AudioTrack.cpp revision a1ebc3b03d4dca534374c19e3c4f32ee687942e3
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31#define WAIT_STREAM_END_TIMEOUT_SEC 120 32 33 34namespace android { 35// --------------------------------------------------------------------------- 36 37// static 38status_t AudioTrack::getMinFrameCount( 39 size_t* frameCount, 40 audio_stream_type_t streamType, 41 uint32_t sampleRate) 42{ 43 if (frameCount == NULL) { 44 return BAD_VALUE; 45 } 46 47 // FIXME merge with similar code in createTrack_l(), except we're missing 48 // some information here that is available in createTrack_l(): 49 // audio_io_handle_t output 50 // audio_format_t format 51 // audio_channel_mask_t channelMask 52 // audio_output_flags_t flags 53 uint32_t afSampleRate; 54 status_t status; 55 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 56 if (status != NO_ERROR) { 57 ALOGE("Unable to query output sample rate for stream type %d; status %d", 58 streamType, status); 59 return status; 60 } 61 size_t afFrameCount; 62 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 63 if (status != NO_ERROR) { 64 ALOGE("Unable to query output frame count for stream type %d; status %d", 65 streamType, status); 66 return status; 67 } 68 uint32_t afLatency; 69 status = AudioSystem::getOutputLatency(&afLatency, streamType); 70 if (status != NO_ERROR) { 71 ALOGE("Unable to query output latency for stream type %d; status %d", 72 streamType, status); 73 return status; 74 } 75 76 // Ensure that buffer depth covers at least audio hardware latency 77 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 78 if (minBufCount < 2) { 79 minBufCount = 2; 80 } 81 82 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 83 afFrameCount * minBufCount * sampleRate / afSampleRate; 84 // The formula above should always produce a non-zero value, but return an error 85 // in the unlikely event that it does not, as that's part of the API contract. 86 if (*frameCount == 0) { 87 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 88 streamType, sampleRate); 89 return BAD_VALUE; 90 } 91 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 92 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 93 return NO_ERROR; 94} 95 96// --------------------------------------------------------------------------- 97 98AudioTrack::AudioTrack() 99 : mStatus(NO_INIT), 100 mIsTimed(false), 101 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 102 mPreviousSchedulingGroup(SP_DEFAULT) 103{ 104} 105 106AudioTrack::AudioTrack( 107 audio_stream_type_t streamType, 108 uint32_t sampleRate, 109 audio_format_t format, 110 audio_channel_mask_t channelMask, 111 int frameCount, 112 audio_output_flags_t flags, 113 callback_t cbf, 114 void* user, 115 int notificationFrames, 116 int sessionId, 117 transfer_type transferType, 118 const audio_offload_info_t *offloadInfo, 119 int uid) 120 : mStatus(NO_INIT), 121 mIsTimed(false), 122 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 123 mPreviousSchedulingGroup(SP_DEFAULT) 124{ 125 mStatus = set(streamType, sampleRate, format, channelMask, 126 frameCount, flags, cbf, user, notificationFrames, 127 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 128 offloadInfo, uid); 129} 130 131AudioTrack::AudioTrack( 132 audio_stream_type_t streamType, 133 uint32_t sampleRate, 134 audio_format_t format, 135 audio_channel_mask_t channelMask, 136 const sp<IMemory>& sharedBuffer, 137 audio_output_flags_t flags, 138 callback_t cbf, 139 void* user, 140 int notificationFrames, 141 int sessionId, 142 transfer_type transferType, 143 const audio_offload_info_t *offloadInfo, 144 int uid) 145 : mStatus(NO_INIT), 146 mIsTimed(false), 147 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 148 mPreviousSchedulingGroup(SP_DEFAULT) 149{ 150 mStatus = set(streamType, sampleRate, format, channelMask, 151 0 /*frameCount*/, flags, cbf, user, notificationFrames, 152 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); 153} 154 155AudioTrack::~AudioTrack() 156{ 157 if (mStatus == NO_ERROR) { 158 // Make sure that callback function exits in the case where 159 // it is looping on buffer full condition in obtainBuffer(). 160 // Otherwise the callback thread will never exit. 161 stop(); 162 if (mAudioTrackThread != 0) { 163 mProxy->interrupt(); 164 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 165 mAudioTrackThread->requestExitAndWait(); 166 mAudioTrackThread.clear(); 167 } 168 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 169 mAudioTrack.clear(); 170 IPCThreadState::self()->flushCommands(); 171 AudioSystem::releaseAudioSessionId(mSessionId); 172 } 173} 174 175status_t AudioTrack::set( 176 audio_stream_type_t streamType, 177 uint32_t sampleRate, 178 audio_format_t format, 179 audio_channel_mask_t channelMask, 180 int frameCountInt, 181 audio_output_flags_t flags, 182 callback_t cbf, 183 void* user, 184 int notificationFrames, 185 const sp<IMemory>& sharedBuffer, 186 bool threadCanCallJava, 187 int sessionId, 188 transfer_type transferType, 189 const audio_offload_info_t *offloadInfo, 190 int uid) 191{ 192 switch (transferType) { 193 case TRANSFER_DEFAULT: 194 if (sharedBuffer != 0) { 195 transferType = TRANSFER_SHARED; 196 } else if (cbf == NULL || threadCanCallJava) { 197 transferType = TRANSFER_SYNC; 198 } else { 199 transferType = TRANSFER_CALLBACK; 200 } 201 break; 202 case TRANSFER_CALLBACK: 203 if (cbf == NULL || sharedBuffer != 0) { 204 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 205 return BAD_VALUE; 206 } 207 break; 208 case TRANSFER_OBTAIN: 209 case TRANSFER_SYNC: 210 if (sharedBuffer != 0) { 211 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 212 return BAD_VALUE; 213 } 214 break; 215 case TRANSFER_SHARED: 216 if (sharedBuffer == 0) { 217 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 218 return BAD_VALUE; 219 } 220 break; 221 default: 222 ALOGE("Invalid transfer type %d", transferType); 223 return BAD_VALUE; 224 } 225 mTransfer = transferType; 226 227 // FIXME "int" here is legacy and will be replaced by size_t later 228 if (frameCountInt < 0) { 229 ALOGE("Invalid frame count %d", frameCountInt); 230 return BAD_VALUE; 231 } 232 size_t frameCount = frameCountInt; 233 234 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 235 sharedBuffer->size()); 236 237 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 238 239 AutoMutex lock(mLock); 240 241 // invariant that mAudioTrack != 0 is true only after set() returns successfully 242 if (mAudioTrack != 0) { 243 ALOGE("Track already in use"); 244 return INVALID_OPERATION; 245 } 246 247 mOutput = 0; 248 249 // handle default values first. 250 if (streamType == AUDIO_STREAM_DEFAULT) { 251 streamType = AUDIO_STREAM_MUSIC; 252 } 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 } 275 276 if (!audio_is_output_channel(channelMask)) { 277 ALOGE("Invalid channel mask %#x", channelMask); 278 return BAD_VALUE; 279 } 280 281 // AudioFlinger does not currently support 8-bit data in shared memory 282 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 283 ALOGE("8-bit data in shared memory is not supported"); 284 return BAD_VALUE; 285 } 286 287 // force direct flag if format is not linear PCM 288 // or offload was requested 289 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 290 || !audio_is_linear_pcm(format)) { 291 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 292 ? "Offload request, forcing to Direct Output" 293 : "Not linear PCM, forcing to Direct Output"); 294 flags = (audio_output_flags_t) 295 // FIXME why can't we allow direct AND fast? 296 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 297 } 298 // only allow deep buffering for music stream type 299 if (streamType != AUDIO_STREAM_MUSIC) { 300 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 301 } 302 303 mChannelMask = channelMask; 304 uint32_t channelCount = popcount(channelMask); 305 mChannelCount = channelCount; 306 307 if (audio_is_linear_pcm(format)) { 308 mFrameSize = channelCount * audio_bytes_per_sample(format); 309 mFrameSizeAF = channelCount * sizeof(int16_t); 310 } else { 311 mFrameSize = sizeof(uint8_t); 312 mFrameSizeAF = sizeof(uint8_t); 313 } 314 315 audio_io_handle_t output = AudioSystem::getOutput( 316 streamType, 317 sampleRate, format, channelMask, 318 flags, 319 offloadInfo); 320 321 if (output == 0) { 322 ALOGE("Could not get audio output for stream type %d", streamType); 323 return BAD_VALUE; 324 } 325 326 mVolume[LEFT] = 1.0f; 327 mVolume[RIGHT] = 1.0f; 328 mSendLevel = 0.0f; 329 // mFrameCount is initialized in createTrack_l 330 mReqFrameCount = frameCount; 331 mNotificationFramesReq = notificationFrames; 332 mNotificationFramesAct = 0; 333 mSessionId = sessionId; 334 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 335 mClientUid = IPCThreadState::self()->getCallingUid(); 336 } else { 337 mClientUid = uid; 338 } 339 mAuxEffectId = 0; 340 mFlags = flags; 341 mCbf = cbf; 342 343 if (cbf != NULL) { 344 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 345 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 346 } 347 348 // create the IAudioTrack 349 status = createTrack_l(streamType, 350 sampleRate, 351 format, 352 frameCount, 353 flags, 354 sharedBuffer, 355 output, 356 0 /*epoch*/); 357 358 if (status != NO_ERROR) { 359 if (mAudioTrackThread != 0) { 360 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 361 mAudioTrackThread->requestExitAndWait(); 362 mAudioTrackThread.clear(); 363 } 364 // Use of direct and offloaded output streams is ref counted by audio policy manager. 365 // As getOutput was called above and resulted in an output stream to be opened, 366 // we need to release it. 367 AudioSystem::releaseOutput(output); 368 return status; 369 } 370 371 mStatus = NO_ERROR; 372 mStreamType = streamType; 373 mFormat = format; 374 mSharedBuffer = sharedBuffer; 375 mState = STATE_STOPPED; 376 mUserData = user; 377 mLoopPeriod = 0; 378 mMarkerPosition = 0; 379 mMarkerReached = false; 380 mNewPosition = 0; 381 mUpdatePeriod = 0; 382 AudioSystem::acquireAudioSessionId(mSessionId); 383 mSequence = 1; 384 mObservedSequence = mSequence; 385 mInUnderrun = false; 386 mOutput = output; 387 388 return NO_ERROR; 389} 390 391// ------------------------------------------------------------------------- 392 393status_t AudioTrack::start() 394{ 395 AutoMutex lock(mLock); 396 397 if (mState == STATE_ACTIVE) { 398 return INVALID_OPERATION; 399 } 400 401 mInUnderrun = true; 402 403 State previousState = mState; 404 if (previousState == STATE_PAUSED_STOPPING) { 405 mState = STATE_STOPPING; 406 } else { 407 mState = STATE_ACTIVE; 408 } 409 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 410 // reset current position as seen by client to 0 411 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 412 // force refresh of remaining frames by processAudioBuffer() as last 413 // write before stop could be partial. 414 mRefreshRemaining = true; 415 } 416 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 417 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 418 419 sp<AudioTrackThread> t = mAudioTrackThread; 420 if (t != 0) { 421 if (previousState == STATE_STOPPING) { 422 mProxy->interrupt(); 423 } else { 424 t->resume(); 425 } 426 } else { 427 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 428 get_sched_policy(0, &mPreviousSchedulingGroup); 429 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 430 } 431 432 status_t status = NO_ERROR; 433 if (!(flags & CBLK_INVALID)) { 434 status = mAudioTrack->start(); 435 if (status == DEAD_OBJECT) { 436 flags |= CBLK_INVALID; 437 } 438 } 439 if (flags & CBLK_INVALID) { 440 status = restoreTrack_l("start"); 441 } 442 443 if (status != NO_ERROR) { 444 ALOGE("start() status %d", status); 445 mState = previousState; 446 if (t != 0) { 447 if (previousState != STATE_STOPPING) { 448 t->pause(); 449 } 450 } else { 451 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 452 set_sched_policy(0, mPreviousSchedulingGroup); 453 } 454 } 455 456 return status; 457} 458 459void AudioTrack::stop() 460{ 461 AutoMutex lock(mLock); 462 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 463 return; 464 } 465 466 if (isOffloaded_l()) { 467 mState = STATE_STOPPING; 468 } else { 469 mState = STATE_STOPPED; 470 } 471 472 mProxy->interrupt(); 473 mAudioTrack->stop(); 474 // the playback head position will reset to 0, so if a marker is set, we need 475 // to activate it again 476 mMarkerReached = false; 477#if 0 478 // Force flush if a shared buffer is used otherwise audioflinger 479 // will not stop before end of buffer is reached. 480 // It may be needed to make sure that we stop playback, likely in case looping is on. 481 if (mSharedBuffer != 0) { 482 flush_l(); 483 } 484#endif 485 486 sp<AudioTrackThread> t = mAudioTrackThread; 487 if (t != 0) { 488 if (!isOffloaded_l()) { 489 t->pause(); 490 } 491 } else { 492 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 493 set_sched_policy(0, mPreviousSchedulingGroup); 494 } 495} 496 497bool AudioTrack::stopped() const 498{ 499 AutoMutex lock(mLock); 500 return mState != STATE_ACTIVE; 501} 502 503void AudioTrack::flush() 504{ 505 if (mSharedBuffer != 0) { 506 return; 507 } 508 AutoMutex lock(mLock); 509 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 510 return; 511 } 512 flush_l(); 513} 514 515void AudioTrack::flush_l() 516{ 517 ALOG_ASSERT(mState != STATE_ACTIVE); 518 519 // clear playback marker and periodic update counter 520 mMarkerPosition = 0; 521 mMarkerReached = false; 522 mUpdatePeriod = 0; 523 mRefreshRemaining = true; 524 525 mState = STATE_FLUSHED; 526 if (isOffloaded_l()) { 527 mProxy->interrupt(); 528 } 529 mProxy->flush(); 530 mAudioTrack->flush(); 531} 532 533void AudioTrack::pause() 534{ 535 AutoMutex lock(mLock); 536 if (mState == STATE_ACTIVE) { 537 mState = STATE_PAUSED; 538 } else if (mState == STATE_STOPPING) { 539 mState = STATE_PAUSED_STOPPING; 540 } else { 541 return; 542 } 543 mProxy->interrupt(); 544 mAudioTrack->pause(); 545} 546 547status_t AudioTrack::setVolume(float left, float right) 548{ 549 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 550 return BAD_VALUE; 551 } 552 553 AutoMutex lock(mLock); 554 mVolume[LEFT] = left; 555 mVolume[RIGHT] = right; 556 557 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 558 559 if (isOffloaded_l()) { 560 mAudioTrack->signal(); 561 } 562 return NO_ERROR; 563} 564 565status_t AudioTrack::setVolume(float volume) 566{ 567 return setVolume(volume, volume); 568} 569 570status_t AudioTrack::setAuxEffectSendLevel(float level) 571{ 572 if (level < 0.0f || level > 1.0f) { 573 return BAD_VALUE; 574 } 575 576 AutoMutex lock(mLock); 577 mSendLevel = level; 578 mProxy->setSendLevel(level); 579 580 return NO_ERROR; 581} 582 583void AudioTrack::getAuxEffectSendLevel(float* level) const 584{ 585 if (level != NULL) { 586 *level = mSendLevel; 587 } 588} 589 590status_t AudioTrack::setSampleRate(uint32_t rate) 591{ 592 if (mIsTimed || isOffloaded()) { 593 return INVALID_OPERATION; 594 } 595 596 uint32_t afSamplingRate; 597 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 598 return NO_INIT; 599 } 600 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 601 if (rate == 0 || rate > afSamplingRate*2 ) { 602 return BAD_VALUE; 603 } 604 605 AutoMutex lock(mLock); 606 mSampleRate = rate; 607 mProxy->setSampleRate(rate); 608 609 return NO_ERROR; 610} 611 612uint32_t AudioTrack::getSampleRate() const 613{ 614 if (mIsTimed) { 615 return 0; 616 } 617 618 AutoMutex lock(mLock); 619 620 // sample rate can be updated during playback by the offloaded decoder so we need to 621 // query the HAL and update if needed. 622// FIXME use Proxy return channel to update the rate from server and avoid polling here 623 if (isOffloaded_l()) { 624 if (mOutput != 0) { 625 uint32_t sampleRate = 0; 626 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 627 if (status == NO_ERROR) { 628 mSampleRate = sampleRate; 629 } 630 } 631 } 632 return mSampleRate; 633} 634 635status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 636{ 637 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 638 return INVALID_OPERATION; 639 } 640 641 if (loopCount == 0) { 642 ; 643 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 644 loopEnd - loopStart >= MIN_LOOP) { 645 ; 646 } else { 647 return BAD_VALUE; 648 } 649 650 AutoMutex lock(mLock); 651 // See setPosition() regarding setting parameters such as loop points or position while active 652 if (mState == STATE_ACTIVE) { 653 return INVALID_OPERATION; 654 } 655 setLoop_l(loopStart, loopEnd, loopCount); 656 return NO_ERROR; 657} 658 659void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 660{ 661 // FIXME If setting a loop also sets position to start of loop, then 662 // this is correct. Otherwise it should be removed. 663 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 664 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 665 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 666} 667 668status_t AudioTrack::setMarkerPosition(uint32_t marker) 669{ 670 // The only purpose of setting marker position is to get a callback 671 if (mCbf == NULL || isOffloaded()) { 672 return INVALID_OPERATION; 673 } 674 675 AutoMutex lock(mLock); 676 mMarkerPosition = marker; 677 mMarkerReached = false; 678 679 return NO_ERROR; 680} 681 682status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 683{ 684 if (isOffloaded()) { 685 return INVALID_OPERATION; 686 } 687 if (marker == NULL) { 688 return BAD_VALUE; 689 } 690 691 AutoMutex lock(mLock); 692 *marker = mMarkerPosition; 693 694 return NO_ERROR; 695} 696 697status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 698{ 699 // The only purpose of setting position update period is to get a callback 700 if (mCbf == NULL || isOffloaded()) { 701 return INVALID_OPERATION; 702 } 703 704 AutoMutex lock(mLock); 705 mNewPosition = mProxy->getPosition() + updatePeriod; 706 mUpdatePeriod = updatePeriod; 707 708 return NO_ERROR; 709} 710 711status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 712{ 713 if (isOffloaded()) { 714 return INVALID_OPERATION; 715 } 716 if (updatePeriod == NULL) { 717 return BAD_VALUE; 718 } 719 720 AutoMutex lock(mLock); 721 *updatePeriod = mUpdatePeriod; 722 723 return NO_ERROR; 724} 725 726status_t AudioTrack::setPosition(uint32_t position) 727{ 728 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 729 return INVALID_OPERATION; 730 } 731 if (position > mFrameCount) { 732 return BAD_VALUE; 733 } 734 735 AutoMutex lock(mLock); 736 // Currently we require that the player is inactive before setting parameters such as position 737 // or loop points. Otherwise, there could be a race condition: the application could read the 738 // current position, compute a new position or loop parameters, and then set that position or 739 // loop parameters but it would do the "wrong" thing since the position has continued to advance 740 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 741 // to specify how it wants to handle such scenarios. 742 if (mState == STATE_ACTIVE) { 743 return INVALID_OPERATION; 744 } 745 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 746 mLoopPeriod = 0; 747 // FIXME Check whether loops and setting position are incompatible in old code. 748 // If we use setLoop for both purposes we lose the capability to set the position while looping. 749 mStaticProxy->setLoop(position, mFrameCount, 0); 750 751 return NO_ERROR; 752} 753 754status_t AudioTrack::getPosition(uint32_t *position) const 755{ 756 if (position == NULL) { 757 return BAD_VALUE; 758 } 759 760 AutoMutex lock(mLock); 761 if (isOffloaded_l()) { 762 uint32_t dspFrames = 0; 763 764 if (mOutput != 0) { 765 uint32_t halFrames; 766 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 767 } 768 *position = dspFrames; 769 } else { 770 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 771 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 772 mProxy->getPosition(); 773 } 774 return NO_ERROR; 775} 776 777status_t AudioTrack::getBufferPosition(size_t *position) 778{ 779 if (mSharedBuffer == 0 || mIsTimed) { 780 return INVALID_OPERATION; 781 } 782 if (position == NULL) { 783 return BAD_VALUE; 784 } 785 786 AutoMutex lock(mLock); 787 *position = mStaticProxy->getBufferPosition(); 788 return NO_ERROR; 789} 790 791status_t AudioTrack::reload() 792{ 793 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 794 return INVALID_OPERATION; 795 } 796 797 AutoMutex lock(mLock); 798 // See setPosition() regarding setting parameters such as loop points or position while active 799 if (mState == STATE_ACTIVE) { 800 return INVALID_OPERATION; 801 } 802 mNewPosition = mUpdatePeriod; 803 mLoopPeriod = 0; 804 // FIXME The new code cannot reload while keeping a loop specified. 805 // Need to check how the old code handled this, and whether it's a significant change. 806 mStaticProxy->setLoop(0, mFrameCount, 0); 807 return NO_ERROR; 808} 809 810audio_io_handle_t AudioTrack::getOutput() 811{ 812 AutoMutex lock(mLock); 813 return mOutput; 814} 815 816// must be called with mLock held 817audio_io_handle_t AudioTrack::getOutput_l() 818{ 819 if (mOutput) { 820 return mOutput; 821 } else { 822 return AudioSystem::getOutput(mStreamType, 823 mSampleRate, mFormat, mChannelMask, mFlags); 824 } 825} 826 827status_t AudioTrack::attachAuxEffect(int effectId) 828{ 829 AutoMutex lock(mLock); 830 status_t status = mAudioTrack->attachAuxEffect(effectId); 831 if (status == NO_ERROR) { 832 mAuxEffectId = effectId; 833 } 834 return status; 835} 836 837// ------------------------------------------------------------------------- 838 839// must be called with mLock held 840status_t AudioTrack::createTrack_l( 841 audio_stream_type_t streamType, 842 uint32_t sampleRate, 843 audio_format_t format, 844 size_t frameCount, 845 audio_output_flags_t flags, 846 const sp<IMemory>& sharedBuffer, 847 audio_io_handle_t output, 848 size_t epoch) 849{ 850 status_t status; 851 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 852 if (audioFlinger == 0) { 853 ALOGE("Could not get audioflinger"); 854 return NO_INIT; 855 } 856 857 // Not all of these values are needed under all conditions, but it is easier to get them all 858 859 uint32_t afLatency; 860 status = AudioSystem::getLatency(output, streamType, &afLatency); 861 if (status != NO_ERROR) { 862 ALOGE("getLatency(%d) failed status %d", output, status); 863 return NO_INIT; 864 } 865 866 size_t afFrameCount; 867 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 868 if (status != NO_ERROR) { 869 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 870 return NO_INIT; 871 } 872 873 uint32_t afSampleRate; 874 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 875 if (status != NO_ERROR) { 876 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 877 return NO_INIT; 878 } 879 880 // Client decides whether the track is TIMED (see below), but can only express a preference 881 // for FAST. Server will perform additional tests. 882 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 883 // either of these use cases: 884 // use case 1: shared buffer 885 (sharedBuffer != 0) || 886 // use case 2: callback handler 887 (mCbf != NULL))) { 888 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 889 // once denied, do not request again if IAudioTrack is re-created 890 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 891 mFlags = flags; 892 } 893 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 894 895 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 896 // n = 1 fast track with single buffering; nBuffering is ignored 897 // n = 2 fast track with double buffering 898 // n = 2 normal track, no sample rate conversion 899 // n = 3 normal track, with sample rate conversion 900 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 901 // n > 3 very high latency or very small notification interval; nBuffering is ignored 902 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 903 904 mNotificationFramesAct = mNotificationFramesReq; 905 906 if (!audio_is_linear_pcm(format)) { 907 908 if (sharedBuffer != 0) { 909 // Same comment as below about ignoring frameCount parameter for set() 910 frameCount = sharedBuffer->size(); 911 } else if (frameCount == 0) { 912 frameCount = afFrameCount; 913 } 914 if (mNotificationFramesAct != frameCount) { 915 mNotificationFramesAct = frameCount; 916 } 917 } else if (sharedBuffer != 0) { 918 919 // Ensure that buffer alignment matches channel count 920 // 8-bit data in shared memory is not currently supported by AudioFlinger 921 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 922 if (mChannelCount > 1) { 923 // More than 2 channels does not require stronger alignment than stereo 924 alignment <<= 1; 925 } 926 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 927 ALOGE("Invalid buffer alignment: address %p, channel count %u", 928 sharedBuffer->pointer(), mChannelCount); 929 return BAD_VALUE; 930 } 931 932 // When initializing a shared buffer AudioTrack via constructors, 933 // there's no frameCount parameter. 934 // But when initializing a shared buffer AudioTrack via set(), 935 // there _is_ a frameCount parameter. We silently ignore it. 936 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 937 938 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 939 940 // FIXME move these calculations and associated checks to server 941 942 // Ensure that buffer depth covers at least audio hardware latency 943 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 944 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 945 afFrameCount, minBufCount, afSampleRate, afLatency); 946 if (minBufCount <= nBuffering) { 947 minBufCount = nBuffering; 948 } 949 950 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 951 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 952 ", afLatency=%d", 953 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 954 955 if (frameCount == 0) { 956 frameCount = minFrameCount; 957 } else if (frameCount < minFrameCount) { 958 // not ALOGW because it happens all the time when playing key clicks over A2DP 959 ALOGV("Minimum buffer size corrected from %d to %d", 960 frameCount, minFrameCount); 961 frameCount = minFrameCount; 962 } 963 // Make sure that application is notified with sufficient margin before underrun 964 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 965 mNotificationFramesAct = frameCount/nBuffering; 966 } 967 968 } else { 969 // For fast tracks, the frame count calculations and checks are done by server 970 } 971 972 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 973 if (mIsTimed) { 974 trackFlags |= IAudioFlinger::TRACK_TIMED; 975 } 976 977 pid_t tid = -1; 978 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 979 trackFlags |= IAudioFlinger::TRACK_FAST; 980 if (mAudioTrackThread != 0) { 981 tid = mAudioTrackThread->getTid(); 982 } 983 } 984 985 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 986 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 987 } 988 989 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 990 sampleRate, 991 // AudioFlinger only sees 16-bit PCM 992 format == AUDIO_FORMAT_PCM_8_BIT ? 993 AUDIO_FORMAT_PCM_16_BIT : format, 994 mChannelMask, 995 frameCount, 996 &trackFlags, 997 sharedBuffer, 998 output, 999 tid, 1000 &mSessionId, 1001 mName, 1002 mClientUid, 1003 &status); 1004 1005 if (track == 0) { 1006 ALOGE("AudioFlinger could not create track, status: %d", status); 1007 return status; 1008 } 1009 sp<IMemory> iMem = track->getCblk(); 1010 if (iMem == 0) { 1011 ALOGE("Could not get control block"); 1012 return NO_INIT; 1013 } 1014 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1015 if (mAudioTrack != 0) { 1016 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1017 mDeathNotifier.clear(); 1018 } 1019 mAudioTrack = track; 1020 mCblkMemory = iMem; 1021 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1022 mCblk = cblk; 1023 size_t temp = cblk->frameCount_; 1024 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1025 // In current design, AudioTrack client checks and ensures frame count validity before 1026 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1027 // for fast track as it uses a special method of assigning frame count. 1028 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1029 } 1030 frameCount = temp; 1031 mAwaitBoost = false; 1032 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1033 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1034 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1035 mAwaitBoost = true; 1036 if (sharedBuffer == 0) { 1037 // Theoretically double-buffering is not required for fast tracks, 1038 // due to tighter scheduling. But in practice, to accommodate kernels with 1039 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1040 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1041 mNotificationFramesAct = frameCount/nBuffering; 1042 } 1043 } 1044 } else { 1045 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1046 // once denied, do not request again if IAudioTrack is re-created 1047 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1048 mFlags = flags; 1049 if (sharedBuffer == 0) { 1050 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1051 mNotificationFramesAct = frameCount/nBuffering; 1052 } 1053 } 1054 } 1055 } 1056 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1057 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1058 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1059 } else { 1060 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1061 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1062 mFlags = flags; 1063 return NO_INIT; 1064 } 1065 } 1066 1067 mRefreshRemaining = true; 1068 1069 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1070 // is the value of pointer() for the shared buffer, otherwise buffers points 1071 // immediately after the control block. This address is for the mapping within client 1072 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1073 void* buffers; 1074 if (sharedBuffer == 0) { 1075 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1076 } else { 1077 buffers = sharedBuffer->pointer(); 1078 } 1079 1080 mAudioTrack->attachAuxEffect(mAuxEffectId); 1081 // FIXME don't believe this lie 1082 mLatency = afLatency + (1000*frameCount) / sampleRate; 1083 mFrameCount = frameCount; 1084 // If IAudioTrack is re-created, don't let the requested frameCount 1085 // decrease. This can confuse clients that cache frameCount(). 1086 if (frameCount > mReqFrameCount) { 1087 mReqFrameCount = frameCount; 1088 } 1089 1090 // update proxy 1091 if (sharedBuffer == 0) { 1092 mStaticProxy.clear(); 1093 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1094 } else { 1095 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1096 mProxy = mStaticProxy; 1097 } 1098 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1099 uint16_t(mVolume[LEFT] * 0x1000)); 1100 mProxy->setSendLevel(mSendLevel); 1101 mProxy->setSampleRate(mSampleRate); 1102 mProxy->setEpoch(epoch); 1103 mProxy->setMinimum(mNotificationFramesAct); 1104 1105 mDeathNotifier = new DeathNotifier(this); 1106 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1107 1108 return NO_ERROR; 1109} 1110 1111status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1112{ 1113 if (audioBuffer == NULL) { 1114 return BAD_VALUE; 1115 } 1116 if (mTransfer != TRANSFER_OBTAIN) { 1117 audioBuffer->frameCount = 0; 1118 audioBuffer->size = 0; 1119 audioBuffer->raw = NULL; 1120 return INVALID_OPERATION; 1121 } 1122 1123 const struct timespec *requested; 1124 if (waitCount == -1) { 1125 requested = &ClientProxy::kForever; 1126 } else if (waitCount == 0) { 1127 requested = &ClientProxy::kNonBlocking; 1128 } else if (waitCount > 0) { 1129 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1130 struct timespec timeout; 1131 timeout.tv_sec = ms / 1000; 1132 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1133 requested = &timeout; 1134 } else { 1135 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1136 requested = NULL; 1137 } 1138 return obtainBuffer(audioBuffer, requested); 1139} 1140 1141status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1142 struct timespec *elapsed, size_t *nonContig) 1143{ 1144 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1145 uint32_t oldSequence = 0; 1146 uint32_t newSequence; 1147 1148 Proxy::Buffer buffer; 1149 status_t status = NO_ERROR; 1150 1151 static const int32_t kMaxTries = 5; 1152 int32_t tryCounter = kMaxTries; 1153 1154 do { 1155 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1156 // keep them from going away if another thread re-creates the track during obtainBuffer() 1157 sp<AudioTrackClientProxy> proxy; 1158 sp<IMemory> iMem; 1159 1160 { // start of lock scope 1161 AutoMutex lock(mLock); 1162 1163 newSequence = mSequence; 1164 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1165 if (status == DEAD_OBJECT) { 1166 // re-create track, unless someone else has already done so 1167 if (newSequence == oldSequence) { 1168 status = restoreTrack_l("obtainBuffer"); 1169 if (status != NO_ERROR) { 1170 buffer.mFrameCount = 0; 1171 buffer.mRaw = NULL; 1172 buffer.mNonContig = 0; 1173 break; 1174 } 1175 } 1176 } 1177 oldSequence = newSequence; 1178 1179 // Keep the extra references 1180 proxy = mProxy; 1181 iMem = mCblkMemory; 1182 1183 if (mState == STATE_STOPPING) { 1184 status = -EINTR; 1185 buffer.mFrameCount = 0; 1186 buffer.mRaw = NULL; 1187 buffer.mNonContig = 0; 1188 break; 1189 } 1190 1191 // Non-blocking if track is stopped or paused 1192 if (mState != STATE_ACTIVE) { 1193 requested = &ClientProxy::kNonBlocking; 1194 } 1195 1196 } // end of lock scope 1197 1198 buffer.mFrameCount = audioBuffer->frameCount; 1199 // FIXME starts the requested timeout and elapsed over from scratch 1200 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1201 1202 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1203 1204 audioBuffer->frameCount = buffer.mFrameCount; 1205 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1206 audioBuffer->raw = buffer.mRaw; 1207 if (nonContig != NULL) { 1208 *nonContig = buffer.mNonContig; 1209 } 1210 return status; 1211} 1212 1213void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1214{ 1215 if (mTransfer == TRANSFER_SHARED) { 1216 return; 1217 } 1218 1219 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1220 if (stepCount == 0) { 1221 return; 1222 } 1223 1224 Proxy::Buffer buffer; 1225 buffer.mFrameCount = stepCount; 1226 buffer.mRaw = audioBuffer->raw; 1227 1228 AutoMutex lock(mLock); 1229 mInUnderrun = false; 1230 mProxy->releaseBuffer(&buffer); 1231 1232 // restart track if it was disabled by audioflinger due to previous underrun 1233 if (mState == STATE_ACTIVE) { 1234 audio_track_cblk_t* cblk = mCblk; 1235 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1236 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1237 this, mName.string()); 1238 // FIXME ignoring status 1239 mAudioTrack->start(); 1240 } 1241 } 1242} 1243 1244// ------------------------------------------------------------------------- 1245 1246ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1247{ 1248 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1249 return INVALID_OPERATION; 1250 } 1251 1252 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1253 // Sanity-check: user is most-likely passing an error code, and it would 1254 // make the return value ambiguous (actualSize vs error). 1255 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1256 return BAD_VALUE; 1257 } 1258 1259 size_t written = 0; 1260 Buffer audioBuffer; 1261 1262 while (userSize >= mFrameSize) { 1263 audioBuffer.frameCount = userSize / mFrameSize; 1264 1265 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1266 if (err < 0) { 1267 if (written > 0) { 1268 break; 1269 } 1270 return ssize_t(err); 1271 } 1272 1273 size_t toWrite; 1274 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1275 // Divide capacity by 2 to take expansion into account 1276 toWrite = audioBuffer.size >> 1; 1277 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1278 } else { 1279 toWrite = audioBuffer.size; 1280 memcpy(audioBuffer.i8, buffer, toWrite); 1281 } 1282 buffer = ((const char *) buffer) + toWrite; 1283 userSize -= toWrite; 1284 written += toWrite; 1285 1286 releaseBuffer(&audioBuffer); 1287 } 1288 1289 return written; 1290} 1291 1292// ------------------------------------------------------------------------- 1293 1294TimedAudioTrack::TimedAudioTrack() { 1295 mIsTimed = true; 1296} 1297 1298status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1299{ 1300 AutoMutex lock(mLock); 1301 status_t result = UNKNOWN_ERROR; 1302 1303#if 1 1304 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1305 // while we are accessing the cblk 1306 sp<IAudioTrack> audioTrack = mAudioTrack; 1307 sp<IMemory> iMem = mCblkMemory; 1308#endif 1309 1310 // If the track is not invalid already, try to allocate a buffer. alloc 1311 // fails indicating that the server is dead, flag the track as invalid so 1312 // we can attempt to restore in just a bit. 1313 audio_track_cblk_t* cblk = mCblk; 1314 if (!(cblk->mFlags & CBLK_INVALID)) { 1315 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1316 if (result == DEAD_OBJECT) { 1317 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1318 } 1319 } 1320 1321 // If the track is invalid at this point, attempt to restore it. and try the 1322 // allocation one more time. 1323 if (cblk->mFlags & CBLK_INVALID) { 1324 result = restoreTrack_l("allocateTimedBuffer"); 1325 1326 if (result == NO_ERROR) { 1327 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1328 } 1329 } 1330 1331 return result; 1332} 1333 1334status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1335 int64_t pts) 1336{ 1337 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1338 { 1339 AutoMutex lock(mLock); 1340 audio_track_cblk_t* cblk = mCblk; 1341 // restart track if it was disabled by audioflinger due to previous underrun 1342 if (buffer->size() != 0 && status == NO_ERROR && 1343 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1344 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1345 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1346 // FIXME ignoring status 1347 mAudioTrack->start(); 1348 } 1349 } 1350 return status; 1351} 1352 1353status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1354 TargetTimeline target) 1355{ 1356 return mAudioTrack->setMediaTimeTransform(xform, target); 1357} 1358 1359// ------------------------------------------------------------------------- 1360 1361nsecs_t AudioTrack::processAudioBuffer() 1362{ 1363 // Currently the AudioTrack thread is not created if there are no callbacks. 1364 // Would it ever make sense to run the thread, even without callbacks? 1365 // If so, then replace this by checks at each use for mCbf != NULL. 1366 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1367 1368 mLock.lock(); 1369 if (mAwaitBoost) { 1370 mAwaitBoost = false; 1371 mLock.unlock(); 1372 static const int32_t kMaxTries = 5; 1373 int32_t tryCounter = kMaxTries; 1374 uint32_t pollUs = 10000; 1375 do { 1376 int policy = sched_getscheduler(0); 1377 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1378 break; 1379 } 1380 usleep(pollUs); 1381 pollUs <<= 1; 1382 } while (tryCounter-- > 0); 1383 if (tryCounter < 0) { 1384 ALOGE("did not receive expected priority boost on time"); 1385 } 1386 // Run again immediately 1387 return 0; 1388 } 1389 1390 // Can only reference mCblk while locked 1391 int32_t flags = android_atomic_and( 1392 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1393 1394 // Check for track invalidation 1395 if (flags & CBLK_INVALID) { 1396 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1397 // AudioSystem cache. We should not exit here but after calling the callback so 1398 // that the upper layers can recreate the track 1399 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1400 status_t status = restoreTrack_l("processAudioBuffer"); 1401 mLock.unlock(); 1402 // Run again immediately, but with a new IAudioTrack 1403 return 0; 1404 } 1405 } 1406 1407 bool waitStreamEnd = mState == STATE_STOPPING; 1408 bool active = mState == STATE_ACTIVE; 1409 1410 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1411 bool newUnderrun = false; 1412 if (flags & CBLK_UNDERRUN) { 1413#if 0 1414 // Currently in shared buffer mode, when the server reaches the end of buffer, 1415 // the track stays active in continuous underrun state. It's up to the application 1416 // to pause or stop the track, or set the position to a new offset within buffer. 1417 // This was some experimental code to auto-pause on underrun. Keeping it here 1418 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1419 if (mTransfer == TRANSFER_SHARED) { 1420 mState = STATE_PAUSED; 1421 active = false; 1422 } 1423#endif 1424 if (!mInUnderrun) { 1425 mInUnderrun = true; 1426 newUnderrun = true; 1427 } 1428 } 1429 1430 // Get current position of server 1431 size_t position = mProxy->getPosition(); 1432 1433 // Manage marker callback 1434 bool markerReached = false; 1435 size_t markerPosition = mMarkerPosition; 1436 // FIXME fails for wraparound, need 64 bits 1437 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1438 mMarkerReached = markerReached = true; 1439 } 1440 1441 // Determine number of new position callback(s) that will be needed, while locked 1442 size_t newPosCount = 0; 1443 size_t newPosition = mNewPosition; 1444 size_t updatePeriod = mUpdatePeriod; 1445 // FIXME fails for wraparound, need 64 bits 1446 if (updatePeriod > 0 && position >= newPosition) { 1447 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1448 mNewPosition += updatePeriod * newPosCount; 1449 } 1450 1451 // Cache other fields that will be needed soon 1452 uint32_t loopPeriod = mLoopPeriod; 1453 uint32_t sampleRate = mSampleRate; 1454 size_t notificationFrames = mNotificationFramesAct; 1455 if (mRefreshRemaining) { 1456 mRefreshRemaining = false; 1457 mRemainingFrames = notificationFrames; 1458 mRetryOnPartialBuffer = false; 1459 } 1460 size_t misalignment = mProxy->getMisalignment(); 1461 uint32_t sequence = mSequence; 1462 1463 // These fields don't need to be cached, because they are assigned only by set(): 1464 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1465 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1466 1467 mLock.unlock(); 1468 1469 if (waitStreamEnd) { 1470 AutoMutex lock(mLock); 1471 1472 sp<AudioTrackClientProxy> proxy = mProxy; 1473 sp<IMemory> iMem = mCblkMemory; 1474 1475 struct timespec timeout; 1476 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1477 timeout.tv_nsec = 0; 1478 1479 mLock.unlock(); 1480 status_t status = mProxy->waitStreamEndDone(&timeout); 1481 mLock.lock(); 1482 switch (status) { 1483 case NO_ERROR: 1484 case DEAD_OBJECT: 1485 case TIMED_OUT: 1486 mLock.unlock(); 1487 mCbf(EVENT_STREAM_END, mUserData, NULL); 1488 mLock.lock(); 1489 if (mState == STATE_STOPPING) { 1490 mState = STATE_STOPPED; 1491 if (status != DEAD_OBJECT) { 1492 return NS_INACTIVE; 1493 } 1494 } 1495 return 0; 1496 default: 1497 return 0; 1498 } 1499 } 1500 1501 // perform callbacks while unlocked 1502 if (newUnderrun) { 1503 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1504 } 1505 // FIXME we will miss loops if loop cycle was signaled several times since last call 1506 // to processAudioBuffer() 1507 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1508 mCbf(EVENT_LOOP_END, mUserData, NULL); 1509 } 1510 if (flags & CBLK_BUFFER_END) { 1511 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1512 } 1513 if (markerReached) { 1514 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1515 } 1516 while (newPosCount > 0) { 1517 size_t temp = newPosition; 1518 mCbf(EVENT_NEW_POS, mUserData, &temp); 1519 newPosition += updatePeriod; 1520 newPosCount--; 1521 } 1522 1523 if (mObservedSequence != sequence) { 1524 mObservedSequence = sequence; 1525 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1526 // for offloaded tracks, just wait for the upper layers to recreate the track 1527 if (isOffloaded()) { 1528 return NS_INACTIVE; 1529 } 1530 } 1531 1532 // if inactive, then don't run me again until re-started 1533 if (!active) { 1534 return NS_INACTIVE; 1535 } 1536 1537 // Compute the estimated time until the next timed event (position, markers, loops) 1538 // FIXME only for non-compressed audio 1539 uint32_t minFrames = ~0; 1540 if (!markerReached && position < markerPosition) { 1541 minFrames = markerPosition - position; 1542 } 1543 if (loopPeriod > 0 && loopPeriod < minFrames) { 1544 minFrames = loopPeriod; 1545 } 1546 if (updatePeriod > 0 && updatePeriod < minFrames) { 1547 minFrames = updatePeriod; 1548 } 1549 1550 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1551 static const uint32_t kPoll = 0; 1552 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1553 minFrames = kPoll * notificationFrames; 1554 } 1555 1556 // Convert frame units to time units 1557 nsecs_t ns = NS_WHENEVER; 1558 if (minFrames != (uint32_t) ~0) { 1559 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1560 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1561 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1562 } 1563 1564 // If not supplying data by EVENT_MORE_DATA, then we're done 1565 if (mTransfer != TRANSFER_CALLBACK) { 1566 return ns; 1567 } 1568 1569 struct timespec timeout; 1570 const struct timespec *requested = &ClientProxy::kForever; 1571 if (ns != NS_WHENEVER) { 1572 timeout.tv_sec = ns / 1000000000LL; 1573 timeout.tv_nsec = ns % 1000000000LL; 1574 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1575 requested = &timeout; 1576 } 1577 1578 while (mRemainingFrames > 0) { 1579 1580 Buffer audioBuffer; 1581 audioBuffer.frameCount = mRemainingFrames; 1582 size_t nonContig; 1583 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1584 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1585 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1586 requested = &ClientProxy::kNonBlocking; 1587 size_t avail = audioBuffer.frameCount + nonContig; 1588 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1589 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1590 if (err != NO_ERROR) { 1591 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1592 (isOffloaded() && (err == DEAD_OBJECT))) { 1593 return 0; 1594 } 1595 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1596 return NS_NEVER; 1597 } 1598 1599 if (mRetryOnPartialBuffer && !isOffloaded()) { 1600 mRetryOnPartialBuffer = false; 1601 if (avail < mRemainingFrames) { 1602 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1603 if (ns < 0 || myns < ns) { 1604 ns = myns; 1605 } 1606 return ns; 1607 } 1608 } 1609 1610 // Divide buffer size by 2 to take into account the expansion 1611 // due to 8 to 16 bit conversion: the callback must fill only half 1612 // of the destination buffer 1613 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1614 audioBuffer.size >>= 1; 1615 } 1616 1617 size_t reqSize = audioBuffer.size; 1618 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1619 size_t writtenSize = audioBuffer.size; 1620 size_t writtenFrames = writtenSize / mFrameSize; 1621 1622 // Sanity check on returned size 1623 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1624 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1625 reqSize, (int) writtenSize); 1626 return NS_NEVER; 1627 } 1628 1629 if (writtenSize == 0) { 1630 // The callback is done filling buffers 1631 // Keep this thread going to handle timed events and 1632 // still try to get more data in intervals of WAIT_PERIOD_MS 1633 // but don't just loop and block the CPU, so wait 1634 return WAIT_PERIOD_MS * 1000000LL; 1635 } 1636 1637 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1638 // 8 to 16 bit conversion, note that source and destination are the same address 1639 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1640 audioBuffer.size <<= 1; 1641 } 1642 1643 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1644 audioBuffer.frameCount = releasedFrames; 1645 mRemainingFrames -= releasedFrames; 1646 if (misalignment >= releasedFrames) { 1647 misalignment -= releasedFrames; 1648 } else { 1649 misalignment = 0; 1650 } 1651 1652 releaseBuffer(&audioBuffer); 1653 1654 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1655 // if callback doesn't like to accept the full chunk 1656 if (writtenSize < reqSize) { 1657 continue; 1658 } 1659 1660 // There could be enough non-contiguous frames available to satisfy the remaining request 1661 if (mRemainingFrames <= nonContig) { 1662 continue; 1663 } 1664 1665#if 0 1666 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1667 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1668 // that total to a sum == notificationFrames. 1669 if (0 < misalignment && misalignment <= mRemainingFrames) { 1670 mRemainingFrames = misalignment; 1671 return (mRemainingFrames * 1100000000LL) / sampleRate; 1672 } 1673#endif 1674 1675 } 1676 mRemainingFrames = notificationFrames; 1677 mRetryOnPartialBuffer = true; 1678 1679 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1680 return 0; 1681} 1682 1683status_t AudioTrack::restoreTrack_l(const char *from) 1684{ 1685 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1686 isOffloaded_l() ? "Offloaded" : "PCM", from); 1687 ++mSequence; 1688 status_t result; 1689 1690 // refresh the audio configuration cache in this process to make sure we get new 1691 // output parameters in getOutput_l() and createTrack_l() 1692 AudioSystem::clearAudioConfigCache(); 1693 1694 if (isOffloaded_l()) { 1695 // FIXME re-creation of offloaded tracks is not yet implemented 1696 return DEAD_OBJECT; 1697 } 1698 1699 // force new output query from audio policy manager; 1700 mOutput = 0; 1701 audio_io_handle_t output = getOutput_l(); 1702 1703 // if the new IAudioTrack is created, createTrack_l() will modify the 1704 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1705 // It will also delete the strong references on previous IAudioTrack and IMemory 1706 1707 // take the frames that will be lost by track recreation into account in saved position 1708 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1709 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1710 result = createTrack_l(mStreamType, 1711 mSampleRate, 1712 mFormat, 1713 mReqFrameCount, // so that frame count never goes down 1714 mFlags, 1715 mSharedBuffer, 1716 output, 1717 position /*epoch*/); 1718 1719 if (result == NO_ERROR) { 1720 // continue playback from last known position, but 1721 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1722 if (mStaticProxy != NULL) { 1723 mLoopPeriod = 0; 1724 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1725 } 1726 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1727 // track destruction have been played? This is critical for SoundPool implementation 1728 // This must be broken, and needs to be tested/debugged. 1729#if 0 1730 // restore write index and set other indexes to reflect empty buffer status 1731 if (!strcmp(from, "start")) { 1732 // Make sure that a client relying on callback events indicating underrun or 1733 // the actual amount of audio frames played (e.g SoundPool) receives them. 1734 if (mSharedBuffer == 0) { 1735 // restart playback even if buffer is not completely filled. 1736 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1737 } 1738 } 1739#endif 1740 if (mState == STATE_ACTIVE) { 1741 result = mAudioTrack->start(); 1742 } 1743 } 1744 if (result != NO_ERROR) { 1745 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1746 // As getOutput was called above and resulted in an output stream to be opened, 1747 // we need to release it. 1748 AudioSystem::releaseOutput(output); 1749 ALOGW("restoreTrack_l() failed status %d", result); 1750 mState = STATE_STOPPED; 1751 } 1752 1753 return result; 1754} 1755 1756status_t AudioTrack::setParameters(const String8& keyValuePairs) 1757{ 1758 AutoMutex lock(mLock); 1759 return mAudioTrack->setParameters(keyValuePairs); 1760} 1761 1762status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1763{ 1764 AutoMutex lock(mLock); 1765 // FIXME not implemented for fast tracks; should use proxy and SSQ 1766 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1767 return INVALID_OPERATION; 1768 } 1769 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1770 return INVALID_OPERATION; 1771 } 1772 status_t status = mAudioTrack->getTimestamp(timestamp); 1773 if (status == NO_ERROR) { 1774 timestamp.mPosition += mProxy->getEpoch(); 1775 } 1776 return status; 1777} 1778 1779String8 AudioTrack::getParameters(const String8& keys) 1780{ 1781 audio_io_handle_t output = getOutput(); 1782 if (output != 0) { 1783 return AudioSystem::getParameters(output, keys); 1784 } else { 1785 return String8::empty(); 1786 } 1787} 1788 1789bool AudioTrack::isOffloaded() const 1790{ 1791 AutoMutex lock(mLock); 1792 return isOffloaded_l(); 1793} 1794 1795status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1796{ 1797 1798 const size_t SIZE = 256; 1799 char buffer[SIZE]; 1800 String8 result; 1801 1802 result.append(" AudioTrack::dump\n"); 1803 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1804 mVolume[0], mVolume[1]); 1805 result.append(buffer); 1806 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1807 mChannelCount, mFrameCount); 1808 result.append(buffer); 1809 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1810 result.append(buffer); 1811 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1812 result.append(buffer); 1813 ::write(fd, result.string(), result.size()); 1814 return NO_ERROR; 1815} 1816 1817uint32_t AudioTrack::getUnderrunFrames() const 1818{ 1819 AutoMutex lock(mLock); 1820 return mProxy->getUnderrunFrames(); 1821} 1822 1823// ========================================================================= 1824 1825void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1826{ 1827 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1828 if (audioTrack != 0) { 1829 AutoMutex lock(audioTrack->mLock); 1830 audioTrack->mProxy->binderDied(); 1831 } 1832} 1833 1834// ========================================================================= 1835 1836AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1837 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1838 mIgnoreNextPausedInt(false) 1839{ 1840} 1841 1842AudioTrack::AudioTrackThread::~AudioTrackThread() 1843{ 1844} 1845 1846bool AudioTrack::AudioTrackThread::threadLoop() 1847{ 1848 { 1849 AutoMutex _l(mMyLock); 1850 if (mPaused) { 1851 mMyCond.wait(mMyLock); 1852 // caller will check for exitPending() 1853 return true; 1854 } 1855 if (mIgnoreNextPausedInt) { 1856 mIgnoreNextPausedInt = false; 1857 mPausedInt = false; 1858 } 1859 if (mPausedInt) { 1860 if (mPausedNs > 0) { 1861 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1862 } else { 1863 mMyCond.wait(mMyLock); 1864 } 1865 mPausedInt = false; 1866 return true; 1867 } 1868 } 1869 nsecs_t ns = mReceiver.processAudioBuffer(); 1870 switch (ns) { 1871 case 0: 1872 return true; 1873 case NS_INACTIVE: 1874 pauseInternal(); 1875 return true; 1876 case NS_NEVER: 1877 return false; 1878 case NS_WHENEVER: 1879 // FIXME increase poll interval, or make event-driven 1880 ns = 1000000000LL; 1881 // fall through 1882 default: 1883 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1884 pauseInternal(ns); 1885 return true; 1886 } 1887} 1888 1889void AudioTrack::AudioTrackThread::requestExit() 1890{ 1891 // must be in this order to avoid a race condition 1892 Thread::requestExit(); 1893 resume(); 1894} 1895 1896void AudioTrack::AudioTrackThread::pause() 1897{ 1898 AutoMutex _l(mMyLock); 1899 mPaused = true; 1900} 1901 1902void AudioTrack::AudioTrackThread::resume() 1903{ 1904 AutoMutex _l(mMyLock); 1905 mIgnoreNextPausedInt = true; 1906 if (mPaused || mPausedInt) { 1907 mPaused = false; 1908 mPausedInt = false; 1909 mMyCond.signal(); 1910 } 1911} 1912 1913void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1914{ 1915 AutoMutex _l(mMyLock); 1916 mPausedInt = true; 1917 mPausedNs = ns; 1918} 1919 1920}; // namespace android 1921