Tracks.cpp revision 59fe010bcc072597852454a2ec53d7b0a2002a3b
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} 290 291 292void AudioFlinger::TrackHandle::signal() 293{ 294 return mTrack->signal(); 295} 296 297status_t AudioFlinger::TrackHandle::onTransact( 298 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 299{ 300 return BnAudioTrack::onTransact(code, data, reply, flags); 301} 302 303// ---------------------------------------------------------------------------- 304 305// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 306AudioFlinger::PlaybackThread::Track::Track( 307 PlaybackThread *thread, 308 const sp<Client>& client, 309 audio_stream_type_t streamType, 310 uint32_t sampleRate, 311 audio_format_t format, 312 audio_channel_mask_t channelMask, 313 size_t frameCount, 314 const sp<IMemory>& sharedBuffer, 315 int sessionId, 316 IAudioFlinger::track_flags_t flags) 317 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 318 sessionId, true /*isOut*/), 319 mFillingUpStatus(FS_INVALID), 320 // mRetryCount initialized later when needed 321 mSharedBuffer(sharedBuffer), 322 mStreamType(streamType), 323 mName(-1), // see note below 324 mMainBuffer(thread->mixBuffer()), 325 mAuxBuffer(NULL), 326 mAuxEffectId(0), mHasVolumeController(false), 327 mPresentationCompleteFrames(0), 328 mFlags(flags), 329 mFastIndex(-1), 330 mCachedVolume(1.0), 331 mIsInvalid(false), 332 mAudioTrackServerProxy(NULL), 333 mResumeToStopping(false) 334{ 335 if (mCblk != NULL) { 336 if (sharedBuffer == 0) { 337 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 338 mFrameSize); 339 } else { 340 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 341 mFrameSize); 342 } 343 mServerProxy = mAudioTrackServerProxy; 344 // to avoid leaking a track name, do not allocate one unless there is an mCblk 345 mName = thread->getTrackName_l(channelMask, sessionId); 346 if (mName < 0) { 347 ALOGE("no more track names available"); 348 return; 349 } 350 // only allocate a fast track index if we were able to allocate a normal track name 351 if (flags & IAudioFlinger::TRACK_FAST) { 352 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 353 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 354 int i = __builtin_ctz(thread->mFastTrackAvailMask); 355 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 356 // FIXME This is too eager. We allocate a fast track index before the 357 // fast track becomes active. Since fast tracks are a scarce resource, 358 // this means we are potentially denying other more important fast tracks from 359 // being created. It would be better to allocate the index dynamically. 360 mFastIndex = i; 361 // Read the initial underruns because this field is never cleared by the fast mixer 362 mObservedUnderruns = thread->getFastTrackUnderruns(i); 363 thread->mFastTrackAvailMask &= ~(1 << i); 364 } 365 } 366 ALOGV("Track constructor name %d, calling pid %d", mName, 367 IPCThreadState::self()->getCallingPid()); 368} 369 370AudioFlinger::PlaybackThread::Track::~Track() 371{ 372 ALOGV("PlaybackThread::Track destructor"); 373 374 // The destructor would clear mSharedBuffer, 375 // but it will not push the decremented reference count, 376 // leaving the client's IMemory dangling indefinitely. 377 // This prevents that leak. 378 if (mSharedBuffer != 0) { 379 mSharedBuffer.clear(); 380 // flush the binder command buffer 381 IPCThreadState::self()->flushCommands(); 382 } 383} 384 385void AudioFlinger::PlaybackThread::Track::destroy() 386{ 387 // NOTE: destroyTrack_l() can remove a strong reference to this Track 388 // by removing it from mTracks vector, so there is a risk that this Tracks's 389 // destructor is called. As the destructor needs to lock mLock, 390 // we must acquire a strong reference on this Track before locking mLock 391 // here so that the destructor is called only when exiting this function. 392 // On the other hand, as long as Track::destroy() is only called by 393 // TrackHandle destructor, the TrackHandle still holds a strong ref on 394 // this Track with its member mTrack. 395 sp<Track> keep(this); 396 { // scope for mLock 397 sp<ThreadBase> thread = mThread.promote(); 398 if (thread != 0) { 399 Mutex::Autolock _l(thread->mLock); 400 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 401 bool wasActive = playbackThread->destroyTrack_l(this); 402 if (!isOutputTrack() && !wasActive) { 403 AudioSystem::releaseOutput(thread->id()); 404 } 405 } 406 } 407} 408 409/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 410{ 411 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 412 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 413} 414 415void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 416{ 417 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 418 if (isFastTrack()) { 419 sprintf(buffer, " F %2d", mFastIndex); 420 } else { 421 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 422 } 423 track_state state = mState; 424 char stateChar; 425 if (isTerminated()) { 426 stateChar = 'T'; 427 } else { 428 switch (state) { 429 case IDLE: 430 stateChar = 'I'; 431 break; 432 case STOPPING_1: 433 stateChar = 's'; 434 break; 435 case STOPPING_2: 436 stateChar = '5'; 437 break; 438 case STOPPED: 439 stateChar = 'S'; 440 break; 441 case RESUMING: 442 stateChar = 'R'; 443 break; 444 case ACTIVE: 445 stateChar = 'A'; 446 break; 447 case PAUSING: 448 stateChar = 'p'; 449 break; 450 case PAUSED: 451 stateChar = 'P'; 452 break; 453 case FLUSHED: 454 stateChar = 'F'; 455 break; 456 default: 457 stateChar = '?'; 458 break; 459 } 460 } 461 char nowInUnderrun; 462 switch (mObservedUnderruns.mBitFields.mMostRecent) { 463 case UNDERRUN_FULL: 464 nowInUnderrun = ' '; 465 break; 466 case UNDERRUN_PARTIAL: 467 nowInUnderrun = '<'; 468 break; 469 case UNDERRUN_EMPTY: 470 nowInUnderrun = '*'; 471 break; 472 default: 473 nowInUnderrun = '?'; 474 break; 475 } 476 snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 477 "%08X %08X %08X 0x%03X %9u%c\n", 478 (mClient == 0) ? getpid_cached : mClient->pid(), 479 mStreamType, 480 mFormat, 481 mChannelMask, 482 mSessionId, 483 mFrameCount, 484 stateChar, 485 mFillingUpStatus, 486 mAudioTrackServerProxy->getSampleRate(), 487 20.0 * log10((vlr & 0xFFFF) / 4096.0), 488 20.0 * log10((vlr >> 16) / 4096.0), 489 mCblk->mServer, 490 (int)mMainBuffer, 491 (int)mAuxBuffer, 492 mCblk->mFlags, 493 mAudioTrackServerProxy->getUnderrunFrames(), 494 nowInUnderrun); 495} 496 497uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 498 return mAudioTrackServerProxy->getSampleRate(); 499} 500 501// AudioBufferProvider interface 502status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 503 AudioBufferProvider::Buffer* buffer, int64_t pts) 504{ 505 ServerProxy::Buffer buf; 506 size_t desiredFrames = buffer->frameCount; 507 buf.mFrameCount = desiredFrames; 508 status_t status = mServerProxy->obtainBuffer(&buf); 509 buffer->frameCount = buf.mFrameCount; 510 buffer->raw = buf.mRaw; 511 if (buf.mFrameCount == 0) { 512 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 513 } 514 return status; 515} 516 517// releaseBuffer() is not overridden 518 519// ExtendedAudioBufferProvider interface 520 521// Note that framesReady() takes a mutex on the control block using tryLock(). 522// This could result in priority inversion if framesReady() is called by the normal mixer, 523// as the normal mixer thread runs at lower 524// priority than the client's callback thread: there is a short window within framesReady() 525// during which the normal mixer could be preempted, and the client callback would block. 526// Another problem can occur if framesReady() is called by the fast mixer: 527// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 528// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 529size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 530 return mAudioTrackServerProxy->framesReady(); 531} 532 533size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 534{ 535 return mAudioTrackServerProxy->framesReleased(); 536} 537 538// Don't call for fast tracks; the framesReady() could result in priority inversion 539bool AudioFlinger::PlaybackThread::Track::isReady() const { 540 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 541 return true; 542 } 543 544 if (framesReady() >= mFrameCount || 545 (mCblk->mFlags & CBLK_FORCEREADY)) { 546 mFillingUpStatus = FS_FILLED; 547 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 548 return true; 549 } 550 return false; 551} 552 553status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 554 int triggerSession) 555{ 556 status_t status = NO_ERROR; 557 ALOGV("start(%d), calling pid %d session %d", 558 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 559 560 sp<ThreadBase> thread = mThread.promote(); 561 if (thread != 0) { 562 if (isOffloaded()) { 563 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 564 Mutex::Autolock _lth(thread->mLock); 565 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 566 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 567 (ec != 0 && ec->isNonOffloadableEnabled())) { 568 invalidate(); 569 return PERMISSION_DENIED; 570 } 571 } 572 Mutex::Autolock _lth(thread->mLock); 573 track_state state = mState; 574 // here the track could be either new, or restarted 575 // in both cases "unstop" the track 576 577 if (state == PAUSED) { 578 if (mResumeToStopping) { 579 // happened we need to resume to STOPPING_1 580 mState = TrackBase::STOPPING_1; 581 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 582 } else { 583 mState = TrackBase::RESUMING; 584 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 585 } 586 } else { 587 mState = TrackBase::ACTIVE; 588 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 589 } 590 591 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 592 status = playbackThread->addTrack_l(this); 593 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 594 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 595 // restore previous state if start was rejected by policy manager 596 if (status == PERMISSION_DENIED) { 597 mState = state; 598 } 599 } 600 // track was already in the active list, not a problem 601 if (status == ALREADY_EXISTS) { 602 status = NO_ERROR; 603 } 604 } else { 605 status = BAD_VALUE; 606 } 607 return status; 608} 609 610void AudioFlinger::PlaybackThread::Track::stop() 611{ 612 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 Mutex::Autolock _l(thread->mLock); 616 track_state state = mState; 617 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 618 // If the track is not active (PAUSED and buffers full), flush buffers 619 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 620 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 621 reset(); 622 mState = STOPPED; 623 } else if (!isFastTrack() && !isOffloaded()) { 624 mState = STOPPED; 625 } else { 626 // For fast tracks prepareTracks_l() will set state to STOPPING_2 627 // presentation is complete 628 // For an offloaded track this starts a drain and state will 629 // move to STOPPING_2 when drain completes and then STOPPED 630 mState = STOPPING_1; 631 } 632 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 633 playbackThread); 634 } 635 } 636} 637 638void AudioFlinger::PlaybackThread::Track::pause() 639{ 640 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 641 sp<ThreadBase> thread = mThread.promote(); 642 if (thread != 0) { 643 Mutex::Autolock _l(thread->mLock); 644 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 645 switch (mState) { 646 case STOPPING_1: 647 case STOPPING_2: 648 if (!isOffloaded()) { 649 /* nothing to do if track is not offloaded */ 650 break; 651 } 652 653 // Offloaded track was draining, we need to carry on draining when resumed 654 mResumeToStopping = true; 655 // fall through... 656 case ACTIVE: 657 case RESUMING: 658 mState = PAUSING; 659 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 660 playbackThread->broadcast_l(); 661 break; 662 663 default: 664 break; 665 } 666 } 667} 668 669void AudioFlinger::PlaybackThread::Track::flush() 670{ 671 ALOGV("flush(%d)", mName); 672 sp<ThreadBase> thread = mThread.promote(); 673 if (thread != 0) { 674 Mutex::Autolock _l(thread->mLock); 675 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 676 677 if (isOffloaded()) { 678 // If offloaded we allow flush during any state except terminated 679 // and keep the track active to avoid problems if user is seeking 680 // rapidly and underlying hardware has a significant delay handling 681 // a pause 682 if (isTerminated()) { 683 return; 684 } 685 686 ALOGV("flush: offload flush"); 687 reset(); 688 689 if (mState == STOPPING_1 || mState == STOPPING_2) { 690 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 691 mState = ACTIVE; 692 } 693 694 if (mState == ACTIVE) { 695 ALOGV("flush called in active state, resetting buffer time out retry count"); 696 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 697 } 698 699 mResumeToStopping = false; 700 } else { 701 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 702 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 703 return; 704 } 705 // No point remaining in PAUSED state after a flush => go to 706 // FLUSHED state 707 mState = FLUSHED; 708 // do not reset the track if it is still in the process of being stopped or paused. 709 // this will be done by prepareTracks_l() when the track is stopped. 710 // prepareTracks_l() will see mState == FLUSHED, then 711 // remove from active track list, reset(), and trigger presentation complete 712 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 713 reset(); 714 } 715 } 716 // Prevent flush being lost if the track is flushed and then resumed 717 // before mixer thread can run. This is important when offloading 718 // because the hardware buffer could hold a large amount of audio 719 playbackThread->flushOutput_l(); 720 playbackThread->broadcast_l(); 721 } 722} 723 724void AudioFlinger::PlaybackThread::Track::reset() 725{ 726 // Do not reset twice to avoid discarding data written just after a flush and before 727 // the audioflinger thread detects the track is stopped. 728 if (!mResetDone) { 729 // Force underrun condition to avoid false underrun callback until first data is 730 // written to buffer 731 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 732 mFillingUpStatus = FS_FILLING; 733 mResetDone = true; 734 if (mState == FLUSHED) { 735 mState = IDLE; 736 } 737 } 738} 739 740status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 741{ 742 sp<ThreadBase> thread = mThread.promote(); 743 if (thread == 0) { 744 ALOGE("thread is dead"); 745 return FAILED_TRANSACTION; 746 } else if ((thread->type() == ThreadBase::DIRECT) || 747 (thread->type() == ThreadBase::OFFLOAD)) { 748 return thread->setParameters(keyValuePairs); 749 } else { 750 return PERMISSION_DENIED; 751 } 752} 753 754status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 755{ 756 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 757 if (isFastTrack()) { 758 return INVALID_OPERATION; 759 } 760 sp<ThreadBase> thread = mThread.promote(); 761 if (thread == 0) { 762 return INVALID_OPERATION; 763 } 764 Mutex::Autolock _l(thread->mLock); 765 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 766 if (!isOffloaded()) { 767 if (!playbackThread->mLatchQValid) { 768 return INVALID_OPERATION; 769 } 770 uint32_t unpresentedFrames = 771 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 772 playbackThread->mSampleRate; 773 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 774 if (framesWritten < unpresentedFrames) { 775 return INVALID_OPERATION; 776 } 777 timestamp.mPosition = framesWritten - unpresentedFrames; 778 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 779 return NO_ERROR; 780 } 781 782 return playbackThread->getTimestamp_l(timestamp); 783} 784 785status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 786{ 787 status_t status = DEAD_OBJECT; 788 sp<ThreadBase> thread = mThread.promote(); 789 if (thread != 0) { 790 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 791 sp<AudioFlinger> af = mClient->audioFlinger(); 792 793 Mutex::Autolock _l(af->mLock); 794 795 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 796 797 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 798 Mutex::Autolock _dl(playbackThread->mLock); 799 Mutex::Autolock _sl(srcThread->mLock); 800 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 801 if (chain == 0) { 802 return INVALID_OPERATION; 803 } 804 805 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 806 if (effect == 0) { 807 return INVALID_OPERATION; 808 } 809 srcThread->removeEffect_l(effect); 810 status = playbackThread->addEffect_l(effect); 811 if (status != NO_ERROR) { 812 srcThread->addEffect_l(effect); 813 return INVALID_OPERATION; 814 } 815 // removeEffect_l() has stopped the effect if it was active so it must be restarted 816 if (effect->state() == EffectModule::ACTIVE || 817 effect->state() == EffectModule::STOPPING) { 818 effect->start(); 819 } 820 821 sp<EffectChain> dstChain = effect->chain().promote(); 822 if (dstChain == 0) { 823 srcThread->addEffect_l(effect); 824 return INVALID_OPERATION; 825 } 826 AudioSystem::unregisterEffect(effect->id()); 827 AudioSystem::registerEffect(&effect->desc(), 828 srcThread->id(), 829 dstChain->strategy(), 830 AUDIO_SESSION_OUTPUT_MIX, 831 effect->id()); 832 } 833 status = playbackThread->attachAuxEffect(this, EffectId); 834 } 835 return status; 836} 837 838void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 839{ 840 mAuxEffectId = EffectId; 841 mAuxBuffer = buffer; 842} 843 844bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 845 size_t audioHalFrames) 846{ 847 // a track is considered presented when the total number of frames written to audio HAL 848 // corresponds to the number of frames written when presentationComplete() is called for the 849 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 850 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 851 // to detect when all frames have been played. In this case framesWritten isn't 852 // useful because it doesn't always reflect whether there is data in the h/w 853 // buffers, particularly if a track has been paused and resumed during draining 854 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 855 mPresentationCompleteFrames, framesWritten); 856 if (mPresentationCompleteFrames == 0) { 857 mPresentationCompleteFrames = framesWritten + audioHalFrames; 858 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 859 mPresentationCompleteFrames, audioHalFrames); 860 } 861 862 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 863 ALOGV("presentationComplete() session %d complete: framesWritten %d", 864 mSessionId, framesWritten); 865 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 866 mAudioTrackServerProxy->setStreamEndDone(); 867 return true; 868 } 869 return false; 870} 871 872void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 873{ 874 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 875 if (mSyncEvents[i]->type() == type) { 876 mSyncEvents[i]->trigger(); 877 mSyncEvents.removeAt(i); 878 i--; 879 } 880 } 881} 882 883// implement VolumeBufferProvider interface 884 885uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 886{ 887 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 888 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 889 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 890 uint32_t vl = vlr & 0xFFFF; 891 uint32_t vr = vlr >> 16; 892 // track volumes come from shared memory, so can't be trusted and must be clamped 893 if (vl > MAX_GAIN_INT) { 894 vl = MAX_GAIN_INT; 895 } 896 if (vr > MAX_GAIN_INT) { 897 vr = MAX_GAIN_INT; 898 } 899 // now apply the cached master volume and stream type volume; 900 // this is trusted but lacks any synchronization or barrier so may be stale 901 float v = mCachedVolume; 902 vl *= v; 903 vr *= v; 904 // re-combine into U4.16 905 vlr = (vr << 16) | (vl & 0xFFFF); 906 // FIXME look at mute, pause, and stop flags 907 return vlr; 908} 909 910status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 911{ 912 if (isTerminated() || mState == PAUSED || 913 ((framesReady() == 0) && ((mSharedBuffer != 0) || 914 (mState == STOPPED)))) { 915 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 916 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 917 event->cancel(); 918 return INVALID_OPERATION; 919 } 920 (void) TrackBase::setSyncEvent(event); 921 return NO_ERROR; 922} 923 924void AudioFlinger::PlaybackThread::Track::invalidate() 925{ 926 // FIXME should use proxy, and needs work 927 audio_track_cblk_t* cblk = mCblk; 928 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 929 android_atomic_release_store(0x40000000, &cblk->mFutex); 930 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 931 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 932 mIsInvalid = true; 933} 934 935void AudioFlinger::PlaybackThread::Track::signal() 936{ 937 sp<ThreadBase> thread = mThread.promote(); 938 if (thread != 0) { 939 PlaybackThread *t = (PlaybackThread *)thread.get(); 940 Mutex::Autolock _l(t->mLock); 941 t->broadcast_l(); 942 } 943} 944 945// ---------------------------------------------------------------------------- 946 947sp<AudioFlinger::PlaybackThread::TimedTrack> 948AudioFlinger::PlaybackThread::TimedTrack::create( 949 PlaybackThread *thread, 950 const sp<Client>& client, 951 audio_stream_type_t streamType, 952 uint32_t sampleRate, 953 audio_format_t format, 954 audio_channel_mask_t channelMask, 955 size_t frameCount, 956 const sp<IMemory>& sharedBuffer, 957 int sessionId) { 958 if (!client->reserveTimedTrack()) 959 return 0; 960 961 return new TimedTrack( 962 thread, client, streamType, sampleRate, format, channelMask, frameCount, 963 sharedBuffer, sessionId); 964} 965 966AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 967 PlaybackThread *thread, 968 const sp<Client>& client, 969 audio_stream_type_t streamType, 970 uint32_t sampleRate, 971 audio_format_t format, 972 audio_channel_mask_t channelMask, 973 size_t frameCount, 974 const sp<IMemory>& sharedBuffer, 975 int sessionId) 976 : Track(thread, client, streamType, sampleRate, format, channelMask, 977 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 978 mQueueHeadInFlight(false), 979 mTrimQueueHeadOnRelease(false), 980 mFramesPendingInQueue(0), 981 mTimedSilenceBuffer(NULL), 982 mTimedSilenceBufferSize(0), 983 mTimedAudioOutputOnTime(false), 984 mMediaTimeTransformValid(false) 985{ 986 LocalClock lc; 987 mLocalTimeFreq = lc.getLocalFreq(); 988 989 mLocalTimeToSampleTransform.a_zero = 0; 990 mLocalTimeToSampleTransform.b_zero = 0; 991 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 992 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 993 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 994 &mLocalTimeToSampleTransform.a_to_b_denom); 995 996 mMediaTimeToSampleTransform.a_zero = 0; 997 mMediaTimeToSampleTransform.b_zero = 0; 998 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 999 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1000 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1001 &mMediaTimeToSampleTransform.a_to_b_denom); 1002} 1003 1004AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1005 mClient->releaseTimedTrack(); 1006 delete [] mTimedSilenceBuffer; 1007} 1008 1009status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1010 size_t size, sp<IMemory>* buffer) { 1011 1012 Mutex::Autolock _l(mTimedBufferQueueLock); 1013 1014 trimTimedBufferQueue_l(); 1015 1016 // lazily initialize the shared memory heap for timed buffers 1017 if (mTimedMemoryDealer == NULL) { 1018 const int kTimedBufferHeapSize = 512 << 10; 1019 1020 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1021 "AudioFlingerTimed"); 1022 if (mTimedMemoryDealer == NULL) 1023 return NO_MEMORY; 1024 } 1025 1026 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1027 if (newBuffer == NULL) { 1028 newBuffer = mTimedMemoryDealer->allocate(size); 1029 if (newBuffer == NULL) 1030 return NO_MEMORY; 1031 } 1032 1033 *buffer = newBuffer; 1034 return NO_ERROR; 1035} 1036 1037// caller must hold mTimedBufferQueueLock 1038void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1039 int64_t mediaTimeNow; 1040 { 1041 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1042 if (!mMediaTimeTransformValid) 1043 return; 1044 1045 int64_t targetTimeNow; 1046 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1047 ? mCCHelper.getCommonTime(&targetTimeNow) 1048 : mCCHelper.getLocalTime(&targetTimeNow); 1049 1050 if (OK != res) 1051 return; 1052 1053 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1054 &mediaTimeNow)) { 1055 return; 1056 } 1057 } 1058 1059 size_t trimEnd; 1060 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1061 int64_t bufEnd; 1062 1063 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1064 // We have a next buffer. Just use its PTS as the PTS of the frame 1065 // following the last frame in this buffer. If the stream is sparse 1066 // (ie, there are deliberate gaps left in the stream which should be 1067 // filled with silence by the TimedAudioTrack), then this can result 1068 // in one extra buffer being left un-trimmed when it could have 1069 // been. In general, this is not typical, and we would rather 1070 // optimized away the TS calculation below for the more common case 1071 // where PTSes are contiguous. 1072 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1073 } else { 1074 // We have no next buffer. Compute the PTS of the frame following 1075 // the last frame in this buffer by computing the duration of of 1076 // this frame in media time units and adding it to the PTS of the 1077 // buffer. 1078 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1079 / mFrameSize; 1080 1081 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1082 &bufEnd)) { 1083 ALOGE("Failed to convert frame count of %lld to media time" 1084 " duration" " (scale factor %d/%u) in %s", 1085 frameCount, 1086 mMediaTimeToSampleTransform.a_to_b_numer, 1087 mMediaTimeToSampleTransform.a_to_b_denom, 1088 __PRETTY_FUNCTION__); 1089 break; 1090 } 1091 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1092 } 1093 1094 if (bufEnd > mediaTimeNow) 1095 break; 1096 1097 // Is the buffer we want to use in the middle of a mix operation right 1098 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1099 // from the mixer which should be coming back shortly. 1100 if (!trimEnd && mQueueHeadInFlight) { 1101 mTrimQueueHeadOnRelease = true; 1102 } 1103 } 1104 1105 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1106 if (trimStart < trimEnd) { 1107 // Update the bookkeeping for framesReady() 1108 for (size_t i = trimStart; i < trimEnd; ++i) { 1109 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1110 } 1111 1112 // Now actually remove the buffers from the queue. 1113 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1114 } 1115} 1116 1117void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1118 const char* logTag) { 1119 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1120 "%s called (reason \"%s\"), but timed buffer queue has no" 1121 " elements to trim.", __FUNCTION__, logTag); 1122 1123 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1124 mTimedBufferQueue.removeAt(0); 1125} 1126 1127void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1128 const TimedBuffer& buf, 1129 const char* logTag) { 1130 uint32_t bufBytes = buf.buffer()->size(); 1131 uint32_t consumedAlready = buf.position(); 1132 1133 ALOG_ASSERT(consumedAlready <= bufBytes, 1134 "Bad bookkeeping while updating frames pending. Timed buffer is" 1135 " only %u bytes long, but claims to have consumed %u" 1136 " bytes. (update reason: \"%s\")", 1137 bufBytes, consumedAlready, logTag); 1138 1139 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1140 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1141 "Bad bookkeeping while updating frames pending. Should have at" 1142 " least %u queued frames, but we think we have only %u. (update" 1143 " reason: \"%s\")", 1144 bufFrames, mFramesPendingInQueue, logTag); 1145 1146 mFramesPendingInQueue -= bufFrames; 1147} 1148 1149status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1150 const sp<IMemory>& buffer, int64_t pts) { 1151 1152 { 1153 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1154 if (!mMediaTimeTransformValid) 1155 return INVALID_OPERATION; 1156 } 1157 1158 Mutex::Autolock _l(mTimedBufferQueueLock); 1159 1160 uint32_t bufFrames = buffer->size() / mFrameSize; 1161 mFramesPendingInQueue += bufFrames; 1162 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1163 1164 return NO_ERROR; 1165} 1166 1167status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1168 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1169 1170 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1171 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1172 target); 1173 1174 if (!(target == TimedAudioTrack::LOCAL_TIME || 1175 target == TimedAudioTrack::COMMON_TIME)) { 1176 return BAD_VALUE; 1177 } 1178 1179 Mutex::Autolock lock(mMediaTimeTransformLock); 1180 mMediaTimeTransform = xform; 1181 mMediaTimeTransformTarget = target; 1182 mMediaTimeTransformValid = true; 1183 1184 return NO_ERROR; 1185} 1186 1187#define min(a, b) ((a) < (b) ? (a) : (b)) 1188 1189// implementation of getNextBuffer for tracks whose buffers have timestamps 1190status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1191 AudioBufferProvider::Buffer* buffer, int64_t pts) 1192{ 1193 if (pts == AudioBufferProvider::kInvalidPTS) { 1194 buffer->raw = NULL; 1195 buffer->frameCount = 0; 1196 mTimedAudioOutputOnTime = false; 1197 return INVALID_OPERATION; 1198 } 1199 1200 Mutex::Autolock _l(mTimedBufferQueueLock); 1201 1202 ALOG_ASSERT(!mQueueHeadInFlight, 1203 "getNextBuffer called without releaseBuffer!"); 1204 1205 while (true) { 1206 1207 // if we have no timed buffers, then fail 1208 if (mTimedBufferQueue.isEmpty()) { 1209 buffer->raw = NULL; 1210 buffer->frameCount = 0; 1211 return NOT_ENOUGH_DATA; 1212 } 1213 1214 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1215 1216 // calculate the PTS of the head of the timed buffer queue expressed in 1217 // local time 1218 int64_t headLocalPTS; 1219 { 1220 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1221 1222 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1223 1224 if (mMediaTimeTransform.a_to_b_denom == 0) { 1225 // the transform represents a pause, so yield silence 1226 timedYieldSilence_l(buffer->frameCount, buffer); 1227 return NO_ERROR; 1228 } 1229 1230 int64_t transformedPTS; 1231 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1232 &transformedPTS)) { 1233 // the transform failed. this shouldn't happen, but if it does 1234 // then just drop this buffer 1235 ALOGW("timedGetNextBuffer transform failed"); 1236 buffer->raw = NULL; 1237 buffer->frameCount = 0; 1238 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1239 return NO_ERROR; 1240 } 1241 1242 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1243 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1244 &headLocalPTS)) { 1245 buffer->raw = NULL; 1246 buffer->frameCount = 0; 1247 return INVALID_OPERATION; 1248 } 1249 } else { 1250 headLocalPTS = transformedPTS; 1251 } 1252 } 1253 1254 uint32_t sr = sampleRate(); 1255 1256 // adjust the head buffer's PTS to reflect the portion of the head buffer 1257 // that has already been consumed 1258 int64_t effectivePTS = headLocalPTS + 1259 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1260 1261 // Calculate the delta in samples between the head of the input buffer 1262 // queue and the start of the next output buffer that will be written. 1263 // If the transformation fails because of over or underflow, it means 1264 // that the sample's position in the output stream is so far out of 1265 // whack that it should just be dropped. 1266 int64_t sampleDelta; 1267 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1268 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1269 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1270 " mix"); 1271 continue; 1272 } 1273 if (!mLocalTimeToSampleTransform.doForwardTransform( 1274 (effectivePTS - pts) << 32, &sampleDelta)) { 1275 ALOGV("*** too late during sample rate transform: dropped buffer"); 1276 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1277 continue; 1278 } 1279 1280 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1281 " sampleDelta=[%d.%08x]", 1282 head.pts(), head.position(), pts, 1283 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1284 + (sampleDelta >> 32)), 1285 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1286 1287 // if the delta between the ideal placement for the next input sample and 1288 // the current output position is within this threshold, then we will 1289 // concatenate the next input samples to the previous output 1290 const int64_t kSampleContinuityThreshold = 1291 (static_cast<int64_t>(sr) << 32) / 250; 1292 1293 // if this is the first buffer of audio that we're emitting from this track 1294 // then it should be almost exactly on time. 1295 const int64_t kSampleStartupThreshold = 1LL << 32; 1296 1297 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1298 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1299 // the next input is close enough to being on time, so concatenate it 1300 // with the last output 1301 timedYieldSamples_l(buffer); 1302 1303 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1304 head.position(), buffer->frameCount); 1305 return NO_ERROR; 1306 } 1307 1308 // Looks like our output is not on time. Reset our on timed status. 1309 // Next time we mix samples from our input queue, then should be within 1310 // the StartupThreshold. 1311 mTimedAudioOutputOnTime = false; 1312 if (sampleDelta > 0) { 1313 // the gap between the current output position and the proper start of 1314 // the next input sample is too big, so fill it with silence 1315 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1316 1317 timedYieldSilence_l(framesUntilNextInput, buffer); 1318 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1319 return NO_ERROR; 1320 } else { 1321 // the next input sample is late 1322 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1323 size_t onTimeSamplePosition = 1324 head.position() + lateFrames * mFrameSize; 1325 1326 if (onTimeSamplePosition > head.buffer()->size()) { 1327 // all the remaining samples in the head are too late, so 1328 // drop it and move on 1329 ALOGV("*** too late: dropped buffer"); 1330 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1331 continue; 1332 } else { 1333 // skip over the late samples 1334 head.setPosition(onTimeSamplePosition); 1335 1336 // yield the available samples 1337 timedYieldSamples_l(buffer); 1338 1339 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1340 return NO_ERROR; 1341 } 1342 } 1343 } 1344} 1345 1346// Yield samples from the timed buffer queue head up to the given output 1347// buffer's capacity. 1348// 1349// Caller must hold mTimedBufferQueueLock 1350void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1351 AudioBufferProvider::Buffer* buffer) { 1352 1353 const TimedBuffer& head = mTimedBufferQueue[0]; 1354 1355 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1356 head.position()); 1357 1358 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1359 mFrameSize); 1360 size_t framesRequested = buffer->frameCount; 1361 buffer->frameCount = min(framesLeftInHead, framesRequested); 1362 1363 mQueueHeadInFlight = true; 1364 mTimedAudioOutputOnTime = true; 1365} 1366 1367// Yield samples of silence up to the given output buffer's capacity 1368// 1369// Caller must hold mTimedBufferQueueLock 1370void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1371 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1372 1373 // lazily allocate a buffer filled with silence 1374 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1375 delete [] mTimedSilenceBuffer; 1376 mTimedSilenceBufferSize = numFrames * mFrameSize; 1377 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1378 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1379 } 1380 1381 buffer->raw = mTimedSilenceBuffer; 1382 size_t framesRequested = buffer->frameCount; 1383 buffer->frameCount = min(numFrames, framesRequested); 1384 1385 mTimedAudioOutputOnTime = false; 1386} 1387 1388// AudioBufferProvider interface 1389void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1390 AudioBufferProvider::Buffer* buffer) { 1391 1392 Mutex::Autolock _l(mTimedBufferQueueLock); 1393 1394 // If the buffer which was just released is part of the buffer at the head 1395 // of the queue, be sure to update the amt of the buffer which has been 1396 // consumed. If the buffer being returned is not part of the head of the 1397 // queue, its either because the buffer is part of the silence buffer, or 1398 // because the head of the timed queue was trimmed after the mixer called 1399 // getNextBuffer but before the mixer called releaseBuffer. 1400 if (buffer->raw == mTimedSilenceBuffer) { 1401 ALOG_ASSERT(!mQueueHeadInFlight, 1402 "Queue head in flight during release of silence buffer!"); 1403 goto done; 1404 } 1405 1406 ALOG_ASSERT(mQueueHeadInFlight, 1407 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1408 " head in flight."); 1409 1410 if (mTimedBufferQueue.size()) { 1411 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1412 1413 void* start = head.buffer()->pointer(); 1414 void* end = reinterpret_cast<void*>( 1415 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1416 + head.buffer()->size()); 1417 1418 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1419 "released buffer not within the head of the timed buffer" 1420 " queue; qHead = [%p, %p], released buffer = %p", 1421 start, end, buffer->raw); 1422 1423 head.setPosition(head.position() + 1424 (buffer->frameCount * mFrameSize)); 1425 mQueueHeadInFlight = false; 1426 1427 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1428 "Bad bookkeeping during releaseBuffer! Should have at" 1429 " least %u queued frames, but we think we have only %u", 1430 buffer->frameCount, mFramesPendingInQueue); 1431 1432 mFramesPendingInQueue -= buffer->frameCount; 1433 1434 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1435 || mTrimQueueHeadOnRelease) { 1436 trimTimedBufferQueueHead_l("releaseBuffer"); 1437 mTrimQueueHeadOnRelease = false; 1438 } 1439 } else { 1440 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1441 " buffers in the timed buffer queue"); 1442 } 1443 1444done: 1445 buffer->raw = 0; 1446 buffer->frameCount = 0; 1447} 1448 1449size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1450 Mutex::Autolock _l(mTimedBufferQueueLock); 1451 return mFramesPendingInQueue; 1452} 1453 1454AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1455 : mPTS(0), mPosition(0) {} 1456 1457AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1458 const sp<IMemory>& buffer, int64_t pts) 1459 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1460 1461 1462// ---------------------------------------------------------------------------- 1463 1464AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1465 PlaybackThread *playbackThread, 1466 DuplicatingThread *sourceThread, 1467 uint32_t sampleRate, 1468 audio_format_t format, 1469 audio_channel_mask_t channelMask, 1470 size_t frameCount) 1471 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1472 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1473 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1474{ 1475 1476 if (mCblk != NULL) { 1477 mOutBuffer.frameCount = 0; 1478 playbackThread->mTracks.add(this); 1479 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1480 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1481 mCblk, mBuffer, 1482 mCblk->frameCount_, mChannelMask); 1483 // since client and server are in the same process, 1484 // the buffer has the same virtual address on both sides 1485 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1486 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1487 mClientProxy->setSendLevel(0.0); 1488 mClientProxy->setSampleRate(sampleRate); 1489 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1490 true /*clientInServer*/); 1491 } else { 1492 ALOGW("Error creating output track on thread %p", playbackThread); 1493 } 1494} 1495 1496AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1497{ 1498 clearBufferQueue(); 1499 delete mClientProxy; 1500 // superclass destructor will now delete the server proxy and shared memory both refer to 1501} 1502 1503status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1504 int triggerSession) 1505{ 1506 status_t status = Track::start(event, triggerSession); 1507 if (status != NO_ERROR) { 1508 return status; 1509 } 1510 1511 mActive = true; 1512 mRetryCount = 127; 1513 return status; 1514} 1515 1516void AudioFlinger::PlaybackThread::OutputTrack::stop() 1517{ 1518 Track::stop(); 1519 clearBufferQueue(); 1520 mOutBuffer.frameCount = 0; 1521 mActive = false; 1522} 1523 1524bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1525{ 1526 Buffer *pInBuffer; 1527 Buffer inBuffer; 1528 uint32_t channelCount = mChannelCount; 1529 bool outputBufferFull = false; 1530 inBuffer.frameCount = frames; 1531 inBuffer.i16 = data; 1532 1533 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1534 1535 if (!mActive && frames != 0) { 1536 start(); 1537 sp<ThreadBase> thread = mThread.promote(); 1538 if (thread != 0) { 1539 MixerThread *mixerThread = (MixerThread *)thread.get(); 1540 if (mFrameCount > frames) { 1541 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1542 uint32_t startFrames = (mFrameCount - frames); 1543 pInBuffer = new Buffer; 1544 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1545 pInBuffer->frameCount = startFrames; 1546 pInBuffer->i16 = pInBuffer->mBuffer; 1547 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1548 mBufferQueue.add(pInBuffer); 1549 } else { 1550 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1551 } 1552 } 1553 } 1554 } 1555 1556 while (waitTimeLeftMs) { 1557 // First write pending buffers, then new data 1558 if (mBufferQueue.size()) { 1559 pInBuffer = mBufferQueue.itemAt(0); 1560 } else { 1561 pInBuffer = &inBuffer; 1562 } 1563 1564 if (pInBuffer->frameCount == 0) { 1565 break; 1566 } 1567 1568 if (mOutBuffer.frameCount == 0) { 1569 mOutBuffer.frameCount = pInBuffer->frameCount; 1570 nsecs_t startTime = systemTime(); 1571 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1572 if (status != NO_ERROR) { 1573 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1574 mThread.unsafe_get(), status); 1575 outputBufferFull = true; 1576 break; 1577 } 1578 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1579 if (waitTimeLeftMs >= waitTimeMs) { 1580 waitTimeLeftMs -= waitTimeMs; 1581 } else { 1582 waitTimeLeftMs = 0; 1583 } 1584 } 1585 1586 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1587 pInBuffer->frameCount; 1588 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1589 Proxy::Buffer buf; 1590 buf.mFrameCount = outFrames; 1591 buf.mRaw = NULL; 1592 mClientProxy->releaseBuffer(&buf); 1593 pInBuffer->frameCount -= outFrames; 1594 pInBuffer->i16 += outFrames * channelCount; 1595 mOutBuffer.frameCount -= outFrames; 1596 mOutBuffer.i16 += outFrames * channelCount; 1597 1598 if (pInBuffer->frameCount == 0) { 1599 if (mBufferQueue.size()) { 1600 mBufferQueue.removeAt(0); 1601 delete [] pInBuffer->mBuffer; 1602 delete pInBuffer; 1603 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1604 mThread.unsafe_get(), mBufferQueue.size()); 1605 } else { 1606 break; 1607 } 1608 } 1609 } 1610 1611 // If we could not write all frames, allocate a buffer and queue it for next time. 1612 if (inBuffer.frameCount) { 1613 sp<ThreadBase> thread = mThread.promote(); 1614 if (thread != 0 && !thread->standby()) { 1615 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1616 pInBuffer = new Buffer; 1617 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1618 pInBuffer->frameCount = inBuffer.frameCount; 1619 pInBuffer->i16 = pInBuffer->mBuffer; 1620 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1621 sizeof(int16_t)); 1622 mBufferQueue.add(pInBuffer); 1623 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1624 mThread.unsafe_get(), mBufferQueue.size()); 1625 } else { 1626 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1627 mThread.unsafe_get(), this); 1628 } 1629 } 1630 } 1631 1632 // Calling write() with a 0 length buffer, means that no more data will be written: 1633 // If no more buffers are pending, fill output track buffer to make sure it is started 1634 // by output mixer. 1635 if (frames == 0 && mBufferQueue.size() == 0) { 1636 // FIXME borken, replace by getting framesReady() from proxy 1637 size_t user = 0; // was mCblk->user 1638 if (user < mFrameCount) { 1639 frames = mFrameCount - user; 1640 pInBuffer = new Buffer; 1641 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1642 pInBuffer->frameCount = frames; 1643 pInBuffer->i16 = pInBuffer->mBuffer; 1644 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1645 mBufferQueue.add(pInBuffer); 1646 } else if (mActive) { 1647 stop(); 1648 } 1649 } 1650 1651 return outputBufferFull; 1652} 1653 1654status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1655 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1656{ 1657 ClientProxy::Buffer buf; 1658 buf.mFrameCount = buffer->frameCount; 1659 struct timespec timeout; 1660 timeout.tv_sec = waitTimeMs / 1000; 1661 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1662 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1663 buffer->frameCount = buf.mFrameCount; 1664 buffer->raw = buf.mRaw; 1665 return status; 1666} 1667 1668void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1669{ 1670 size_t size = mBufferQueue.size(); 1671 1672 for (size_t i = 0; i < size; i++) { 1673 Buffer *pBuffer = mBufferQueue.itemAt(i); 1674 delete [] pBuffer->mBuffer; 1675 delete pBuffer; 1676 } 1677 mBufferQueue.clear(); 1678} 1679 1680 1681// ---------------------------------------------------------------------------- 1682// Record 1683// ---------------------------------------------------------------------------- 1684 1685AudioFlinger::RecordHandle::RecordHandle( 1686 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1687 : BnAudioRecord(), 1688 mRecordTrack(recordTrack) 1689{ 1690} 1691 1692AudioFlinger::RecordHandle::~RecordHandle() { 1693 stop_nonvirtual(); 1694 mRecordTrack->destroy(); 1695} 1696 1697sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1698 return mRecordTrack->getCblk(); 1699} 1700 1701status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1702 int triggerSession) { 1703 ALOGV("RecordHandle::start()"); 1704 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1705} 1706 1707void AudioFlinger::RecordHandle::stop() { 1708 stop_nonvirtual(); 1709} 1710 1711void AudioFlinger::RecordHandle::stop_nonvirtual() { 1712 ALOGV("RecordHandle::stop()"); 1713 mRecordTrack->stop(); 1714} 1715 1716status_t AudioFlinger::RecordHandle::onTransact( 1717 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1718{ 1719 return BnAudioRecord::onTransact(code, data, reply, flags); 1720} 1721 1722// ---------------------------------------------------------------------------- 1723 1724// RecordTrack constructor must be called with AudioFlinger::mLock held 1725AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1726 RecordThread *thread, 1727 const sp<Client>& client, 1728 uint32_t sampleRate, 1729 audio_format_t format, 1730 audio_channel_mask_t channelMask, 1731 size_t frameCount, 1732 int sessionId) 1733 : TrackBase(thread, client, sampleRate, format, 1734 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1735 mOverflow(false) 1736{ 1737 ALOGV("RecordTrack constructor"); 1738 if (mCblk != NULL) { 1739 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1740 mFrameSize); 1741 mServerProxy = mAudioRecordServerProxy; 1742 } 1743} 1744 1745AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1746{ 1747 ALOGV("%s", __func__); 1748} 1749 1750// AudioBufferProvider interface 1751status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1752 int64_t pts) 1753{ 1754 ServerProxy::Buffer buf; 1755 buf.mFrameCount = buffer->frameCount; 1756 status_t status = mServerProxy->obtainBuffer(&buf); 1757 buffer->frameCount = buf.mFrameCount; 1758 buffer->raw = buf.mRaw; 1759 if (buf.mFrameCount == 0) { 1760 // FIXME also wake futex so that overrun is noticed more quickly 1761 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1762 } 1763 return status; 1764} 1765 1766status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1767 int triggerSession) 1768{ 1769 sp<ThreadBase> thread = mThread.promote(); 1770 if (thread != 0) { 1771 RecordThread *recordThread = (RecordThread *)thread.get(); 1772 return recordThread->start(this, event, triggerSession); 1773 } else { 1774 return BAD_VALUE; 1775 } 1776} 1777 1778void AudioFlinger::RecordThread::RecordTrack::stop() 1779{ 1780 sp<ThreadBase> thread = mThread.promote(); 1781 if (thread != 0) { 1782 RecordThread *recordThread = (RecordThread *)thread.get(); 1783 if (recordThread->stop(this)) { 1784 AudioSystem::stopInput(recordThread->id()); 1785 } 1786 } 1787} 1788 1789void AudioFlinger::RecordThread::RecordTrack::destroy() 1790{ 1791 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1792 sp<RecordTrack> keep(this); 1793 { 1794 sp<ThreadBase> thread = mThread.promote(); 1795 if (thread != 0) { 1796 if (mState == ACTIVE || mState == RESUMING) { 1797 AudioSystem::stopInput(thread->id()); 1798 } 1799 AudioSystem::releaseInput(thread->id()); 1800 Mutex::Autolock _l(thread->mLock); 1801 RecordThread *recordThread = (RecordThread *) thread.get(); 1802 recordThread->destroyTrack_l(this); 1803 } 1804 } 1805} 1806 1807void AudioFlinger::RecordThread::RecordTrack::invalidate() 1808{ 1809 // FIXME should use proxy, and needs work 1810 audio_track_cblk_t* cblk = mCblk; 1811 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1812 android_atomic_release_store(0x40000000, &cblk->mFutex); 1813 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1814 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1815} 1816 1817 1818/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1819{ 1820 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1821} 1822 1823void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1824{ 1825 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1826 (mClient == 0) ? getpid_cached : mClient->pid(), 1827 mFormat, 1828 mChannelMask, 1829 mSessionId, 1830 mState, 1831 mCblk->mServer, 1832 mFrameCount); 1833} 1834 1835}; // namespace android 1836