AudioTrack.cpp revision 2b2165c75790050810460c8de3f414876bce4c0e
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31#define WAIT_STREAM_END_TIMEOUT_SEC 120 32 33 34namespace android { 35// --------------------------------------------------------------------------- 36 37// static 38status_t AudioTrack::getMinFrameCount( 39 size_t* frameCount, 40 audio_stream_type_t streamType, 41 uint32_t sampleRate) 42{ 43 if (frameCount == NULL) { 44 return BAD_VALUE; 45 } 46 47 // FIXME merge with similar code in createTrack_l(), except we're missing 48 // some information here that is available in createTrack_l(): 49 // audio_io_handle_t output 50 // audio_format_t format 51 // audio_channel_mask_t channelMask 52 // audio_output_flags_t flags 53 uint32_t afSampleRate; 54 status_t status; 55 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 56 if (status != NO_ERROR) { 57 return status; 58 } 59 size_t afFrameCount; 60 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 61 if (status != NO_ERROR) { 62 return status; 63 } 64 uint32_t afLatency; 65 status = AudioSystem::getOutputLatency(&afLatency, streamType); 66 if (status != NO_ERROR) { 67 return status; 68 } 69 70 // Ensure that buffer depth covers at least audio hardware latency 71 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 72 if (minBufCount < 2) { 73 minBufCount = 2; 74 } 75 76 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 77 afFrameCount * minBufCount * sampleRate / afSampleRate; 78 // The formula above should always produce a non-zero value, but return an error 79 // in the unlikely event that it does not, as that's part of the API contract. 80 if (*frameCount == 0) { 81 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 82 streamType, sampleRate); 83 return BAD_VALUE; 84 } 85 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 86 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 87 return NO_ERROR; 88} 89 90// --------------------------------------------------------------------------- 91 92AudioTrack::AudioTrack() 93 : mStatus(NO_INIT), 94 mIsTimed(false), 95 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 96 mPreviousSchedulingGroup(SP_DEFAULT) 97{ 98} 99 100AudioTrack::AudioTrack( 101 audio_stream_type_t streamType, 102 uint32_t sampleRate, 103 audio_format_t format, 104 audio_channel_mask_t channelMask, 105 int frameCount, 106 audio_output_flags_t flags, 107 callback_t cbf, 108 void* user, 109 int notificationFrames, 110 int sessionId, 111 transfer_type transferType, 112 const audio_offload_info_t *offloadInfo, 113 int uid) 114 : mStatus(NO_INIT), 115 mIsTimed(false), 116 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 117 mPreviousSchedulingGroup(SP_DEFAULT) 118{ 119 mStatus = set(streamType, sampleRate, format, channelMask, 120 frameCount, flags, cbf, user, notificationFrames, 121 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 122 offloadInfo, uid); 123} 124 125AudioTrack::AudioTrack( 126 audio_stream_type_t streamType, 127 uint32_t sampleRate, 128 audio_format_t format, 129 audio_channel_mask_t channelMask, 130 const sp<IMemory>& sharedBuffer, 131 audio_output_flags_t flags, 132 callback_t cbf, 133 void* user, 134 int notificationFrames, 135 int sessionId, 136 transfer_type transferType, 137 const audio_offload_info_t *offloadInfo, 138 int uid) 139 : mStatus(NO_INIT), 140 mIsTimed(false), 141 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 142 mPreviousSchedulingGroup(SP_DEFAULT) 143{ 144 mStatus = set(streamType, sampleRate, format, channelMask, 145 0 /*frameCount*/, flags, cbf, user, notificationFrames, 146 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); 147} 148 149AudioTrack::~AudioTrack() 150{ 151 if (mStatus == NO_ERROR) { 152 // Make sure that callback function exits in the case where 153 // it is looping on buffer full condition in obtainBuffer(). 154 // Otherwise the callback thread will never exit. 155 stop(); 156 if (mAudioTrackThread != 0) { 157 mProxy->interrupt(); 158 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 159 mAudioTrackThread->requestExitAndWait(); 160 mAudioTrackThread.clear(); 161 } 162 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 163 mAudioTrack.clear(); 164 IPCThreadState::self()->flushCommands(); 165 AudioSystem::releaseAudioSessionId(mSessionId); 166 } 167} 168 169status_t AudioTrack::set( 170 audio_stream_type_t streamType, 171 uint32_t sampleRate, 172 audio_format_t format, 173 audio_channel_mask_t channelMask, 174 int frameCountInt, 175 audio_output_flags_t flags, 176 callback_t cbf, 177 void* user, 178 int notificationFrames, 179 const sp<IMemory>& sharedBuffer, 180 bool threadCanCallJava, 181 int sessionId, 182 transfer_type transferType, 183 const audio_offload_info_t *offloadInfo, 184 int uid) 185{ 186 switch (transferType) { 187 case TRANSFER_DEFAULT: 188 if (sharedBuffer != 0) { 189 transferType = TRANSFER_SHARED; 190 } else if (cbf == NULL || threadCanCallJava) { 191 transferType = TRANSFER_SYNC; 192 } else { 193 transferType = TRANSFER_CALLBACK; 194 } 195 break; 196 case TRANSFER_CALLBACK: 197 if (cbf == NULL || sharedBuffer != 0) { 198 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 199 return BAD_VALUE; 200 } 201 break; 202 case TRANSFER_OBTAIN: 203 case TRANSFER_SYNC: 204 if (sharedBuffer != 0) { 205 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 206 return BAD_VALUE; 207 } 208 break; 209 case TRANSFER_SHARED: 210 if (sharedBuffer == 0) { 211 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 212 return BAD_VALUE; 213 } 214 break; 215 default: 216 ALOGE("Invalid transfer type %d", transferType); 217 return BAD_VALUE; 218 } 219 mTransfer = transferType; 220 221 // FIXME "int" here is legacy and will be replaced by size_t later 222 if (frameCountInt < 0) { 223 ALOGE("Invalid frame count %d", frameCountInt); 224 return BAD_VALUE; 225 } 226 size_t frameCount = frameCountInt; 227 228 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 229 sharedBuffer->size()); 230 231 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 232 233 AutoMutex lock(mLock); 234 235 // invariant that mAudioTrack != 0 is true only after set() returns successfully 236 if (mAudioTrack != 0) { 237 ALOGE("Track already in use"); 238 return INVALID_OPERATION; 239 } 240 241 mOutput = 0; 242 243 // handle default values first. 244 if (streamType == AUDIO_STREAM_DEFAULT) { 245 streamType = AUDIO_STREAM_MUSIC; 246 } 247 248 status_t status; 249 if (sampleRate == 0) { 250 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 251 if (status != NO_ERROR) { 252 ALOGE("Could not get output sample rate for stream type %d; status %d", 253 streamType, status); 254 return status; 255 } 256 } 257 mSampleRate = sampleRate; 258 259 // these below should probably come from the audioFlinger too... 260 if (format == AUDIO_FORMAT_DEFAULT) { 261 format = AUDIO_FORMAT_PCM_16_BIT; 262 } 263 264 // validate parameters 265 if (!audio_is_valid_format(format)) { 266 ALOGE("Invalid format %d", format); 267 return BAD_VALUE; 268 } 269 270 if (!audio_is_output_channel(channelMask)) { 271 ALOGE("Invalid channel mask %#x", channelMask); 272 return BAD_VALUE; 273 } 274 275 // AudioFlinger does not currently support 8-bit data in shared memory 276 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 277 ALOGE("8-bit data in shared memory is not supported"); 278 return BAD_VALUE; 279 } 280 281 // force direct flag if format is not linear PCM 282 // or offload was requested 283 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 284 || !audio_is_linear_pcm(format)) { 285 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 286 ? "Offload request, forcing to Direct Output" 287 : "Not linear PCM, forcing to Direct Output"); 288 flags = (audio_output_flags_t) 289 // FIXME why can't we allow direct AND fast? 290 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 291 } 292 // only allow deep buffering for music stream type 293 if (streamType != AUDIO_STREAM_MUSIC) { 294 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 295 } 296 297 mChannelMask = channelMask; 298 uint32_t channelCount = popcount(channelMask); 299 mChannelCount = channelCount; 300 301 if (audio_is_linear_pcm(format)) { 302 mFrameSize = channelCount * audio_bytes_per_sample(format); 303 mFrameSizeAF = channelCount * sizeof(int16_t); 304 } else { 305 mFrameSize = sizeof(uint8_t); 306 mFrameSizeAF = sizeof(uint8_t); 307 } 308 309 audio_io_handle_t output = AudioSystem::getOutput( 310 streamType, 311 sampleRate, format, channelMask, 312 flags, 313 offloadInfo); 314 315 if (output == 0) { 316 ALOGE("Could not get audio output for stream type %d", streamType); 317 return BAD_VALUE; 318 } 319 320 mVolume[LEFT] = 1.0f; 321 mVolume[RIGHT] = 1.0f; 322 mSendLevel = 0.0f; 323 mFrameCount = frameCount; 324 mReqFrameCount = frameCount; 325 mNotificationFramesReq = notificationFrames; 326 mNotificationFramesAct = 0; 327 mSessionId = sessionId; 328 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 329 mClientUid = IPCThreadState::self()->getCallingUid(); 330 } else { 331 mClientUid = uid; 332 } 333 mAuxEffectId = 0; 334 mFlags = flags; 335 mCbf = cbf; 336 337 if (cbf != NULL) { 338 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 339 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 340 } 341 342 // create the IAudioTrack 343 status = createTrack_l(streamType, 344 sampleRate, 345 format, 346 frameCount, 347 flags, 348 sharedBuffer, 349 output, 350 0 /*epoch*/); 351 352 if (status != NO_ERROR) { 353 if (mAudioTrackThread != 0) { 354 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 355 mAudioTrackThread->requestExitAndWait(); 356 mAudioTrackThread.clear(); 357 } 358 // Use of direct and offloaded output streams is ref counted by audio policy manager. 359 // As getOutput was called above and resulted in an output stream to be opened, 360 // we need to release it. 361 AudioSystem::releaseOutput(output); 362 return status; 363 } 364 365 mStatus = NO_ERROR; 366 mStreamType = streamType; 367 mFormat = format; 368 mSharedBuffer = sharedBuffer; 369 mState = STATE_STOPPED; 370 mUserData = user; 371 mLoopPeriod = 0; 372 mMarkerPosition = 0; 373 mMarkerReached = false; 374 mNewPosition = 0; 375 mUpdatePeriod = 0; 376 AudioSystem::acquireAudioSessionId(mSessionId); 377 mSequence = 1; 378 mObservedSequence = mSequence; 379 mInUnderrun = false; 380 mOutput = output; 381 382 return NO_ERROR; 383} 384 385// ------------------------------------------------------------------------- 386 387status_t AudioTrack::start() 388{ 389 AutoMutex lock(mLock); 390 391 if (mState == STATE_ACTIVE) { 392 return INVALID_OPERATION; 393 } 394 395 mInUnderrun = true; 396 397 State previousState = mState; 398 if (previousState == STATE_PAUSED_STOPPING) { 399 mState = STATE_STOPPING; 400 } else { 401 mState = STATE_ACTIVE; 402 } 403 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 404 // reset current position as seen by client to 0 405 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 406 // force refresh of remaining frames by processAudioBuffer() as last 407 // write before stop could be partial. 408 mRefreshRemaining = true; 409 } 410 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 411 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 412 413 sp<AudioTrackThread> t = mAudioTrackThread; 414 if (t != 0) { 415 if (previousState == STATE_STOPPING) { 416 mProxy->interrupt(); 417 } else { 418 t->resume(); 419 } 420 } else { 421 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 422 get_sched_policy(0, &mPreviousSchedulingGroup); 423 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 424 } 425 426 status_t status = NO_ERROR; 427 if (!(flags & CBLK_INVALID)) { 428 status = mAudioTrack->start(); 429 if (status == DEAD_OBJECT) { 430 flags |= CBLK_INVALID; 431 } 432 } 433 if (flags & CBLK_INVALID) { 434 status = restoreTrack_l("start"); 435 } 436 437 if (status != NO_ERROR) { 438 ALOGE("start() status %d", status); 439 mState = previousState; 440 if (t != 0) { 441 if (previousState != STATE_STOPPING) { 442 t->pause(); 443 } 444 } else { 445 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 446 set_sched_policy(0, mPreviousSchedulingGroup); 447 } 448 } 449 450 return status; 451} 452 453void AudioTrack::stop() 454{ 455 AutoMutex lock(mLock); 456 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 457 return; 458 } 459 460 if (isOffloaded_l()) { 461 mState = STATE_STOPPING; 462 } else { 463 mState = STATE_STOPPED; 464 } 465 466 mProxy->interrupt(); 467 mAudioTrack->stop(); 468 // the playback head position will reset to 0, so if a marker is set, we need 469 // to activate it again 470 mMarkerReached = false; 471#if 0 472 // Force flush if a shared buffer is used otherwise audioflinger 473 // will not stop before end of buffer is reached. 474 // It may be needed to make sure that we stop playback, likely in case looping is on. 475 if (mSharedBuffer != 0) { 476 flush_l(); 477 } 478#endif 479 480 sp<AudioTrackThread> t = mAudioTrackThread; 481 if (t != 0) { 482 if (!isOffloaded_l()) { 483 t->pause(); 484 } 485 } else { 486 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 487 set_sched_policy(0, mPreviousSchedulingGroup); 488 } 489} 490 491bool AudioTrack::stopped() const 492{ 493 AutoMutex lock(mLock); 494 return mState != STATE_ACTIVE; 495} 496 497void AudioTrack::flush() 498{ 499 if (mSharedBuffer != 0) { 500 return; 501 } 502 AutoMutex lock(mLock); 503 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 504 return; 505 } 506 flush_l(); 507} 508 509void AudioTrack::flush_l() 510{ 511 ALOG_ASSERT(mState != STATE_ACTIVE); 512 513 // clear playback marker and periodic update counter 514 mMarkerPosition = 0; 515 mMarkerReached = false; 516 mUpdatePeriod = 0; 517 mRefreshRemaining = true; 518 519 mState = STATE_FLUSHED; 520 if (isOffloaded_l()) { 521 mProxy->interrupt(); 522 } 523 mProxy->flush(); 524 mAudioTrack->flush(); 525} 526 527void AudioTrack::pause() 528{ 529 AutoMutex lock(mLock); 530 if (mState == STATE_ACTIVE) { 531 mState = STATE_PAUSED; 532 } else if (mState == STATE_STOPPING) { 533 mState = STATE_PAUSED_STOPPING; 534 } else { 535 return; 536 } 537 mProxy->interrupt(); 538 mAudioTrack->pause(); 539} 540 541status_t AudioTrack::setVolume(float left, float right) 542{ 543 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 544 return BAD_VALUE; 545 } 546 547 AutoMutex lock(mLock); 548 mVolume[LEFT] = left; 549 mVolume[RIGHT] = right; 550 551 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 552 553 if (isOffloaded_l()) { 554 mAudioTrack->signal(); 555 } 556 return NO_ERROR; 557} 558 559status_t AudioTrack::setVolume(float volume) 560{ 561 return setVolume(volume, volume); 562} 563 564status_t AudioTrack::setAuxEffectSendLevel(float level) 565{ 566 if (level < 0.0f || level > 1.0f) { 567 return BAD_VALUE; 568 } 569 570 AutoMutex lock(mLock); 571 mSendLevel = level; 572 mProxy->setSendLevel(level); 573 574 return NO_ERROR; 575} 576 577void AudioTrack::getAuxEffectSendLevel(float* level) const 578{ 579 if (level != NULL) { 580 *level = mSendLevel; 581 } 582} 583 584status_t AudioTrack::setSampleRate(uint32_t rate) 585{ 586 if (mIsTimed || isOffloaded()) { 587 return INVALID_OPERATION; 588 } 589 590 uint32_t afSamplingRate; 591 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 592 return NO_INIT; 593 } 594 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 595 if (rate == 0 || rate > afSamplingRate*2 ) { 596 return BAD_VALUE; 597 } 598 599 AutoMutex lock(mLock); 600 mSampleRate = rate; 601 mProxy->setSampleRate(rate); 602 603 return NO_ERROR; 604} 605 606uint32_t AudioTrack::getSampleRate() const 607{ 608 if (mIsTimed) { 609 return 0; 610 } 611 612 AutoMutex lock(mLock); 613 614 // sample rate can be updated during playback by the offloaded decoder so we need to 615 // query the HAL and update if needed. 616// FIXME use Proxy return channel to update the rate from server and avoid polling here 617 if (isOffloaded_l()) { 618 if (mOutput != 0) { 619 uint32_t sampleRate = 0; 620 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 621 if (status == NO_ERROR) { 622 mSampleRate = sampleRate; 623 } 624 } 625 } 626 return mSampleRate; 627} 628 629status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 630{ 631 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 632 return INVALID_OPERATION; 633 } 634 635 if (loopCount == 0) { 636 ; 637 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 638 loopEnd - loopStart >= MIN_LOOP) { 639 ; 640 } else { 641 return BAD_VALUE; 642 } 643 644 AutoMutex lock(mLock); 645 // See setPosition() regarding setting parameters such as loop points or position while active 646 if (mState == STATE_ACTIVE) { 647 return INVALID_OPERATION; 648 } 649 setLoop_l(loopStart, loopEnd, loopCount); 650 return NO_ERROR; 651} 652 653void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 654{ 655 // FIXME If setting a loop also sets position to start of loop, then 656 // this is correct. Otherwise it should be removed. 657 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 658 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 659 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 660} 661 662status_t AudioTrack::setMarkerPosition(uint32_t marker) 663{ 664 // The only purpose of setting marker position is to get a callback 665 if (mCbf == NULL || isOffloaded()) { 666 return INVALID_OPERATION; 667 } 668 669 AutoMutex lock(mLock); 670 mMarkerPosition = marker; 671 mMarkerReached = false; 672 673 return NO_ERROR; 674} 675 676status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 677{ 678 if (isOffloaded()) { 679 return INVALID_OPERATION; 680 } 681 if (marker == NULL) { 682 return BAD_VALUE; 683 } 684 685 AutoMutex lock(mLock); 686 *marker = mMarkerPosition; 687 688 return NO_ERROR; 689} 690 691status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 692{ 693 // The only purpose of setting position update period is to get a callback 694 if (mCbf == NULL || isOffloaded()) { 695 return INVALID_OPERATION; 696 } 697 698 AutoMutex lock(mLock); 699 mNewPosition = mProxy->getPosition() + updatePeriod; 700 mUpdatePeriod = updatePeriod; 701 702 return NO_ERROR; 703} 704 705status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 706{ 707 if (isOffloaded()) { 708 return INVALID_OPERATION; 709 } 710 if (updatePeriod == NULL) { 711 return BAD_VALUE; 712 } 713 714 AutoMutex lock(mLock); 715 *updatePeriod = mUpdatePeriod; 716 717 return NO_ERROR; 718} 719 720status_t AudioTrack::setPosition(uint32_t position) 721{ 722 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 723 return INVALID_OPERATION; 724 } 725 if (position > mFrameCount) { 726 return BAD_VALUE; 727 } 728 729 AutoMutex lock(mLock); 730 // Currently we require that the player is inactive before setting parameters such as position 731 // or loop points. Otherwise, there could be a race condition: the application could read the 732 // current position, compute a new position or loop parameters, and then set that position or 733 // loop parameters but it would do the "wrong" thing since the position has continued to advance 734 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 735 // to specify how it wants to handle such scenarios. 736 if (mState == STATE_ACTIVE) { 737 return INVALID_OPERATION; 738 } 739 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 740 mLoopPeriod = 0; 741 // FIXME Check whether loops and setting position are incompatible in old code. 742 // If we use setLoop for both purposes we lose the capability to set the position while looping. 743 mStaticProxy->setLoop(position, mFrameCount, 0); 744 745 return NO_ERROR; 746} 747 748status_t AudioTrack::getPosition(uint32_t *position) const 749{ 750 if (position == NULL) { 751 return BAD_VALUE; 752 } 753 754 AutoMutex lock(mLock); 755 if (isOffloaded_l()) { 756 uint32_t dspFrames = 0; 757 758 if (mOutput != 0) { 759 uint32_t halFrames; 760 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 761 } 762 *position = dspFrames; 763 } else { 764 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 765 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 766 mProxy->getPosition(); 767 } 768 return NO_ERROR; 769} 770 771status_t AudioTrack::getBufferPosition(size_t *position) 772{ 773 if (mSharedBuffer == 0 || mIsTimed) { 774 return INVALID_OPERATION; 775 } 776 if (position == NULL) { 777 return BAD_VALUE; 778 } 779 780 AutoMutex lock(mLock); 781 *position = mStaticProxy->getBufferPosition(); 782 return NO_ERROR; 783} 784 785status_t AudioTrack::reload() 786{ 787 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 788 return INVALID_OPERATION; 789 } 790 791 AutoMutex lock(mLock); 792 // See setPosition() regarding setting parameters such as loop points or position while active 793 if (mState == STATE_ACTIVE) { 794 return INVALID_OPERATION; 795 } 796 mNewPosition = mUpdatePeriod; 797 mLoopPeriod = 0; 798 // FIXME The new code cannot reload while keeping a loop specified. 799 // Need to check how the old code handled this, and whether it's a significant change. 800 mStaticProxy->setLoop(0, mFrameCount, 0); 801 return NO_ERROR; 802} 803 804audio_io_handle_t AudioTrack::getOutput() 805{ 806 AutoMutex lock(mLock); 807 return mOutput; 808} 809 810// must be called with mLock held 811audio_io_handle_t AudioTrack::getOutput_l() 812{ 813 if (mOutput) { 814 return mOutput; 815 } else { 816 return AudioSystem::getOutput(mStreamType, 817 mSampleRate, mFormat, mChannelMask, mFlags); 818 } 819} 820 821status_t AudioTrack::attachAuxEffect(int effectId) 822{ 823 AutoMutex lock(mLock); 824 status_t status = mAudioTrack->attachAuxEffect(effectId); 825 if (status == NO_ERROR) { 826 mAuxEffectId = effectId; 827 } 828 return status; 829} 830 831// ------------------------------------------------------------------------- 832 833// must be called with mLock held 834status_t AudioTrack::createTrack_l( 835 audio_stream_type_t streamType, 836 uint32_t sampleRate, 837 audio_format_t format, 838 size_t frameCount, 839 audio_output_flags_t flags, 840 const sp<IMemory>& sharedBuffer, 841 audio_io_handle_t output, 842 size_t epoch) 843{ 844 status_t status; 845 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 846 if (audioFlinger == 0) { 847 ALOGE("Could not get audioflinger"); 848 return NO_INIT; 849 } 850 851 // Not all of these values are needed under all conditions, but it is easier to get them all 852 853 uint32_t afLatency; 854 status = AudioSystem::getLatency(output, streamType, &afLatency); 855 if (status != NO_ERROR) { 856 ALOGE("getLatency(%d) failed status %d", output, status); 857 return NO_INIT; 858 } 859 860 size_t afFrameCount; 861 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 862 if (status != NO_ERROR) { 863 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 864 return NO_INIT; 865 } 866 867 uint32_t afSampleRate; 868 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 869 if (status != NO_ERROR) { 870 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 871 return NO_INIT; 872 } 873 874 // Client decides whether the track is TIMED (see below), but can only express a preference 875 // for FAST. Server will perform additional tests. 876 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 877 // either of these use cases: 878 // use case 1: shared buffer 879 (sharedBuffer != 0) || 880 // use case 2: callback handler 881 (mCbf != NULL))) { 882 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 883 // once denied, do not request again if IAudioTrack is re-created 884 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 885 mFlags = flags; 886 } 887 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 888 889 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 890 // n = 1 fast track with single buffering; nBuffering is ignored 891 // n = 2 fast track with double buffering 892 // n = 2 normal track, no sample rate conversion 893 // n = 3 normal track, with sample rate conversion 894 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 895 // n > 3 very high latency or very small notification interval; nBuffering is ignored 896 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 897 898 mNotificationFramesAct = mNotificationFramesReq; 899 900 if (!audio_is_linear_pcm(format)) { 901 902 if (sharedBuffer != 0) { 903 // Same comment as below about ignoring frameCount parameter for set() 904 frameCount = sharedBuffer->size(); 905 } else if (frameCount == 0) { 906 frameCount = afFrameCount; 907 } 908 if (mNotificationFramesAct != frameCount) { 909 mNotificationFramesAct = frameCount; 910 } 911 } else if (sharedBuffer != 0) { 912 913 // Ensure that buffer alignment matches channel count 914 // 8-bit data in shared memory is not currently supported by AudioFlinger 915 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 916 if (mChannelCount > 1) { 917 // More than 2 channels does not require stronger alignment than stereo 918 alignment <<= 1; 919 } 920 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 921 ALOGE("Invalid buffer alignment: address %p, channel count %u", 922 sharedBuffer->pointer(), mChannelCount); 923 return BAD_VALUE; 924 } 925 926 // When initializing a shared buffer AudioTrack via constructors, 927 // there's no frameCount parameter. 928 // But when initializing a shared buffer AudioTrack via set(), 929 // there _is_ a frameCount parameter. We silently ignore it. 930 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 931 932 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 933 934 // FIXME move these calculations and associated checks to server 935 936 // Ensure that buffer depth covers at least audio hardware latency 937 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 938 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 939 afFrameCount, minBufCount, afSampleRate, afLatency); 940 if (minBufCount <= nBuffering) { 941 minBufCount = nBuffering; 942 } 943 944 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 945 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 946 ", afLatency=%d", 947 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 948 949 if (frameCount == 0) { 950 frameCount = minFrameCount; 951 } else if (frameCount < minFrameCount) { 952 // not ALOGW because it happens all the time when playing key clicks over A2DP 953 ALOGV("Minimum buffer size corrected from %d to %d", 954 frameCount, minFrameCount); 955 frameCount = minFrameCount; 956 } 957 // Make sure that application is notified with sufficient margin before underrun 958 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 959 mNotificationFramesAct = frameCount/nBuffering; 960 } 961 962 } else { 963 // For fast tracks, the frame count calculations and checks are done by server 964 } 965 966 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 967 if (mIsTimed) { 968 trackFlags |= IAudioFlinger::TRACK_TIMED; 969 } 970 971 pid_t tid = -1; 972 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 973 trackFlags |= IAudioFlinger::TRACK_FAST; 974 if (mAudioTrackThread != 0) { 975 tid = mAudioTrackThread->getTid(); 976 } 977 } 978 979 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 980 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 981 } 982 983 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 984 sampleRate, 985 // AudioFlinger only sees 16-bit PCM 986 format == AUDIO_FORMAT_PCM_8_BIT ? 987 AUDIO_FORMAT_PCM_16_BIT : format, 988 mChannelMask, 989 frameCount, 990 &trackFlags, 991 sharedBuffer, 992 output, 993 tid, 994 &mSessionId, 995 mName, 996 mClientUid, 997 &status); 998 999 if (track == 0) { 1000 ALOGE("AudioFlinger could not create track, status: %d", status); 1001 return status; 1002 } 1003 sp<IMemory> iMem = track->getCblk(); 1004 if (iMem == 0) { 1005 ALOGE("Could not get control block"); 1006 return NO_INIT; 1007 } 1008 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1009 if (mAudioTrack != 0) { 1010 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1011 mDeathNotifier.clear(); 1012 } 1013 mAudioTrack = track; 1014 mCblkMemory = iMem; 1015 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1016 mCblk = cblk; 1017 size_t temp = cblk->frameCount_; 1018 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1019 // In current design, AudioTrack client checks and ensures frame count validity before 1020 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1021 // for fast track as it uses a special method of assigning frame count. 1022 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1023 } 1024 frameCount = temp; 1025 mAwaitBoost = false; 1026 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1027 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1028 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1029 mAwaitBoost = true; 1030 if (sharedBuffer == 0) { 1031 // Theoretically double-buffering is not required for fast tracks, 1032 // due to tighter scheduling. But in practice, to accommodate kernels with 1033 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1034 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1035 mNotificationFramesAct = frameCount/nBuffering; 1036 } 1037 } 1038 } else { 1039 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1040 // once denied, do not request again if IAudioTrack is re-created 1041 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1042 mFlags = flags; 1043 if (sharedBuffer == 0) { 1044 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1045 mNotificationFramesAct = frameCount/nBuffering; 1046 } 1047 } 1048 } 1049 } 1050 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1051 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1052 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1053 } else { 1054 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1055 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1056 mFlags = flags; 1057 return NO_INIT; 1058 } 1059 } 1060 1061 mRefreshRemaining = true; 1062 1063 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1064 // is the value of pointer() for the shared buffer, otherwise buffers points 1065 // immediately after the control block. This address is for the mapping within client 1066 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1067 void* buffers; 1068 if (sharedBuffer == 0) { 1069 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1070 } else { 1071 buffers = sharedBuffer->pointer(); 1072 } 1073 1074 mAudioTrack->attachAuxEffect(mAuxEffectId); 1075 // FIXME don't believe this lie 1076 mLatency = afLatency + (1000*frameCount) / sampleRate; 1077 mFrameCount = frameCount; 1078 // If IAudioTrack is re-created, don't let the requested frameCount 1079 // decrease. This can confuse clients that cache frameCount(). 1080 if (frameCount > mReqFrameCount) { 1081 mReqFrameCount = frameCount; 1082 } 1083 1084 // update proxy 1085 if (sharedBuffer == 0) { 1086 mStaticProxy.clear(); 1087 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1088 } else { 1089 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1090 mProxy = mStaticProxy; 1091 } 1092 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1093 uint16_t(mVolume[LEFT] * 0x1000)); 1094 mProxy->setSendLevel(mSendLevel); 1095 mProxy->setSampleRate(mSampleRate); 1096 mProxy->setEpoch(epoch); 1097 mProxy->setMinimum(mNotificationFramesAct); 1098 1099 mDeathNotifier = new DeathNotifier(this); 1100 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1101 1102 return NO_ERROR; 1103} 1104 1105status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1106{ 1107 if (audioBuffer == NULL) { 1108 return BAD_VALUE; 1109 } 1110 if (mTransfer != TRANSFER_OBTAIN) { 1111 audioBuffer->frameCount = 0; 1112 audioBuffer->size = 0; 1113 audioBuffer->raw = NULL; 1114 return INVALID_OPERATION; 1115 } 1116 1117 const struct timespec *requested; 1118 if (waitCount == -1) { 1119 requested = &ClientProxy::kForever; 1120 } else if (waitCount == 0) { 1121 requested = &ClientProxy::kNonBlocking; 1122 } else if (waitCount > 0) { 1123 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1124 struct timespec timeout; 1125 timeout.tv_sec = ms / 1000; 1126 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1127 requested = &timeout; 1128 } else { 1129 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1130 requested = NULL; 1131 } 1132 return obtainBuffer(audioBuffer, requested); 1133} 1134 1135status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1136 struct timespec *elapsed, size_t *nonContig) 1137{ 1138 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1139 uint32_t oldSequence = 0; 1140 uint32_t newSequence; 1141 1142 Proxy::Buffer buffer; 1143 status_t status = NO_ERROR; 1144 1145 static const int32_t kMaxTries = 5; 1146 int32_t tryCounter = kMaxTries; 1147 1148 do { 1149 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1150 // keep them from going away if another thread re-creates the track during obtainBuffer() 1151 sp<AudioTrackClientProxy> proxy; 1152 sp<IMemory> iMem; 1153 1154 { // start of lock scope 1155 AutoMutex lock(mLock); 1156 1157 newSequence = mSequence; 1158 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1159 if (status == DEAD_OBJECT) { 1160 // re-create track, unless someone else has already done so 1161 if (newSequence == oldSequence) { 1162 status = restoreTrack_l("obtainBuffer"); 1163 if (status != NO_ERROR) { 1164 buffer.mFrameCount = 0; 1165 buffer.mRaw = NULL; 1166 buffer.mNonContig = 0; 1167 break; 1168 } 1169 } 1170 } 1171 oldSequence = newSequence; 1172 1173 // Keep the extra references 1174 proxy = mProxy; 1175 iMem = mCblkMemory; 1176 1177 if (mState == STATE_STOPPING) { 1178 status = -EINTR; 1179 buffer.mFrameCount = 0; 1180 buffer.mRaw = NULL; 1181 buffer.mNonContig = 0; 1182 break; 1183 } 1184 1185 // Non-blocking if track is stopped or paused 1186 if (mState != STATE_ACTIVE) { 1187 requested = &ClientProxy::kNonBlocking; 1188 } 1189 1190 } // end of lock scope 1191 1192 buffer.mFrameCount = audioBuffer->frameCount; 1193 // FIXME starts the requested timeout and elapsed over from scratch 1194 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1195 1196 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1197 1198 audioBuffer->frameCount = buffer.mFrameCount; 1199 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1200 audioBuffer->raw = buffer.mRaw; 1201 if (nonContig != NULL) { 1202 *nonContig = buffer.mNonContig; 1203 } 1204 return status; 1205} 1206 1207void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1208{ 1209 if (mTransfer == TRANSFER_SHARED) { 1210 return; 1211 } 1212 1213 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1214 if (stepCount == 0) { 1215 return; 1216 } 1217 1218 Proxy::Buffer buffer; 1219 buffer.mFrameCount = stepCount; 1220 buffer.mRaw = audioBuffer->raw; 1221 1222 AutoMutex lock(mLock); 1223 mInUnderrun = false; 1224 mProxy->releaseBuffer(&buffer); 1225 1226 // restart track if it was disabled by audioflinger due to previous underrun 1227 if (mState == STATE_ACTIVE) { 1228 audio_track_cblk_t* cblk = mCblk; 1229 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1230 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1231 this, mName.string()); 1232 // FIXME ignoring status 1233 mAudioTrack->start(); 1234 } 1235 } 1236} 1237 1238// ------------------------------------------------------------------------- 1239 1240ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1241{ 1242 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1243 return INVALID_OPERATION; 1244 } 1245 1246 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1247 // Sanity-check: user is most-likely passing an error code, and it would 1248 // make the return value ambiguous (actualSize vs error). 1249 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1250 return BAD_VALUE; 1251 } 1252 1253 size_t written = 0; 1254 Buffer audioBuffer; 1255 1256 while (userSize >= mFrameSize) { 1257 audioBuffer.frameCount = userSize / mFrameSize; 1258 1259 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1260 if (err < 0) { 1261 if (written > 0) { 1262 break; 1263 } 1264 return ssize_t(err); 1265 } 1266 1267 size_t toWrite; 1268 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1269 // Divide capacity by 2 to take expansion into account 1270 toWrite = audioBuffer.size >> 1; 1271 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1272 } else { 1273 toWrite = audioBuffer.size; 1274 memcpy(audioBuffer.i8, buffer, toWrite); 1275 } 1276 buffer = ((const char *) buffer) + toWrite; 1277 userSize -= toWrite; 1278 written += toWrite; 1279 1280 releaseBuffer(&audioBuffer); 1281 } 1282 1283 return written; 1284} 1285 1286// ------------------------------------------------------------------------- 1287 1288TimedAudioTrack::TimedAudioTrack() { 1289 mIsTimed = true; 1290} 1291 1292status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1293{ 1294 AutoMutex lock(mLock); 1295 status_t result = UNKNOWN_ERROR; 1296 1297#if 1 1298 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1299 // while we are accessing the cblk 1300 sp<IAudioTrack> audioTrack = mAudioTrack; 1301 sp<IMemory> iMem = mCblkMemory; 1302#endif 1303 1304 // If the track is not invalid already, try to allocate a buffer. alloc 1305 // fails indicating that the server is dead, flag the track as invalid so 1306 // we can attempt to restore in just a bit. 1307 audio_track_cblk_t* cblk = mCblk; 1308 if (!(cblk->mFlags & CBLK_INVALID)) { 1309 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1310 if (result == DEAD_OBJECT) { 1311 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1312 } 1313 } 1314 1315 // If the track is invalid at this point, attempt to restore it. and try the 1316 // allocation one more time. 1317 if (cblk->mFlags & CBLK_INVALID) { 1318 result = restoreTrack_l("allocateTimedBuffer"); 1319 1320 if (result == NO_ERROR) { 1321 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1322 } 1323 } 1324 1325 return result; 1326} 1327 1328status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1329 int64_t pts) 1330{ 1331 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1332 { 1333 AutoMutex lock(mLock); 1334 audio_track_cblk_t* cblk = mCblk; 1335 // restart track if it was disabled by audioflinger due to previous underrun 1336 if (buffer->size() != 0 && status == NO_ERROR && 1337 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1338 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1339 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1340 // FIXME ignoring status 1341 mAudioTrack->start(); 1342 } 1343 } 1344 return status; 1345} 1346 1347status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1348 TargetTimeline target) 1349{ 1350 return mAudioTrack->setMediaTimeTransform(xform, target); 1351} 1352 1353// ------------------------------------------------------------------------- 1354 1355nsecs_t AudioTrack::processAudioBuffer() 1356{ 1357 // Currently the AudioTrack thread is not created if there are no callbacks. 1358 // Would it ever make sense to run the thread, even without callbacks? 1359 // If so, then replace this by checks at each use for mCbf != NULL. 1360 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1361 1362 mLock.lock(); 1363 if (mAwaitBoost) { 1364 mAwaitBoost = false; 1365 mLock.unlock(); 1366 static const int32_t kMaxTries = 5; 1367 int32_t tryCounter = kMaxTries; 1368 uint32_t pollUs = 10000; 1369 do { 1370 int policy = sched_getscheduler(0); 1371 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1372 break; 1373 } 1374 usleep(pollUs); 1375 pollUs <<= 1; 1376 } while (tryCounter-- > 0); 1377 if (tryCounter < 0) { 1378 ALOGE("did not receive expected priority boost on time"); 1379 } 1380 // Run again immediately 1381 return 0; 1382 } 1383 1384 // Can only reference mCblk while locked 1385 int32_t flags = android_atomic_and( 1386 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1387 1388 // Check for track invalidation 1389 if (flags & CBLK_INVALID) { 1390 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1391 // AudioSystem cache. We should not exit here but after calling the callback so 1392 // that the upper layers can recreate the track 1393 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1394 status_t status = restoreTrack_l("processAudioBuffer"); 1395 mLock.unlock(); 1396 // Run again immediately, but with a new IAudioTrack 1397 return 0; 1398 } 1399 } 1400 1401 bool waitStreamEnd = mState == STATE_STOPPING; 1402 bool active = mState == STATE_ACTIVE; 1403 1404 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1405 bool newUnderrun = false; 1406 if (flags & CBLK_UNDERRUN) { 1407#if 0 1408 // Currently in shared buffer mode, when the server reaches the end of buffer, 1409 // the track stays active in continuous underrun state. It's up to the application 1410 // to pause or stop the track, or set the position to a new offset within buffer. 1411 // This was some experimental code to auto-pause on underrun. Keeping it here 1412 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1413 if (mTransfer == TRANSFER_SHARED) { 1414 mState = STATE_PAUSED; 1415 active = false; 1416 } 1417#endif 1418 if (!mInUnderrun) { 1419 mInUnderrun = true; 1420 newUnderrun = true; 1421 } 1422 } 1423 1424 // Get current position of server 1425 size_t position = mProxy->getPosition(); 1426 1427 // Manage marker callback 1428 bool markerReached = false; 1429 size_t markerPosition = mMarkerPosition; 1430 // FIXME fails for wraparound, need 64 bits 1431 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1432 mMarkerReached = markerReached = true; 1433 } 1434 1435 // Determine number of new position callback(s) that will be needed, while locked 1436 size_t newPosCount = 0; 1437 size_t newPosition = mNewPosition; 1438 size_t updatePeriod = mUpdatePeriod; 1439 // FIXME fails for wraparound, need 64 bits 1440 if (updatePeriod > 0 && position >= newPosition) { 1441 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1442 mNewPosition += updatePeriod * newPosCount; 1443 } 1444 1445 // Cache other fields that will be needed soon 1446 uint32_t loopPeriod = mLoopPeriod; 1447 uint32_t sampleRate = mSampleRate; 1448 size_t notificationFrames = mNotificationFramesAct; 1449 if (mRefreshRemaining) { 1450 mRefreshRemaining = false; 1451 mRemainingFrames = notificationFrames; 1452 mRetryOnPartialBuffer = false; 1453 } 1454 size_t misalignment = mProxy->getMisalignment(); 1455 uint32_t sequence = mSequence; 1456 1457 // These fields don't need to be cached, because they are assigned only by set(): 1458 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1459 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1460 1461 mLock.unlock(); 1462 1463 if (waitStreamEnd) { 1464 AutoMutex lock(mLock); 1465 1466 sp<AudioTrackClientProxy> proxy = mProxy; 1467 sp<IMemory> iMem = mCblkMemory; 1468 1469 struct timespec timeout; 1470 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1471 timeout.tv_nsec = 0; 1472 1473 mLock.unlock(); 1474 status_t status = mProxy->waitStreamEndDone(&timeout); 1475 mLock.lock(); 1476 switch (status) { 1477 case NO_ERROR: 1478 case DEAD_OBJECT: 1479 case TIMED_OUT: 1480 mLock.unlock(); 1481 mCbf(EVENT_STREAM_END, mUserData, NULL); 1482 mLock.lock(); 1483 if (mState == STATE_STOPPING) { 1484 mState = STATE_STOPPED; 1485 if (status != DEAD_OBJECT) { 1486 return NS_INACTIVE; 1487 } 1488 } 1489 return 0; 1490 default: 1491 return 0; 1492 } 1493 } 1494 1495 // perform callbacks while unlocked 1496 if (newUnderrun) { 1497 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1498 } 1499 // FIXME we will miss loops if loop cycle was signaled several times since last call 1500 // to processAudioBuffer() 1501 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1502 mCbf(EVENT_LOOP_END, mUserData, NULL); 1503 } 1504 if (flags & CBLK_BUFFER_END) { 1505 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1506 } 1507 if (markerReached) { 1508 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1509 } 1510 while (newPosCount > 0) { 1511 size_t temp = newPosition; 1512 mCbf(EVENT_NEW_POS, mUserData, &temp); 1513 newPosition += updatePeriod; 1514 newPosCount--; 1515 } 1516 1517 if (mObservedSequence != sequence) { 1518 mObservedSequence = sequence; 1519 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1520 // for offloaded tracks, just wait for the upper layers to recreate the track 1521 if (isOffloaded()) { 1522 return NS_INACTIVE; 1523 } 1524 } 1525 1526 // if inactive, then don't run me again until re-started 1527 if (!active) { 1528 return NS_INACTIVE; 1529 } 1530 1531 // Compute the estimated time until the next timed event (position, markers, loops) 1532 // FIXME only for non-compressed audio 1533 uint32_t minFrames = ~0; 1534 if (!markerReached && position < markerPosition) { 1535 minFrames = markerPosition - position; 1536 } 1537 if (loopPeriod > 0 && loopPeriod < minFrames) { 1538 minFrames = loopPeriod; 1539 } 1540 if (updatePeriod > 0 && updatePeriod < minFrames) { 1541 minFrames = updatePeriod; 1542 } 1543 1544 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1545 static const uint32_t kPoll = 0; 1546 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1547 minFrames = kPoll * notificationFrames; 1548 } 1549 1550 // Convert frame units to time units 1551 nsecs_t ns = NS_WHENEVER; 1552 if (minFrames != (uint32_t) ~0) { 1553 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1554 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1555 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1556 } 1557 1558 // If not supplying data by EVENT_MORE_DATA, then we're done 1559 if (mTransfer != TRANSFER_CALLBACK) { 1560 return ns; 1561 } 1562 1563 struct timespec timeout; 1564 const struct timespec *requested = &ClientProxy::kForever; 1565 if (ns != NS_WHENEVER) { 1566 timeout.tv_sec = ns / 1000000000LL; 1567 timeout.tv_nsec = ns % 1000000000LL; 1568 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1569 requested = &timeout; 1570 } 1571 1572 while (mRemainingFrames > 0) { 1573 1574 Buffer audioBuffer; 1575 audioBuffer.frameCount = mRemainingFrames; 1576 size_t nonContig; 1577 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1578 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1579 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1580 requested = &ClientProxy::kNonBlocking; 1581 size_t avail = audioBuffer.frameCount + nonContig; 1582 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1583 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1584 if (err != NO_ERROR) { 1585 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1586 (isOffloaded() && (err == DEAD_OBJECT))) { 1587 return 0; 1588 } 1589 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1590 return NS_NEVER; 1591 } 1592 1593 if (mRetryOnPartialBuffer && !isOffloaded()) { 1594 mRetryOnPartialBuffer = false; 1595 if (avail < mRemainingFrames) { 1596 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1597 if (ns < 0 || myns < ns) { 1598 ns = myns; 1599 } 1600 return ns; 1601 } 1602 } 1603 1604 // Divide buffer size by 2 to take into account the expansion 1605 // due to 8 to 16 bit conversion: the callback must fill only half 1606 // of the destination buffer 1607 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1608 audioBuffer.size >>= 1; 1609 } 1610 1611 size_t reqSize = audioBuffer.size; 1612 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1613 size_t writtenSize = audioBuffer.size; 1614 size_t writtenFrames = writtenSize / mFrameSize; 1615 1616 // Sanity check on returned size 1617 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1618 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1619 reqSize, (int) writtenSize); 1620 return NS_NEVER; 1621 } 1622 1623 if (writtenSize == 0) { 1624 // The callback is done filling buffers 1625 // Keep this thread going to handle timed events and 1626 // still try to get more data in intervals of WAIT_PERIOD_MS 1627 // but don't just loop and block the CPU, so wait 1628 return WAIT_PERIOD_MS * 1000000LL; 1629 } 1630 1631 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1632 // 8 to 16 bit conversion, note that source and destination are the same address 1633 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1634 audioBuffer.size <<= 1; 1635 } 1636 1637 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1638 audioBuffer.frameCount = releasedFrames; 1639 mRemainingFrames -= releasedFrames; 1640 if (misalignment >= releasedFrames) { 1641 misalignment -= releasedFrames; 1642 } else { 1643 misalignment = 0; 1644 } 1645 1646 releaseBuffer(&audioBuffer); 1647 1648 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1649 // if callback doesn't like to accept the full chunk 1650 if (writtenSize < reqSize) { 1651 continue; 1652 } 1653 1654 // There could be enough non-contiguous frames available to satisfy the remaining request 1655 if (mRemainingFrames <= nonContig) { 1656 continue; 1657 } 1658 1659#if 0 1660 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1661 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1662 // that total to a sum == notificationFrames. 1663 if (0 < misalignment && misalignment <= mRemainingFrames) { 1664 mRemainingFrames = misalignment; 1665 return (mRemainingFrames * 1100000000LL) / sampleRate; 1666 } 1667#endif 1668 1669 } 1670 mRemainingFrames = notificationFrames; 1671 mRetryOnPartialBuffer = true; 1672 1673 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1674 return 0; 1675} 1676 1677status_t AudioTrack::restoreTrack_l(const char *from) 1678{ 1679 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1680 isOffloaded_l() ? "Offloaded" : "PCM", from); 1681 ++mSequence; 1682 status_t result; 1683 1684 // refresh the audio configuration cache in this process to make sure we get new 1685 // output parameters in getOutput_l() and createTrack_l() 1686 AudioSystem::clearAudioConfigCache(); 1687 1688 if (isOffloaded_l()) { 1689 // FIXME re-creation of offloaded tracks is not yet implemented 1690 return DEAD_OBJECT; 1691 } 1692 1693 // force new output query from audio policy manager; 1694 mOutput = 0; 1695 audio_io_handle_t output = getOutput_l(); 1696 1697 // if the new IAudioTrack is created, createTrack_l() will modify the 1698 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1699 // It will also delete the strong references on previous IAudioTrack and IMemory 1700 1701 // take the frames that will be lost by track recreation into account in saved position 1702 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1703 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1704 result = createTrack_l(mStreamType, 1705 mSampleRate, 1706 mFormat, 1707 mReqFrameCount, // so that frame count never goes down 1708 mFlags, 1709 mSharedBuffer, 1710 output, 1711 position /*epoch*/); 1712 1713 if (result == NO_ERROR) { 1714 // continue playback from last known position, but 1715 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1716 if (mStaticProxy != NULL) { 1717 mLoopPeriod = 0; 1718 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1719 } 1720 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1721 // track destruction have been played? This is critical for SoundPool implementation 1722 // This must be broken, and needs to be tested/debugged. 1723#if 0 1724 // restore write index and set other indexes to reflect empty buffer status 1725 if (!strcmp(from, "start")) { 1726 // Make sure that a client relying on callback events indicating underrun or 1727 // the actual amount of audio frames played (e.g SoundPool) receives them. 1728 if (mSharedBuffer == 0) { 1729 // restart playback even if buffer is not completely filled. 1730 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1731 } 1732 } 1733#endif 1734 if (mState == STATE_ACTIVE) { 1735 result = mAudioTrack->start(); 1736 } 1737 } 1738 if (result != NO_ERROR) { 1739 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1740 // As getOutput was called above and resulted in an output stream to be opened, 1741 // we need to release it. 1742 AudioSystem::releaseOutput(output); 1743 ALOGW("restoreTrack_l() failed status %d", result); 1744 mState = STATE_STOPPED; 1745 } 1746 1747 return result; 1748} 1749 1750status_t AudioTrack::setParameters(const String8& keyValuePairs) 1751{ 1752 AutoMutex lock(mLock); 1753 return mAudioTrack->setParameters(keyValuePairs); 1754} 1755 1756status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1757{ 1758 AutoMutex lock(mLock); 1759 // FIXME not implemented for fast tracks; should use proxy and SSQ 1760 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1761 return INVALID_OPERATION; 1762 } 1763 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1764 return INVALID_OPERATION; 1765 } 1766 status_t status = mAudioTrack->getTimestamp(timestamp); 1767 if (status == NO_ERROR) { 1768 timestamp.mPosition += mProxy->getEpoch(); 1769 } 1770 return status; 1771} 1772 1773String8 AudioTrack::getParameters(const String8& keys) 1774{ 1775 audio_io_handle_t output = getOutput(); 1776 if (output != 0) { 1777 return AudioSystem::getParameters(output, keys); 1778 } else { 1779 return String8::empty(); 1780 } 1781} 1782 1783bool AudioTrack::isOffloaded() const 1784{ 1785 AutoMutex lock(mLock); 1786 return isOffloaded_l(); 1787} 1788 1789status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1790{ 1791 1792 const size_t SIZE = 256; 1793 char buffer[SIZE]; 1794 String8 result; 1795 1796 result.append(" AudioTrack::dump\n"); 1797 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1798 mVolume[0], mVolume[1]); 1799 result.append(buffer); 1800 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1801 mChannelCount, mFrameCount); 1802 result.append(buffer); 1803 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1804 result.append(buffer); 1805 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1806 result.append(buffer); 1807 ::write(fd, result.string(), result.size()); 1808 return NO_ERROR; 1809} 1810 1811uint32_t AudioTrack::getUnderrunFrames() const 1812{ 1813 AutoMutex lock(mLock); 1814 return mProxy->getUnderrunFrames(); 1815} 1816 1817// ========================================================================= 1818 1819void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1820{ 1821 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1822 if (audioTrack != 0) { 1823 AutoMutex lock(audioTrack->mLock); 1824 audioTrack->mProxy->binderDied(); 1825 } 1826} 1827 1828// ========================================================================= 1829 1830AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1831 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1832 mIgnoreNextPausedInt(false) 1833{ 1834} 1835 1836AudioTrack::AudioTrackThread::~AudioTrackThread() 1837{ 1838} 1839 1840bool AudioTrack::AudioTrackThread::threadLoop() 1841{ 1842 { 1843 AutoMutex _l(mMyLock); 1844 if (mPaused) { 1845 mMyCond.wait(mMyLock); 1846 // caller will check for exitPending() 1847 return true; 1848 } 1849 if (mIgnoreNextPausedInt) { 1850 mIgnoreNextPausedInt = false; 1851 mPausedInt = false; 1852 } 1853 if (mPausedInt) { 1854 if (mPausedNs > 0) { 1855 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1856 } else { 1857 mMyCond.wait(mMyLock); 1858 } 1859 mPausedInt = false; 1860 return true; 1861 } 1862 } 1863 nsecs_t ns = mReceiver.processAudioBuffer(); 1864 switch (ns) { 1865 case 0: 1866 return true; 1867 case NS_INACTIVE: 1868 pauseInternal(); 1869 return true; 1870 case NS_NEVER: 1871 return false; 1872 case NS_WHENEVER: 1873 // FIXME increase poll interval, or make event-driven 1874 ns = 1000000000LL; 1875 // fall through 1876 default: 1877 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1878 pauseInternal(ns); 1879 return true; 1880 } 1881} 1882 1883void AudioTrack::AudioTrackThread::requestExit() 1884{ 1885 // must be in this order to avoid a race condition 1886 Thread::requestExit(); 1887 resume(); 1888} 1889 1890void AudioTrack::AudioTrackThread::pause() 1891{ 1892 AutoMutex _l(mMyLock); 1893 mPaused = true; 1894} 1895 1896void AudioTrack::AudioTrackThread::resume() 1897{ 1898 AutoMutex _l(mMyLock); 1899 mIgnoreNextPausedInt = true; 1900 if (mPaused || mPausedInt) { 1901 mPaused = false; 1902 mPausedInt = false; 1903 mMyCond.signal(); 1904 } 1905} 1906 1907void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1908{ 1909 AutoMutex _l(mMyLock); 1910 mPausedInt = true; 1911 mPausedNs = ns; 1912} 1913 1914}; // namespace android 1915