Tracks.cpp revision 5baf2af52cd186633b7173196c1e4a4cd3435f22
1/* 2** 3** Copyright 2012, 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_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <math.h> 24#include <utils/Log.h> 25 26#include <private/media/AudioTrackShared.h> 27 28#include <common_time/cc_helper.h> 29#include <common_time/local_clock.h> 30 31#include "AudioMixer.h" 32#include "AudioFlinger.h" 33#include "ServiceUtilities.h" 34 35#include <media/nbaio/Pipe.h> 36#include <media/nbaio/PipeReader.h> 37 38// ---------------------------------------------------------------------------- 39 40// Note: the following macro is used for extremely verbose logging message. In 41// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 42// 0; but one side effect of this is to turn all LOGV's as well. Some messages 43// are so verbose that we want to suppress them even when we have ALOG_ASSERT 44// turned on. Do not uncomment the #def below unless you really know what you 45// are doing and want to see all of the extremely verbose messages. 46//#define VERY_VERY_VERBOSE_LOGGING 47#ifdef VERY_VERY_VERBOSE_LOGGING 48#define ALOGVV ALOGV 49#else 50#define ALOGVV(a...) do { } while(0) 51#endif 52 53namespace android { 54 55// ---------------------------------------------------------------------------- 56// TrackBase 57// ---------------------------------------------------------------------------- 58 59static volatile int32_t nextTrackId = 55; 60 61// TrackBase constructor must be called with AudioFlinger::mLock held 62AudioFlinger::ThreadBase::TrackBase::TrackBase( 63 ThreadBase *thread, 64 const sp<Client>& client, 65 uint32_t sampleRate, 66 audio_format_t format, 67 audio_channel_mask_t channelMask, 68 size_t frameCount, 69 const sp<IMemory>& sharedBuffer, 70 int sessionId, 71 bool isOut) 72 : RefBase(), 73 mThread(thread), 74 mClient(client), 75 mCblk(NULL), 76 // mBuffer 77 mState(IDLE), 78 mSampleRate(sampleRate), 79 mFormat(format), 80 mChannelMask(channelMask), 81 mChannelCount(popcount(channelMask)), 82 mFrameSize(audio_is_linear_pcm(format) ? 83 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 84 mFrameCount(frameCount), 85 mSessionId(sessionId), 86 mIsOut(isOut), 87 mServerProxy(NULL), 88 mId(android_atomic_inc(&nextTrackId)), 89 mTerminated(false) 90{ 91 // client == 0 implies sharedBuffer == 0 92 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 93 94 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 95 sharedBuffer->size()); 96 97 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 98 size_t size = sizeof(audio_track_cblk_t); 99 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 100 if (sharedBuffer == 0) { 101 size += bufferSize; 102 } 103 104 if (client != 0) { 105 mCblkMemory = client->heap()->allocate(size); 106 if (mCblkMemory != 0) { 107 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 108 // can't assume mCblk != NULL 109 } else { 110 ALOGE("not enough memory for AudioTrack size=%u", size); 111 client->heap()->dump("AudioTrack"); 112 return; 113 } 114 } else { 115 // this syntax avoids calling the audio_track_cblk_t constructor twice 116 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 117 // assume mCblk != NULL 118 } 119 120 // construct the shared structure in-place. 121 if (mCblk != NULL) { 122 new(mCblk) audio_track_cblk_t(); 123 // clear all buffers 124 mCblk->frameCount_ = frameCount; 125 if (sharedBuffer == 0) { 126 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 127 memset(mBuffer, 0, bufferSize); 128 } else { 129 mBuffer = sharedBuffer->pointer(); 130#if 0 131 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 132#endif 133 } 134 135#ifdef TEE_SINK 136 if (mTeeSinkTrackEnabled) { 137 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 138 if (pipeFormat != Format_Invalid) { 139 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 140 size_t numCounterOffers = 0; 141 const NBAIO_Format offers[1] = {pipeFormat}; 142 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 143 ALOG_ASSERT(index == 0); 144 PipeReader *pipeReader = new PipeReader(*pipe); 145 numCounterOffers = 0; 146 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 147 ALOG_ASSERT(index == 0); 148 mTeeSink = pipe; 149 mTeeSource = pipeReader; 150 } 151 } 152#endif 153 154 } 155} 156 157AudioFlinger::ThreadBase::TrackBase::~TrackBase() 158{ 159#ifdef TEE_SINK 160 dumpTee(-1, mTeeSource, mId); 161#endif 162 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 163 delete mServerProxy; 164 if (mCblk != NULL) { 165 if (mClient == 0) { 166 delete mCblk; 167 } else { 168 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 169 } 170 } 171 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 172 if (mClient != 0) { 173 // Client destructor must run with AudioFlinger mutex locked 174 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 175 // If the client's reference count drops to zero, the associated destructor 176 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 177 // relying on the automatic clear() at end of scope. 178 mClient.clear(); 179 } 180} 181 182// AudioBufferProvider interface 183// getNextBuffer() = 0; 184// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 185void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 186{ 187#ifdef TEE_SINK 188 if (mTeeSink != 0) { 189 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 190 } 191#endif 192 193 ServerProxy::Buffer buf; 194 buf.mFrameCount = buffer->frameCount; 195 buf.mRaw = buffer->raw; 196 buffer->frameCount = 0; 197 buffer->raw = NULL; 198 mServerProxy->releaseBuffer(&buf); 199} 200 201status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 202{ 203 mSyncEvents.add(event); 204 return NO_ERROR; 205} 206 207// ---------------------------------------------------------------------------- 208// Playback 209// ---------------------------------------------------------------------------- 210 211AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 212 : BnAudioTrack(), 213 mTrack(track) 214{ 215} 216 217AudioFlinger::TrackHandle::~TrackHandle() { 218 // just stop the track on deletion, associated resources 219 // will be freed from the main thread once all pending buffers have 220 // been played. Unless it's not in the active track list, in which 221 // case we free everything now... 222 mTrack->destroy(); 223} 224 225sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 226 return mTrack->getCblk(); 227} 228 229status_t AudioFlinger::TrackHandle::start() { 230 return mTrack->start(); 231} 232 233void AudioFlinger::TrackHandle::stop() { 234 mTrack->stop(); 235} 236 237void AudioFlinger::TrackHandle::flush() { 238 mTrack->flush(); 239} 240 241void AudioFlinger::TrackHandle::pause() { 242 mTrack->pause(); 243} 244 245status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 246{ 247 return mTrack->attachAuxEffect(EffectId); 248} 249 250status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 251 sp<IMemory>* buffer) { 252 if (!mTrack->isTimedTrack()) 253 return INVALID_OPERATION; 254 255 PlaybackThread::TimedTrack* tt = 256 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 257 return tt->allocateTimedBuffer(size, buffer); 258} 259 260status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 261 int64_t pts) { 262 if (!mTrack->isTimedTrack()) 263 return INVALID_OPERATION; 264 265 PlaybackThread::TimedTrack* tt = 266 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 267 return tt->queueTimedBuffer(buffer, pts); 268} 269 270status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 271 const LinearTransform& xform, int target) { 272 273 if (!mTrack->isTimedTrack()) 274 return INVALID_OPERATION; 275 276 PlaybackThread::TimedTrack* tt = 277 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 278 return tt->setMediaTimeTransform( 279 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 280} 281 282status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 283 return mTrack->setParameters(keyValuePairs); 284} 285 286status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 287{ 288 return mTrack->getTimestamp(timestamp); 289} 290 291status_t AudioFlinger::TrackHandle::onTransact( 292 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 293{ 294 return BnAudioTrack::onTransact(code, data, reply, flags); 295} 296 297// ---------------------------------------------------------------------------- 298 299// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 300AudioFlinger::PlaybackThread::Track::Track( 301 PlaybackThread *thread, 302 const sp<Client>& client, 303 audio_stream_type_t streamType, 304 uint32_t sampleRate, 305 audio_format_t format, 306 audio_channel_mask_t channelMask, 307 size_t frameCount, 308 const sp<IMemory>& sharedBuffer, 309 int sessionId, 310 IAudioFlinger::track_flags_t flags) 311 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 312 sessionId, true /*isOut*/), 313 mFillingUpStatus(FS_INVALID), 314 // mRetryCount initialized later when needed 315 mSharedBuffer(sharedBuffer), 316 mStreamType(streamType), 317 mName(-1), // see note below 318 mMainBuffer(thread->mixBuffer()), 319 mAuxBuffer(NULL), 320 mAuxEffectId(0), mHasVolumeController(false), 321 mPresentationCompleteFrames(0), 322 mFlags(flags), 323 mFastIndex(-1), 324 mCachedVolume(1.0), 325 mIsInvalid(false), 326 mAudioTrackServerProxy(NULL), 327 mResumeToStopping(false) 328{ 329 if (mCblk != NULL) { 330 if (sharedBuffer == 0) { 331 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 332 mFrameSize); 333 } else { 334 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 335 mFrameSize); 336 } 337 mServerProxy = mAudioTrackServerProxy; 338 // to avoid leaking a track name, do not allocate one unless there is an mCblk 339 mName = thread->getTrackName_l(channelMask, sessionId); 340 if (mName < 0) { 341 ALOGE("no more track names available"); 342 return; 343 } 344 // only allocate a fast track index if we were able to allocate a normal track name 345 if (flags & IAudioFlinger::TRACK_FAST) { 346 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 347 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 348 int i = __builtin_ctz(thread->mFastTrackAvailMask); 349 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 350 // FIXME This is too eager. We allocate a fast track index before the 351 // fast track becomes active. Since fast tracks are a scarce resource, 352 // this means we are potentially denying other more important fast tracks from 353 // being created. It would be better to allocate the index dynamically. 354 mFastIndex = i; 355 // Read the initial underruns because this field is never cleared by the fast mixer 356 mObservedUnderruns = thread->getFastTrackUnderruns(i); 357 thread->mFastTrackAvailMask &= ~(1 << i); 358 } 359 } 360 ALOGV("Track constructor name %d, calling pid %d", mName, 361 IPCThreadState::self()->getCallingPid()); 362} 363 364AudioFlinger::PlaybackThread::Track::~Track() 365{ 366 ALOGV("PlaybackThread::Track destructor"); 367 368 // The destructor would clear mSharedBuffer, 369 // but it will not push the decremented reference count, 370 // leaving the client's IMemory dangling indefinitely. 371 // This prevents that leak. 372 if (mSharedBuffer != 0) { 373 mSharedBuffer.clear(); 374 // flush the binder command buffer 375 IPCThreadState::self()->flushCommands(); 376 } 377} 378 379void AudioFlinger::PlaybackThread::Track::destroy() 380{ 381 // NOTE: destroyTrack_l() can remove a strong reference to this Track 382 // by removing it from mTracks vector, so there is a risk that this Tracks's 383 // destructor is called. As the destructor needs to lock mLock, 384 // we must acquire a strong reference on this Track before locking mLock 385 // here so that the destructor is called only when exiting this function. 386 // On the other hand, as long as Track::destroy() is only called by 387 // TrackHandle destructor, the TrackHandle still holds a strong ref on 388 // this Track with its member mTrack. 389 sp<Track> keep(this); 390 { // scope for mLock 391 sp<ThreadBase> thread = mThread.promote(); 392 if (thread != 0) { 393 Mutex::Autolock _l(thread->mLock); 394 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 395 bool wasActive = playbackThread->destroyTrack_l(this); 396 if (!isOutputTrack() && !wasActive) { 397 AudioSystem::releaseOutput(thread->id()); 398 } 399 } 400 } 401} 402 403/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 404{ 405 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 406 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 407} 408 409void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 410{ 411 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 412 if (isFastTrack()) { 413 sprintf(buffer, " F %2d", mFastIndex); 414 } else { 415 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 416 } 417 track_state state = mState; 418 char stateChar; 419 if (isTerminated()) { 420 stateChar = 'T'; 421 } else { 422 switch (state) { 423 case IDLE: 424 stateChar = 'I'; 425 break; 426 case STOPPING_1: 427 stateChar = 's'; 428 break; 429 case STOPPING_2: 430 stateChar = '5'; 431 break; 432 case STOPPED: 433 stateChar = 'S'; 434 break; 435 case RESUMING: 436 stateChar = 'R'; 437 break; 438 case ACTIVE: 439 stateChar = 'A'; 440 break; 441 case PAUSING: 442 stateChar = 'p'; 443 break; 444 case PAUSED: 445 stateChar = 'P'; 446 break; 447 case FLUSHED: 448 stateChar = 'F'; 449 break; 450 default: 451 stateChar = '?'; 452 break; 453 } 454 } 455 char nowInUnderrun; 456 switch (mObservedUnderruns.mBitFields.mMostRecent) { 457 case UNDERRUN_FULL: 458 nowInUnderrun = ' '; 459 break; 460 case UNDERRUN_PARTIAL: 461 nowInUnderrun = '<'; 462 break; 463 case UNDERRUN_EMPTY: 464 nowInUnderrun = '*'; 465 break; 466 default: 467 nowInUnderrun = '?'; 468 break; 469 } 470 snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 471 "%08X %08X %08X 0x%03X %9u%c\n", 472 (mClient == 0) ? getpid_cached : mClient->pid(), 473 mStreamType, 474 mFormat, 475 mChannelMask, 476 mSessionId, 477 mFrameCount, 478 stateChar, 479 mFillingUpStatus, 480 mAudioTrackServerProxy->getSampleRate(), 481 20.0 * log10((vlr & 0xFFFF) / 4096.0), 482 20.0 * log10((vlr >> 16) / 4096.0), 483 mCblk->mServer, 484 (int)mMainBuffer, 485 (int)mAuxBuffer, 486 mCblk->mFlags, 487 mAudioTrackServerProxy->getUnderrunFrames(), 488 nowInUnderrun); 489} 490 491uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 492 return mAudioTrackServerProxy->getSampleRate(); 493} 494 495// AudioBufferProvider interface 496status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 497 AudioBufferProvider::Buffer* buffer, int64_t pts) 498{ 499 ServerProxy::Buffer buf; 500 size_t desiredFrames = buffer->frameCount; 501 buf.mFrameCount = desiredFrames; 502 status_t status = mServerProxy->obtainBuffer(&buf); 503 buffer->frameCount = buf.mFrameCount; 504 buffer->raw = buf.mRaw; 505 if (buf.mFrameCount == 0) { 506 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 507 } 508 return status; 509} 510 511// releaseBuffer() is not overridden 512 513// ExtendedAudioBufferProvider interface 514 515// Note that framesReady() takes a mutex on the control block using tryLock(). 516// This could result in priority inversion if framesReady() is called by the normal mixer, 517// as the normal mixer thread runs at lower 518// priority than the client's callback thread: there is a short window within framesReady() 519// during which the normal mixer could be preempted, and the client callback would block. 520// Another problem can occur if framesReady() is called by the fast mixer: 521// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 522// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 523size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 524 return mAudioTrackServerProxy->framesReady(); 525} 526 527size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 528{ 529 return mAudioTrackServerProxy->framesReleased(); 530} 531 532// Don't call for fast tracks; the framesReady() could result in priority inversion 533bool AudioFlinger::PlaybackThread::Track::isReady() const { 534 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 535 return true; 536 } 537 538 if (framesReady() >= mFrameCount || 539 (mCblk->mFlags & CBLK_FORCEREADY)) { 540 mFillingUpStatus = FS_FILLED; 541 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 542 return true; 543 } 544 return false; 545} 546 547status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 548 int triggerSession) 549{ 550 status_t status = NO_ERROR; 551 ALOGV("start(%d), calling pid %d session %d", 552 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 553 554 sp<ThreadBase> thread = mThread.promote(); 555 if (thread != 0) { 556 if (isOffloaded()) { 557 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 558 Mutex::Autolock _lth(thread->mLock); 559 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 560 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 561 (ec != 0 && ec->isNonOffloadableEnabled())) { 562 invalidate(); 563 return PERMISSION_DENIED; 564 } 565 } 566 Mutex::Autolock _lth(thread->mLock); 567 track_state state = mState; 568 // here the track could be either new, or restarted 569 // in both cases "unstop" the track 570 571 if (state == PAUSED) { 572 if (mResumeToStopping) { 573 // happened we need to resume to STOPPING_1 574 mState = TrackBase::STOPPING_1; 575 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 576 } else { 577 mState = TrackBase::RESUMING; 578 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 579 } 580 } else { 581 mState = TrackBase::ACTIVE; 582 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 583 } 584 585 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 586 status = playbackThread->addTrack_l(this); 587 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 588 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 589 // restore previous state if start was rejected by policy manager 590 if (status == PERMISSION_DENIED) { 591 mState = state; 592 } 593 } 594 // track was already in the active list, not a problem 595 if (status == ALREADY_EXISTS) { 596 status = NO_ERROR; 597 } 598 } else { 599 status = BAD_VALUE; 600 } 601 return status; 602} 603 604void AudioFlinger::PlaybackThread::Track::stop() 605{ 606 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 607 sp<ThreadBase> thread = mThread.promote(); 608 if (thread != 0) { 609 Mutex::Autolock _l(thread->mLock); 610 track_state state = mState; 611 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 612 // If the track is not active (PAUSED and buffers full), flush buffers 613 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 614 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 615 reset(); 616 mState = STOPPED; 617 } else if (!isFastTrack() && !isOffloaded()) { 618 mState = STOPPED; 619 } else { 620 // For fast tracks prepareTracks_l() will set state to STOPPING_2 621 // presentation is complete 622 // For an offloaded track this starts a drain and state will 623 // move to STOPPING_2 when drain completes and then STOPPED 624 mState = STOPPING_1; 625 } 626 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 627 playbackThread); 628 } 629 } 630} 631 632void AudioFlinger::PlaybackThread::Track::pause() 633{ 634 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 635 sp<ThreadBase> thread = mThread.promote(); 636 if (thread != 0) { 637 Mutex::Autolock _l(thread->mLock); 638 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 639 switch (mState) { 640 case STOPPING_1: 641 case STOPPING_2: 642 if (!isOffloaded()) { 643 /* nothing to do if track is not offloaded */ 644 break; 645 } 646 647 // Offloaded track was draining, we need to carry on draining when resumed 648 mResumeToStopping = true; 649 // fall through... 650 case ACTIVE: 651 case RESUMING: 652 mState = PAUSING; 653 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 654 playbackThread->signal_l(); 655 break; 656 657 default: 658 break; 659 } 660 } 661} 662 663void AudioFlinger::PlaybackThread::Track::flush() 664{ 665 ALOGV("flush(%d)", mName); 666 sp<ThreadBase> thread = mThread.promote(); 667 if (thread != 0) { 668 Mutex::Autolock _l(thread->mLock); 669 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 670 671 if (isOffloaded()) { 672 // If offloaded we allow flush during any state except terminated 673 // and keep the track active to avoid problems if user is seeking 674 // rapidly and underlying hardware has a significant delay handling 675 // a pause 676 if (isTerminated()) { 677 return; 678 } 679 680 ALOGV("flush: offload flush"); 681 reset(); 682 683 if (mState == STOPPING_1 || mState == STOPPING_2) { 684 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 685 mState = ACTIVE; 686 } 687 688 if (mState == ACTIVE) { 689 ALOGV("flush called in active state, resetting buffer time out retry count"); 690 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 691 } 692 693 mResumeToStopping = false; 694 } else { 695 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 696 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 697 return; 698 } 699 // No point remaining in PAUSED state after a flush => go to 700 // FLUSHED state 701 mState = FLUSHED; 702 // do not reset the track if it is still in the process of being stopped or paused. 703 // this will be done by prepareTracks_l() when the track is stopped. 704 // prepareTracks_l() will see mState == FLUSHED, then 705 // remove from active track list, reset(), and trigger presentation complete 706 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 707 reset(); 708 } 709 } 710 // Prevent flush being lost if the track is flushed and then resumed 711 // before mixer thread can run. This is important when offloading 712 // because the hardware buffer could hold a large amount of audio 713 playbackThread->flushOutput_l(); 714 playbackThread->signal_l(); 715 } 716} 717 718void AudioFlinger::PlaybackThread::Track::reset() 719{ 720 // Do not reset twice to avoid discarding data written just after a flush and before 721 // the audioflinger thread detects the track is stopped. 722 if (!mResetDone) { 723 // Force underrun condition to avoid false underrun callback until first data is 724 // written to buffer 725 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 726 mFillingUpStatus = FS_FILLING; 727 mResetDone = true; 728 if (mState == FLUSHED) { 729 mState = IDLE; 730 } 731 } 732} 733 734status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 735{ 736 sp<ThreadBase> thread = mThread.promote(); 737 if (thread == 0) { 738 ALOGE("thread is dead"); 739 return FAILED_TRANSACTION; 740 } else if ((thread->type() == ThreadBase::DIRECT) || 741 (thread->type() == ThreadBase::OFFLOAD)) { 742 return thread->setParameters(keyValuePairs); 743 } else { 744 return PERMISSION_DENIED; 745 } 746} 747 748status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 749{ 750 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 751 if (isFastTrack()) { 752 return INVALID_OPERATION; 753 } 754 sp<ThreadBase> thread = mThread.promote(); 755 if (thread == 0) { 756 return INVALID_OPERATION; 757 } 758 Mutex::Autolock _l(thread->mLock); 759 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 760 if (!playbackThread->mLatchQValid) { 761 return INVALID_OPERATION; 762 } 763 uint32_t unpresentedFrames = 764 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 765 playbackThread->mSampleRate; 766 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 767 if (framesWritten < unpresentedFrames) { 768 return INVALID_OPERATION; 769 } 770 timestamp.mPosition = framesWritten - unpresentedFrames; 771 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 772 return NO_ERROR; 773} 774 775status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 776{ 777 status_t status = DEAD_OBJECT; 778 sp<ThreadBase> thread = mThread.promote(); 779 if (thread != 0) { 780 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 781 sp<AudioFlinger> af = mClient->audioFlinger(); 782 783 Mutex::Autolock _l(af->mLock); 784 785 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 786 787 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 788 Mutex::Autolock _dl(playbackThread->mLock); 789 Mutex::Autolock _sl(srcThread->mLock); 790 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 791 if (chain == 0) { 792 return INVALID_OPERATION; 793 } 794 795 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 796 if (effect == 0) { 797 return INVALID_OPERATION; 798 } 799 srcThread->removeEffect_l(effect); 800 status = playbackThread->addEffect_l(effect); 801 if (status != NO_ERROR) { 802 srcThread->addEffect_l(effect); 803 return INVALID_OPERATION; 804 } 805 // removeEffect_l() has stopped the effect if it was active so it must be restarted 806 if (effect->state() == EffectModule::ACTIVE || 807 effect->state() == EffectModule::STOPPING) { 808 effect->start(); 809 } 810 811 sp<EffectChain> dstChain = effect->chain().promote(); 812 if (dstChain == 0) { 813 srcThread->addEffect_l(effect); 814 return INVALID_OPERATION; 815 } 816 AudioSystem::unregisterEffect(effect->id()); 817 AudioSystem::registerEffect(&effect->desc(), 818 srcThread->id(), 819 dstChain->strategy(), 820 AUDIO_SESSION_OUTPUT_MIX, 821 effect->id()); 822 } 823 status = playbackThread->attachAuxEffect(this, EffectId); 824 } 825 return status; 826} 827 828void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 829{ 830 mAuxEffectId = EffectId; 831 mAuxBuffer = buffer; 832} 833 834bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 835 size_t audioHalFrames) 836{ 837 // a track is considered presented when the total number of frames written to audio HAL 838 // corresponds to the number of frames written when presentationComplete() is called for the 839 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 840 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 841 // to detect when all frames have been played. In this case framesWritten isn't 842 // useful because it doesn't always reflect whether there is data in the h/w 843 // buffers, particularly if a track has been paused and resumed during draining 844 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 845 mPresentationCompleteFrames, framesWritten); 846 if (mPresentationCompleteFrames == 0) { 847 mPresentationCompleteFrames = framesWritten + audioHalFrames; 848 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 849 mPresentationCompleteFrames, audioHalFrames); 850 } 851 852 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 853 ALOGV("presentationComplete() session %d complete: framesWritten %d", 854 mSessionId, framesWritten); 855 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 856 mAudioTrackServerProxy->setStreamEndDone(); 857 return true; 858 } 859 return false; 860} 861 862void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 863{ 864 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 865 if (mSyncEvents[i]->type() == type) { 866 mSyncEvents[i]->trigger(); 867 mSyncEvents.removeAt(i); 868 i--; 869 } 870 } 871} 872 873// implement VolumeBufferProvider interface 874 875uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 876{ 877 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 878 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 879 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 880 uint32_t vl = vlr & 0xFFFF; 881 uint32_t vr = vlr >> 16; 882 // track volumes come from shared memory, so can't be trusted and must be clamped 883 if (vl > MAX_GAIN_INT) { 884 vl = MAX_GAIN_INT; 885 } 886 if (vr > MAX_GAIN_INT) { 887 vr = MAX_GAIN_INT; 888 } 889 // now apply the cached master volume and stream type volume; 890 // this is trusted but lacks any synchronization or barrier so may be stale 891 float v = mCachedVolume; 892 vl *= v; 893 vr *= v; 894 // re-combine into U4.16 895 vlr = (vr << 16) | (vl & 0xFFFF); 896 // FIXME look at mute, pause, and stop flags 897 return vlr; 898} 899 900status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 901{ 902 if (isTerminated() || mState == PAUSED || 903 ((framesReady() == 0) && ((mSharedBuffer != 0) || 904 (mState == STOPPED)))) { 905 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 906 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 907 event->cancel(); 908 return INVALID_OPERATION; 909 } 910 (void) TrackBase::setSyncEvent(event); 911 return NO_ERROR; 912} 913 914void AudioFlinger::PlaybackThread::Track::invalidate() 915{ 916 // FIXME should use proxy, and needs work 917 audio_track_cblk_t* cblk = mCblk; 918 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 919 android_atomic_release_store(0x40000000, &cblk->mFutex); 920 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 921 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 922 mIsInvalid = true; 923} 924 925// ---------------------------------------------------------------------------- 926 927sp<AudioFlinger::PlaybackThread::TimedTrack> 928AudioFlinger::PlaybackThread::TimedTrack::create( 929 PlaybackThread *thread, 930 const sp<Client>& client, 931 audio_stream_type_t streamType, 932 uint32_t sampleRate, 933 audio_format_t format, 934 audio_channel_mask_t channelMask, 935 size_t frameCount, 936 const sp<IMemory>& sharedBuffer, 937 int sessionId) { 938 if (!client->reserveTimedTrack()) 939 return 0; 940 941 return new TimedTrack( 942 thread, client, streamType, sampleRate, format, channelMask, frameCount, 943 sharedBuffer, sessionId); 944} 945 946AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 947 PlaybackThread *thread, 948 const sp<Client>& client, 949 audio_stream_type_t streamType, 950 uint32_t sampleRate, 951 audio_format_t format, 952 audio_channel_mask_t channelMask, 953 size_t frameCount, 954 const sp<IMemory>& sharedBuffer, 955 int sessionId) 956 : Track(thread, client, streamType, sampleRate, format, channelMask, 957 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 958 mQueueHeadInFlight(false), 959 mTrimQueueHeadOnRelease(false), 960 mFramesPendingInQueue(0), 961 mTimedSilenceBuffer(NULL), 962 mTimedSilenceBufferSize(0), 963 mTimedAudioOutputOnTime(false), 964 mMediaTimeTransformValid(false) 965{ 966 LocalClock lc; 967 mLocalTimeFreq = lc.getLocalFreq(); 968 969 mLocalTimeToSampleTransform.a_zero = 0; 970 mLocalTimeToSampleTransform.b_zero = 0; 971 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 972 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 973 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 974 &mLocalTimeToSampleTransform.a_to_b_denom); 975 976 mMediaTimeToSampleTransform.a_zero = 0; 977 mMediaTimeToSampleTransform.b_zero = 0; 978 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 979 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 980 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 981 &mMediaTimeToSampleTransform.a_to_b_denom); 982} 983 984AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 985 mClient->releaseTimedTrack(); 986 delete [] mTimedSilenceBuffer; 987} 988 989status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 990 size_t size, sp<IMemory>* buffer) { 991 992 Mutex::Autolock _l(mTimedBufferQueueLock); 993 994 trimTimedBufferQueue_l(); 995 996 // lazily initialize the shared memory heap for timed buffers 997 if (mTimedMemoryDealer == NULL) { 998 const int kTimedBufferHeapSize = 512 << 10; 999 1000 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1001 "AudioFlingerTimed"); 1002 if (mTimedMemoryDealer == NULL) 1003 return NO_MEMORY; 1004 } 1005 1006 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1007 if (newBuffer == NULL) { 1008 newBuffer = mTimedMemoryDealer->allocate(size); 1009 if (newBuffer == NULL) 1010 return NO_MEMORY; 1011 } 1012 1013 *buffer = newBuffer; 1014 return NO_ERROR; 1015} 1016 1017// caller must hold mTimedBufferQueueLock 1018void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1019 int64_t mediaTimeNow; 1020 { 1021 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1022 if (!mMediaTimeTransformValid) 1023 return; 1024 1025 int64_t targetTimeNow; 1026 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1027 ? mCCHelper.getCommonTime(&targetTimeNow) 1028 : mCCHelper.getLocalTime(&targetTimeNow); 1029 1030 if (OK != res) 1031 return; 1032 1033 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1034 &mediaTimeNow)) { 1035 return; 1036 } 1037 } 1038 1039 size_t trimEnd; 1040 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1041 int64_t bufEnd; 1042 1043 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1044 // We have a next buffer. Just use its PTS as the PTS of the frame 1045 // following the last frame in this buffer. If the stream is sparse 1046 // (ie, there are deliberate gaps left in the stream which should be 1047 // filled with silence by the TimedAudioTrack), then this can result 1048 // in one extra buffer being left un-trimmed when it could have 1049 // been. In general, this is not typical, and we would rather 1050 // optimized away the TS calculation below for the more common case 1051 // where PTSes are contiguous. 1052 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1053 } else { 1054 // We have no next buffer. Compute the PTS of the frame following 1055 // the last frame in this buffer by computing the duration of of 1056 // this frame in media time units and adding it to the PTS of the 1057 // buffer. 1058 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1059 / mFrameSize; 1060 1061 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1062 &bufEnd)) { 1063 ALOGE("Failed to convert frame count of %lld to media time" 1064 " duration" " (scale factor %d/%u) in %s", 1065 frameCount, 1066 mMediaTimeToSampleTransform.a_to_b_numer, 1067 mMediaTimeToSampleTransform.a_to_b_denom, 1068 __PRETTY_FUNCTION__); 1069 break; 1070 } 1071 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1072 } 1073 1074 if (bufEnd > mediaTimeNow) 1075 break; 1076 1077 // Is the buffer we want to use in the middle of a mix operation right 1078 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1079 // from the mixer which should be coming back shortly. 1080 if (!trimEnd && mQueueHeadInFlight) { 1081 mTrimQueueHeadOnRelease = true; 1082 } 1083 } 1084 1085 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1086 if (trimStart < trimEnd) { 1087 // Update the bookkeeping for framesReady() 1088 for (size_t i = trimStart; i < trimEnd; ++i) { 1089 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1090 } 1091 1092 // Now actually remove the buffers from the queue. 1093 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1094 } 1095} 1096 1097void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1098 const char* logTag) { 1099 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1100 "%s called (reason \"%s\"), but timed buffer queue has no" 1101 " elements to trim.", __FUNCTION__, logTag); 1102 1103 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1104 mTimedBufferQueue.removeAt(0); 1105} 1106 1107void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1108 const TimedBuffer& buf, 1109 const char* logTag) { 1110 uint32_t bufBytes = buf.buffer()->size(); 1111 uint32_t consumedAlready = buf.position(); 1112 1113 ALOG_ASSERT(consumedAlready <= bufBytes, 1114 "Bad bookkeeping while updating frames pending. Timed buffer is" 1115 " only %u bytes long, but claims to have consumed %u" 1116 " bytes. (update reason: \"%s\")", 1117 bufBytes, consumedAlready, logTag); 1118 1119 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1120 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1121 "Bad bookkeeping while updating frames pending. Should have at" 1122 " least %u queued frames, but we think we have only %u. (update" 1123 " reason: \"%s\")", 1124 bufFrames, mFramesPendingInQueue, logTag); 1125 1126 mFramesPendingInQueue -= bufFrames; 1127} 1128 1129status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1130 const sp<IMemory>& buffer, int64_t pts) { 1131 1132 { 1133 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1134 if (!mMediaTimeTransformValid) 1135 return INVALID_OPERATION; 1136 } 1137 1138 Mutex::Autolock _l(mTimedBufferQueueLock); 1139 1140 uint32_t bufFrames = buffer->size() / mFrameSize; 1141 mFramesPendingInQueue += bufFrames; 1142 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1143 1144 return NO_ERROR; 1145} 1146 1147status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1148 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1149 1150 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1151 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1152 target); 1153 1154 if (!(target == TimedAudioTrack::LOCAL_TIME || 1155 target == TimedAudioTrack::COMMON_TIME)) { 1156 return BAD_VALUE; 1157 } 1158 1159 Mutex::Autolock lock(mMediaTimeTransformLock); 1160 mMediaTimeTransform = xform; 1161 mMediaTimeTransformTarget = target; 1162 mMediaTimeTransformValid = true; 1163 1164 return NO_ERROR; 1165} 1166 1167#define min(a, b) ((a) < (b) ? (a) : (b)) 1168 1169// implementation of getNextBuffer for tracks whose buffers have timestamps 1170status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1171 AudioBufferProvider::Buffer* buffer, int64_t pts) 1172{ 1173 if (pts == AudioBufferProvider::kInvalidPTS) { 1174 buffer->raw = NULL; 1175 buffer->frameCount = 0; 1176 mTimedAudioOutputOnTime = false; 1177 return INVALID_OPERATION; 1178 } 1179 1180 Mutex::Autolock _l(mTimedBufferQueueLock); 1181 1182 ALOG_ASSERT(!mQueueHeadInFlight, 1183 "getNextBuffer called without releaseBuffer!"); 1184 1185 while (true) { 1186 1187 // if we have no timed buffers, then fail 1188 if (mTimedBufferQueue.isEmpty()) { 1189 buffer->raw = NULL; 1190 buffer->frameCount = 0; 1191 return NOT_ENOUGH_DATA; 1192 } 1193 1194 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1195 1196 // calculate the PTS of the head of the timed buffer queue expressed in 1197 // local time 1198 int64_t headLocalPTS; 1199 { 1200 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1201 1202 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1203 1204 if (mMediaTimeTransform.a_to_b_denom == 0) { 1205 // the transform represents a pause, so yield silence 1206 timedYieldSilence_l(buffer->frameCount, buffer); 1207 return NO_ERROR; 1208 } 1209 1210 int64_t transformedPTS; 1211 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1212 &transformedPTS)) { 1213 // the transform failed. this shouldn't happen, but if it does 1214 // then just drop this buffer 1215 ALOGW("timedGetNextBuffer transform failed"); 1216 buffer->raw = NULL; 1217 buffer->frameCount = 0; 1218 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1219 return NO_ERROR; 1220 } 1221 1222 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1223 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1224 &headLocalPTS)) { 1225 buffer->raw = NULL; 1226 buffer->frameCount = 0; 1227 return INVALID_OPERATION; 1228 } 1229 } else { 1230 headLocalPTS = transformedPTS; 1231 } 1232 } 1233 1234 uint32_t sr = sampleRate(); 1235 1236 // adjust the head buffer's PTS to reflect the portion of the head buffer 1237 // that has already been consumed 1238 int64_t effectivePTS = headLocalPTS + 1239 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1240 1241 // Calculate the delta in samples between the head of the input buffer 1242 // queue and the start of the next output buffer that will be written. 1243 // If the transformation fails because of over or underflow, it means 1244 // that the sample's position in the output stream is so far out of 1245 // whack that it should just be dropped. 1246 int64_t sampleDelta; 1247 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1248 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1249 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1250 " mix"); 1251 continue; 1252 } 1253 if (!mLocalTimeToSampleTransform.doForwardTransform( 1254 (effectivePTS - pts) << 32, &sampleDelta)) { 1255 ALOGV("*** too late during sample rate transform: dropped buffer"); 1256 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1257 continue; 1258 } 1259 1260 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1261 " sampleDelta=[%d.%08x]", 1262 head.pts(), head.position(), pts, 1263 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1264 + (sampleDelta >> 32)), 1265 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1266 1267 // if the delta between the ideal placement for the next input sample and 1268 // the current output position is within this threshold, then we will 1269 // concatenate the next input samples to the previous output 1270 const int64_t kSampleContinuityThreshold = 1271 (static_cast<int64_t>(sr) << 32) / 250; 1272 1273 // if this is the first buffer of audio that we're emitting from this track 1274 // then it should be almost exactly on time. 1275 const int64_t kSampleStartupThreshold = 1LL << 32; 1276 1277 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1278 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1279 // the next input is close enough to being on time, so concatenate it 1280 // with the last output 1281 timedYieldSamples_l(buffer); 1282 1283 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1284 head.position(), buffer->frameCount); 1285 return NO_ERROR; 1286 } 1287 1288 // Looks like our output is not on time. Reset our on timed status. 1289 // Next time we mix samples from our input queue, then should be within 1290 // the StartupThreshold. 1291 mTimedAudioOutputOnTime = false; 1292 if (sampleDelta > 0) { 1293 // the gap between the current output position and the proper start of 1294 // the next input sample is too big, so fill it with silence 1295 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1296 1297 timedYieldSilence_l(framesUntilNextInput, buffer); 1298 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1299 return NO_ERROR; 1300 } else { 1301 // the next input sample is late 1302 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1303 size_t onTimeSamplePosition = 1304 head.position() + lateFrames * mFrameSize; 1305 1306 if (onTimeSamplePosition > head.buffer()->size()) { 1307 // all the remaining samples in the head are too late, so 1308 // drop it and move on 1309 ALOGV("*** too late: dropped buffer"); 1310 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1311 continue; 1312 } else { 1313 // skip over the late samples 1314 head.setPosition(onTimeSamplePosition); 1315 1316 // yield the available samples 1317 timedYieldSamples_l(buffer); 1318 1319 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1320 return NO_ERROR; 1321 } 1322 } 1323 } 1324} 1325 1326// Yield samples from the timed buffer queue head up to the given output 1327// buffer's capacity. 1328// 1329// Caller must hold mTimedBufferQueueLock 1330void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1331 AudioBufferProvider::Buffer* buffer) { 1332 1333 const TimedBuffer& head = mTimedBufferQueue[0]; 1334 1335 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1336 head.position()); 1337 1338 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1339 mFrameSize); 1340 size_t framesRequested = buffer->frameCount; 1341 buffer->frameCount = min(framesLeftInHead, framesRequested); 1342 1343 mQueueHeadInFlight = true; 1344 mTimedAudioOutputOnTime = true; 1345} 1346 1347// Yield samples of silence up to the given output buffer's capacity 1348// 1349// Caller must hold mTimedBufferQueueLock 1350void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1351 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1352 1353 // lazily allocate a buffer filled with silence 1354 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1355 delete [] mTimedSilenceBuffer; 1356 mTimedSilenceBufferSize = numFrames * mFrameSize; 1357 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1358 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1359 } 1360 1361 buffer->raw = mTimedSilenceBuffer; 1362 size_t framesRequested = buffer->frameCount; 1363 buffer->frameCount = min(numFrames, framesRequested); 1364 1365 mTimedAudioOutputOnTime = false; 1366} 1367 1368// AudioBufferProvider interface 1369void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1370 AudioBufferProvider::Buffer* buffer) { 1371 1372 Mutex::Autolock _l(mTimedBufferQueueLock); 1373 1374 // If the buffer which was just released is part of the buffer at the head 1375 // of the queue, be sure to update the amt of the buffer which has been 1376 // consumed. If the buffer being returned is not part of the head of the 1377 // queue, its either because the buffer is part of the silence buffer, or 1378 // because the head of the timed queue was trimmed after the mixer called 1379 // getNextBuffer but before the mixer called releaseBuffer. 1380 if (buffer->raw == mTimedSilenceBuffer) { 1381 ALOG_ASSERT(!mQueueHeadInFlight, 1382 "Queue head in flight during release of silence buffer!"); 1383 goto done; 1384 } 1385 1386 ALOG_ASSERT(mQueueHeadInFlight, 1387 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1388 " head in flight."); 1389 1390 if (mTimedBufferQueue.size()) { 1391 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1392 1393 void* start = head.buffer()->pointer(); 1394 void* end = reinterpret_cast<void*>( 1395 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1396 + head.buffer()->size()); 1397 1398 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1399 "released buffer not within the head of the timed buffer" 1400 " queue; qHead = [%p, %p], released buffer = %p", 1401 start, end, buffer->raw); 1402 1403 head.setPosition(head.position() + 1404 (buffer->frameCount * mFrameSize)); 1405 mQueueHeadInFlight = false; 1406 1407 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1408 "Bad bookkeeping during releaseBuffer! Should have at" 1409 " least %u queued frames, but we think we have only %u", 1410 buffer->frameCount, mFramesPendingInQueue); 1411 1412 mFramesPendingInQueue -= buffer->frameCount; 1413 1414 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1415 || mTrimQueueHeadOnRelease) { 1416 trimTimedBufferQueueHead_l("releaseBuffer"); 1417 mTrimQueueHeadOnRelease = false; 1418 } 1419 } else { 1420 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1421 " buffers in the timed buffer queue"); 1422 } 1423 1424done: 1425 buffer->raw = 0; 1426 buffer->frameCount = 0; 1427} 1428 1429size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1430 Mutex::Autolock _l(mTimedBufferQueueLock); 1431 return mFramesPendingInQueue; 1432} 1433 1434AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1435 : mPTS(0), mPosition(0) {} 1436 1437AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1438 const sp<IMemory>& buffer, int64_t pts) 1439 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1440 1441 1442// ---------------------------------------------------------------------------- 1443 1444AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1445 PlaybackThread *playbackThread, 1446 DuplicatingThread *sourceThread, 1447 uint32_t sampleRate, 1448 audio_format_t format, 1449 audio_channel_mask_t channelMask, 1450 size_t frameCount) 1451 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1452 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1453 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1454{ 1455 1456 if (mCblk != NULL) { 1457 mOutBuffer.frameCount = 0; 1458 playbackThread->mTracks.add(this); 1459 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1460 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1461 mCblk, mBuffer, 1462 mCblk->frameCount_, mChannelMask); 1463 // since client and server are in the same process, 1464 // the buffer has the same virtual address on both sides 1465 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1466 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1467 mClientProxy->setSendLevel(0.0); 1468 mClientProxy->setSampleRate(sampleRate); 1469 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1470 true /*clientInServer*/); 1471 } else { 1472 ALOGW("Error creating output track on thread %p", playbackThread); 1473 } 1474} 1475 1476AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1477{ 1478 clearBufferQueue(); 1479 delete mClientProxy; 1480 // superclass destructor will now delete the server proxy and shared memory both refer to 1481} 1482 1483status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1484 int triggerSession) 1485{ 1486 status_t status = Track::start(event, triggerSession); 1487 if (status != NO_ERROR) { 1488 return status; 1489 } 1490 1491 mActive = true; 1492 mRetryCount = 127; 1493 return status; 1494} 1495 1496void AudioFlinger::PlaybackThread::OutputTrack::stop() 1497{ 1498 Track::stop(); 1499 clearBufferQueue(); 1500 mOutBuffer.frameCount = 0; 1501 mActive = false; 1502} 1503 1504bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1505{ 1506 Buffer *pInBuffer; 1507 Buffer inBuffer; 1508 uint32_t channelCount = mChannelCount; 1509 bool outputBufferFull = false; 1510 inBuffer.frameCount = frames; 1511 inBuffer.i16 = data; 1512 1513 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1514 1515 if (!mActive && frames != 0) { 1516 start(); 1517 sp<ThreadBase> thread = mThread.promote(); 1518 if (thread != 0) { 1519 MixerThread *mixerThread = (MixerThread *)thread.get(); 1520 if (mFrameCount > frames) { 1521 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1522 uint32_t startFrames = (mFrameCount - frames); 1523 pInBuffer = new Buffer; 1524 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1525 pInBuffer->frameCount = startFrames; 1526 pInBuffer->i16 = pInBuffer->mBuffer; 1527 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1528 mBufferQueue.add(pInBuffer); 1529 } else { 1530 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1531 } 1532 } 1533 } 1534 } 1535 1536 while (waitTimeLeftMs) { 1537 // First write pending buffers, then new data 1538 if (mBufferQueue.size()) { 1539 pInBuffer = mBufferQueue.itemAt(0); 1540 } else { 1541 pInBuffer = &inBuffer; 1542 } 1543 1544 if (pInBuffer->frameCount == 0) { 1545 break; 1546 } 1547 1548 if (mOutBuffer.frameCount == 0) { 1549 mOutBuffer.frameCount = pInBuffer->frameCount; 1550 nsecs_t startTime = systemTime(); 1551 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1552 if (status != NO_ERROR) { 1553 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1554 mThread.unsafe_get(), status); 1555 outputBufferFull = true; 1556 break; 1557 } 1558 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1559 if (waitTimeLeftMs >= waitTimeMs) { 1560 waitTimeLeftMs -= waitTimeMs; 1561 } else { 1562 waitTimeLeftMs = 0; 1563 } 1564 } 1565 1566 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1567 pInBuffer->frameCount; 1568 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1569 Proxy::Buffer buf; 1570 buf.mFrameCount = outFrames; 1571 buf.mRaw = NULL; 1572 mClientProxy->releaseBuffer(&buf); 1573 pInBuffer->frameCount -= outFrames; 1574 pInBuffer->i16 += outFrames * channelCount; 1575 mOutBuffer.frameCount -= outFrames; 1576 mOutBuffer.i16 += outFrames * channelCount; 1577 1578 if (pInBuffer->frameCount == 0) { 1579 if (mBufferQueue.size()) { 1580 mBufferQueue.removeAt(0); 1581 delete [] pInBuffer->mBuffer; 1582 delete pInBuffer; 1583 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1584 mThread.unsafe_get(), mBufferQueue.size()); 1585 } else { 1586 break; 1587 } 1588 } 1589 } 1590 1591 // If we could not write all frames, allocate a buffer and queue it for next time. 1592 if (inBuffer.frameCount) { 1593 sp<ThreadBase> thread = mThread.promote(); 1594 if (thread != 0 && !thread->standby()) { 1595 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1596 pInBuffer = new Buffer; 1597 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1598 pInBuffer->frameCount = inBuffer.frameCount; 1599 pInBuffer->i16 = pInBuffer->mBuffer; 1600 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1601 sizeof(int16_t)); 1602 mBufferQueue.add(pInBuffer); 1603 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1604 mThread.unsafe_get(), mBufferQueue.size()); 1605 } else { 1606 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1607 mThread.unsafe_get(), this); 1608 } 1609 } 1610 } 1611 1612 // Calling write() with a 0 length buffer, means that no more data will be written: 1613 // If no more buffers are pending, fill output track buffer to make sure it is started 1614 // by output mixer. 1615 if (frames == 0 && mBufferQueue.size() == 0) { 1616 // FIXME borken, replace by getting framesReady() from proxy 1617 size_t user = 0; // was mCblk->user 1618 if (user < mFrameCount) { 1619 frames = mFrameCount - user; 1620 pInBuffer = new Buffer; 1621 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1622 pInBuffer->frameCount = frames; 1623 pInBuffer->i16 = pInBuffer->mBuffer; 1624 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1625 mBufferQueue.add(pInBuffer); 1626 } else if (mActive) { 1627 stop(); 1628 } 1629 } 1630 1631 return outputBufferFull; 1632} 1633 1634status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1635 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1636{ 1637 ClientProxy::Buffer buf; 1638 buf.mFrameCount = buffer->frameCount; 1639 struct timespec timeout; 1640 timeout.tv_sec = waitTimeMs / 1000; 1641 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1642 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1643 buffer->frameCount = buf.mFrameCount; 1644 buffer->raw = buf.mRaw; 1645 return status; 1646} 1647 1648void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1649{ 1650 size_t size = mBufferQueue.size(); 1651 1652 for (size_t i = 0; i < size; i++) { 1653 Buffer *pBuffer = mBufferQueue.itemAt(i); 1654 delete [] pBuffer->mBuffer; 1655 delete pBuffer; 1656 } 1657 mBufferQueue.clear(); 1658} 1659 1660 1661// ---------------------------------------------------------------------------- 1662// Record 1663// ---------------------------------------------------------------------------- 1664 1665AudioFlinger::RecordHandle::RecordHandle( 1666 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1667 : BnAudioRecord(), 1668 mRecordTrack(recordTrack) 1669{ 1670} 1671 1672AudioFlinger::RecordHandle::~RecordHandle() { 1673 stop_nonvirtual(); 1674 mRecordTrack->destroy(); 1675} 1676 1677sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1678 return mRecordTrack->getCblk(); 1679} 1680 1681status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1682 int triggerSession) { 1683 ALOGV("RecordHandle::start()"); 1684 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1685} 1686 1687void AudioFlinger::RecordHandle::stop() { 1688 stop_nonvirtual(); 1689} 1690 1691void AudioFlinger::RecordHandle::stop_nonvirtual() { 1692 ALOGV("RecordHandle::stop()"); 1693 mRecordTrack->stop(); 1694} 1695 1696status_t AudioFlinger::RecordHandle::onTransact( 1697 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1698{ 1699 return BnAudioRecord::onTransact(code, data, reply, flags); 1700} 1701 1702// ---------------------------------------------------------------------------- 1703 1704// RecordTrack constructor must be called with AudioFlinger::mLock held 1705AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1706 RecordThread *thread, 1707 const sp<Client>& client, 1708 uint32_t sampleRate, 1709 audio_format_t format, 1710 audio_channel_mask_t channelMask, 1711 size_t frameCount, 1712 int sessionId) 1713 : TrackBase(thread, client, sampleRate, format, 1714 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1715 mOverflow(false) 1716{ 1717 ALOGV("RecordTrack constructor"); 1718 if (mCblk != NULL) { 1719 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1720 mFrameSize); 1721 mServerProxy = mAudioRecordServerProxy; 1722 } 1723} 1724 1725AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1726{ 1727 ALOGV("%s", __func__); 1728} 1729 1730// AudioBufferProvider interface 1731status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1732 int64_t pts) 1733{ 1734 ServerProxy::Buffer buf; 1735 buf.mFrameCount = buffer->frameCount; 1736 status_t status = mServerProxy->obtainBuffer(&buf); 1737 buffer->frameCount = buf.mFrameCount; 1738 buffer->raw = buf.mRaw; 1739 if (buf.mFrameCount == 0) { 1740 // FIXME also wake futex so that overrun is noticed more quickly 1741 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1742 } 1743 return status; 1744} 1745 1746status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1747 int triggerSession) 1748{ 1749 sp<ThreadBase> thread = mThread.promote(); 1750 if (thread != 0) { 1751 RecordThread *recordThread = (RecordThread *)thread.get(); 1752 return recordThread->start(this, event, triggerSession); 1753 } else { 1754 return BAD_VALUE; 1755 } 1756} 1757 1758void AudioFlinger::RecordThread::RecordTrack::stop() 1759{ 1760 sp<ThreadBase> thread = mThread.promote(); 1761 if (thread != 0) { 1762 RecordThread *recordThread = (RecordThread *)thread.get(); 1763 if (recordThread->stop(this)) { 1764 AudioSystem::stopInput(recordThread->id()); 1765 } 1766 } 1767} 1768 1769void AudioFlinger::RecordThread::RecordTrack::destroy() 1770{ 1771 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1772 sp<RecordTrack> keep(this); 1773 { 1774 sp<ThreadBase> thread = mThread.promote(); 1775 if (thread != 0) { 1776 if (mState == ACTIVE || mState == RESUMING) { 1777 AudioSystem::stopInput(thread->id()); 1778 } 1779 AudioSystem::releaseInput(thread->id()); 1780 Mutex::Autolock _l(thread->mLock); 1781 RecordThread *recordThread = (RecordThread *) thread.get(); 1782 recordThread->destroyTrack_l(this); 1783 } 1784 } 1785} 1786 1787void AudioFlinger::RecordThread::RecordTrack::invalidate() 1788{ 1789 // FIXME should use proxy, and needs work 1790 audio_track_cblk_t* cblk = mCblk; 1791 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1792 android_atomic_release_store(0x40000000, &cblk->mFutex); 1793 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1794 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1795} 1796 1797 1798/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1799{ 1800 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1801} 1802 1803void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1804{ 1805 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1806 (mClient == 0) ? getpid_cached : mClient->pid(), 1807 mFormat, 1808 mChannelMask, 1809 mSessionId, 1810 mState, 1811 mCblk->mServer, 1812 mFrameCount); 1813} 1814 1815}; // namespace android 1816