AudioTrack.cpp revision 70c0bfbe5ec88dcc3efa2bd8df26f36cff1cf03a
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 return NO_ERROR; 708} 709 710status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 711{ 712 if (isOffloaded()) { 713 return INVALID_OPERATION; 714 } 715 if (updatePeriod == NULL) { 716 return BAD_VALUE; 717 } 718 719 AutoMutex lock(mLock); 720 *updatePeriod = mUpdatePeriod; 721 722 return NO_ERROR; 723} 724 725status_t AudioTrack::setPosition(uint32_t position) 726{ 727 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 728 return INVALID_OPERATION; 729 } 730 if (position > mFrameCount) { 731 return BAD_VALUE; 732 } 733 734 AutoMutex lock(mLock); 735 // Currently we require that the player is inactive before setting parameters such as position 736 // or loop points. Otherwise, there could be a race condition: the application could read the 737 // current position, compute a new position or loop parameters, and then set that position or 738 // loop parameters but it would do the "wrong" thing since the position has continued to advance 739 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 740 // to specify how it wants to handle such scenarios. 741 if (mState == STATE_ACTIVE) { 742 return INVALID_OPERATION; 743 } 744 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 745 mLoopPeriod = 0; 746 // FIXME Check whether loops and setting position are incompatible in old code. 747 // If we use setLoop for both purposes we lose the capability to set the position while looping. 748 mStaticProxy->setLoop(position, mFrameCount, 0); 749 750 return NO_ERROR; 751} 752 753status_t AudioTrack::getPosition(uint32_t *position) const 754{ 755 if (position == NULL) { 756 return BAD_VALUE; 757 } 758 759 AutoMutex lock(mLock); 760 if (isOffloaded_l()) { 761 uint32_t dspFrames = 0; 762 763 if (mOutput != 0) { 764 uint32_t halFrames; 765 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 766 } 767 *position = dspFrames; 768 } else { 769 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 770 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 771 mProxy->getPosition(); 772 } 773 return NO_ERROR; 774} 775 776status_t AudioTrack::getBufferPosition(size_t *position) 777{ 778 if (mSharedBuffer == 0 || mIsTimed) { 779 return INVALID_OPERATION; 780 } 781 if (position == NULL) { 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 *position = mStaticProxy->getBufferPosition(); 787 return NO_ERROR; 788} 789 790status_t AudioTrack::reload() 791{ 792 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 793 return INVALID_OPERATION; 794 } 795 796 AutoMutex lock(mLock); 797 // See setPosition() regarding setting parameters such as loop points or position while active 798 if (mState == STATE_ACTIVE) { 799 return INVALID_OPERATION; 800 } 801 mNewPosition = mUpdatePeriod; 802 mLoopPeriod = 0; 803 // FIXME The new code cannot reload while keeping a loop specified. 804 // Need to check how the old code handled this, and whether it's a significant change. 805 mStaticProxy->setLoop(0, mFrameCount, 0); 806 return NO_ERROR; 807} 808 809audio_io_handle_t AudioTrack::getOutput() 810{ 811 AutoMutex lock(mLock); 812 return mOutput; 813} 814 815// must be called with mLock held 816audio_io_handle_t AudioTrack::getOutput_l() 817{ 818 if (mOutput) { 819 return mOutput; 820 } else { 821 return AudioSystem::getOutput(mStreamType, 822 mSampleRate, mFormat, mChannelMask, mFlags); 823 } 824} 825 826status_t AudioTrack::attachAuxEffect(int effectId) 827{ 828 AutoMutex lock(mLock); 829 status_t status = mAudioTrack->attachAuxEffect(effectId); 830 if (status == NO_ERROR) { 831 mAuxEffectId = effectId; 832 } 833 return status; 834} 835 836// ------------------------------------------------------------------------- 837 838// must be called with mLock held 839status_t AudioTrack::createTrack_l( 840 audio_stream_type_t streamType, 841 uint32_t sampleRate, 842 audio_format_t format, 843 size_t frameCount, 844 audio_output_flags_t flags, 845 const sp<IMemory>& sharedBuffer, 846 audio_io_handle_t output, 847 size_t epoch) 848{ 849 status_t status; 850 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 851 if (audioFlinger == 0) { 852 ALOGE("Could not get audioflinger"); 853 return NO_INIT; 854 } 855 856 // Not all of these values are needed under all conditions, but it is easier to get them all 857 858 uint32_t afLatency; 859 status = AudioSystem::getLatency(output, streamType, &afLatency); 860 if (status != NO_ERROR) { 861 ALOGE("getLatency(%d) failed status %d", output, status); 862 return NO_INIT; 863 } 864 865 size_t afFrameCount; 866 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 867 if (status != NO_ERROR) { 868 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 869 return NO_INIT; 870 } 871 872 uint32_t afSampleRate; 873 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 874 if (status != NO_ERROR) { 875 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status); 876 return NO_INIT; 877 } 878 879 // Client decides whether the track is TIMED (see below), but can only express a preference 880 // for FAST. Server will perform additional tests. 881 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 882 // either of these use cases: 883 // use case 1: shared buffer 884 (sharedBuffer != 0) || 885 // use case 2: callback handler 886 (mCbf != NULL))) { 887 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 888 // once denied, do not request again if IAudioTrack is re-created 889 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 890 mFlags = flags; 891 } 892 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 893 894 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 895 // n = 1 fast track with single buffering; nBuffering is ignored 896 // n = 2 fast track with double buffering 897 // n = 2 normal track, no sample rate conversion 898 // n = 3 normal track, with sample rate conversion 899 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 900 // n > 3 very high latency or very small notification interval; nBuffering is ignored 901 const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3; 902 903 mNotificationFramesAct = mNotificationFramesReq; 904 905 if (!audio_is_linear_pcm(format)) { 906 907 if (sharedBuffer != 0) { 908 // Same comment as below about ignoring frameCount parameter for set() 909 frameCount = sharedBuffer->size(); 910 } else if (frameCount == 0) { 911 frameCount = afFrameCount; 912 } 913 if (mNotificationFramesAct != frameCount) { 914 mNotificationFramesAct = frameCount; 915 } 916 } else if (sharedBuffer != 0) { 917 918 // Ensure that buffer alignment matches channel count 919 // 8-bit data in shared memory is not currently supported by AudioFlinger 920 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 921 if (mChannelCount > 1) { 922 // More than 2 channels does not require stronger alignment than stereo 923 alignment <<= 1; 924 } 925 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 926 ALOGE("Invalid buffer alignment: address %p, channel count %u", 927 sharedBuffer->pointer(), mChannelCount); 928 return BAD_VALUE; 929 } 930 931 // When initializing a shared buffer AudioTrack via constructors, 932 // there's no frameCount parameter. 933 // But when initializing a shared buffer AudioTrack via set(), 934 // there _is_ a frameCount parameter. We silently ignore it. 935 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 936 937 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 938 939 // FIXME move these calculations and associated checks to server 940 941 // Ensure that buffer depth covers at least audio hardware latency 942 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 943 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 944 afFrameCount, minBufCount, afSampleRate, afLatency); 945 if (minBufCount <= nBuffering) { 946 minBufCount = nBuffering; 947 } 948 949 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 950 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 951 ", afLatency=%d", 952 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 953 954 if (frameCount == 0) { 955 frameCount = minFrameCount; 956 } else if (frameCount < minFrameCount) { 957 // not ALOGW because it happens all the time when playing key clicks over A2DP 958 ALOGV("Minimum buffer size corrected from %d to %d", 959 frameCount, minFrameCount); 960 frameCount = minFrameCount; 961 } 962 // Make sure that application is notified with sufficient margin before underrun 963 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 964 mNotificationFramesAct = frameCount/nBuffering; 965 } 966 967 } else { 968 // For fast tracks, the frame count calculations and checks are done by server 969 } 970 971 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 972 if (mIsTimed) { 973 trackFlags |= IAudioFlinger::TRACK_TIMED; 974 } 975 976 pid_t tid = -1; 977 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 978 trackFlags |= IAudioFlinger::TRACK_FAST; 979 if (mAudioTrackThread != 0) { 980 tid = mAudioTrackThread->getTid(); 981 } 982 } 983 984 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 985 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 986 } 987 988 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 989 sampleRate, 990 // AudioFlinger only sees 16-bit PCM 991 format == AUDIO_FORMAT_PCM_8_BIT ? 992 AUDIO_FORMAT_PCM_16_BIT : format, 993 mChannelMask, 994 frameCount, 995 &trackFlags, 996 sharedBuffer, 997 output, 998 tid, 999 &mSessionId, 1000 mName, 1001 mClientUid, 1002 &status); 1003 1004 if (track == 0) { 1005 ALOGE("AudioFlinger could not create track, status: %d", status); 1006 return status; 1007 } 1008 sp<IMemory> iMem = track->getCblk(); 1009 if (iMem == 0) { 1010 ALOGE("Could not get control block"); 1011 return NO_INIT; 1012 } 1013 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1014 if (mAudioTrack != 0) { 1015 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1016 mDeathNotifier.clear(); 1017 } 1018 mAudioTrack = track; 1019 mCblkMemory = iMem; 1020 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 1021 mCblk = cblk; 1022 size_t temp = cblk->frameCount_; 1023 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1024 // In current design, AudioTrack client checks and ensures frame count validity before 1025 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1026 // for fast track as it uses a special method of assigning frame count. 1027 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1028 } 1029 frameCount = temp; 1030 mAwaitBoost = false; 1031 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1032 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1033 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1034 mAwaitBoost = true; 1035 if (sharedBuffer == 0) { 1036 // Theoretically double-buffering is not required for fast tracks, 1037 // due to tighter scheduling. But in practice, to accommodate kernels with 1038 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1039 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1040 mNotificationFramesAct = frameCount/nBuffering; 1041 } 1042 } 1043 } else { 1044 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1045 // once denied, do not request again if IAudioTrack is re-created 1046 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 1047 mFlags = flags; 1048 if (sharedBuffer == 0) { 1049 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1050 mNotificationFramesAct = frameCount/nBuffering; 1051 } 1052 } 1053 } 1054 } 1055 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1056 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1057 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1058 } else { 1059 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1060 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1061 mFlags = flags; 1062 return NO_INIT; 1063 } 1064 } 1065 1066 mRefreshRemaining = true; 1067 1068 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1069 // is the value of pointer() for the shared buffer, otherwise buffers points 1070 // immediately after the control block. This address is for the mapping within client 1071 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1072 void* buffers; 1073 if (sharedBuffer == 0) { 1074 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1075 } else { 1076 buffers = sharedBuffer->pointer(); 1077 } 1078 1079 mAudioTrack->attachAuxEffect(mAuxEffectId); 1080 // FIXME don't believe this lie 1081 mLatency = afLatency + (1000*frameCount) / sampleRate; 1082 mFrameCount = frameCount; 1083 // If IAudioTrack is re-created, don't let the requested frameCount 1084 // decrease. This can confuse clients that cache frameCount(). 1085 if (frameCount > mReqFrameCount) { 1086 mReqFrameCount = frameCount; 1087 } 1088 1089 // update proxy 1090 if (sharedBuffer == 0) { 1091 mStaticProxy.clear(); 1092 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1093 } else { 1094 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1095 mProxy = mStaticProxy; 1096 } 1097 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1098 uint16_t(mVolume[LEFT] * 0x1000)); 1099 mProxy->setSendLevel(mSendLevel); 1100 mProxy->setSampleRate(mSampleRate); 1101 mProxy->setEpoch(epoch); 1102 mProxy->setMinimum(mNotificationFramesAct); 1103 1104 mDeathNotifier = new DeathNotifier(this); 1105 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1106 1107 return NO_ERROR; 1108} 1109 1110status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1111{ 1112 if (audioBuffer == NULL) { 1113 return BAD_VALUE; 1114 } 1115 if (mTransfer != TRANSFER_OBTAIN) { 1116 audioBuffer->frameCount = 0; 1117 audioBuffer->size = 0; 1118 audioBuffer->raw = NULL; 1119 return INVALID_OPERATION; 1120 } 1121 1122 const struct timespec *requested; 1123 if (waitCount == -1) { 1124 requested = &ClientProxy::kForever; 1125 } else if (waitCount == 0) { 1126 requested = &ClientProxy::kNonBlocking; 1127 } else if (waitCount > 0) { 1128 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1129 struct timespec timeout; 1130 timeout.tv_sec = ms / 1000; 1131 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1132 requested = &timeout; 1133 } else { 1134 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1135 requested = NULL; 1136 } 1137 return obtainBuffer(audioBuffer, requested); 1138} 1139 1140status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1141 struct timespec *elapsed, size_t *nonContig) 1142{ 1143 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1144 uint32_t oldSequence = 0; 1145 uint32_t newSequence; 1146 1147 Proxy::Buffer buffer; 1148 status_t status = NO_ERROR; 1149 1150 static const int32_t kMaxTries = 5; 1151 int32_t tryCounter = kMaxTries; 1152 1153 do { 1154 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1155 // keep them from going away if another thread re-creates the track during obtainBuffer() 1156 sp<AudioTrackClientProxy> proxy; 1157 sp<IMemory> iMem; 1158 1159 { // start of lock scope 1160 AutoMutex lock(mLock); 1161 1162 newSequence = mSequence; 1163 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1164 if (status == DEAD_OBJECT) { 1165 // re-create track, unless someone else has already done so 1166 if (newSequence == oldSequence) { 1167 status = restoreTrack_l("obtainBuffer"); 1168 if (status != NO_ERROR) { 1169 buffer.mFrameCount = 0; 1170 buffer.mRaw = NULL; 1171 buffer.mNonContig = 0; 1172 break; 1173 } 1174 } 1175 } 1176 oldSequence = newSequence; 1177 1178 // Keep the extra references 1179 proxy = mProxy; 1180 iMem = mCblkMemory; 1181 1182 if (mState == STATE_STOPPING) { 1183 status = -EINTR; 1184 buffer.mFrameCount = 0; 1185 buffer.mRaw = NULL; 1186 buffer.mNonContig = 0; 1187 break; 1188 } 1189 1190 // Non-blocking if track is stopped or paused 1191 if (mState != STATE_ACTIVE) { 1192 requested = &ClientProxy::kNonBlocking; 1193 } 1194 1195 } // end of lock scope 1196 1197 buffer.mFrameCount = audioBuffer->frameCount; 1198 // FIXME starts the requested timeout and elapsed over from scratch 1199 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1200 1201 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1202 1203 audioBuffer->frameCount = buffer.mFrameCount; 1204 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1205 audioBuffer->raw = buffer.mRaw; 1206 if (nonContig != NULL) { 1207 *nonContig = buffer.mNonContig; 1208 } 1209 return status; 1210} 1211 1212void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1213{ 1214 if (mTransfer == TRANSFER_SHARED) { 1215 return; 1216 } 1217 1218 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1219 if (stepCount == 0) { 1220 return; 1221 } 1222 1223 Proxy::Buffer buffer; 1224 buffer.mFrameCount = stepCount; 1225 buffer.mRaw = audioBuffer->raw; 1226 1227 AutoMutex lock(mLock); 1228 mInUnderrun = false; 1229 mProxy->releaseBuffer(&buffer); 1230 1231 // restart track if it was disabled by audioflinger due to previous underrun 1232 if (mState == STATE_ACTIVE) { 1233 audio_track_cblk_t* cblk = mCblk; 1234 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1235 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1236 this, mName.string()); 1237 // FIXME ignoring status 1238 mAudioTrack->start(); 1239 } 1240 } 1241} 1242 1243// ------------------------------------------------------------------------- 1244 1245ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1246{ 1247 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1248 return INVALID_OPERATION; 1249 } 1250 1251 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1252 // Sanity-check: user is most-likely passing an error code, and it would 1253 // make the return value ambiguous (actualSize vs error). 1254 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1255 return BAD_VALUE; 1256 } 1257 1258 size_t written = 0; 1259 Buffer audioBuffer; 1260 1261 while (userSize >= mFrameSize) { 1262 audioBuffer.frameCount = userSize / mFrameSize; 1263 1264 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1265 if (err < 0) { 1266 if (written > 0) { 1267 break; 1268 } 1269 return ssize_t(err); 1270 } 1271 1272 size_t toWrite; 1273 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1274 // Divide capacity by 2 to take expansion into account 1275 toWrite = audioBuffer.size >> 1; 1276 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1277 } else { 1278 toWrite = audioBuffer.size; 1279 memcpy(audioBuffer.i8, buffer, toWrite); 1280 } 1281 buffer = ((const char *) buffer) + toWrite; 1282 userSize -= toWrite; 1283 written += toWrite; 1284 1285 releaseBuffer(&audioBuffer); 1286 } 1287 1288 return written; 1289} 1290 1291// ------------------------------------------------------------------------- 1292 1293TimedAudioTrack::TimedAudioTrack() { 1294 mIsTimed = true; 1295} 1296 1297status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1298{ 1299 AutoMutex lock(mLock); 1300 status_t result = UNKNOWN_ERROR; 1301 1302#if 1 1303 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1304 // while we are accessing the cblk 1305 sp<IAudioTrack> audioTrack = mAudioTrack; 1306 sp<IMemory> iMem = mCblkMemory; 1307#endif 1308 1309 // If the track is not invalid already, try to allocate a buffer. alloc 1310 // fails indicating that the server is dead, flag the track as invalid so 1311 // we can attempt to restore in just a bit. 1312 audio_track_cblk_t* cblk = mCblk; 1313 if (!(cblk->mFlags & CBLK_INVALID)) { 1314 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1315 if (result == DEAD_OBJECT) { 1316 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1317 } 1318 } 1319 1320 // If the track is invalid at this point, attempt to restore it. and try the 1321 // allocation one more time. 1322 if (cblk->mFlags & CBLK_INVALID) { 1323 result = restoreTrack_l("allocateTimedBuffer"); 1324 1325 if (result == NO_ERROR) { 1326 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1327 } 1328 } 1329 1330 return result; 1331} 1332 1333status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1334 int64_t pts) 1335{ 1336 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1337 { 1338 AutoMutex lock(mLock); 1339 audio_track_cblk_t* cblk = mCblk; 1340 // restart track if it was disabled by audioflinger due to previous underrun 1341 if (buffer->size() != 0 && status == NO_ERROR && 1342 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1343 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1344 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1345 // FIXME ignoring status 1346 mAudioTrack->start(); 1347 } 1348 } 1349 return status; 1350} 1351 1352status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1353 TargetTimeline target) 1354{ 1355 return mAudioTrack->setMediaTimeTransform(xform, target); 1356} 1357 1358// ------------------------------------------------------------------------- 1359 1360nsecs_t AudioTrack::processAudioBuffer() 1361{ 1362 // Currently the AudioTrack thread is not created if there are no callbacks. 1363 // Would it ever make sense to run the thread, even without callbacks? 1364 // If so, then replace this by checks at each use for mCbf != NULL. 1365 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1366 1367 mLock.lock(); 1368 if (mAwaitBoost) { 1369 mAwaitBoost = false; 1370 mLock.unlock(); 1371 static const int32_t kMaxTries = 5; 1372 int32_t tryCounter = kMaxTries; 1373 uint32_t pollUs = 10000; 1374 do { 1375 int policy = sched_getscheduler(0); 1376 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1377 break; 1378 } 1379 usleep(pollUs); 1380 pollUs <<= 1; 1381 } while (tryCounter-- > 0); 1382 if (tryCounter < 0) { 1383 ALOGE("did not receive expected priority boost on time"); 1384 } 1385 // Run again immediately 1386 return 0; 1387 } 1388 1389 // Can only reference mCblk while locked 1390 int32_t flags = android_atomic_and( 1391 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1392 1393 // Check for track invalidation 1394 if (flags & CBLK_INVALID) { 1395 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1396 // AudioSystem cache. We should not exit here but after calling the callback so 1397 // that the upper layers can recreate the track 1398 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1399 status_t status = restoreTrack_l("processAudioBuffer"); 1400 mLock.unlock(); 1401 // Run again immediately, but with a new IAudioTrack 1402 return 0; 1403 } 1404 } 1405 1406 bool waitStreamEnd = mState == STATE_STOPPING; 1407 bool active = mState == STATE_ACTIVE; 1408 1409 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1410 bool newUnderrun = false; 1411 if (flags & CBLK_UNDERRUN) { 1412#if 0 1413 // Currently in shared buffer mode, when the server reaches the end of buffer, 1414 // the track stays active in continuous underrun state. It's up to the application 1415 // to pause or stop the track, or set the position to a new offset within buffer. 1416 // This was some experimental code to auto-pause on underrun. Keeping it here 1417 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1418 if (mTransfer == TRANSFER_SHARED) { 1419 mState = STATE_PAUSED; 1420 active = false; 1421 } 1422#endif 1423 if (!mInUnderrun) { 1424 mInUnderrun = true; 1425 newUnderrun = true; 1426 } 1427 } 1428 1429 // Get current position of server 1430 size_t position = mProxy->getPosition(); 1431 1432 // Manage marker callback 1433 bool markerReached = false; 1434 size_t markerPosition = mMarkerPosition; 1435 // FIXME fails for wraparound, need 64 bits 1436 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1437 mMarkerReached = markerReached = true; 1438 } 1439 1440 // Determine number of new position callback(s) that will be needed, while locked 1441 size_t newPosCount = 0; 1442 size_t newPosition = mNewPosition; 1443 size_t updatePeriod = mUpdatePeriod; 1444 // FIXME fails for wraparound, need 64 bits 1445 if (updatePeriod > 0 && position >= newPosition) { 1446 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1447 mNewPosition += updatePeriod * newPosCount; 1448 } 1449 1450 // Cache other fields that will be needed soon 1451 uint32_t loopPeriod = mLoopPeriod; 1452 uint32_t sampleRate = mSampleRate; 1453 size_t notificationFrames = mNotificationFramesAct; 1454 if (mRefreshRemaining) { 1455 mRefreshRemaining = false; 1456 mRemainingFrames = notificationFrames; 1457 mRetryOnPartialBuffer = false; 1458 } 1459 size_t misalignment = mProxy->getMisalignment(); 1460 uint32_t sequence = mSequence; 1461 1462 // These fields don't need to be cached, because they are assigned only by set(): 1463 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1464 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1465 1466 mLock.unlock(); 1467 1468 if (waitStreamEnd) { 1469 AutoMutex lock(mLock); 1470 1471 sp<AudioTrackClientProxy> proxy = mProxy; 1472 sp<IMemory> iMem = mCblkMemory; 1473 1474 struct timespec timeout; 1475 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1476 timeout.tv_nsec = 0; 1477 1478 mLock.unlock(); 1479 status_t status = mProxy->waitStreamEndDone(&timeout); 1480 mLock.lock(); 1481 switch (status) { 1482 case NO_ERROR: 1483 case DEAD_OBJECT: 1484 case TIMED_OUT: 1485 mLock.unlock(); 1486 mCbf(EVENT_STREAM_END, mUserData, NULL); 1487 mLock.lock(); 1488 if (mState == STATE_STOPPING) { 1489 mState = STATE_STOPPED; 1490 if (status != DEAD_OBJECT) { 1491 return NS_INACTIVE; 1492 } 1493 } 1494 return 0; 1495 default: 1496 return 0; 1497 } 1498 } 1499 1500 // perform callbacks while unlocked 1501 if (newUnderrun) { 1502 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1503 } 1504 // FIXME we will miss loops if loop cycle was signaled several times since last call 1505 // to processAudioBuffer() 1506 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1507 mCbf(EVENT_LOOP_END, mUserData, NULL); 1508 } 1509 if (flags & CBLK_BUFFER_END) { 1510 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1511 } 1512 if (markerReached) { 1513 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1514 } 1515 while (newPosCount > 0) { 1516 size_t temp = newPosition; 1517 mCbf(EVENT_NEW_POS, mUserData, &temp); 1518 newPosition += updatePeriod; 1519 newPosCount--; 1520 } 1521 1522 if (mObservedSequence != sequence) { 1523 mObservedSequence = sequence; 1524 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1525 // for offloaded tracks, just wait for the upper layers to recreate the track 1526 if (isOffloaded()) { 1527 return NS_INACTIVE; 1528 } 1529 } 1530 1531 // if inactive, then don't run me again until re-started 1532 if (!active) { 1533 return NS_INACTIVE; 1534 } 1535 1536 // Compute the estimated time until the next timed event (position, markers, loops) 1537 // FIXME only for non-compressed audio 1538 uint32_t minFrames = ~0; 1539 if (!markerReached && position < markerPosition) { 1540 minFrames = markerPosition - position; 1541 } 1542 if (loopPeriod > 0 && loopPeriod < minFrames) { 1543 minFrames = loopPeriod; 1544 } 1545 if (updatePeriod > 0 && updatePeriod < minFrames) { 1546 minFrames = updatePeriod; 1547 } 1548 1549 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1550 static const uint32_t kPoll = 0; 1551 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1552 minFrames = kPoll * notificationFrames; 1553 } 1554 1555 // Convert frame units to time units 1556 nsecs_t ns = NS_WHENEVER; 1557 if (minFrames != (uint32_t) ~0) { 1558 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1559 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1560 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1561 } 1562 1563 // If not supplying data by EVENT_MORE_DATA, then we're done 1564 if (mTransfer != TRANSFER_CALLBACK) { 1565 return ns; 1566 } 1567 1568 struct timespec timeout; 1569 const struct timespec *requested = &ClientProxy::kForever; 1570 if (ns != NS_WHENEVER) { 1571 timeout.tv_sec = ns / 1000000000LL; 1572 timeout.tv_nsec = ns % 1000000000LL; 1573 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1574 requested = &timeout; 1575 } 1576 1577 while (mRemainingFrames > 0) { 1578 1579 Buffer audioBuffer; 1580 audioBuffer.frameCount = mRemainingFrames; 1581 size_t nonContig; 1582 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1583 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1584 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1585 requested = &ClientProxy::kNonBlocking; 1586 size_t avail = audioBuffer.frameCount + nonContig; 1587 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1588 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1589 if (err != NO_ERROR) { 1590 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1591 (isOffloaded() && (err == DEAD_OBJECT))) { 1592 return 0; 1593 } 1594 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1595 return NS_NEVER; 1596 } 1597 1598 if (mRetryOnPartialBuffer && !isOffloaded()) { 1599 mRetryOnPartialBuffer = false; 1600 if (avail < mRemainingFrames) { 1601 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1602 if (ns < 0 || myns < ns) { 1603 ns = myns; 1604 } 1605 return ns; 1606 } 1607 } 1608 1609 // Divide buffer size by 2 to take into account the expansion 1610 // due to 8 to 16 bit conversion: the callback must fill only half 1611 // of the destination buffer 1612 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1613 audioBuffer.size >>= 1; 1614 } 1615 1616 size_t reqSize = audioBuffer.size; 1617 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1618 size_t writtenSize = audioBuffer.size; 1619 size_t writtenFrames = writtenSize / mFrameSize; 1620 1621 // Sanity check on returned size 1622 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1623 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1624 reqSize, (int) writtenSize); 1625 return NS_NEVER; 1626 } 1627 1628 if (writtenSize == 0) { 1629 // The callback is done filling buffers 1630 // Keep this thread going to handle timed events and 1631 // still try to get more data in intervals of WAIT_PERIOD_MS 1632 // but don't just loop and block the CPU, so wait 1633 return WAIT_PERIOD_MS * 1000000LL; 1634 } 1635 1636 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1637 // 8 to 16 bit conversion, note that source and destination are the same address 1638 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1639 audioBuffer.size <<= 1; 1640 } 1641 1642 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1643 audioBuffer.frameCount = releasedFrames; 1644 mRemainingFrames -= releasedFrames; 1645 if (misalignment >= releasedFrames) { 1646 misalignment -= releasedFrames; 1647 } else { 1648 misalignment = 0; 1649 } 1650 1651 releaseBuffer(&audioBuffer); 1652 1653 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1654 // if callback doesn't like to accept the full chunk 1655 if (writtenSize < reqSize) { 1656 continue; 1657 } 1658 1659 // There could be enough non-contiguous frames available to satisfy the remaining request 1660 if (mRemainingFrames <= nonContig) { 1661 continue; 1662 } 1663 1664#if 0 1665 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1666 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1667 // that total to a sum == notificationFrames. 1668 if (0 < misalignment && misalignment <= mRemainingFrames) { 1669 mRemainingFrames = misalignment; 1670 return (mRemainingFrames * 1100000000LL) / sampleRate; 1671 } 1672#endif 1673 1674 } 1675 mRemainingFrames = notificationFrames; 1676 mRetryOnPartialBuffer = true; 1677 1678 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1679 return 0; 1680} 1681 1682status_t AudioTrack::restoreTrack_l(const char *from) 1683{ 1684 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1685 isOffloaded_l() ? "Offloaded" : "PCM", from); 1686 ++mSequence; 1687 status_t result; 1688 1689 // refresh the audio configuration cache in this process to make sure we get new 1690 // output parameters in getOutput_l() and createTrack_l() 1691 AudioSystem::clearAudioConfigCache(); 1692 1693 if (isOffloaded_l()) { 1694 // FIXME re-creation of offloaded tracks is not yet implemented 1695 return DEAD_OBJECT; 1696 } 1697 1698 // force new output query from audio policy manager; 1699 mOutput = 0; 1700 audio_io_handle_t output = getOutput_l(); 1701 1702 // if the new IAudioTrack is created, createTrack_l() will modify the 1703 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1704 // It will also delete the strong references on previous IAudioTrack and IMemory 1705 1706 // take the frames that will be lost by track recreation into account in saved position 1707 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1708 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1709 result = createTrack_l(mStreamType, 1710 mSampleRate, 1711 mFormat, 1712 mReqFrameCount, // so that frame count never goes down 1713 mFlags, 1714 mSharedBuffer, 1715 output, 1716 position /*epoch*/); 1717 1718 if (result == NO_ERROR) { 1719 // continue playback from last known position, but 1720 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1721 if (mStaticProxy != NULL) { 1722 mLoopPeriod = 0; 1723 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1724 } 1725 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1726 // track destruction have been played? This is critical for SoundPool implementation 1727 // This must be broken, and needs to be tested/debugged. 1728#if 0 1729 // restore write index and set other indexes to reflect empty buffer status 1730 if (!strcmp(from, "start")) { 1731 // Make sure that a client relying on callback events indicating underrun or 1732 // the actual amount of audio frames played (e.g SoundPool) receives them. 1733 if (mSharedBuffer == 0) { 1734 // restart playback even if buffer is not completely filled. 1735 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1736 } 1737 } 1738#endif 1739 if (mState == STATE_ACTIVE) { 1740 result = mAudioTrack->start(); 1741 } 1742 } 1743 if (result != NO_ERROR) { 1744 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1745 // As getOutput was called above and resulted in an output stream to be opened, 1746 // we need to release it. 1747 AudioSystem::releaseOutput(output); 1748 ALOGW("restoreTrack_l() failed status %d", result); 1749 mState = STATE_STOPPED; 1750 } 1751 1752 return result; 1753} 1754 1755status_t AudioTrack::setParameters(const String8& keyValuePairs) 1756{ 1757 AutoMutex lock(mLock); 1758 return mAudioTrack->setParameters(keyValuePairs); 1759} 1760 1761status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1762{ 1763 AutoMutex lock(mLock); 1764 // FIXME not implemented for fast tracks; should use proxy and SSQ 1765 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1766 return INVALID_OPERATION; 1767 } 1768 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1769 return INVALID_OPERATION; 1770 } 1771 status_t status = mAudioTrack->getTimestamp(timestamp); 1772 if (status == NO_ERROR) { 1773 timestamp.mPosition += mProxy->getEpoch(); 1774 } 1775 return status; 1776} 1777 1778String8 AudioTrack::getParameters(const String8& keys) 1779{ 1780 audio_io_handle_t output = getOutput(); 1781 if (output != 0) { 1782 return AudioSystem::getParameters(output, keys); 1783 } else { 1784 return String8::empty(); 1785 } 1786} 1787 1788bool AudioTrack::isOffloaded() const 1789{ 1790 AutoMutex lock(mLock); 1791 return isOffloaded_l(); 1792} 1793 1794status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1795{ 1796 1797 const size_t SIZE = 256; 1798 char buffer[SIZE]; 1799 String8 result; 1800 1801 result.append(" AudioTrack::dump\n"); 1802 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1803 mVolume[0], mVolume[1]); 1804 result.append(buffer); 1805 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1806 mChannelCount, mFrameCount); 1807 result.append(buffer); 1808 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1809 result.append(buffer); 1810 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1811 result.append(buffer); 1812 ::write(fd, result.string(), result.size()); 1813 return NO_ERROR; 1814} 1815 1816uint32_t AudioTrack::getUnderrunFrames() const 1817{ 1818 AutoMutex lock(mLock); 1819 return mProxy->getUnderrunFrames(); 1820} 1821 1822// ========================================================================= 1823 1824void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1825{ 1826 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1827 if (audioTrack != 0) { 1828 AutoMutex lock(audioTrack->mLock); 1829 audioTrack->mProxy->binderDied(); 1830 } 1831} 1832 1833// ========================================================================= 1834 1835AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1836 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1837 mIgnoreNextPausedInt(false) 1838{ 1839} 1840 1841AudioTrack::AudioTrackThread::~AudioTrackThread() 1842{ 1843} 1844 1845bool AudioTrack::AudioTrackThread::threadLoop() 1846{ 1847 { 1848 AutoMutex _l(mMyLock); 1849 if (mPaused) { 1850 mMyCond.wait(mMyLock); 1851 // caller will check for exitPending() 1852 return true; 1853 } 1854 if (mIgnoreNextPausedInt) { 1855 mIgnoreNextPausedInt = false; 1856 mPausedInt = false; 1857 } 1858 if (mPausedInt) { 1859 if (mPausedNs > 0) { 1860 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1861 } else { 1862 mMyCond.wait(mMyLock); 1863 } 1864 mPausedInt = false; 1865 return true; 1866 } 1867 } 1868 nsecs_t ns = mReceiver.processAudioBuffer(); 1869 switch (ns) { 1870 case 0: 1871 return true; 1872 case NS_INACTIVE: 1873 pauseInternal(); 1874 return true; 1875 case NS_NEVER: 1876 return false; 1877 case NS_WHENEVER: 1878 // FIXME increase poll interval, or make event-driven 1879 ns = 1000000000LL; 1880 // fall through 1881 default: 1882 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1883 pauseInternal(ns); 1884 return true; 1885 } 1886} 1887 1888void AudioTrack::AudioTrackThread::requestExit() 1889{ 1890 // must be in this order to avoid a race condition 1891 Thread::requestExit(); 1892 resume(); 1893} 1894 1895void AudioTrack::AudioTrackThread::pause() 1896{ 1897 AutoMutex _l(mMyLock); 1898 mPaused = true; 1899} 1900 1901void AudioTrack::AudioTrackThread::resume() 1902{ 1903 AutoMutex _l(mMyLock); 1904 mIgnoreNextPausedInt = true; 1905 if (mPaused || mPausedInt) { 1906 mPaused = false; 1907 mPausedInt = false; 1908 mMyCond.signal(); 1909 } 1910} 1911 1912void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1913{ 1914 AutoMutex _l(mMyLock); 1915 mPausedInt = true; 1916 mPausedNs = ns; 1917} 1918 1919}; // namespace android 1920