AudioTrack.cpp revision 398f21348e5100289f6e5be30c8b5257fa04aaf9
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 if (channelMask == 0) { 254 channelMask = AUDIO_CHANNEL_OUT_STEREO; 255 } 256 257 // validate parameters 258 if (!audio_is_valid_format(format)) { 259 ALOGE("Invalid format %d", format); 260 return BAD_VALUE; 261 } 262 263 // AudioFlinger does not currently support 8-bit data in shared memory 264 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 265 ALOGE("8-bit data in shared memory is not supported"); 266 return BAD_VALUE; 267 } 268 269 // force direct flag if format is not linear PCM 270 // or offload was requested 271 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 272 || !audio_is_linear_pcm(format)) { 273 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 274 ? "Offload request, forcing to Direct Output" 275 : "Not linear PCM, forcing to Direct Output"); 276 flags = (audio_output_flags_t) 277 // FIXME why can't we allow direct AND fast? 278 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 279 } 280 // only allow deep buffering for music stream type 281 if (streamType != AUDIO_STREAM_MUSIC) { 282 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 283 } 284 285 if (!audio_is_output_channel(channelMask)) { 286 ALOGE("Invalid channel mask %#x", channelMask); 287 return BAD_VALUE; 288 } 289 mChannelMask = channelMask; 290 uint32_t channelCount = popcount(channelMask); 291 mChannelCount = channelCount; 292 293 if (audio_is_linear_pcm(format)) { 294 mFrameSize = channelCount * audio_bytes_per_sample(format); 295 mFrameSizeAF = channelCount * sizeof(int16_t); 296 } else { 297 mFrameSize = sizeof(uint8_t); 298 mFrameSizeAF = sizeof(uint8_t); 299 } 300 301 audio_io_handle_t output = AudioSystem::getOutput( 302 streamType, 303 sampleRate, format, channelMask, 304 flags, 305 offloadInfo); 306 307 if (output == 0) { 308 ALOGE("Could not get audio output for stream type %d", streamType); 309 return BAD_VALUE; 310 } 311 312 mVolume[LEFT] = 1.0f; 313 mVolume[RIGHT] = 1.0f; 314 mSendLevel = 0.0f; 315 mFrameCount = frameCount; 316 mReqFrameCount = frameCount; 317 mNotificationFramesReq = notificationFrames; 318 mNotificationFramesAct = 0; 319 mSessionId = sessionId; 320 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 321 mClientUid = IPCThreadState::self()->getCallingUid(); 322 } else { 323 mClientUid = uid; 324 } 325 mAuxEffectId = 0; 326 mFlags = flags; 327 mCbf = cbf; 328 329 if (cbf != NULL) { 330 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 331 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 332 } 333 334 // create the IAudioTrack 335 status_t status = createTrack_l(streamType, 336 sampleRate, 337 format, 338 frameCount, 339 flags, 340 sharedBuffer, 341 output, 342 0 /*epoch*/); 343 344 if (status != NO_ERROR) { 345 if (mAudioTrackThread != 0) { 346 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 347 mAudioTrackThread->requestExitAndWait(); 348 mAudioTrackThread.clear(); 349 } 350 //Use of direct and offloaded output streams is ref counted by audio policy manager. 351 // As getOutput was called above and resulted in an output stream to be opened, 352 // we need to release it. 353 AudioSystem::releaseOutput(output); 354 return status; 355 } 356 357 mStatus = NO_ERROR; 358 mStreamType = streamType; 359 mFormat = format; 360 mSharedBuffer = sharedBuffer; 361 mState = STATE_STOPPED; 362 mUserData = user; 363 mLoopPeriod = 0; 364 mMarkerPosition = 0; 365 mMarkerReached = false; 366 mNewPosition = 0; 367 mUpdatePeriod = 0; 368 AudioSystem::acquireAudioSessionId(mSessionId); 369 mSequence = 1; 370 mObservedSequence = mSequence; 371 mInUnderrun = false; 372 mOutput = output; 373 374 return NO_ERROR; 375} 376 377// ------------------------------------------------------------------------- 378 379status_t AudioTrack::start() 380{ 381 AutoMutex lock(mLock); 382 383 if (mState == STATE_ACTIVE) { 384 return INVALID_OPERATION; 385 } 386 387 mInUnderrun = true; 388 389 State previousState = mState; 390 if (previousState == STATE_PAUSED_STOPPING) { 391 mState = STATE_STOPPING; 392 } else { 393 mState = STATE_ACTIVE; 394 } 395 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 396 // reset current position as seen by client to 0 397 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 398 // force refresh of remaining frames by processAudioBuffer() as last 399 // write before stop could be partial. 400 mRefreshRemaining = true; 401 } 402 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 403 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 404 405 sp<AudioTrackThread> t = mAudioTrackThread; 406 if (t != 0) { 407 if (previousState == STATE_STOPPING) { 408 mProxy->interrupt(); 409 } else { 410 t->resume(); 411 } 412 } else { 413 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 414 get_sched_policy(0, &mPreviousSchedulingGroup); 415 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 416 } 417 418 status_t status = NO_ERROR; 419 if (!(flags & CBLK_INVALID)) { 420 status = mAudioTrack->start(); 421 if (status == DEAD_OBJECT) { 422 flags |= CBLK_INVALID; 423 } 424 } 425 if (flags & CBLK_INVALID) { 426 status = restoreTrack_l("start"); 427 } 428 429 if (status != NO_ERROR) { 430 ALOGE("start() status %d", status); 431 mState = previousState; 432 if (t != 0) { 433 if (previousState != STATE_STOPPING) { 434 t->pause(); 435 } 436 } else { 437 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 438 set_sched_policy(0, mPreviousSchedulingGroup); 439 } 440 } 441 442 return status; 443} 444 445void AudioTrack::stop() 446{ 447 AutoMutex lock(mLock); 448 // FIXME pause then stop should not be a nop 449 if (mState != STATE_ACTIVE) { 450 return; 451 } 452 453 if (isOffloaded()) { 454 mState = STATE_STOPPING; 455 } else { 456 mState = STATE_STOPPED; 457 } 458 459 mProxy->interrupt(); 460 mAudioTrack->stop(); 461 // the playback head position will reset to 0, so if a marker is set, we need 462 // to activate it again 463 mMarkerReached = false; 464#if 0 465 // Force flush if a shared buffer is used otherwise audioflinger 466 // will not stop before end of buffer is reached. 467 // It may be needed to make sure that we stop playback, likely in case looping is on. 468 if (mSharedBuffer != 0) { 469 flush_l(); 470 } 471#endif 472 473 sp<AudioTrackThread> t = mAudioTrackThread; 474 if (t != 0) { 475 if (!isOffloaded()) { 476 t->pause(); 477 } 478 } else { 479 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 480 set_sched_policy(0, mPreviousSchedulingGroup); 481 } 482} 483 484bool AudioTrack::stopped() const 485{ 486 AutoMutex lock(mLock); 487 return mState != STATE_ACTIVE; 488} 489 490void AudioTrack::flush() 491{ 492 if (mSharedBuffer != 0) { 493 return; 494 } 495 AutoMutex lock(mLock); 496 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 497 return; 498 } 499 flush_l(); 500} 501 502void AudioTrack::flush_l() 503{ 504 ALOG_ASSERT(mState != STATE_ACTIVE); 505 506 // clear playback marker and periodic update counter 507 mMarkerPosition = 0; 508 mMarkerReached = false; 509 mUpdatePeriod = 0; 510 mRefreshRemaining = true; 511 512 mState = STATE_FLUSHED; 513 if (isOffloaded()) { 514 mProxy->interrupt(); 515 } 516 mProxy->flush(); 517 mAudioTrack->flush(); 518} 519 520void AudioTrack::pause() 521{ 522 AutoMutex lock(mLock); 523 if (mState == STATE_ACTIVE) { 524 mState = STATE_PAUSED; 525 } else if (mState == STATE_STOPPING) { 526 mState = STATE_PAUSED_STOPPING; 527 } else { 528 return; 529 } 530 mProxy->interrupt(); 531 mAudioTrack->pause(); 532} 533 534status_t AudioTrack::setVolume(float left, float right) 535{ 536 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 537 return BAD_VALUE; 538 } 539 540 AutoMutex lock(mLock); 541 mVolume[LEFT] = left; 542 mVolume[RIGHT] = right; 543 544 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 545 546 if (isOffloaded()) { 547 mAudioTrack->signal(); 548 } 549 return NO_ERROR; 550} 551 552status_t AudioTrack::setVolume(float volume) 553{ 554 return setVolume(volume, volume); 555} 556 557status_t AudioTrack::setAuxEffectSendLevel(float level) 558{ 559 if (level < 0.0f || level > 1.0f) { 560 return BAD_VALUE; 561 } 562 563 AutoMutex lock(mLock); 564 mSendLevel = level; 565 mProxy->setSendLevel(level); 566 567 return NO_ERROR; 568} 569 570void AudioTrack::getAuxEffectSendLevel(float* level) const 571{ 572 if (level != NULL) { 573 *level = mSendLevel; 574 } 575} 576 577status_t AudioTrack::setSampleRate(uint32_t rate) 578{ 579 if (mIsTimed || isOffloaded()) { 580 return INVALID_OPERATION; 581 } 582 583 uint32_t afSamplingRate; 584 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 585 return NO_INIT; 586 } 587 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 588 if (rate == 0 || rate > afSamplingRate*2 ) { 589 return BAD_VALUE; 590 } 591 592 AutoMutex lock(mLock); 593 mSampleRate = rate; 594 mProxy->setSampleRate(rate); 595 596 return NO_ERROR; 597} 598 599uint32_t AudioTrack::getSampleRate() const 600{ 601 if (mIsTimed) { 602 return 0; 603 } 604 605 AutoMutex lock(mLock); 606 607 // sample rate can be updated during playback by the offloaded decoder so we need to 608 // query the HAL and update if needed. 609// FIXME use Proxy return channel to update the rate from server and avoid polling here 610 if (isOffloaded()) { 611 if (mOutput != 0) { 612 uint32_t sampleRate = 0; 613 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 614 if (status == NO_ERROR) { 615 mSampleRate = sampleRate; 616 } 617 } 618 } 619 return mSampleRate; 620} 621 622status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 623{ 624 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 625 return INVALID_OPERATION; 626 } 627 628 if (loopCount == 0) { 629 ; 630 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 631 loopEnd - loopStart >= MIN_LOOP) { 632 ; 633 } else { 634 return BAD_VALUE; 635 } 636 637 AutoMutex lock(mLock); 638 // See setPosition() regarding setting parameters such as loop points or position while active 639 if (mState == STATE_ACTIVE) { 640 return INVALID_OPERATION; 641 } 642 setLoop_l(loopStart, loopEnd, loopCount); 643 return NO_ERROR; 644} 645 646void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 647{ 648 // FIXME If setting a loop also sets position to start of loop, then 649 // this is correct. Otherwise it should be removed. 650 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 651 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 652 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 653} 654 655status_t AudioTrack::setMarkerPosition(uint32_t marker) 656{ 657 // The only purpose of setting marker position is to get a callback 658 if (mCbf == NULL || isOffloaded()) { 659 return INVALID_OPERATION; 660 } 661 662 AutoMutex lock(mLock); 663 mMarkerPosition = marker; 664 mMarkerReached = false; 665 666 return NO_ERROR; 667} 668 669status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 670{ 671 if (isOffloaded()) { 672 return INVALID_OPERATION; 673 } 674 if (marker == NULL) { 675 return BAD_VALUE; 676 } 677 678 AutoMutex lock(mLock); 679 *marker = mMarkerPosition; 680 681 return NO_ERROR; 682} 683 684status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 685{ 686 // The only purpose of setting position update period is to get a callback 687 if (mCbf == NULL || isOffloaded()) { 688 return INVALID_OPERATION; 689 } 690 691 AutoMutex lock(mLock); 692 mNewPosition = mProxy->getPosition() + updatePeriod; 693 mUpdatePeriod = updatePeriod; 694 return NO_ERROR; 695} 696 697status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 698{ 699 if (isOffloaded()) { 700 return INVALID_OPERATION; 701 } 702 if (updatePeriod == NULL) { 703 return BAD_VALUE; 704 } 705 706 AutoMutex lock(mLock); 707 *updatePeriod = mUpdatePeriod; 708 709 return NO_ERROR; 710} 711 712status_t AudioTrack::setPosition(uint32_t position) 713{ 714 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 715 return INVALID_OPERATION; 716 } 717 if (position > mFrameCount) { 718 return BAD_VALUE; 719 } 720 721 AutoMutex lock(mLock); 722 // Currently we require that the player is inactive before setting parameters such as position 723 // or loop points. Otherwise, there could be a race condition: the application could read the 724 // current position, compute a new position or loop parameters, and then set that position or 725 // loop parameters but it would do the "wrong" thing since the position has continued to advance 726 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 727 // to specify how it wants to handle such scenarios. 728 if (mState == STATE_ACTIVE) { 729 return INVALID_OPERATION; 730 } 731 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 732 mLoopPeriod = 0; 733 // FIXME Check whether loops and setting position are incompatible in old code. 734 // If we use setLoop for both purposes we lose the capability to set the position while looping. 735 mStaticProxy->setLoop(position, mFrameCount, 0); 736 737 return NO_ERROR; 738} 739 740status_t AudioTrack::getPosition(uint32_t *position) const 741{ 742 if (position == NULL) { 743 return BAD_VALUE; 744 } 745 746 AutoMutex lock(mLock); 747 if (isOffloaded()) { 748 uint32_t dspFrames = 0; 749 750 if (mOutput != 0) { 751 uint32_t halFrames; 752 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 753 } 754 *position = dspFrames; 755 } else { 756 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 757 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 758 mProxy->getPosition(); 759 } 760 return NO_ERROR; 761} 762 763status_t AudioTrack::getBufferPosition(size_t *position) 764{ 765 if (mSharedBuffer == 0 || mIsTimed) { 766 return INVALID_OPERATION; 767 } 768 if (position == NULL) { 769 return BAD_VALUE; 770 } 771 772 AutoMutex lock(mLock); 773 *position = mStaticProxy->getBufferPosition(); 774 return NO_ERROR; 775} 776 777status_t AudioTrack::reload() 778{ 779 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 780 return INVALID_OPERATION; 781 } 782 783 AutoMutex lock(mLock); 784 // See setPosition() regarding setting parameters such as loop points or position while active 785 if (mState == STATE_ACTIVE) { 786 return INVALID_OPERATION; 787 } 788 mNewPosition = mUpdatePeriod; 789 mLoopPeriod = 0; 790 // FIXME The new code cannot reload while keeping a loop specified. 791 // Need to check how the old code handled this, and whether it's a significant change. 792 mStaticProxy->setLoop(0, mFrameCount, 0); 793 return NO_ERROR; 794} 795 796audio_io_handle_t AudioTrack::getOutput() 797{ 798 AutoMutex lock(mLock); 799 return mOutput; 800} 801 802// must be called with mLock held 803audio_io_handle_t AudioTrack::getOutput_l() 804{ 805 if (mOutput) { 806 return mOutput; 807 } else { 808 return AudioSystem::getOutput(mStreamType, 809 mSampleRate, mFormat, mChannelMask, mFlags); 810 } 811} 812 813status_t AudioTrack::attachAuxEffect(int effectId) 814{ 815 AutoMutex lock(mLock); 816 status_t status = mAudioTrack->attachAuxEffect(effectId); 817 if (status == NO_ERROR) { 818 mAuxEffectId = effectId; 819 } 820 return status; 821} 822 823// ------------------------------------------------------------------------- 824 825// must be called with mLock held 826status_t AudioTrack::createTrack_l( 827 audio_stream_type_t streamType, 828 uint32_t sampleRate, 829 audio_format_t format, 830 size_t frameCount, 831 audio_output_flags_t flags, 832 const sp<IMemory>& sharedBuffer, 833 audio_io_handle_t output, 834 size_t epoch) 835{ 836 status_t status; 837 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 838 if (audioFlinger == 0) { 839 ALOGE("Could not get audioflinger"); 840 return NO_INIT; 841 } 842 843 // Not all of these values are needed under all conditions, but it is easier to get them all 844 845 uint32_t afLatency; 846 status = AudioSystem::getLatency(output, streamType, &afLatency); 847 if (status != NO_ERROR) { 848 ALOGE("getLatency(%d) failed status %d", output, status); 849 return NO_INIT; 850 } 851 852 size_t afFrameCount; 853 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 854 if (status != NO_ERROR) { 855 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 856 return NO_INIT; 857 } 858 859 uint32_t afSampleRate; 860 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 861 if (status != NO_ERROR) { 862 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 863 return NO_INIT; 864 } 865 866 // Client decides whether the track is TIMED (see below), but can only express a preference 867 // for FAST. Server will perform additional tests. 868 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 869 // either of these use cases: 870 // use case 1: shared buffer 871 (sharedBuffer != 0) || 872 // use case 2: callback handler 873 (mCbf != NULL))) { 874 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 875 // once denied, do not request again if IAudioTrack is re-created 876 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 877 mFlags = flags; 878 } 879 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 880 881 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && sampleRate != afSampleRate) { 882 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client due to mismatching sample rate (%d vs %d)", 883 sampleRate, afSampleRate); 884 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 885 } 886 887 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 888 // n = 1 fast track with single buffering; nBuffering is ignored 889 // n = 2 fast track with double buffering 890 // n = 2 normal track, no sample rate conversion 891 // n = 3 normal track, with sample rate conversion 892 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 893 // n > 3 very high latency or very small notification interval; nBuffering is ignored 894 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 895 896 mNotificationFramesAct = mNotificationFramesReq; 897 898 if (!audio_is_linear_pcm(format)) { 899 900 if (sharedBuffer != 0) { 901 // Same comment as below about ignoring frameCount parameter for set() 902 frameCount = sharedBuffer->size(); 903 } else if (frameCount == 0) { 904 frameCount = afFrameCount; 905 } 906 if (mNotificationFramesAct != frameCount) { 907 mNotificationFramesAct = frameCount; 908 } 909 } else if (sharedBuffer != 0) { 910 911 // Ensure that buffer alignment matches channel count 912 // 8-bit data in shared memory is not currently supported by AudioFlinger 913 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 914 if (mChannelCount > 1) { 915 // More than 2 channels does not require stronger alignment than stereo 916 alignment <<= 1; 917 } 918 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 919 ALOGE("Invalid buffer alignment: address %p, channel count %u", 920 sharedBuffer->pointer(), mChannelCount); 921 return BAD_VALUE; 922 } 923 924 // When initializing a shared buffer AudioTrack via constructors, 925 // there's no frameCount parameter. 926 // But when initializing a shared buffer AudioTrack via set(), 927 // there _is_ a frameCount parameter. We silently ignore it. 928 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 929 930 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 931 932 // FIXME move these calculations and associated checks to server 933 934 // Ensure that buffer depth covers at least audio hardware latency 935 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 936 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 937 afFrameCount, minBufCount, afSampleRate, afLatency); 938 if (minBufCount <= nBuffering) { 939 minBufCount = nBuffering; 940 } 941 942 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 943 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 944 ", afLatency=%d", 945 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 946 947 if (frameCount == 0) { 948 frameCount = minFrameCount; 949 } else if (frameCount < minFrameCount) { 950 // not ALOGW because it happens all the time when playing key clicks over A2DP 951 ALOGV("Minimum buffer size corrected from %d to %d", 952 frameCount, minFrameCount); 953 frameCount = minFrameCount; 954 } 955 // Make sure that application is notified with sufficient margin before underrun 956 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 957 mNotificationFramesAct = frameCount/nBuffering; 958 } 959 960 } else { 961 // For fast tracks, the frame count calculations and checks are done by server 962 } 963 964 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 965 if (mIsTimed) { 966 trackFlags |= IAudioFlinger::TRACK_TIMED; 967 } 968 969 pid_t tid = -1; 970 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 971 trackFlags |= IAudioFlinger::TRACK_FAST; 972 if (mAudioTrackThread != 0) { 973 tid = mAudioTrackThread->getTid(); 974 } 975 } 976 977 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 978 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 979 } 980 981 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 982 sampleRate, 983 // AudioFlinger only sees 16-bit PCM 984 format == AUDIO_FORMAT_PCM_8_BIT ? 985 AUDIO_FORMAT_PCM_16_BIT : format, 986 mChannelMask, 987 frameCount, 988 &trackFlags, 989 sharedBuffer, 990 output, 991 tid, 992 &mSessionId, 993 mName, 994 mClientUid, 995 &status); 996 997 if (track == 0) { 998 ALOGE("AudioFlinger could not create track, status: %d", status); 999 return status; 1000 } 1001 sp<IMemory> iMem = track->getCblk(); 1002 if (iMem == 0) { 1003 ALOGE("Could not get control block"); 1004 return NO_INIT; 1005 } 1006 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1007 if (mAudioTrack != 0) { 1008 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1009 mDeathNotifier.clear(); 1010 } 1011 mAudioTrack = track; 1012 mCblkMemory = iMem; 1013 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1014 mCblk = cblk; 1015 size_t temp = cblk->frameCount_; 1016 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1017 // In current design, AudioTrack client checks and ensures frame count validity before 1018 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1019 // for fast track as it uses a special method of assigning frame count. 1020 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1021 } 1022 frameCount = temp; 1023 mAwaitBoost = false; 1024 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1025 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1026 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1027 mAwaitBoost = true; 1028 if (sharedBuffer == 0) { 1029 // Theoretically double-buffering is not required for fast tracks, 1030 // due to tighter scheduling. But in practice, to accommodate kernels with 1031 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1032 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1033 mNotificationFramesAct = frameCount/nBuffering; 1034 } 1035 } 1036 } else { 1037 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1038 // once denied, do not request again if IAudioTrack is re-created 1039 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1040 mFlags = flags; 1041 if (sharedBuffer == 0) { 1042 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1043 mNotificationFramesAct = frameCount/nBuffering; 1044 } 1045 } 1046 } 1047 } 1048 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1049 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1050 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1051 } else { 1052 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1053 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1054 mFlags = flags; 1055 return NO_INIT; 1056 } 1057 } 1058 1059 mRefreshRemaining = true; 1060 1061 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1062 // is the value of pointer() for the shared buffer, otherwise buffers points 1063 // immediately after the control block. This address is for the mapping within client 1064 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1065 void* buffers; 1066 if (sharedBuffer == 0) { 1067 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1068 } else { 1069 buffers = sharedBuffer->pointer(); 1070 } 1071 1072 mAudioTrack->attachAuxEffect(mAuxEffectId); 1073 // FIXME don't believe this lie 1074 mLatency = afLatency + (1000*frameCount) / sampleRate; 1075 mFrameCount = frameCount; 1076 // If IAudioTrack is re-created, don't let the requested frameCount 1077 // decrease. This can confuse clients that cache frameCount(). 1078 if (frameCount > mReqFrameCount) { 1079 mReqFrameCount = frameCount; 1080 } 1081 1082 // update proxy 1083 if (sharedBuffer == 0) { 1084 mStaticProxy.clear(); 1085 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1086 } else { 1087 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1088 mProxy = mStaticProxy; 1089 } 1090 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1091 uint16_t(mVolume[LEFT] * 0x1000)); 1092 mProxy->setSendLevel(mSendLevel); 1093 mProxy->setSampleRate(mSampleRate); 1094 mProxy->setEpoch(epoch); 1095 mProxy->setMinimum(mNotificationFramesAct); 1096 1097 mDeathNotifier = new DeathNotifier(this); 1098 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1099 1100 return NO_ERROR; 1101} 1102 1103status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1104{ 1105 if (audioBuffer == NULL) { 1106 return BAD_VALUE; 1107 } 1108 if (mTransfer != TRANSFER_OBTAIN) { 1109 audioBuffer->frameCount = 0; 1110 audioBuffer->size = 0; 1111 audioBuffer->raw = NULL; 1112 return INVALID_OPERATION; 1113 } 1114 1115 const struct timespec *requested; 1116 if (waitCount == -1) { 1117 requested = &ClientProxy::kForever; 1118 } else if (waitCount == 0) { 1119 requested = &ClientProxy::kNonBlocking; 1120 } else if (waitCount > 0) { 1121 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1122 struct timespec timeout; 1123 timeout.tv_sec = ms / 1000; 1124 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1125 requested = &timeout; 1126 } else { 1127 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1128 requested = NULL; 1129 } 1130 return obtainBuffer(audioBuffer, requested); 1131} 1132 1133status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1134 struct timespec *elapsed, size_t *nonContig) 1135{ 1136 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1137 uint32_t oldSequence = 0; 1138 uint32_t newSequence; 1139 1140 Proxy::Buffer buffer; 1141 status_t status = NO_ERROR; 1142 1143 static const int32_t kMaxTries = 5; 1144 int32_t tryCounter = kMaxTries; 1145 1146 do { 1147 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1148 // keep them from going away if another thread re-creates the track during obtainBuffer() 1149 sp<AudioTrackClientProxy> proxy; 1150 sp<IMemory> iMem; 1151 1152 { // start of lock scope 1153 AutoMutex lock(mLock); 1154 1155 newSequence = mSequence; 1156 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1157 if (status == DEAD_OBJECT) { 1158 // re-create track, unless someone else has already done so 1159 if (newSequence == oldSequence) { 1160 status = restoreTrack_l("obtainBuffer"); 1161 if (status != NO_ERROR) { 1162 buffer.mFrameCount = 0; 1163 buffer.mRaw = NULL; 1164 buffer.mNonContig = 0; 1165 break; 1166 } 1167 } 1168 } 1169 oldSequence = newSequence; 1170 1171 // Keep the extra references 1172 proxy = mProxy; 1173 iMem = mCblkMemory; 1174 1175 if (mState == STATE_STOPPING) { 1176 status = -EINTR; 1177 buffer.mFrameCount = 0; 1178 buffer.mRaw = NULL; 1179 buffer.mNonContig = 0; 1180 break; 1181 } 1182 1183 // Non-blocking if track is stopped or paused 1184 if (mState != STATE_ACTIVE) { 1185 requested = &ClientProxy::kNonBlocking; 1186 } 1187 1188 } // end of lock scope 1189 1190 buffer.mFrameCount = audioBuffer->frameCount; 1191 // FIXME starts the requested timeout and elapsed over from scratch 1192 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1193 1194 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1195 1196 audioBuffer->frameCount = buffer.mFrameCount; 1197 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1198 audioBuffer->raw = buffer.mRaw; 1199 if (nonContig != NULL) { 1200 *nonContig = buffer.mNonContig; 1201 } 1202 return status; 1203} 1204 1205void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1206{ 1207 if (mTransfer == TRANSFER_SHARED) { 1208 return; 1209 } 1210 1211 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1212 if (stepCount == 0) { 1213 return; 1214 } 1215 1216 Proxy::Buffer buffer; 1217 buffer.mFrameCount = stepCount; 1218 buffer.mRaw = audioBuffer->raw; 1219 1220 AutoMutex lock(mLock); 1221 mInUnderrun = false; 1222 mProxy->releaseBuffer(&buffer); 1223 1224 // restart track if it was disabled by audioflinger due to previous underrun 1225 if (mState == STATE_ACTIVE) { 1226 audio_track_cblk_t* cblk = mCblk; 1227 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1228 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1229 this, mName.string()); 1230 // FIXME ignoring status 1231 mAudioTrack->start(); 1232 } 1233 } 1234} 1235 1236// ------------------------------------------------------------------------- 1237 1238ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1239{ 1240 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1241 return INVALID_OPERATION; 1242 } 1243 1244 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1245 // Sanity-check: user is most-likely passing an error code, and it would 1246 // make the return value ambiguous (actualSize vs error). 1247 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1248 return BAD_VALUE; 1249 } 1250 1251 size_t written = 0; 1252 Buffer audioBuffer; 1253 1254 while (userSize >= mFrameSize) { 1255 audioBuffer.frameCount = userSize / mFrameSize; 1256 1257 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1258 if (err < 0) { 1259 if (written > 0) { 1260 break; 1261 } 1262 return ssize_t(err); 1263 } 1264 1265 size_t toWrite; 1266 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1267 // Divide capacity by 2 to take expansion into account 1268 toWrite = audioBuffer.size >> 1; 1269 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1270 } else { 1271 toWrite = audioBuffer.size; 1272 memcpy(audioBuffer.i8, buffer, toWrite); 1273 } 1274 buffer = ((const char *) buffer) + toWrite; 1275 userSize -= toWrite; 1276 written += toWrite; 1277 1278 releaseBuffer(&audioBuffer); 1279 } 1280 1281 return written; 1282} 1283 1284// ------------------------------------------------------------------------- 1285 1286TimedAudioTrack::TimedAudioTrack() { 1287 mIsTimed = true; 1288} 1289 1290status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1291{ 1292 AutoMutex lock(mLock); 1293 status_t result = UNKNOWN_ERROR; 1294 1295#if 1 1296 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1297 // while we are accessing the cblk 1298 sp<IAudioTrack> audioTrack = mAudioTrack; 1299 sp<IMemory> iMem = mCblkMemory; 1300#endif 1301 1302 // If the track is not invalid already, try to allocate a buffer. alloc 1303 // fails indicating that the server is dead, flag the track as invalid so 1304 // we can attempt to restore in just a bit. 1305 audio_track_cblk_t* cblk = mCblk; 1306 if (!(cblk->mFlags & CBLK_INVALID)) { 1307 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1308 if (result == DEAD_OBJECT) { 1309 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1310 } 1311 } 1312 1313 // If the track is invalid at this point, attempt to restore it. and try the 1314 // allocation one more time. 1315 if (cblk->mFlags & CBLK_INVALID) { 1316 result = restoreTrack_l("allocateTimedBuffer"); 1317 1318 if (result == NO_ERROR) { 1319 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1320 } 1321 } 1322 1323 return result; 1324} 1325 1326status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1327 int64_t pts) 1328{ 1329 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1330 { 1331 AutoMutex lock(mLock); 1332 audio_track_cblk_t* cblk = mCblk; 1333 // restart track if it was disabled by audioflinger due to previous underrun 1334 if (buffer->size() != 0 && status == NO_ERROR && 1335 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1336 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1337 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1338 // FIXME ignoring status 1339 mAudioTrack->start(); 1340 } 1341 } 1342 return status; 1343} 1344 1345status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1346 TargetTimeline target) 1347{ 1348 return mAudioTrack->setMediaTimeTransform(xform, target); 1349} 1350 1351// ------------------------------------------------------------------------- 1352 1353nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1354{ 1355 // Currently the AudioTrack thread is not created if there are no callbacks. 1356 // Would it ever make sense to run the thread, even without callbacks? 1357 // If so, then replace this by checks at each use for mCbf != NULL. 1358 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1359 1360 mLock.lock(); 1361 if (mAwaitBoost) { 1362 mAwaitBoost = false; 1363 mLock.unlock(); 1364 static const int32_t kMaxTries = 5; 1365 int32_t tryCounter = kMaxTries; 1366 uint32_t pollUs = 10000; 1367 do { 1368 int policy = sched_getscheduler(0); 1369 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1370 break; 1371 } 1372 usleep(pollUs); 1373 pollUs <<= 1; 1374 } while (tryCounter-- > 0); 1375 if (tryCounter < 0) { 1376 ALOGE("did not receive expected priority boost on time"); 1377 } 1378 // Run again immediately 1379 return 0; 1380 } 1381 1382 // Can only reference mCblk while locked 1383 int32_t flags = android_atomic_and( 1384 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1385 1386 // Check for track invalidation 1387 if (flags & CBLK_INVALID) { 1388 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1389 // AudioSystem cache. We should not exit here but after calling the callback so 1390 // that the upper layers can recreate the track 1391 if (!isOffloaded() || (mSequence == mObservedSequence)) { 1392 status_t status = restoreTrack_l("processAudioBuffer"); 1393 mLock.unlock(); 1394 // Run again immediately, but with a new IAudioTrack 1395 return 0; 1396 } 1397 } 1398 1399 bool waitStreamEnd = mState == STATE_STOPPING; 1400 bool active = mState == STATE_ACTIVE; 1401 1402 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1403 bool newUnderrun = false; 1404 if (flags & CBLK_UNDERRUN) { 1405#if 0 1406 // Currently in shared buffer mode, when the server reaches the end of buffer, 1407 // the track stays active in continuous underrun state. It's up to the application 1408 // to pause or stop the track, or set the position to a new offset within buffer. 1409 // This was some experimental code to auto-pause on underrun. Keeping it here 1410 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1411 if (mTransfer == TRANSFER_SHARED) { 1412 mState = STATE_PAUSED; 1413 active = false; 1414 } 1415#endif 1416 if (!mInUnderrun) { 1417 mInUnderrun = true; 1418 newUnderrun = true; 1419 } 1420 } 1421 1422 // Get current position of server 1423 size_t position = mProxy->getPosition(); 1424 1425 // Manage marker callback 1426 bool markerReached = false; 1427 size_t markerPosition = mMarkerPosition; 1428 // FIXME fails for wraparound, need 64 bits 1429 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1430 mMarkerReached = markerReached = true; 1431 } 1432 1433 // Determine number of new position callback(s) that will be needed, while locked 1434 size_t newPosCount = 0; 1435 size_t newPosition = mNewPosition; 1436 size_t updatePeriod = mUpdatePeriod; 1437 // FIXME fails for wraparound, need 64 bits 1438 if (updatePeriod > 0 && position >= newPosition) { 1439 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1440 mNewPosition += updatePeriod * newPosCount; 1441 } 1442 1443 // Cache other fields that will be needed soon 1444 uint32_t loopPeriod = mLoopPeriod; 1445 uint32_t sampleRate = mSampleRate; 1446 size_t notificationFrames = mNotificationFramesAct; 1447 if (mRefreshRemaining) { 1448 mRefreshRemaining = false; 1449 mRemainingFrames = notificationFrames; 1450 mRetryOnPartialBuffer = false; 1451 } 1452 size_t misalignment = mProxy->getMisalignment(); 1453 uint32_t sequence = mSequence; 1454 1455 // These fields don't need to be cached, because they are assigned only by set(): 1456 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1457 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1458 1459 mLock.unlock(); 1460 1461 if (waitStreamEnd) { 1462 AutoMutex lock(mLock); 1463 1464 sp<AudioTrackClientProxy> proxy = mProxy; 1465 sp<IMemory> iMem = mCblkMemory; 1466 1467 struct timespec timeout; 1468 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1469 timeout.tv_nsec = 0; 1470 1471 mLock.unlock(); 1472 status_t status = mProxy->waitStreamEndDone(&timeout); 1473 mLock.lock(); 1474 switch (status) { 1475 case NO_ERROR: 1476 case DEAD_OBJECT: 1477 case TIMED_OUT: 1478 mLock.unlock(); 1479 mCbf(EVENT_STREAM_END, mUserData, NULL); 1480 mLock.lock(); 1481 if (mState == STATE_STOPPING) { 1482 mState = STATE_STOPPED; 1483 if (status != DEAD_OBJECT) { 1484 return NS_INACTIVE; 1485 } 1486 } 1487 return 0; 1488 default: 1489 return 0; 1490 } 1491 } 1492 1493 // perform callbacks while unlocked 1494 if (newUnderrun) { 1495 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1496 } 1497 // FIXME we will miss loops if loop cycle was signaled several times since last call 1498 // to processAudioBuffer() 1499 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1500 mCbf(EVENT_LOOP_END, mUserData, NULL); 1501 } 1502 if (flags & CBLK_BUFFER_END) { 1503 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1504 } 1505 if (markerReached) { 1506 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1507 } 1508 while (newPosCount > 0) { 1509 size_t temp = newPosition; 1510 mCbf(EVENT_NEW_POS, mUserData, &temp); 1511 newPosition += updatePeriod; 1512 newPosCount--; 1513 } 1514 1515 if (mObservedSequence != sequence) { 1516 mObservedSequence = sequence; 1517 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1518 // for offloaded tracks, just wait for the upper layers to recreate the track 1519 if (isOffloaded()) { 1520 return NS_INACTIVE; 1521 } 1522 } 1523 1524 // if inactive, then don't run me again until re-started 1525 if (!active) { 1526 return NS_INACTIVE; 1527 } 1528 1529 // Compute the estimated time until the next timed event (position, markers, loops) 1530 // FIXME only for non-compressed audio 1531 uint32_t minFrames = ~0; 1532 if (!markerReached && position < markerPosition) { 1533 minFrames = markerPosition - position; 1534 } 1535 if (loopPeriod > 0 && loopPeriod < minFrames) { 1536 minFrames = loopPeriod; 1537 } 1538 if (updatePeriod > 0 && updatePeriod < minFrames) { 1539 minFrames = updatePeriod; 1540 } 1541 1542 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1543 static const uint32_t kPoll = 0; 1544 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1545 minFrames = kPoll * notificationFrames; 1546 } 1547 1548 // Convert frame units to time units 1549 nsecs_t ns = NS_WHENEVER; 1550 if (minFrames != (uint32_t) ~0) { 1551 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1552 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1553 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1554 } 1555 1556 // If not supplying data by EVENT_MORE_DATA, then we're done 1557 if (mTransfer != TRANSFER_CALLBACK) { 1558 return ns; 1559 } 1560 1561 struct timespec timeout; 1562 const struct timespec *requested = &ClientProxy::kForever; 1563 if (ns != NS_WHENEVER) { 1564 timeout.tv_sec = ns / 1000000000LL; 1565 timeout.tv_nsec = ns % 1000000000LL; 1566 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1567 requested = &timeout; 1568 } 1569 1570 while (mRemainingFrames > 0) { 1571 1572 Buffer audioBuffer; 1573 audioBuffer.frameCount = mRemainingFrames; 1574 size_t nonContig; 1575 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1576 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1577 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1578 requested = &ClientProxy::kNonBlocking; 1579 size_t avail = audioBuffer.frameCount + nonContig; 1580 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1581 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1582 if (err != NO_ERROR) { 1583 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1584 (isOffloaded() && (err == DEAD_OBJECT))) { 1585 return 0; 1586 } 1587 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1588 return NS_NEVER; 1589 } 1590 1591 if (mRetryOnPartialBuffer && !isOffloaded()) { 1592 mRetryOnPartialBuffer = false; 1593 if (avail < mRemainingFrames) { 1594 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1595 if (ns < 0 || myns < ns) { 1596 ns = myns; 1597 } 1598 return ns; 1599 } 1600 } 1601 1602 // Divide buffer size by 2 to take into account the expansion 1603 // due to 8 to 16 bit conversion: the callback must fill only half 1604 // of the destination buffer 1605 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1606 audioBuffer.size >>= 1; 1607 } 1608 1609 size_t reqSize = audioBuffer.size; 1610 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1611 size_t writtenSize = audioBuffer.size; 1612 size_t writtenFrames = writtenSize / mFrameSize; 1613 1614 // Sanity check on returned size 1615 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1616 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1617 reqSize, (int) writtenSize); 1618 return NS_NEVER; 1619 } 1620 1621 if (writtenSize == 0) { 1622 // The callback is done filling buffers 1623 // Keep this thread going to handle timed events and 1624 // still try to get more data in intervals of WAIT_PERIOD_MS 1625 // but don't just loop and block the CPU, so wait 1626 return WAIT_PERIOD_MS * 1000000LL; 1627 } 1628 1629 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1630 // 8 to 16 bit conversion, note that source and destination are the same address 1631 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1632 audioBuffer.size <<= 1; 1633 } 1634 1635 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1636 audioBuffer.frameCount = releasedFrames; 1637 mRemainingFrames -= releasedFrames; 1638 if (misalignment >= releasedFrames) { 1639 misalignment -= releasedFrames; 1640 } else { 1641 misalignment = 0; 1642 } 1643 1644 releaseBuffer(&audioBuffer); 1645 1646 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1647 // if callback doesn't like to accept the full chunk 1648 if (writtenSize < reqSize) { 1649 continue; 1650 } 1651 1652 // There could be enough non-contiguous frames available to satisfy the remaining request 1653 if (mRemainingFrames <= nonContig) { 1654 continue; 1655 } 1656 1657#if 0 1658 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1659 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1660 // that total to a sum == notificationFrames. 1661 if (0 < misalignment && misalignment <= mRemainingFrames) { 1662 mRemainingFrames = misalignment; 1663 return (mRemainingFrames * 1100000000LL) / sampleRate; 1664 } 1665#endif 1666 1667 } 1668 mRemainingFrames = notificationFrames; 1669 mRetryOnPartialBuffer = true; 1670 1671 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1672 return 0; 1673} 1674 1675status_t AudioTrack::restoreTrack_l(const char *from) 1676{ 1677 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1678 isOffloaded() ? "Offloaded" : "PCM", from); 1679 ++mSequence; 1680 status_t result; 1681 1682 // refresh the audio configuration cache in this process to make sure we get new 1683 // output parameters in getOutput_l() and createTrack_l() 1684 AudioSystem::clearAudioConfigCache(); 1685 1686 if (isOffloaded()) { 1687 return DEAD_OBJECT; 1688 } 1689 1690 // force new output query from audio policy manager; 1691 mOutput = 0; 1692 audio_io_handle_t output = getOutput_l(); 1693 1694 // if the new IAudioTrack is created, createTrack_l() will modify the 1695 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1696 // It will also delete the strong references on previous IAudioTrack and IMemory 1697 1698 // take the frames that will be lost by track recreation into account in saved position 1699 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1700 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1701 result = createTrack_l(mStreamType, 1702 mSampleRate, 1703 mFormat, 1704 mReqFrameCount, // so that frame count never goes down 1705 mFlags, 1706 mSharedBuffer, 1707 output, 1708 position /*epoch*/); 1709 1710 if (result == NO_ERROR) { 1711 // continue playback from last known position, but 1712 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1713 if (mStaticProxy != NULL) { 1714 mLoopPeriod = 0; 1715 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1716 } 1717 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1718 // track destruction have been played? This is critical for SoundPool implementation 1719 // This must be broken, and needs to be tested/debugged. 1720#if 0 1721 // restore write index and set other indexes to reflect empty buffer status 1722 if (!strcmp(from, "start")) { 1723 // Make sure that a client relying on callback events indicating underrun or 1724 // the actual amount of audio frames played (e.g SoundPool) receives them. 1725 if (mSharedBuffer == 0) { 1726 // restart playback even if buffer is not completely filled. 1727 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1728 } 1729 } 1730#endif 1731 if (mState == STATE_ACTIVE) { 1732 result = mAudioTrack->start(); 1733 } 1734 } 1735 if (result != NO_ERROR) { 1736 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1737 // As getOutput was called above and resulted in an output stream to be opened, 1738 // we need to release it. 1739 AudioSystem::releaseOutput(output); 1740 ALOGW("restoreTrack_l() failed status %d", result); 1741 mState = STATE_STOPPED; 1742 } 1743 1744 return result; 1745} 1746 1747status_t AudioTrack::setParameters(const String8& keyValuePairs) 1748{ 1749 AutoMutex lock(mLock); 1750 return mAudioTrack->setParameters(keyValuePairs); 1751} 1752 1753status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1754{ 1755 AutoMutex lock(mLock); 1756 // FIXME not implemented for fast tracks; should use proxy and SSQ 1757 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1758 return INVALID_OPERATION; 1759 } 1760 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1761 return INVALID_OPERATION; 1762 } 1763 status_t status = mAudioTrack->getTimestamp(timestamp); 1764 if (status == NO_ERROR) { 1765 timestamp.mPosition += mProxy->getEpoch(); 1766 } 1767 return status; 1768} 1769 1770String8 AudioTrack::getParameters(const String8& keys) 1771{ 1772 if (mOutput) { 1773 return AudioSystem::getParameters(mOutput, keys); 1774 } else { 1775 return String8::empty(); 1776 } 1777} 1778 1779status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1780{ 1781 1782 const size_t SIZE = 256; 1783 char buffer[SIZE]; 1784 String8 result; 1785 1786 result.append(" AudioTrack::dump\n"); 1787 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1788 mVolume[0], mVolume[1]); 1789 result.append(buffer); 1790 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1791 mChannelCount, mFrameCount); 1792 result.append(buffer); 1793 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1794 result.append(buffer); 1795 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1796 result.append(buffer); 1797 ::write(fd, result.string(), result.size()); 1798 return NO_ERROR; 1799} 1800 1801uint32_t AudioTrack::getUnderrunFrames() const 1802{ 1803 AutoMutex lock(mLock); 1804 return mProxy->getUnderrunFrames(); 1805} 1806 1807// ========================================================================= 1808 1809void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1810{ 1811 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1812 if (audioTrack != 0) { 1813 AutoMutex lock(audioTrack->mLock); 1814 audioTrack->mProxy->binderDied(); 1815 } 1816} 1817 1818// ========================================================================= 1819 1820AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1821 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1822 mIgnoreNextPausedInt(false) 1823{ 1824} 1825 1826AudioTrack::AudioTrackThread::~AudioTrackThread() 1827{ 1828} 1829 1830bool AudioTrack::AudioTrackThread::threadLoop() 1831{ 1832 { 1833 AutoMutex _l(mMyLock); 1834 if (mPaused) { 1835 mMyCond.wait(mMyLock); 1836 // caller will check for exitPending() 1837 return true; 1838 } 1839 if (mIgnoreNextPausedInt) { 1840 mIgnoreNextPausedInt = false; 1841 mPausedInt = false; 1842 } 1843 if (mPausedInt) { 1844 if (mPausedNs > 0) { 1845 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1846 } else { 1847 mMyCond.wait(mMyLock); 1848 } 1849 mPausedInt = false; 1850 return true; 1851 } 1852 } 1853 nsecs_t ns = mReceiver.processAudioBuffer(this); 1854 switch (ns) { 1855 case 0: 1856 return true; 1857 case NS_INACTIVE: 1858 pauseInternal(); 1859 return true; 1860 case NS_NEVER: 1861 return false; 1862 case NS_WHENEVER: 1863 // FIXME increase poll interval, or make event-driven 1864 ns = 1000000000LL; 1865 // fall through 1866 default: 1867 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1868 pauseInternal(ns); 1869 return true; 1870 } 1871} 1872 1873void AudioTrack::AudioTrackThread::requestExit() 1874{ 1875 // must be in this order to avoid a race condition 1876 Thread::requestExit(); 1877 resume(); 1878} 1879 1880void AudioTrack::AudioTrackThread::pause() 1881{ 1882 AutoMutex _l(mMyLock); 1883 mPaused = true; 1884} 1885 1886void AudioTrack::AudioTrackThread::resume() 1887{ 1888 AutoMutex _l(mMyLock); 1889 mIgnoreNextPausedInt = true; 1890 if (mPaused || mPausedInt) { 1891 mPaused = false; 1892 mPausedInt = false; 1893 mMyCond.signal(); 1894 } 1895} 1896 1897void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1898{ 1899 AutoMutex _l(mMyLock); 1900 mPausedInt = true; 1901 mPausedNs = ns; 1902} 1903 1904}; // namespace android 1905