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