AudioTrack.cpp revision 43bdc1de363a3c72c7dcf9c9a898bac109dc7cb5
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 mSharedBuffer = sharedBuffer; 226 mTransfer = transferType; 227 228 // FIXME "int" here is legacy and will be replaced by size_t later 229 if (frameCountInt < 0) { 230 ALOGE("Invalid frame count %d", frameCountInt); 231 return BAD_VALUE; 232 } 233 size_t frameCount = frameCountInt; 234 235 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 236 sharedBuffer->size()); 237 238 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 239 240 AutoMutex lock(mLock); 241 242 // invariant that mAudioTrack != 0 is true only after set() returns successfully 243 if (mAudioTrack != 0) { 244 ALOGE("Track already in use"); 245 return INVALID_OPERATION; 246 } 247 248 // handle default values first. 249 if (streamType == AUDIO_STREAM_DEFAULT) { 250 streamType = AUDIO_STREAM_MUSIC; 251 } 252 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 253 ALOGE("Invalid stream type %d", streamType); 254 return BAD_VALUE; 255 } 256 mStreamType = streamType; 257 258 status_t status; 259 if (sampleRate == 0) { 260 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 261 if (status != NO_ERROR) { 262 ALOGE("Could not get output sample rate for stream type %d; status %d", 263 streamType, status); 264 return status; 265 } 266 } 267 mSampleRate = sampleRate; 268 269 // these below should probably come from the audioFlinger too... 270 if (format == AUDIO_FORMAT_DEFAULT) { 271 format = AUDIO_FORMAT_PCM_16_BIT; 272 } 273 274 // validate parameters 275 if (!audio_is_valid_format(format)) { 276 ALOGE("Invalid format %#x", format); 277 return BAD_VALUE; 278 } 279 mFormat = format; 280 281 if (!audio_is_output_channel(channelMask)) { 282 ALOGE("Invalid channel mask %#x", channelMask); 283 return BAD_VALUE; 284 } 285 286 // AudioFlinger does not currently support 8-bit data in shared memory 287 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 288 ALOGE("8-bit data in shared memory is not supported"); 289 return BAD_VALUE; 290 } 291 292 // force direct flag if format is not linear PCM 293 // or offload was requested 294 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 295 || !audio_is_linear_pcm(format)) { 296 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 297 ? "Offload request, forcing to Direct Output" 298 : "Not linear PCM, forcing to Direct Output"); 299 flags = (audio_output_flags_t) 300 // FIXME why can't we allow direct AND fast? 301 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 302 } 303 // only allow deep buffering for music stream type 304 if (streamType != AUDIO_STREAM_MUSIC) { 305 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 306 } 307 308 mChannelMask = channelMask; 309 uint32_t channelCount = popcount(channelMask); 310 mChannelCount = channelCount; 311 312 if (audio_is_linear_pcm(format)) { 313 mFrameSize = channelCount * audio_bytes_per_sample(format); 314 mFrameSizeAF = channelCount * sizeof(int16_t); 315 } else { 316 mFrameSize = sizeof(uint8_t); 317 mFrameSizeAF = sizeof(uint8_t); 318 } 319 320 // Make copy of input parameter offloadInfo so that in the future: 321 // (a) createTrack_l doesn't need it as an input parameter 322 // (b) we can support re-creation of offloaded tracks 323 if (offloadInfo != NULL) { 324 mOffloadInfoCopy = *offloadInfo; 325 mOffloadInfo = &mOffloadInfoCopy; 326 } else { 327 mOffloadInfo = NULL; 328 } 329 330 mVolume[LEFT] = 1.0f; 331 mVolume[RIGHT] = 1.0f; 332 mSendLevel = 0.0f; 333 // mFrameCount is initialized in createTrack_l 334 mReqFrameCount = frameCount; 335 mNotificationFramesReq = notificationFrames; 336 mNotificationFramesAct = 0; 337 mSessionId = sessionId; 338 if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { 339 mClientUid = IPCThreadState::self()->getCallingUid(); 340 } else { 341 mClientUid = uid; 342 } 343 mAuxEffectId = 0; 344 mFlags = flags; 345 mCbf = cbf; 346 347 if (cbf != NULL) { 348 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 349 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 350 } 351 352 // create the IAudioTrack 353 status = createTrack_l(0 /*epoch*/); 354 355 if (status != NO_ERROR) { 356 if (mAudioTrackThread != 0) { 357 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 358 mAudioTrackThread->requestExitAndWait(); 359 mAudioTrackThread.clear(); 360 } 361 // Use of direct and offloaded output streams is ref counted by audio policy manager. 362#if 0 // FIXME This should no longer be needed 363 //Use of direct and offloaded output streams is ref counted by audio policy manager. 364 // As getOutput was called above and resulted in an output stream to be opened, 365 // we need to release it. 366 if (mOutput != 0) { 367 AudioSystem::releaseOutput(mOutput); 368 mOutput = 0; 369 } 370#endif 371 return status; 372 } 373 374 mStatus = NO_ERROR; 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; 386 387 return NO_ERROR; 388} 389 390// ------------------------------------------------------------------------- 391 392status_t AudioTrack::start() 393{ 394 AutoMutex lock(mLock); 395 396 if (mState == STATE_ACTIVE) { 397 return INVALID_OPERATION; 398 } 399 400 mInUnderrun = true; 401 402 State previousState = mState; 403 if (previousState == STATE_PAUSED_STOPPING) { 404 mState = STATE_STOPPING; 405 } else { 406 mState = STATE_ACTIVE; 407 } 408 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 409 // reset current position as seen by client to 0 410 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 411 // force refresh of remaining frames by processAudioBuffer() as last 412 // write before stop could be partial. 413 mRefreshRemaining = true; 414 } 415 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 416 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 417 418 sp<AudioTrackThread> t = mAudioTrackThread; 419 if (t != 0) { 420 if (previousState == STATE_STOPPING) { 421 mProxy->interrupt(); 422 } else { 423 t->resume(); 424 } 425 } else { 426 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 427 get_sched_policy(0, &mPreviousSchedulingGroup); 428 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 429 } 430 431 status_t status = NO_ERROR; 432 if (!(flags & CBLK_INVALID)) { 433 status = mAudioTrack->start(); 434 if (status == DEAD_OBJECT) { 435 flags |= CBLK_INVALID; 436 } 437 } 438 if (flags & CBLK_INVALID) { 439 status = restoreTrack_l("start"); 440 } 441 442 if (status != NO_ERROR) { 443 ALOGE("start() status %d", status); 444 mState = previousState; 445 if (t != 0) { 446 if (previousState != STATE_STOPPING) { 447 t->pause(); 448 } 449 } else { 450 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 451 set_sched_policy(0, mPreviousSchedulingGroup); 452 } 453 } 454 455 return status; 456} 457 458void AudioTrack::stop() 459{ 460 AutoMutex lock(mLock); 461 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 462 return; 463 } 464 465 if (isOffloaded_l()) { 466 mState = STATE_STOPPING; 467 } else { 468 mState = STATE_STOPPED; 469 } 470 471 mProxy->interrupt(); 472 mAudioTrack->stop(); 473 // the playback head position will reset to 0, so if a marker is set, we need 474 // to activate it again 475 mMarkerReached = false; 476#if 0 477 // Force flush if a shared buffer is used otherwise audioflinger 478 // will not stop before end of buffer is reached. 479 // It may be needed to make sure that we stop playback, likely in case looping is on. 480 if (mSharedBuffer != 0) { 481 flush_l(); 482 } 483#endif 484 485 sp<AudioTrackThread> t = mAudioTrackThread; 486 if (t != 0) { 487 if (!isOffloaded_l()) { 488 t->pause(); 489 } 490 } else { 491 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 492 set_sched_policy(0, mPreviousSchedulingGroup); 493 } 494} 495 496bool AudioTrack::stopped() const 497{ 498 AutoMutex lock(mLock); 499 return mState != STATE_ACTIVE; 500} 501 502void AudioTrack::flush() 503{ 504 if (mSharedBuffer != 0) { 505 return; 506 } 507 AutoMutex lock(mLock); 508 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 509 return; 510 } 511 flush_l(); 512} 513 514void AudioTrack::flush_l() 515{ 516 ALOG_ASSERT(mState != STATE_ACTIVE); 517 518 // clear playback marker and periodic update counter 519 mMarkerPosition = 0; 520 mMarkerReached = false; 521 mUpdatePeriod = 0; 522 mRefreshRemaining = true; 523 524 mState = STATE_FLUSHED; 525 if (isOffloaded_l()) { 526 mProxy->interrupt(); 527 } 528 mProxy->flush(); 529 mAudioTrack->flush(); 530} 531 532void AudioTrack::pause() 533{ 534 AutoMutex lock(mLock); 535 if (mState == STATE_ACTIVE) { 536 mState = STATE_PAUSED; 537 } else if (mState == STATE_STOPPING) { 538 mState = STATE_PAUSED_STOPPING; 539 } else { 540 return; 541 } 542 mProxy->interrupt(); 543 mAudioTrack->pause(); 544} 545 546status_t AudioTrack::setVolume(float left, float right) 547{ 548 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 549 return BAD_VALUE; 550 } 551 552 AutoMutex lock(mLock); 553 mVolume[LEFT] = left; 554 mVolume[RIGHT] = right; 555 556 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 557 558 if (isOffloaded_l()) { 559 mAudioTrack->signal(); 560 } 561 return NO_ERROR; 562} 563 564status_t AudioTrack::setVolume(float volume) 565{ 566 return setVolume(volume, volume); 567} 568 569status_t AudioTrack::setAuxEffectSendLevel(float level) 570{ 571 if (level < 0.0f || level > 1.0f) { 572 return BAD_VALUE; 573 } 574 575 AutoMutex lock(mLock); 576 mSendLevel = level; 577 mProxy->setSendLevel(level); 578 579 return NO_ERROR; 580} 581 582void AudioTrack::getAuxEffectSendLevel(float* level) const 583{ 584 if (level != NULL) { 585 *level = mSendLevel; 586 } 587} 588 589status_t AudioTrack::setSampleRate(uint32_t rate) 590{ 591 if (mIsTimed || isOffloaded()) { 592 return INVALID_OPERATION; 593 } 594 595 uint32_t afSamplingRate; 596 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 597 return NO_INIT; 598 } 599 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 600 if (rate == 0 || rate > afSamplingRate*2 ) { 601 return BAD_VALUE; 602 } 603 604 AutoMutex lock(mLock); 605 mSampleRate = rate; 606 mProxy->setSampleRate(rate); 607 608 return NO_ERROR; 609} 610 611uint32_t AudioTrack::getSampleRate() const 612{ 613 if (mIsTimed) { 614 return 0; 615 } 616 617 AutoMutex lock(mLock); 618 619 // sample rate can be updated during playback by the offloaded decoder so we need to 620 // query the HAL and update if needed. 621// FIXME use Proxy return channel to update the rate from server and avoid polling here 622 if (isOffloaded_l()) { 623 if (mOutput != 0) { 624 uint32_t sampleRate = 0; 625 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 626 if (status == NO_ERROR) { 627 mSampleRate = sampleRate; 628 } 629 } 630 } 631 return mSampleRate; 632} 633 634status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 635{ 636 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 637 return INVALID_OPERATION; 638 } 639 640 if (loopCount == 0) { 641 ; 642 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 643 loopEnd - loopStart >= MIN_LOOP) { 644 ; 645 } else { 646 return BAD_VALUE; 647 } 648 649 AutoMutex lock(mLock); 650 // See setPosition() regarding setting parameters such as loop points or position while active 651 if (mState == STATE_ACTIVE) { 652 return INVALID_OPERATION; 653 } 654 setLoop_l(loopStart, loopEnd, loopCount); 655 return NO_ERROR; 656} 657 658void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 659{ 660 // FIXME If setting a loop also sets position to start of loop, then 661 // this is correct. Otherwise it should be removed. 662 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 663 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 664 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 665} 666 667status_t AudioTrack::setMarkerPosition(uint32_t marker) 668{ 669 // The only purpose of setting marker position is to get a callback 670 if (mCbf == NULL || isOffloaded()) { 671 return INVALID_OPERATION; 672 } 673 674 AutoMutex lock(mLock); 675 mMarkerPosition = marker; 676 mMarkerReached = false; 677 678 return NO_ERROR; 679} 680 681status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 682{ 683 if (isOffloaded()) { 684 return INVALID_OPERATION; 685 } 686 if (marker == NULL) { 687 return BAD_VALUE; 688 } 689 690 AutoMutex lock(mLock); 691 *marker = mMarkerPosition; 692 693 return NO_ERROR; 694} 695 696status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 697{ 698 // The only purpose of setting position update period is to get a callback 699 if (mCbf == NULL || isOffloaded()) { 700 return INVALID_OPERATION; 701 } 702 703 AutoMutex lock(mLock); 704 mNewPosition = mProxy->getPosition() + updatePeriod; 705 mUpdatePeriod = updatePeriod; 706 707 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() const 810{ 811 AutoMutex lock(mLock); 812 return mOutput; 813} 814 815status_t AudioTrack::attachAuxEffect(int effectId) 816{ 817 AutoMutex lock(mLock); 818 status_t status = mAudioTrack->attachAuxEffect(effectId); 819 if (status == NO_ERROR) { 820 mAuxEffectId = effectId; 821 } 822 return status; 823} 824 825// ------------------------------------------------------------------------- 826 827// must be called with mLock held 828status_t AudioTrack::createTrack_l(size_t epoch) 829{ 830 status_t status; 831 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 832 if (audioFlinger == 0) { 833 ALOGE("Could not get audioflinger"); 834 return NO_INIT; 835 } 836 837 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 838 mChannelMask, mFlags, mOffloadInfo); 839 if (output == 0) { 840 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 841 "channel mask %#x, flags %#x", 842 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 843 return BAD_VALUE; 844 } 845 { 846 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 847 // we must release it ourselves if anything goes wrong. 848 849 // Not all of these values are needed under all conditions, but it is easier to get them all 850 851 uint32_t afLatency; 852 status = AudioSystem::getLatency(output, mStreamType, &afLatency); 853 if (status != NO_ERROR) { 854 ALOGE("getLatency(%d) failed status %d", output, status); 855 goto release; 856 } 857 858 size_t afFrameCount; 859 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 860 if (status != NO_ERROR) { 861 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 862 goto release; 863 } 864 865 uint32_t afSampleRate; 866 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 867 if (status != NO_ERROR) { 868 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 869 goto release; 870 } 871 872 // Client decides whether the track is TIMED (see below), but can only express a preference 873 // for FAST. Server will perform additional tests. 874 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 875 // either of these use cases: 876 // use case 1: shared buffer 877 (mSharedBuffer != 0) || 878 // use case 2: callback handler 879 (mCbf != NULL)) && 880 // matching sample rate 881 (mSampleRate == afSampleRate))) { 882 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 883 // once denied, do not request again if IAudioTrack is re-created 884 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 885 } 886 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 887 888 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 889 // n = 1 fast track with single buffering; nBuffering is ignored 890 // n = 2 fast track with double buffering 891 // n = 2 normal track, no sample rate conversion 892 // n = 3 normal track, with sample rate conversion 893 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 894 // n > 3 very high latency or very small notification interval; nBuffering is ignored 895 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 896 897 mNotificationFramesAct = mNotificationFramesReq; 898 899 size_t frameCount = mReqFrameCount; 900 if (!audio_is_linear_pcm(mFormat)) { 901 902 if (mSharedBuffer != 0) { 903 // Same comment as below about ignoring frameCount parameter for set() 904 frameCount = mSharedBuffer->size(); 905 } else if (frameCount == 0) { 906 frameCount = afFrameCount; 907 } 908 if (mNotificationFramesAct != frameCount) { 909 mNotificationFramesAct = frameCount; 910 } 911 } else if (mSharedBuffer != 0) { 912 913 // Ensure that buffer alignment matches channel count 914 // 8-bit data in shared memory is not currently supported by AudioFlinger 915 size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 916 if (mChannelCount > 1) { 917 // More than 2 channels does not require stronger alignment than stereo 918 alignment <<= 1; 919 } 920 if (((size_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 921 ALOGE("Invalid buffer alignment: address %p, channel count %u", 922 mSharedBuffer->pointer(), mChannelCount); 923 status = BAD_VALUE; 924 goto release; 925 } 926 927 // When initializing a shared buffer AudioTrack via constructors, 928 // there's no frameCount parameter. 929 // But when initializing a shared buffer AudioTrack via set(), 930 // there _is_ a frameCount parameter. We silently ignore it. 931 frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); 932 933 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 934 935 // FIXME move these calculations and associated checks to server 936 937 // Ensure that buffer depth covers at least audio hardware latency 938 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 939 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 940 afFrameCount, minBufCount, afSampleRate, afLatency); 941 if (minBufCount <= nBuffering) { 942 minBufCount = nBuffering; 943 } 944 945 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 946 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 947 ", afLatency=%d", 948 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 949 950 if (frameCount == 0) { 951 frameCount = minFrameCount; 952 } else if (frameCount < minFrameCount) { 953 // not ALOGW because it happens all the time when playing key clicks over A2DP 954 ALOGV("Minimum buffer size corrected from %d to %d", 955 frameCount, minFrameCount); 956 frameCount = minFrameCount; 957 } 958 // Make sure that application is notified with sufficient margin before underrun 959 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 960 mNotificationFramesAct = frameCount/nBuffering; 961 } 962 963 } else { 964 // For fast tracks, the frame count calculations and checks are done by server 965 } 966 967 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 968 if (mIsTimed) { 969 trackFlags |= IAudioFlinger::TRACK_TIMED; 970 } 971 972 pid_t tid = -1; 973 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 974 trackFlags |= IAudioFlinger::TRACK_FAST; 975 if (mAudioTrackThread != 0) { 976 tid = mAudioTrackThread->getTid(); 977 } 978 } 979 980 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 981 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 982 } 983 984 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 985 // but we will still need the original value also 986 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 987 mSampleRate, 988 // AudioFlinger only sees 16-bit PCM 989 mFormat == AUDIO_FORMAT_PCM_8_BIT ? 990 AUDIO_FORMAT_PCM_16_BIT : mFormat, 991 mChannelMask, 992 &temp, 993 &trackFlags, 994 mSharedBuffer, 995 output, 996 tid, 997 &mSessionId, 998 mName, 999 mClientUid, 1000 &status); 1001 1002 if (track == 0) { 1003 ALOGE("AudioFlinger could not create track, status: %d", status); 1004 goto release; 1005 } 1006 // AudioFlinger now owns the reference to the I/O handle, 1007 // so we are no longer responsible for releasing it. 1008 1009 sp<IMemory> iMem = track->getCblk(); 1010 if (iMem == 0) { 1011 ALOGE("Could not get control block"); 1012 return NO_INIT; 1013 } 1014 void *iMemPointer = iMem->pointer(); 1015 if (iMemPointer == NULL) { 1016 ALOGE("Could not get control block pointer"); 1017 return NO_INIT; 1018 } 1019 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1020 if (mAudioTrack != 0) { 1021 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1022 mDeathNotifier.clear(); 1023 } 1024 mAudioTrack = track; 1025 mCblkMemory = iMem; 1026 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1027 mCblk = cblk; 1028 // note that temp is the (possibly revised) value of frameCount 1029 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1030 // In current design, AudioTrack client checks and ensures frame count validity before 1031 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1032 // for fast track as it uses a special method of assigning frame count. 1033 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1034 } 1035 frameCount = temp; 1036 mAwaitBoost = false; 1037 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1038 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1039 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1040 mAwaitBoost = true; 1041 if (mSharedBuffer == 0) { 1042 // Theoretically double-buffering is not required for fast tracks, 1043 // due to tighter scheduling. But in practice, to accommodate kernels with 1044 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1045 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1046 mNotificationFramesAct = frameCount/nBuffering; 1047 } 1048 } 1049 } else { 1050 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1051 // once denied, do not request again if IAudioTrack is re-created 1052 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1053 if (mSharedBuffer == 0) { 1054 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1055 mNotificationFramesAct = frameCount/nBuffering; 1056 } 1057 } 1058 } 1059 } 1060 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1061 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1062 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1063 } else { 1064 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1065 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1066 // FIXME This is a warning, not an error, so don't return error status 1067 //return NO_INIT; 1068 } 1069 } 1070 1071 // We retain a copy of the I/O handle, but don't own the reference 1072 mOutput = output; 1073 mRefreshRemaining = true; 1074 1075 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1076 // is the value of pointer() for the shared buffer, otherwise buffers points 1077 // immediately after the control block. This address is for the mapping within client 1078 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1079 void* buffers; 1080 if (mSharedBuffer == 0) { 1081 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1082 } else { 1083 buffers = mSharedBuffer->pointer(); 1084 } 1085 1086 mAudioTrack->attachAuxEffect(mAuxEffectId); 1087 // FIXME don't believe this lie 1088 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1089 mFrameCount = frameCount; 1090 // If IAudioTrack is re-created, don't let the requested frameCount 1091 // decrease. This can confuse clients that cache frameCount(). 1092 if (frameCount > mReqFrameCount) { 1093 mReqFrameCount = frameCount; 1094 } 1095 1096 // update proxy 1097 if (mSharedBuffer == 0) { 1098 mStaticProxy.clear(); 1099 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1100 } else { 1101 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1102 mProxy = mStaticProxy; 1103 } 1104 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 1105 uint16_t(mVolume[LEFT] * 0x1000)); 1106 mProxy->setSendLevel(mSendLevel); 1107 mProxy->setSampleRate(mSampleRate); 1108 mProxy->setEpoch(epoch); 1109 mProxy->setMinimum(mNotificationFramesAct); 1110 1111 mDeathNotifier = new DeathNotifier(this); 1112 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1113 1114 return NO_ERROR; 1115 } 1116 1117release: 1118 AudioSystem::releaseOutput(output); 1119 if (status == NO_ERROR) { 1120 status = NO_INIT; 1121 } 1122 return status; 1123} 1124 1125status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1126{ 1127 if (audioBuffer == NULL) { 1128 return BAD_VALUE; 1129 } 1130 if (mTransfer != TRANSFER_OBTAIN) { 1131 audioBuffer->frameCount = 0; 1132 audioBuffer->size = 0; 1133 audioBuffer->raw = NULL; 1134 return INVALID_OPERATION; 1135 } 1136 1137 const struct timespec *requested; 1138 struct timespec timeout; 1139 if (waitCount == -1) { 1140 requested = &ClientProxy::kForever; 1141 } else if (waitCount == 0) { 1142 requested = &ClientProxy::kNonBlocking; 1143 } else if (waitCount > 0) { 1144 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1145 timeout.tv_sec = ms / 1000; 1146 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1147 requested = &timeout; 1148 } else { 1149 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1150 requested = NULL; 1151 } 1152 return obtainBuffer(audioBuffer, requested); 1153} 1154 1155status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1156 struct timespec *elapsed, size_t *nonContig) 1157{ 1158 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1159 uint32_t oldSequence = 0; 1160 uint32_t newSequence; 1161 1162 Proxy::Buffer buffer; 1163 status_t status = NO_ERROR; 1164 1165 static const int32_t kMaxTries = 5; 1166 int32_t tryCounter = kMaxTries; 1167 1168 do { 1169 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1170 // keep them from going away if another thread re-creates the track during obtainBuffer() 1171 sp<AudioTrackClientProxy> proxy; 1172 sp<IMemory> iMem; 1173 1174 { // start of lock scope 1175 AutoMutex lock(mLock); 1176 1177 newSequence = mSequence; 1178 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1179 if (status == DEAD_OBJECT) { 1180 // re-create track, unless someone else has already done so 1181 if (newSequence == oldSequence) { 1182 status = restoreTrack_l("obtainBuffer"); 1183 if (status != NO_ERROR) { 1184 buffer.mFrameCount = 0; 1185 buffer.mRaw = NULL; 1186 buffer.mNonContig = 0; 1187 break; 1188 } 1189 } 1190 } 1191 oldSequence = newSequence; 1192 1193 // Keep the extra references 1194 proxy = mProxy; 1195 iMem = mCblkMemory; 1196 1197 if (mState == STATE_STOPPING) { 1198 status = -EINTR; 1199 buffer.mFrameCount = 0; 1200 buffer.mRaw = NULL; 1201 buffer.mNonContig = 0; 1202 break; 1203 } 1204 1205 // Non-blocking if track is stopped or paused 1206 if (mState != STATE_ACTIVE) { 1207 requested = &ClientProxy::kNonBlocking; 1208 } 1209 1210 } // end of lock scope 1211 1212 buffer.mFrameCount = audioBuffer->frameCount; 1213 // FIXME starts the requested timeout and elapsed over from scratch 1214 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1215 1216 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1217 1218 audioBuffer->frameCount = buffer.mFrameCount; 1219 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1220 audioBuffer->raw = buffer.mRaw; 1221 if (nonContig != NULL) { 1222 *nonContig = buffer.mNonContig; 1223 } 1224 return status; 1225} 1226 1227void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1228{ 1229 if (mTransfer == TRANSFER_SHARED) { 1230 return; 1231 } 1232 1233 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1234 if (stepCount == 0) { 1235 return; 1236 } 1237 1238 Proxy::Buffer buffer; 1239 buffer.mFrameCount = stepCount; 1240 buffer.mRaw = audioBuffer->raw; 1241 1242 AutoMutex lock(mLock); 1243 mInUnderrun = false; 1244 mProxy->releaseBuffer(&buffer); 1245 1246 // restart track if it was disabled by audioflinger due to previous underrun 1247 if (mState == STATE_ACTIVE) { 1248 audio_track_cblk_t* cblk = mCblk; 1249 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1250 ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", 1251 this, mName.string()); 1252 // FIXME ignoring status 1253 mAudioTrack->start(); 1254 } 1255 } 1256} 1257 1258// ------------------------------------------------------------------------- 1259 1260ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1261{ 1262 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1263 return INVALID_OPERATION; 1264 } 1265 1266 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1267 // Sanity-check: user is most-likely passing an error code, and it would 1268 // make the return value ambiguous (actualSize vs error). 1269 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1270 return BAD_VALUE; 1271 } 1272 1273 size_t written = 0; 1274 Buffer audioBuffer; 1275 1276 while (userSize >= mFrameSize) { 1277 audioBuffer.frameCount = userSize / mFrameSize; 1278 1279 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1280 if (err < 0) { 1281 if (written > 0) { 1282 break; 1283 } 1284 return ssize_t(err); 1285 } 1286 1287 size_t toWrite; 1288 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1289 // Divide capacity by 2 to take expansion into account 1290 toWrite = audioBuffer.size >> 1; 1291 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1292 } else { 1293 toWrite = audioBuffer.size; 1294 memcpy(audioBuffer.i8, buffer, toWrite); 1295 } 1296 buffer = ((const char *) buffer) + toWrite; 1297 userSize -= toWrite; 1298 written += toWrite; 1299 1300 releaseBuffer(&audioBuffer); 1301 } 1302 1303 return written; 1304} 1305 1306// ------------------------------------------------------------------------- 1307 1308TimedAudioTrack::TimedAudioTrack() { 1309 mIsTimed = true; 1310} 1311 1312status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1313{ 1314 AutoMutex lock(mLock); 1315 status_t result = UNKNOWN_ERROR; 1316 1317#if 1 1318 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1319 // while we are accessing the cblk 1320 sp<IAudioTrack> audioTrack = mAudioTrack; 1321 sp<IMemory> iMem = mCblkMemory; 1322#endif 1323 1324 // If the track is not invalid already, try to allocate a buffer. alloc 1325 // fails indicating that the server is dead, flag the track as invalid so 1326 // we can attempt to restore in just a bit. 1327 audio_track_cblk_t* cblk = mCblk; 1328 if (!(cblk->mFlags & CBLK_INVALID)) { 1329 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1330 if (result == DEAD_OBJECT) { 1331 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1332 } 1333 } 1334 1335 // If the track is invalid at this point, attempt to restore it. and try the 1336 // allocation one more time. 1337 if (cblk->mFlags & CBLK_INVALID) { 1338 result = restoreTrack_l("allocateTimedBuffer"); 1339 1340 if (result == NO_ERROR) { 1341 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1342 } 1343 } 1344 1345 return result; 1346} 1347 1348status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1349 int64_t pts) 1350{ 1351 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1352 { 1353 AutoMutex lock(mLock); 1354 audio_track_cblk_t* cblk = mCblk; 1355 // restart track if it was disabled by audioflinger due to previous underrun 1356 if (buffer->size() != 0 && status == NO_ERROR && 1357 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1358 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1359 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1360 // FIXME ignoring status 1361 mAudioTrack->start(); 1362 } 1363 } 1364 return status; 1365} 1366 1367status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1368 TargetTimeline target) 1369{ 1370 return mAudioTrack->setMediaTimeTransform(xform, target); 1371} 1372 1373// ------------------------------------------------------------------------- 1374 1375nsecs_t AudioTrack::processAudioBuffer() 1376{ 1377 // Currently the AudioTrack thread is not created if there are no callbacks. 1378 // Would it ever make sense to run the thread, even without callbacks? 1379 // If so, then replace this by checks at each use for mCbf != NULL. 1380 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1381 1382 mLock.lock(); 1383 if (mAwaitBoost) { 1384 mAwaitBoost = false; 1385 mLock.unlock(); 1386 static const int32_t kMaxTries = 5; 1387 int32_t tryCounter = kMaxTries; 1388 uint32_t pollUs = 10000; 1389 do { 1390 int policy = sched_getscheduler(0); 1391 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1392 break; 1393 } 1394 usleep(pollUs); 1395 pollUs <<= 1; 1396 } while (tryCounter-- > 0); 1397 if (tryCounter < 0) { 1398 ALOGE("did not receive expected priority boost on time"); 1399 } 1400 // Run again immediately 1401 return 0; 1402 } 1403 1404 // Can only reference mCblk while locked 1405 int32_t flags = android_atomic_and( 1406 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1407 1408 // Check for track invalidation 1409 if (flags & CBLK_INVALID) { 1410 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1411 // AudioSystem cache. We should not exit here but after calling the callback so 1412 // that the upper layers can recreate the track 1413 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1414 status_t status = restoreTrack_l("processAudioBuffer"); 1415 mLock.unlock(); 1416 // Run again immediately, but with a new IAudioTrack 1417 return 0; 1418 } 1419 } 1420 1421 bool waitStreamEnd = mState == STATE_STOPPING; 1422 bool active = mState == STATE_ACTIVE; 1423 1424 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1425 bool newUnderrun = false; 1426 if (flags & CBLK_UNDERRUN) { 1427#if 0 1428 // Currently in shared buffer mode, when the server reaches the end of buffer, 1429 // the track stays active in continuous underrun state. It's up to the application 1430 // to pause or stop the track, or set the position to a new offset within buffer. 1431 // This was some experimental code to auto-pause on underrun. Keeping it here 1432 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1433 if (mTransfer == TRANSFER_SHARED) { 1434 mState = STATE_PAUSED; 1435 active = false; 1436 } 1437#endif 1438 if (!mInUnderrun) { 1439 mInUnderrun = true; 1440 newUnderrun = true; 1441 } 1442 } 1443 1444 // Get current position of server 1445 size_t position = mProxy->getPosition(); 1446 1447 // Manage marker callback 1448 bool markerReached = false; 1449 size_t markerPosition = mMarkerPosition; 1450 // FIXME fails for wraparound, need 64 bits 1451 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1452 mMarkerReached = markerReached = true; 1453 } 1454 1455 // Determine number of new position callback(s) that will be needed, while locked 1456 size_t newPosCount = 0; 1457 size_t newPosition = mNewPosition; 1458 size_t updatePeriod = mUpdatePeriod; 1459 // FIXME fails for wraparound, need 64 bits 1460 if (updatePeriod > 0 && position >= newPosition) { 1461 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1462 mNewPosition += updatePeriod * newPosCount; 1463 } 1464 1465 // Cache other fields that will be needed soon 1466 uint32_t loopPeriod = mLoopPeriod; 1467 uint32_t sampleRate = mSampleRate; 1468 size_t notificationFrames = mNotificationFramesAct; 1469 if (mRefreshRemaining) { 1470 mRefreshRemaining = false; 1471 mRemainingFrames = notificationFrames; 1472 mRetryOnPartialBuffer = false; 1473 } 1474 size_t misalignment = mProxy->getMisalignment(); 1475 uint32_t sequence = mSequence; 1476 1477 // These fields don't need to be cached, because they are assigned only by set(): 1478 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1479 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1480 1481 mLock.unlock(); 1482 1483 if (waitStreamEnd) { 1484 AutoMutex lock(mLock); 1485 1486 sp<AudioTrackClientProxy> proxy = mProxy; 1487 sp<IMemory> iMem = mCblkMemory; 1488 1489 struct timespec timeout; 1490 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1491 timeout.tv_nsec = 0; 1492 1493 mLock.unlock(); 1494 status_t status = mProxy->waitStreamEndDone(&timeout); 1495 mLock.lock(); 1496 switch (status) { 1497 case NO_ERROR: 1498 case DEAD_OBJECT: 1499 case TIMED_OUT: 1500 mLock.unlock(); 1501 mCbf(EVENT_STREAM_END, mUserData, NULL); 1502 mLock.lock(); 1503 if (mState == STATE_STOPPING) { 1504 mState = STATE_STOPPED; 1505 if (status != DEAD_OBJECT) { 1506 return NS_INACTIVE; 1507 } 1508 } 1509 return 0; 1510 default: 1511 return 0; 1512 } 1513 } 1514 1515 // perform callbacks while unlocked 1516 if (newUnderrun) { 1517 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1518 } 1519 // FIXME we will miss loops if loop cycle was signaled several times since last call 1520 // to processAudioBuffer() 1521 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1522 mCbf(EVENT_LOOP_END, mUserData, NULL); 1523 } 1524 if (flags & CBLK_BUFFER_END) { 1525 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1526 } 1527 if (markerReached) { 1528 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1529 } 1530 while (newPosCount > 0) { 1531 size_t temp = newPosition; 1532 mCbf(EVENT_NEW_POS, mUserData, &temp); 1533 newPosition += updatePeriod; 1534 newPosCount--; 1535 } 1536 1537 if (mObservedSequence != sequence) { 1538 mObservedSequence = sequence; 1539 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1540 // for offloaded tracks, just wait for the upper layers to recreate the track 1541 if (isOffloaded()) { 1542 return NS_INACTIVE; 1543 } 1544 } 1545 1546 // if inactive, then don't run me again until re-started 1547 if (!active) { 1548 return NS_INACTIVE; 1549 } 1550 1551 // Compute the estimated time until the next timed event (position, markers, loops) 1552 // FIXME only for non-compressed audio 1553 uint32_t minFrames = ~0; 1554 if (!markerReached && position < markerPosition) { 1555 minFrames = markerPosition - position; 1556 } 1557 if (loopPeriod > 0 && loopPeriod < minFrames) { 1558 minFrames = loopPeriod; 1559 } 1560 if (updatePeriod > 0 && updatePeriod < minFrames) { 1561 minFrames = updatePeriod; 1562 } 1563 1564 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1565 static const uint32_t kPoll = 0; 1566 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1567 minFrames = kPoll * notificationFrames; 1568 } 1569 1570 // Convert frame units to time units 1571 nsecs_t ns = NS_WHENEVER; 1572 if (minFrames != (uint32_t) ~0) { 1573 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1574 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1575 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1576 } 1577 1578 // If not supplying data by EVENT_MORE_DATA, then we're done 1579 if (mTransfer != TRANSFER_CALLBACK) { 1580 return ns; 1581 } 1582 1583 struct timespec timeout; 1584 const struct timespec *requested = &ClientProxy::kForever; 1585 if (ns != NS_WHENEVER) { 1586 timeout.tv_sec = ns / 1000000000LL; 1587 timeout.tv_nsec = ns % 1000000000LL; 1588 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1589 requested = &timeout; 1590 } 1591 1592 while (mRemainingFrames > 0) { 1593 1594 Buffer audioBuffer; 1595 audioBuffer.frameCount = mRemainingFrames; 1596 size_t nonContig; 1597 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1598 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1599 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1600 requested = &ClientProxy::kNonBlocking; 1601 size_t avail = audioBuffer.frameCount + nonContig; 1602 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1603 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1604 if (err != NO_ERROR) { 1605 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1606 (isOffloaded() && (err == DEAD_OBJECT))) { 1607 return 0; 1608 } 1609 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1610 return NS_NEVER; 1611 } 1612 1613 if (mRetryOnPartialBuffer && !isOffloaded()) { 1614 mRetryOnPartialBuffer = false; 1615 if (avail < mRemainingFrames) { 1616 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1617 if (ns < 0 || myns < ns) { 1618 ns = myns; 1619 } 1620 return ns; 1621 } 1622 } 1623 1624 // Divide buffer size by 2 to take into account the expansion 1625 // due to 8 to 16 bit conversion: the callback must fill only half 1626 // of the destination buffer 1627 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1628 audioBuffer.size >>= 1; 1629 } 1630 1631 size_t reqSize = audioBuffer.size; 1632 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1633 size_t writtenSize = audioBuffer.size; 1634 1635 // Sanity check on returned size 1636 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1637 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1638 reqSize, (int) writtenSize); 1639 return NS_NEVER; 1640 } 1641 1642 if (writtenSize == 0) { 1643 // The callback is done filling buffers 1644 // Keep this thread going to handle timed events and 1645 // still try to get more data in intervals of WAIT_PERIOD_MS 1646 // but don't just loop and block the CPU, so wait 1647 return WAIT_PERIOD_MS * 1000000LL; 1648 } 1649 1650 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1651 // 8 to 16 bit conversion, note that source and destination are the same address 1652 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1653 audioBuffer.size <<= 1; 1654 } 1655 1656 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1657 audioBuffer.frameCount = releasedFrames; 1658 mRemainingFrames -= releasedFrames; 1659 if (misalignment >= releasedFrames) { 1660 misalignment -= releasedFrames; 1661 } else { 1662 misalignment = 0; 1663 } 1664 1665 releaseBuffer(&audioBuffer); 1666 1667 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1668 // if callback doesn't like to accept the full chunk 1669 if (writtenSize < reqSize) { 1670 continue; 1671 } 1672 1673 // There could be enough non-contiguous frames available to satisfy the remaining request 1674 if (mRemainingFrames <= nonContig) { 1675 continue; 1676 } 1677 1678#if 0 1679 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1680 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1681 // that total to a sum == notificationFrames. 1682 if (0 < misalignment && misalignment <= mRemainingFrames) { 1683 mRemainingFrames = misalignment; 1684 return (mRemainingFrames * 1100000000LL) / sampleRate; 1685 } 1686#endif 1687 1688 } 1689 mRemainingFrames = notificationFrames; 1690 mRetryOnPartialBuffer = true; 1691 1692 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1693 return 0; 1694} 1695 1696status_t AudioTrack::restoreTrack_l(const char *from) 1697{ 1698 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1699 isOffloaded_l() ? "Offloaded" : "PCM", from); 1700 ++mSequence; 1701 status_t result; 1702 1703 // refresh the audio configuration cache in this process to make sure we get new 1704 // output parameters in createTrack_l() 1705 AudioSystem::clearAudioConfigCache(); 1706 1707 if (isOffloaded_l()) { 1708 // FIXME re-creation of offloaded tracks is not yet implemented 1709 return DEAD_OBJECT; 1710 } 1711 1712 // if the new IAudioTrack is created, createTrack_l() will modify the 1713 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1714 // It will also delete the strong references on previous IAudioTrack and IMemory 1715 1716 // take the frames that will be lost by track recreation into account in saved position 1717 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1718 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1719 result = createTrack_l(position /*epoch*/); 1720 1721 if (result == NO_ERROR) { 1722 // continue playback from last known position, but 1723 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1724 if (mStaticProxy != NULL) { 1725 mLoopPeriod = 0; 1726 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1727 } 1728 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1729 // track destruction have been played? This is critical for SoundPool implementation 1730 // This must be broken, and needs to be tested/debugged. 1731#if 0 1732 // restore write index and set other indexes to reflect empty buffer status 1733 if (!strcmp(from, "start")) { 1734 // Make sure that a client relying on callback events indicating underrun or 1735 // the actual amount of audio frames played (e.g SoundPool) receives them. 1736 if (mSharedBuffer == 0) { 1737 // restart playback even if buffer is not completely filled. 1738 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1739 } 1740 } 1741#endif 1742 if (mState == STATE_ACTIVE) { 1743 result = mAudioTrack->start(); 1744 } 1745 } 1746 if (result != NO_ERROR) { 1747 // Use of direct and offloaded output streams is ref counted by audio policy manager. 1748#if 0 // FIXME This should no longer be needed 1749 //Use of direct and offloaded output streams is ref counted by audio policy manager. 1750 // As getOutput was called above and resulted in an output stream to be opened, 1751 // we need to release it. 1752 if (mOutput != 0) { 1753 AudioSystem::releaseOutput(mOutput); 1754 mOutput = 0; 1755 } 1756#endif 1757 ALOGW("restoreTrack_l() failed status %d", result); 1758 mState = STATE_STOPPED; 1759 } 1760 1761 return result; 1762} 1763 1764status_t AudioTrack::setParameters(const String8& keyValuePairs) 1765{ 1766 AutoMutex lock(mLock); 1767 return mAudioTrack->setParameters(keyValuePairs); 1768} 1769 1770status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1771{ 1772 AutoMutex lock(mLock); 1773 // FIXME not implemented for fast tracks; should use proxy and SSQ 1774 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1775 return INVALID_OPERATION; 1776 } 1777 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1778 return INVALID_OPERATION; 1779 } 1780 status_t status = mAudioTrack->getTimestamp(timestamp); 1781 if (status == NO_ERROR) { 1782 timestamp.mPosition += mProxy->getEpoch(); 1783 } 1784 return status; 1785} 1786 1787String8 AudioTrack::getParameters(const String8& keys) 1788{ 1789 audio_io_handle_t output = getOutput(); 1790 if (output != 0) { 1791 return AudioSystem::getParameters(output, keys); 1792 } else { 1793 return String8::empty(); 1794 } 1795} 1796 1797bool AudioTrack::isOffloaded() const 1798{ 1799 AutoMutex lock(mLock); 1800 return isOffloaded_l(); 1801} 1802 1803status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1804{ 1805 1806 const size_t SIZE = 256; 1807 char buffer[SIZE]; 1808 String8 result; 1809 1810 result.append(" AudioTrack::dump\n"); 1811 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1812 mVolume[0], mVolume[1]); 1813 result.append(buffer); 1814 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1815 mChannelCount, mFrameCount); 1816 result.append(buffer); 1817 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1818 result.append(buffer); 1819 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1820 result.append(buffer); 1821 ::write(fd, result.string(), result.size()); 1822 return NO_ERROR; 1823} 1824 1825uint32_t AudioTrack::getUnderrunFrames() const 1826{ 1827 AutoMutex lock(mLock); 1828 return mProxy->getUnderrunFrames(); 1829} 1830 1831// ========================================================================= 1832 1833void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1834{ 1835 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1836 if (audioTrack != 0) { 1837 AutoMutex lock(audioTrack->mLock); 1838 audioTrack->mProxy->binderDied(); 1839 } 1840} 1841 1842// ========================================================================= 1843 1844AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1845 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1846 mIgnoreNextPausedInt(false) 1847{ 1848} 1849 1850AudioTrack::AudioTrackThread::~AudioTrackThread() 1851{ 1852} 1853 1854bool AudioTrack::AudioTrackThread::threadLoop() 1855{ 1856 { 1857 AutoMutex _l(mMyLock); 1858 if (mPaused) { 1859 mMyCond.wait(mMyLock); 1860 // caller will check for exitPending() 1861 return true; 1862 } 1863 if (mIgnoreNextPausedInt) { 1864 mIgnoreNextPausedInt = false; 1865 mPausedInt = false; 1866 } 1867 if (mPausedInt) { 1868 if (mPausedNs > 0) { 1869 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1870 } else { 1871 mMyCond.wait(mMyLock); 1872 } 1873 mPausedInt = false; 1874 return true; 1875 } 1876 } 1877 nsecs_t ns = mReceiver.processAudioBuffer(); 1878 switch (ns) { 1879 case 0: 1880 return true; 1881 case NS_INACTIVE: 1882 pauseInternal(); 1883 return true; 1884 case NS_NEVER: 1885 return false; 1886 case NS_WHENEVER: 1887 // FIXME increase poll interval, or make event-driven 1888 ns = 1000000000LL; 1889 // fall through 1890 default: 1891 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1892 pauseInternal(ns); 1893 return true; 1894 } 1895} 1896 1897void AudioTrack::AudioTrackThread::requestExit() 1898{ 1899 // must be in this order to avoid a race condition 1900 Thread::requestExit(); 1901 resume(); 1902} 1903 1904void AudioTrack::AudioTrackThread::pause() 1905{ 1906 AutoMutex _l(mMyLock); 1907 mPaused = true; 1908} 1909 1910void AudioTrack::AudioTrackThread::resume() 1911{ 1912 AutoMutex _l(mMyLock); 1913 mIgnoreNextPausedInt = true; 1914 if (mPaused || mPausedInt) { 1915 mPaused = false; 1916 mPausedInt = false; 1917 mMyCond.signal(); 1918 } 1919} 1920 1921void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1922{ 1923 AutoMutex _l(mMyLock); 1924 mPausedInt = true; 1925 mPausedNs = ns; 1926} 1927 1928}; // namespace android 1929