AudioTrack.cpp revision 23a7545c4de71e989c2d8ebf1d5b9dcf463c36a9
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 status_t status; 249 if (sampleRate == 0) { 250 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 251 if (status != NO_ERROR) { 252 ALOGE("Could not get output sample rate for stream type %d; status %d", 253 streamType, status); 254 return status; 255 } 256 } 257 mSampleRate = sampleRate; 258 259 // these below should probably come from the audioFlinger too... 260 if (format == AUDIO_FORMAT_DEFAULT) { 261 format = AUDIO_FORMAT_PCM_16_BIT; 262 } 263 264 // validate parameters 265 if (!audio_is_valid_format(format)) { 266 ALOGE("Invalid format %d", format); 267 return BAD_VALUE; 268 } 269 270 if (!audio_is_output_channel(channelMask)) { 271 ALOGE("Invalid channel mask %#x", channelMask); 272 return BAD_VALUE; 273 } 274 275 // AudioFlinger does not currently support 8-bit data in shared memory 276 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 277 ALOGE("8-bit data in shared memory is not supported"); 278 return BAD_VALUE; 279 } 280 281 // force direct flag if format is not linear PCM 282 // or offload was requested 283 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 284 || !audio_is_linear_pcm(format)) { 285 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 286 ? "Offload request, forcing to Direct Output" 287 : "Not linear PCM, forcing to Direct Output"); 288 flags = (audio_output_flags_t) 289 // FIXME why can't we allow direct AND fast? 290 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 291 } 292 // only allow deep buffering for music stream type 293 if (streamType != AUDIO_STREAM_MUSIC) { 294 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 295 } 296 297 mChannelMask = channelMask; 298 uint32_t channelCount = popcount(channelMask); 299 mChannelCount = channelCount; 300 301 if (audio_is_linear_pcm(format)) { 302 mFrameSize = channelCount * audio_bytes_per_sample(format); 303 mFrameSizeAF = channelCount * sizeof(int16_t); 304 } else { 305 mFrameSize = sizeof(uint8_t); 306 mFrameSizeAF = sizeof(uint8_t); 307 } 308 309 audio_io_handle_t output = AudioSystem::getOutput( 310 streamType, 311 sampleRate, format, channelMask, 312 flags, 313 offloadInfo); 314 315 if (output == 0) { 316 ALOGE("Could not get audio output for stream type %d", streamType); 317 return BAD_VALUE; 318 } 319 320 mVolume[LEFT] = 1.0f; 321 mVolume[RIGHT] = 1.0f; 322 mSendLevel = 0.0f; 323 mFrameCount = frameCount; 324 mReqFrameCount = frameCount; 325 mNotificationFramesReq = notificationFrames; 326 mNotificationFramesAct = 0; 327 mSessionId = sessionId; 328 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 329 mClientUid = IPCThreadState::self()->getCallingUid(); 330 } else { 331 mClientUid = uid; 332 } 333 mAuxEffectId = 0; 334 mFlags = flags; 335 mCbf = cbf; 336 337 if (cbf != NULL) { 338 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 339 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 340 } 341 342 // create the IAudioTrack 343 status = createTrack_l(streamType, 344 sampleRate, 345 format, 346 frameCount, 347 flags, 348 sharedBuffer, 349 output, 350 0 /*epoch*/); 351 352 if (status != NO_ERROR) { 353 if (mAudioTrackThread != 0) { 354 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 355 mAudioTrackThread->requestExitAndWait(); 356 mAudioTrackThread.clear(); 357 } 358 //Use of direct and offloaded output streams is ref counted by audio policy manager. 359 // As getOutput was called above and resulted in an output stream to be opened, 360 // we need to release it. 361 AudioSystem::releaseOutput(output); 362 return status; 363 } 364 365 mStatus = NO_ERROR; 366 mStreamType = streamType; 367 mFormat = format; 368 mSharedBuffer = sharedBuffer; 369 mState = STATE_STOPPED; 370 mUserData = user; 371 mLoopPeriod = 0; 372 mMarkerPosition = 0; 373 mMarkerReached = false; 374 mNewPosition = 0; 375 mUpdatePeriod = 0; 376 AudioSystem::acquireAudioSessionId(mSessionId); 377 mSequence = 1; 378 mObservedSequence = mSequence; 379 mInUnderrun = false; 380 mOutput = output; 381 382 return NO_ERROR; 383} 384 385// ------------------------------------------------------------------------- 386 387status_t AudioTrack::start() 388{ 389 AutoMutex lock(mLock); 390 391 if (mState == STATE_ACTIVE) { 392 return INVALID_OPERATION; 393 } 394 395 mInUnderrun = true; 396 397 State previousState = mState; 398 if (previousState == STATE_PAUSED_STOPPING) { 399 mState = STATE_STOPPING; 400 } else { 401 mState = STATE_ACTIVE; 402 } 403 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 404 // reset current position as seen by client to 0 405 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 406 // force refresh of remaining frames by processAudioBuffer() as last 407 // write before stop could be partial. 408 mRefreshRemaining = true; 409 } 410 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 411 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 412 413 sp<AudioTrackThread> t = mAudioTrackThread; 414 if (t != 0) { 415 if (previousState == STATE_STOPPING) { 416 mProxy->interrupt(); 417 } else { 418 t->resume(); 419 } 420 } else { 421 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 422 get_sched_policy(0, &mPreviousSchedulingGroup); 423 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 424 } 425 426 status_t status = NO_ERROR; 427 if (!(flags & CBLK_INVALID)) { 428 status = mAudioTrack->start(); 429 if (status == DEAD_OBJECT) { 430 flags |= CBLK_INVALID; 431 } 432 } 433 if (flags & CBLK_INVALID) { 434 status = restoreTrack_l("start"); 435 } 436 437 if (status != NO_ERROR) { 438 ALOGE("start() status %d", status); 439 mState = previousState; 440 if (t != 0) { 441 if (previousState != STATE_STOPPING) { 442 t->pause(); 443 } 444 } else { 445 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 446 set_sched_policy(0, mPreviousSchedulingGroup); 447 } 448 } 449 450 return status; 451} 452 453void AudioTrack::stop() 454{ 455 AutoMutex lock(mLock); 456 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 457 return; 458 } 459 460 if (isOffloaded_l()) { 461 mState = STATE_STOPPING; 462 } else { 463 mState = STATE_STOPPED; 464 } 465 466 mProxy->interrupt(); 467 mAudioTrack->stop(); 468 // the playback head position will reset to 0, so if a marker is set, we need 469 // to activate it again 470 mMarkerReached = false; 471#if 0 472 // Force flush if a shared buffer is used otherwise audioflinger 473 // will not stop before end of buffer is reached. 474 // It may be needed to make sure that we stop playback, likely in case looping is on. 475 if (mSharedBuffer != 0) { 476 flush_l(); 477 } 478#endif 479 480 sp<AudioTrackThread> t = mAudioTrackThread; 481 if (t != 0) { 482 if (!isOffloaded_l()) { 483 t->pause(); 484 } 485 } else { 486 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 487 set_sched_policy(0, mPreviousSchedulingGroup); 488 } 489} 490 491bool AudioTrack::stopped() const 492{ 493 AutoMutex lock(mLock); 494 return mState != STATE_ACTIVE; 495} 496 497void AudioTrack::flush() 498{ 499 if (mSharedBuffer != 0) { 500 return; 501 } 502 AutoMutex lock(mLock); 503 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 504 return; 505 } 506 flush_l(); 507} 508 509void AudioTrack::flush_l() 510{ 511 ALOG_ASSERT(mState != STATE_ACTIVE); 512 513 // clear playback marker and periodic update counter 514 mMarkerPosition = 0; 515 mMarkerReached = false; 516 mUpdatePeriod = 0; 517 mRefreshRemaining = true; 518 519 mState = STATE_FLUSHED; 520 if (isOffloaded_l()) { 521 mProxy->interrupt(); 522 } 523 mProxy->flush(); 524 mAudioTrack->flush(); 525} 526 527void AudioTrack::pause() 528{ 529 AutoMutex lock(mLock); 530 if (mState == STATE_ACTIVE) { 531 mState = STATE_PAUSED; 532 } else if (mState == STATE_STOPPING) { 533 mState = STATE_PAUSED_STOPPING; 534 } else { 535 return; 536 } 537 mProxy->interrupt(); 538 mAudioTrack->pause(); 539} 540 541status_t AudioTrack::setVolume(float left, float right) 542{ 543 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 544 return BAD_VALUE; 545 } 546 547 AutoMutex lock(mLock); 548 mVolume[LEFT] = left; 549 mVolume[RIGHT] = right; 550 551 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 552 553 if (isOffloaded_l()) { 554 mAudioTrack->signal(); 555 } 556 return NO_ERROR; 557} 558 559status_t AudioTrack::setVolume(float volume) 560{ 561 return setVolume(volume, volume); 562} 563 564status_t AudioTrack::setAuxEffectSendLevel(float level) 565{ 566 if (level < 0.0f || level > 1.0f) { 567 return BAD_VALUE; 568 } 569 570 AutoMutex lock(mLock); 571 mSendLevel = level; 572 mProxy->setSendLevel(level); 573 574 return NO_ERROR; 575} 576 577void AudioTrack::getAuxEffectSendLevel(float* level) const 578{ 579 if (level != NULL) { 580 *level = mSendLevel; 581 } 582} 583 584status_t AudioTrack::setSampleRate(uint32_t rate) 585{ 586 if (mIsTimed || isOffloaded()) { 587 return INVALID_OPERATION; 588 } 589 590 uint32_t afSamplingRate; 591 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 592 return NO_INIT; 593 } 594 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 595 if (rate == 0 || rate > afSamplingRate*2 ) { 596 return BAD_VALUE; 597 } 598 599 AutoMutex lock(mLock); 600 mSampleRate = rate; 601 mProxy->setSampleRate(rate); 602 603 return NO_ERROR; 604} 605 606uint32_t AudioTrack::getSampleRate() const 607{ 608 if (mIsTimed) { 609 return 0; 610 } 611 612 AutoMutex lock(mLock); 613 614 // sample rate can be updated during playback by the offloaded decoder so we need to 615 // query the HAL and update if needed. 616// FIXME use Proxy return channel to update the rate from server and avoid polling here 617 if (isOffloaded_l()) { 618 if (mOutput != 0) { 619 uint32_t sampleRate = 0; 620 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 621 if (status == NO_ERROR) { 622 mSampleRate = sampleRate; 623 } 624 } 625 } 626 return mSampleRate; 627} 628 629status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 630{ 631 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 632 return INVALID_OPERATION; 633 } 634 635 if (loopCount == 0) { 636 ; 637 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 638 loopEnd - loopStart >= MIN_LOOP) { 639 ; 640 } else { 641 return BAD_VALUE; 642 } 643 644 AutoMutex lock(mLock); 645 // See setPosition() regarding setting parameters such as loop points or position while active 646 if (mState == STATE_ACTIVE) { 647 return INVALID_OPERATION; 648 } 649 setLoop_l(loopStart, loopEnd, loopCount); 650 return NO_ERROR; 651} 652 653void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 654{ 655 // FIXME If setting a loop also sets position to start of loop, then 656 // this is correct. Otherwise it should be removed. 657 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 658 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 659 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 660} 661 662status_t AudioTrack::setMarkerPosition(uint32_t marker) 663{ 664 // The only purpose of setting marker position is to get a callback 665 if (mCbf == NULL || isOffloaded()) { 666 return INVALID_OPERATION; 667 } 668 669 AutoMutex lock(mLock); 670 mMarkerPosition = marker; 671 mMarkerReached = false; 672 673 return NO_ERROR; 674} 675 676status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 677{ 678 if (isOffloaded()) { 679 return INVALID_OPERATION; 680 } 681 if (marker == NULL) { 682 return BAD_VALUE; 683 } 684 685 AutoMutex lock(mLock); 686 *marker = mMarkerPosition; 687 688 return NO_ERROR; 689} 690 691status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 692{ 693 // The only purpose of setting position update period is to get a callback 694 if (mCbf == NULL || isOffloaded()) { 695 return INVALID_OPERATION; 696 } 697 698 AutoMutex lock(mLock); 699 mNewPosition = mProxy->getPosition() + updatePeriod; 700 mUpdatePeriod = updatePeriod; 701 return NO_ERROR; 702} 703 704status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 705{ 706 if (isOffloaded()) { 707 return INVALID_OPERATION; 708 } 709 if (updatePeriod == NULL) { 710 return BAD_VALUE; 711 } 712 713 AutoMutex lock(mLock); 714 *updatePeriod = mUpdatePeriod; 715 716 return NO_ERROR; 717} 718 719status_t AudioTrack::setPosition(uint32_t position) 720{ 721 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 722 return INVALID_OPERATION; 723 } 724 if (position > mFrameCount) { 725 return BAD_VALUE; 726 } 727 728 AutoMutex lock(mLock); 729 // Currently we require that the player is inactive before setting parameters such as position 730 // or loop points. Otherwise, there could be a race condition: the application could read the 731 // current position, compute a new position or loop parameters, and then set that position or 732 // loop parameters but it would do the "wrong" thing since the position has continued to advance 733 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 734 // to specify how it wants to handle such scenarios. 735 if (mState == STATE_ACTIVE) { 736 return INVALID_OPERATION; 737 } 738 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 739 mLoopPeriod = 0; 740 // FIXME Check whether loops and setting position are incompatible in old code. 741 // If we use setLoop for both purposes we lose the capability to set the position while looping. 742 mStaticProxy->setLoop(position, mFrameCount, 0); 743 744 return NO_ERROR; 745} 746 747status_t AudioTrack::getPosition(uint32_t *position) const 748{ 749 if (position == NULL) { 750 return BAD_VALUE; 751 } 752 753 AutoMutex lock(mLock); 754 if (isOffloaded_l()) { 755 uint32_t dspFrames = 0; 756 757 if (mOutput != 0) { 758 uint32_t halFrames; 759 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 760 } 761 *position = dspFrames; 762 } else { 763 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 764 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 765 mProxy->getPosition(); 766 } 767 return NO_ERROR; 768} 769 770status_t AudioTrack::getBufferPosition(size_t *position) 771{ 772 if (mSharedBuffer == 0 || mIsTimed) { 773 return INVALID_OPERATION; 774 } 775 if (position == NULL) { 776 return BAD_VALUE; 777 } 778 779 AutoMutex lock(mLock); 780 *position = mStaticProxy->getBufferPosition(); 781 return NO_ERROR; 782} 783 784status_t AudioTrack::reload() 785{ 786 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 787 return INVALID_OPERATION; 788 } 789 790 AutoMutex lock(mLock); 791 // See setPosition() regarding setting parameters such as loop points or position while active 792 if (mState == STATE_ACTIVE) { 793 return INVALID_OPERATION; 794 } 795 mNewPosition = mUpdatePeriod; 796 mLoopPeriod = 0; 797 // FIXME The new code cannot reload while keeping a loop specified. 798 // Need to check how the old code handled this, and whether it's a significant change. 799 mStaticProxy->setLoop(0, mFrameCount, 0); 800 return NO_ERROR; 801} 802 803audio_io_handle_t AudioTrack::getOutput() 804{ 805 AutoMutex lock(mLock); 806 return mOutput; 807} 808 809// must be called with mLock held 810audio_io_handle_t AudioTrack::getOutput_l() 811{ 812 if (mOutput) { 813 return mOutput; 814 } else { 815 return AudioSystem::getOutput(mStreamType, 816 mSampleRate, mFormat, mChannelMask, mFlags); 817 } 818} 819 820status_t AudioTrack::attachAuxEffect(int effectId) 821{ 822 AutoMutex lock(mLock); 823 status_t status = mAudioTrack->attachAuxEffect(effectId); 824 if (status == NO_ERROR) { 825 mAuxEffectId = effectId; 826 } 827 return status; 828} 829 830// ------------------------------------------------------------------------- 831 832// must be called with mLock held 833status_t AudioTrack::createTrack_l( 834 audio_stream_type_t streamType, 835 uint32_t sampleRate, 836 audio_format_t format, 837 size_t frameCount, 838 audio_output_flags_t flags, 839 const sp<IMemory>& sharedBuffer, 840 audio_io_handle_t output, 841 size_t epoch) 842{ 843 status_t status; 844 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 845 if (audioFlinger == 0) { 846 ALOGE("Could not get audioflinger"); 847 return NO_INIT; 848 } 849 850 // Not all of these values are needed under all conditions, but it is easier to get them all 851 852 uint32_t afLatency; 853 status = AudioSystem::getLatency(output, streamType, &afLatency); 854 if (status != NO_ERROR) { 855 ALOGE("getLatency(%d) failed status %d", output, status); 856 return NO_INIT; 857 } 858 859 size_t afFrameCount; 860 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 861 if (status != NO_ERROR) { 862 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 863 return NO_INIT; 864 } 865 866 uint32_t afSampleRate; 867 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 868 if (status != NO_ERROR) { 869 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 870 return NO_INIT; 871 } 872 873 // Client decides whether the track is TIMED (see below), but can only express a preference 874 // for FAST. Server will perform additional tests. 875 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 876 // either of these use cases: 877 // use case 1: shared buffer 878 (sharedBuffer != 0) || 879 // use case 2: callback handler 880 (mCbf != NULL))) { 881 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 882 // once denied, do not request again if IAudioTrack is re-created 883 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 884 mFlags = flags; 885 } 886 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 887 888 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 889 // n = 1 fast track with single buffering; nBuffering is ignored 890 // n = 2 fast track with double buffering 891 // n = 2 normal track, no sample rate conversion 892 // n = 3 normal track, with sample rate conversion 893 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 894 // n > 3 very high latency or very small notification interval; nBuffering is ignored 895 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 896 897 mNotificationFramesAct = mNotificationFramesReq; 898 899 if (!audio_is_linear_pcm(format)) { 900 901 if (sharedBuffer != 0) { 902 // Same comment as below about ignoring frameCount parameter for set() 903 frameCount = sharedBuffer->size(); 904 } else if (frameCount == 0) { 905 frameCount = afFrameCount; 906 } 907 if (mNotificationFramesAct != frameCount) { 908 mNotificationFramesAct = frameCount; 909 } 910 } else if (sharedBuffer != 0) { 911 912 // Ensure that buffer alignment matches channel count 913 // 8-bit data in shared memory is not currently supported by AudioFlinger 914 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 915 if (mChannelCount > 1) { 916 // More than 2 channels does not require stronger alignment than stereo 917 alignment <<= 1; 918 } 919 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 920 ALOGE("Invalid buffer alignment: address %p, channel count %u", 921 sharedBuffer->pointer(), mChannelCount); 922 return BAD_VALUE; 923 } 924 925 // When initializing a shared buffer AudioTrack via constructors, 926 // there's no frameCount parameter. 927 // But when initializing a shared buffer AudioTrack via set(), 928 // there _is_ a frameCount parameter. We silently ignore it. 929 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 930 931 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 932 933 // FIXME move these calculations and associated checks to server 934 935 // Ensure that buffer depth covers at least audio hardware latency 936 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 937 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 938 afFrameCount, minBufCount, afSampleRate, afLatency); 939 if (minBufCount <= nBuffering) { 940 minBufCount = nBuffering; 941 } 942 943 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 944 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 945 ", afLatency=%d", 946 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 947 948 if (frameCount == 0) { 949 frameCount = minFrameCount; 950 } else if (frameCount < minFrameCount) { 951 // not ALOGW because it happens all the time when playing key clicks over A2DP 952 ALOGV("Minimum buffer size corrected from %d to %d", 953 frameCount, minFrameCount); 954 frameCount = minFrameCount; 955 } 956 // Make sure that application is notified with sufficient margin before underrun 957 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 958 mNotificationFramesAct = frameCount/nBuffering; 959 } 960 961 } else { 962 // For fast tracks, the frame count calculations and checks are done by server 963 } 964 965 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 966 if (mIsTimed) { 967 trackFlags |= IAudioFlinger::TRACK_TIMED; 968 } 969 970 pid_t tid = -1; 971 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 972 trackFlags |= IAudioFlinger::TRACK_FAST; 973 if (mAudioTrackThread != 0) { 974 tid = mAudioTrackThread->getTid(); 975 } 976 } 977 978 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 979 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 980 } 981 982 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 983 sampleRate, 984 // AudioFlinger only sees 16-bit PCM 985 format == AUDIO_FORMAT_PCM_8_BIT ? 986 AUDIO_FORMAT_PCM_16_BIT : format, 987 mChannelMask, 988 frameCount, 989 &trackFlags, 990 sharedBuffer, 991 output, 992 tid, 993 &mSessionId, 994 mName, 995 mClientUid, 996 &status); 997 998 if (track == 0) { 999 ALOGE("AudioFlinger could not create track, status: %d", status); 1000 return status; 1001 } 1002 sp<IMemory> iMem = track->getCblk(); 1003 if (iMem == 0) { 1004 ALOGE("Could not get control block"); 1005 return NO_INIT; 1006 } 1007 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1008 if (mAudioTrack != 0) { 1009 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1010 mDeathNotifier.clear(); 1011 } 1012 mAudioTrack = track; 1013 mCblkMemory = iMem; 1014 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1015 mCblk = cblk; 1016 size_t temp = cblk->frameCount_; 1017 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1018 // In current design, AudioTrack client checks and ensures frame count validity before 1019 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1020 // for fast track as it uses a special method of assigning frame count. 1021 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1022 } 1023 frameCount = temp; 1024 mAwaitBoost = false; 1025 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1026 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1027 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1028 mAwaitBoost = true; 1029 if (sharedBuffer == 0) { 1030 // Theoretically double-buffering is not required for fast tracks, 1031 // due to tighter scheduling. But in practice, to accommodate kernels with 1032 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1033 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1034 mNotificationFramesAct = frameCount/nBuffering; 1035 } 1036 } 1037 } else { 1038 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1039 // once denied, do not request again if IAudioTrack is re-created 1040 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1041 mFlags = flags; 1042 if (sharedBuffer == 0) { 1043 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1044 mNotificationFramesAct = frameCount/nBuffering; 1045 } 1046 } 1047 } 1048 } 1049 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1050 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1051 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1052 } else { 1053 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1054 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1055 mFlags = flags; 1056 return NO_INIT; 1057 } 1058 } 1059 1060 mRefreshRemaining = true; 1061 1062 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1063 // is the value of pointer() for the shared buffer, otherwise buffers points 1064 // immediately after the control block. This address is for the mapping within client 1065 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1066 void* buffers; 1067 if (sharedBuffer == 0) { 1068 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1069 } else { 1070 buffers = sharedBuffer->pointer(); 1071 } 1072 1073 mAudioTrack->attachAuxEffect(mAuxEffectId); 1074 // FIXME don't believe this lie 1075 mLatency = afLatency + (1000*frameCount) / sampleRate; 1076 mFrameCount = frameCount; 1077 // If IAudioTrack is re-created, don't let the requested frameCount 1078 // decrease. This can confuse clients that cache frameCount(). 1079 if (frameCount > mReqFrameCount) { 1080 mReqFrameCount = frameCount; 1081 } 1082 1083 // update proxy 1084 if (sharedBuffer == 0) { 1085 mStaticProxy.clear(); 1086 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1087 } else { 1088 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1089 mProxy = mStaticProxy; 1090 } 1091 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1092 uint16_t(mVolume[LEFT] * 0x1000)); 1093 mProxy->setSendLevel(mSendLevel); 1094 mProxy->setSampleRate(mSampleRate); 1095 mProxy->setEpoch(epoch); 1096 mProxy->setMinimum(mNotificationFramesAct); 1097 1098 mDeathNotifier = new DeathNotifier(this); 1099 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1100 1101 return NO_ERROR; 1102} 1103 1104status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1105{ 1106 if (audioBuffer == NULL) { 1107 return BAD_VALUE; 1108 } 1109 if (mTransfer != TRANSFER_OBTAIN) { 1110 audioBuffer->frameCount = 0; 1111 audioBuffer->size = 0; 1112 audioBuffer->raw = NULL; 1113 return INVALID_OPERATION; 1114 } 1115 1116 const struct timespec *requested; 1117 if (waitCount == -1) { 1118 requested = &ClientProxy::kForever; 1119 } else if (waitCount == 0) { 1120 requested = &ClientProxy::kNonBlocking; 1121 } else if (waitCount > 0) { 1122 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1123 struct timespec timeout; 1124 timeout.tv_sec = ms / 1000; 1125 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1126 requested = &timeout; 1127 } else { 1128 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1129 requested = NULL; 1130 } 1131 return obtainBuffer(audioBuffer, requested); 1132} 1133 1134status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1135 struct timespec *elapsed, size_t *nonContig) 1136{ 1137 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1138 uint32_t oldSequence = 0; 1139 uint32_t newSequence; 1140 1141 Proxy::Buffer buffer; 1142 status_t status = NO_ERROR; 1143 1144 static const int32_t kMaxTries = 5; 1145 int32_t tryCounter = kMaxTries; 1146 1147 do { 1148 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1149 // keep them from going away if another thread re-creates the track during obtainBuffer() 1150 sp<AudioTrackClientProxy> proxy; 1151 sp<IMemory> iMem; 1152 1153 { // start of lock scope 1154 AutoMutex lock(mLock); 1155 1156 newSequence = mSequence; 1157 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1158 if (status == DEAD_OBJECT) { 1159 // re-create track, unless someone else has already done so 1160 if (newSequence == oldSequence) { 1161 status = restoreTrack_l("obtainBuffer"); 1162 if (status != NO_ERROR) { 1163 buffer.mFrameCount = 0; 1164 buffer.mRaw = NULL; 1165 buffer.mNonContig = 0; 1166 break; 1167 } 1168 } 1169 } 1170 oldSequence = newSequence; 1171 1172 // Keep the extra references 1173 proxy = mProxy; 1174 iMem = mCblkMemory; 1175 1176 if (mState == STATE_STOPPING) { 1177 status = -EINTR; 1178 buffer.mFrameCount = 0; 1179 buffer.mRaw = NULL; 1180 buffer.mNonContig = 0; 1181 break; 1182 } 1183 1184 // Non-blocking if track is stopped or paused 1185 if (mState != STATE_ACTIVE) { 1186 requested = &ClientProxy::kNonBlocking; 1187 } 1188 1189 } // end of lock scope 1190 1191 buffer.mFrameCount = audioBuffer->frameCount; 1192 // FIXME starts the requested timeout and elapsed over from scratch 1193 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1194 1195 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1196 1197 audioBuffer->frameCount = buffer.mFrameCount; 1198 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1199 audioBuffer->raw = buffer.mRaw; 1200 if (nonContig != NULL) { 1201 *nonContig = buffer.mNonContig; 1202 } 1203 return status; 1204} 1205 1206void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1207{ 1208 if (mTransfer == TRANSFER_SHARED) { 1209 return; 1210 } 1211 1212 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1213 if (stepCount == 0) { 1214 return; 1215 } 1216 1217 Proxy::Buffer buffer; 1218 buffer.mFrameCount = stepCount; 1219 buffer.mRaw = audioBuffer->raw; 1220 1221 AutoMutex lock(mLock); 1222 mInUnderrun = false; 1223 mProxy->releaseBuffer(&buffer); 1224 1225 // restart track if it was disabled by audioflinger due to previous underrun 1226 if (mState == STATE_ACTIVE) { 1227 audio_track_cblk_t* cblk = mCblk; 1228 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1229 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1230 this, mName.string()); 1231 // FIXME ignoring status 1232 mAudioTrack->start(); 1233 } 1234 } 1235} 1236 1237// ------------------------------------------------------------------------- 1238 1239ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1240{ 1241 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1242 return INVALID_OPERATION; 1243 } 1244 1245 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1246 // Sanity-check: user is most-likely passing an error code, and it would 1247 // make the return value ambiguous (actualSize vs error). 1248 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1249 return BAD_VALUE; 1250 } 1251 1252 size_t written = 0; 1253 Buffer audioBuffer; 1254 1255 while (userSize >= mFrameSize) { 1256 audioBuffer.frameCount = userSize / mFrameSize; 1257 1258 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1259 if (err < 0) { 1260 if (written > 0) { 1261 break; 1262 } 1263 return ssize_t(err); 1264 } 1265 1266 size_t toWrite; 1267 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1268 // Divide capacity by 2 to take expansion into account 1269 toWrite = audioBuffer.size >> 1; 1270 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1271 } else { 1272 toWrite = audioBuffer.size; 1273 memcpy(audioBuffer.i8, buffer, toWrite); 1274 } 1275 buffer = ((const char *) buffer) + toWrite; 1276 userSize -= toWrite; 1277 written += toWrite; 1278 1279 releaseBuffer(&audioBuffer); 1280 } 1281 1282 return written; 1283} 1284 1285// ------------------------------------------------------------------------- 1286 1287TimedAudioTrack::TimedAudioTrack() { 1288 mIsTimed = true; 1289} 1290 1291status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1292{ 1293 AutoMutex lock(mLock); 1294 status_t result = UNKNOWN_ERROR; 1295 1296#if 1 1297 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1298 // while we are accessing the cblk 1299 sp<IAudioTrack> audioTrack = mAudioTrack; 1300 sp<IMemory> iMem = mCblkMemory; 1301#endif 1302 1303 // If the track is not invalid already, try to allocate a buffer. alloc 1304 // fails indicating that the server is dead, flag the track as invalid so 1305 // we can attempt to restore in just a bit. 1306 audio_track_cblk_t* cblk = mCblk; 1307 if (!(cblk->mFlags & CBLK_INVALID)) { 1308 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1309 if (result == DEAD_OBJECT) { 1310 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1311 } 1312 } 1313 1314 // If the track is invalid at this point, attempt to restore it. and try the 1315 // allocation one more time. 1316 if (cblk->mFlags & CBLK_INVALID) { 1317 result = restoreTrack_l("allocateTimedBuffer"); 1318 1319 if (result == NO_ERROR) { 1320 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1321 } 1322 } 1323 1324 return result; 1325} 1326 1327status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1328 int64_t pts) 1329{ 1330 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1331 { 1332 AutoMutex lock(mLock); 1333 audio_track_cblk_t* cblk = mCblk; 1334 // restart track if it was disabled by audioflinger due to previous underrun 1335 if (buffer->size() != 0 && status == NO_ERROR && 1336 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1337 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1338 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1339 // FIXME ignoring status 1340 mAudioTrack->start(); 1341 } 1342 } 1343 return status; 1344} 1345 1346status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1347 TargetTimeline target) 1348{ 1349 return mAudioTrack->setMediaTimeTransform(xform, target); 1350} 1351 1352// ------------------------------------------------------------------------- 1353 1354nsecs_t AudioTrack::processAudioBuffer() 1355{ 1356 // Currently the AudioTrack thread is not created if there are no callbacks. 1357 // Would it ever make sense to run the thread, even without callbacks? 1358 // If so, then replace this by checks at each use for mCbf != NULL. 1359 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1360 1361 mLock.lock(); 1362 if (mAwaitBoost) { 1363 mAwaitBoost = false; 1364 mLock.unlock(); 1365 static const int32_t kMaxTries = 5; 1366 int32_t tryCounter = kMaxTries; 1367 uint32_t pollUs = 10000; 1368 do { 1369 int policy = sched_getscheduler(0); 1370 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1371 break; 1372 } 1373 usleep(pollUs); 1374 pollUs <<= 1; 1375 } while (tryCounter-- > 0); 1376 if (tryCounter < 0) { 1377 ALOGE("did not receive expected priority boost on time"); 1378 } 1379 // Run again immediately 1380 return 0; 1381 } 1382 1383 // Can only reference mCblk while locked 1384 int32_t flags = android_atomic_and( 1385 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1386 1387 // Check for track invalidation 1388 if (flags & CBLK_INVALID) { 1389 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1390 // AudioSystem cache. We should not exit here but after calling the callback so 1391 // that the upper layers can recreate the track 1392 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1393 status_t status = restoreTrack_l("processAudioBuffer"); 1394 mLock.unlock(); 1395 // Run again immediately, but with a new IAudioTrack 1396 return 0; 1397 } 1398 } 1399 1400 bool waitStreamEnd = mState == STATE_STOPPING; 1401 bool active = mState == STATE_ACTIVE; 1402 1403 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1404 bool newUnderrun = false; 1405 if (flags & CBLK_UNDERRUN) { 1406#if 0 1407 // Currently in shared buffer mode, when the server reaches the end of buffer, 1408 // the track stays active in continuous underrun state. It's up to the application 1409 // to pause or stop the track, or set the position to a new offset within buffer. 1410 // This was some experimental code to auto-pause on underrun. Keeping it here 1411 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1412 if (mTransfer == TRANSFER_SHARED) { 1413 mState = STATE_PAUSED; 1414 active = false; 1415 } 1416#endif 1417 if (!mInUnderrun) { 1418 mInUnderrun = true; 1419 newUnderrun = true; 1420 } 1421 } 1422 1423 // Get current position of server 1424 size_t position = mProxy->getPosition(); 1425 1426 // Manage marker callback 1427 bool markerReached = false; 1428 size_t markerPosition = mMarkerPosition; 1429 // FIXME fails for wraparound, need 64 bits 1430 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1431 mMarkerReached = markerReached = true; 1432 } 1433 1434 // Determine number of new position callback(s) that will be needed, while locked 1435 size_t newPosCount = 0; 1436 size_t newPosition = mNewPosition; 1437 size_t updatePeriod = mUpdatePeriod; 1438 // FIXME fails for wraparound, need 64 bits 1439 if (updatePeriod > 0 && position >= newPosition) { 1440 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1441 mNewPosition += updatePeriod * newPosCount; 1442 } 1443 1444 // Cache other fields that will be needed soon 1445 uint32_t loopPeriod = mLoopPeriod; 1446 uint32_t sampleRate = mSampleRate; 1447 size_t notificationFrames = mNotificationFramesAct; 1448 if (mRefreshRemaining) { 1449 mRefreshRemaining = false; 1450 mRemainingFrames = notificationFrames; 1451 mRetryOnPartialBuffer = false; 1452 } 1453 size_t misalignment = mProxy->getMisalignment(); 1454 uint32_t sequence = mSequence; 1455 1456 // These fields don't need to be cached, because they are assigned only by set(): 1457 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1458 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1459 1460 mLock.unlock(); 1461 1462 if (waitStreamEnd) { 1463 AutoMutex lock(mLock); 1464 1465 sp<AudioTrackClientProxy> proxy = mProxy; 1466 sp<IMemory> iMem = mCblkMemory; 1467 1468 struct timespec timeout; 1469 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1470 timeout.tv_nsec = 0; 1471 1472 mLock.unlock(); 1473 status_t status = mProxy->waitStreamEndDone(&timeout); 1474 mLock.lock(); 1475 switch (status) { 1476 case NO_ERROR: 1477 case DEAD_OBJECT: 1478 case TIMED_OUT: 1479 mLock.unlock(); 1480 mCbf(EVENT_STREAM_END, mUserData, NULL); 1481 mLock.lock(); 1482 if (mState == STATE_STOPPING) { 1483 mState = STATE_STOPPED; 1484 if (status != DEAD_OBJECT) { 1485 return NS_INACTIVE; 1486 } 1487 } 1488 return 0; 1489 default: 1490 return 0; 1491 } 1492 } 1493 1494 // perform callbacks while unlocked 1495 if (newUnderrun) { 1496 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1497 } 1498 // FIXME we will miss loops if loop cycle was signaled several times since last call 1499 // to processAudioBuffer() 1500 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1501 mCbf(EVENT_LOOP_END, mUserData, NULL); 1502 } 1503 if (flags & CBLK_BUFFER_END) { 1504 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1505 } 1506 if (markerReached) { 1507 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1508 } 1509 while (newPosCount > 0) { 1510 size_t temp = newPosition; 1511 mCbf(EVENT_NEW_POS, mUserData, &temp); 1512 newPosition += updatePeriod; 1513 newPosCount--; 1514 } 1515 1516 if (mObservedSequence != sequence) { 1517 mObservedSequence = sequence; 1518 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1519 // for offloaded tracks, just wait for the upper layers to recreate the track 1520 if (isOffloaded()) { 1521 return NS_INACTIVE; 1522 } 1523 } 1524 1525 // if inactive, then don't run me again until re-started 1526 if (!active) { 1527 return NS_INACTIVE; 1528 } 1529 1530 // Compute the estimated time until the next timed event (position, markers, loops) 1531 // FIXME only for non-compressed audio 1532 uint32_t minFrames = ~0; 1533 if (!markerReached && position < markerPosition) { 1534 minFrames = markerPosition - position; 1535 } 1536 if (loopPeriod > 0 && loopPeriod < minFrames) { 1537 minFrames = loopPeriod; 1538 } 1539 if (updatePeriod > 0 && updatePeriod < minFrames) { 1540 minFrames = updatePeriod; 1541 } 1542 1543 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1544 static const uint32_t kPoll = 0; 1545 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1546 minFrames = kPoll * notificationFrames; 1547 } 1548 1549 // Convert frame units to time units 1550 nsecs_t ns = NS_WHENEVER; 1551 if (minFrames != (uint32_t) ~0) { 1552 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1553 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1554 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1555 } 1556 1557 // If not supplying data by EVENT_MORE_DATA, then we're done 1558 if (mTransfer != TRANSFER_CALLBACK) { 1559 return ns; 1560 } 1561 1562 struct timespec timeout; 1563 const struct timespec *requested = &ClientProxy::kForever; 1564 if (ns != NS_WHENEVER) { 1565 timeout.tv_sec = ns / 1000000000LL; 1566 timeout.tv_nsec = ns % 1000000000LL; 1567 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1568 requested = &timeout; 1569 } 1570 1571 while (mRemainingFrames > 0) { 1572 1573 Buffer audioBuffer; 1574 audioBuffer.frameCount = mRemainingFrames; 1575 size_t nonContig; 1576 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1577 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1578 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1579 requested = &ClientProxy::kNonBlocking; 1580 size_t avail = audioBuffer.frameCount + nonContig; 1581 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1582 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1583 if (err != NO_ERROR) { 1584 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1585 (isOffloaded() && (err == DEAD_OBJECT))) { 1586 return 0; 1587 } 1588 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1589 return NS_NEVER; 1590 } 1591 1592 if (mRetryOnPartialBuffer && !isOffloaded()) { 1593 mRetryOnPartialBuffer = false; 1594 if (avail < mRemainingFrames) { 1595 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1596 if (ns < 0 || myns < ns) { 1597 ns = myns; 1598 } 1599 return ns; 1600 } 1601 } 1602 1603 // Divide buffer size by 2 to take into account the expansion 1604 // due to 8 to 16 bit conversion: the callback must fill only half 1605 // of the destination buffer 1606 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1607 audioBuffer.size >>= 1; 1608 } 1609 1610 size_t reqSize = audioBuffer.size; 1611 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1612 size_t writtenSize = audioBuffer.size; 1613 size_t writtenFrames = writtenSize / mFrameSize; 1614 1615 // Sanity check on returned size 1616 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1617 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1618 reqSize, (int) writtenSize); 1619 return NS_NEVER; 1620 } 1621 1622 if (writtenSize == 0) { 1623 // The callback is done filling buffers 1624 // Keep this thread going to handle timed events and 1625 // still try to get more data in intervals of WAIT_PERIOD_MS 1626 // but don't just loop and block the CPU, so wait 1627 return WAIT_PERIOD_MS * 1000000LL; 1628 } 1629 1630 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1631 // 8 to 16 bit conversion, note that source and destination are the same address 1632 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1633 audioBuffer.size <<= 1; 1634 } 1635 1636 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1637 audioBuffer.frameCount = releasedFrames; 1638 mRemainingFrames -= releasedFrames; 1639 if (misalignment >= releasedFrames) { 1640 misalignment -= releasedFrames; 1641 } else { 1642 misalignment = 0; 1643 } 1644 1645 releaseBuffer(&audioBuffer); 1646 1647 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1648 // if callback doesn't like to accept the full chunk 1649 if (writtenSize < reqSize) { 1650 continue; 1651 } 1652 1653 // There could be enough non-contiguous frames available to satisfy the remaining request 1654 if (mRemainingFrames <= nonContig) { 1655 continue; 1656 } 1657 1658#if 0 1659 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1660 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1661 // that total to a sum == notificationFrames. 1662 if (0 < misalignment && misalignment <= mRemainingFrames) { 1663 mRemainingFrames = misalignment; 1664 return (mRemainingFrames * 1100000000LL) / sampleRate; 1665 } 1666#endif 1667 1668 } 1669 mRemainingFrames = notificationFrames; 1670 mRetryOnPartialBuffer = true; 1671 1672 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1673 return 0; 1674} 1675 1676status_t AudioTrack::restoreTrack_l(const char *from) 1677{ 1678 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1679 isOffloaded_l() ? "Offloaded" : "PCM", from); 1680 ++mSequence; 1681 status_t result; 1682 1683 // refresh the audio configuration cache in this process to make sure we get new 1684 // output parameters in getOutput_l() and createTrack_l() 1685 AudioSystem::clearAudioConfigCache(); 1686 1687 if (isOffloaded_l()) { 1688 // FIXME re-creation of offloaded tracks is not yet implemented 1689 return DEAD_OBJECT; 1690 } 1691 1692 // force new output query from audio policy manager; 1693 mOutput = 0; 1694 audio_io_handle_t output = getOutput_l(); 1695 1696 // if the new IAudioTrack is created, createTrack_l() will modify the 1697 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1698 // It will also delete the strong references on previous IAudioTrack and IMemory 1699 1700 // take the frames that will be lost by track recreation into account in saved position 1701 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1702 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1703 result = createTrack_l(mStreamType, 1704 mSampleRate, 1705 mFormat, 1706 mReqFrameCount, // so that frame count never goes down 1707 mFlags, 1708 mSharedBuffer, 1709 output, 1710 position /*epoch*/); 1711 1712 if (result == NO_ERROR) { 1713 // continue playback from last known position, but 1714 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1715 if (mStaticProxy != NULL) { 1716 mLoopPeriod = 0; 1717 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1718 } 1719 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1720 // track destruction have been played? This is critical for SoundPool implementation 1721 // This must be broken, and needs to be tested/debugged. 1722#if 0 1723 // restore write index and set other indexes to reflect empty buffer status 1724 if (!strcmp(from, "start")) { 1725 // Make sure that a client relying on callback events indicating underrun or 1726 // the actual amount of audio frames played (e.g SoundPool) receives them. 1727 if (mSharedBuffer == 0) { 1728 // restart playback even if buffer is not completely filled. 1729 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1730 } 1731 } 1732#endif 1733 if (mState == STATE_ACTIVE) { 1734 result = mAudioTrack->start(); 1735 } 1736 } 1737 if (result != NO_ERROR) { 1738 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1739 // As getOutput was called above and resulted in an output stream to be opened, 1740 // we need to release it. 1741 AudioSystem::releaseOutput(output); 1742 ALOGW("restoreTrack_l() failed status %d", result); 1743 mState = STATE_STOPPED; 1744 } 1745 1746 return result; 1747} 1748 1749status_t AudioTrack::setParameters(const String8& keyValuePairs) 1750{ 1751 AutoMutex lock(mLock); 1752 return mAudioTrack->setParameters(keyValuePairs); 1753} 1754 1755status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1756{ 1757 AutoMutex lock(mLock); 1758 // FIXME not implemented for fast tracks; should use proxy and SSQ 1759 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1760 return INVALID_OPERATION; 1761 } 1762 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1763 return INVALID_OPERATION; 1764 } 1765 status_t status = mAudioTrack->getTimestamp(timestamp); 1766 if (status == NO_ERROR) { 1767 timestamp.mPosition += mProxy->getEpoch(); 1768 } 1769 return status; 1770} 1771 1772String8 AudioTrack::getParameters(const String8& keys) 1773{ 1774 audio_io_handle_t output = getOutput(); 1775 if (output != 0) { 1776 return AudioSystem::getParameters(output, keys); 1777 } else { 1778 return String8::empty(); 1779 } 1780} 1781 1782bool AudioTrack::isOffloaded() const 1783{ 1784 AutoMutex lock(mLock); 1785 return isOffloaded_l(); 1786} 1787 1788status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1789{ 1790 1791 const size_t SIZE = 256; 1792 char buffer[SIZE]; 1793 String8 result; 1794 1795 result.append(" AudioTrack::dump\n"); 1796 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1797 mVolume[0], mVolume[1]); 1798 result.append(buffer); 1799 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1800 mChannelCount, mFrameCount); 1801 result.append(buffer); 1802 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1803 result.append(buffer); 1804 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1805 result.append(buffer); 1806 ::write(fd, result.string(), result.size()); 1807 return NO_ERROR; 1808} 1809 1810uint32_t AudioTrack::getUnderrunFrames() const 1811{ 1812 AutoMutex lock(mLock); 1813 return mProxy->getUnderrunFrames(); 1814} 1815 1816// ========================================================================= 1817 1818void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1819{ 1820 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1821 if (audioTrack != 0) { 1822 AutoMutex lock(audioTrack->mLock); 1823 audioTrack->mProxy->binderDied(); 1824 } 1825} 1826 1827// ========================================================================= 1828 1829AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1830 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1831 mIgnoreNextPausedInt(false) 1832{ 1833} 1834 1835AudioTrack::AudioTrackThread::~AudioTrackThread() 1836{ 1837} 1838 1839bool AudioTrack::AudioTrackThread::threadLoop() 1840{ 1841 { 1842 AutoMutex _l(mMyLock); 1843 if (mPaused) { 1844 mMyCond.wait(mMyLock); 1845 // caller will check for exitPending() 1846 return true; 1847 } 1848 if (mIgnoreNextPausedInt) { 1849 mIgnoreNextPausedInt = false; 1850 mPausedInt = false; 1851 } 1852 if (mPausedInt) { 1853 if (mPausedNs > 0) { 1854 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1855 } else { 1856 mMyCond.wait(mMyLock); 1857 } 1858 mPausedInt = false; 1859 return true; 1860 } 1861 } 1862 nsecs_t ns = mReceiver.processAudioBuffer(); 1863 switch (ns) { 1864 case 0: 1865 return true; 1866 case NS_INACTIVE: 1867 pauseInternal(); 1868 return true; 1869 case NS_NEVER: 1870 return false; 1871 case NS_WHENEVER: 1872 // FIXME increase poll interval, or make event-driven 1873 ns = 1000000000LL; 1874 // fall through 1875 default: 1876 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1877 pauseInternal(ns); 1878 return true; 1879 } 1880} 1881 1882void AudioTrack::AudioTrackThread::requestExit() 1883{ 1884 // must be in this order to avoid a race condition 1885 Thread::requestExit(); 1886 resume(); 1887} 1888 1889void AudioTrack::AudioTrackThread::pause() 1890{ 1891 AutoMutex _l(mMyLock); 1892 mPaused = true; 1893} 1894 1895void AudioTrack::AudioTrackThread::resume() 1896{ 1897 AutoMutex _l(mMyLock); 1898 mIgnoreNextPausedInt = true; 1899 if (mPaused || mPausedInt) { 1900 mPaused = false; 1901 mPausedInt = false; 1902 mMyCond.signal(); 1903 } 1904} 1905 1906void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1907{ 1908 AutoMutex _l(mMyLock); 1909 mPausedInt = true; 1910 mPausedNs = ns; 1911} 1912 1913}; // namespace android 1914