AudioTrack.cpp revision 74935e44734c1ec235c2b6677db3e0dbefa5ddb8
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31#define WAIT_STREAM_END_TIMEOUT_SEC 120 32 33 34namespace android { 35// --------------------------------------------------------------------------- 36 37// static 38status_t AudioTrack::getMinFrameCount( 39 size_t* frameCount, 40 audio_stream_type_t streamType, 41 uint32_t sampleRate) 42{ 43 if (frameCount == NULL) { 44 return BAD_VALUE; 45 } 46 47 // default to 0 in case of error 48 *frameCount = 0; 49 50 // FIXME merge with similar code in createTrack_l(), except we're missing 51 // some information here that is available in createTrack_l(): 52 // audio_io_handle_t output 53 // audio_format_t format 54 // audio_channel_mask_t channelMask 55 // audio_output_flags_t flags 56 uint32_t afSampleRate; 57 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 58 return NO_INIT; 59 } 60 size_t afFrameCount; 61 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 62 return NO_INIT; 63 } 64 uint32_t afLatency; 65 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 66 return NO_INIT; 67 } 68 69 // Ensure that buffer depth covers at least audio hardware latency 70 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 71 if (minBufCount < 2) { 72 minBufCount = 2; 73 } 74 75 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 76 afFrameCount * minBufCount * sampleRate / afSampleRate; 77 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 78 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 79 return NO_ERROR; 80} 81 82// --------------------------------------------------------------------------- 83 84AudioTrack::AudioTrack() 85 : mStatus(NO_INIT), 86 mIsTimed(false), 87 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 88 mPreviousSchedulingGroup(SP_DEFAULT) 89{ 90} 91 92AudioTrack::AudioTrack( 93 audio_stream_type_t streamType, 94 uint32_t sampleRate, 95 audio_format_t format, 96 audio_channel_mask_t channelMask, 97 int frameCount, 98 audio_output_flags_t flags, 99 callback_t cbf, 100 void* user, 101 int notificationFrames, 102 int sessionId, 103 transfer_type transferType, 104 const audio_offload_info_t *offloadInfo, 105 int uid) 106 : mStatus(NO_INIT), 107 mIsTimed(false), 108 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 109 mPreviousSchedulingGroup(SP_DEFAULT) 110{ 111 mStatus = set(streamType, sampleRate, format, channelMask, 112 frameCount, flags, cbf, user, notificationFrames, 113 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 114 offloadInfo, uid); 115} 116 117AudioTrack::AudioTrack( 118 audio_stream_type_t streamType, 119 uint32_t sampleRate, 120 audio_format_t format, 121 audio_channel_mask_t channelMask, 122 const sp<IMemory>& sharedBuffer, 123 audio_output_flags_t flags, 124 callback_t cbf, 125 void* user, 126 int notificationFrames, 127 int sessionId, 128 transfer_type transferType, 129 const audio_offload_info_t *offloadInfo, 130 int uid) 131 : mStatus(NO_INIT), 132 mIsTimed(false), 133 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 134 mPreviousSchedulingGroup(SP_DEFAULT) 135{ 136 mStatus = set(streamType, sampleRate, format, channelMask, 137 0 /*frameCount*/, flags, cbf, user, notificationFrames, 138 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); 139} 140 141AudioTrack::~AudioTrack() 142{ 143 if (mStatus == NO_ERROR) { 144 // Make sure that callback function exits in the case where 145 // it is looping on buffer full condition in obtainBuffer(). 146 // Otherwise the callback thread will never exit. 147 stop(); 148 if (mAudioTrackThread != 0) { 149 mProxy->interrupt(); 150 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 151 mAudioTrackThread->requestExitAndWait(); 152 mAudioTrackThread.clear(); 153 } 154 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 155 mAudioTrack.clear(); 156 IPCThreadState::self()->flushCommands(); 157 AudioSystem::releaseAudioSessionId(mSessionId); 158 } 159} 160 161status_t AudioTrack::set( 162 audio_stream_type_t streamType, 163 uint32_t sampleRate, 164 audio_format_t format, 165 audio_channel_mask_t channelMask, 166 int frameCountInt, 167 audio_output_flags_t flags, 168 callback_t cbf, 169 void* user, 170 int notificationFrames, 171 const sp<IMemory>& sharedBuffer, 172 bool threadCanCallJava, 173 int sessionId, 174 transfer_type transferType, 175 const audio_offload_info_t *offloadInfo, 176 int uid) 177{ 178 switch (transferType) { 179 case TRANSFER_DEFAULT: 180 if (sharedBuffer != 0) { 181 transferType = TRANSFER_SHARED; 182 } else if (cbf == NULL || threadCanCallJava) { 183 transferType = TRANSFER_SYNC; 184 } else { 185 transferType = TRANSFER_CALLBACK; 186 } 187 break; 188 case TRANSFER_CALLBACK: 189 if (cbf == NULL || sharedBuffer != 0) { 190 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 191 return BAD_VALUE; 192 } 193 break; 194 case TRANSFER_OBTAIN: 195 case TRANSFER_SYNC: 196 if (sharedBuffer != 0) { 197 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 198 return BAD_VALUE; 199 } 200 break; 201 case TRANSFER_SHARED: 202 if (sharedBuffer == 0) { 203 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 204 return BAD_VALUE; 205 } 206 break; 207 default: 208 ALOGE("Invalid transfer type %d", transferType); 209 return BAD_VALUE; 210 } 211 mTransfer = transferType; 212 213 // FIXME "int" here is legacy and will be replaced by size_t later 214 if (frameCountInt < 0) { 215 ALOGE("Invalid frame count %d", frameCountInt); 216 return BAD_VALUE; 217 } 218 size_t frameCount = frameCountInt; 219 220 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 221 sharedBuffer->size()); 222 223 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 224 225 AutoMutex lock(mLock); 226 227 // invariant that mAudioTrack != 0 is true only after set() returns successfully 228 if (mAudioTrack != 0) { 229 ALOGE("Track already in use"); 230 return INVALID_OPERATION; 231 } 232 233 mOutput = 0; 234 235 // handle default values first. 236 if (streamType == AUDIO_STREAM_DEFAULT) { 237 streamType = AUDIO_STREAM_MUSIC; 238 } 239 240 if (sampleRate == 0) { 241 uint32_t afSampleRate; 242 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 243 return NO_INIT; 244 } 245 sampleRate = afSampleRate; 246 } 247 mSampleRate = sampleRate; 248 249 // these below should probably come from the audioFlinger too... 250 if (format == AUDIO_FORMAT_DEFAULT) { 251 format = AUDIO_FORMAT_PCM_16_BIT; 252 } 253 254 // validate parameters 255 if (!audio_is_valid_format(format)) { 256 ALOGE("Invalid format %d", format); 257 return BAD_VALUE; 258 } 259 260 if (!audio_is_output_channel(channelMask)) { 261 ALOGE("Invalid channel mask %#x", channelMask); 262 return BAD_VALUE; 263 } 264 265 // AudioFlinger does not currently support 8-bit data in shared memory 266 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 267 ALOGE("8-bit data in shared memory is not supported"); 268 return BAD_VALUE; 269 } 270 271 // force direct flag if format is not linear PCM 272 // or offload was requested 273 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 274 || !audio_is_linear_pcm(format)) { 275 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 276 ? "Offload request, forcing to Direct Output" 277 : "Not linear PCM, forcing to Direct Output"); 278 flags = (audio_output_flags_t) 279 // FIXME why can't we allow direct AND fast? 280 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 281 } 282 // only allow deep buffering for music stream type 283 if (streamType != AUDIO_STREAM_MUSIC) { 284 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 285 } 286 287 mChannelMask = channelMask; 288 uint32_t channelCount = popcount(channelMask); 289 mChannelCount = channelCount; 290 291 if (audio_is_linear_pcm(format)) { 292 mFrameSize = channelCount * audio_bytes_per_sample(format); 293 mFrameSizeAF = channelCount * sizeof(int16_t); 294 } else { 295 mFrameSize = sizeof(uint8_t); 296 mFrameSizeAF = sizeof(uint8_t); 297 } 298 299 audio_io_handle_t output = AudioSystem::getOutput( 300 streamType, 301 sampleRate, format, channelMask, 302 flags, 303 offloadInfo); 304 305 if (output == 0) { 306 ALOGE("Could not get audio output for stream type %d", streamType); 307 return BAD_VALUE; 308 } 309 310 mVolume[LEFT] = 1.0f; 311 mVolume[RIGHT] = 1.0f; 312 mSendLevel = 0.0f; 313 mFrameCount = frameCount; 314 mReqFrameCount = frameCount; 315 mNotificationFramesReq = notificationFrames; 316 mNotificationFramesAct = 0; 317 mSessionId = sessionId; 318 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 319 mClientUid = IPCThreadState::self()->getCallingUid(); 320 } else { 321 mClientUid = uid; 322 } 323 mAuxEffectId = 0; 324 mFlags = flags; 325 mCbf = cbf; 326 327 if (cbf != NULL) { 328 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 329 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 330 } 331 332 // create the IAudioTrack 333 status_t status = createTrack_l(streamType, 334 sampleRate, 335 format, 336 frameCount, 337 flags, 338 sharedBuffer, 339 output, 340 0 /*epoch*/); 341 342 if (status != NO_ERROR) { 343 if (mAudioTrackThread != 0) { 344 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 345 mAudioTrackThread->requestExitAndWait(); 346 mAudioTrackThread.clear(); 347 } 348 //Use of direct and offloaded output streams is ref counted by audio policy manager. 349 // As getOutput was called above and resulted in an output stream to be opened, 350 // we need to release it. 351 AudioSystem::releaseOutput(output); 352 return status; 353 } 354 355 mStatus = NO_ERROR; 356 mStreamType = streamType; 357 mFormat = format; 358 mSharedBuffer = sharedBuffer; 359 mState = STATE_STOPPED; 360 mUserData = user; 361 mLoopPeriod = 0; 362 mMarkerPosition = 0; 363 mMarkerReached = false; 364 mNewPosition = 0; 365 mUpdatePeriod = 0; 366 AudioSystem::acquireAudioSessionId(mSessionId); 367 mSequence = 1; 368 mObservedSequence = mSequence; 369 mInUnderrun = false; 370 mOutput = output; 371 372 return NO_ERROR; 373} 374 375// ------------------------------------------------------------------------- 376 377status_t AudioTrack::start() 378{ 379 AutoMutex lock(mLock); 380 381 if (mState == STATE_ACTIVE) { 382 return INVALID_OPERATION; 383 } 384 385 mInUnderrun = true; 386 387 State previousState = mState; 388 if (previousState == STATE_PAUSED_STOPPING) { 389 mState = STATE_STOPPING; 390 } else { 391 mState = STATE_ACTIVE; 392 } 393 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 394 // reset current position as seen by client to 0 395 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 396 // force refresh of remaining frames by processAudioBuffer() as last 397 // write before stop could be partial. 398 mRefreshRemaining = true; 399 } 400 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 401 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 402 403 sp<AudioTrackThread> t = mAudioTrackThread; 404 if (t != 0) { 405 if (previousState == STATE_STOPPING) { 406 mProxy->interrupt(); 407 } else { 408 t->resume(); 409 } 410 } else { 411 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 412 get_sched_policy(0, &mPreviousSchedulingGroup); 413 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 414 } 415 416 status_t status = NO_ERROR; 417 if (!(flags & CBLK_INVALID)) { 418 status = mAudioTrack->start(); 419 if (status == DEAD_OBJECT) { 420 flags |= CBLK_INVALID; 421 } 422 } 423 if (flags & CBLK_INVALID) { 424 status = restoreTrack_l("start"); 425 } 426 427 if (status != NO_ERROR) { 428 ALOGE("start() status %d", status); 429 mState = previousState; 430 if (t != 0) { 431 if (previousState != STATE_STOPPING) { 432 t->pause(); 433 } 434 } else { 435 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 436 set_sched_policy(0, mPreviousSchedulingGroup); 437 } 438 } 439 440 return status; 441} 442 443void AudioTrack::stop() 444{ 445 AutoMutex lock(mLock); 446 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 447 return; 448 } 449 450 if (isOffloaded()) { 451 mState = STATE_STOPPING; 452 } else { 453 mState = STATE_STOPPED; 454 } 455 456 mProxy->interrupt(); 457 mAudioTrack->stop(); 458 // the playback head position will reset to 0, so if a marker is set, we need 459 // to activate it again 460 mMarkerReached = false; 461#if 0 462 // Force flush if a shared buffer is used otherwise audioflinger 463 // will not stop before end of buffer is reached. 464 // It may be needed to make sure that we stop playback, likely in case looping is on. 465 if (mSharedBuffer != 0) { 466 flush_l(); 467 } 468#endif 469 470 sp<AudioTrackThread> t = mAudioTrackThread; 471 if (t != 0) { 472 if (!isOffloaded()) { 473 t->pause(); 474 } 475 } else { 476 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 477 set_sched_policy(0, mPreviousSchedulingGroup); 478 } 479} 480 481bool AudioTrack::stopped() const 482{ 483 AutoMutex lock(mLock); 484 return mState != STATE_ACTIVE; 485} 486 487void AudioTrack::flush() 488{ 489 if (mSharedBuffer != 0) { 490 return; 491 } 492 AutoMutex lock(mLock); 493 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 494 return; 495 } 496 flush_l(); 497} 498 499void AudioTrack::flush_l() 500{ 501 ALOG_ASSERT(mState != STATE_ACTIVE); 502 503 // clear playback marker and periodic update counter 504 mMarkerPosition = 0; 505 mMarkerReached = false; 506 mUpdatePeriod = 0; 507 mRefreshRemaining = true; 508 509 mState = STATE_FLUSHED; 510 if (isOffloaded()) { 511 mProxy->interrupt(); 512 } 513 mProxy->flush(); 514 mAudioTrack->flush(); 515} 516 517void AudioTrack::pause() 518{ 519 AutoMutex lock(mLock); 520 if (mState == STATE_ACTIVE) { 521 mState = STATE_PAUSED; 522 } else if (mState == STATE_STOPPING) { 523 mState = STATE_PAUSED_STOPPING; 524 } else { 525 return; 526 } 527 mProxy->interrupt(); 528 mAudioTrack->pause(); 529} 530 531status_t AudioTrack::setVolume(float left, float right) 532{ 533 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 534 return BAD_VALUE; 535 } 536 537 AutoMutex lock(mLock); 538 mVolume[LEFT] = left; 539 mVolume[RIGHT] = right; 540 541 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 542 543 if (isOffloaded()) { 544 mAudioTrack->signal(); 545 } 546 return NO_ERROR; 547} 548 549status_t AudioTrack::setVolume(float volume) 550{ 551 return setVolume(volume, volume); 552} 553 554status_t AudioTrack::setAuxEffectSendLevel(float level) 555{ 556 if (level < 0.0f || level > 1.0f) { 557 return BAD_VALUE; 558 } 559 560 AutoMutex lock(mLock); 561 mSendLevel = level; 562 mProxy->setSendLevel(level); 563 564 return NO_ERROR; 565} 566 567void AudioTrack::getAuxEffectSendLevel(float* level) const 568{ 569 if (level != NULL) { 570 *level = mSendLevel; 571 } 572} 573 574status_t AudioTrack::setSampleRate(uint32_t rate) 575{ 576 if (mIsTimed || isOffloaded()) { 577 return INVALID_OPERATION; 578 } 579 580 uint32_t afSamplingRate; 581 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 582 return NO_INIT; 583 } 584 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 585 if (rate == 0 || rate > afSamplingRate*2 ) { 586 return BAD_VALUE; 587 } 588 589 AutoMutex lock(mLock); 590 mSampleRate = rate; 591 mProxy->setSampleRate(rate); 592 593 return NO_ERROR; 594} 595 596uint32_t AudioTrack::getSampleRate() const 597{ 598 if (mIsTimed) { 599 return 0; 600 } 601 602 AutoMutex lock(mLock); 603 604 // sample rate can be updated during playback by the offloaded decoder so we need to 605 // query the HAL and update if needed. 606// FIXME use Proxy return channel to update the rate from server and avoid polling here 607 if (isOffloaded()) { 608 if (mOutput != 0) { 609 uint32_t sampleRate = 0; 610 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 611 if (status == NO_ERROR) { 612 mSampleRate = sampleRate; 613 } 614 } 615 } 616 return mSampleRate; 617} 618 619status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 620{ 621 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 622 return INVALID_OPERATION; 623 } 624 625 if (loopCount == 0) { 626 ; 627 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 628 loopEnd - loopStart >= MIN_LOOP) { 629 ; 630 } else { 631 return BAD_VALUE; 632 } 633 634 AutoMutex lock(mLock); 635 // See setPosition() regarding setting parameters such as loop points or position while active 636 if (mState == STATE_ACTIVE) { 637 return INVALID_OPERATION; 638 } 639 setLoop_l(loopStart, loopEnd, loopCount); 640 return NO_ERROR; 641} 642 643void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 644{ 645 // FIXME If setting a loop also sets position to start of loop, then 646 // this is correct. Otherwise it should be removed. 647 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 648 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 649 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 650} 651 652status_t AudioTrack::setMarkerPosition(uint32_t marker) 653{ 654 // The only purpose of setting marker position is to get a callback 655 if (mCbf == NULL || isOffloaded()) { 656 return INVALID_OPERATION; 657 } 658 659 AutoMutex lock(mLock); 660 mMarkerPosition = marker; 661 mMarkerReached = false; 662 663 return NO_ERROR; 664} 665 666status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 667{ 668 if (isOffloaded()) { 669 return INVALID_OPERATION; 670 } 671 if (marker == NULL) { 672 return BAD_VALUE; 673 } 674 675 AutoMutex lock(mLock); 676 *marker = mMarkerPosition; 677 678 return NO_ERROR; 679} 680 681status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 682{ 683 // The only purpose of setting position update period is to get a callback 684 if (mCbf == NULL || isOffloaded()) { 685 return INVALID_OPERATION; 686 } 687 688 AutoMutex lock(mLock); 689 mNewPosition = mProxy->getPosition() + updatePeriod; 690 mUpdatePeriod = updatePeriod; 691 return NO_ERROR; 692} 693 694status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 695{ 696 if (isOffloaded()) { 697 return INVALID_OPERATION; 698 } 699 if (updatePeriod == NULL) { 700 return BAD_VALUE; 701 } 702 703 AutoMutex lock(mLock); 704 *updatePeriod = mUpdatePeriod; 705 706 return NO_ERROR; 707} 708 709status_t AudioTrack::setPosition(uint32_t position) 710{ 711 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 712 return INVALID_OPERATION; 713 } 714 if (position > mFrameCount) { 715 return BAD_VALUE; 716 } 717 718 AutoMutex lock(mLock); 719 // Currently we require that the player is inactive before setting parameters such as position 720 // or loop points. Otherwise, there could be a race condition: the application could read the 721 // current position, compute a new position or loop parameters, and then set that position or 722 // loop parameters but it would do the "wrong" thing since the position has continued to advance 723 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 724 // to specify how it wants to handle such scenarios. 725 if (mState == STATE_ACTIVE) { 726 return INVALID_OPERATION; 727 } 728 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 729 mLoopPeriod = 0; 730 // FIXME Check whether loops and setting position are incompatible in old code. 731 // If we use setLoop for both purposes we lose the capability to set the position while looping. 732 mStaticProxy->setLoop(position, mFrameCount, 0); 733 734 return NO_ERROR; 735} 736 737status_t AudioTrack::getPosition(uint32_t *position) const 738{ 739 if (position == NULL) { 740 return BAD_VALUE; 741 } 742 743 AutoMutex lock(mLock); 744 if (isOffloaded()) { 745 uint32_t dspFrames = 0; 746 747 if (mOutput != 0) { 748 uint32_t halFrames; 749 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 750 } 751 *position = dspFrames; 752 } else { 753 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 754 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 755 mProxy->getPosition(); 756 } 757 return NO_ERROR; 758} 759 760status_t AudioTrack::getBufferPosition(size_t *position) 761{ 762 if (mSharedBuffer == 0 || mIsTimed) { 763 return INVALID_OPERATION; 764 } 765 if (position == NULL) { 766 return BAD_VALUE; 767 } 768 769 AutoMutex lock(mLock); 770 *position = mStaticProxy->getBufferPosition(); 771 return NO_ERROR; 772} 773 774status_t AudioTrack::reload() 775{ 776 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 777 return INVALID_OPERATION; 778 } 779 780 AutoMutex lock(mLock); 781 // See setPosition() regarding setting parameters such as loop points or position while active 782 if (mState == STATE_ACTIVE) { 783 return INVALID_OPERATION; 784 } 785 mNewPosition = mUpdatePeriod; 786 mLoopPeriod = 0; 787 // FIXME The new code cannot reload while keeping a loop specified. 788 // Need to check how the old code handled this, and whether it's a significant change. 789 mStaticProxy->setLoop(0, mFrameCount, 0); 790 return NO_ERROR; 791} 792 793audio_io_handle_t AudioTrack::getOutput() 794{ 795 AutoMutex lock(mLock); 796 return mOutput; 797} 798 799// must be called with mLock held 800audio_io_handle_t AudioTrack::getOutput_l() 801{ 802 if (mOutput) { 803 return mOutput; 804 } else { 805 return AudioSystem::getOutput(mStreamType, 806 mSampleRate, mFormat, mChannelMask, mFlags); 807 } 808} 809 810status_t AudioTrack::attachAuxEffect(int effectId) 811{ 812 AutoMutex lock(mLock); 813 status_t status = mAudioTrack->attachAuxEffect(effectId); 814 if (status == NO_ERROR) { 815 mAuxEffectId = effectId; 816 } 817 return status; 818} 819 820// ------------------------------------------------------------------------- 821 822// must be called with mLock held 823status_t AudioTrack::createTrack_l( 824 audio_stream_type_t streamType, 825 uint32_t sampleRate, 826 audio_format_t format, 827 size_t frameCount, 828 audio_output_flags_t flags, 829 const sp<IMemory>& sharedBuffer, 830 audio_io_handle_t output, 831 size_t epoch) 832{ 833 status_t status; 834 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 835 if (audioFlinger == 0) { 836 ALOGE("Could not get audioflinger"); 837 return NO_INIT; 838 } 839 840 // Not all of these values are needed under all conditions, but it is easier to get them all 841 842 uint32_t afLatency; 843 status = AudioSystem::getLatency(output, streamType, &afLatency); 844 if (status != NO_ERROR) { 845 ALOGE("getLatency(%d) failed status %d", output, status); 846 return NO_INIT; 847 } 848 849 size_t afFrameCount; 850 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 851 if (status != NO_ERROR) { 852 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 853 return NO_INIT; 854 } 855 856 uint32_t afSampleRate; 857 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 858 if (status != NO_ERROR) { 859 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 860 return NO_INIT; 861 } 862 863 // Client decides whether the track is TIMED (see below), but can only express a preference 864 // for FAST. Server will perform additional tests. 865 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 866 // either of these use cases: 867 // use case 1: shared buffer 868 (sharedBuffer != 0) || 869 // use case 2: callback handler 870 (mCbf != NULL))) { 871 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 872 // once denied, do not request again if IAudioTrack is re-created 873 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 874 mFlags = flags; 875 } 876 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 877 878 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 879 // n = 1 fast track with single buffering; nBuffering is ignored 880 // n = 2 fast track with double buffering 881 // n = 2 normal track, no sample rate conversion 882 // n = 3 normal track, with sample rate conversion 883 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 884 // n > 3 very high latency or very small notification interval; nBuffering is ignored 885 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 886 887 mNotificationFramesAct = mNotificationFramesReq; 888 889 if (!audio_is_linear_pcm(format)) { 890 891 if (sharedBuffer != 0) { 892 // Same comment as below about ignoring frameCount parameter for set() 893 frameCount = sharedBuffer->size(); 894 } else if (frameCount == 0) { 895 frameCount = afFrameCount; 896 } 897 if (mNotificationFramesAct != frameCount) { 898 mNotificationFramesAct = frameCount; 899 } 900 } else if (sharedBuffer != 0) { 901 902 // Ensure that buffer alignment matches channel count 903 // 8-bit data in shared memory is not currently supported by AudioFlinger 904 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 905 if (mChannelCount > 1) { 906 // More than 2 channels does not require stronger alignment than stereo 907 alignment <<= 1; 908 } 909 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 910 ALOGE("Invalid buffer alignment: address %p, channel count %u", 911 sharedBuffer->pointer(), mChannelCount); 912 return BAD_VALUE; 913 } 914 915 // When initializing a shared buffer AudioTrack via constructors, 916 // there's no frameCount parameter. 917 // But when initializing a shared buffer AudioTrack via set(), 918 // there _is_ a frameCount parameter. We silently ignore it. 919 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 920 921 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 922 923 // FIXME move these calculations and associated checks to server 924 925 // Ensure that buffer depth covers at least audio hardware latency 926 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 927 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 928 afFrameCount, minBufCount, afSampleRate, afLatency); 929 if (minBufCount <= nBuffering) { 930 minBufCount = nBuffering; 931 } 932 933 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 934 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 935 ", afLatency=%d", 936 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 937 938 if (frameCount == 0) { 939 frameCount = minFrameCount; 940 } else if (frameCount < minFrameCount) { 941 // not ALOGW because it happens all the time when playing key clicks over A2DP 942 ALOGV("Minimum buffer size corrected from %d to %d", 943 frameCount, minFrameCount); 944 frameCount = minFrameCount; 945 } 946 // Make sure that application is notified with sufficient margin before underrun 947 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 948 mNotificationFramesAct = frameCount/nBuffering; 949 } 950 951 } else { 952 // For fast tracks, the frame count calculations and checks are done by server 953 } 954 955 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 956 if (mIsTimed) { 957 trackFlags |= IAudioFlinger::TRACK_TIMED; 958 } 959 960 pid_t tid = -1; 961 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 962 trackFlags |= IAudioFlinger::TRACK_FAST; 963 if (mAudioTrackThread != 0) { 964 tid = mAudioTrackThread->getTid(); 965 } 966 } 967 968 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 969 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 970 } 971 972 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 973 // but we will still need the original value also 974 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 975 sampleRate, 976 // AudioFlinger only sees 16-bit PCM 977 format == AUDIO_FORMAT_PCM_8_BIT ? 978 AUDIO_FORMAT_PCM_16_BIT : format, 979 mChannelMask, 980 &temp, 981 &trackFlags, 982 sharedBuffer, 983 output, 984 tid, 985 &mSessionId, 986 mName, 987 mClientUid, 988 &status); 989 990 if (track == 0) { 991 ALOGE("AudioFlinger could not create track, status: %d", status); 992 return status; 993 } 994 sp<IMemory> iMem = track->getCblk(); 995 if (iMem == 0) { 996 ALOGE("Could not get control block"); 997 return NO_INIT; 998 } 999 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1000 if (mAudioTrack != 0) { 1001 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1002 mDeathNotifier.clear(); 1003 } 1004 mAudioTrack = track; 1005 mCblkMemory = iMem; 1006 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1007 mCblk = cblk; 1008 // note that temp is the (possibly revised) value of frameCount 1009 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1010 // In current design, AudioTrack client checks and ensures frame count validity before 1011 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1012 // for fast track as it uses a special method of assigning frame count. 1013 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1014 } 1015 frameCount = temp; 1016 mAwaitBoost = false; 1017 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1018 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1019 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1020 mAwaitBoost = true; 1021 if (sharedBuffer == 0) { 1022 // Theoretically double-buffering is not required for fast tracks, 1023 // due to tighter scheduling. But in practice, to accommodate kernels with 1024 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1025 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1026 mNotificationFramesAct = frameCount/nBuffering; 1027 } 1028 } 1029 } else { 1030 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1031 // once denied, do not request again if IAudioTrack is re-created 1032 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1033 mFlags = flags; 1034 if (sharedBuffer == 0) { 1035 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1036 mNotificationFramesAct = frameCount/nBuffering; 1037 } 1038 } 1039 } 1040 } 1041 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1042 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1043 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1044 } else { 1045 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1046 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1047 mFlags = flags; 1048 return NO_INIT; 1049 } 1050 } 1051 1052 mRefreshRemaining = true; 1053 1054 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1055 // is the value of pointer() for the shared buffer, otherwise buffers points 1056 // immediately after the control block. This address is for the mapping within client 1057 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1058 void* buffers; 1059 if (sharedBuffer == 0) { 1060 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1061 } else { 1062 buffers = sharedBuffer->pointer(); 1063 } 1064 1065 mAudioTrack->attachAuxEffect(mAuxEffectId); 1066 // FIXME don't believe this lie 1067 mLatency = afLatency + (1000*frameCount) / sampleRate; 1068 mFrameCount = frameCount; 1069 // If IAudioTrack is re-created, don't let the requested frameCount 1070 // decrease. This can confuse clients that cache frameCount(). 1071 if (frameCount > mReqFrameCount) { 1072 mReqFrameCount = frameCount; 1073 } 1074 1075 // update proxy 1076 if (sharedBuffer == 0) { 1077 mStaticProxy.clear(); 1078 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1079 } else { 1080 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1081 mProxy = mStaticProxy; 1082 } 1083 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1084 uint16_t(mVolume[LEFT] * 0x1000)); 1085 mProxy->setSendLevel(mSendLevel); 1086 mProxy->setSampleRate(mSampleRate); 1087 mProxy->setEpoch(epoch); 1088 mProxy->setMinimum(mNotificationFramesAct); 1089 1090 mDeathNotifier = new DeathNotifier(this); 1091 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1092 1093 return NO_ERROR; 1094} 1095 1096status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1097{ 1098 if (audioBuffer == NULL) { 1099 return BAD_VALUE; 1100 } 1101 if (mTransfer != TRANSFER_OBTAIN) { 1102 audioBuffer->frameCount = 0; 1103 audioBuffer->size = 0; 1104 audioBuffer->raw = NULL; 1105 return INVALID_OPERATION; 1106 } 1107 1108 const struct timespec *requested; 1109 if (waitCount == -1) { 1110 requested = &ClientProxy::kForever; 1111 } else if (waitCount == 0) { 1112 requested = &ClientProxy::kNonBlocking; 1113 } else if (waitCount > 0) { 1114 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1115 struct timespec timeout; 1116 timeout.tv_sec = ms / 1000; 1117 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1118 requested = &timeout; 1119 } else { 1120 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1121 requested = NULL; 1122 } 1123 return obtainBuffer(audioBuffer, requested); 1124} 1125 1126status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1127 struct timespec *elapsed, size_t *nonContig) 1128{ 1129 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1130 uint32_t oldSequence = 0; 1131 uint32_t newSequence; 1132 1133 Proxy::Buffer buffer; 1134 status_t status = NO_ERROR; 1135 1136 static const int32_t kMaxTries = 5; 1137 int32_t tryCounter = kMaxTries; 1138 1139 do { 1140 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1141 // keep them from going away if another thread re-creates the track during obtainBuffer() 1142 sp<AudioTrackClientProxy> proxy; 1143 sp<IMemory> iMem; 1144 1145 { // start of lock scope 1146 AutoMutex lock(mLock); 1147 1148 newSequence = mSequence; 1149 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1150 if (status == DEAD_OBJECT) { 1151 // re-create track, unless someone else has already done so 1152 if (newSequence == oldSequence) { 1153 status = restoreTrack_l("obtainBuffer"); 1154 if (status != NO_ERROR) { 1155 buffer.mFrameCount = 0; 1156 buffer.mRaw = NULL; 1157 buffer.mNonContig = 0; 1158 break; 1159 } 1160 } 1161 } 1162 oldSequence = newSequence; 1163 1164 // Keep the extra references 1165 proxy = mProxy; 1166 iMem = mCblkMemory; 1167 1168 if (mState == STATE_STOPPING) { 1169 status = -EINTR; 1170 buffer.mFrameCount = 0; 1171 buffer.mRaw = NULL; 1172 buffer.mNonContig = 0; 1173 break; 1174 } 1175 1176 // Non-blocking if track is stopped or paused 1177 if (mState != STATE_ACTIVE) { 1178 requested = &ClientProxy::kNonBlocking; 1179 } 1180 1181 } // end of lock scope 1182 1183 buffer.mFrameCount = audioBuffer->frameCount; 1184 // FIXME starts the requested timeout and elapsed over from scratch 1185 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1186 1187 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1188 1189 audioBuffer->frameCount = buffer.mFrameCount; 1190 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1191 audioBuffer->raw = buffer.mRaw; 1192 if (nonContig != NULL) { 1193 *nonContig = buffer.mNonContig; 1194 } 1195 return status; 1196} 1197 1198void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1199{ 1200 if (mTransfer == TRANSFER_SHARED) { 1201 return; 1202 } 1203 1204 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1205 if (stepCount == 0) { 1206 return; 1207 } 1208 1209 Proxy::Buffer buffer; 1210 buffer.mFrameCount = stepCount; 1211 buffer.mRaw = audioBuffer->raw; 1212 1213 AutoMutex lock(mLock); 1214 mInUnderrun = false; 1215 mProxy->releaseBuffer(&buffer); 1216 1217 // restart track if it was disabled by audioflinger due to previous underrun 1218 if (mState == STATE_ACTIVE) { 1219 audio_track_cblk_t* cblk = mCblk; 1220 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1221 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1222 this, mName.string()); 1223 // FIXME ignoring status 1224 mAudioTrack->start(); 1225 } 1226 } 1227} 1228 1229// ------------------------------------------------------------------------- 1230 1231ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1232{ 1233 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1234 return INVALID_OPERATION; 1235 } 1236 1237 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1238 // Sanity-check: user is most-likely passing an error code, and it would 1239 // make the return value ambiguous (actualSize vs error). 1240 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1241 return BAD_VALUE; 1242 } 1243 1244 size_t written = 0; 1245 Buffer audioBuffer; 1246 1247 while (userSize >= mFrameSize) { 1248 audioBuffer.frameCount = userSize / mFrameSize; 1249 1250 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1251 if (err < 0) { 1252 if (written > 0) { 1253 break; 1254 } 1255 return ssize_t(err); 1256 } 1257 1258 size_t toWrite; 1259 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1260 // Divide capacity by 2 to take expansion into account 1261 toWrite = audioBuffer.size >> 1; 1262 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1263 } else { 1264 toWrite = audioBuffer.size; 1265 memcpy(audioBuffer.i8, buffer, toWrite); 1266 } 1267 buffer = ((const char *) buffer) + toWrite; 1268 userSize -= toWrite; 1269 written += toWrite; 1270 1271 releaseBuffer(&audioBuffer); 1272 } 1273 1274 return written; 1275} 1276 1277// ------------------------------------------------------------------------- 1278 1279TimedAudioTrack::TimedAudioTrack() { 1280 mIsTimed = true; 1281} 1282 1283status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1284{ 1285 AutoMutex lock(mLock); 1286 status_t result = UNKNOWN_ERROR; 1287 1288#if 1 1289 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1290 // while we are accessing the cblk 1291 sp<IAudioTrack> audioTrack = mAudioTrack; 1292 sp<IMemory> iMem = mCblkMemory; 1293#endif 1294 1295 // If the track is not invalid already, try to allocate a buffer. alloc 1296 // fails indicating that the server is dead, flag the track as invalid so 1297 // we can attempt to restore in just a bit. 1298 audio_track_cblk_t* cblk = mCblk; 1299 if (!(cblk->mFlags & CBLK_INVALID)) { 1300 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1301 if (result == DEAD_OBJECT) { 1302 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1303 } 1304 } 1305 1306 // If the track is invalid at this point, attempt to restore it. and try the 1307 // allocation one more time. 1308 if (cblk->mFlags & CBLK_INVALID) { 1309 result = restoreTrack_l("allocateTimedBuffer"); 1310 1311 if (result == NO_ERROR) { 1312 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1313 } 1314 } 1315 1316 return result; 1317} 1318 1319status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1320 int64_t pts) 1321{ 1322 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1323 { 1324 AutoMutex lock(mLock); 1325 audio_track_cblk_t* cblk = mCblk; 1326 // restart track if it was disabled by audioflinger due to previous underrun 1327 if (buffer->size() != 0 && status == NO_ERROR && 1328 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1329 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1330 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1331 // FIXME ignoring status 1332 mAudioTrack->start(); 1333 } 1334 } 1335 return status; 1336} 1337 1338status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1339 TargetTimeline target) 1340{ 1341 return mAudioTrack->setMediaTimeTransform(xform, target); 1342} 1343 1344// ------------------------------------------------------------------------- 1345 1346nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1347{ 1348 // Currently the AudioTrack thread is not created if there are no callbacks. 1349 // Would it ever make sense to run the thread, even without callbacks? 1350 // If so, then replace this by checks at each use for mCbf != NULL. 1351 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1352 1353 mLock.lock(); 1354 if (mAwaitBoost) { 1355 mAwaitBoost = false; 1356 mLock.unlock(); 1357 static const int32_t kMaxTries = 5; 1358 int32_t tryCounter = kMaxTries; 1359 uint32_t pollUs = 10000; 1360 do { 1361 int policy = sched_getscheduler(0); 1362 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1363 break; 1364 } 1365 usleep(pollUs); 1366 pollUs <<= 1; 1367 } while (tryCounter-- > 0); 1368 if (tryCounter < 0) { 1369 ALOGE("did not receive expected priority boost on time"); 1370 } 1371 // Run again immediately 1372 return 0; 1373 } 1374 1375 // Can only reference mCblk while locked 1376 int32_t flags = android_atomic_and( 1377 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1378 1379 // Check for track invalidation 1380 if (flags & CBLK_INVALID) { 1381 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1382 // AudioSystem cache. We should not exit here but after calling the callback so 1383 // that the upper layers can recreate the track 1384 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1385 status_t status = restoreTrack_l("processAudioBuffer"); 1386 mLock.unlock(); 1387 // Run again immediately, but with a new IAudioTrack 1388 return 0; 1389 } 1390 } 1391 1392 bool waitStreamEnd = mState == STATE_STOPPING; 1393 bool active = mState == STATE_ACTIVE; 1394 1395 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1396 bool newUnderrun = false; 1397 if (flags & CBLK_UNDERRUN) { 1398#if 0 1399 // Currently in shared buffer mode, when the server reaches the end of buffer, 1400 // the track stays active in continuous underrun state. It's up to the application 1401 // to pause or stop the track, or set the position to a new offset within buffer. 1402 // This was some experimental code to auto-pause on underrun. Keeping it here 1403 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1404 if (mTransfer == TRANSFER_SHARED) { 1405 mState = STATE_PAUSED; 1406 active = false; 1407 } 1408#endif 1409 if (!mInUnderrun) { 1410 mInUnderrun = true; 1411 newUnderrun = true; 1412 } 1413 } 1414 1415 // Get current position of server 1416 size_t position = mProxy->getPosition(); 1417 1418 // Manage marker callback 1419 bool markerReached = false; 1420 size_t markerPosition = mMarkerPosition; 1421 // FIXME fails for wraparound, need 64 bits 1422 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1423 mMarkerReached = markerReached = true; 1424 } 1425 1426 // Determine number of new position callback(s) that will be needed, while locked 1427 size_t newPosCount = 0; 1428 size_t newPosition = mNewPosition; 1429 size_t updatePeriod = mUpdatePeriod; 1430 // FIXME fails for wraparound, need 64 bits 1431 if (updatePeriod > 0 && position >= newPosition) { 1432 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1433 mNewPosition += updatePeriod * newPosCount; 1434 } 1435 1436 // Cache other fields that will be needed soon 1437 uint32_t loopPeriod = mLoopPeriod; 1438 uint32_t sampleRate = mSampleRate; 1439 size_t notificationFrames = mNotificationFramesAct; 1440 if (mRefreshRemaining) { 1441 mRefreshRemaining = false; 1442 mRemainingFrames = notificationFrames; 1443 mRetryOnPartialBuffer = false; 1444 } 1445 size_t misalignment = mProxy->getMisalignment(); 1446 uint32_t sequence = mSequence; 1447 1448 // These fields don't need to be cached, because they are assigned only by set(): 1449 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1450 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1451 1452 mLock.unlock(); 1453 1454 if (waitStreamEnd) { 1455 AutoMutex lock(mLock); 1456 1457 sp<AudioTrackClientProxy> proxy = mProxy; 1458 sp<IMemory> iMem = mCblkMemory; 1459 1460 struct timespec timeout; 1461 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1462 timeout.tv_nsec = 0; 1463 1464 mLock.unlock(); 1465 status_t status = mProxy->waitStreamEndDone(&timeout); 1466 mLock.lock(); 1467 switch (status) { 1468 case NO_ERROR: 1469 case DEAD_OBJECT: 1470 case TIMED_OUT: 1471 mLock.unlock(); 1472 mCbf(EVENT_STREAM_END, mUserData, NULL); 1473 mLock.lock(); 1474 if (mState == STATE_STOPPING) { 1475 mState = STATE_STOPPED; 1476 if (status != DEAD_OBJECT) { 1477 return NS_INACTIVE; 1478 } 1479 } 1480 return 0; 1481 default: 1482 return 0; 1483 } 1484 } 1485 1486 // perform callbacks while unlocked 1487 if (newUnderrun) { 1488 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1489 } 1490 // FIXME we will miss loops if loop cycle was signaled several times since last call 1491 // to processAudioBuffer() 1492 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1493 mCbf(EVENT_LOOP_END, mUserData, NULL); 1494 } 1495 if (flags & CBLK_BUFFER_END) { 1496 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1497 } 1498 if (markerReached) { 1499 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1500 } 1501 while (newPosCount > 0) { 1502 size_t temp = newPosition; 1503 mCbf(EVENT_NEW_POS, mUserData, &temp); 1504 newPosition += updatePeriod; 1505 newPosCount--; 1506 } 1507 1508 if (mObservedSequence != sequence) { 1509 mObservedSequence = sequence; 1510 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1511 // for offloaded tracks, just wait for the upper layers to recreate the track 1512 if (isOffloaded()) { 1513 return NS_INACTIVE; 1514 } 1515 } 1516 1517 // if inactive, then don't run me again until re-started 1518 if (!active) { 1519 return NS_INACTIVE; 1520 } 1521 1522 // Compute the estimated time until the next timed event (position, markers, loops) 1523 // FIXME only for non-compressed audio 1524 uint32_t minFrames = ~0; 1525 if (!markerReached && position < markerPosition) { 1526 minFrames = markerPosition - position; 1527 } 1528 if (loopPeriod > 0 && loopPeriod < minFrames) { 1529 minFrames = loopPeriod; 1530 } 1531 if (updatePeriod > 0 && updatePeriod < minFrames) { 1532 minFrames = updatePeriod; 1533 } 1534 1535 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1536 static const uint32_t kPoll = 0; 1537 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1538 minFrames = kPoll * notificationFrames; 1539 } 1540 1541 // Convert frame units to time units 1542 nsecs_t ns = NS_WHENEVER; 1543 if (minFrames != (uint32_t) ~0) { 1544 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1545 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1546 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1547 } 1548 1549 // If not supplying data by EVENT_MORE_DATA, then we're done 1550 if (mTransfer != TRANSFER_CALLBACK) { 1551 return ns; 1552 } 1553 1554 struct timespec timeout; 1555 const struct timespec *requested = &ClientProxy::kForever; 1556 if (ns != NS_WHENEVER) { 1557 timeout.tv_sec = ns / 1000000000LL; 1558 timeout.tv_nsec = ns % 1000000000LL; 1559 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1560 requested = &timeout; 1561 } 1562 1563 while (mRemainingFrames > 0) { 1564 1565 Buffer audioBuffer; 1566 audioBuffer.frameCount = mRemainingFrames; 1567 size_t nonContig; 1568 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1569 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1570 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1571 requested = &ClientProxy::kNonBlocking; 1572 size_t avail = audioBuffer.frameCount + nonContig; 1573 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1574 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1575 if (err != NO_ERROR) { 1576 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1577 (isOffloaded() && (err == DEAD_OBJECT))) { 1578 return 0; 1579 } 1580 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1581 return NS_NEVER; 1582 } 1583 1584 if (mRetryOnPartialBuffer && !isOffloaded()) { 1585 mRetryOnPartialBuffer = false; 1586 if (avail < mRemainingFrames) { 1587 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1588 if (ns < 0 || myns < ns) { 1589 ns = myns; 1590 } 1591 return ns; 1592 } 1593 } 1594 1595 // Divide buffer size by 2 to take into account the expansion 1596 // due to 8 to 16 bit conversion: the callback must fill only half 1597 // of the destination buffer 1598 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1599 audioBuffer.size >>= 1; 1600 } 1601 1602 size_t reqSize = audioBuffer.size; 1603 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1604 size_t writtenSize = audioBuffer.size; 1605 size_t writtenFrames = writtenSize / mFrameSize; 1606 1607 // Sanity check on returned size 1608 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1609 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1610 reqSize, (int) writtenSize); 1611 return NS_NEVER; 1612 } 1613 1614 if (writtenSize == 0) { 1615 // The callback is done filling buffers 1616 // Keep this thread going to handle timed events and 1617 // still try to get more data in intervals of WAIT_PERIOD_MS 1618 // but don't just loop and block the CPU, so wait 1619 return WAIT_PERIOD_MS * 1000000LL; 1620 } 1621 1622 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1623 // 8 to 16 bit conversion, note that source and destination are the same address 1624 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1625 audioBuffer.size <<= 1; 1626 } 1627 1628 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1629 audioBuffer.frameCount = releasedFrames; 1630 mRemainingFrames -= releasedFrames; 1631 if (misalignment >= releasedFrames) { 1632 misalignment -= releasedFrames; 1633 } else { 1634 misalignment = 0; 1635 } 1636 1637 releaseBuffer(&audioBuffer); 1638 1639 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1640 // if callback doesn't like to accept the full chunk 1641 if (writtenSize < reqSize) { 1642 continue; 1643 } 1644 1645 // There could be enough non-contiguous frames available to satisfy the remaining request 1646 if (mRemainingFrames <= nonContig) { 1647 continue; 1648 } 1649 1650#if 0 1651 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1652 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1653 // that total to a sum == notificationFrames. 1654 if (0 < misalignment && misalignment <= mRemainingFrames) { 1655 mRemainingFrames = misalignment; 1656 return (mRemainingFrames * 1100000000LL) / sampleRate; 1657 } 1658#endif 1659 1660 } 1661 mRemainingFrames = notificationFrames; 1662 mRetryOnPartialBuffer = true; 1663 1664 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1665 return 0; 1666} 1667 1668status_t AudioTrack::restoreTrack_l(const char *from) 1669{ 1670 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1671 isOffloaded() ? "Offloaded" : "PCM", from); 1672 ++mSequence; 1673 status_t result; 1674 1675 // refresh the audio configuration cache in this process to make sure we get new 1676 // output parameters in getOutput_l() and createTrack_l() 1677 AudioSystem::clearAudioConfigCache(); 1678 1679 if (isOffloaded()) { 1680 return DEAD_OBJECT; 1681 } 1682 1683 // force new output query from audio policy manager; 1684 mOutput = 0; 1685 audio_io_handle_t output = getOutput_l(); 1686 1687 // if the new IAudioTrack is created, createTrack_l() will modify the 1688 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1689 // It will also delete the strong references on previous IAudioTrack and IMemory 1690 1691 // take the frames that will be lost by track recreation into account in saved position 1692 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1693 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1694 result = createTrack_l(mStreamType, 1695 mSampleRate, 1696 mFormat, 1697 mReqFrameCount, // so that frame count never goes down 1698 mFlags, 1699 mSharedBuffer, 1700 output, 1701 position /*epoch*/); 1702 1703 if (result == NO_ERROR) { 1704 // continue playback from last known position, but 1705 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1706 if (mStaticProxy != NULL) { 1707 mLoopPeriod = 0; 1708 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1709 } 1710 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1711 // track destruction have been played? This is critical for SoundPool implementation 1712 // This must be broken, and needs to be tested/debugged. 1713#if 0 1714 // restore write index and set other indexes to reflect empty buffer status 1715 if (!strcmp(from, "start")) { 1716 // Make sure that a client relying on callback events indicating underrun or 1717 // the actual amount of audio frames played (e.g SoundPool) receives them. 1718 if (mSharedBuffer == 0) { 1719 // restart playback even if buffer is not completely filled. 1720 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1721 } 1722 } 1723#endif 1724 if (mState == STATE_ACTIVE) { 1725 result = mAudioTrack->start(); 1726 } 1727 } 1728 if (result != NO_ERROR) { 1729 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1730 // As getOutput was called above and resulted in an output stream to be opened, 1731 // we need to release it. 1732 AudioSystem::releaseOutput(output); 1733 ALOGW("restoreTrack_l() failed status %d", result); 1734 mState = STATE_STOPPED; 1735 } 1736 1737 return result; 1738} 1739 1740status_t AudioTrack::setParameters(const String8& keyValuePairs) 1741{ 1742 AutoMutex lock(mLock); 1743 return mAudioTrack->setParameters(keyValuePairs); 1744} 1745 1746status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1747{ 1748 AutoMutex lock(mLock); 1749 // FIXME not implemented for fast tracks; should use proxy and SSQ 1750 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1751 return INVALID_OPERATION; 1752 } 1753 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1754 return INVALID_OPERATION; 1755 } 1756 status_t status = mAudioTrack->getTimestamp(timestamp); 1757 if (status == NO_ERROR) { 1758 timestamp.mPosition += mProxy->getEpoch(); 1759 } 1760 return status; 1761} 1762 1763String8 AudioTrack::getParameters(const String8& keys) 1764{ 1765 if (mOutput) { 1766 return AudioSystem::getParameters(mOutput, keys); 1767 } else { 1768 return String8::empty(); 1769 } 1770} 1771 1772status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1773{ 1774 1775 const size_t SIZE = 256; 1776 char buffer[SIZE]; 1777 String8 result; 1778 1779 result.append(" AudioTrack::dump\n"); 1780 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1781 mVolume[0], mVolume[1]); 1782 result.append(buffer); 1783 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1784 mChannelCount, mFrameCount); 1785 result.append(buffer); 1786 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1787 result.append(buffer); 1788 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1789 result.append(buffer); 1790 ::write(fd, result.string(), result.size()); 1791 return NO_ERROR; 1792} 1793 1794uint32_t AudioTrack::getUnderrunFrames() const 1795{ 1796 AutoMutex lock(mLock); 1797 return mProxy->getUnderrunFrames(); 1798} 1799 1800// ========================================================================= 1801 1802void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1803{ 1804 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1805 if (audioTrack != 0) { 1806 AutoMutex lock(audioTrack->mLock); 1807 audioTrack->mProxy->binderDied(); 1808 } 1809} 1810 1811// ========================================================================= 1812 1813AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1814 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1815 mIgnoreNextPausedInt(false) 1816{ 1817} 1818 1819AudioTrack::AudioTrackThread::~AudioTrackThread() 1820{ 1821} 1822 1823bool AudioTrack::AudioTrackThread::threadLoop() 1824{ 1825 { 1826 AutoMutex _l(mMyLock); 1827 if (mPaused) { 1828 mMyCond.wait(mMyLock); 1829 // caller will check for exitPending() 1830 return true; 1831 } 1832 if (mIgnoreNextPausedInt) { 1833 mIgnoreNextPausedInt = false; 1834 mPausedInt = false; 1835 } 1836 if (mPausedInt) { 1837 if (mPausedNs > 0) { 1838 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1839 } else { 1840 mMyCond.wait(mMyLock); 1841 } 1842 mPausedInt = false; 1843 return true; 1844 } 1845 } 1846 nsecs_t ns = mReceiver.processAudioBuffer(this); 1847 switch (ns) { 1848 case 0: 1849 return true; 1850 case NS_INACTIVE: 1851 pauseInternal(); 1852 return true; 1853 case NS_NEVER: 1854 return false; 1855 case NS_WHENEVER: 1856 // FIXME increase poll interval, or make event-driven 1857 ns = 1000000000LL; 1858 // fall through 1859 default: 1860 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1861 pauseInternal(ns); 1862 return true; 1863 } 1864} 1865 1866void AudioTrack::AudioTrackThread::requestExit() 1867{ 1868 // must be in this order to avoid a race condition 1869 Thread::requestExit(); 1870 resume(); 1871} 1872 1873void AudioTrack::AudioTrackThread::pause() 1874{ 1875 AutoMutex _l(mMyLock); 1876 mPaused = true; 1877} 1878 1879void AudioTrack::AudioTrackThread::resume() 1880{ 1881 AutoMutex _l(mMyLock); 1882 mIgnoreNextPausedInt = true; 1883 if (mPaused || mPausedInt) { 1884 mPaused = false; 1885 mPausedInt = false; 1886 mMyCond.signal(); 1887 } 1888} 1889 1890void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1891{ 1892 AutoMutex _l(mMyLock); 1893 mPausedInt = true; 1894 mPausedNs = ns; 1895} 1896 1897}; // namespace android 1898