AudioTrack.cpp revision 720ad9ddb2ac6b55b0dfbfcd2d8360151d8ac427
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 ALOGE("Unable to query output sample rate for stream type %d; status %d", 58 streamType, status); 59 return status; 60 } 61 size_t afFrameCount; 62 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 63 if (status != NO_ERROR) { 64 ALOGE("Unable to query output frame count for stream type %d; status %d", 65 streamType, status); 66 return status; 67 } 68 uint32_t afLatency; 69 status = AudioSystem::getOutputLatency(&afLatency, streamType); 70 if (status != NO_ERROR) { 71 ALOGE("Unable to query output latency for stream type %d; status %d", 72 streamType, status); 73 return status; 74 } 75 76 // Ensure that buffer depth covers at least audio hardware latency 77 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 78 if (minBufCount < 2) { 79 minBufCount = 2; 80 } 81 82 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 83 afFrameCount * minBufCount * sampleRate / afSampleRate; 84 // The formula above should always produce a non-zero value, but return an error 85 // in the unlikely event that it does not, as that's part of the API contract. 86 if (*frameCount == 0) { 87 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 88 streamType, sampleRate); 89 return BAD_VALUE; 90 } 91 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 92 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 93 return NO_ERROR; 94} 95 96// --------------------------------------------------------------------------- 97 98AudioTrack::AudioTrack() 99 : mStatus(NO_INIT), 100 mIsTimed(false), 101 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 102 mPreviousSchedulingGroup(SP_DEFAULT) 103{ 104} 105 106AudioTrack::AudioTrack( 107 audio_stream_type_t streamType, 108 uint32_t sampleRate, 109 audio_format_t format, 110 audio_channel_mask_t channelMask, 111 int frameCount, 112 audio_output_flags_t flags, 113 callback_t cbf, 114 void* user, 115 int notificationFrames, 116 int sessionId, 117 transfer_type transferType, 118 const audio_offload_info_t *offloadInfo, 119 int uid, 120 pid_t pid) 121 : mStatus(NO_INIT), 122 mIsTimed(false), 123 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 124 mPreviousSchedulingGroup(SP_DEFAULT) 125{ 126 mStatus = set(streamType, sampleRate, format, channelMask, 127 frameCount, flags, cbf, user, notificationFrames, 128 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 129 offloadInfo, uid, pid); 130} 131 132AudioTrack::AudioTrack( 133 audio_stream_type_t streamType, 134 uint32_t sampleRate, 135 audio_format_t format, 136 audio_channel_mask_t channelMask, 137 const sp<IMemory>& sharedBuffer, 138 audio_output_flags_t flags, 139 callback_t cbf, 140 void* user, 141 int notificationFrames, 142 int sessionId, 143 transfer_type transferType, 144 const audio_offload_info_t *offloadInfo, 145 int uid, 146 pid_t pid) 147 : mStatus(NO_INIT), 148 mIsTimed(false), 149 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 150 mPreviousSchedulingGroup(SP_DEFAULT) 151{ 152 mStatus = set(streamType, sampleRate, format, channelMask, 153 0 /*frameCount*/, flags, cbf, user, notificationFrames, 154 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 155 uid, pid); 156} 157 158AudioTrack::~AudioTrack() 159{ 160 if (mStatus == NO_ERROR) { 161 // Make sure that callback function exits in the case where 162 // it is looping on buffer full condition in obtainBuffer(). 163 // Otherwise the callback thread will never exit. 164 stop(); 165 if (mAudioTrackThread != 0) { 166 mProxy->interrupt(); 167 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 168 mAudioTrackThread->requestExitAndWait(); 169 mAudioTrackThread.clear(); 170 } 171 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 172 mAudioTrack.clear(); 173 IPCThreadState::self()->flushCommands(); 174 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 175 IPCThreadState::self()->getCallingPid(), mClientPid); 176 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 177 } 178} 179 180status_t AudioTrack::set( 181 audio_stream_type_t streamType, 182 uint32_t sampleRate, 183 audio_format_t format, 184 audio_channel_mask_t channelMask, 185 int frameCountInt, 186 audio_output_flags_t flags, 187 callback_t cbf, 188 void* user, 189 int notificationFrames, 190 const sp<IMemory>& sharedBuffer, 191 bool threadCanCallJava, 192 int sessionId, 193 transfer_type transferType, 194 const audio_offload_info_t *offloadInfo, 195 int uid, 196 pid_t pid) 197{ 198 switch (transferType) { 199 case TRANSFER_DEFAULT: 200 if (sharedBuffer != 0) { 201 transferType = TRANSFER_SHARED; 202 } else if (cbf == NULL || threadCanCallJava) { 203 transferType = TRANSFER_SYNC; 204 } else { 205 transferType = TRANSFER_CALLBACK; 206 } 207 break; 208 case TRANSFER_CALLBACK: 209 if (cbf == NULL || sharedBuffer != 0) { 210 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 211 return BAD_VALUE; 212 } 213 break; 214 case TRANSFER_OBTAIN: 215 case TRANSFER_SYNC: 216 if (sharedBuffer != 0) { 217 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 218 return BAD_VALUE; 219 } 220 break; 221 case TRANSFER_SHARED: 222 if (sharedBuffer == 0) { 223 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 224 return BAD_VALUE; 225 } 226 break; 227 default: 228 ALOGE("Invalid transfer type %d", transferType); 229 return BAD_VALUE; 230 } 231 mSharedBuffer = sharedBuffer; 232 mTransfer = transferType; 233 234 // FIXME "int" here is legacy and will be replaced by size_t later 235 if (frameCountInt < 0) { 236 ALOGE("Invalid frame count %d", frameCountInt); 237 return BAD_VALUE; 238 } 239 size_t frameCount = frameCountInt; 240 241 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 242 sharedBuffer->size()); 243 244 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 245 246 AutoMutex lock(mLock); 247 248 // invariant that mAudioTrack != 0 is true only after set() returns successfully 249 if (mAudioTrack != 0) { 250 ALOGE("Track already in use"); 251 return INVALID_OPERATION; 252 } 253 254 // handle default values first. 255 if (streamType == AUDIO_STREAM_DEFAULT) { 256 streamType = AUDIO_STREAM_MUSIC; 257 } 258 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 259 ALOGE("Invalid stream type %d", streamType); 260 return BAD_VALUE; 261 } 262 mStreamType = streamType; 263 264 status_t status; 265 if (sampleRate == 0) { 266 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 267 if (status != NO_ERROR) { 268 ALOGE("Could not get output sample rate for stream type %d; status %d", 269 streamType, status); 270 return status; 271 } 272 } 273 mSampleRate = sampleRate; 274 275 // these below should probably come from the audioFlinger too... 276 if (format == AUDIO_FORMAT_DEFAULT) { 277 format = AUDIO_FORMAT_PCM_16_BIT; 278 } 279 280 // validate parameters 281 if (!audio_is_valid_format(format)) { 282 ALOGE("Invalid format %#x", format); 283 return BAD_VALUE; 284 } 285 mFormat = format; 286 287 if (!audio_is_output_channel(channelMask)) { 288 ALOGE("Invalid channel mask %#x", channelMask); 289 return BAD_VALUE; 290 } 291 292 // AudioFlinger does not currently support 8-bit data in shared memory 293 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 294 ALOGE("8-bit data in shared memory is not supported"); 295 return BAD_VALUE; 296 } 297 298 // force direct flag if format is not linear PCM 299 // or offload was requested 300 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 301 || !audio_is_linear_pcm(format)) { 302 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 303 ? "Offload request, forcing to Direct Output" 304 : "Not linear PCM, forcing to Direct Output"); 305 flags = (audio_output_flags_t) 306 // FIXME why can't we allow direct AND fast? 307 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 308 } 309 // only allow deep buffering for music stream type 310 if (streamType != AUDIO_STREAM_MUSIC) { 311 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 312 } 313 314 mChannelMask = channelMask; 315 uint32_t channelCount = popcount(channelMask); 316 mChannelCount = channelCount; 317 318 if (audio_is_linear_pcm(format)) { 319 mFrameSize = channelCount * audio_bytes_per_sample(format); 320 mFrameSizeAF = channelCount * sizeof(int16_t); 321 } else { 322 mFrameSize = sizeof(uint8_t); 323 mFrameSizeAF = sizeof(uint8_t); 324 } 325 326 // Make copy of input parameter offloadInfo so that in the future: 327 // (a) createTrack_l doesn't need it as an input parameter 328 // (b) we can support re-creation of offloaded tracks 329 if (offloadInfo != NULL) { 330 mOffloadInfoCopy = *offloadInfo; 331 mOffloadInfo = &mOffloadInfoCopy; 332 } else { 333 mOffloadInfo = NULL; 334 } 335 336 mVolume[LEFT] = 1.0f; 337 mVolume[RIGHT] = 1.0f; 338 mSendLevel = 0.0f; 339 // mFrameCount is initialized in createTrack_l 340 mReqFrameCount = frameCount; 341 mNotificationFramesReq = notificationFrames; 342 mNotificationFramesAct = 0; 343 mSessionId = sessionId; 344 int callingpid = IPCThreadState::self()->getCallingPid(); 345 int mypid = getpid(); 346 if (uid == -1 || (callingpid != mypid)) { 347 mClientUid = IPCThreadState::self()->getCallingUid(); 348 } else { 349 mClientUid = uid; 350 } 351 if (pid == -1 || (callingpid != mypid)) { 352 mClientPid = callingpid; 353 } else { 354 mClientPid = pid; 355 } 356 mAuxEffectId = 0; 357 mFlags = flags; 358 mCbf = cbf; 359 360 if (cbf != NULL) { 361 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 362 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 363 } 364 365 // create the IAudioTrack 366 status = createTrack_l(0 /*epoch*/); 367 368 if (status != NO_ERROR) { 369 if (mAudioTrackThread != 0) { 370 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 371 mAudioTrackThread->requestExitAndWait(); 372 mAudioTrackThread.clear(); 373 } 374 // Use of direct and offloaded output streams is ref counted by audio policy manager. 375#if 0 // FIXME This should no longer be needed 376 //Use of direct and offloaded output streams is ref counted by audio policy manager. 377 // As getOutput was called above and resulted in an output stream to be opened, 378 // we need to release it. 379 if (mOutput != 0) { 380 AudioSystem::releaseOutput(mOutput); 381 mOutput = 0; 382 } 383#endif 384 return status; 385 } 386 387 mStatus = NO_ERROR; 388 mState = STATE_STOPPED; 389 mUserData = user; 390 mLoopPeriod = 0; 391 mMarkerPosition = 0; 392 mMarkerReached = false; 393 mNewPosition = 0; 394 mUpdatePeriod = 0; 395 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 396 mSequence = 1; 397 mObservedSequence = mSequence; 398 mInUnderrun = false; 399 400 return NO_ERROR; 401} 402 403// ------------------------------------------------------------------------- 404 405status_t AudioTrack::start() 406{ 407 AutoMutex lock(mLock); 408 409 if (mState == STATE_ACTIVE) { 410 return INVALID_OPERATION; 411 } 412 413 mInUnderrun = true; 414 415 State previousState = mState; 416 if (previousState == STATE_PAUSED_STOPPING) { 417 mState = STATE_STOPPING; 418 } else { 419 mState = STATE_ACTIVE; 420 } 421 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 422 // reset current position as seen by client to 0 423 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 424 // force refresh of remaining frames by processAudioBuffer() as last 425 // write before stop could be partial. 426 mRefreshRemaining = true; 427 } 428 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 429 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 430 431 sp<AudioTrackThread> t = mAudioTrackThread; 432 if (t != 0) { 433 if (previousState == STATE_STOPPING) { 434 mProxy->interrupt(); 435 } else { 436 t->resume(); 437 } 438 } else { 439 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 440 get_sched_policy(0, &mPreviousSchedulingGroup); 441 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 442 } 443 444 status_t status = NO_ERROR; 445 if (!(flags & CBLK_INVALID)) { 446 status = mAudioTrack->start(); 447 if (status == DEAD_OBJECT) { 448 flags |= CBLK_INVALID; 449 } 450 } 451 if (flags & CBLK_INVALID) { 452 status = restoreTrack_l("start"); 453 } 454 455 if (status != NO_ERROR) { 456 ALOGE("start() status %d", status); 457 mState = previousState; 458 if (t != 0) { 459 if (previousState != STATE_STOPPING) { 460 t->pause(); 461 } 462 } else { 463 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 464 set_sched_policy(0, mPreviousSchedulingGroup); 465 } 466 } 467 468 return status; 469} 470 471void AudioTrack::stop() 472{ 473 AutoMutex lock(mLock); 474 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 475 return; 476 } 477 478 if (isOffloaded_l()) { 479 mState = STATE_STOPPING; 480 } else { 481 mState = STATE_STOPPED; 482 } 483 484 mProxy->interrupt(); 485 mAudioTrack->stop(); 486 // the playback head position will reset to 0, so if a marker is set, we need 487 // to activate it again 488 mMarkerReached = false; 489#if 0 490 // Force flush if a shared buffer is used otherwise audioflinger 491 // will not stop before end of buffer is reached. 492 // It may be needed to make sure that we stop playback, likely in case looping is on. 493 if (mSharedBuffer != 0) { 494 flush_l(); 495 } 496#endif 497 498 sp<AudioTrackThread> t = mAudioTrackThread; 499 if (t != 0) { 500 if (!isOffloaded_l()) { 501 t->pause(); 502 } 503 } else { 504 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 505 set_sched_policy(0, mPreviousSchedulingGroup); 506 } 507} 508 509bool AudioTrack::stopped() const 510{ 511 AutoMutex lock(mLock); 512 return mState != STATE_ACTIVE; 513} 514 515void AudioTrack::flush() 516{ 517 if (mSharedBuffer != 0) { 518 return; 519 } 520 AutoMutex lock(mLock); 521 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 522 return; 523 } 524 flush_l(); 525} 526 527void AudioTrack::flush_l() 528{ 529 ALOG_ASSERT(mState != STATE_ACTIVE); 530 531 // clear playback marker and periodic update counter 532 mMarkerPosition = 0; 533 mMarkerReached = false; 534 mUpdatePeriod = 0; 535 mRefreshRemaining = true; 536 537 mState = STATE_FLUSHED; 538 if (isOffloaded_l()) { 539 mProxy->interrupt(); 540 } 541 mProxy->flush(); 542 mAudioTrack->flush(); 543} 544 545void AudioTrack::pause() 546{ 547 AutoMutex lock(mLock); 548 if (mState == STATE_ACTIVE) { 549 mState = STATE_PAUSED; 550 } else if (mState == STATE_STOPPING) { 551 mState = STATE_PAUSED_STOPPING; 552 } else { 553 return; 554 } 555 mProxy->interrupt(); 556 mAudioTrack->pause(); 557} 558 559status_t AudioTrack::setVolume(float left, float right) 560{ 561 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 562 return BAD_VALUE; 563 } 564 565 AutoMutex lock(mLock); 566 mVolume[LEFT] = left; 567 mVolume[RIGHT] = right; 568 569 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 570 571 if (isOffloaded_l()) { 572 mAudioTrack->signal(); 573 } 574 return NO_ERROR; 575} 576 577status_t AudioTrack::setVolume(float volume) 578{ 579 return setVolume(volume, volume); 580} 581 582status_t AudioTrack::setAuxEffectSendLevel(float level) 583{ 584 if (level < 0.0f || level > 1.0f) { 585 return BAD_VALUE; 586 } 587 588 AutoMutex lock(mLock); 589 mSendLevel = level; 590 mProxy->setSendLevel(level); 591 592 return NO_ERROR; 593} 594 595void AudioTrack::getAuxEffectSendLevel(float* level) const 596{ 597 if (level != NULL) { 598 *level = mSendLevel; 599 } 600} 601 602status_t AudioTrack::setSampleRate(uint32_t rate) 603{ 604 if (mIsTimed || isOffloaded()) { 605 return INVALID_OPERATION; 606 } 607 608 uint32_t afSamplingRate; 609 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 610 return NO_INIT; 611 } 612 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 613 if (rate == 0 || rate > afSamplingRate*2 ) { 614 return BAD_VALUE; 615 } 616 617 AutoMutex lock(mLock); 618 mSampleRate = rate; 619 mProxy->setSampleRate(rate); 620 621 return NO_ERROR; 622} 623 624uint32_t AudioTrack::getSampleRate() const 625{ 626 if (mIsTimed) { 627 return 0; 628 } 629 630 AutoMutex lock(mLock); 631 632 // sample rate can be updated during playback by the offloaded decoder so we need to 633 // query the HAL and update if needed. 634// FIXME use Proxy return channel to update the rate from server and avoid polling here 635 if (isOffloaded_l()) { 636 if (mOutput != 0) { 637 uint32_t sampleRate = 0; 638 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 639 if (status == NO_ERROR) { 640 mSampleRate = sampleRate; 641 } 642 } 643 } 644 return mSampleRate; 645} 646 647status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 648{ 649 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 650 return INVALID_OPERATION; 651 } 652 653 if (loopCount == 0) { 654 ; 655 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 656 loopEnd - loopStart >= MIN_LOOP) { 657 ; 658 } else { 659 return BAD_VALUE; 660 } 661 662 AutoMutex lock(mLock); 663 // See setPosition() regarding setting parameters such as loop points or position while active 664 if (mState == STATE_ACTIVE) { 665 return INVALID_OPERATION; 666 } 667 setLoop_l(loopStart, loopEnd, loopCount); 668 return NO_ERROR; 669} 670 671void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 672{ 673 // FIXME If setting a loop also sets position to start of loop, then 674 // this is correct. Otherwise it should be removed. 675 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 676 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 677 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 678} 679 680status_t AudioTrack::setMarkerPosition(uint32_t marker) 681{ 682 // The only purpose of setting marker position is to get a callback 683 if (mCbf == NULL || isOffloaded()) { 684 return INVALID_OPERATION; 685 } 686 687 AutoMutex lock(mLock); 688 mMarkerPosition = marker; 689 mMarkerReached = false; 690 691 return NO_ERROR; 692} 693 694status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 695{ 696 if (isOffloaded()) { 697 return INVALID_OPERATION; 698 } 699 if (marker == NULL) { 700 return BAD_VALUE; 701 } 702 703 AutoMutex lock(mLock); 704 *marker = mMarkerPosition; 705 706 return NO_ERROR; 707} 708 709status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 710{ 711 // The only purpose of setting position update period is to get a callback 712 if (mCbf == NULL || isOffloaded()) { 713 return INVALID_OPERATION; 714 } 715 716 AutoMutex lock(mLock); 717 mNewPosition = mProxy->getPosition() + updatePeriod; 718 mUpdatePeriod = updatePeriod; 719 720 return NO_ERROR; 721} 722 723status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 724{ 725 if (isOffloaded()) { 726 return INVALID_OPERATION; 727 } 728 if (updatePeriod == NULL) { 729 return BAD_VALUE; 730 } 731 732 AutoMutex lock(mLock); 733 *updatePeriod = mUpdatePeriod; 734 735 return NO_ERROR; 736} 737 738status_t AudioTrack::setPosition(uint32_t position) 739{ 740 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 741 return INVALID_OPERATION; 742 } 743 if (position > mFrameCount) { 744 return BAD_VALUE; 745 } 746 747 AutoMutex lock(mLock); 748 // Currently we require that the player is inactive before setting parameters such as position 749 // or loop points. Otherwise, there could be a race condition: the application could read the 750 // current position, compute a new position or loop parameters, and then set that position or 751 // loop parameters but it would do the "wrong" thing since the position has continued to advance 752 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 753 // to specify how it wants to handle such scenarios. 754 if (mState == STATE_ACTIVE) { 755 return INVALID_OPERATION; 756 } 757 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 758 mLoopPeriod = 0; 759 // FIXME Check whether loops and setting position are incompatible in old code. 760 // If we use setLoop for both purposes we lose the capability to set the position while looping. 761 mStaticProxy->setLoop(position, mFrameCount, 0); 762 763 return NO_ERROR; 764} 765 766status_t AudioTrack::getPosition(uint32_t *position) const 767{ 768 if (position == NULL) { 769 return BAD_VALUE; 770 } 771 772 AutoMutex lock(mLock); 773 if (isOffloaded_l()) { 774 uint32_t dspFrames = 0; 775 776 if (mOutput != 0) { 777 uint32_t halFrames; 778 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 779 } 780 *position = dspFrames; 781 } else { 782 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 783 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 784 mProxy->getPosition(); 785 } 786 return NO_ERROR; 787} 788 789status_t AudioTrack::getBufferPosition(uint32_t *position) 790{ 791 if (mSharedBuffer == 0 || mIsTimed) { 792 return INVALID_OPERATION; 793 } 794 if (position == NULL) { 795 return BAD_VALUE; 796 } 797 798 AutoMutex lock(mLock); 799 *position = mStaticProxy->getBufferPosition(); 800 return NO_ERROR; 801} 802 803status_t AudioTrack::reload() 804{ 805 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 806 return INVALID_OPERATION; 807 } 808 809 AutoMutex lock(mLock); 810 // See setPosition() regarding setting parameters such as loop points or position while active 811 if (mState == STATE_ACTIVE) { 812 return INVALID_OPERATION; 813 } 814 mNewPosition = mUpdatePeriod; 815 mLoopPeriod = 0; 816 // FIXME The new code cannot reload while keeping a loop specified. 817 // Need to check how the old code handled this, and whether it's a significant change. 818 mStaticProxy->setLoop(0, mFrameCount, 0); 819 return NO_ERROR; 820} 821 822audio_io_handle_t AudioTrack::getOutput() const 823{ 824 AutoMutex lock(mLock); 825 return mOutput; 826} 827 828status_t AudioTrack::attachAuxEffect(int effectId) 829{ 830 AutoMutex lock(mLock); 831 status_t status = mAudioTrack->attachAuxEffect(effectId); 832 if (status == NO_ERROR) { 833 mAuxEffectId = effectId; 834 } 835 return status; 836} 837 838// ------------------------------------------------------------------------- 839 840// must be called with mLock held 841status_t AudioTrack::createTrack_l(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 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 851 mChannelMask, mFlags, mOffloadInfo); 852 if (output == 0) { 853 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 854 "channel mask %#x, flags %#x", 855 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 856 return BAD_VALUE; 857 } 858 { 859 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 860 // we must release it ourselves if anything goes wrong. 861 862 // Not all of these values are needed under all conditions, but it is easier to get them all 863 864 uint32_t afLatency; 865 status = AudioSystem::getLatency(output, mStreamType, &afLatency); 866 if (status != NO_ERROR) { 867 ALOGE("getLatency(%d) failed status %d", output, status); 868 goto release; 869 } 870 871 size_t afFrameCount; 872 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 873 if (status != NO_ERROR) { 874 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 875 goto release; 876 } 877 878 uint32_t afSampleRate; 879 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 880 if (status != NO_ERROR) { 881 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 882 goto release; 883 } 884 885 // Client decides whether the track is TIMED (see below), but can only express a preference 886 // for FAST. Server will perform additional tests. 887 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 888 // either of these use cases: 889 // use case 1: shared buffer 890 (mSharedBuffer != 0) || 891 // use case 2: callback handler 892 (mCbf != NULL)) && 893 // matching sample rate 894 (mSampleRate == afSampleRate))) { 895 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 896 // once denied, do not request again if IAudioTrack is re-created 897 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 898 } 899 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 900 901 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 902 // n = 1 fast track with single buffering; nBuffering is ignored 903 // n = 2 fast track with double buffering 904 // n = 2 normal track, no sample rate conversion 905 // n = 3 normal track, with sample rate conversion 906 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 907 // n > 3 very high latency or very small notification interval; nBuffering is ignored 908 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 909 910 mNotificationFramesAct = mNotificationFramesReq; 911 912 size_t frameCount = mReqFrameCount; 913 if (!audio_is_linear_pcm(mFormat)) { 914 915 if (mSharedBuffer != 0) { 916 // Same comment as below about ignoring frameCount parameter for set() 917 frameCount = mSharedBuffer->size(); 918 } else if (frameCount == 0) { 919 frameCount = afFrameCount; 920 } 921 if (mNotificationFramesAct != frameCount) { 922 mNotificationFramesAct = frameCount; 923 } 924 } else if (mSharedBuffer != 0) { 925 926 // Ensure that buffer alignment matches channel count 927 // 8-bit data in shared memory is not currently supported by AudioFlinger 928 size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 929 if (mChannelCount > 1) { 930 // More than 2 channels does not require stronger alignment than stereo 931 alignment <<= 1; 932 } 933 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 934 ALOGE("Invalid buffer alignment: address %p, channel count %u", 935 mSharedBuffer->pointer(), mChannelCount); 936 status = BAD_VALUE; 937 goto release; 938 } 939 940 // When initializing a shared buffer AudioTrack via constructors, 941 // there's no frameCount parameter. 942 // But when initializing a shared buffer AudioTrack via set(), 943 // there _is_ a frameCount parameter. We silently ignore it. 944 frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); 945 946 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 947 948 // FIXME move these calculations and associated checks to server 949 950 // Ensure that buffer depth covers at least audio hardware latency 951 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 952 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 953 afFrameCount, minBufCount, afSampleRate, afLatency); 954 if (minBufCount <= nBuffering) { 955 minBufCount = nBuffering; 956 } 957 958 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 959 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 960 ", afLatency=%d", 961 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 962 963 if (frameCount == 0) { 964 frameCount = minFrameCount; 965 } else if (frameCount < minFrameCount) { 966 // not ALOGW because it happens all the time when playing key clicks over A2DP 967 ALOGV("Minimum buffer size corrected from %d to %d", 968 frameCount, minFrameCount); 969 frameCount = minFrameCount; 970 } 971 // Make sure that application is notified with sufficient margin before underrun 972 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 973 mNotificationFramesAct = frameCount/nBuffering; 974 } 975 976 } else { 977 // For fast tracks, the frame count calculations and checks are done by server 978 } 979 980 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 981 if (mIsTimed) { 982 trackFlags |= IAudioFlinger::TRACK_TIMED; 983 } 984 985 pid_t tid = -1; 986 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 987 trackFlags |= IAudioFlinger::TRACK_FAST; 988 if (mAudioTrackThread != 0) { 989 tid = mAudioTrackThread->getTid(); 990 } 991 } 992 993 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 994 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 995 } 996 997 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 998 // but we will still need the original value also 999 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1000 mSampleRate, 1001 // AudioFlinger only sees 16-bit PCM 1002 mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1003 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1004 mChannelMask, 1005 &temp, 1006 &trackFlags, 1007 mSharedBuffer, 1008 output, 1009 tid, 1010 &mSessionId, 1011 mName, 1012 mClientUid, 1013 &status); 1014 1015 if (track == 0) { 1016 ALOGE("AudioFlinger could not create track, status: %d", status); 1017 goto release; 1018 } 1019 // AudioFlinger now owns the reference to the I/O handle, 1020 // so we are no longer responsible for releasing it. 1021 1022 sp<IMemory> iMem = track->getCblk(); 1023 if (iMem == 0) { 1024 ALOGE("Could not get control block"); 1025 return NO_INIT; 1026 } 1027 void *iMemPointer = iMem->pointer(); 1028 if (iMemPointer == NULL) { 1029 ALOGE("Could not get control block pointer"); 1030 return NO_INIT; 1031 } 1032 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1033 if (mAudioTrack != 0) { 1034 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1035 mDeathNotifier.clear(); 1036 } 1037 mAudioTrack = track; 1038 mCblkMemory = iMem; 1039 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1040 mCblk = cblk; 1041 // note that temp is the (possibly revised) value of frameCount 1042 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1043 // In current design, AudioTrack client checks and ensures frame count validity before 1044 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1045 // for fast track as it uses a special method of assigning frame count. 1046 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1047 } 1048 frameCount = temp; 1049 mAwaitBoost = false; 1050 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1051 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1052 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1053 mAwaitBoost = true; 1054 if (mSharedBuffer == 0) { 1055 // Theoretically double-buffering is not required for fast tracks, 1056 // due to tighter scheduling. But in practice, to accommodate kernels with 1057 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1058 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1059 mNotificationFramesAct = frameCount/nBuffering; 1060 } 1061 } 1062 } else { 1063 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1064 // once denied, do not request again if IAudioTrack is re-created 1065 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1066 if (mSharedBuffer == 0) { 1067 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1068 mNotificationFramesAct = frameCount/nBuffering; 1069 } 1070 } 1071 } 1072 } 1073 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1074 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1075 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1076 } else { 1077 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1078 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1079 // FIXME This is a warning, not an error, so don't return error status 1080 //return NO_INIT; 1081 } 1082 } 1083 1084 // We retain a copy of the I/O handle, but don't own the reference 1085 mOutput = output; 1086 mRefreshRemaining = true; 1087 1088 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1089 // is the value of pointer() for the shared buffer, otherwise buffers points 1090 // immediately after the control block. This address is for the mapping within client 1091 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1092 void* buffers; 1093 if (mSharedBuffer == 0) { 1094 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1095 } else { 1096 buffers = mSharedBuffer->pointer(); 1097 } 1098 1099 mAudioTrack->attachAuxEffect(mAuxEffectId); 1100 // FIXME don't believe this lie 1101 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1102 mFrameCount = frameCount; 1103 // If IAudioTrack is re-created, don't let the requested frameCount 1104 // decrease. This can confuse clients that cache frameCount(). 1105 if (frameCount > mReqFrameCount) { 1106 mReqFrameCount = frameCount; 1107 } 1108 1109 // update proxy 1110 if (mSharedBuffer == 0) { 1111 mStaticProxy.clear(); 1112 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1113 } else { 1114 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1115 mProxy = mStaticProxy; 1116 } 1117 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1118 uint16_t(mVolume[LEFT] * 0x1000)); 1119 mProxy->setSendLevel(mSendLevel); 1120 mProxy->setSampleRate(mSampleRate); 1121 mProxy->setEpoch(epoch); 1122 mProxy->setMinimum(mNotificationFramesAct); 1123 1124 mDeathNotifier = new DeathNotifier(this); 1125 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1126 1127 return NO_ERROR; 1128 } 1129 1130release: 1131 AudioSystem::releaseOutput(output); 1132 if (status == NO_ERROR) { 1133 status = NO_INIT; 1134 } 1135 return status; 1136} 1137 1138status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1139{ 1140 if (audioBuffer == NULL) { 1141 return BAD_VALUE; 1142 } 1143 if (mTransfer != TRANSFER_OBTAIN) { 1144 audioBuffer->frameCount = 0; 1145 audioBuffer->size = 0; 1146 audioBuffer->raw = NULL; 1147 return INVALID_OPERATION; 1148 } 1149 1150 const struct timespec *requested; 1151 struct timespec timeout; 1152 if (waitCount == -1) { 1153 requested = &ClientProxy::kForever; 1154 } else if (waitCount == 0) { 1155 requested = &ClientProxy::kNonBlocking; 1156 } else if (waitCount > 0) { 1157 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1158 timeout.tv_sec = ms / 1000; 1159 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1160 requested = &timeout; 1161 } else { 1162 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1163 requested = NULL; 1164 } 1165 return obtainBuffer(audioBuffer, requested); 1166} 1167 1168status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1169 struct timespec *elapsed, size_t *nonContig) 1170{ 1171 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1172 uint32_t oldSequence = 0; 1173 uint32_t newSequence; 1174 1175 Proxy::Buffer buffer; 1176 status_t status = NO_ERROR; 1177 1178 static const int32_t kMaxTries = 5; 1179 int32_t tryCounter = kMaxTries; 1180 1181 do { 1182 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1183 // keep them from going away if another thread re-creates the track during obtainBuffer() 1184 sp<AudioTrackClientProxy> proxy; 1185 sp<IMemory> iMem; 1186 1187 { // start of lock scope 1188 AutoMutex lock(mLock); 1189 1190 newSequence = mSequence; 1191 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1192 if (status == DEAD_OBJECT) { 1193 // re-create track, unless someone else has already done so 1194 if (newSequence == oldSequence) { 1195 status = restoreTrack_l("obtainBuffer"); 1196 if (status != NO_ERROR) { 1197 buffer.mFrameCount = 0; 1198 buffer.mRaw = NULL; 1199 buffer.mNonContig = 0; 1200 break; 1201 } 1202 } 1203 } 1204 oldSequence = newSequence; 1205 1206 // Keep the extra references 1207 proxy = mProxy; 1208 iMem = mCblkMemory; 1209 1210 if (mState == STATE_STOPPING) { 1211 status = -EINTR; 1212 buffer.mFrameCount = 0; 1213 buffer.mRaw = NULL; 1214 buffer.mNonContig = 0; 1215 break; 1216 } 1217 1218 // Non-blocking if track is stopped or paused 1219 if (mState != STATE_ACTIVE) { 1220 requested = &ClientProxy::kNonBlocking; 1221 } 1222 1223 } // end of lock scope 1224 1225 buffer.mFrameCount = audioBuffer->frameCount; 1226 // FIXME starts the requested timeout and elapsed over from scratch 1227 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1228 1229 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1230 1231 audioBuffer->frameCount = buffer.mFrameCount; 1232 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1233 audioBuffer->raw = buffer.mRaw; 1234 if (nonContig != NULL) { 1235 *nonContig = buffer.mNonContig; 1236 } 1237 return status; 1238} 1239 1240void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1241{ 1242 if (mTransfer == TRANSFER_SHARED) { 1243 return; 1244 } 1245 1246 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1247 if (stepCount == 0) { 1248 return; 1249 } 1250 1251 Proxy::Buffer buffer; 1252 buffer.mFrameCount = stepCount; 1253 buffer.mRaw = audioBuffer->raw; 1254 1255 AutoMutex lock(mLock); 1256 mInUnderrun = false; 1257 mProxy->releaseBuffer(&buffer); 1258 1259 // restart track if it was disabled by audioflinger due to previous underrun 1260 if (mState == STATE_ACTIVE) { 1261 audio_track_cblk_t* cblk = mCblk; 1262 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1263 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1264 this, mName.string()); 1265 // FIXME ignoring status 1266 mAudioTrack->start(); 1267 } 1268 } 1269} 1270 1271// ------------------------------------------------------------------------- 1272 1273ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1274{ 1275 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1276 return INVALID_OPERATION; 1277 } 1278 1279 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1280 // Sanity-check: user is most-likely passing an error code, and it would 1281 // make the return value ambiguous (actualSize vs error). 1282 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1283 return BAD_VALUE; 1284 } 1285 1286 size_t written = 0; 1287 Buffer audioBuffer; 1288 1289 while (userSize >= mFrameSize) { 1290 audioBuffer.frameCount = userSize / mFrameSize; 1291 1292 status_t err = obtainBuffer(&audioBuffer, 1293 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1294 if (err < 0) { 1295 if (written > 0) { 1296 break; 1297 } 1298 return ssize_t(err); 1299 } 1300 1301 size_t toWrite; 1302 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1303 // Divide capacity by 2 to take expansion into account 1304 toWrite = audioBuffer.size >> 1; 1305 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1306 } else { 1307 toWrite = audioBuffer.size; 1308 memcpy(audioBuffer.i8, buffer, toWrite); 1309 } 1310 buffer = ((const char *) buffer) + toWrite; 1311 userSize -= toWrite; 1312 written += toWrite; 1313 1314 releaseBuffer(&audioBuffer); 1315 } 1316 1317 return written; 1318} 1319 1320// ------------------------------------------------------------------------- 1321 1322TimedAudioTrack::TimedAudioTrack() { 1323 mIsTimed = true; 1324} 1325 1326status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1327{ 1328 AutoMutex lock(mLock); 1329 status_t result = UNKNOWN_ERROR; 1330 1331#if 1 1332 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1333 // while we are accessing the cblk 1334 sp<IAudioTrack> audioTrack = mAudioTrack; 1335 sp<IMemory> iMem = mCblkMemory; 1336#endif 1337 1338 // If the track is not invalid already, try to allocate a buffer. alloc 1339 // fails indicating that the server is dead, flag the track as invalid so 1340 // we can attempt to restore in just a bit. 1341 audio_track_cblk_t* cblk = mCblk; 1342 if (!(cblk->mFlags & CBLK_INVALID)) { 1343 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1344 if (result == DEAD_OBJECT) { 1345 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1346 } 1347 } 1348 1349 // If the track is invalid at this point, attempt to restore it. and try the 1350 // allocation one more time. 1351 if (cblk->mFlags & CBLK_INVALID) { 1352 result = restoreTrack_l("allocateTimedBuffer"); 1353 1354 if (result == NO_ERROR) { 1355 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1356 } 1357 } 1358 1359 return result; 1360} 1361 1362status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1363 int64_t pts) 1364{ 1365 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1366 { 1367 AutoMutex lock(mLock); 1368 audio_track_cblk_t* cblk = mCblk; 1369 // restart track if it was disabled by audioflinger due to previous underrun 1370 if (buffer->size() != 0 && status == NO_ERROR && 1371 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1372 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1373 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1374 // FIXME ignoring status 1375 mAudioTrack->start(); 1376 } 1377 } 1378 return status; 1379} 1380 1381status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1382 TargetTimeline target) 1383{ 1384 return mAudioTrack->setMediaTimeTransform(xform, target); 1385} 1386 1387// ------------------------------------------------------------------------- 1388 1389nsecs_t AudioTrack::processAudioBuffer() 1390{ 1391 // Currently the AudioTrack thread is not created if there are no callbacks. 1392 // Would it ever make sense to run the thread, even without callbacks? 1393 // If so, then replace this by checks at each use for mCbf != NULL. 1394 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1395 1396 mLock.lock(); 1397 if (mAwaitBoost) { 1398 mAwaitBoost = false; 1399 mLock.unlock(); 1400 static const int32_t kMaxTries = 5; 1401 int32_t tryCounter = kMaxTries; 1402 uint32_t pollUs = 10000; 1403 do { 1404 int policy = sched_getscheduler(0); 1405 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1406 break; 1407 } 1408 usleep(pollUs); 1409 pollUs <<= 1; 1410 } while (tryCounter-- > 0); 1411 if (tryCounter < 0) { 1412 ALOGE("did not receive expected priority boost on time"); 1413 } 1414 // Run again immediately 1415 return 0; 1416 } 1417 1418 // Can only reference mCblk while locked 1419 int32_t flags = android_atomic_and( 1420 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1421 1422 // Check for track invalidation 1423 if (flags & CBLK_INVALID) { 1424 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1425 // AudioSystem cache. We should not exit here but after calling the callback so 1426 // that the upper layers can recreate the track 1427 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1428 status_t status = restoreTrack_l("processAudioBuffer"); 1429 mLock.unlock(); 1430 // Run again immediately, but with a new IAudioTrack 1431 return 0; 1432 } 1433 } 1434 1435 bool waitStreamEnd = mState == STATE_STOPPING; 1436 bool active = mState == STATE_ACTIVE; 1437 1438 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1439 bool newUnderrun = false; 1440 if (flags & CBLK_UNDERRUN) { 1441#if 0 1442 // Currently in shared buffer mode, when the server reaches the end of buffer, 1443 // the track stays active in continuous underrun state. It's up to the application 1444 // to pause or stop the track, or set the position to a new offset within buffer. 1445 // This was some experimental code to auto-pause on underrun. Keeping it here 1446 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1447 if (mTransfer == TRANSFER_SHARED) { 1448 mState = STATE_PAUSED; 1449 active = false; 1450 } 1451#endif 1452 if (!mInUnderrun) { 1453 mInUnderrun = true; 1454 newUnderrun = true; 1455 } 1456 } 1457 1458 // Get current position of server 1459 size_t position = mProxy->getPosition(); 1460 1461 // Manage marker callback 1462 bool markerReached = false; 1463 size_t markerPosition = mMarkerPosition; 1464 // FIXME fails for wraparound, need 64 bits 1465 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1466 mMarkerReached = markerReached = true; 1467 } 1468 1469 // Determine number of new position callback(s) that will be needed, while locked 1470 size_t newPosCount = 0; 1471 size_t newPosition = mNewPosition; 1472 size_t updatePeriod = mUpdatePeriod; 1473 // FIXME fails for wraparound, need 64 bits 1474 if (updatePeriod > 0 && position >= newPosition) { 1475 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1476 mNewPosition += updatePeriod * newPosCount; 1477 } 1478 1479 // Cache other fields that will be needed soon 1480 uint32_t loopPeriod = mLoopPeriod; 1481 uint32_t sampleRate = mSampleRate; 1482 size_t notificationFrames = mNotificationFramesAct; 1483 if (mRefreshRemaining) { 1484 mRefreshRemaining = false; 1485 mRemainingFrames = notificationFrames; 1486 mRetryOnPartialBuffer = false; 1487 } 1488 size_t misalignment = mProxy->getMisalignment(); 1489 uint32_t sequence = mSequence; 1490 1491 // These fields don't need to be cached, because they are assigned only by set(): 1492 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1493 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1494 1495 mLock.unlock(); 1496 1497 if (waitStreamEnd) { 1498 AutoMutex lock(mLock); 1499 1500 sp<AudioTrackClientProxy> proxy = mProxy; 1501 sp<IMemory> iMem = mCblkMemory; 1502 1503 struct timespec timeout; 1504 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1505 timeout.tv_nsec = 0; 1506 1507 mLock.unlock(); 1508 status_t status = mProxy->waitStreamEndDone(&timeout); 1509 mLock.lock(); 1510 switch (status) { 1511 case NO_ERROR: 1512 case DEAD_OBJECT: 1513 case TIMED_OUT: 1514 mLock.unlock(); 1515 mCbf(EVENT_STREAM_END, mUserData, NULL); 1516 mLock.lock(); 1517 if (mState == STATE_STOPPING) { 1518 mState = STATE_STOPPED; 1519 if (status != DEAD_OBJECT) { 1520 return NS_INACTIVE; 1521 } 1522 } 1523 return 0; 1524 default: 1525 return 0; 1526 } 1527 } 1528 1529 // perform callbacks while unlocked 1530 if (newUnderrun) { 1531 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1532 } 1533 // FIXME we will miss loops if loop cycle was signaled several times since last call 1534 // to processAudioBuffer() 1535 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1536 mCbf(EVENT_LOOP_END, mUserData, NULL); 1537 } 1538 if (flags & CBLK_BUFFER_END) { 1539 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1540 } 1541 if (markerReached) { 1542 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1543 } 1544 while (newPosCount > 0) { 1545 size_t temp = newPosition; 1546 mCbf(EVENT_NEW_POS, mUserData, &temp); 1547 newPosition += updatePeriod; 1548 newPosCount--; 1549 } 1550 1551 if (mObservedSequence != sequence) { 1552 mObservedSequence = sequence; 1553 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1554 // for offloaded tracks, just wait for the upper layers to recreate the track 1555 if (isOffloaded()) { 1556 return NS_INACTIVE; 1557 } 1558 } 1559 1560 // if inactive, then don't run me again until re-started 1561 if (!active) { 1562 return NS_INACTIVE; 1563 } 1564 1565 // Compute the estimated time until the next timed event (position, markers, loops) 1566 // FIXME only for non-compressed audio 1567 uint32_t minFrames = ~0; 1568 if (!markerReached && position < markerPosition) { 1569 minFrames = markerPosition - position; 1570 } 1571 if (loopPeriod > 0 && loopPeriod < minFrames) { 1572 minFrames = loopPeriod; 1573 } 1574 if (updatePeriod > 0 && updatePeriod < minFrames) { 1575 minFrames = updatePeriod; 1576 } 1577 1578 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1579 static const uint32_t kPoll = 0; 1580 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1581 minFrames = kPoll * notificationFrames; 1582 } 1583 1584 // Convert frame units to time units 1585 nsecs_t ns = NS_WHENEVER; 1586 if (minFrames != (uint32_t) ~0) { 1587 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1588 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1589 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1590 } 1591 1592 // If not supplying data by EVENT_MORE_DATA, then we're done 1593 if (mTransfer != TRANSFER_CALLBACK) { 1594 return ns; 1595 } 1596 1597 struct timespec timeout; 1598 const struct timespec *requested = &ClientProxy::kForever; 1599 if (ns != NS_WHENEVER) { 1600 timeout.tv_sec = ns / 1000000000LL; 1601 timeout.tv_nsec = ns % 1000000000LL; 1602 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1603 requested = &timeout; 1604 } 1605 1606 while (mRemainingFrames > 0) { 1607 1608 Buffer audioBuffer; 1609 audioBuffer.frameCount = mRemainingFrames; 1610 size_t nonContig; 1611 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1612 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1613 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1614 requested = &ClientProxy::kNonBlocking; 1615 size_t avail = audioBuffer.frameCount + nonContig; 1616 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1617 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1618 if (err != NO_ERROR) { 1619 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1620 (isOffloaded() && (err == DEAD_OBJECT))) { 1621 return 0; 1622 } 1623 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1624 return NS_NEVER; 1625 } 1626 1627 if (mRetryOnPartialBuffer && !isOffloaded()) { 1628 mRetryOnPartialBuffer = false; 1629 if (avail < mRemainingFrames) { 1630 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1631 if (ns < 0 || myns < ns) { 1632 ns = myns; 1633 } 1634 return ns; 1635 } 1636 } 1637 1638 // Divide buffer size by 2 to take into account the expansion 1639 // due to 8 to 16 bit conversion: the callback must fill only half 1640 // of the destination buffer 1641 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1642 audioBuffer.size >>= 1; 1643 } 1644 1645 size_t reqSize = audioBuffer.size; 1646 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1647 size_t writtenSize = audioBuffer.size; 1648 1649 // Sanity check on returned size 1650 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1651 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1652 reqSize, (int) writtenSize); 1653 return NS_NEVER; 1654 } 1655 1656 if (writtenSize == 0) { 1657 // The callback is done filling buffers 1658 // Keep this thread going to handle timed events and 1659 // still try to get more data in intervals of WAIT_PERIOD_MS 1660 // but don't just loop and block the CPU, so wait 1661 return WAIT_PERIOD_MS * 1000000LL; 1662 } 1663 1664 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1665 // 8 to 16 bit conversion, note that source and destination are the same address 1666 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1667 audioBuffer.size <<= 1; 1668 } 1669 1670 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1671 audioBuffer.frameCount = releasedFrames; 1672 mRemainingFrames -= releasedFrames; 1673 if (misalignment >= releasedFrames) { 1674 misalignment -= releasedFrames; 1675 } else { 1676 misalignment = 0; 1677 } 1678 1679 releaseBuffer(&audioBuffer); 1680 1681 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1682 // if callback doesn't like to accept the full chunk 1683 if (writtenSize < reqSize) { 1684 continue; 1685 } 1686 1687 // There could be enough non-contiguous frames available to satisfy the remaining request 1688 if (mRemainingFrames <= nonContig) { 1689 continue; 1690 } 1691 1692#if 0 1693 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1694 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1695 // that total to a sum == notificationFrames. 1696 if (0 < misalignment && misalignment <= mRemainingFrames) { 1697 mRemainingFrames = misalignment; 1698 return (mRemainingFrames * 1100000000LL) / sampleRate; 1699 } 1700#endif 1701 1702 } 1703 mRemainingFrames = notificationFrames; 1704 mRetryOnPartialBuffer = true; 1705 1706 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1707 return 0; 1708} 1709 1710status_t AudioTrack::restoreTrack_l(const char *from) 1711{ 1712 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1713 isOffloaded_l() ? "Offloaded" : "PCM", from); 1714 ++mSequence; 1715 status_t result; 1716 1717 // refresh the audio configuration cache in this process to make sure we get new 1718 // output parameters in createTrack_l() 1719 AudioSystem::clearAudioConfigCache(); 1720 1721 if (isOffloaded_l()) { 1722 // FIXME re-creation of offloaded tracks is not yet implemented 1723 return DEAD_OBJECT; 1724 } 1725 1726 // if the new IAudioTrack is created, createTrack_l() will modify the 1727 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1728 // It will also delete the strong references on previous IAudioTrack and IMemory 1729 1730 // take the frames that will be lost by track recreation into account in saved position 1731 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1732 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1733 result = createTrack_l(position /*epoch*/); 1734 1735 if (result == NO_ERROR) { 1736 // continue playback from last known position, but 1737 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1738 if (mStaticProxy != NULL) { 1739 mLoopPeriod = 0; 1740 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1741 } 1742 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1743 // track destruction have been played? This is critical for SoundPool implementation 1744 // This must be broken, and needs to be tested/debugged. 1745#if 0 1746 // restore write index and set other indexes to reflect empty buffer status 1747 if (!strcmp(from, "start")) { 1748 // Make sure that a client relying on callback events indicating underrun or 1749 // the actual amount of audio frames played (e.g SoundPool) receives them. 1750 if (mSharedBuffer == 0) { 1751 // restart playback even if buffer is not completely filled. 1752 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1753 } 1754 } 1755#endif 1756 if (mState == STATE_ACTIVE) { 1757 result = mAudioTrack->start(); 1758 } 1759 } 1760 if (result != NO_ERROR) { 1761 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1762#if 0 // FIXME This should no longer be needed 1763 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1764 // As getOutput was called above and resulted in an output stream to be opened, 1765 // we need to release it. 1766 if (mOutput != 0) { 1767 AudioSystem::releaseOutput(mOutput); 1768 mOutput = 0; 1769 } 1770#endif 1771 ALOGW("restoreTrack_l() failed status %d", result); 1772 mState = STATE_STOPPED; 1773 } 1774 1775 return result; 1776} 1777 1778status_t AudioTrack::setParameters(const String8& keyValuePairs) 1779{ 1780 AutoMutex lock(mLock); 1781 return mAudioTrack->setParameters(keyValuePairs); 1782} 1783 1784status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1785{ 1786 AutoMutex lock(mLock); 1787 // FIXME not implemented for fast tracks; should use proxy and SSQ 1788 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1789 return INVALID_OPERATION; 1790 } 1791 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1792 return INVALID_OPERATION; 1793 } 1794 status_t status = mAudioTrack->getTimestamp(timestamp); 1795 if (status == NO_ERROR) { 1796 timestamp.mPosition += mProxy->getEpoch(); 1797 } 1798 return status; 1799} 1800 1801String8 AudioTrack::getParameters(const String8& keys) 1802{ 1803 audio_io_handle_t output = getOutput(); 1804 if (output != 0) { 1805 return AudioSystem::getParameters(output, keys); 1806 } else { 1807 return String8::empty(); 1808 } 1809} 1810 1811bool AudioTrack::isOffloaded() const 1812{ 1813 AutoMutex lock(mLock); 1814 return isOffloaded_l(); 1815} 1816 1817status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1818{ 1819 1820 const size_t SIZE = 256; 1821 char buffer[SIZE]; 1822 String8 result; 1823 1824 result.append(" AudioTrack::dump\n"); 1825 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1826 mVolume[0], mVolume[1]); 1827 result.append(buffer); 1828 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1829 mChannelCount, mFrameCount); 1830 result.append(buffer); 1831 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1832 result.append(buffer); 1833 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1834 result.append(buffer); 1835 ::write(fd, result.string(), result.size()); 1836 return NO_ERROR; 1837} 1838 1839uint32_t AudioTrack::getUnderrunFrames() const 1840{ 1841 AutoMutex lock(mLock); 1842 return mProxy->getUnderrunFrames(); 1843} 1844 1845// ========================================================================= 1846 1847void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1848{ 1849 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1850 if (audioTrack != 0) { 1851 AutoMutex lock(audioTrack->mLock); 1852 audioTrack->mProxy->binderDied(); 1853 } 1854} 1855 1856// ========================================================================= 1857 1858AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1859 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1860 mIgnoreNextPausedInt(false) 1861{ 1862} 1863 1864AudioTrack::AudioTrackThread::~AudioTrackThread() 1865{ 1866} 1867 1868bool AudioTrack::AudioTrackThread::threadLoop() 1869{ 1870 { 1871 AutoMutex _l(mMyLock); 1872 if (mPaused) { 1873 mMyCond.wait(mMyLock); 1874 // caller will check for exitPending() 1875 return true; 1876 } 1877 if (mIgnoreNextPausedInt) { 1878 mIgnoreNextPausedInt = false; 1879 mPausedInt = false; 1880 } 1881 if (mPausedInt) { 1882 if (mPausedNs > 0) { 1883 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1884 } else { 1885 mMyCond.wait(mMyLock); 1886 } 1887 mPausedInt = false; 1888 return true; 1889 } 1890 } 1891 nsecs_t ns = mReceiver.processAudioBuffer(); 1892 switch (ns) { 1893 case 0: 1894 return true; 1895 case NS_INACTIVE: 1896 pauseInternal(); 1897 return true; 1898 case NS_NEVER: 1899 return false; 1900 case NS_WHENEVER: 1901 // FIXME increase poll interval, or make event-driven 1902 ns = 1000000000LL; 1903 // fall through 1904 default: 1905 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1906 pauseInternal(ns); 1907 return true; 1908 } 1909} 1910 1911void AudioTrack::AudioTrackThread::requestExit() 1912{ 1913 // must be in this order to avoid a race condition 1914 Thread::requestExit(); 1915 resume(); 1916} 1917 1918void AudioTrack::AudioTrackThread::pause() 1919{ 1920 AutoMutex _l(mMyLock); 1921 mPaused = true; 1922} 1923 1924void AudioTrack::AudioTrackThread::resume() 1925{ 1926 AutoMutex _l(mMyLock); 1927 mIgnoreNextPausedInt = true; 1928 if (mPaused || mPausedInt) { 1929 mPaused = false; 1930 mPausedInt = false; 1931 mMyCond.signal(); 1932 } 1933} 1934 1935void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1936{ 1937 AutoMutex _l(mMyLock); 1938 mPausedInt = true; 1939 mPausedNs = ns; 1940} 1941 1942}; // namespace android 1943