AudioTrack.cpp revision dd5f4c8c4059f890e81b28b026a688febb4e1dd9
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 mSharedBuffer = sharedBuffer; 220 mTransfer = transferType; 221 222 // FIXME "int" here is legacy and will be replaced by size_t later 223 if (frameCountInt < 0) { 224 ALOGE("Invalid frame count %d", frameCountInt); 225 return BAD_VALUE; 226 } 227 size_t frameCount = frameCountInt; 228 229 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 230 sharedBuffer->size()); 231 232 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 233 234 AutoMutex lock(mLock); 235 236 // invariant that mAudioTrack != 0 is true only after set() returns successfully 237 if (mAudioTrack != 0) { 238 ALOGE("Track already in use"); 239 return INVALID_OPERATION; 240 } 241 242 mOutput = 0; 243 244 // handle default values first. 245 if (streamType == AUDIO_STREAM_DEFAULT) { 246 streamType = AUDIO_STREAM_MUSIC; 247 } 248 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 249 ALOGE("Invalid stream type %d", streamType); 250 return BAD_VALUE; 251 } 252 mStreamType = streamType; 253 254 status_t status; 255 if (sampleRate == 0) { 256 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 257 if (status != NO_ERROR) { 258 ALOGE("Could not get output sample rate for stream type %d; status %d", 259 streamType, status); 260 return status; 261 } 262 } 263 mSampleRate = sampleRate; 264 265 // these below should probably come from the audioFlinger too... 266 if (format == AUDIO_FORMAT_DEFAULT) { 267 format = AUDIO_FORMAT_PCM_16_BIT; 268 } 269 270 // validate parameters 271 if (!audio_is_valid_format(format)) { 272 ALOGE("Invalid format %d", format); 273 return BAD_VALUE; 274 } 275 mFormat = format; 276 277 if (!audio_is_output_channel(channelMask)) { 278 ALOGE("Invalid channel mask %#x", channelMask); 279 return BAD_VALUE; 280 } 281 282 // AudioFlinger does not currently support 8-bit data in shared memory 283 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 284 ALOGE("8-bit data in shared memory is not supported"); 285 return BAD_VALUE; 286 } 287 288 // force direct flag if format is not linear PCM 289 // or offload was requested 290 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 291 || !audio_is_linear_pcm(format)) { 292 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 293 ? "Offload request, forcing to Direct Output" 294 : "Not linear PCM, forcing to Direct Output"); 295 flags = (audio_output_flags_t) 296 // FIXME why can't we allow direct AND fast? 297 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 298 } 299 // only allow deep buffering for music stream type 300 if (streamType != AUDIO_STREAM_MUSIC) { 301 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 302 } 303 304 mChannelMask = channelMask; 305 uint32_t channelCount = popcount(channelMask); 306 mChannelCount = channelCount; 307 308 if (audio_is_linear_pcm(format)) { 309 mFrameSize = channelCount * audio_bytes_per_sample(format); 310 mFrameSizeAF = channelCount * sizeof(int16_t); 311 } else { 312 mFrameSize = sizeof(uint8_t); 313 mFrameSizeAF = sizeof(uint8_t); 314 } 315 316 audio_io_handle_t output = AudioSystem::getOutput( 317 streamType, 318 sampleRate, format, channelMask, 319 flags, 320 offloadInfo); 321 322 if (output == 0) { 323 ALOGE("Could not get audio output for stream type %d", streamType); 324 return BAD_VALUE; 325 } 326 327 mVolume[LEFT] = 1.0f; 328 mVolume[RIGHT] = 1.0f; 329 mSendLevel = 0.0f; 330 // mFrameCount is initialized in createTrack_l 331 mReqFrameCount = frameCount; 332 mNotificationFramesReq = notificationFrames; 333 mNotificationFramesAct = 0; 334 mSessionId = sessionId; 335 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 336 mClientUid = IPCThreadState::self()->getCallingUid(); 337 } else { 338 mClientUid = uid; 339 } 340 mAuxEffectId = 0; 341 mFlags = flags; 342 mCbf = cbf; 343 344 if (cbf != NULL) { 345 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 346 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 347 } 348 349 // create the IAudioTrack 350 status = createTrack_l(streamType, 351 sampleRate, 352 format, 353 frameCount, 354 flags, 355 sharedBuffer, 356 output, 357 0 /*epoch*/); 358 359 if (status != NO_ERROR) { 360 if (mAudioTrackThread != 0) { 361 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 362 mAudioTrackThread->requestExitAndWait(); 363 mAudioTrackThread.clear(); 364 } 365 //Use of direct and offloaded output streams is ref counted by audio policy manager. 366 // As getOutput was called above and resulted in an output stream to be opened, 367 // we need to release it. 368 AudioSystem::releaseOutput(output); 369 return status; 370 } 371 372 mStatus = NO_ERROR; 373 mState = STATE_STOPPED; 374 mUserData = user; 375 mLoopPeriod = 0; 376 mMarkerPosition = 0; 377 mMarkerReached = false; 378 mNewPosition = 0; 379 mUpdatePeriod = 0; 380 AudioSystem::acquireAudioSessionId(mSessionId); 381 mSequence = 1; 382 mObservedSequence = mSequence; 383 mInUnderrun = false; 384 mOutput = output; 385 386 return NO_ERROR; 387} 388 389// ------------------------------------------------------------------------- 390 391status_t AudioTrack::start() 392{ 393 AutoMutex lock(mLock); 394 395 if (mState == STATE_ACTIVE) { 396 return INVALID_OPERATION; 397 } 398 399 mInUnderrun = true; 400 401 State previousState = mState; 402 if (previousState == STATE_PAUSED_STOPPING) { 403 mState = STATE_STOPPING; 404 } else { 405 mState = STATE_ACTIVE; 406 } 407 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 408 // reset current position as seen by client to 0 409 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 410 // force refresh of remaining frames by processAudioBuffer() as last 411 // write before stop could be partial. 412 mRefreshRemaining = true; 413 } 414 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 415 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 416 417 sp<AudioTrackThread> t = mAudioTrackThread; 418 if (t != 0) { 419 if (previousState == STATE_STOPPING) { 420 mProxy->interrupt(); 421 } else { 422 t->resume(); 423 } 424 } else { 425 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 426 get_sched_policy(0, &mPreviousSchedulingGroup); 427 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 428 } 429 430 status_t status = NO_ERROR; 431 if (!(flags & CBLK_INVALID)) { 432 status = mAudioTrack->start(); 433 if (status == DEAD_OBJECT) { 434 flags |= CBLK_INVALID; 435 } 436 } 437 if (flags & CBLK_INVALID) { 438 status = restoreTrack_l("start"); 439 } 440 441 if (status != NO_ERROR) { 442 ALOGE("start() status %d", status); 443 mState = previousState; 444 if (t != 0) { 445 if (previousState != STATE_STOPPING) { 446 t->pause(); 447 } 448 } else { 449 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 450 set_sched_policy(0, mPreviousSchedulingGroup); 451 } 452 } 453 454 return status; 455} 456 457void AudioTrack::stop() 458{ 459 AutoMutex lock(mLock); 460 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 461 return; 462 } 463 464 if (isOffloaded_l()) { 465 mState = STATE_STOPPING; 466 } else { 467 mState = STATE_STOPPED; 468 } 469 470 mProxy->interrupt(); 471 mAudioTrack->stop(); 472 // the playback head position will reset to 0, so if a marker is set, we need 473 // to activate it again 474 mMarkerReached = false; 475#if 0 476 // Force flush if a shared buffer is used otherwise audioflinger 477 // will not stop before end of buffer is reached. 478 // It may be needed to make sure that we stop playback, likely in case looping is on. 479 if (mSharedBuffer != 0) { 480 flush_l(); 481 } 482#endif 483 484 sp<AudioTrackThread> t = mAudioTrackThread; 485 if (t != 0) { 486 if (!isOffloaded_l()) { 487 t->pause(); 488 } 489 } else { 490 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 491 set_sched_policy(0, mPreviousSchedulingGroup); 492 } 493} 494 495bool AudioTrack::stopped() const 496{ 497 AutoMutex lock(mLock); 498 return mState != STATE_ACTIVE; 499} 500 501void AudioTrack::flush() 502{ 503 if (mSharedBuffer != 0) { 504 return; 505 } 506 AutoMutex lock(mLock); 507 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 508 return; 509 } 510 flush_l(); 511} 512 513void AudioTrack::flush_l() 514{ 515 ALOG_ASSERT(mState != STATE_ACTIVE); 516 517 // clear playback marker and periodic update counter 518 mMarkerPosition = 0; 519 mMarkerReached = false; 520 mUpdatePeriod = 0; 521 mRefreshRemaining = true; 522 523 mState = STATE_FLUSHED; 524 if (isOffloaded_l()) { 525 mProxy->interrupt(); 526 } 527 mProxy->flush(); 528 mAudioTrack->flush(); 529} 530 531void AudioTrack::pause() 532{ 533 AutoMutex lock(mLock); 534 if (mState == STATE_ACTIVE) { 535 mState = STATE_PAUSED; 536 } else if (mState == STATE_STOPPING) { 537 mState = STATE_PAUSED_STOPPING; 538 } else { 539 return; 540 } 541 mProxy->interrupt(); 542 mAudioTrack->pause(); 543} 544 545status_t AudioTrack::setVolume(float left, float right) 546{ 547 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 548 return BAD_VALUE; 549 } 550 551 AutoMutex lock(mLock); 552 mVolume[LEFT] = left; 553 mVolume[RIGHT] = right; 554 555 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 556 557 if (isOffloaded_l()) { 558 mAudioTrack->signal(); 559 } 560 return NO_ERROR; 561} 562 563status_t AudioTrack::setVolume(float volume) 564{ 565 return setVolume(volume, volume); 566} 567 568status_t AudioTrack::setAuxEffectSendLevel(float level) 569{ 570 if (level < 0.0f || level > 1.0f) { 571 return BAD_VALUE; 572 } 573 574 AutoMutex lock(mLock); 575 mSendLevel = level; 576 mProxy->setSendLevel(level); 577 578 return NO_ERROR; 579} 580 581void AudioTrack::getAuxEffectSendLevel(float* level) const 582{ 583 if (level != NULL) { 584 *level = mSendLevel; 585 } 586} 587 588status_t AudioTrack::setSampleRate(uint32_t rate) 589{ 590 if (mIsTimed || isOffloaded()) { 591 return INVALID_OPERATION; 592 } 593 594 uint32_t afSamplingRate; 595 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 596 return NO_INIT; 597 } 598 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 599 if (rate == 0 || rate > afSamplingRate*2 ) { 600 return BAD_VALUE; 601 } 602 603 AutoMutex lock(mLock); 604 mSampleRate = rate; 605 mProxy->setSampleRate(rate); 606 607 return NO_ERROR; 608} 609 610uint32_t AudioTrack::getSampleRate() const 611{ 612 if (mIsTimed) { 613 return 0; 614 } 615 616 AutoMutex lock(mLock); 617 618 // sample rate can be updated during playback by the offloaded decoder so we need to 619 // query the HAL and update if needed. 620// FIXME use Proxy return channel to update the rate from server and avoid polling here 621 if (isOffloaded_l()) { 622 if (mOutput != 0) { 623 uint32_t sampleRate = 0; 624 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 625 if (status == NO_ERROR) { 626 mSampleRate = sampleRate; 627 } 628 } 629 } 630 return mSampleRate; 631} 632 633status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 634{ 635 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 636 return INVALID_OPERATION; 637 } 638 639 if (loopCount == 0) { 640 ; 641 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 642 loopEnd - loopStart >= MIN_LOOP) { 643 ; 644 } else { 645 return BAD_VALUE; 646 } 647 648 AutoMutex lock(mLock); 649 // See setPosition() regarding setting parameters such as loop points or position while active 650 if (mState == STATE_ACTIVE) { 651 return INVALID_OPERATION; 652 } 653 setLoop_l(loopStart, loopEnd, loopCount); 654 return NO_ERROR; 655} 656 657void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 658{ 659 // FIXME If setting a loop also sets position to start of loop, then 660 // this is correct. Otherwise it should be removed. 661 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 662 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 663 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 664} 665 666status_t AudioTrack::setMarkerPosition(uint32_t marker) 667{ 668 // The only purpose of setting marker position is to get a callback 669 if (mCbf == NULL || isOffloaded()) { 670 return INVALID_OPERATION; 671 } 672 673 AutoMutex lock(mLock); 674 mMarkerPosition = marker; 675 mMarkerReached = false; 676 677 return NO_ERROR; 678} 679 680status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 681{ 682 if (isOffloaded()) { 683 return INVALID_OPERATION; 684 } 685 if (marker == NULL) { 686 return BAD_VALUE; 687 } 688 689 AutoMutex lock(mLock); 690 *marker = mMarkerPosition; 691 692 return NO_ERROR; 693} 694 695status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 696{ 697 // The only purpose of setting position update period is to get a callback 698 if (mCbf == NULL || isOffloaded()) { 699 return INVALID_OPERATION; 700 } 701 702 AutoMutex lock(mLock); 703 mNewPosition = mProxy->getPosition() + updatePeriod; 704 mUpdatePeriod = updatePeriod; 705 return NO_ERROR; 706} 707 708status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 709{ 710 if (isOffloaded()) { 711 return INVALID_OPERATION; 712 } 713 if (updatePeriod == NULL) { 714 return BAD_VALUE; 715 } 716 717 AutoMutex lock(mLock); 718 *updatePeriod = mUpdatePeriod; 719 720 return NO_ERROR; 721} 722 723status_t AudioTrack::setPosition(uint32_t position) 724{ 725 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 726 return INVALID_OPERATION; 727 } 728 if (position > mFrameCount) { 729 return BAD_VALUE; 730 } 731 732 AutoMutex lock(mLock); 733 // Currently we require that the player is inactive before setting parameters such as position 734 // or loop points. Otherwise, there could be a race condition: the application could read the 735 // current position, compute a new position or loop parameters, and then set that position or 736 // loop parameters but it would do the "wrong" thing since the position has continued to advance 737 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 738 // to specify how it wants to handle such scenarios. 739 if (mState == STATE_ACTIVE) { 740 return INVALID_OPERATION; 741 } 742 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 743 mLoopPeriod = 0; 744 // FIXME Check whether loops and setting position are incompatible in old code. 745 // If we use setLoop for both purposes we lose the capability to set the position while looping. 746 mStaticProxy->setLoop(position, mFrameCount, 0); 747 748 return NO_ERROR; 749} 750 751status_t AudioTrack::getPosition(uint32_t *position) const 752{ 753 if (position == NULL) { 754 return BAD_VALUE; 755 } 756 757 AutoMutex lock(mLock); 758 if (isOffloaded_l()) { 759 uint32_t dspFrames = 0; 760 761 if (mOutput != 0) { 762 uint32_t halFrames; 763 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 764 } 765 *position = dspFrames; 766 } else { 767 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 768 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 769 mProxy->getPosition(); 770 } 771 return NO_ERROR; 772} 773 774status_t AudioTrack::getBufferPosition(size_t *position) 775{ 776 if (mSharedBuffer == 0 || mIsTimed) { 777 return INVALID_OPERATION; 778 } 779 if (position == NULL) { 780 return BAD_VALUE; 781 } 782 783 AutoMutex lock(mLock); 784 *position = mStaticProxy->getBufferPosition(); 785 return NO_ERROR; 786} 787 788status_t AudioTrack::reload() 789{ 790 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 791 return INVALID_OPERATION; 792 } 793 794 AutoMutex lock(mLock); 795 // See setPosition() regarding setting parameters such as loop points or position while active 796 if (mState == STATE_ACTIVE) { 797 return INVALID_OPERATION; 798 } 799 mNewPosition = mUpdatePeriod; 800 mLoopPeriod = 0; 801 // FIXME The new code cannot reload while keeping a loop specified. 802 // Need to check how the old code handled this, and whether it's a significant change. 803 mStaticProxy->setLoop(0, mFrameCount, 0); 804 return NO_ERROR; 805} 806 807audio_io_handle_t AudioTrack::getOutput() 808{ 809 AutoMutex lock(mLock); 810 return mOutput; 811} 812 813// must be called with mLock held 814audio_io_handle_t AudioTrack::getOutput_l() 815{ 816 if (mOutput) { 817 return mOutput; 818 } else { 819 return AudioSystem::getOutput(mStreamType, 820 mSampleRate, mFormat, mChannelMask, mFlags); 821 } 822} 823 824status_t AudioTrack::attachAuxEffect(int effectId) 825{ 826 AutoMutex lock(mLock); 827 status_t status = mAudioTrack->attachAuxEffect(effectId); 828 if (status == NO_ERROR) { 829 mAuxEffectId = effectId; 830 } 831 return status; 832} 833 834// ------------------------------------------------------------------------- 835 836// must be called with mLock held 837status_t AudioTrack::createTrack_l( 838 audio_stream_type_t streamType, 839 uint32_t sampleRate, 840 audio_format_t format, 841 size_t frameCount, 842 audio_output_flags_t flags, 843 const sp<IMemory>& sharedBuffer, 844 audio_io_handle_t output, 845 size_t epoch) 846{ 847 status_t status; 848 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 849 if (audioFlinger == 0) { 850 ALOGE("Could not get audioflinger"); 851 return NO_INIT; 852 } 853 854 // Not all of these values are needed under all conditions, but it is easier to get them all 855 856 uint32_t afLatency; 857 status = AudioSystem::getLatency(output, streamType, &afLatency); 858 if (status != NO_ERROR) { 859 ALOGE("getLatency(%d) failed status %d", output, status); 860 return NO_INIT; 861 } 862 863 size_t afFrameCount; 864 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 865 if (status != NO_ERROR) { 866 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 867 return NO_INIT; 868 } 869 870 uint32_t afSampleRate; 871 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 872 if (status != NO_ERROR) { 873 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 874 return NO_INIT; 875 } 876 877 // Client decides whether the track is TIMED (see below), but can only express a preference 878 // for FAST. Server will perform additional tests. 879 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 880 // either of these use cases: 881 // use case 1: shared buffer 882 (sharedBuffer != 0) || 883 // use case 2: callback handler 884 (mCbf != NULL))) { 885 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 886 // once denied, do not request again if IAudioTrack is re-created 887 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 888 mFlags = flags; 889 } 890 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 891 892 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 893 // n = 1 fast track with single buffering; nBuffering is ignored 894 // n = 2 fast track with double buffering 895 // n = 2 normal track, no sample rate conversion 896 // n = 3 normal track, with sample rate conversion 897 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 898 // n > 3 very high latency or very small notification interval; nBuffering is ignored 899 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 900 901 mNotificationFramesAct = mNotificationFramesReq; 902 903 if (!audio_is_linear_pcm(format)) { 904 905 if (sharedBuffer != 0) { 906 // Same comment as below about ignoring frameCount parameter for set() 907 frameCount = sharedBuffer->size(); 908 } else if (frameCount == 0) { 909 frameCount = afFrameCount; 910 } 911 if (mNotificationFramesAct != frameCount) { 912 mNotificationFramesAct = frameCount; 913 } 914 } else if (sharedBuffer != 0) { 915 916 // Ensure that buffer alignment matches channel count 917 // 8-bit data in shared memory is not currently supported by AudioFlinger 918 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 919 if (mChannelCount > 1) { 920 // More than 2 channels does not require stronger alignment than stereo 921 alignment <<= 1; 922 } 923 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 924 ALOGE("Invalid buffer alignment: address %p, channel count %u", 925 sharedBuffer->pointer(), mChannelCount); 926 return BAD_VALUE; 927 } 928 929 // When initializing a shared buffer AudioTrack via constructors, 930 // there's no frameCount parameter. 931 // But when initializing a shared buffer AudioTrack via set(), 932 // there _is_ a frameCount parameter. We silently ignore it. 933 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 934 935 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 936 937 // FIXME move these calculations and associated checks to server 938 939 // Ensure that buffer depth covers at least audio hardware latency 940 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 941 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 942 afFrameCount, minBufCount, afSampleRate, afLatency); 943 if (minBufCount <= nBuffering) { 944 minBufCount = nBuffering; 945 } 946 947 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 948 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 949 ", afLatency=%d", 950 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 951 952 if (frameCount == 0) { 953 frameCount = minFrameCount; 954 } else if (frameCount < minFrameCount) { 955 // not ALOGW because it happens all the time when playing key clicks over A2DP 956 ALOGV("Minimum buffer size corrected from %d to %d", 957 frameCount, minFrameCount); 958 frameCount = minFrameCount; 959 } 960 // Make sure that application is notified with sufficient margin before underrun 961 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 962 mNotificationFramesAct = frameCount/nBuffering; 963 } 964 965 } else { 966 // For fast tracks, the frame count calculations and checks are done by server 967 } 968 969 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 970 if (mIsTimed) { 971 trackFlags |= IAudioFlinger::TRACK_TIMED; 972 } 973 974 pid_t tid = -1; 975 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 976 trackFlags |= IAudioFlinger::TRACK_FAST; 977 if (mAudioTrackThread != 0) { 978 tid = mAudioTrackThread->getTid(); 979 } 980 } 981 982 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 983 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 984 } 985 986 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 987 sampleRate, 988 // AudioFlinger only sees 16-bit PCM 989 format == AUDIO_FORMAT_PCM_8_BIT ? 990 AUDIO_FORMAT_PCM_16_BIT : format, 991 mChannelMask, 992 frameCount, 993 &trackFlags, 994 sharedBuffer, 995 output, 996 tid, 997 &mSessionId, 998 mName, 999 mClientUid, 1000 &status); 1001 1002 if (track == 0) { 1003 ALOGE("AudioFlinger could not create track, status: %d", status); 1004 return status; 1005 } 1006 sp<IMemory> iMem = track->getCblk(); 1007 if (iMem == 0) { 1008 ALOGE("Could not get control block"); 1009 return NO_INIT; 1010 } 1011 void *iMemPointer = iMem->pointer(); 1012 if (iMemPointer == NULL) { 1013 ALOGE("Could not get control block pointer"); 1014 return NO_INIT; 1015 } 1016 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1017 if (mAudioTrack != 0) { 1018 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1019 mDeathNotifier.clear(); 1020 } 1021 mAudioTrack = track; 1022 mCblkMemory = iMem; 1023 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1024 mCblk = cblk; 1025 size_t temp = cblk->frameCount_; 1026 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1027 // In current design, AudioTrack client checks and ensures frame count validity before 1028 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1029 // for fast track as it uses a special method of assigning frame count. 1030 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1031 } 1032 frameCount = temp; 1033 mAwaitBoost = false; 1034 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1035 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1036 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1037 mAwaitBoost = true; 1038 if (sharedBuffer == 0) { 1039 // Theoretically double-buffering is not required for fast tracks, 1040 // due to tighter scheduling. But in practice, to accommodate kernels with 1041 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1042 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1043 mNotificationFramesAct = frameCount/nBuffering; 1044 } 1045 } 1046 } else { 1047 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1048 // once denied, do not request again if IAudioTrack is re-created 1049 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1050 mFlags = flags; 1051 if (sharedBuffer == 0) { 1052 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1053 mNotificationFramesAct = frameCount/nBuffering; 1054 } 1055 } 1056 } 1057 } 1058 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1059 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1060 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1061 } else { 1062 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1063 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1064 mFlags = flags; 1065 return NO_INIT; 1066 } 1067 } 1068 1069 mRefreshRemaining = true; 1070 1071 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1072 // is the value of pointer() for the shared buffer, otherwise buffers points 1073 // immediately after the control block. This address is for the mapping within client 1074 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1075 void* buffers; 1076 if (sharedBuffer == 0) { 1077 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1078 } else { 1079 buffers = sharedBuffer->pointer(); 1080 } 1081 1082 mAudioTrack->attachAuxEffect(mAuxEffectId); 1083 // FIXME don't believe this lie 1084 mLatency = afLatency + (1000*frameCount) / sampleRate; 1085 mFrameCount = frameCount; 1086 // If IAudioTrack is re-created, don't let the requested frameCount 1087 // decrease. This can confuse clients that cache frameCount(). 1088 if (frameCount > mReqFrameCount) { 1089 mReqFrameCount = frameCount; 1090 } 1091 1092 // update proxy 1093 if (sharedBuffer == 0) { 1094 mStaticProxy.clear(); 1095 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1096 } else { 1097 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1098 mProxy = mStaticProxy; 1099 } 1100 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1101 uint16_t(mVolume[LEFT] * 0x1000)); 1102 mProxy->setSendLevel(mSendLevel); 1103 mProxy->setSampleRate(mSampleRate); 1104 mProxy->setEpoch(epoch); 1105 mProxy->setMinimum(mNotificationFramesAct); 1106 1107 mDeathNotifier = new DeathNotifier(this); 1108 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1109 1110 return NO_ERROR; 1111} 1112 1113status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1114{ 1115 if (audioBuffer == NULL) { 1116 return BAD_VALUE; 1117 } 1118 if (mTransfer != TRANSFER_OBTAIN) { 1119 audioBuffer->frameCount = 0; 1120 audioBuffer->size = 0; 1121 audioBuffer->raw = NULL; 1122 return INVALID_OPERATION; 1123 } 1124 1125 const struct timespec *requested; 1126 if (waitCount == -1) { 1127 requested = &ClientProxy::kForever; 1128 } else if (waitCount == 0) { 1129 requested = &ClientProxy::kNonBlocking; 1130 } else if (waitCount > 0) { 1131 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1132 struct timespec timeout; 1133 timeout.tv_sec = ms / 1000; 1134 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1135 requested = &timeout; 1136 } else { 1137 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1138 requested = NULL; 1139 } 1140 return obtainBuffer(audioBuffer, requested); 1141} 1142 1143status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1144 struct timespec *elapsed, size_t *nonContig) 1145{ 1146 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1147 uint32_t oldSequence = 0; 1148 uint32_t newSequence; 1149 1150 Proxy::Buffer buffer; 1151 status_t status = NO_ERROR; 1152 1153 static const int32_t kMaxTries = 5; 1154 int32_t tryCounter = kMaxTries; 1155 1156 do { 1157 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1158 // keep them from going away if another thread re-creates the track during obtainBuffer() 1159 sp<AudioTrackClientProxy> proxy; 1160 sp<IMemory> iMem; 1161 1162 { // start of lock scope 1163 AutoMutex lock(mLock); 1164 1165 newSequence = mSequence; 1166 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1167 if (status == DEAD_OBJECT) { 1168 // re-create track, unless someone else has already done so 1169 if (newSequence == oldSequence) { 1170 status = restoreTrack_l("obtainBuffer"); 1171 if (status != NO_ERROR) { 1172 buffer.mFrameCount = 0; 1173 buffer.mRaw = NULL; 1174 buffer.mNonContig = 0; 1175 break; 1176 } 1177 } 1178 } 1179 oldSequence = newSequence; 1180 1181 // Keep the extra references 1182 proxy = mProxy; 1183 iMem = mCblkMemory; 1184 1185 if (mState == STATE_STOPPING) { 1186 status = -EINTR; 1187 buffer.mFrameCount = 0; 1188 buffer.mRaw = NULL; 1189 buffer.mNonContig = 0; 1190 break; 1191 } 1192 1193 // Non-blocking if track is stopped or paused 1194 if (mState != STATE_ACTIVE) { 1195 requested = &ClientProxy::kNonBlocking; 1196 } 1197 1198 } // end of lock scope 1199 1200 buffer.mFrameCount = audioBuffer->frameCount; 1201 // FIXME starts the requested timeout and elapsed over from scratch 1202 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1203 1204 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1205 1206 audioBuffer->frameCount = buffer.mFrameCount; 1207 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1208 audioBuffer->raw = buffer.mRaw; 1209 if (nonContig != NULL) { 1210 *nonContig = buffer.mNonContig; 1211 } 1212 return status; 1213} 1214 1215void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1216{ 1217 if (mTransfer == TRANSFER_SHARED) { 1218 return; 1219 } 1220 1221 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1222 if (stepCount == 0) { 1223 return; 1224 } 1225 1226 Proxy::Buffer buffer; 1227 buffer.mFrameCount = stepCount; 1228 buffer.mRaw = audioBuffer->raw; 1229 1230 AutoMutex lock(mLock); 1231 mInUnderrun = false; 1232 mProxy->releaseBuffer(&buffer); 1233 1234 // restart track if it was disabled by audioflinger due to previous underrun 1235 if (mState == STATE_ACTIVE) { 1236 audio_track_cblk_t* cblk = mCblk; 1237 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1238 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1239 this, mName.string()); 1240 // FIXME ignoring status 1241 mAudioTrack->start(); 1242 } 1243 } 1244} 1245 1246// ------------------------------------------------------------------------- 1247 1248ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1249{ 1250 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1251 return INVALID_OPERATION; 1252 } 1253 1254 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1255 // Sanity-check: user is most-likely passing an error code, and it would 1256 // make the return value ambiguous (actualSize vs error). 1257 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1258 return BAD_VALUE; 1259 } 1260 1261 size_t written = 0; 1262 Buffer audioBuffer; 1263 1264 while (userSize >= mFrameSize) { 1265 audioBuffer.frameCount = userSize / mFrameSize; 1266 1267 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1268 if (err < 0) { 1269 if (written > 0) { 1270 break; 1271 } 1272 return ssize_t(err); 1273 } 1274 1275 size_t toWrite; 1276 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1277 // Divide capacity by 2 to take expansion into account 1278 toWrite = audioBuffer.size >> 1; 1279 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1280 } else { 1281 toWrite = audioBuffer.size; 1282 memcpy(audioBuffer.i8, buffer, toWrite); 1283 } 1284 buffer = ((const char *) buffer) + toWrite; 1285 userSize -= toWrite; 1286 written += toWrite; 1287 1288 releaseBuffer(&audioBuffer); 1289 } 1290 1291 return written; 1292} 1293 1294// ------------------------------------------------------------------------- 1295 1296TimedAudioTrack::TimedAudioTrack() { 1297 mIsTimed = true; 1298} 1299 1300status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1301{ 1302 AutoMutex lock(mLock); 1303 status_t result = UNKNOWN_ERROR; 1304 1305#if 1 1306 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1307 // while we are accessing the cblk 1308 sp<IAudioTrack> audioTrack = mAudioTrack; 1309 sp<IMemory> iMem = mCblkMemory; 1310#endif 1311 1312 // If the track is not invalid already, try to allocate a buffer. alloc 1313 // fails indicating that the server is dead, flag the track as invalid so 1314 // we can attempt to restore in just a bit. 1315 audio_track_cblk_t* cblk = mCblk; 1316 if (!(cblk->mFlags & CBLK_INVALID)) { 1317 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1318 if (result == DEAD_OBJECT) { 1319 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1320 } 1321 } 1322 1323 // If the track is invalid at this point, attempt to restore it. and try the 1324 // allocation one more time. 1325 if (cblk->mFlags & CBLK_INVALID) { 1326 result = restoreTrack_l("allocateTimedBuffer"); 1327 1328 if (result == NO_ERROR) { 1329 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1330 } 1331 } 1332 1333 return result; 1334} 1335 1336status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1337 int64_t pts) 1338{ 1339 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1340 { 1341 AutoMutex lock(mLock); 1342 audio_track_cblk_t* cblk = mCblk; 1343 // restart track if it was disabled by audioflinger due to previous underrun 1344 if (buffer->size() != 0 && status == NO_ERROR && 1345 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1346 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1347 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1348 // FIXME ignoring status 1349 mAudioTrack->start(); 1350 } 1351 } 1352 return status; 1353} 1354 1355status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1356 TargetTimeline target) 1357{ 1358 return mAudioTrack->setMediaTimeTransform(xform, target); 1359} 1360 1361// ------------------------------------------------------------------------- 1362 1363nsecs_t AudioTrack::processAudioBuffer() 1364{ 1365 // Currently the AudioTrack thread is not created if there are no callbacks. 1366 // Would it ever make sense to run the thread, even without callbacks? 1367 // If so, then replace this by checks at each use for mCbf != NULL. 1368 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1369 1370 mLock.lock(); 1371 if (mAwaitBoost) { 1372 mAwaitBoost = false; 1373 mLock.unlock(); 1374 static const int32_t kMaxTries = 5; 1375 int32_t tryCounter = kMaxTries; 1376 uint32_t pollUs = 10000; 1377 do { 1378 int policy = sched_getscheduler(0); 1379 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1380 break; 1381 } 1382 usleep(pollUs); 1383 pollUs <<= 1; 1384 } while (tryCounter-- > 0); 1385 if (tryCounter < 0) { 1386 ALOGE("did not receive expected priority boost on time"); 1387 } 1388 // Run again immediately 1389 return 0; 1390 } 1391 1392 // Can only reference mCblk while locked 1393 int32_t flags = android_atomic_and( 1394 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1395 1396 // Check for track invalidation 1397 if (flags & CBLK_INVALID) { 1398 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1399 // AudioSystem cache. We should not exit here but after calling the callback so 1400 // that the upper layers can recreate the track 1401 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1402 status_t status = restoreTrack_l("processAudioBuffer"); 1403 mLock.unlock(); 1404 // Run again immediately, but with a new IAudioTrack 1405 return 0; 1406 } 1407 } 1408 1409 bool waitStreamEnd = mState == STATE_STOPPING; 1410 bool active = mState == STATE_ACTIVE; 1411 1412 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1413 bool newUnderrun = false; 1414 if (flags & CBLK_UNDERRUN) { 1415#if 0 1416 // Currently in shared buffer mode, when the server reaches the end of buffer, 1417 // the track stays active in continuous underrun state. It's up to the application 1418 // to pause or stop the track, or set the position to a new offset within buffer. 1419 // This was some experimental code to auto-pause on underrun. Keeping it here 1420 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1421 if (mTransfer == TRANSFER_SHARED) { 1422 mState = STATE_PAUSED; 1423 active = false; 1424 } 1425#endif 1426 if (!mInUnderrun) { 1427 mInUnderrun = true; 1428 newUnderrun = true; 1429 } 1430 } 1431 1432 // Get current position of server 1433 size_t position = mProxy->getPosition(); 1434 1435 // Manage marker callback 1436 bool markerReached = false; 1437 size_t markerPosition = mMarkerPosition; 1438 // FIXME fails for wraparound, need 64 bits 1439 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1440 mMarkerReached = markerReached = true; 1441 } 1442 1443 // Determine number of new position callback(s) that will be needed, while locked 1444 size_t newPosCount = 0; 1445 size_t newPosition = mNewPosition; 1446 size_t updatePeriod = mUpdatePeriod; 1447 // FIXME fails for wraparound, need 64 bits 1448 if (updatePeriod > 0 && position >= newPosition) { 1449 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1450 mNewPosition += updatePeriod * newPosCount; 1451 } 1452 1453 // Cache other fields that will be needed soon 1454 uint32_t loopPeriod = mLoopPeriod; 1455 uint32_t sampleRate = mSampleRate; 1456 size_t notificationFrames = mNotificationFramesAct; 1457 if (mRefreshRemaining) { 1458 mRefreshRemaining = false; 1459 mRemainingFrames = notificationFrames; 1460 mRetryOnPartialBuffer = false; 1461 } 1462 size_t misalignment = mProxy->getMisalignment(); 1463 uint32_t sequence = mSequence; 1464 1465 // These fields don't need to be cached, because they are assigned only by set(): 1466 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1467 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1468 1469 mLock.unlock(); 1470 1471 if (waitStreamEnd) { 1472 AutoMutex lock(mLock); 1473 1474 sp<AudioTrackClientProxy> proxy = mProxy; 1475 sp<IMemory> iMem = mCblkMemory; 1476 1477 struct timespec timeout; 1478 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1479 timeout.tv_nsec = 0; 1480 1481 mLock.unlock(); 1482 status_t status = mProxy->waitStreamEndDone(&timeout); 1483 mLock.lock(); 1484 switch (status) { 1485 case NO_ERROR: 1486 case DEAD_OBJECT: 1487 case TIMED_OUT: 1488 mLock.unlock(); 1489 mCbf(EVENT_STREAM_END, mUserData, NULL); 1490 mLock.lock(); 1491 if (mState == STATE_STOPPING) { 1492 mState = STATE_STOPPED; 1493 if (status != DEAD_OBJECT) { 1494 return NS_INACTIVE; 1495 } 1496 } 1497 return 0; 1498 default: 1499 return 0; 1500 } 1501 } 1502 1503 // perform callbacks while unlocked 1504 if (newUnderrun) { 1505 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1506 } 1507 // FIXME we will miss loops if loop cycle was signaled several times since last call 1508 // to processAudioBuffer() 1509 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1510 mCbf(EVENT_LOOP_END, mUserData, NULL); 1511 } 1512 if (flags & CBLK_BUFFER_END) { 1513 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1514 } 1515 if (markerReached) { 1516 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1517 } 1518 while (newPosCount > 0) { 1519 size_t temp = newPosition; 1520 mCbf(EVENT_NEW_POS, mUserData, &temp); 1521 newPosition += updatePeriod; 1522 newPosCount--; 1523 } 1524 1525 if (mObservedSequence != sequence) { 1526 mObservedSequence = sequence; 1527 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1528 // for offloaded tracks, just wait for the upper layers to recreate the track 1529 if (isOffloaded()) { 1530 return NS_INACTIVE; 1531 } 1532 } 1533 1534 // if inactive, then don't run me again until re-started 1535 if (!active) { 1536 return NS_INACTIVE; 1537 } 1538 1539 // Compute the estimated time until the next timed event (position, markers, loops) 1540 // FIXME only for non-compressed audio 1541 uint32_t minFrames = ~0; 1542 if (!markerReached && position < markerPosition) { 1543 minFrames = markerPosition - position; 1544 } 1545 if (loopPeriod > 0 && loopPeriod < minFrames) { 1546 minFrames = loopPeriod; 1547 } 1548 if (updatePeriod > 0 && updatePeriod < minFrames) { 1549 minFrames = updatePeriod; 1550 } 1551 1552 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1553 static const uint32_t kPoll = 0; 1554 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1555 minFrames = kPoll * notificationFrames; 1556 } 1557 1558 // Convert frame units to time units 1559 nsecs_t ns = NS_WHENEVER; 1560 if (minFrames != (uint32_t) ~0) { 1561 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1562 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1563 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1564 } 1565 1566 // If not supplying data by EVENT_MORE_DATA, then we're done 1567 if (mTransfer != TRANSFER_CALLBACK) { 1568 return ns; 1569 } 1570 1571 struct timespec timeout; 1572 const struct timespec *requested = &ClientProxy::kForever; 1573 if (ns != NS_WHENEVER) { 1574 timeout.tv_sec = ns / 1000000000LL; 1575 timeout.tv_nsec = ns % 1000000000LL; 1576 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1577 requested = &timeout; 1578 } 1579 1580 while (mRemainingFrames > 0) { 1581 1582 Buffer audioBuffer; 1583 audioBuffer.frameCount = mRemainingFrames; 1584 size_t nonContig; 1585 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1586 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1587 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1588 requested = &ClientProxy::kNonBlocking; 1589 size_t avail = audioBuffer.frameCount + nonContig; 1590 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1591 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1592 if (err != NO_ERROR) { 1593 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1594 (isOffloaded() && (err == DEAD_OBJECT))) { 1595 return 0; 1596 } 1597 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1598 return NS_NEVER; 1599 } 1600 1601 if (mRetryOnPartialBuffer && !isOffloaded()) { 1602 mRetryOnPartialBuffer = false; 1603 if (avail < mRemainingFrames) { 1604 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1605 if (ns < 0 || myns < ns) { 1606 ns = myns; 1607 } 1608 return ns; 1609 } 1610 } 1611 1612 // Divide buffer size by 2 to take into account the expansion 1613 // due to 8 to 16 bit conversion: the callback must fill only half 1614 // of the destination buffer 1615 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1616 audioBuffer.size >>= 1; 1617 } 1618 1619 size_t reqSize = audioBuffer.size; 1620 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1621 size_t writtenSize = audioBuffer.size; 1622 size_t writtenFrames = writtenSize / mFrameSize; 1623 1624 // Sanity check on returned size 1625 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1626 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1627 reqSize, (int) writtenSize); 1628 return NS_NEVER; 1629 } 1630 1631 if (writtenSize == 0) { 1632 // The callback is done filling buffers 1633 // Keep this thread going to handle timed events and 1634 // still try to get more data in intervals of WAIT_PERIOD_MS 1635 // but don't just loop and block the CPU, so wait 1636 return WAIT_PERIOD_MS * 1000000LL; 1637 } 1638 1639 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1640 // 8 to 16 bit conversion, note that source and destination are the same address 1641 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1642 audioBuffer.size <<= 1; 1643 } 1644 1645 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1646 audioBuffer.frameCount = releasedFrames; 1647 mRemainingFrames -= releasedFrames; 1648 if (misalignment >= releasedFrames) { 1649 misalignment -= releasedFrames; 1650 } else { 1651 misalignment = 0; 1652 } 1653 1654 releaseBuffer(&audioBuffer); 1655 1656 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1657 // if callback doesn't like to accept the full chunk 1658 if (writtenSize < reqSize) { 1659 continue; 1660 } 1661 1662 // There could be enough non-contiguous frames available to satisfy the remaining request 1663 if (mRemainingFrames <= nonContig) { 1664 continue; 1665 } 1666 1667#if 0 1668 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1669 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1670 // that total to a sum == notificationFrames. 1671 if (0 < misalignment && misalignment <= mRemainingFrames) { 1672 mRemainingFrames = misalignment; 1673 return (mRemainingFrames * 1100000000LL) / sampleRate; 1674 } 1675#endif 1676 1677 } 1678 mRemainingFrames = notificationFrames; 1679 mRetryOnPartialBuffer = true; 1680 1681 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1682 return 0; 1683} 1684 1685status_t AudioTrack::restoreTrack_l(const char *from) 1686{ 1687 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1688 isOffloaded_l() ? "Offloaded" : "PCM", from); 1689 ++mSequence; 1690 status_t result; 1691 1692 // refresh the audio configuration cache in this process to make sure we get new 1693 // output parameters in getOutput_l() and createTrack_l() 1694 AudioSystem::clearAudioConfigCache(); 1695 1696 if (isOffloaded_l()) { 1697 // FIXME re-creation of offloaded tracks is not yet implemented 1698 return DEAD_OBJECT; 1699 } 1700 1701 // force new output query from audio policy manager; 1702 mOutput = 0; 1703 audio_io_handle_t output = getOutput_l(); 1704 1705 // if the new IAudioTrack is created, createTrack_l() will modify the 1706 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1707 // It will also delete the strong references on previous IAudioTrack and IMemory 1708 1709 // take the frames that will be lost by track recreation into account in saved position 1710 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1711 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1712 result = createTrack_l(mStreamType, 1713 mSampleRate, 1714 mFormat, 1715 mReqFrameCount, // so that frame count never goes down 1716 mFlags, 1717 mSharedBuffer, 1718 output, 1719 position /*epoch*/); 1720 1721 if (result == NO_ERROR) { 1722 // continue playback from last known position, but 1723 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1724 if (mStaticProxy != NULL) { 1725 mLoopPeriod = 0; 1726 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1727 } 1728 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1729 // track destruction have been played? This is critical for SoundPool implementation 1730 // This must be broken, and needs to be tested/debugged. 1731#if 0 1732 // restore write index and set other indexes to reflect empty buffer status 1733 if (!strcmp(from, "start")) { 1734 // Make sure that a client relying on callback events indicating underrun or 1735 // the actual amount of audio frames played (e.g SoundPool) receives them. 1736 if (mSharedBuffer == 0) { 1737 // restart playback even if buffer is not completely filled. 1738 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1739 } 1740 } 1741#endif 1742 if (mState == STATE_ACTIVE) { 1743 result = mAudioTrack->start(); 1744 } 1745 } 1746 if (result != NO_ERROR) { 1747 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1748 // As getOutput was called above and resulted in an output stream to be opened, 1749 // we need to release it. 1750 AudioSystem::releaseOutput(output); 1751 ALOGW("restoreTrack_l() failed status %d", result); 1752 mState = STATE_STOPPED; 1753 } 1754 1755 return result; 1756} 1757 1758status_t AudioTrack::setParameters(const String8& keyValuePairs) 1759{ 1760 AutoMutex lock(mLock); 1761 return mAudioTrack->setParameters(keyValuePairs); 1762} 1763 1764status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1765{ 1766 AutoMutex lock(mLock); 1767 // FIXME not implemented for fast tracks; should use proxy and SSQ 1768 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1769 return INVALID_OPERATION; 1770 } 1771 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1772 return INVALID_OPERATION; 1773 } 1774 status_t status = mAudioTrack->getTimestamp(timestamp); 1775 if (status == NO_ERROR) { 1776 timestamp.mPosition += mProxy->getEpoch(); 1777 } 1778 return status; 1779} 1780 1781String8 AudioTrack::getParameters(const String8& keys) 1782{ 1783 audio_io_handle_t output = getOutput(); 1784 if (output != 0) { 1785 return AudioSystem::getParameters(output, keys); 1786 } else { 1787 return String8::empty(); 1788 } 1789} 1790 1791bool AudioTrack::isOffloaded() const 1792{ 1793 AutoMutex lock(mLock); 1794 return isOffloaded_l(); 1795} 1796 1797status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1798{ 1799 1800 const size_t SIZE = 256; 1801 char buffer[SIZE]; 1802 String8 result; 1803 1804 result.append(" AudioTrack::dump\n"); 1805 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1806 mVolume[0], mVolume[1]); 1807 result.append(buffer); 1808 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1809 mChannelCount, mFrameCount); 1810 result.append(buffer); 1811 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1812 result.append(buffer); 1813 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1814 result.append(buffer); 1815 ::write(fd, result.string(), result.size()); 1816 return NO_ERROR; 1817} 1818 1819uint32_t AudioTrack::getUnderrunFrames() const 1820{ 1821 AutoMutex lock(mLock); 1822 return mProxy->getUnderrunFrames(); 1823} 1824 1825// ========================================================================= 1826 1827void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1828{ 1829 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1830 if (audioTrack != 0) { 1831 AutoMutex lock(audioTrack->mLock); 1832 audioTrack->mProxy->binderDied(); 1833 } 1834} 1835 1836// ========================================================================= 1837 1838AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1839 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1840 mIgnoreNextPausedInt(false) 1841{ 1842} 1843 1844AudioTrack::AudioTrackThread::~AudioTrackThread() 1845{ 1846} 1847 1848bool AudioTrack::AudioTrackThread::threadLoop() 1849{ 1850 { 1851 AutoMutex _l(mMyLock); 1852 if (mPaused) { 1853 mMyCond.wait(mMyLock); 1854 // caller will check for exitPending() 1855 return true; 1856 } 1857 if (mIgnoreNextPausedInt) { 1858 mIgnoreNextPausedInt = false; 1859 mPausedInt = false; 1860 } 1861 if (mPausedInt) { 1862 if (mPausedNs > 0) { 1863 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1864 } else { 1865 mMyCond.wait(mMyLock); 1866 } 1867 mPausedInt = false; 1868 return true; 1869 } 1870 } 1871 nsecs_t ns = mReceiver.processAudioBuffer(); 1872 switch (ns) { 1873 case 0: 1874 return true; 1875 case NS_INACTIVE: 1876 pauseInternal(); 1877 return true; 1878 case NS_NEVER: 1879 return false; 1880 case NS_WHENEVER: 1881 // FIXME increase poll interval, or make event-driven 1882 ns = 1000000000LL; 1883 // fall through 1884 default: 1885 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1886 pauseInternal(ns); 1887 return true; 1888 } 1889} 1890 1891void AudioTrack::AudioTrackThread::requestExit() 1892{ 1893 // must be in this order to avoid a race condition 1894 Thread::requestExit(); 1895 resume(); 1896} 1897 1898void AudioTrack::AudioTrackThread::pause() 1899{ 1900 AutoMutex _l(mMyLock); 1901 mPaused = true; 1902} 1903 1904void AudioTrack::AudioTrackThread::resume() 1905{ 1906 AutoMutex _l(mMyLock); 1907 mIgnoreNextPausedInt = true; 1908 if (mPaused || mPausedInt) { 1909 mPaused = false; 1910 mPausedInt = false; 1911 mMyCond.signal(); 1912 } 1913} 1914 1915void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1916{ 1917 AutoMutex _l(mMyLock); 1918 mPausedInt = true; 1919 mPausedNs = ns; 1920} 1921 1922}; // namespace android 1923