Tracks.cpp revision bfb1b832079bbb9426f72f3863199a54aefd02da
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 <cutils/compiler.h> 25#include <utils/Log.h> 26 27#include <private/media/AudioTrackShared.h> 28 29#include <common_time/cc_helper.h> 30#include <common_time/local_clock.h> 31 32#include "AudioMixer.h" 33#include "AudioFlinger.h" 34#include "ServiceUtilities.h" 35 36#include <media/nbaio/Pipe.h> 37#include <media/nbaio/PipeReader.h> 38 39// ---------------------------------------------------------------------------- 40 41// Note: the following macro is used for extremely verbose logging message. In 42// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 43// 0; but one side effect of this is to turn all LOGV's as well. Some messages 44// are so verbose that we want to suppress them even when we have ALOG_ASSERT 45// turned on. Do not uncomment the #def below unless you really know what you 46// are doing and want to see all of the extremely verbose messages. 47//#define VERY_VERY_VERBOSE_LOGGING 48#ifdef VERY_VERY_VERBOSE_LOGGING 49#define ALOGVV ALOGV 50#else 51#define ALOGVV(a...) do { } while(0) 52#endif 53 54namespace android { 55 56// ---------------------------------------------------------------------------- 57// TrackBase 58// ---------------------------------------------------------------------------- 59 60static volatile int32_t nextTrackId = 55; 61 62// TrackBase constructor must be called with AudioFlinger::mLock held 63AudioFlinger::ThreadBase::TrackBase::TrackBase( 64 ThreadBase *thread, 65 const sp<Client>& client, 66 uint32_t sampleRate, 67 audio_format_t format, 68 audio_channel_mask_t channelMask, 69 size_t frameCount, 70 const sp<IMemory>& sharedBuffer, 71 int sessionId, 72 bool isOut) 73 : RefBase(), 74 mThread(thread), 75 mClient(client), 76 mCblk(NULL), 77 // mBuffer 78 // mBufferEnd 79 mStepCount(0), 80 mState(IDLE), 81 mSampleRate(sampleRate), 82 mFormat(format), 83 mChannelMask(channelMask), 84 mChannelCount(popcount(channelMask)), 85 mFrameSize(audio_is_linear_pcm(format) ? 86 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 87 mFrameCount(frameCount), 88 mStepServerFailed(false), 89 mSessionId(sessionId), 90 mIsOut(isOut), 91 mServerProxy(NULL), 92 mId(android_atomic_inc(&nextTrackId)), 93 mTerminated(false) 94{ 95 // client == 0 implies sharedBuffer == 0 96 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 97 98 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 99 sharedBuffer->size()); 100 101 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 102 size_t size = sizeof(audio_track_cblk_t); 103 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 104 if (sharedBuffer == 0) { 105 size += bufferSize; 106 } 107 108 if (client != 0) { 109 mCblkMemory = client->heap()->allocate(size); 110 if (mCblkMemory != 0) { 111 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 112 // can't assume mCblk != NULL 113 } else { 114 ALOGE("not enough memory for AudioTrack size=%u", size); 115 client->heap()->dump("AudioTrack"); 116 return; 117 } 118 } else { 119 // this syntax avoids calling the audio_track_cblk_t constructor twice 120 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 121 // assume mCblk != NULL 122 } 123 124 // construct the shared structure in-place. 125 if (mCblk != NULL) { 126 new(mCblk) audio_track_cblk_t(); 127 // clear all buffers 128 mCblk->frameCount_ = frameCount; 129 if (sharedBuffer == 0) { 130 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 131 memset(mBuffer, 0, bufferSize); 132 } else { 133 mBuffer = sharedBuffer->pointer(); 134#if 0 135 mCblk->flags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 136#endif 137 } 138 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 139 140#ifdef TEE_SINK 141 if (mTeeSinkTrackEnabled) { 142 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 143 if (pipeFormat != Format_Invalid) { 144 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 145 size_t numCounterOffers = 0; 146 const NBAIO_Format offers[1] = {pipeFormat}; 147 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 148 ALOG_ASSERT(index == 0); 149 PipeReader *pipeReader = new PipeReader(*pipe); 150 numCounterOffers = 0; 151 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 152 ALOG_ASSERT(index == 0); 153 mTeeSink = pipe; 154 mTeeSource = pipeReader; 155 } 156 } 157#endif 158 159 } 160} 161 162AudioFlinger::ThreadBase::TrackBase::~TrackBase() 163{ 164#ifdef TEE_SINK 165 dumpTee(-1, mTeeSource, mId); 166#endif 167 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 168 delete mServerProxy; 169 if (mCblk != NULL) { 170 if (mClient == 0) { 171 delete mCblk; 172 } else { 173 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 174 } 175 } 176 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 177 if (mClient != 0) { 178 // Client destructor must run with AudioFlinger mutex locked 179 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 180 // If the client's reference count drops to zero, the associated destructor 181 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 182 // relying on the automatic clear() at end of scope. 183 mClient.clear(); 184 } 185} 186 187// AudioBufferProvider interface 188// getNextBuffer() = 0; 189// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 190void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 191{ 192#ifdef TEE_SINK 193 if (mTeeSink != 0) { 194 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 195 } 196#endif 197 198 ServerProxy::Buffer buf; 199 buf.mFrameCount = buffer->frameCount; 200 buf.mRaw = buffer->raw; 201 buffer->frameCount = 0; 202 buffer->raw = NULL; 203 mServerProxy->releaseBuffer(&buf); 204} 205 206void AudioFlinger::ThreadBase::TrackBase::reset() { 207 ALOGV("TrackBase::reset"); 208 // FIXME still needed? 209} 210 211status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 212{ 213 mSyncEvents.add(event); 214 return NO_ERROR; 215} 216 217// ---------------------------------------------------------------------------- 218// Playback 219// ---------------------------------------------------------------------------- 220 221AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 222 : BnAudioTrack(), 223 mTrack(track) 224{ 225} 226 227AudioFlinger::TrackHandle::~TrackHandle() { 228 // just stop the track on deletion, associated resources 229 // will be freed from the main thread once all pending buffers have 230 // been played. Unless it's not in the active track list, in which 231 // case we free everything now... 232 mTrack->destroy(); 233} 234 235sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 236 return mTrack->getCblk(); 237} 238 239status_t AudioFlinger::TrackHandle::start() { 240 return mTrack->start(); 241} 242 243void AudioFlinger::TrackHandle::stop() { 244 mTrack->stop(); 245} 246 247void AudioFlinger::TrackHandle::flush() { 248 mTrack->flush(); 249} 250 251void AudioFlinger::TrackHandle::pause() { 252 mTrack->pause(); 253} 254 255status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 256 return mTrack->setParameters(keyValuePairs); 257} 258 259status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 260{ 261 return mTrack->attachAuxEffect(EffectId); 262} 263 264status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 265 sp<IMemory>* buffer) { 266 if (!mTrack->isTimedTrack()) 267 return INVALID_OPERATION; 268 269 PlaybackThread::TimedTrack* tt = 270 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 271 return tt->allocateTimedBuffer(size, buffer); 272} 273 274status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 275 int64_t pts) { 276 if (!mTrack->isTimedTrack()) 277 return INVALID_OPERATION; 278 279 PlaybackThread::TimedTrack* tt = 280 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 281 return tt->queueTimedBuffer(buffer, pts); 282} 283 284status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 285 const LinearTransform& xform, int target) { 286 287 if (!mTrack->isTimedTrack()) 288 return INVALID_OPERATION; 289 290 PlaybackThread::TimedTrack* tt = 291 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 292 return tt->setMediaTimeTransform( 293 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 294} 295 296status_t AudioFlinger::TrackHandle::onTransact( 297 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 298{ 299 return BnAudioTrack::onTransact(code, data, reply, flags); 300} 301 302// ---------------------------------------------------------------------------- 303 304// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 305AudioFlinger::PlaybackThread::Track::Track( 306 PlaybackThread *thread, 307 const sp<Client>& client, 308 audio_stream_type_t streamType, 309 uint32_t sampleRate, 310 audio_format_t format, 311 audio_channel_mask_t channelMask, 312 size_t frameCount, 313 const sp<IMemory>& sharedBuffer, 314 int sessionId, 315 IAudioFlinger::track_flags_t flags) 316 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 317 sessionId, true /*isOut*/), 318 mFillingUpStatus(FS_INVALID), 319 // mRetryCount initialized later when needed 320 mSharedBuffer(sharedBuffer), 321 mStreamType(streamType), 322 mName(-1), // see note below 323 mMainBuffer(thread->mixBuffer()), 324 mAuxBuffer(NULL), 325 mAuxEffectId(0), mHasVolumeController(false), 326 mPresentationCompleteFrames(0), 327 mFlags(flags), 328 mFastIndex(-1), 329 mUnderrunCount(0), 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 mCblk->mName = mName; 347 if (mName < 0) { 348 ALOGE("no more track names available"); 349 return; 350 } 351 // only allocate a fast track index if we were able to allocate a normal track name 352 if (flags & IAudioFlinger::TRACK_FAST) { 353 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 354 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 355 int i = __builtin_ctz(thread->mFastTrackAvailMask); 356 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 357 // FIXME This is too eager. We allocate a fast track index before the 358 // fast track becomes active. Since fast tracks are a scarce resource, 359 // this means we are potentially denying other more important fast tracks from 360 // being created. It would be better to allocate the index dynamically. 361 mFastIndex = i; 362 mCblk->mName = i; 363 // Read the initial underruns because this field is never cleared by the fast mixer 364 mObservedUnderruns = thread->getFastTrackUnderruns(i); 365 thread->mFastTrackAvailMask &= ~(1 << i); 366 } 367 } 368 ALOGV("Track constructor name %d, calling pid %d", mName, 369 IPCThreadState::self()->getCallingPid()); 370} 371 372AudioFlinger::PlaybackThread::Track::~Track() 373{ 374 ALOGV("PlaybackThread::Track destructor"); 375} 376 377void AudioFlinger::PlaybackThread::Track::destroy() 378{ 379 // NOTE: destroyTrack_l() can remove a strong reference to this Track 380 // by removing it from mTracks vector, so there is a risk that this Tracks's 381 // destructor is called. As the destructor needs to lock mLock, 382 // we must acquire a strong reference on this Track before locking mLock 383 // here so that the destructor is called only when exiting this function. 384 // On the other hand, as long as Track::destroy() is only called by 385 // TrackHandle destructor, the TrackHandle still holds a strong ref on 386 // this Track with its member mTrack. 387 sp<Track> keep(this); 388 { // scope for mLock 389 sp<ThreadBase> thread = mThread.promote(); 390 if (thread != 0) { 391 Mutex::Autolock _l(thread->mLock); 392 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 393 bool wasActive = playbackThread->destroyTrack_l(this); 394 if (!isOutputTrack() && !wasActive) { 395 AudioSystem::releaseOutput(thread->id()); 396 } 397 } 398 } 399} 400 401/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 402{ 403 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate " 404 "L dB R dB Server Main buf Aux Buf Flags Underruns\n"); 405} 406 407void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 408{ 409 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 410 if (isFastTrack()) { 411 sprintf(buffer, " F %2d", mFastIndex); 412 } else { 413 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 414 } 415 track_state state = mState; 416 char stateChar; 417 if (isTerminated()) { 418 stateChar = 'T'; 419 } else { 420 switch (state) { 421 case IDLE: 422 stateChar = 'I'; 423 break; 424 case STOPPING_1: 425 stateChar = 's'; 426 break; 427 case STOPPING_2: 428 stateChar = '5'; 429 break; 430 case STOPPED: 431 stateChar = 'S'; 432 break; 433 case RESUMING: 434 stateChar = 'R'; 435 break; 436 case ACTIVE: 437 stateChar = 'A'; 438 break; 439 case PAUSING: 440 stateChar = 'p'; 441 break; 442 case PAUSED: 443 stateChar = 'P'; 444 break; 445 case FLUSHED: 446 stateChar = 'F'; 447 break; 448 default: 449 stateChar = '?'; 450 break; 451 } 452 } 453 char nowInUnderrun; 454 switch (mObservedUnderruns.mBitFields.mMostRecent) { 455 case UNDERRUN_FULL: 456 nowInUnderrun = ' '; 457 break; 458 case UNDERRUN_PARTIAL: 459 nowInUnderrun = '<'; 460 break; 461 case UNDERRUN_EMPTY: 462 nowInUnderrun = '*'; 463 break; 464 default: 465 nowInUnderrun = '?'; 466 break; 467 } 468 snprintf(&buffer[7], size-7, " %6d %4u 0x%08x 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g " 469 "0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 470 (mClient == 0) ? getpid_cached : mClient->pid(), 471 mStreamType, 472 mFormat, 473 mChannelMask, 474 mSessionId, 475 mStepCount, 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->server, 483 (int)mMainBuffer, 484 (int)mAuxBuffer, 485 mCblk->flags, 486 mUnderrunCount, 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 // only implemented so far for normal tracks, not fast tracks 506 mCblk->u.mStreaming.mUnderrunFrames += desiredFrames; 507 // FIXME also wake futex so that underrun is noticed more quickly 508 (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 509 } 510 return status; 511} 512 513// Note that framesReady() takes a mutex on the control block using tryLock(). 514// This could result in priority inversion if framesReady() is called by the normal mixer, 515// as the normal mixer thread runs at lower 516// priority than the client's callback thread: there is a short window within framesReady() 517// during which the normal mixer could be preempted, and the client callback would block. 518// Another problem can occur if framesReady() is called by the fast mixer: 519// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 520// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 521size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 522 return mAudioTrackServerProxy->framesReady(); 523} 524 525// Don't call for fast tracks; the framesReady() could result in priority inversion 526bool AudioFlinger::PlaybackThread::Track::isReady() const { 527 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 528 return true; 529 } 530 531 if (framesReady() >= mFrameCount || 532 (mCblk->flags & CBLK_FORCEREADY)) { 533 mFillingUpStatus = FS_FILLED; 534 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 535 return true; 536 } 537 return false; 538} 539 540status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 541 int triggerSession) 542{ 543 status_t status = NO_ERROR; 544 ALOGV("start(%d), calling pid %d session %d", 545 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 546 547 sp<ThreadBase> thread = mThread.promote(); 548 if (thread != 0) { 549 Mutex::Autolock _l(thread->mLock); 550 track_state state = mState; 551 // here the track could be either new, or restarted 552 // in both cases "unstop" the track 553 554 if (state == PAUSED) { 555 if (mResumeToStopping) { 556 // happened we need to resume to STOPPING_1 557 mState = TrackBase::STOPPING_1; 558 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 559 } else { 560 mState = TrackBase::RESUMING; 561 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 562 } 563 } else { 564 mState = TrackBase::ACTIVE; 565 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 566 } 567 568 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 569 status = playbackThread->addTrack_l(this); 570 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 571 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 572 // restore previous state if start was rejected by policy manager 573 if (status == PERMISSION_DENIED) { 574 mState = state; 575 } 576 } 577 // track was already in the active list, not a problem 578 if (status == ALREADY_EXISTS) { 579 status = NO_ERROR; 580 } 581 } else { 582 status = BAD_VALUE; 583 } 584 return status; 585} 586 587void AudioFlinger::PlaybackThread::Track::stop() 588{ 589 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 590 sp<ThreadBase> thread = mThread.promote(); 591 if (thread != 0) { 592 Mutex::Autolock _l(thread->mLock); 593 track_state state = mState; 594 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 595 // If the track is not active (PAUSED and buffers full), flush buffers 596 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 597 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 598 reset(); 599 mState = STOPPED; 600 } else if (!isFastTrack() && !isOffloaded()) { 601 mState = STOPPED; 602 } else { 603 // For fast tracks prepareTracks_l() will set state to STOPPING_2 604 // presentation is complete 605 // For an offloaded track this starts a drain and state will 606 // move to STOPPING_2 when drain completes and then STOPPED 607 mState = STOPPING_1; 608 } 609 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 610 playbackThread); 611 } 612 } 613} 614 615void AudioFlinger::PlaybackThread::Track::pause() 616{ 617 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 618 sp<ThreadBase> thread = mThread.promote(); 619 if (thread != 0) { 620 Mutex::Autolock _l(thread->mLock); 621 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 622 switch (mState) { 623 case STOPPING_1: 624 case STOPPING_2: 625 if (!isOffloaded()) { 626 /* nothing to do if track is not offloaded */ 627 break; 628 } 629 630 // Offloaded track was draining, we need to carry on draining when resumed 631 mResumeToStopping = true; 632 // fall through... 633 case ACTIVE: 634 case RESUMING: 635 mState = PAUSING; 636 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 637 playbackThread->signal_l(); 638 break; 639 640 default: 641 break; 642 } 643 } 644} 645 646void AudioFlinger::PlaybackThread::Track::flush() 647{ 648 ALOGV("flush(%d)", mName); 649 sp<ThreadBase> thread = mThread.promote(); 650 if (thread != 0) { 651 Mutex::Autolock _l(thread->mLock); 652 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 653 654 if (isOffloaded()) { 655 // If offloaded we allow flush during any state except terminated 656 // and keep the track active to avoid problems if user is seeking 657 // rapidly and underlying hardware has a significant delay handling 658 // a pause 659 if (isTerminated()) { 660 return; 661 } 662 663 ALOGV("flush: offload flush"); 664 reset(); 665 666 if (mState == STOPPING_1 || mState == STOPPING_2) { 667 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 668 mState = ACTIVE; 669 } 670 671 if (mState == ACTIVE) { 672 ALOGV("flush called in active state, resetting buffer time out retry count"); 673 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 674 } 675 676 mResumeToStopping = false; 677 } else { 678 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 679 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 680 return; 681 } 682 // No point remaining in PAUSED state after a flush => go to 683 // FLUSHED state 684 mState = FLUSHED; 685 // do not reset the track if it is still in the process of being stopped or paused. 686 // this will be done by prepareTracks_l() when the track is stopped. 687 // prepareTracks_l() will see mState == FLUSHED, then 688 // remove from active track list, reset(), and trigger presentation complete 689 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 690 reset(); 691 } 692 } 693 // Prevent flush being lost if the track is flushed and then resumed 694 // before mixer thread can run. This is important when offloading 695 // because the hardware buffer could hold a large amount of audio 696 playbackThread->flushOutput_l(); 697 playbackThread->signal_l(); 698 } 699} 700 701void AudioFlinger::PlaybackThread::Track::reset() 702{ 703 // Do not reset twice to avoid discarding data written just after a flush and before 704 // the audioflinger thread detects the track is stopped. 705 if (!mResetDone) { 706 TrackBase::reset(); 707 // Force underrun condition to avoid false underrun callback until first data is 708 // written to buffer 709 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 710 mFillingUpStatus = FS_FILLING; 711 mResetDone = true; 712 if (mState == FLUSHED) { 713 mState = IDLE; 714 } 715 } 716} 717 718status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 719{ 720 sp<ThreadBase> thread = mThread.promote(); 721 if (thread == 0) { 722 ALOGE("thread is dead"); 723 return FAILED_TRANSACTION; 724 } else if ((thread->type() == ThreadBase::DIRECT) || 725 (thread->type() == ThreadBase::OFFLOAD)) { 726 return thread->setParameters(keyValuePairs); 727 } else { 728 return PERMISSION_DENIED; 729 } 730} 731 732status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 733{ 734 status_t status = DEAD_OBJECT; 735 sp<ThreadBase> thread = mThread.promote(); 736 if (thread != 0) { 737 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 738 sp<AudioFlinger> af = mClient->audioFlinger(); 739 740 Mutex::Autolock _l(af->mLock); 741 742 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 743 744 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 745 Mutex::Autolock _dl(playbackThread->mLock); 746 Mutex::Autolock _sl(srcThread->mLock); 747 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 748 if (chain == 0) { 749 return INVALID_OPERATION; 750 } 751 752 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 753 if (effect == 0) { 754 return INVALID_OPERATION; 755 } 756 srcThread->removeEffect_l(effect); 757 playbackThread->addEffect_l(effect); 758 // removeEffect_l() has stopped the effect if it was active so it must be restarted 759 if (effect->state() == EffectModule::ACTIVE || 760 effect->state() == EffectModule::STOPPING) { 761 effect->start(); 762 } 763 764 sp<EffectChain> dstChain = effect->chain().promote(); 765 if (dstChain == 0) { 766 srcThread->addEffect_l(effect); 767 return INVALID_OPERATION; 768 } 769 AudioSystem::unregisterEffect(effect->id()); 770 AudioSystem::registerEffect(&effect->desc(), 771 srcThread->id(), 772 dstChain->strategy(), 773 AUDIO_SESSION_OUTPUT_MIX, 774 effect->id()); 775 } 776 status = playbackThread->attachAuxEffect(this, EffectId); 777 } 778 return status; 779} 780 781void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 782{ 783 mAuxEffectId = EffectId; 784 mAuxBuffer = buffer; 785} 786 787bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 788 size_t audioHalFrames) 789{ 790 // a track is considered presented when the total number of frames written to audio HAL 791 // corresponds to the number of frames written when presentationComplete() is called for the 792 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 793 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 794 // to detect when all frames have been played. In this case framesWritten isn't 795 // useful because it doesn't always reflect whether there is data in the h/w 796 // buffers, particularly if a track has been paused and resumed during draining 797 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 798 mPresentationCompleteFrames, framesWritten); 799 if (mPresentationCompleteFrames == 0) { 800 mPresentationCompleteFrames = framesWritten + audioHalFrames; 801 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 802 mPresentationCompleteFrames, audioHalFrames); 803 } 804 805 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 806 ALOGV("presentationComplete() session %d complete: framesWritten %d", 807 mSessionId, framesWritten); 808 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 809 mAudioTrackServerProxy->setStreamEndDone(); 810 return true; 811 } 812 return false; 813} 814 815void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 816{ 817 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 818 if (mSyncEvents[i]->type() == type) { 819 mSyncEvents[i]->trigger(); 820 mSyncEvents.removeAt(i); 821 i--; 822 } 823 } 824} 825 826// implement VolumeBufferProvider interface 827 828uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 829{ 830 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 831 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 832 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 833 uint32_t vl = vlr & 0xFFFF; 834 uint32_t vr = vlr >> 16; 835 // track volumes come from shared memory, so can't be trusted and must be clamped 836 if (vl > MAX_GAIN_INT) { 837 vl = MAX_GAIN_INT; 838 } 839 if (vr > MAX_GAIN_INT) { 840 vr = MAX_GAIN_INT; 841 } 842 // now apply the cached master volume and stream type volume; 843 // this is trusted but lacks any synchronization or barrier so may be stale 844 float v = mCachedVolume; 845 vl *= v; 846 vr *= v; 847 // re-combine into U4.16 848 vlr = (vr << 16) | (vl & 0xFFFF); 849 // FIXME look at mute, pause, and stop flags 850 return vlr; 851} 852 853status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 854{ 855 if (isTerminated() || mState == PAUSED || 856 ((framesReady() == 0) && ((mSharedBuffer != 0) || 857 (mState == STOPPED)))) { 858 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 859 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 860 event->cancel(); 861 return INVALID_OPERATION; 862 } 863 (void) TrackBase::setSyncEvent(event); 864 return NO_ERROR; 865} 866 867void AudioFlinger::PlaybackThread::Track::invalidate() 868{ 869 // FIXME should use proxy, and needs work 870 audio_track_cblk_t* cblk = mCblk; 871 android_atomic_or(CBLK_INVALID, &cblk->flags); 872 android_atomic_release_store(0x40000000, &cblk->mFutex); 873 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 874 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 875 mIsInvalid = true; 876} 877 878// ---------------------------------------------------------------------------- 879 880sp<AudioFlinger::PlaybackThread::TimedTrack> 881AudioFlinger::PlaybackThread::TimedTrack::create( 882 PlaybackThread *thread, 883 const sp<Client>& client, 884 audio_stream_type_t streamType, 885 uint32_t sampleRate, 886 audio_format_t format, 887 audio_channel_mask_t channelMask, 888 size_t frameCount, 889 const sp<IMemory>& sharedBuffer, 890 int sessionId) { 891 if (!client->reserveTimedTrack()) 892 return 0; 893 894 return new TimedTrack( 895 thread, client, streamType, sampleRate, format, channelMask, frameCount, 896 sharedBuffer, sessionId); 897} 898 899AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 900 PlaybackThread *thread, 901 const sp<Client>& client, 902 audio_stream_type_t streamType, 903 uint32_t sampleRate, 904 audio_format_t format, 905 audio_channel_mask_t channelMask, 906 size_t frameCount, 907 const sp<IMemory>& sharedBuffer, 908 int sessionId) 909 : Track(thread, client, streamType, sampleRate, format, channelMask, 910 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 911 mQueueHeadInFlight(false), 912 mTrimQueueHeadOnRelease(false), 913 mFramesPendingInQueue(0), 914 mTimedSilenceBuffer(NULL), 915 mTimedSilenceBufferSize(0), 916 mTimedAudioOutputOnTime(false), 917 mMediaTimeTransformValid(false) 918{ 919 LocalClock lc; 920 mLocalTimeFreq = lc.getLocalFreq(); 921 922 mLocalTimeToSampleTransform.a_zero = 0; 923 mLocalTimeToSampleTransform.b_zero = 0; 924 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 925 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 926 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 927 &mLocalTimeToSampleTransform.a_to_b_denom); 928 929 mMediaTimeToSampleTransform.a_zero = 0; 930 mMediaTimeToSampleTransform.b_zero = 0; 931 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 932 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 933 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 934 &mMediaTimeToSampleTransform.a_to_b_denom); 935} 936 937AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 938 mClient->releaseTimedTrack(); 939 delete [] mTimedSilenceBuffer; 940} 941 942status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 943 size_t size, sp<IMemory>* buffer) { 944 945 Mutex::Autolock _l(mTimedBufferQueueLock); 946 947 trimTimedBufferQueue_l(); 948 949 // lazily initialize the shared memory heap for timed buffers 950 if (mTimedMemoryDealer == NULL) { 951 const int kTimedBufferHeapSize = 512 << 10; 952 953 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 954 "AudioFlingerTimed"); 955 if (mTimedMemoryDealer == NULL) 956 return NO_MEMORY; 957 } 958 959 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 960 if (newBuffer == NULL) { 961 newBuffer = mTimedMemoryDealer->allocate(size); 962 if (newBuffer == NULL) 963 return NO_MEMORY; 964 } 965 966 *buffer = newBuffer; 967 return NO_ERROR; 968} 969 970// caller must hold mTimedBufferQueueLock 971void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 972 int64_t mediaTimeNow; 973 { 974 Mutex::Autolock mttLock(mMediaTimeTransformLock); 975 if (!mMediaTimeTransformValid) 976 return; 977 978 int64_t targetTimeNow; 979 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 980 ? mCCHelper.getCommonTime(&targetTimeNow) 981 : mCCHelper.getLocalTime(&targetTimeNow); 982 983 if (OK != res) 984 return; 985 986 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 987 &mediaTimeNow)) { 988 return; 989 } 990 } 991 992 size_t trimEnd; 993 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 994 int64_t bufEnd; 995 996 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 997 // We have a next buffer. Just use its PTS as the PTS of the frame 998 // following the last frame in this buffer. If the stream is sparse 999 // (ie, there are deliberate gaps left in the stream which should be 1000 // filled with silence by the TimedAudioTrack), then this can result 1001 // in one extra buffer being left un-trimmed when it could have 1002 // been. In general, this is not typical, and we would rather 1003 // optimized away the TS calculation below for the more common case 1004 // where PTSes are contiguous. 1005 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1006 } else { 1007 // We have no next buffer. Compute the PTS of the frame following 1008 // the last frame in this buffer by computing the duration of of 1009 // this frame in media time units and adding it to the PTS of the 1010 // buffer. 1011 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1012 / mFrameSize; 1013 1014 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1015 &bufEnd)) { 1016 ALOGE("Failed to convert frame count of %lld to media time" 1017 " duration" " (scale factor %d/%u) in %s", 1018 frameCount, 1019 mMediaTimeToSampleTransform.a_to_b_numer, 1020 mMediaTimeToSampleTransform.a_to_b_denom, 1021 __PRETTY_FUNCTION__); 1022 break; 1023 } 1024 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1025 } 1026 1027 if (bufEnd > mediaTimeNow) 1028 break; 1029 1030 // Is the buffer we want to use in the middle of a mix operation right 1031 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1032 // from the mixer which should be coming back shortly. 1033 if (!trimEnd && mQueueHeadInFlight) { 1034 mTrimQueueHeadOnRelease = true; 1035 } 1036 } 1037 1038 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1039 if (trimStart < trimEnd) { 1040 // Update the bookkeeping for framesReady() 1041 for (size_t i = trimStart; i < trimEnd; ++i) { 1042 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1043 } 1044 1045 // Now actually remove the buffers from the queue. 1046 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1047 } 1048} 1049 1050void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1051 const char* logTag) { 1052 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1053 "%s called (reason \"%s\"), but timed buffer queue has no" 1054 " elements to trim.", __FUNCTION__, logTag); 1055 1056 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1057 mTimedBufferQueue.removeAt(0); 1058} 1059 1060void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1061 const TimedBuffer& buf, 1062 const char* logTag) { 1063 uint32_t bufBytes = buf.buffer()->size(); 1064 uint32_t consumedAlready = buf.position(); 1065 1066 ALOG_ASSERT(consumedAlready <= bufBytes, 1067 "Bad bookkeeping while updating frames pending. Timed buffer is" 1068 " only %u bytes long, but claims to have consumed %u" 1069 " bytes. (update reason: \"%s\")", 1070 bufBytes, consumedAlready, logTag); 1071 1072 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1073 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1074 "Bad bookkeeping while updating frames pending. Should have at" 1075 " least %u queued frames, but we think we have only %u. (update" 1076 " reason: \"%s\")", 1077 bufFrames, mFramesPendingInQueue, logTag); 1078 1079 mFramesPendingInQueue -= bufFrames; 1080} 1081 1082status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1083 const sp<IMemory>& buffer, int64_t pts) { 1084 1085 { 1086 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1087 if (!mMediaTimeTransformValid) 1088 return INVALID_OPERATION; 1089 } 1090 1091 Mutex::Autolock _l(mTimedBufferQueueLock); 1092 1093 uint32_t bufFrames = buffer->size() / mFrameSize; 1094 mFramesPendingInQueue += bufFrames; 1095 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1096 1097 return NO_ERROR; 1098} 1099 1100status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1101 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1102 1103 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1104 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1105 target); 1106 1107 if (!(target == TimedAudioTrack::LOCAL_TIME || 1108 target == TimedAudioTrack::COMMON_TIME)) { 1109 return BAD_VALUE; 1110 } 1111 1112 Mutex::Autolock lock(mMediaTimeTransformLock); 1113 mMediaTimeTransform = xform; 1114 mMediaTimeTransformTarget = target; 1115 mMediaTimeTransformValid = true; 1116 1117 return NO_ERROR; 1118} 1119 1120#define min(a, b) ((a) < (b) ? (a) : (b)) 1121 1122// implementation of getNextBuffer for tracks whose buffers have timestamps 1123status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1124 AudioBufferProvider::Buffer* buffer, int64_t pts) 1125{ 1126 if (pts == AudioBufferProvider::kInvalidPTS) { 1127 buffer->raw = NULL; 1128 buffer->frameCount = 0; 1129 mTimedAudioOutputOnTime = false; 1130 return INVALID_OPERATION; 1131 } 1132 1133 Mutex::Autolock _l(mTimedBufferQueueLock); 1134 1135 ALOG_ASSERT(!mQueueHeadInFlight, 1136 "getNextBuffer called without releaseBuffer!"); 1137 1138 while (true) { 1139 1140 // if we have no timed buffers, then fail 1141 if (mTimedBufferQueue.isEmpty()) { 1142 buffer->raw = NULL; 1143 buffer->frameCount = 0; 1144 return NOT_ENOUGH_DATA; 1145 } 1146 1147 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1148 1149 // calculate the PTS of the head of the timed buffer queue expressed in 1150 // local time 1151 int64_t headLocalPTS; 1152 { 1153 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1154 1155 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1156 1157 if (mMediaTimeTransform.a_to_b_denom == 0) { 1158 // the transform represents a pause, so yield silence 1159 timedYieldSilence_l(buffer->frameCount, buffer); 1160 return NO_ERROR; 1161 } 1162 1163 int64_t transformedPTS; 1164 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1165 &transformedPTS)) { 1166 // the transform failed. this shouldn't happen, but if it does 1167 // then just drop this buffer 1168 ALOGW("timedGetNextBuffer transform failed"); 1169 buffer->raw = NULL; 1170 buffer->frameCount = 0; 1171 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1172 return NO_ERROR; 1173 } 1174 1175 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1176 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1177 &headLocalPTS)) { 1178 buffer->raw = NULL; 1179 buffer->frameCount = 0; 1180 return INVALID_OPERATION; 1181 } 1182 } else { 1183 headLocalPTS = transformedPTS; 1184 } 1185 } 1186 1187 // adjust the head buffer's PTS to reflect the portion of the head buffer 1188 // that has already been consumed 1189 int64_t effectivePTS = headLocalPTS + 1190 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 1191 1192 // Calculate the delta in samples between the head of the input buffer 1193 // queue and the start of the next output buffer that will be written. 1194 // If the transformation fails because of over or underflow, it means 1195 // that the sample's position in the output stream is so far out of 1196 // whack that it should just be dropped. 1197 int64_t sampleDelta; 1198 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1199 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1200 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1201 " mix"); 1202 continue; 1203 } 1204 if (!mLocalTimeToSampleTransform.doForwardTransform( 1205 (effectivePTS - pts) << 32, &sampleDelta)) { 1206 ALOGV("*** too late during sample rate transform: dropped buffer"); 1207 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1208 continue; 1209 } 1210 1211 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1212 " sampleDelta=[%d.%08x]", 1213 head.pts(), head.position(), pts, 1214 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1215 + (sampleDelta >> 32)), 1216 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1217 1218 // if the delta between the ideal placement for the next input sample and 1219 // the current output position is within this threshold, then we will 1220 // concatenate the next input samples to the previous output 1221 const int64_t kSampleContinuityThreshold = 1222 (static_cast<int64_t>(sampleRate()) << 32) / 250; 1223 1224 // if this is the first buffer of audio that we're emitting from this track 1225 // then it should be almost exactly on time. 1226 const int64_t kSampleStartupThreshold = 1LL << 32; 1227 1228 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1229 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1230 // the next input is close enough to being on time, so concatenate it 1231 // with the last output 1232 timedYieldSamples_l(buffer); 1233 1234 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1235 head.position(), buffer->frameCount); 1236 return NO_ERROR; 1237 } 1238 1239 // Looks like our output is not on time. Reset our on timed status. 1240 // Next time we mix samples from our input queue, then should be within 1241 // the StartupThreshold. 1242 mTimedAudioOutputOnTime = false; 1243 if (sampleDelta > 0) { 1244 // the gap between the current output position and the proper start of 1245 // the next input sample is too big, so fill it with silence 1246 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1247 1248 timedYieldSilence_l(framesUntilNextInput, buffer); 1249 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1250 return NO_ERROR; 1251 } else { 1252 // the next input sample is late 1253 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1254 size_t onTimeSamplePosition = 1255 head.position() + lateFrames * mFrameSize; 1256 1257 if (onTimeSamplePosition > head.buffer()->size()) { 1258 // all the remaining samples in the head are too late, so 1259 // drop it and move on 1260 ALOGV("*** too late: dropped buffer"); 1261 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1262 continue; 1263 } else { 1264 // skip over the late samples 1265 head.setPosition(onTimeSamplePosition); 1266 1267 // yield the available samples 1268 timedYieldSamples_l(buffer); 1269 1270 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1271 return NO_ERROR; 1272 } 1273 } 1274 } 1275} 1276 1277// Yield samples from the timed buffer queue head up to the given output 1278// buffer's capacity. 1279// 1280// Caller must hold mTimedBufferQueueLock 1281void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1282 AudioBufferProvider::Buffer* buffer) { 1283 1284 const TimedBuffer& head = mTimedBufferQueue[0]; 1285 1286 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1287 head.position()); 1288 1289 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1290 mFrameSize); 1291 size_t framesRequested = buffer->frameCount; 1292 buffer->frameCount = min(framesLeftInHead, framesRequested); 1293 1294 mQueueHeadInFlight = true; 1295 mTimedAudioOutputOnTime = true; 1296} 1297 1298// Yield samples of silence up to the given output buffer's capacity 1299// 1300// Caller must hold mTimedBufferQueueLock 1301void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1302 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1303 1304 // lazily allocate a buffer filled with silence 1305 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1306 delete [] mTimedSilenceBuffer; 1307 mTimedSilenceBufferSize = numFrames * mFrameSize; 1308 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1309 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1310 } 1311 1312 buffer->raw = mTimedSilenceBuffer; 1313 size_t framesRequested = buffer->frameCount; 1314 buffer->frameCount = min(numFrames, framesRequested); 1315 1316 mTimedAudioOutputOnTime = false; 1317} 1318 1319// AudioBufferProvider interface 1320void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1321 AudioBufferProvider::Buffer* buffer) { 1322 1323 Mutex::Autolock _l(mTimedBufferQueueLock); 1324 1325 // If the buffer which was just released is part of the buffer at the head 1326 // of the queue, be sure to update the amt of the buffer which has been 1327 // consumed. If the buffer being returned is not part of the head of the 1328 // queue, its either because the buffer is part of the silence buffer, or 1329 // because the head of the timed queue was trimmed after the mixer called 1330 // getNextBuffer but before the mixer called releaseBuffer. 1331 if (buffer->raw == mTimedSilenceBuffer) { 1332 ALOG_ASSERT(!mQueueHeadInFlight, 1333 "Queue head in flight during release of silence buffer!"); 1334 goto done; 1335 } 1336 1337 ALOG_ASSERT(mQueueHeadInFlight, 1338 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1339 " head in flight."); 1340 1341 if (mTimedBufferQueue.size()) { 1342 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1343 1344 void* start = head.buffer()->pointer(); 1345 void* end = reinterpret_cast<void*>( 1346 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1347 + head.buffer()->size()); 1348 1349 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1350 "released buffer not within the head of the timed buffer" 1351 " queue; qHead = [%p, %p], released buffer = %p", 1352 start, end, buffer->raw); 1353 1354 head.setPosition(head.position() + 1355 (buffer->frameCount * mFrameSize)); 1356 mQueueHeadInFlight = false; 1357 1358 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1359 "Bad bookkeeping during releaseBuffer! Should have at" 1360 " least %u queued frames, but we think we have only %u", 1361 buffer->frameCount, mFramesPendingInQueue); 1362 1363 mFramesPendingInQueue -= buffer->frameCount; 1364 1365 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1366 || mTrimQueueHeadOnRelease) { 1367 trimTimedBufferQueueHead_l("releaseBuffer"); 1368 mTrimQueueHeadOnRelease = false; 1369 } 1370 } else { 1371 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1372 " buffers in the timed buffer queue"); 1373 } 1374 1375done: 1376 buffer->raw = 0; 1377 buffer->frameCount = 0; 1378} 1379 1380size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1381 Mutex::Autolock _l(mTimedBufferQueueLock); 1382 return mFramesPendingInQueue; 1383} 1384 1385AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1386 : mPTS(0), mPosition(0) {} 1387 1388AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1389 const sp<IMemory>& buffer, int64_t pts) 1390 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1391 1392 1393// ---------------------------------------------------------------------------- 1394 1395AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1396 PlaybackThread *playbackThread, 1397 DuplicatingThread *sourceThread, 1398 uint32_t sampleRate, 1399 audio_format_t format, 1400 audio_channel_mask_t channelMask, 1401 size_t frameCount) 1402 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1403 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1404 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1405{ 1406 1407 if (mCblk != NULL) { 1408 mOutBuffer.frameCount = 0; 1409 playbackThread->mTracks.add(this); 1410 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1411 "mCblk->frameCount_ %u, mChannelMask 0x%08x mBufferEnd %p", 1412 mCblk, mBuffer, 1413 mCblk->frameCount_, mChannelMask, mBufferEnd); 1414 // since client and server are in the same process, 1415 // the buffer has the same virtual address on both sides 1416 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1417 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1418 mClientProxy->setSendLevel(0.0); 1419 mClientProxy->setSampleRate(sampleRate); 1420 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1421 true /*clientInServer*/); 1422 } else { 1423 ALOGW("Error creating output track on thread %p", playbackThread); 1424 } 1425} 1426 1427AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1428{ 1429 clearBufferQueue(); 1430 delete mClientProxy; 1431 // superclass destructor will now delete the server proxy and shared memory both refer to 1432} 1433 1434status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1435 int triggerSession) 1436{ 1437 status_t status = Track::start(event, triggerSession); 1438 if (status != NO_ERROR) { 1439 return status; 1440 } 1441 1442 mActive = true; 1443 mRetryCount = 127; 1444 return status; 1445} 1446 1447void AudioFlinger::PlaybackThread::OutputTrack::stop() 1448{ 1449 Track::stop(); 1450 clearBufferQueue(); 1451 mOutBuffer.frameCount = 0; 1452 mActive = false; 1453} 1454 1455bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1456{ 1457 Buffer *pInBuffer; 1458 Buffer inBuffer; 1459 uint32_t channelCount = mChannelCount; 1460 bool outputBufferFull = false; 1461 inBuffer.frameCount = frames; 1462 inBuffer.i16 = data; 1463 1464 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1465 1466 if (!mActive && frames != 0) { 1467 start(); 1468 sp<ThreadBase> thread = mThread.promote(); 1469 if (thread != 0) { 1470 MixerThread *mixerThread = (MixerThread *)thread.get(); 1471 if (mFrameCount > frames) { 1472 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1473 uint32_t startFrames = (mFrameCount - frames); 1474 pInBuffer = new Buffer; 1475 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1476 pInBuffer->frameCount = startFrames; 1477 pInBuffer->i16 = pInBuffer->mBuffer; 1478 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1479 mBufferQueue.add(pInBuffer); 1480 } else { 1481 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1482 } 1483 } 1484 } 1485 } 1486 1487 while (waitTimeLeftMs) { 1488 // First write pending buffers, then new data 1489 if (mBufferQueue.size()) { 1490 pInBuffer = mBufferQueue.itemAt(0); 1491 } else { 1492 pInBuffer = &inBuffer; 1493 } 1494 1495 if (pInBuffer->frameCount == 0) { 1496 break; 1497 } 1498 1499 if (mOutBuffer.frameCount == 0) { 1500 mOutBuffer.frameCount = pInBuffer->frameCount; 1501 nsecs_t startTime = systemTime(); 1502 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1503 if (status != NO_ERROR) { 1504 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1505 mThread.unsafe_get(), status); 1506 outputBufferFull = true; 1507 break; 1508 } 1509 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1510 if (waitTimeLeftMs >= waitTimeMs) { 1511 waitTimeLeftMs -= waitTimeMs; 1512 } else { 1513 waitTimeLeftMs = 0; 1514 } 1515 } 1516 1517 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1518 pInBuffer->frameCount; 1519 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1520 Proxy::Buffer buf; 1521 buf.mFrameCount = outFrames; 1522 buf.mRaw = NULL; 1523 mClientProxy->releaseBuffer(&buf); 1524 pInBuffer->frameCount -= outFrames; 1525 pInBuffer->i16 += outFrames * channelCount; 1526 mOutBuffer.frameCount -= outFrames; 1527 mOutBuffer.i16 += outFrames * channelCount; 1528 1529 if (pInBuffer->frameCount == 0) { 1530 if (mBufferQueue.size()) { 1531 mBufferQueue.removeAt(0); 1532 delete [] pInBuffer->mBuffer; 1533 delete pInBuffer; 1534 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1535 mThread.unsafe_get(), mBufferQueue.size()); 1536 } else { 1537 break; 1538 } 1539 } 1540 } 1541 1542 // If we could not write all frames, allocate a buffer and queue it for next time. 1543 if (inBuffer.frameCount) { 1544 sp<ThreadBase> thread = mThread.promote(); 1545 if (thread != 0 && !thread->standby()) { 1546 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1547 pInBuffer = new Buffer; 1548 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1549 pInBuffer->frameCount = inBuffer.frameCount; 1550 pInBuffer->i16 = pInBuffer->mBuffer; 1551 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1552 sizeof(int16_t)); 1553 mBufferQueue.add(pInBuffer); 1554 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1555 mThread.unsafe_get(), mBufferQueue.size()); 1556 } else { 1557 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1558 mThread.unsafe_get(), this); 1559 } 1560 } 1561 } 1562 1563 // Calling write() with a 0 length buffer, means that no more data will be written: 1564 // If no more buffers are pending, fill output track buffer to make sure it is started 1565 // by output mixer. 1566 if (frames == 0 && mBufferQueue.size() == 0) { 1567 // FIXME borken, replace by getting framesReady() from proxy 1568 size_t user = 0; // was mCblk->user 1569 if (user < mFrameCount) { 1570 frames = mFrameCount - user; 1571 pInBuffer = new Buffer; 1572 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1573 pInBuffer->frameCount = frames; 1574 pInBuffer->i16 = pInBuffer->mBuffer; 1575 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1576 mBufferQueue.add(pInBuffer); 1577 } else if (mActive) { 1578 stop(); 1579 } 1580 } 1581 1582 return outputBufferFull; 1583} 1584 1585status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1586 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1587{ 1588 ClientProxy::Buffer buf; 1589 buf.mFrameCount = buffer->frameCount; 1590 struct timespec timeout; 1591 timeout.tv_sec = waitTimeMs / 1000; 1592 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1593 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1594 buffer->frameCount = buf.mFrameCount; 1595 buffer->raw = buf.mRaw; 1596 return status; 1597} 1598 1599void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1600{ 1601 size_t size = mBufferQueue.size(); 1602 1603 for (size_t i = 0; i < size; i++) { 1604 Buffer *pBuffer = mBufferQueue.itemAt(i); 1605 delete [] pBuffer->mBuffer; 1606 delete pBuffer; 1607 } 1608 mBufferQueue.clear(); 1609} 1610 1611 1612// ---------------------------------------------------------------------------- 1613// Record 1614// ---------------------------------------------------------------------------- 1615 1616AudioFlinger::RecordHandle::RecordHandle( 1617 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1618 : BnAudioRecord(), 1619 mRecordTrack(recordTrack) 1620{ 1621} 1622 1623AudioFlinger::RecordHandle::~RecordHandle() { 1624 stop_nonvirtual(); 1625 mRecordTrack->destroy(); 1626} 1627 1628sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1629 return mRecordTrack->getCblk(); 1630} 1631 1632status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1633 int triggerSession) { 1634 ALOGV("RecordHandle::start()"); 1635 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1636} 1637 1638void AudioFlinger::RecordHandle::stop() { 1639 stop_nonvirtual(); 1640} 1641 1642void AudioFlinger::RecordHandle::stop_nonvirtual() { 1643 ALOGV("RecordHandle::stop()"); 1644 mRecordTrack->stop(); 1645} 1646 1647status_t AudioFlinger::RecordHandle::onTransact( 1648 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1649{ 1650 return BnAudioRecord::onTransact(code, data, reply, flags); 1651} 1652 1653// ---------------------------------------------------------------------------- 1654 1655// RecordTrack constructor must be called with AudioFlinger::mLock held 1656AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1657 RecordThread *thread, 1658 const sp<Client>& client, 1659 uint32_t sampleRate, 1660 audio_format_t format, 1661 audio_channel_mask_t channelMask, 1662 size_t frameCount, 1663 int sessionId) 1664 : TrackBase(thread, client, sampleRate, format, 1665 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1666 mOverflow(false) 1667{ 1668 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 1669 if (mCblk != NULL) { 1670 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1671 mFrameSize); 1672 mServerProxy = mAudioRecordServerProxy; 1673 } 1674} 1675 1676AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1677{ 1678 ALOGV("%s", __func__); 1679} 1680 1681// AudioBufferProvider interface 1682status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1683 int64_t pts) 1684{ 1685 ServerProxy::Buffer buf; 1686 buf.mFrameCount = buffer->frameCount; 1687 status_t status = mServerProxy->obtainBuffer(&buf); 1688 buffer->frameCount = buf.mFrameCount; 1689 buffer->raw = buf.mRaw; 1690 if (buf.mFrameCount == 0) { 1691 // FIXME also wake futex so that overrun is noticed more quickly 1692 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->flags); 1693 } 1694 return status; 1695} 1696 1697status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1698 int triggerSession) 1699{ 1700 sp<ThreadBase> thread = mThread.promote(); 1701 if (thread != 0) { 1702 RecordThread *recordThread = (RecordThread *)thread.get(); 1703 return recordThread->start(this, event, triggerSession); 1704 } else { 1705 return BAD_VALUE; 1706 } 1707} 1708 1709void AudioFlinger::RecordThread::RecordTrack::stop() 1710{ 1711 sp<ThreadBase> thread = mThread.promote(); 1712 if (thread != 0) { 1713 RecordThread *recordThread = (RecordThread *)thread.get(); 1714 recordThread->mLock.lock(); 1715 bool doStop = recordThread->stop_l(this); 1716 if (doStop) { 1717 TrackBase::reset(); 1718 // Force overrun condition to avoid false overrun callback until first data is 1719 // read from buffer 1720 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 1721 } 1722 recordThread->mLock.unlock(); 1723 if (doStop) { 1724 AudioSystem::stopInput(recordThread->id()); 1725 } 1726 } 1727} 1728 1729void AudioFlinger::RecordThread::RecordTrack::destroy() 1730{ 1731 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1732 sp<RecordTrack> keep(this); 1733 { 1734 sp<ThreadBase> thread = mThread.promote(); 1735 if (thread != 0) { 1736 if (mState == ACTIVE || mState == RESUMING) { 1737 AudioSystem::stopInput(thread->id()); 1738 } 1739 AudioSystem::releaseInput(thread->id()); 1740 Mutex::Autolock _l(thread->mLock); 1741 RecordThread *recordThread = (RecordThread *) thread.get(); 1742 recordThread->destroyTrack_l(this); 1743 } 1744 } 1745} 1746 1747 1748/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1749{ 1750 result.append(" Clien Fmt Chn mask Session Step S Serv FrameCount\n"); 1751} 1752 1753void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1754{ 1755 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %05d\n", 1756 (mClient == 0) ? getpid_cached : mClient->pid(), 1757 mFormat, 1758 mChannelMask, 1759 mSessionId, 1760 mStepCount, 1761 mState, 1762 mCblk->server, 1763 mFrameCount); 1764} 1765 1766}; // namespace android 1767