Tracks.cpp revision d5577f26de1ae3a0dc6fbea9c60a07d585f894bf
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 369status_t AudioFlinger::PlaybackThread::Track::initCheck() const 370{ 371 status_t status = TrackBase::initCheck(); 372 if (status == NO_ERROR && mName < 0) { 373 status = NO_MEMORY; 374 } 375 return status; 376} 377 378void AudioFlinger::PlaybackThread::Track::destroy() 379{ 380 // NOTE: destroyTrack_l() can remove a strong reference to this Track 381 // by removing it from mTracks vector, so there is a risk that this Tracks's 382 // destructor is called. As the destructor needs to lock mLock, 383 // we must acquire a strong reference on this Track before locking mLock 384 // here so that the destructor is called only when exiting this function. 385 // On the other hand, as long as Track::destroy() is only called by 386 // TrackHandle destructor, the TrackHandle still holds a strong ref on 387 // this Track with its member mTrack. 388 sp<Track> keep(this); 389 { // scope for mLock 390 sp<ThreadBase> thread = mThread.promote(); 391 if (thread != 0) { 392 Mutex::Autolock _l(thread->mLock); 393 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 394 bool wasActive = playbackThread->destroyTrack_l(this); 395 if (!isOutputTrack() && !wasActive) { 396 AudioSystem::releaseOutput(thread->id()); 397 } 398 } 399 } 400} 401 402/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 403{ 404 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 405 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 406} 407 408void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 409{ 410 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 411 if (isFastTrack()) { 412 sprintf(buffer, " F %2d", mFastIndex); 413 } else { 414 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 415 } 416 track_state state = mState; 417 char stateChar; 418 if (isTerminated()) { 419 stateChar = 'T'; 420 } else { 421 switch (state) { 422 case IDLE: 423 stateChar = 'I'; 424 break; 425 case STOPPING_1: 426 stateChar = 's'; 427 break; 428 case STOPPING_2: 429 stateChar = '5'; 430 break; 431 case STOPPED: 432 stateChar = 'S'; 433 break; 434 case RESUMING: 435 stateChar = 'R'; 436 break; 437 case ACTIVE: 438 stateChar = 'A'; 439 break; 440 case PAUSING: 441 stateChar = 'p'; 442 break; 443 case PAUSED: 444 stateChar = 'P'; 445 break; 446 case FLUSHED: 447 stateChar = 'F'; 448 break; 449 default: 450 stateChar = '?'; 451 break; 452 } 453 } 454 char nowInUnderrun; 455 switch (mObservedUnderruns.mBitFields.mMostRecent) { 456 case UNDERRUN_FULL: 457 nowInUnderrun = ' '; 458 break; 459 case UNDERRUN_PARTIAL: 460 nowInUnderrun = '<'; 461 break; 462 case UNDERRUN_EMPTY: 463 nowInUnderrun = '*'; 464 break; 465 default: 466 nowInUnderrun = '?'; 467 break; 468 } 469 snprintf(&buffer[7], size-7, " %6u %4u %3u %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 470 "%08X %08X %08X 0x%03X %9u%c\n", 471 (mClient == 0) ? getpid_cached : mClient->pid(), 472 mStreamType, 473 mFormat, 474 mChannelMask, 475 mSessionId, 476 mFrameCount, 477 stateChar, 478 mFillingUpStatus, 479 mAudioTrackServerProxy->getSampleRate(), 480 20.0 * log10((vlr & 0xFFFF) / 4096.0), 481 20.0 * log10((vlr >> 16) / 4096.0), 482 mCblk->mServer, 483 (int)mMainBuffer, 484 (int)mAuxBuffer, 485 mCblk->mFlags, 486 mAudioTrackServerProxy->getUnderrunFrames(), 487 nowInUnderrun); 488} 489 490uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 491 return mAudioTrackServerProxy->getSampleRate(); 492} 493 494// AudioBufferProvider interface 495status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 496 AudioBufferProvider::Buffer* buffer, int64_t pts) 497{ 498 ServerProxy::Buffer buf; 499 size_t desiredFrames = buffer->frameCount; 500 buf.mFrameCount = desiredFrames; 501 status_t status = mServerProxy->obtainBuffer(&buf); 502 buffer->frameCount = buf.mFrameCount; 503 buffer->raw = buf.mRaw; 504 if (buf.mFrameCount == 0) { 505 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 506 } 507 return status; 508} 509 510// releaseBuffer() is not overridden 511 512// ExtendedAudioBufferProvider interface 513 514// Note that framesReady() takes a mutex on the control block using tryLock(). 515// This could result in priority inversion if framesReady() is called by the normal mixer, 516// as the normal mixer thread runs at lower 517// priority than the client's callback thread: there is a short window within framesReady() 518// during which the normal mixer could be preempted, and the client callback would block. 519// Another problem can occur if framesReady() is called by the fast mixer: 520// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 521// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 522size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 523 return mAudioTrackServerProxy->framesReady(); 524} 525 526size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 527{ 528 return mAudioTrackServerProxy->framesReleased(); 529} 530 531// Don't call for fast tracks; the framesReady() could result in priority inversion 532bool AudioFlinger::PlaybackThread::Track::isReady() const { 533 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 534 return true; 535 } 536 537 if (framesReady() >= mFrameCount || 538 (mCblk->mFlags & CBLK_FORCEREADY)) { 539 mFillingUpStatus = FS_FILLED; 540 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 541 return true; 542 } 543 return false; 544} 545 546status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 547 int triggerSession) 548{ 549 status_t status = NO_ERROR; 550 ALOGV("start(%d), calling pid %d session %d", 551 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 552 553 sp<ThreadBase> thread = mThread.promote(); 554 if (thread != 0) { 555 //TODO: remove when effect offload is implemented 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->isGlobalEffectEnabled_l() || (ec != 0 && ec->isEnabled())) { 561 invalidate(); 562 return PERMISSION_DENIED; 563 } 564 } 565 Mutex::Autolock _lth(thread->mLock); 566 track_state state = mState; 567 // here the track could be either new, or restarted 568 // in both cases "unstop" the track 569 570 if (state == PAUSED) { 571 if (mResumeToStopping) { 572 // happened we need to resume to STOPPING_1 573 mState = TrackBase::STOPPING_1; 574 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 575 } else { 576 mState = TrackBase::RESUMING; 577 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 578 } 579 } else { 580 mState = TrackBase::ACTIVE; 581 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 582 } 583 584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 585 status = playbackThread->addTrack_l(this); 586 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 587 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 588 // restore previous state if start was rejected by policy manager 589 if (status == PERMISSION_DENIED) { 590 mState = state; 591 } 592 } 593 // track was already in the active list, not a problem 594 if (status == ALREADY_EXISTS) { 595 status = NO_ERROR; 596 } 597 } else { 598 status = BAD_VALUE; 599 } 600 return status; 601} 602 603void AudioFlinger::PlaybackThread::Track::stop() 604{ 605 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 606 sp<ThreadBase> thread = mThread.promote(); 607 if (thread != 0) { 608 Mutex::Autolock _l(thread->mLock); 609 track_state state = mState; 610 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 611 // If the track is not active (PAUSED and buffers full), flush buffers 612 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 613 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 614 reset(); 615 mState = STOPPED; 616 } else if (!isFastTrack() && !isOffloaded()) { 617 mState = STOPPED; 618 } else { 619 // For fast tracks prepareTracks_l() will set state to STOPPING_2 620 // presentation is complete 621 // For an offloaded track this starts a drain and state will 622 // move to STOPPING_2 when drain completes and then STOPPED 623 mState = STOPPING_1; 624 } 625 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 626 playbackThread); 627 } 628 } 629} 630 631void AudioFlinger::PlaybackThread::Track::pause() 632{ 633 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 634 sp<ThreadBase> thread = mThread.promote(); 635 if (thread != 0) { 636 Mutex::Autolock _l(thread->mLock); 637 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 638 switch (mState) { 639 case STOPPING_1: 640 case STOPPING_2: 641 if (!isOffloaded()) { 642 /* nothing to do if track is not offloaded */ 643 break; 644 } 645 646 // Offloaded track was draining, we need to carry on draining when resumed 647 mResumeToStopping = true; 648 // fall through... 649 case ACTIVE: 650 case RESUMING: 651 mState = PAUSING; 652 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 653 playbackThread->signal_l(); 654 break; 655 656 default: 657 break; 658 } 659 } 660} 661 662void AudioFlinger::PlaybackThread::Track::flush() 663{ 664 ALOGV("flush(%d)", mName); 665 sp<ThreadBase> thread = mThread.promote(); 666 if (thread != 0) { 667 Mutex::Autolock _l(thread->mLock); 668 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 669 670 if (isOffloaded()) { 671 // If offloaded we allow flush during any state except terminated 672 // and keep the track active to avoid problems if user is seeking 673 // rapidly and underlying hardware has a significant delay handling 674 // a pause 675 if (isTerminated()) { 676 return; 677 } 678 679 ALOGV("flush: offload flush"); 680 reset(); 681 682 if (mState == STOPPING_1 || mState == STOPPING_2) { 683 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 684 mState = ACTIVE; 685 } 686 687 if (mState == ACTIVE) { 688 ALOGV("flush called in active state, resetting buffer time out retry count"); 689 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 690 } 691 692 mResumeToStopping = false; 693 } else { 694 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 695 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 696 return; 697 } 698 // No point remaining in PAUSED state after a flush => go to 699 // FLUSHED state 700 mState = FLUSHED; 701 // do not reset the track if it is still in the process of being stopped or paused. 702 // this will be done by prepareTracks_l() when the track is stopped. 703 // prepareTracks_l() will see mState == FLUSHED, then 704 // remove from active track list, reset(), and trigger presentation complete 705 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 706 reset(); 707 } 708 } 709 // Prevent flush being lost if the track is flushed and then resumed 710 // before mixer thread can run. This is important when offloading 711 // because the hardware buffer could hold a large amount of audio 712 playbackThread->flushOutput_l(); 713 playbackThread->signal_l(); 714 } 715} 716 717void AudioFlinger::PlaybackThread::Track::reset() 718{ 719 // Do not reset twice to avoid discarding data written just after a flush and before 720 // the audioflinger thread detects the track is stopped. 721 if (!mResetDone) { 722 // Force underrun condition to avoid false underrun callback until first data is 723 // written to buffer 724 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 725 mFillingUpStatus = FS_FILLING; 726 mResetDone = true; 727 if (mState == FLUSHED) { 728 mState = IDLE; 729 } 730 } 731} 732 733status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 734{ 735 sp<ThreadBase> thread = mThread.promote(); 736 if (thread == 0) { 737 ALOGE("thread is dead"); 738 return FAILED_TRANSACTION; 739 } else if ((thread->type() == ThreadBase::DIRECT) || 740 (thread->type() == ThreadBase::OFFLOAD)) { 741 return thread->setParameters(keyValuePairs); 742 } else { 743 return PERMISSION_DENIED; 744 } 745} 746 747status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 748{ 749 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 750 if (isFastTrack()) { 751 return INVALID_OPERATION; 752 } 753 sp<ThreadBase> thread = mThread.promote(); 754 if (thread == 0) { 755 return INVALID_OPERATION; 756 } 757 Mutex::Autolock _l(thread->mLock); 758 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 759 if (!playbackThread->mLatchQValid) { 760 return INVALID_OPERATION; 761 } 762 uint32_t unpresentedFrames = 763 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 764 playbackThread->mSampleRate; 765 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 766 if (framesWritten < unpresentedFrames) { 767 return INVALID_OPERATION; 768 } 769 timestamp.mPosition = framesWritten - unpresentedFrames; 770 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 771 return NO_ERROR; 772} 773 774status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 775{ 776 status_t status = DEAD_OBJECT; 777 sp<ThreadBase> thread = mThread.promote(); 778 if (thread != 0) { 779 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 780 sp<AudioFlinger> af = mClient->audioFlinger(); 781 782 Mutex::Autolock _l(af->mLock); 783 784 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 785 786 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 787 Mutex::Autolock _dl(playbackThread->mLock); 788 Mutex::Autolock _sl(srcThread->mLock); 789 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 790 if (chain == 0) { 791 return INVALID_OPERATION; 792 } 793 794 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 795 if (effect == 0) { 796 return INVALID_OPERATION; 797 } 798 srcThread->removeEffect_l(effect); 799 playbackThread->addEffect_l(effect); 800 // removeEffect_l() has stopped the effect if it was active so it must be restarted 801 if (effect->state() == EffectModule::ACTIVE || 802 effect->state() == EffectModule::STOPPING) { 803 effect->start(); 804 } 805 806 sp<EffectChain> dstChain = effect->chain().promote(); 807 if (dstChain == 0) { 808 srcThread->addEffect_l(effect); 809 return INVALID_OPERATION; 810 } 811 AudioSystem::unregisterEffect(effect->id()); 812 AudioSystem::registerEffect(&effect->desc(), 813 srcThread->id(), 814 dstChain->strategy(), 815 AUDIO_SESSION_OUTPUT_MIX, 816 effect->id()); 817 } 818 status = playbackThread->attachAuxEffect(this, EffectId); 819 } 820 return status; 821} 822 823void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 824{ 825 mAuxEffectId = EffectId; 826 mAuxBuffer = buffer; 827} 828 829bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 830 size_t audioHalFrames) 831{ 832 // a track is considered presented when the total number of frames written to audio HAL 833 // corresponds to the number of frames written when presentationComplete() is called for the 834 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 835 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 836 // to detect when all frames have been played. In this case framesWritten isn't 837 // useful because it doesn't always reflect whether there is data in the h/w 838 // buffers, particularly if a track has been paused and resumed during draining 839 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 840 mPresentationCompleteFrames, framesWritten); 841 if (mPresentationCompleteFrames == 0) { 842 mPresentationCompleteFrames = framesWritten + audioHalFrames; 843 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 844 mPresentationCompleteFrames, audioHalFrames); 845 } 846 847 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 848 ALOGV("presentationComplete() session %d complete: framesWritten %d", 849 mSessionId, framesWritten); 850 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 851 mAudioTrackServerProxy->setStreamEndDone(); 852 return true; 853 } 854 return false; 855} 856 857void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 858{ 859 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 860 if (mSyncEvents[i]->type() == type) { 861 mSyncEvents[i]->trigger(); 862 mSyncEvents.removeAt(i); 863 i--; 864 } 865 } 866} 867 868// implement VolumeBufferProvider interface 869 870uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 871{ 872 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 873 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 874 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 875 uint32_t vl = vlr & 0xFFFF; 876 uint32_t vr = vlr >> 16; 877 // track volumes come from shared memory, so can't be trusted and must be clamped 878 if (vl > MAX_GAIN_INT) { 879 vl = MAX_GAIN_INT; 880 } 881 if (vr > MAX_GAIN_INT) { 882 vr = MAX_GAIN_INT; 883 } 884 // now apply the cached master volume and stream type volume; 885 // this is trusted but lacks any synchronization or barrier so may be stale 886 float v = mCachedVolume; 887 vl *= v; 888 vr *= v; 889 // re-combine into U4.16 890 vlr = (vr << 16) | (vl & 0xFFFF); 891 // FIXME look at mute, pause, and stop flags 892 return vlr; 893} 894 895status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 896{ 897 if (isTerminated() || mState == PAUSED || 898 ((framesReady() == 0) && ((mSharedBuffer != 0) || 899 (mState == STOPPED)))) { 900 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 901 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 902 event->cancel(); 903 return INVALID_OPERATION; 904 } 905 (void) TrackBase::setSyncEvent(event); 906 return NO_ERROR; 907} 908 909void AudioFlinger::PlaybackThread::Track::invalidate() 910{ 911 // FIXME should use proxy, and needs work 912 audio_track_cblk_t* cblk = mCblk; 913 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 914 android_atomic_release_store(0x40000000, &cblk->mFutex); 915 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 916 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 917 mIsInvalid = true; 918} 919 920// ---------------------------------------------------------------------------- 921 922sp<AudioFlinger::PlaybackThread::TimedTrack> 923AudioFlinger::PlaybackThread::TimedTrack::create( 924 PlaybackThread *thread, 925 const sp<Client>& client, 926 audio_stream_type_t streamType, 927 uint32_t sampleRate, 928 audio_format_t format, 929 audio_channel_mask_t channelMask, 930 size_t frameCount, 931 const sp<IMemory>& sharedBuffer, 932 int sessionId) { 933 if (!client->reserveTimedTrack()) 934 return 0; 935 936 return new TimedTrack( 937 thread, client, streamType, sampleRate, format, channelMask, frameCount, 938 sharedBuffer, sessionId); 939} 940 941AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 942 PlaybackThread *thread, 943 const sp<Client>& client, 944 audio_stream_type_t streamType, 945 uint32_t sampleRate, 946 audio_format_t format, 947 audio_channel_mask_t channelMask, 948 size_t frameCount, 949 const sp<IMemory>& sharedBuffer, 950 int sessionId) 951 : Track(thread, client, streamType, sampleRate, format, channelMask, 952 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 953 mQueueHeadInFlight(false), 954 mTrimQueueHeadOnRelease(false), 955 mFramesPendingInQueue(0), 956 mTimedSilenceBuffer(NULL), 957 mTimedSilenceBufferSize(0), 958 mTimedAudioOutputOnTime(false), 959 mMediaTimeTransformValid(false) 960{ 961 LocalClock lc; 962 mLocalTimeFreq = lc.getLocalFreq(); 963 964 mLocalTimeToSampleTransform.a_zero = 0; 965 mLocalTimeToSampleTransform.b_zero = 0; 966 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 967 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 968 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 969 &mLocalTimeToSampleTransform.a_to_b_denom); 970 971 mMediaTimeToSampleTransform.a_zero = 0; 972 mMediaTimeToSampleTransform.b_zero = 0; 973 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 974 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 975 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 976 &mMediaTimeToSampleTransform.a_to_b_denom); 977} 978 979AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 980 mClient->releaseTimedTrack(); 981 delete [] mTimedSilenceBuffer; 982} 983 984status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 985 size_t size, sp<IMemory>* buffer) { 986 987 Mutex::Autolock _l(mTimedBufferQueueLock); 988 989 trimTimedBufferQueue_l(); 990 991 // lazily initialize the shared memory heap for timed buffers 992 if (mTimedMemoryDealer == NULL) { 993 const int kTimedBufferHeapSize = 512 << 10; 994 995 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 996 "AudioFlingerTimed"); 997 if (mTimedMemoryDealer == NULL) { 998 return NO_MEMORY; 999 } 1000 } 1001 1002 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1003 if (newBuffer == NULL) { 1004 newBuffer = mTimedMemoryDealer->allocate(size); 1005 if (newBuffer == NULL) { 1006 return NO_MEMORY; 1007 } 1008 } 1009 1010 *buffer = newBuffer; 1011 return NO_ERROR; 1012} 1013 1014// caller must hold mTimedBufferQueueLock 1015void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1016 int64_t mediaTimeNow; 1017 { 1018 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1019 if (!mMediaTimeTransformValid) 1020 return; 1021 1022 int64_t targetTimeNow; 1023 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1024 ? mCCHelper.getCommonTime(&targetTimeNow) 1025 : mCCHelper.getLocalTime(&targetTimeNow); 1026 1027 if (OK != res) 1028 return; 1029 1030 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1031 &mediaTimeNow)) { 1032 return; 1033 } 1034 } 1035 1036 size_t trimEnd; 1037 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1038 int64_t bufEnd; 1039 1040 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1041 // We have a next buffer. Just use its PTS as the PTS of the frame 1042 // following the last frame in this buffer. If the stream is sparse 1043 // (ie, there are deliberate gaps left in the stream which should be 1044 // filled with silence by the TimedAudioTrack), then this can result 1045 // in one extra buffer being left un-trimmed when it could have 1046 // been. In general, this is not typical, and we would rather 1047 // optimized away the TS calculation below for the more common case 1048 // where PTSes are contiguous. 1049 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1050 } else { 1051 // We have no next buffer. Compute the PTS of the frame following 1052 // the last frame in this buffer by computing the duration of of 1053 // this frame in media time units and adding it to the PTS of the 1054 // buffer. 1055 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1056 / mFrameSize; 1057 1058 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1059 &bufEnd)) { 1060 ALOGE("Failed to convert frame count of %lld to media time" 1061 " duration" " (scale factor %d/%u) in %s", 1062 frameCount, 1063 mMediaTimeToSampleTransform.a_to_b_numer, 1064 mMediaTimeToSampleTransform.a_to_b_denom, 1065 __PRETTY_FUNCTION__); 1066 break; 1067 } 1068 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1069 } 1070 1071 if (bufEnd > mediaTimeNow) 1072 break; 1073 1074 // Is the buffer we want to use in the middle of a mix operation right 1075 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1076 // from the mixer which should be coming back shortly. 1077 if (!trimEnd && mQueueHeadInFlight) { 1078 mTrimQueueHeadOnRelease = true; 1079 } 1080 } 1081 1082 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1083 if (trimStart < trimEnd) { 1084 // Update the bookkeeping for framesReady() 1085 for (size_t i = trimStart; i < trimEnd; ++i) { 1086 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1087 } 1088 1089 // Now actually remove the buffers from the queue. 1090 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1091 } 1092} 1093 1094void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1095 const char* logTag) { 1096 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1097 "%s called (reason \"%s\"), but timed buffer queue has no" 1098 " elements to trim.", __FUNCTION__, logTag); 1099 1100 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1101 mTimedBufferQueue.removeAt(0); 1102} 1103 1104void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1105 const TimedBuffer& buf, 1106 const char* logTag) { 1107 uint32_t bufBytes = buf.buffer()->size(); 1108 uint32_t consumedAlready = buf.position(); 1109 1110 ALOG_ASSERT(consumedAlready <= bufBytes, 1111 "Bad bookkeeping while updating frames pending. Timed buffer is" 1112 " only %u bytes long, but claims to have consumed %u" 1113 " bytes. (update reason: \"%s\")", 1114 bufBytes, consumedAlready, logTag); 1115 1116 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1117 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1118 "Bad bookkeeping while updating frames pending. Should have at" 1119 " least %u queued frames, but we think we have only %u. (update" 1120 " reason: \"%s\")", 1121 bufFrames, mFramesPendingInQueue, logTag); 1122 1123 mFramesPendingInQueue -= bufFrames; 1124} 1125 1126status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1127 const sp<IMemory>& buffer, int64_t pts) { 1128 1129 { 1130 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1131 if (!mMediaTimeTransformValid) 1132 return INVALID_OPERATION; 1133 } 1134 1135 Mutex::Autolock _l(mTimedBufferQueueLock); 1136 1137 uint32_t bufFrames = buffer->size() / mFrameSize; 1138 mFramesPendingInQueue += bufFrames; 1139 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1140 1141 return NO_ERROR; 1142} 1143 1144status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1145 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1146 1147 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1148 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1149 target); 1150 1151 if (!(target == TimedAudioTrack::LOCAL_TIME || 1152 target == TimedAudioTrack::COMMON_TIME)) { 1153 return BAD_VALUE; 1154 } 1155 1156 Mutex::Autolock lock(mMediaTimeTransformLock); 1157 mMediaTimeTransform = xform; 1158 mMediaTimeTransformTarget = target; 1159 mMediaTimeTransformValid = true; 1160 1161 return NO_ERROR; 1162} 1163 1164#define min(a, b) ((a) < (b) ? (a) : (b)) 1165 1166// implementation of getNextBuffer for tracks whose buffers have timestamps 1167status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1168 AudioBufferProvider::Buffer* buffer, int64_t pts) 1169{ 1170 if (pts == AudioBufferProvider::kInvalidPTS) { 1171 buffer->raw = NULL; 1172 buffer->frameCount = 0; 1173 mTimedAudioOutputOnTime = false; 1174 return INVALID_OPERATION; 1175 } 1176 1177 Mutex::Autolock _l(mTimedBufferQueueLock); 1178 1179 ALOG_ASSERT(!mQueueHeadInFlight, 1180 "getNextBuffer called without releaseBuffer!"); 1181 1182 while (true) { 1183 1184 // if we have no timed buffers, then fail 1185 if (mTimedBufferQueue.isEmpty()) { 1186 buffer->raw = NULL; 1187 buffer->frameCount = 0; 1188 return NOT_ENOUGH_DATA; 1189 } 1190 1191 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1192 1193 // calculate the PTS of the head of the timed buffer queue expressed in 1194 // local time 1195 int64_t headLocalPTS; 1196 { 1197 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1198 1199 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1200 1201 if (mMediaTimeTransform.a_to_b_denom == 0) { 1202 // the transform represents a pause, so yield silence 1203 timedYieldSilence_l(buffer->frameCount, buffer); 1204 return NO_ERROR; 1205 } 1206 1207 int64_t transformedPTS; 1208 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1209 &transformedPTS)) { 1210 // the transform failed. this shouldn't happen, but if it does 1211 // then just drop this buffer 1212 ALOGW("timedGetNextBuffer transform failed"); 1213 buffer->raw = NULL; 1214 buffer->frameCount = 0; 1215 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1216 return NO_ERROR; 1217 } 1218 1219 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1220 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1221 &headLocalPTS)) { 1222 buffer->raw = NULL; 1223 buffer->frameCount = 0; 1224 return INVALID_OPERATION; 1225 } 1226 } else { 1227 headLocalPTS = transformedPTS; 1228 } 1229 } 1230 1231 uint32_t sr = sampleRate(); 1232 1233 // adjust the head buffer's PTS to reflect the portion of the head buffer 1234 // that has already been consumed 1235 int64_t effectivePTS = headLocalPTS + 1236 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1237 1238 // Calculate the delta in samples between the head of the input buffer 1239 // queue and the start of the next output buffer that will be written. 1240 // If the transformation fails because of over or underflow, it means 1241 // that the sample's position in the output stream is so far out of 1242 // whack that it should just be dropped. 1243 int64_t sampleDelta; 1244 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1245 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1246 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1247 " mix"); 1248 continue; 1249 } 1250 if (!mLocalTimeToSampleTransform.doForwardTransform( 1251 (effectivePTS - pts) << 32, &sampleDelta)) { 1252 ALOGV("*** too late during sample rate transform: dropped buffer"); 1253 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1254 continue; 1255 } 1256 1257 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1258 " sampleDelta=[%d.%08x]", 1259 head.pts(), head.position(), pts, 1260 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1261 + (sampleDelta >> 32)), 1262 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1263 1264 // if the delta between the ideal placement for the next input sample and 1265 // the current output position is within this threshold, then we will 1266 // concatenate the next input samples to the previous output 1267 const int64_t kSampleContinuityThreshold = 1268 (static_cast<int64_t>(sr) << 32) / 250; 1269 1270 // if this is the first buffer of audio that we're emitting from this track 1271 // then it should be almost exactly on time. 1272 const int64_t kSampleStartupThreshold = 1LL << 32; 1273 1274 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1275 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1276 // the next input is close enough to being on time, so concatenate it 1277 // with the last output 1278 timedYieldSamples_l(buffer); 1279 1280 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1281 head.position(), buffer->frameCount); 1282 return NO_ERROR; 1283 } 1284 1285 // Looks like our output is not on time. Reset our on timed status. 1286 // Next time we mix samples from our input queue, then should be within 1287 // the StartupThreshold. 1288 mTimedAudioOutputOnTime = false; 1289 if (sampleDelta > 0) { 1290 // the gap between the current output position and the proper start of 1291 // the next input sample is too big, so fill it with silence 1292 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1293 1294 timedYieldSilence_l(framesUntilNextInput, buffer); 1295 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1296 return NO_ERROR; 1297 } else { 1298 // the next input sample is late 1299 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1300 size_t onTimeSamplePosition = 1301 head.position() + lateFrames * mFrameSize; 1302 1303 if (onTimeSamplePosition > head.buffer()->size()) { 1304 // all the remaining samples in the head are too late, so 1305 // drop it and move on 1306 ALOGV("*** too late: dropped buffer"); 1307 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1308 continue; 1309 } else { 1310 // skip over the late samples 1311 head.setPosition(onTimeSamplePosition); 1312 1313 // yield the available samples 1314 timedYieldSamples_l(buffer); 1315 1316 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1317 return NO_ERROR; 1318 } 1319 } 1320 } 1321} 1322 1323// Yield samples from the timed buffer queue head up to the given output 1324// buffer's capacity. 1325// 1326// Caller must hold mTimedBufferQueueLock 1327void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1328 AudioBufferProvider::Buffer* buffer) { 1329 1330 const TimedBuffer& head = mTimedBufferQueue[0]; 1331 1332 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1333 head.position()); 1334 1335 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1336 mFrameSize); 1337 size_t framesRequested = buffer->frameCount; 1338 buffer->frameCount = min(framesLeftInHead, framesRequested); 1339 1340 mQueueHeadInFlight = true; 1341 mTimedAudioOutputOnTime = true; 1342} 1343 1344// Yield samples of silence up to the given output buffer's capacity 1345// 1346// Caller must hold mTimedBufferQueueLock 1347void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1348 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1349 1350 // lazily allocate a buffer filled with silence 1351 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1352 delete [] mTimedSilenceBuffer; 1353 mTimedSilenceBufferSize = numFrames * mFrameSize; 1354 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1355 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1356 } 1357 1358 buffer->raw = mTimedSilenceBuffer; 1359 size_t framesRequested = buffer->frameCount; 1360 buffer->frameCount = min(numFrames, framesRequested); 1361 1362 mTimedAudioOutputOnTime = false; 1363} 1364 1365// AudioBufferProvider interface 1366void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1367 AudioBufferProvider::Buffer* buffer) { 1368 1369 Mutex::Autolock _l(mTimedBufferQueueLock); 1370 1371 // If the buffer which was just released is part of the buffer at the head 1372 // of the queue, be sure to update the amt of the buffer which has been 1373 // consumed. If the buffer being returned is not part of the head of the 1374 // queue, its either because the buffer is part of the silence buffer, or 1375 // because the head of the timed queue was trimmed after the mixer called 1376 // getNextBuffer but before the mixer called releaseBuffer. 1377 if (buffer->raw == mTimedSilenceBuffer) { 1378 ALOG_ASSERT(!mQueueHeadInFlight, 1379 "Queue head in flight during release of silence buffer!"); 1380 goto done; 1381 } 1382 1383 ALOG_ASSERT(mQueueHeadInFlight, 1384 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1385 " head in flight."); 1386 1387 if (mTimedBufferQueue.size()) { 1388 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1389 1390 void* start = head.buffer()->pointer(); 1391 void* end = reinterpret_cast<void*>( 1392 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1393 + head.buffer()->size()); 1394 1395 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1396 "released buffer not within the head of the timed buffer" 1397 " queue; qHead = [%p, %p], released buffer = %p", 1398 start, end, buffer->raw); 1399 1400 head.setPosition(head.position() + 1401 (buffer->frameCount * mFrameSize)); 1402 mQueueHeadInFlight = false; 1403 1404 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1405 "Bad bookkeeping during releaseBuffer! Should have at" 1406 " least %u queued frames, but we think we have only %u", 1407 buffer->frameCount, mFramesPendingInQueue); 1408 1409 mFramesPendingInQueue -= buffer->frameCount; 1410 1411 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1412 || mTrimQueueHeadOnRelease) { 1413 trimTimedBufferQueueHead_l("releaseBuffer"); 1414 mTrimQueueHeadOnRelease = false; 1415 } 1416 } else { 1417 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1418 " buffers in the timed buffer queue"); 1419 } 1420 1421done: 1422 buffer->raw = 0; 1423 buffer->frameCount = 0; 1424} 1425 1426size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1427 Mutex::Autolock _l(mTimedBufferQueueLock); 1428 return mFramesPendingInQueue; 1429} 1430 1431AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1432 : mPTS(0), mPosition(0) {} 1433 1434AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1435 const sp<IMemory>& buffer, int64_t pts) 1436 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1437 1438 1439// ---------------------------------------------------------------------------- 1440 1441AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1442 PlaybackThread *playbackThread, 1443 DuplicatingThread *sourceThread, 1444 uint32_t sampleRate, 1445 audio_format_t format, 1446 audio_channel_mask_t channelMask, 1447 size_t frameCount) 1448 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1449 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1450 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1451{ 1452 1453 if (mCblk != NULL) { 1454 mOutBuffer.frameCount = 0; 1455 playbackThread->mTracks.add(this); 1456 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1457 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1458 mCblk, mBuffer, 1459 mCblk->frameCount_, mChannelMask); 1460 // since client and server are in the same process, 1461 // the buffer has the same virtual address on both sides 1462 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1463 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1464 mClientProxy->setSendLevel(0.0); 1465 mClientProxy->setSampleRate(sampleRate); 1466 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1467 true /*clientInServer*/); 1468 } else { 1469 ALOGW("Error creating output track on thread %p", playbackThread); 1470 } 1471} 1472 1473AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1474{ 1475 clearBufferQueue(); 1476 delete mClientProxy; 1477 // superclass destructor will now delete the server proxy and shared memory both refer to 1478} 1479 1480status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1481 int triggerSession) 1482{ 1483 status_t status = Track::start(event, triggerSession); 1484 if (status != NO_ERROR) { 1485 return status; 1486 } 1487 1488 mActive = true; 1489 mRetryCount = 127; 1490 return status; 1491} 1492 1493void AudioFlinger::PlaybackThread::OutputTrack::stop() 1494{ 1495 Track::stop(); 1496 clearBufferQueue(); 1497 mOutBuffer.frameCount = 0; 1498 mActive = false; 1499} 1500 1501bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1502{ 1503 Buffer *pInBuffer; 1504 Buffer inBuffer; 1505 uint32_t channelCount = mChannelCount; 1506 bool outputBufferFull = false; 1507 inBuffer.frameCount = frames; 1508 inBuffer.i16 = data; 1509 1510 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1511 1512 if (!mActive && frames != 0) { 1513 start(); 1514 sp<ThreadBase> thread = mThread.promote(); 1515 if (thread != 0) { 1516 MixerThread *mixerThread = (MixerThread *)thread.get(); 1517 if (mFrameCount > frames) { 1518 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1519 uint32_t startFrames = (mFrameCount - frames); 1520 pInBuffer = new Buffer; 1521 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1522 pInBuffer->frameCount = startFrames; 1523 pInBuffer->i16 = pInBuffer->mBuffer; 1524 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1525 mBufferQueue.add(pInBuffer); 1526 } else { 1527 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1528 } 1529 } 1530 } 1531 } 1532 1533 while (waitTimeLeftMs) { 1534 // First write pending buffers, then new data 1535 if (mBufferQueue.size()) { 1536 pInBuffer = mBufferQueue.itemAt(0); 1537 } else { 1538 pInBuffer = &inBuffer; 1539 } 1540 1541 if (pInBuffer->frameCount == 0) { 1542 break; 1543 } 1544 1545 if (mOutBuffer.frameCount == 0) { 1546 mOutBuffer.frameCount = pInBuffer->frameCount; 1547 nsecs_t startTime = systemTime(); 1548 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1549 if (status != NO_ERROR) { 1550 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1551 mThread.unsafe_get(), status); 1552 outputBufferFull = true; 1553 break; 1554 } 1555 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1556 if (waitTimeLeftMs >= waitTimeMs) { 1557 waitTimeLeftMs -= waitTimeMs; 1558 } else { 1559 waitTimeLeftMs = 0; 1560 } 1561 } 1562 1563 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1564 pInBuffer->frameCount; 1565 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1566 Proxy::Buffer buf; 1567 buf.mFrameCount = outFrames; 1568 buf.mRaw = NULL; 1569 mClientProxy->releaseBuffer(&buf); 1570 pInBuffer->frameCount -= outFrames; 1571 pInBuffer->i16 += outFrames * channelCount; 1572 mOutBuffer.frameCount -= outFrames; 1573 mOutBuffer.i16 += outFrames * channelCount; 1574 1575 if (pInBuffer->frameCount == 0) { 1576 if (mBufferQueue.size()) { 1577 mBufferQueue.removeAt(0); 1578 delete [] pInBuffer->mBuffer; 1579 delete pInBuffer; 1580 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1581 mThread.unsafe_get(), mBufferQueue.size()); 1582 } else { 1583 break; 1584 } 1585 } 1586 } 1587 1588 // If we could not write all frames, allocate a buffer and queue it for next time. 1589 if (inBuffer.frameCount) { 1590 sp<ThreadBase> thread = mThread.promote(); 1591 if (thread != 0 && !thread->standby()) { 1592 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1593 pInBuffer = new Buffer; 1594 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1595 pInBuffer->frameCount = inBuffer.frameCount; 1596 pInBuffer->i16 = pInBuffer->mBuffer; 1597 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1598 sizeof(int16_t)); 1599 mBufferQueue.add(pInBuffer); 1600 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1601 mThread.unsafe_get(), mBufferQueue.size()); 1602 } else { 1603 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1604 mThread.unsafe_get(), this); 1605 } 1606 } 1607 } 1608 1609 // Calling write() with a 0 length buffer, means that no more data will be written: 1610 // If no more buffers are pending, fill output track buffer to make sure it is started 1611 // by output mixer. 1612 if (frames == 0 && mBufferQueue.size() == 0) { 1613 // FIXME borken, replace by getting framesReady() from proxy 1614 size_t user = 0; // was mCblk->user 1615 if (user < mFrameCount) { 1616 frames = mFrameCount - user; 1617 pInBuffer = new Buffer; 1618 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1619 pInBuffer->frameCount = frames; 1620 pInBuffer->i16 = pInBuffer->mBuffer; 1621 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1622 mBufferQueue.add(pInBuffer); 1623 } else if (mActive) { 1624 stop(); 1625 } 1626 } 1627 1628 return outputBufferFull; 1629} 1630 1631status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1632 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1633{ 1634 ClientProxy::Buffer buf; 1635 buf.mFrameCount = buffer->frameCount; 1636 struct timespec timeout; 1637 timeout.tv_sec = waitTimeMs / 1000; 1638 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1639 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1640 buffer->frameCount = buf.mFrameCount; 1641 buffer->raw = buf.mRaw; 1642 return status; 1643} 1644 1645void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1646{ 1647 size_t size = mBufferQueue.size(); 1648 1649 for (size_t i = 0; i < size; i++) { 1650 Buffer *pBuffer = mBufferQueue.itemAt(i); 1651 delete [] pBuffer->mBuffer; 1652 delete pBuffer; 1653 } 1654 mBufferQueue.clear(); 1655} 1656 1657 1658// ---------------------------------------------------------------------------- 1659// Record 1660// ---------------------------------------------------------------------------- 1661 1662AudioFlinger::RecordHandle::RecordHandle( 1663 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1664 : BnAudioRecord(), 1665 mRecordTrack(recordTrack) 1666{ 1667} 1668 1669AudioFlinger::RecordHandle::~RecordHandle() { 1670 stop_nonvirtual(); 1671 mRecordTrack->destroy(); 1672} 1673 1674sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1675 return mRecordTrack->getCblk(); 1676} 1677 1678status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1679 int triggerSession) { 1680 ALOGV("RecordHandle::start()"); 1681 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1682} 1683 1684void AudioFlinger::RecordHandle::stop() { 1685 stop_nonvirtual(); 1686} 1687 1688void AudioFlinger::RecordHandle::stop_nonvirtual() { 1689 ALOGV("RecordHandle::stop()"); 1690 mRecordTrack->stop(); 1691} 1692 1693status_t AudioFlinger::RecordHandle::onTransact( 1694 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1695{ 1696 return BnAudioRecord::onTransact(code, data, reply, flags); 1697} 1698 1699// ---------------------------------------------------------------------------- 1700 1701// RecordTrack constructor must be called with AudioFlinger::mLock held 1702AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1703 RecordThread *thread, 1704 const sp<Client>& client, 1705 uint32_t sampleRate, 1706 audio_format_t format, 1707 audio_channel_mask_t channelMask, 1708 size_t frameCount, 1709 int sessionId) 1710 : TrackBase(thread, client, sampleRate, format, 1711 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1712 mOverflow(false) 1713{ 1714 ALOGV("RecordTrack constructor"); 1715 if (mCblk != NULL) { 1716 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1717 } 1718} 1719 1720AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1721{ 1722 ALOGV("%s", __func__); 1723} 1724 1725// AudioBufferProvider interface 1726status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1727 int64_t pts) 1728{ 1729 ServerProxy::Buffer buf; 1730 buf.mFrameCount = buffer->frameCount; 1731 status_t status = mServerProxy->obtainBuffer(&buf); 1732 buffer->frameCount = buf.mFrameCount; 1733 buffer->raw = buf.mRaw; 1734 if (buf.mFrameCount == 0) { 1735 // FIXME also wake futex so that overrun is noticed more quickly 1736 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1737 } 1738 return status; 1739} 1740 1741status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1742 int triggerSession) 1743{ 1744 sp<ThreadBase> thread = mThread.promote(); 1745 if (thread != 0) { 1746 RecordThread *recordThread = (RecordThread *)thread.get(); 1747 return recordThread->start(this, event, triggerSession); 1748 } else { 1749 return BAD_VALUE; 1750 } 1751} 1752 1753void AudioFlinger::RecordThread::RecordTrack::stop() 1754{ 1755 sp<ThreadBase> thread = mThread.promote(); 1756 if (thread != 0) { 1757 RecordThread *recordThread = (RecordThread *)thread.get(); 1758 if (recordThread->stop(this)) { 1759 AudioSystem::stopInput(recordThread->id()); 1760 } 1761 } 1762} 1763 1764void AudioFlinger::RecordThread::RecordTrack::destroy() 1765{ 1766 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1767 sp<RecordTrack> keep(this); 1768 { 1769 sp<ThreadBase> thread = mThread.promote(); 1770 if (thread != 0) { 1771 if (mState == ACTIVE || mState == RESUMING) { 1772 AudioSystem::stopInput(thread->id()); 1773 } 1774 AudioSystem::releaseInput(thread->id()); 1775 Mutex::Autolock _l(thread->mLock); 1776 RecordThread *recordThread = (RecordThread *) thread.get(); 1777 recordThread->destroyTrack_l(this); 1778 } 1779 } 1780} 1781 1782void AudioFlinger::RecordThread::RecordTrack::invalidate() 1783{ 1784 // FIXME should use proxy, and needs work 1785 audio_track_cblk_t* cblk = mCblk; 1786 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1787 android_atomic_release_store(0x40000000, &cblk->mFutex); 1788 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1789 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1790} 1791 1792 1793/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1794{ 1795 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1796} 1797 1798void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1799{ 1800 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1801 (mClient == 0) ? getpid_cached : mClient->pid(), 1802 mFormat, 1803 mChannelMask, 1804 mSessionId, 1805 mState, 1806 mCblk->mServer, 1807 mFrameCount); 1808} 1809 1810}; // namespace android 1811