 1092 mProxy->setSampleRate(mSampleRate); 1093 mProxy->setEpoch(epoch); 1094 mProxy->setMinimum(mNotificationFramesAct); 1095 1096 mDeathNotifier = new DeathNotifier(this); 1097 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1098 1099 return NO_ERROR; 1100} 1101 1102status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1103{ 1104 if (audioBuffer == NULL) { 1105 return BAD_VALUE; 1106 } 1107 if (mTransfer != TRANSFER_OBTAIN) { 1108 audioBuffer->frameCount = 0; 1109 audioBuffer->size = 0; 1110 audioBuffer->raw = NULL; 1111 return INVALID_OPERATION; 1112 } 1113 1114 const struct timespec *requested; 1115 if (waitCount == -1) { 1116 requested = &ClientProxy::kForever; 1117 } else if (waitCount == 0) { 1118 requested = &ClientProxy::kNonBlocking; 1119 } else if (waitCount > 0) { 1120 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1121 struct timespec timeout; 1122 timeout.tv_sec = ms / 1000; 1123 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1124 requested = &timeout; 1125 } else { 1126 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1127 requested = NULL; 1128 } 1129 return obtainBuffer(audioBuffer, requested); 1130} 1131 1132status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1133 struct timespec *elapsed, size_t *nonContig) 1134{ 1135 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1136 uint32_t oldSequence = 0; 1137 uint32_t newSequence; 1138 1139 Proxy::Buffer buffer; 1140 status_t status = NO_ERROR; 1141 1142 static const int32_t kMaxTries = 5; 1143 int32_t tryCounter = kMaxTries; 1144 1145 do { 1146 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1147 // keep them from going away if another thread re-creates the track during obtainBuffer() 1148 sp<AudioTrackClientProxy> proxy; 1149 sp<IMemory> iMem; 1150 1151 { // start of lock scope 1152 AutoMutex lock(mLock); 1153 1154 newSequence = mSequence; 1155 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1156 if (status == DEAD_OBJECT) { 1157 // re-create track, unless someone else has already done so 1158 if (newSequence == oldSequence) { 1159 status = restoreTrack_l("obtainBuffer"); 1160 if (status != NO_ERROR) { 1161 buffer.mFrameCount = 0; 1162 buffer.mRaw = NULL; 1163 buffer.mNonContig = 0; 1164 break; 1165 } 1166 } 1167 } 1168 oldSequence = newSequence; 1169 1170 // Keep the extra references 1171 proxy = mProxy; 1172 iMem = mCblkMemory; 1173 1174 if (mState == STATE_STOPPING) { 1175 status = -EINTR; 1176 buffer.mFrameCount = 0; 1177 buffer.mRaw = NULL; 1178 buffer.mNonContig = 0; 1179 break; 1180 } 1181 1182 // Non-blocking if track is stopped or paused 1183 if (mState != STATE_ACTIVE) { 1184 requested = &ClientProxy::kNonBlocking; 1185 } 1186 1187 } // end of lock scope 1188 1189 buffer.mFrameCount = audioBuffer->frameCount; 1190 // FIXME starts the requested timeout and elapsed over from scratch 1191 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1192 1193 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1194 1195 audioBuffer->frameCount = buffer.mFrameCount; 1196 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1197 audioBuffer->raw = buffer.mRaw; 1198 if (nonContig != NULL) { 1199 *nonContig = buffer.mNonContig; 1200 } 1201 return status; 1202} 1203 1204void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1205{ 1206 if (mTransfer == TRANSFER_SHARED) { 1207 return; 1208 } 1209 1210 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1211 if (stepCount == 0) { 1212 return; 1213 } 1214 1215 Proxy::Buffer buffer; 1216 buffer.mFrameCount = stepCount; 1217 buffer.mRaw = audioBuffer->raw; 1218 1219 AutoMutex lock(mLock); 1220 mInUnderrun = false; 1221 mProxy->releaseBuffer(&buffer); 1222 1223 // restart track if it was disabled by audioflinger due to previous underrun 1224 if (mState == STATE_ACTIVE) { 1225 audio_track_cblk_t* cblk = mCblk; 1226 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1227 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1228 this, mName.string()); 1229 // FIXME ignoring status 1230 mAudioTrack->start(); 1231 } 1232 } 1233} 1234 1235// ------------------------------------------------------------------------- 1236 1237ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1238{ 1239 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1240 return INVALID_OPERATION; 1241 } 1242 1243 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1244 // Sanity-check: user is most-likely passing an error code, and it would 1245 // make the return value ambiguous (actualSize vs error). 1246 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1247 return BAD_VALUE; 1248 } 1249 1250 size_t written = 0; 1251 Buffer audioBuffer; 1252 1253 while (userSize >= mFrameSize) { 1254 audioBuffer.frameCount = userSize / mFrameSize; 1255 1256 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1257 if (err < 0) { 1258 if (written > 0) { 1259 break; 1260 } 1261 return ssize_t(err); 1262 } 1263 1264 size_t toWrite; 1265 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1266 // Divide capacity by 2 to take expansion into account 1267 toWrite = audioBuffer.size >> 1; 1268 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1269 } else { 1270 toWrite = audioBuffer.size; 1271 memcpy(audioBuffer.i8, buffer, toWrite); 1272 } 1273 buffer = ((const char *) buffer) + toWrite; 1274 userSize -= toWrite; 1275 written += toWrite; 1276 1277 releaseBuffer(&audioBuffer); 1278 } 1279 1280 return written; 1281} 1282 1283// ------------------------------------------------------------------------- 1284 1285TimedAudioTrack::TimedAudioTrack() { 1286 mIsTimed = true; 1287} 1288 1289status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1290{ 1291 AutoMutex lock(mLock); 1292 status_t result = UNKNOWN_ERROR; 1293 1294#if 1 1295 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1296 // while we are accessing the cblk 1297 sp<IAudioTrack> audioTrack = mAudioTrack; 1298 sp<IMemory> iMem = mCblkMemory; 1299#endif 1300 1301 // If the track is not invalid already, try to allocate a buffer. alloc 1302 // fails indicating that the server is dead, flag the track as invalid so 1303 // we can attempt to restore in just a bit. 1304 audio_track_cblk_t* cblk = mCblk; 1305 if (!(cblk->mFlags & CBLK_INVALID)) { 1306 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1307 if (result == DEAD_OBJECT) { 1308 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1309 } 1310 } 1311 1312 // If the track is invalid at this point, attempt to restore it. and try the 1313 // allocation one more time. 1314 if (cblk->mFlags & CBLK_INVALID) { 1315 result = restoreTrack_l("allocateTimedBuffer"); 1316 1317 if (result == NO_ERROR) { 1318 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1319 } 1320 } 1321 1322 return result; 1323} 1324 1325status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1326 int64_t pts) 1327{ 1328 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1329 { 1330 AutoMutex lock(mLock); 1331 audio_track_cblk_t* cblk = mCblk; 1332 // restart track if it was disabled by audioflinger due to previous underrun 1333 if (buffer->size() != 0 && status == NO_ERROR && 1334 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1335 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1336 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1337 // FIXME ignoring status 1338 mAudioTrack->start(); 1339 } 1340 } 1341 return status; 1342} 1343 1344status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1345 TargetTimeline target) 1346{ 1347 return mAudioTrack->setMediaTimeTransform(xform, target); 1348} 1349 1350// ------------------------------------------------------------------------- 1351 1352nsecs_t AudioTrack::processAudioBuffer() 1353{ 1354 // Currently the AudioTrack thread is not created if there are no callbacks. 1355 // Would it ever make sense to run the thread, even without callbacks? 1356 // If so, then replace this by checks at each use for mCbf != NULL. 1357 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1358 1359 mLock.lock(); 1360 if (mAwaitBoost) { 1361 mAwaitBoost = false; 1362 mLock.unlock(); 1363 static const int32_t kMaxTries = 5; 1364 int32_t tryCounter = kMaxTries; 1365 uint32_t pollUs = 10000; 1366 do { 1367 int policy = sched_getscheduler(0); 1368 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1369 break; 1370 } 1371 usleep(pollUs); 1372 pollUs <<= 1; 1373 } while (tryCounter-- > 0); 1374 if (tryCounter < 0) { 1375 ALOGE("did not receive expected priority boost on time"); 1376 } 1377 // Run again immediately 1378 return 0; 1379 } 1380 1381 // Can only reference mCblk while locked 1382 int32_t flags = android_atomic_and( 1383 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1384 1385 // Check for track invalidation 1386 if (flags & CBLK_INVALID) { 1387 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1388 // AudioSystem cache. We should not exit here but after calling the callback so 1389 // that the upper layers can recreate the track 1390 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1391 status_t status = restoreTrack_l("processAudioBuffer"); 1392 mLock.unlock(); 1393 // Run again immediately, but with a new IAudioTrack 1394 return 0; 1395 } 1396 } 1397 1398 bool waitStreamEnd = mState == STATE_STOPPING; 1399 bool active = mState == STATE_ACTIVE; 1400 1401 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1402 bool newUnderrun = false; 1403 if (flags & CBLK_UNDERRUN) { 1404#if 0 1405 // Currently in shared buffer mode, when the server reaches the end of buffer, 1406 // the track stays active in continuous underrun state. It's up to the application 1407 // to pause or stop the track, or set the position to a new offset within buffer. 1408 // This was some experimental code to auto-pause on underrun. Keeping it here 1409 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1410 if (mTransfer == TRANSFER_SHARED) { 1411 mState = STATE_PAUSED; 1412 active = false; 1413 } 1414#endif 1415 if (!mInUnderrun) { 1416 mInUnderrun = true; 1417 newUnderrun = true; 1418 } 1419 } 1420 1421 // Get current position of server 1422 size_t position = mProxy->getPosition(); 1423 1424 // Manage marker callback 1425 bool markerReached = false; 1426 size_t markerPosition = mMarkerPosition; 1427 // FIXME fails for wraparound, need 64 bits 1428 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1429 mMarkerReached = markerReached = true; 1430 } 1431 1432 // Determine number of new position callback(s) that will be needed, while locked 1433 size_t newPosCount = 0; 1434 size_t newPosition = mNewPosition; 1435 size_t updatePeriod = mUpdatePeriod; 1436 // FIXME fails for wraparound, need 64 bits 1437 if (updatePeriod > 0 && position >= newPosition) { 1438 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1439 mNewPosition += updatePeriod * newPosCount; 1440 } 1441 1442 // Cache other fields that will be needed soon 1443 uint32_t loopPeriod = mLoopPeriod; 1444 uint32_t sampleRate = mSampleRate; 1445 size_t notificationFrames = mNotificationFramesAct; 1446 if (mRefreshRemaining) { 1447 mRefreshRemaining = false; 1448 mRemainingFrames = notificationFrames; 1449 mRetryOnPartialBuffer = false; 1450 } 1451 size_t misalignment = mProxy->getMisalignment(); 1452 uint32_t sequence = mSequence; 1453 1454 // These fields don't need to be cached, because they are assigned only by set(): 1455 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1456 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1457 1458 mLock.unlock(); 1459 1460 if (waitStreamEnd) { 1461 AutoMutex lock(mLock); 1462 1463 sp<AudioTrackClientProxy> proxy = mProxy; 1464 sp<IMemory> iMem = mCblkMemory; 1465 1466 struct timespec timeout; 1467 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1468 timeout.tv_nsec = 0; 1469 1470 mLock.unlock(); 1471 status_t status = mProxy->waitStreamEndDone(&timeout); 1472 mLock.lock(); 1473 switch (status) { 1474 case NO_ERROR: 1475 case DEAD_OBJECT: 1476 case TIMED_OUT: 1477 mLock.unlock(); 1478 mCbf(EVENT_STREAM_END, mUserData, NULL); 1479 mLock.lock(); 1480 if (mState == STATE_STOPPING) { 1481 mState = STATE_STOPPED; 1482 if (status != DEAD_OBJECT) { 1483 return NS_INACTIVE; 1484 } 1485 } 1486 return 0; 1487 default: 1488 return 0; 1489 } 1490 } 1491 1492 // perform callbacks while unlocked 1493 if (newUnderrun) { 1494 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1495 } 1496 // FIXME we will miss loops if loop cycle was signaled several times since last call 1497 // to processAudioBuffer() 1498 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1499 mCbf(EVENT_LOOP_END, mUserData, NULL); 1500 } 1501 if (flags & CBLK_BUFFER_END) { 1502 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1503 } 1504 if (markerReached) { 1505 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1506 } 1507 while (newPosCount > 0) { 1508 size_t temp = newPosition; 1509 mCbf(EVENT_NEW_POS, mUserData, &temp); 1510 newPosition += updatePeriod; 1511 newPosCount--; 1512 } 1513 1514 if (mObservedSequence != sequence) { 1515 mObservedSequence = sequence; 1516 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1517 // for offloaded tracks, just wait for the upper layers to recreate the track 1518 if (isOffloaded()) { 1519 return NS_INACTIVE; 1520 } 1521 } 1522 1523 // if inactive, then don't run me again until re-started 1524 if (!active) { 1525 return NS_INACTIVE; 1526 } 1527 1528 // Compute the estimated time until the next timed event (position, markers, loops) 1529 // FIXME only for non-compressed audio 1530 uint32_t minFrames = ~0; 1531 if (!markerReached && position < markerPosition) { 1532 minFrames = markerPosition - position; 1533 } 1534 if (loopPeriod > 0 && loopPeriod < minFrames) { 1535 minFrames = loopPeriod; 1536 } 1537 if (updatePeriod > 0 && updatePeriod < minFrames) { 1538 minFrames = updatePeriod; 1539 } 1540 1541 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1542 static const uint32_t kPoll = 0; 1543 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1544 minFrames = kPoll * notificationFrames; 1545 } 1546 1547 // Convert frame units to time units 1548 nsecs_t ns = NS_WHENEVER; 1549 if (minFrames != (uint32_t) ~0) { 1550 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1551 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1552 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1553 } 1554 1555 // If not supplying data by EVENT_MORE_DATA, then we're done 1556 if (mTransfer != TRANSFER_CALLBACK) { 1557 return ns; 1558 } 1559 1560 struct timespec timeout; 1561 const struct timespec *requested = &ClientProxy::kForever; 1562 if (ns != NS_WHENEVER) { 1563 timeout.tv_sec = ns / 1000000000LL; 1564 timeout.tv_nsec = ns % 1000000000LL; 1565 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1566 requested = &timeout; 1567 } 1568 1569 while (mRemainingFrames > 0) { 1570 1571 Buffer audioBuffer; 1572 audioBuffer.frameCount = mRemainingFrames; 1573 size_t nonContig; 1574 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1575 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1576 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1577 requested = &ClientProxy::kNonBlocking; 1578 size_t avail = audioBuffer.frameCount + nonContig; 1579 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1580 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1581 if (err != NO_ERROR) { 1582 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1583 (isOffloaded() && (err == DEAD_OBJECT))) { 1584 return 0; 1585 } 1586 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1587 return NS_NEVER; 1588 } 1589 1590 if (mRetryOnPartialBuffer && !isOffloaded()) { 1591 mRetryOnPartialBuffer = false; 1592 if (avail < mRemainingFrames) { 1593 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1594 if (ns < 0 || myns < ns) { 1595 ns = myns; 1596 } 1597 return ns; 1598 } 1599 } 1600 1601 // Divide buffer size by 2 to take into account the expansion 1602 // due to 8 to 16 bit conversion: the callback must fill only half 1603 // of the destination buffer 1604 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1605 audioBuffer.size >>= 1; 1606 } 1607 1608 size_t reqSize = audioBuffer.size; 1609 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1610 size_t writtenSize = audioBuffer.size; 1611 size_t writtenFrames = writtenSize / mFrameSize; 1612 1613 // Sanity check on returned size 1614 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1615 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1616 reqSize, (int) writtenSize); 1617 return NS_NEVER; 1618 } 1619 1620 if (writtenSize == 0) { 1621 // The callback is done filling buffers 1622 // Keep this thread going to handle timed events and 1623 // still try to get more data in intervals of WAIT_PERIOD_MS 1624 // but don't just loop and block the CPU, so wait 1625 return WAIT_PERIOD_MS * 1000000LL; 1626 } 1627 1628 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1629 // 8 to 16 bit conversion, note that source and destination are the same address 1630 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1631 audioBuffer.size <<= 1; 1632 } 1633 1634 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1635 audioBuffer.frameCount = releasedFrames; 1636 mRemainingFrames -= releasedFrames; 1637 if (misalignment >= releasedFrames) { 1638 misalignment -= releasedFrames; 1639 } else { 1640 misalignment = 0; 1641 } 1642 1643 releaseBuffer(&audioBuffer); 1644 1645 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1646 // if callback doesn't like to accept the full chunk 1647 if (writtenSize < reqSize) { 1648 continue; 1649 } 1650 1651 // There could be enough non-contiguous frames available to satisfy the remaining request 1652 if (mRemainingFrames <= nonContig) { 1653 continue; 1654 } 1655 1656#if 0 1657 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1658 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1659 // that total to a sum == notificationFrames. 1660 if (0 < misalignment && misalignment <= mRemainingFrames) { 1661 mRemainingFrames = misalignment; 1662 return (mRemainingFrames * 1100000000LL) / sampleRate; 1663 } 1664#endif 1665 1666 } 1667 mRemainingFrames = notificationFrames; 1668 mRetryOnPartialBuffer = true; 1669 1670 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1671 return 0; 1672} 1673 1674status_t AudioTrack::restoreTrack_l(const char *from) 1675{ 1676 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1677 isOffloaded() ? "Offloaded" : "PCM", from); 1678 ++mSequence; 1679 status_t result; 1680 1681 // refresh the audio configuration cache in this process to make sure we get new 1682 // output parameters in getOutput_l() and createTrack_l() 1683 AudioSystem::clearAudioConfigCache(); 1684 1685 if (isOffloaded()) { 1686 return DEAD_OBJECT; 1687 } 1688 1689 // force new output query from audio policy manager; 1690 mOutput = 0; 1691 audio_io_handle_t output = getOutput_l(); 1692 1693 // if the new IAudioTrack is created, createTrack_l() will modify the 1694 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1695 // It will also delete the strong references on previous IAudioTrack and IMemory 1696 1697 // take the frames that will be lost by track recreation into account in saved position 1698 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1699 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1700 result = createTrack_l(mStreamType, 1701 mSampleRate, 1702 mFormat, 1703 mReqFrameCount, // so that frame count never goes down 1704 mFlags, 1705 mSharedBuffer, 1706 output, 1707 position /*epoch*/); 1708 1709 if (result == NO_ERROR) { 1710 // continue playback from last known position, but 1711 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1712 if (mStaticProxy != NULL) { 1713 mLoopPeriod = 0; 1714 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1715 } 1716 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1717 // track destruction have been played? This is critical for SoundPool implementation 1718 // This must be broken, and needs to be tested/debugged. 1719#if 0 1720 // restore write index and set other indexes to reflect empty buffer status 1721 if (!strcmp(from, "start")) { 1722 // Make sure that a client relying on callback events indicating underrun or 1723 // the actual amount of audio frames played (e.g SoundPool) receives them. 1724 if (mSharedBuffer == 0) { 1725 // restart playback even if buffer is not completely filled. 1726 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1727 } 1728 } 1729#endif 1730 if (mState == STATE_ACTIVE) { 1731 result = mAudioTrack->start(); 1732 } 1733 } 1734 if (result != NO_ERROR) { 1735 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1736 // As getOutput was called above and resulted in an output stream to be opened, 1737 // we need to release it. 1738 AudioSystem::releaseOutput(output); 1739 ALOGW("restoreTrack_l() failed status %d", result); 1740 mState = STATE_STOPPED; 1741 } 1742 1743 return result; 1744} 1745 1746status_t AudioTrack::setParameters(const String8& keyValuePairs) 1747{ 1748 AutoMutex lock(mLock); 1749 return mAudioTrack->setParameters(keyValuePairs); 1750} 1751 1752status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1753{ 1754 AutoMutex lock(mLock); 1755 // FIXME not implemented for fast tracks; should use proxy and SSQ 1756 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1757 return INVALID_OPERATION; 1758 } 1759 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1760 return INVALID_OPERATION; 1761 } 1762 status_t status = mAudioTrack->getTimestamp(timestamp); 1763 if (status == NO_ERROR) { 1764 timestamp.mPosition += mProxy->getEpoch(); 1765 } 1766 return status; 1767} 1768 1769String8 AudioTrack::getParameters(const String8& keys) 1770{ 1771 if (mOutput) { 1772 return AudioSystem::getParameters(mOutput, keys); 1773 } else { 1774 return String8::empty(); 1775 } 1776} 1777 1778status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1779{ 1780 1781 const size_t SIZE = 256; 1782 char buffer[SIZE]; 1783 String8 result; 1784 1785 result.append(" AudioTrack::dump\n"); 1786 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1787 mVolume[0], mVolume[1]); 1788 result.append(buffer); 1789 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1790 mChannelCount, mFrameCount); 1791 result.append(buffer); 1792 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1793 result.append(buffer); 1794 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1795 result.append(buffer); 1796 ::write(fd, result.string(), result.size()); 1797 return NO_ERROR; 1798} 1799 1800uint32_t AudioTrack::getUnderrunFrames() const 1801{ 1802 AutoMutex lock(mLock); 1803 return mProxy->getUnderrunFrames(); 1804} 1805 1806// ========================================================================= 1807 1808void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1809{ 1810 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1811 if (audioTrack != 0) { 1812 AutoMutex lock(audioTrack->mLock); 1813 audioTrack->mProxy->binderDied(); 1814 } 1815} 1816 1817// ========================================================================= 1818 1819AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1820 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1821 mIgnoreNextPausedInt(false) 1822{ 1823} 1824 1825AudioTrack::AudioTrackThread::~AudioTrackThread() 1826{ 1827} 1828 1829bool AudioTrack::AudioTrackThread::threadLoop() 1830{ 1831 { 1832 AutoMutex _l(mMyLock); 1833 if (mPaused) { 1834 mMyCond.wait(mMyLock); 1835 // caller will check for exitPending() 1836 return true; 1837 } 1838 if (mIgnoreNextPausedInt) { 1839 mIgnoreNextPausedInt = false; 1840 mPausedInt = false; 1841 } 1842 if (mPausedInt) { 1843 if (mPausedNs > 0) { 1844 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1845 } else { 1846 mMyCond.wait(mMyLock); 1847 } 1848 mPausedInt = false; 1849 return true; 1850 } 1851 } 1852 nsecs_t ns = mReceiver.processAudioBuffer(); 1853 switch (ns) { 1854 case 0: 1855 return true; 1856 case NS_INACTIVE: 1857 pauseInternal(); 1858 return true; 1859 case NS_NEVER: 1860 return false; 1861 case NS_WHENEVER: 1862 // FIXME increase poll interval, or make event-driven 1863 ns = 1000000000LL; 1864 // fall through 1865 default: 1866 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1867 pauseInternal(ns); 1868 return true; 1869 } 1870} 1871 1872void AudioTrack::AudioTrackThread::requestExit() 1873{ 1874 // must be in this order to avoid a race condition 1875 Thread::requestExit(); 1876 resume(); 1877} 1878 1879void AudioTrack::AudioTrackThread::pause() 1880{ 1881 AutoMutex _l(mMyLock); 1882 mPaused = true; 1883} 1884 1885void AudioTrack::AudioTrackThread::resume() 1886{ 1887 AutoMutex _l(mMyLock); 1888 mIgnoreNextPausedInt = true; 1889 if (mPaused || mPausedInt) { 1890 mPaused = false; 1891 mPausedInt = false; 1892 mMyCond.signal(); 1893 } 1894} 1895 1896void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1897{ 1898 AutoMutex _l(mMyLock); 1899 mPausedInt = true; 1900 mPausedNs = ns; 1901} 1902 1903}; // namespace android 1904