Tracks.cpp revision 4944acb7355b3aa25748fd25945a363a69d65444
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 int clientUid, 72 bool isOut) 73 : RefBase(), 74 mThread(thread), 75 mClient(client), 76 mCblk(NULL), 77 // mBuffer 78 mState(IDLE), 79 mSampleRate(sampleRate), 80 mFormat(format), 81 mChannelMask(channelMask), 82 mChannelCount(popcount(channelMask)), 83 mFrameSize(audio_is_linear_pcm(format) ? 84 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 85 mFrameCount(frameCount), 86 mSessionId(sessionId), 87 mIsOut(isOut), 88 mServerProxy(NULL), 89 mId(android_atomic_inc(&nextTrackId)), 90 mTerminated(false) 91{ 92 // if the caller is us, trust the specified uid 93 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 94 int newclientUid = IPCThreadState::self()->getCallingUid(); 95 if (clientUid != -1 && clientUid != newclientUid) { 96 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 97 } 98 clientUid = newclientUid; 99 } 100 // clientUid contains the uid of the app that is responsible for this track, so we can blame 101 // battery usage on it. 102 mUid = clientUid; 103 104 // client == 0 implies sharedBuffer == 0 105 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 106 107 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 108 sharedBuffer->size()); 109 110 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 111 size_t size = sizeof(audio_track_cblk_t); 112 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 113 if (sharedBuffer == 0) { 114 size += bufferSize; 115 } 116 117 if (client != 0) { 118 mCblkMemory = client->heap()->allocate(size); 119 if (mCblkMemory == 0 || 120 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 121 ALOGE("not enough memory for AudioTrack size=%u", size); 122 client->heap()->dump("AudioTrack"); 123 mCblkMemory.clear(); 124 return; 125 } 126 } else { 127 // this syntax avoids calling the audio_track_cblk_t constructor twice 128 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 129 // assume mCblk != NULL 130 } 131 132 // construct the shared structure in-place. 133 if (mCblk != NULL) { 134 new(mCblk) audio_track_cblk_t(); 135 // clear all buffers 136 if (sharedBuffer == 0) { 137 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 138 memset(mBuffer, 0, bufferSize); 139 } else { 140 mBuffer = sharedBuffer->pointer(); 141#if 0 142 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 143#endif 144 } 145 146#ifdef TEE_SINK 147 if (mTeeSinkTrackEnabled) { 148 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 149 if (pipeFormat != Format_Invalid) { 150 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 151 size_t numCounterOffers = 0; 152 const NBAIO_Format offers[1] = {pipeFormat}; 153 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 154 ALOG_ASSERT(index == 0); 155 PipeReader *pipeReader = new PipeReader(*pipe); 156 numCounterOffers = 0; 157 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 158 ALOG_ASSERT(index == 0); 159 mTeeSink = pipe; 160 mTeeSource = pipeReader; 161 } 162 } 163#endif 164 165 } 166} 167 168AudioFlinger::ThreadBase::TrackBase::~TrackBase() 169{ 170#ifdef TEE_SINK 171 dumpTee(-1, mTeeSource, mId); 172#endif 173 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 174 delete mServerProxy; 175 if (mCblk != NULL) { 176 if (mClient == 0) { 177 delete mCblk; 178 } else { 179 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 180 } 181 } 182 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 183 if (mClient != 0) { 184 // Client destructor must run with AudioFlinger mutex locked 185 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 186 // If the client's reference count drops to zero, the associated destructor 187 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 188 // relying on the automatic clear() at end of scope. 189 mClient.clear(); 190 } 191} 192 193// AudioBufferProvider interface 194// getNextBuffer() = 0; 195// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 196void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 197{ 198#ifdef TEE_SINK 199 if (mTeeSink != 0) { 200 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 201 } 202#endif 203 204 ServerProxy::Buffer buf; 205 buf.mFrameCount = buffer->frameCount; 206 buf.mRaw = buffer->raw; 207 buffer->frameCount = 0; 208 buffer->raw = NULL; 209 mServerProxy->releaseBuffer(&buf); 210} 211 212status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 213{ 214 mSyncEvents.add(event); 215 return NO_ERROR; 216} 217 218// ---------------------------------------------------------------------------- 219// Playback 220// ---------------------------------------------------------------------------- 221 222AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 223 : BnAudioTrack(), 224 mTrack(track) 225{ 226} 227 228AudioFlinger::TrackHandle::~TrackHandle() { 229 // just stop the track on deletion, associated resources 230 // will be freed from the main thread once all pending buffers have 231 // been played. Unless it's not in the active track list, in which 232 // case we free everything now... 233 mTrack->destroy(); 234} 235 236sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 237 return mTrack->getCblk(); 238} 239 240status_t AudioFlinger::TrackHandle::start() { 241 return mTrack->start(); 242} 243 244void AudioFlinger::TrackHandle::stop() { 245 mTrack->stop(); 246} 247 248void AudioFlinger::TrackHandle::flush() { 249 mTrack->flush(); 250} 251 252void AudioFlinger::TrackHandle::pause() { 253 mTrack->pause(); 254} 255 256status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 257{ 258 return mTrack->attachAuxEffect(EffectId); 259} 260 261status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 262 sp<IMemory>* buffer) { 263 if (!mTrack->isTimedTrack()) 264 return INVALID_OPERATION; 265 266 PlaybackThread::TimedTrack* tt = 267 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 268 return tt->allocateTimedBuffer(size, buffer); 269} 270 271status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 272 int64_t pts) { 273 if (!mTrack->isTimedTrack()) 274 return INVALID_OPERATION; 275 276 if (buffer == 0 || buffer->pointer() == NULL) { 277 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 278 return BAD_VALUE; 279 } 280 281 PlaybackThread::TimedTrack* tt = 282 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 283 return tt->queueTimedBuffer(buffer, pts); 284} 285 286status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 287 const LinearTransform& xform, int target) { 288 289 if (!mTrack->isTimedTrack()) 290 return INVALID_OPERATION; 291 292 PlaybackThread::TimedTrack* tt = 293 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 294 return tt->setMediaTimeTransform( 295 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 296} 297 298status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 299 return mTrack->setParameters(keyValuePairs); 300} 301 302status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 303{ 304 return mTrack->getTimestamp(timestamp); 305} 306 307 308void AudioFlinger::TrackHandle::signal() 309{ 310 return mTrack->signal(); 311} 312 313status_t AudioFlinger::TrackHandle::onTransact( 314 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 315{ 316 return BnAudioTrack::onTransact(code, data, reply, flags); 317} 318 319// ---------------------------------------------------------------------------- 320 321// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 322AudioFlinger::PlaybackThread::Track::Track( 323 PlaybackThread *thread, 324 const sp<Client>& client, 325 audio_stream_type_t streamType, 326 uint32_t sampleRate, 327 audio_format_t format, 328 audio_channel_mask_t channelMask, 329 size_t frameCount, 330 const sp<IMemory>& sharedBuffer, 331 int sessionId, 332 int uid, 333 IAudioFlinger::track_flags_t flags) 334 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 335 sessionId, uid, true /*isOut*/), 336 mFillingUpStatus(FS_INVALID), 337 // mRetryCount initialized later when needed 338 mSharedBuffer(sharedBuffer), 339 mStreamType(streamType), 340 mName(-1), // see note below 341 mMainBuffer(thread->mixBuffer()), 342 mAuxBuffer(NULL), 343 mAuxEffectId(0), mHasVolumeController(false), 344 mPresentationCompleteFrames(0), 345 mFlags(flags), 346 mFastIndex(-1), 347 mCachedVolume(1.0), 348 mIsInvalid(false), 349 mAudioTrackServerProxy(NULL), 350 mResumeToStopping(false), 351 mFlushHwPending(false) 352{ 353 if (mCblk != NULL) { 354 if (sharedBuffer == 0) { 355 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 356 mFrameSize); 357 } else { 358 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 359 mFrameSize); 360 } 361 mServerProxy = mAudioTrackServerProxy; 362 // to avoid leaking a track name, do not allocate one unless there is an mCblk 363 mName = thread->getTrackName_l(channelMask, sessionId); 364 if (mName < 0) { 365 ALOGE("no more track names available"); 366 return; 367 } 368 // only allocate a fast track index if we were able to allocate a normal track name 369 if (flags & IAudioFlinger::TRACK_FAST) { 370 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 371 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 372 int i = __builtin_ctz(thread->mFastTrackAvailMask); 373 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 374 // FIXME This is too eager. We allocate a fast track index before the 375 // fast track becomes active. Since fast tracks are a scarce resource, 376 // this means we are potentially denying other more important fast tracks from 377 // being created. It would be better to allocate the index dynamically. 378 mFastIndex = i; 379 // Read the initial underruns because this field is never cleared by the fast mixer 380 mObservedUnderruns = thread->getFastTrackUnderruns(i); 381 thread->mFastTrackAvailMask &= ~(1 << i); 382 } 383 } 384 ALOGV("Track constructor name %d, calling pid %d", mName, 385 IPCThreadState::self()->getCallingPid()); 386} 387 388AudioFlinger::PlaybackThread::Track::~Track() 389{ 390 ALOGV("PlaybackThread::Track destructor"); 391 392 // The destructor would clear mSharedBuffer, 393 // but it will not push the decremented reference count, 394 // leaving the client's IMemory dangling indefinitely. 395 // This prevents that leak. 396 if (mSharedBuffer != 0) { 397 mSharedBuffer.clear(); 398 // flush the binder command buffer 399 IPCThreadState::self()->flushCommands(); 400 } 401} 402 403status_t AudioFlinger::PlaybackThread::Track::initCheck() const 404{ 405 status_t status = TrackBase::initCheck(); 406 if (status == NO_ERROR && mName < 0) { 407 status = NO_MEMORY; 408 } 409 return status; 410} 411 412void AudioFlinger::PlaybackThread::Track::destroy() 413{ 414 // NOTE: destroyTrack_l() can remove a strong reference to this Track 415 // by removing it from mTracks vector, so there is a risk that this Tracks's 416 // destructor is called. As the destructor needs to lock mLock, 417 // we must acquire a strong reference on this Track before locking mLock 418 // here so that the destructor is called only when exiting this function. 419 // On the other hand, as long as Track::destroy() is only called by 420 // TrackHandle destructor, the TrackHandle still holds a strong ref on 421 // this Track with its member mTrack. 422 sp<Track> keep(this); 423 { // scope for mLock 424 sp<ThreadBase> thread = mThread.promote(); 425 if (thread != 0) { 426 Mutex::Autolock _l(thread->mLock); 427 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 428 bool wasActive = playbackThread->destroyTrack_l(this); 429 if (!isOutputTrack() && !wasActive) { 430 AudioSystem::releaseOutput(thread->id()); 431 } 432 } 433 } 434} 435 436/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 437{ 438 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 439 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 440} 441 442void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 443{ 444 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 445 if (isFastTrack()) { 446 sprintf(buffer, " F %2d", mFastIndex); 447 } else { 448 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 449 } 450 track_state state = mState; 451 char stateChar; 452 if (isTerminated()) { 453 stateChar = 'T'; 454 } else { 455 switch (state) { 456 case IDLE: 457 stateChar = 'I'; 458 break; 459 case STOPPING_1: 460 stateChar = 's'; 461 break; 462 case STOPPING_2: 463 stateChar = '5'; 464 break; 465 case STOPPED: 466 stateChar = 'S'; 467 break; 468 case RESUMING: 469 stateChar = 'R'; 470 break; 471 case ACTIVE: 472 stateChar = 'A'; 473 break; 474 case PAUSING: 475 stateChar = 'p'; 476 break; 477 case PAUSED: 478 stateChar = 'P'; 479 break; 480 case FLUSHED: 481 stateChar = 'F'; 482 break; 483 default: 484 stateChar = '?'; 485 break; 486 } 487 } 488 char nowInUnderrun; 489 switch (mObservedUnderruns.mBitFields.mMostRecent) { 490 case UNDERRUN_FULL: 491 nowInUnderrun = ' '; 492 break; 493 case UNDERRUN_PARTIAL: 494 nowInUnderrun = '<'; 495 break; 496 case UNDERRUN_EMPTY: 497 nowInUnderrun = '*'; 498 break; 499 default: 500 nowInUnderrun = '?'; 501 break; 502 } 503 snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 504 "%08X %08X %08X 0x%03X %9u%c\n", 505 (mClient == 0) ? getpid_cached : mClient->pid(), 506 mStreamType, 507 mFormat, 508 mChannelMask, 509 mSessionId, 510 mFrameCount, 511 stateChar, 512 mFillingUpStatus, 513 mAudioTrackServerProxy->getSampleRate(), 514 20.0 * log10((vlr & 0xFFFF) / 4096.0), 515 20.0 * log10((vlr >> 16) / 4096.0), 516 mCblk->mServer, 517 (int)mMainBuffer, 518 (int)mAuxBuffer, 519 mCblk->mFlags, 520 mAudioTrackServerProxy->getUnderrunFrames(), 521 nowInUnderrun); 522} 523 524uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 525 return mAudioTrackServerProxy->getSampleRate(); 526} 527 528// AudioBufferProvider interface 529status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 530 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 531{ 532 ServerProxy::Buffer buf; 533 size_t desiredFrames = buffer->frameCount; 534 buf.mFrameCount = desiredFrames; 535 status_t status = mServerProxy->obtainBuffer(&buf); 536 buffer->frameCount = buf.mFrameCount; 537 buffer->raw = buf.mRaw; 538 if (buf.mFrameCount == 0) { 539 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 540 } 541 return status; 542} 543 544// releaseBuffer() is not overridden 545 546// ExtendedAudioBufferProvider interface 547 548// Note that framesReady() takes a mutex on the control block using tryLock(). 549// This could result in priority inversion if framesReady() is called by the normal mixer, 550// as the normal mixer thread runs at lower 551// priority than the client's callback thread: there is a short window within framesReady() 552// during which the normal mixer could be preempted, and the client callback would block. 553// Another problem can occur if framesReady() is called by the fast mixer: 554// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 555// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 556size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 557 return mAudioTrackServerProxy->framesReady(); 558} 559 560size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 561{ 562 return mAudioTrackServerProxy->framesReleased(); 563} 564 565// Don't call for fast tracks; the framesReady() could result in priority inversion 566bool AudioFlinger::PlaybackThread::Track::isReady() const { 567 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) { 568 return true; 569 } 570 571 if (framesReady() >= mFrameCount || 572 (mCblk->mFlags & CBLK_FORCEREADY)) { 573 mFillingUpStatus = FS_FILLED; 574 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 575 return true; 576 } 577 return false; 578} 579 580status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 581 int triggerSession __unused) 582{ 583 status_t status = NO_ERROR; 584 ALOGV("start(%d), calling pid %d session %d", 585 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 586 587 sp<ThreadBase> thread = mThread.promote(); 588 if (thread != 0) { 589 if (isOffloaded()) { 590 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 591 Mutex::Autolock _lth(thread->mLock); 592 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 593 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 594 (ec != 0 && ec->isNonOffloadableEnabled())) { 595 invalidate(); 596 return PERMISSION_DENIED; 597 } 598 } 599 Mutex::Autolock _lth(thread->mLock); 600 track_state state = mState; 601 // here the track could be either new, or restarted 602 // in both cases "unstop" the track 603 604 if (state == PAUSED) { 605 if (mResumeToStopping) { 606 // happened we need to resume to STOPPING_1 607 mState = TrackBase::STOPPING_1; 608 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 609 } else { 610 mState = TrackBase::RESUMING; 611 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 612 } 613 } else { 614 mState = TrackBase::ACTIVE; 615 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 616 } 617 618 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 619 status = playbackThread->addTrack_l(this); 620 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 621 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 622 // restore previous state if start was rejected by policy manager 623 if (status == PERMISSION_DENIED) { 624 mState = state; 625 } 626 } 627 // track was already in the active list, not a problem 628 if (status == ALREADY_EXISTS) { 629 status = NO_ERROR; 630 } else { 631 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 632 // It is usually unsafe to access the server proxy from a binder thread. 633 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 634 // isn't looking at this track yet: we still hold the normal mixer thread lock, 635 // and for fast tracks the track is not yet in the fast mixer thread's active set. 636 ServerProxy::Buffer buffer; 637 buffer.mFrameCount = 1; 638 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 639 } 640 } else { 641 status = BAD_VALUE; 642 } 643 return status; 644} 645 646void AudioFlinger::PlaybackThread::Track::stop() 647{ 648 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 649 sp<ThreadBase> thread = mThread.promote(); 650 if (thread != 0) { 651 Mutex::Autolock _l(thread->mLock); 652 track_state state = mState; 653 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 654 // If the track is not active (PAUSED and buffers full), flush buffers 655 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 656 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 657 reset(); 658 mState = STOPPED; 659 } else if (!isFastTrack() && !isOffloaded()) { 660 mState = STOPPED; 661 } else { 662 // For fast tracks prepareTracks_l() will set state to STOPPING_2 663 // presentation is complete 664 // For an offloaded track this starts a drain and state will 665 // move to STOPPING_2 when drain completes and then STOPPED 666 mState = STOPPING_1; 667 } 668 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 669 playbackThread); 670 } 671 } 672} 673 674void AudioFlinger::PlaybackThread::Track::pause() 675{ 676 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 677 sp<ThreadBase> thread = mThread.promote(); 678 if (thread != 0) { 679 Mutex::Autolock _l(thread->mLock); 680 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 681 switch (mState) { 682 case STOPPING_1: 683 case STOPPING_2: 684 if (!isOffloaded()) { 685 /* nothing to do if track is not offloaded */ 686 break; 687 } 688 689 // Offloaded track was draining, we need to carry on draining when resumed 690 mResumeToStopping = true; 691 // fall through... 692 case ACTIVE: 693 case RESUMING: 694 mState = PAUSING; 695 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 696 playbackThread->broadcast_l(); 697 break; 698 699 default: 700 break; 701 } 702 } 703} 704 705void AudioFlinger::PlaybackThread::Track::flush() 706{ 707 ALOGV("flush(%d)", mName); 708 sp<ThreadBase> thread = mThread.promote(); 709 if (thread != 0) { 710 Mutex::Autolock _l(thread->mLock); 711 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 712 713 if (isOffloaded()) { 714 // If offloaded we allow flush during any state except terminated 715 // and keep the track active to avoid problems if user is seeking 716 // rapidly and underlying hardware has a significant delay handling 717 // a pause 718 if (isTerminated()) { 719 return; 720 } 721 722 ALOGV("flush: offload flush"); 723 reset(); 724 725 if (mState == STOPPING_1 || mState == STOPPING_2) { 726 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 727 mState = ACTIVE; 728 } 729 730 if (mState == ACTIVE) { 731 ALOGV("flush called in active state, resetting buffer time out retry count"); 732 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 733 } 734 735 mFlushHwPending = true; 736 mResumeToStopping = false; 737 } else { 738 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 739 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 740 return; 741 } 742 // No point remaining in PAUSED state after a flush => go to 743 // FLUSHED state 744 mState = FLUSHED; 745 // do not reset the track if it is still in the process of being stopped or paused. 746 // this will be done by prepareTracks_l() when the track is stopped. 747 // prepareTracks_l() will see mState == FLUSHED, then 748 // remove from active track list, reset(), and trigger presentation complete 749 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 750 reset(); 751 } 752 } 753 // Prevent flush being lost if the track is flushed and then resumed 754 // before mixer thread can run. This is important when offloading 755 // because the hardware buffer could hold a large amount of audio 756 playbackThread->broadcast_l(); 757 } 758} 759 760// must be called with thread lock held 761void AudioFlinger::PlaybackThread::Track::flushAck() 762{ 763 if (!isOffloaded()) 764 return; 765 766 mFlushHwPending = false; 767} 768 769void AudioFlinger::PlaybackThread::Track::reset() 770{ 771 // Do not reset twice to avoid discarding data written just after a flush and before 772 // the audioflinger thread detects the track is stopped. 773 if (!mResetDone) { 774 // Force underrun condition to avoid false underrun callback until first data is 775 // written to buffer 776 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 777 mFillingUpStatus = FS_FILLING; 778 mResetDone = true; 779 if (mState == FLUSHED) { 780 mState = IDLE; 781 } 782 } 783} 784 785status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 786{ 787 sp<ThreadBase> thread = mThread.promote(); 788 if (thread == 0) { 789 ALOGE("thread is dead"); 790 return FAILED_TRANSACTION; 791 } else if ((thread->type() == ThreadBase::DIRECT) || 792 (thread->type() == ThreadBase::OFFLOAD)) { 793 return thread->setParameters(keyValuePairs); 794 } else { 795 return PERMISSION_DENIED; 796 } 797} 798 799status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 800{ 801 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 802 if (isFastTrack()) { 803 return INVALID_OPERATION; 804 } 805 sp<ThreadBase> thread = mThread.promote(); 806 if (thread == 0) { 807 return INVALID_OPERATION; 808 } 809 Mutex::Autolock _l(thread->mLock); 810 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 811 if (!isOffloaded()) { 812 if (!playbackThread->mLatchQValid) { 813 return INVALID_OPERATION; 814 } 815 uint32_t unpresentedFrames = 816 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 817 playbackThread->mSampleRate; 818 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 819 if (framesWritten < unpresentedFrames) { 820 return INVALID_OPERATION; 821 } 822 timestamp.mPosition = framesWritten - unpresentedFrames; 823 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 824 return NO_ERROR; 825 } 826 827 return playbackThread->getTimestamp_l(timestamp); 828} 829 830status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 831{ 832 status_t status = DEAD_OBJECT; 833 sp<ThreadBase> thread = mThread.promote(); 834 if (thread != 0) { 835 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 836 sp<AudioFlinger> af = mClient->audioFlinger(); 837 838 Mutex::Autolock _l(af->mLock); 839 840 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 841 842 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 843 Mutex::Autolock _dl(playbackThread->mLock); 844 Mutex::Autolock _sl(srcThread->mLock); 845 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 846 if (chain == 0) { 847 return INVALID_OPERATION; 848 } 849 850 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 851 if (effect == 0) { 852 return INVALID_OPERATION; 853 } 854 srcThread->removeEffect_l(effect); 855 status = playbackThread->addEffect_l(effect); 856 if (status != NO_ERROR) { 857 srcThread->addEffect_l(effect); 858 return INVALID_OPERATION; 859 } 860 // removeEffect_l() has stopped the effect if it was active so it must be restarted 861 if (effect->state() == EffectModule::ACTIVE || 862 effect->state() == EffectModule::STOPPING) { 863 effect->start(); 864 } 865 866 sp<EffectChain> dstChain = effect->chain().promote(); 867 if (dstChain == 0) { 868 srcThread->addEffect_l(effect); 869 return INVALID_OPERATION; 870 } 871 AudioSystem::unregisterEffect(effect->id()); 872 AudioSystem::registerEffect(&effect->desc(), 873 srcThread->id(), 874 dstChain->strategy(), 875 AUDIO_SESSION_OUTPUT_MIX, 876 effect->id()); 877 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 878 } 879 status = playbackThread->attachAuxEffect(this, EffectId); 880 } 881 return status; 882} 883 884void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 885{ 886 mAuxEffectId = EffectId; 887 mAuxBuffer = buffer; 888} 889 890bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 891 size_t audioHalFrames) 892{ 893 // a track is considered presented when the total number of frames written to audio HAL 894 // corresponds to the number of frames written when presentationComplete() is called for the 895 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 896 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 897 // to detect when all frames have been played. In this case framesWritten isn't 898 // useful because it doesn't always reflect whether there is data in the h/w 899 // buffers, particularly if a track has been paused and resumed during draining 900 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 901 mPresentationCompleteFrames, framesWritten); 902 if (mPresentationCompleteFrames == 0) { 903 mPresentationCompleteFrames = framesWritten + audioHalFrames; 904 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 905 mPresentationCompleteFrames, audioHalFrames); 906 } 907 908 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 909 ALOGV("presentationComplete() session %d complete: framesWritten %d", 910 mSessionId, framesWritten); 911 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 912 mAudioTrackServerProxy->setStreamEndDone(); 913 return true; 914 } 915 return false; 916} 917 918void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 919{ 920 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 921 if (mSyncEvents[i]->type() == type) { 922 mSyncEvents[i]->trigger(); 923 mSyncEvents.removeAt(i); 924 i--; 925 } 926 } 927} 928 929// implement VolumeBufferProvider interface 930 931uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 932{ 933 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 934 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 935 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 936 uint32_t vl = vlr & 0xFFFF; 937 uint32_t vr = vlr >> 16; 938 // track volumes come from shared memory, so can't be trusted and must be clamped 939 if (vl > MAX_GAIN_INT) { 940 vl = MAX_GAIN_INT; 941 } 942 if (vr > MAX_GAIN_INT) { 943 vr = MAX_GAIN_INT; 944 } 945 // now apply the cached master volume and stream type volume; 946 // this is trusted but lacks any synchronization or barrier so may be stale 947 float v = mCachedVolume; 948 vl *= v; 949 vr *= v; 950 // re-combine into U4.16 951 vlr = (vr << 16) | (vl & 0xFFFF); 952 // FIXME look at mute, pause, and stop flags 953 return vlr; 954} 955 956status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 957{ 958 if (isTerminated() || mState == PAUSED || 959 ((framesReady() == 0) && ((mSharedBuffer != 0) || 960 (mState == STOPPED)))) { 961 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 962 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 963 event->cancel(); 964 return INVALID_OPERATION; 965 } 966 (void) TrackBase::setSyncEvent(event); 967 return NO_ERROR; 968} 969 970void AudioFlinger::PlaybackThread::Track::invalidate() 971{ 972 // FIXME should use proxy, and needs work 973 audio_track_cblk_t* cblk = mCblk; 974 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 975 android_atomic_release_store(0x40000000, &cblk->mFutex); 976 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 977 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 978 mIsInvalid = true; 979} 980 981void AudioFlinger::PlaybackThread::Track::signal() 982{ 983 sp<ThreadBase> thread = mThread.promote(); 984 if (thread != 0) { 985 PlaybackThread *t = (PlaybackThread *)thread.get(); 986 Mutex::Autolock _l(t->mLock); 987 t->broadcast_l(); 988 } 989} 990 991// ---------------------------------------------------------------------------- 992 993sp<AudioFlinger::PlaybackThread::TimedTrack> 994AudioFlinger::PlaybackThread::TimedTrack::create( 995 PlaybackThread *thread, 996 const sp<Client>& client, 997 audio_stream_type_t streamType, 998 uint32_t sampleRate, 999 audio_format_t format, 1000 audio_channel_mask_t channelMask, 1001 size_t frameCount, 1002 const sp<IMemory>& sharedBuffer, 1003 int sessionId, 1004 int uid) 1005{ 1006 if (!client->reserveTimedTrack()) 1007 return 0; 1008 1009 return new TimedTrack( 1010 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1011 sharedBuffer, sessionId, uid); 1012} 1013 1014AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1015 PlaybackThread *thread, 1016 const sp<Client>& client, 1017 audio_stream_type_t streamType, 1018 uint32_t sampleRate, 1019 audio_format_t format, 1020 audio_channel_mask_t channelMask, 1021 size_t frameCount, 1022 const sp<IMemory>& sharedBuffer, 1023 int sessionId, 1024 int uid) 1025 : Track(thread, client, streamType, sampleRate, format, channelMask, 1026 frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED), 1027 mQueueHeadInFlight(false), 1028 mTrimQueueHeadOnRelease(false), 1029 mFramesPendingInQueue(0), 1030 mTimedSilenceBuffer(NULL), 1031 mTimedSilenceBufferSize(0), 1032 mTimedAudioOutputOnTime(false), 1033 mMediaTimeTransformValid(false) 1034{ 1035 LocalClock lc; 1036 mLocalTimeFreq = lc.getLocalFreq(); 1037 1038 mLocalTimeToSampleTransform.a_zero = 0; 1039 mLocalTimeToSampleTransform.b_zero = 0; 1040 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1041 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1042 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1043 &mLocalTimeToSampleTransform.a_to_b_denom); 1044 1045 mMediaTimeToSampleTransform.a_zero = 0; 1046 mMediaTimeToSampleTransform.b_zero = 0; 1047 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1048 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1049 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1050 &mMediaTimeToSampleTransform.a_to_b_denom); 1051} 1052 1053AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1054 mClient->releaseTimedTrack(); 1055 delete [] mTimedSilenceBuffer; 1056} 1057 1058status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1059 size_t size, sp<IMemory>* buffer) { 1060 1061 Mutex::Autolock _l(mTimedBufferQueueLock); 1062 1063 trimTimedBufferQueue_l(); 1064 1065 // lazily initialize the shared memory heap for timed buffers 1066 if (mTimedMemoryDealer == NULL) { 1067 const int kTimedBufferHeapSize = 512 << 10; 1068 1069 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1070 "AudioFlingerTimed"); 1071 if (mTimedMemoryDealer == NULL) { 1072 return NO_MEMORY; 1073 } 1074 } 1075 1076 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1077 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1078 return NO_MEMORY; 1079 } 1080 1081 *buffer = newBuffer; 1082 return NO_ERROR; 1083} 1084 1085// caller must hold mTimedBufferQueueLock 1086void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1087 int64_t mediaTimeNow; 1088 { 1089 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1090 if (!mMediaTimeTransformValid) 1091 return; 1092 1093 int64_t targetTimeNow; 1094 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1095 ? mCCHelper.getCommonTime(&targetTimeNow) 1096 : mCCHelper.getLocalTime(&targetTimeNow); 1097 1098 if (OK != res) 1099 return; 1100 1101 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1102 &mediaTimeNow)) { 1103 return; 1104 } 1105 } 1106 1107 size_t trimEnd; 1108 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1109 int64_t bufEnd; 1110 1111 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1112 // We have a next buffer. Just use its PTS as the PTS of the frame 1113 // following the last frame in this buffer. If the stream is sparse 1114 // (ie, there are deliberate gaps left in the stream which should be 1115 // filled with silence by the TimedAudioTrack), then this can result 1116 // in one extra buffer being left un-trimmed when it could have 1117 // been. In general, this is not typical, and we would rather 1118 // optimized away the TS calculation below for the more common case 1119 // where PTSes are contiguous. 1120 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1121 } else { 1122 // We have no next buffer. Compute the PTS of the frame following 1123 // the last frame in this buffer by computing the duration of of 1124 // this frame in media time units and adding it to the PTS of the 1125 // buffer. 1126 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1127 / mFrameSize; 1128 1129 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1130 &bufEnd)) { 1131 ALOGE("Failed to convert frame count of %lld to media time" 1132 " duration" " (scale factor %d/%u) in %s", 1133 frameCount, 1134 mMediaTimeToSampleTransform.a_to_b_numer, 1135 mMediaTimeToSampleTransform.a_to_b_denom, 1136 __PRETTY_FUNCTION__); 1137 break; 1138 } 1139 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1140 } 1141 1142 if (bufEnd > mediaTimeNow) 1143 break; 1144 1145 // Is the buffer we want to use in the middle of a mix operation right 1146 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1147 // from the mixer which should be coming back shortly. 1148 if (!trimEnd && mQueueHeadInFlight) { 1149 mTrimQueueHeadOnRelease = true; 1150 } 1151 } 1152 1153 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1154 if (trimStart < trimEnd) { 1155 // Update the bookkeeping for framesReady() 1156 for (size_t i = trimStart; i < trimEnd; ++i) { 1157 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1158 } 1159 1160 // Now actually remove the buffers from the queue. 1161 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1162 } 1163} 1164 1165void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1166 const char* logTag) { 1167 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1168 "%s called (reason \"%s\"), but timed buffer queue has no" 1169 " elements to trim.", __FUNCTION__, logTag); 1170 1171 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1172 mTimedBufferQueue.removeAt(0); 1173} 1174 1175void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1176 const TimedBuffer& buf, 1177 const char* logTag __unused) { 1178 uint32_t bufBytes = buf.buffer()->size(); 1179 uint32_t consumedAlready = buf.position(); 1180 1181 ALOG_ASSERT(consumedAlready <= bufBytes, 1182 "Bad bookkeeping while updating frames pending. Timed buffer is" 1183 " only %u bytes long, but claims to have consumed %u" 1184 " bytes. (update reason: \"%s\")", 1185 bufBytes, consumedAlready, logTag); 1186 1187 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1188 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1189 "Bad bookkeeping while updating frames pending. Should have at" 1190 " least %u queued frames, but we think we have only %u. (update" 1191 " reason: \"%s\")", 1192 bufFrames, mFramesPendingInQueue, logTag); 1193 1194 mFramesPendingInQueue -= bufFrames; 1195} 1196 1197status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1198 const sp<IMemory>& buffer, int64_t pts) { 1199 1200 { 1201 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1202 if (!mMediaTimeTransformValid) 1203 return INVALID_OPERATION; 1204 } 1205 1206 Mutex::Autolock _l(mTimedBufferQueueLock); 1207 1208 uint32_t bufFrames = buffer->size() / mFrameSize; 1209 mFramesPendingInQueue += bufFrames; 1210 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1211 1212 return NO_ERROR; 1213} 1214 1215status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1216 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1217 1218 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1219 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1220 target); 1221 1222 if (!(target == TimedAudioTrack::LOCAL_TIME || 1223 target == TimedAudioTrack::COMMON_TIME)) { 1224 return BAD_VALUE; 1225 } 1226 1227 Mutex::Autolock lock(mMediaTimeTransformLock); 1228 mMediaTimeTransform = xform; 1229 mMediaTimeTransformTarget = target; 1230 mMediaTimeTransformValid = true; 1231 1232 return NO_ERROR; 1233} 1234 1235#define min(a, b) ((a) < (b) ? (a) : (b)) 1236 1237// implementation of getNextBuffer for tracks whose buffers have timestamps 1238status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1239 AudioBufferProvider::Buffer* buffer, int64_t pts) 1240{ 1241 if (pts == AudioBufferProvider::kInvalidPTS) { 1242 buffer->raw = NULL; 1243 buffer->frameCount = 0; 1244 mTimedAudioOutputOnTime = false; 1245 return INVALID_OPERATION; 1246 } 1247 1248 Mutex::Autolock _l(mTimedBufferQueueLock); 1249 1250 ALOG_ASSERT(!mQueueHeadInFlight, 1251 "getNextBuffer called without releaseBuffer!"); 1252 1253 while (true) { 1254 1255 // if we have no timed buffers, then fail 1256 if (mTimedBufferQueue.isEmpty()) { 1257 buffer->raw = NULL; 1258 buffer->frameCount = 0; 1259 return NOT_ENOUGH_DATA; 1260 } 1261 1262 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1263 1264 // calculate the PTS of the head of the timed buffer queue expressed in 1265 // local time 1266 int64_t headLocalPTS; 1267 { 1268 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1269 1270 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1271 1272 if (mMediaTimeTransform.a_to_b_denom == 0) { 1273 // the transform represents a pause, so yield silence 1274 timedYieldSilence_l(buffer->frameCount, buffer); 1275 return NO_ERROR; 1276 } 1277 1278 int64_t transformedPTS; 1279 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1280 &transformedPTS)) { 1281 // the transform failed. this shouldn't happen, but if it does 1282 // then just drop this buffer 1283 ALOGW("timedGetNextBuffer transform failed"); 1284 buffer->raw = NULL; 1285 buffer->frameCount = 0; 1286 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1287 return NO_ERROR; 1288 } 1289 1290 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1291 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1292 &headLocalPTS)) { 1293 buffer->raw = NULL; 1294 buffer->frameCount = 0; 1295 return INVALID_OPERATION; 1296 } 1297 } else { 1298 headLocalPTS = transformedPTS; 1299 } 1300 } 1301 1302 uint32_t sr = sampleRate(); 1303 1304 // adjust the head buffer's PTS to reflect the portion of the head buffer 1305 // that has already been consumed 1306 int64_t effectivePTS = headLocalPTS + 1307 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1308 1309 // Calculate the delta in samples between the head of the input buffer 1310 // queue and the start of the next output buffer that will be written. 1311 // If the transformation fails because of over or underflow, it means 1312 // that the sample's position in the output stream is so far out of 1313 // whack that it should just be dropped. 1314 int64_t sampleDelta; 1315 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1316 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1317 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1318 " mix"); 1319 continue; 1320 } 1321 if (!mLocalTimeToSampleTransform.doForwardTransform( 1322 (effectivePTS - pts) << 32, &sampleDelta)) { 1323 ALOGV("*** too late during sample rate transform: dropped buffer"); 1324 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1325 continue; 1326 } 1327 1328 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1329 " sampleDelta=[%d.%08x]", 1330 head.pts(), head.position(), pts, 1331 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1332 + (sampleDelta >> 32)), 1333 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1334 1335 // if the delta between the ideal placement for the next input sample and 1336 // the current output position is within this threshold, then we will 1337 // concatenate the next input samples to the previous output 1338 const int64_t kSampleContinuityThreshold = 1339 (static_cast<int64_t>(sr) << 32) / 250; 1340 1341 // if this is the first buffer of audio that we're emitting from this track 1342 // then it should be almost exactly on time. 1343 const int64_t kSampleStartupThreshold = 1LL << 32; 1344 1345 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1346 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1347 // the next input is close enough to being on time, so concatenate it 1348 // with the last output 1349 timedYieldSamples_l(buffer); 1350 1351 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1352 head.position(), buffer->frameCount); 1353 return NO_ERROR; 1354 } 1355 1356 // Looks like our output is not on time. Reset our on timed status. 1357 // Next time we mix samples from our input queue, then should be within 1358 // the StartupThreshold. 1359 mTimedAudioOutputOnTime = false; 1360 if (sampleDelta > 0) { 1361 // the gap between the current output position and the proper start of 1362 // the next input sample is too big, so fill it with silence 1363 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1364 1365 timedYieldSilence_l(framesUntilNextInput, buffer); 1366 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1367 return NO_ERROR; 1368 } else { 1369 // the next input sample is late 1370 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1371 size_t onTimeSamplePosition = 1372 head.position() + lateFrames * mFrameSize; 1373 1374 if (onTimeSamplePosition > head.buffer()->size()) { 1375 // all the remaining samples in the head are too late, so 1376 // drop it and move on 1377 ALOGV("*** too late: dropped buffer"); 1378 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1379 continue; 1380 } else { 1381 // skip over the late samples 1382 head.setPosition(onTimeSamplePosition); 1383 1384 // yield the available samples 1385 timedYieldSamples_l(buffer); 1386 1387 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1388 return NO_ERROR; 1389 } 1390 } 1391 } 1392} 1393 1394// Yield samples from the timed buffer queue head up to the given output 1395// buffer's capacity. 1396// 1397// Caller must hold mTimedBufferQueueLock 1398void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1399 AudioBufferProvider::Buffer* buffer) { 1400 1401 const TimedBuffer& head = mTimedBufferQueue[0]; 1402 1403 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1404 head.position()); 1405 1406 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1407 mFrameSize); 1408 size_t framesRequested = buffer->frameCount; 1409 buffer->frameCount = min(framesLeftInHead, framesRequested); 1410 1411 mQueueHeadInFlight = true; 1412 mTimedAudioOutputOnTime = true; 1413} 1414 1415// Yield samples of silence up to the given output buffer's capacity 1416// 1417// Caller must hold mTimedBufferQueueLock 1418void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1419 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1420 1421 // lazily allocate a buffer filled with silence 1422 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1423 delete [] mTimedSilenceBuffer; 1424 mTimedSilenceBufferSize = numFrames * mFrameSize; 1425 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1426 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1427 } 1428 1429 buffer->raw = mTimedSilenceBuffer; 1430 size_t framesRequested = buffer->frameCount; 1431 buffer->frameCount = min(numFrames, framesRequested); 1432 1433 mTimedAudioOutputOnTime = false; 1434} 1435 1436// AudioBufferProvider interface 1437void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1438 AudioBufferProvider::Buffer* buffer) { 1439 1440 Mutex::Autolock _l(mTimedBufferQueueLock); 1441 1442 // If the buffer which was just released is part of the buffer at the head 1443 // of the queue, be sure to update the amt of the buffer which has been 1444 // consumed. If the buffer being returned is not part of the head of the 1445 // queue, its either because the buffer is part of the silence buffer, or 1446 // because the head of the timed queue was trimmed after the mixer called 1447 // getNextBuffer but before the mixer called releaseBuffer. 1448 if (buffer->raw == mTimedSilenceBuffer) { 1449 ALOG_ASSERT(!mQueueHeadInFlight, 1450 "Queue head in flight during release of silence buffer!"); 1451 goto done; 1452 } 1453 1454 ALOG_ASSERT(mQueueHeadInFlight, 1455 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1456 " head in flight."); 1457 1458 if (mTimedBufferQueue.size()) { 1459 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1460 1461 void* start = head.buffer()->pointer(); 1462 void* end = reinterpret_cast<void*>( 1463 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1464 + head.buffer()->size()); 1465 1466 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1467 "released buffer not within the head of the timed buffer" 1468 " queue; qHead = [%p, %p], released buffer = %p", 1469 start, end, buffer->raw); 1470 1471 head.setPosition(head.position() + 1472 (buffer->frameCount * mFrameSize)); 1473 mQueueHeadInFlight = false; 1474 1475 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1476 "Bad bookkeeping during releaseBuffer! Should have at" 1477 " least %u queued frames, but we think we have only %u", 1478 buffer->frameCount, mFramesPendingInQueue); 1479 1480 mFramesPendingInQueue -= buffer->frameCount; 1481 1482 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1483 || mTrimQueueHeadOnRelease) { 1484 trimTimedBufferQueueHead_l("releaseBuffer"); 1485 mTrimQueueHeadOnRelease = false; 1486 } 1487 } else { 1488 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1489 " buffers in the timed buffer queue"); 1490 } 1491 1492done: 1493 buffer->raw = 0; 1494 buffer->frameCount = 0; 1495} 1496 1497size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1498 Mutex::Autolock _l(mTimedBufferQueueLock); 1499 return mFramesPendingInQueue; 1500} 1501 1502AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1503 : mPTS(0), mPosition(0) {} 1504 1505AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1506 const sp<IMemory>& buffer, int64_t pts) 1507 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1508 1509 1510// ---------------------------------------------------------------------------- 1511 1512AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1513 PlaybackThread *playbackThread, 1514 DuplicatingThread *sourceThread, 1515 uint32_t sampleRate, 1516 audio_format_t format, 1517 audio_channel_mask_t channelMask, 1518 size_t frameCount, 1519 int uid) 1520 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1521 NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT), 1522 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1523{ 1524 1525 if (mCblk != NULL) { 1526 mOutBuffer.frameCount = 0; 1527 playbackThread->mTracks.add(this); 1528 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1529 "frameCount %u, mChannelMask 0x%08x", 1530 mCblk, mBuffer, 1531 frameCount, mChannelMask); 1532 // since client and server are in the same process, 1533 // the buffer has the same virtual address on both sides 1534 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1535 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1536 mClientProxy->setSendLevel(0.0); 1537 mClientProxy->setSampleRate(sampleRate); 1538 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1539 true /*clientInServer*/); 1540 } else { 1541 ALOGW("Error creating output track on thread %p", playbackThread); 1542 } 1543} 1544 1545AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1546{ 1547 clearBufferQueue(); 1548 delete mClientProxy; 1549 // superclass destructor will now delete the server proxy and shared memory both refer to 1550} 1551 1552status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1553 int triggerSession) 1554{ 1555 status_t status = Track::start(event, triggerSession); 1556 if (status != NO_ERROR) { 1557 return status; 1558 } 1559 1560 mActive = true; 1561 mRetryCount = 127; 1562 return status; 1563} 1564 1565void AudioFlinger::PlaybackThread::OutputTrack::stop() 1566{ 1567 Track::stop(); 1568 clearBufferQueue(); 1569 mOutBuffer.frameCount = 0; 1570 mActive = false; 1571} 1572 1573bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1574{ 1575 Buffer *pInBuffer; 1576 Buffer inBuffer; 1577 uint32_t channelCount = mChannelCount; 1578 bool outputBufferFull = false; 1579 inBuffer.frameCount = frames; 1580 inBuffer.i16 = data; 1581 1582 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1583 1584 if (!mActive && frames != 0) { 1585 start(); 1586 sp<ThreadBase> thread = mThread.promote(); 1587 if (thread != 0) { 1588 MixerThread *mixerThread = (MixerThread *)thread.get(); 1589 if (mFrameCount > frames) { 1590 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1591 uint32_t startFrames = (mFrameCount - frames); 1592 pInBuffer = new Buffer; 1593 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1594 pInBuffer->frameCount = startFrames; 1595 pInBuffer->i16 = pInBuffer->mBuffer; 1596 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1597 mBufferQueue.add(pInBuffer); 1598 } else { 1599 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1600 } 1601 } 1602 } 1603 } 1604 1605 while (waitTimeLeftMs) { 1606 // First write pending buffers, then new data 1607 if (mBufferQueue.size()) { 1608 pInBuffer = mBufferQueue.itemAt(0); 1609 } else { 1610 pInBuffer = &inBuffer; 1611 } 1612 1613 if (pInBuffer->frameCount == 0) { 1614 break; 1615 } 1616 1617 if (mOutBuffer.frameCount == 0) { 1618 mOutBuffer.frameCount = pInBuffer->frameCount; 1619 nsecs_t startTime = systemTime(); 1620 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1621 if (status != NO_ERROR) { 1622 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1623 mThread.unsafe_get(), status); 1624 outputBufferFull = true; 1625 break; 1626 } 1627 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1628 if (waitTimeLeftMs >= waitTimeMs) { 1629 waitTimeLeftMs -= waitTimeMs; 1630 } else { 1631 waitTimeLeftMs = 0; 1632 } 1633 } 1634 1635 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1636 pInBuffer->frameCount; 1637 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1638 Proxy::Buffer buf; 1639 buf.mFrameCount = outFrames; 1640 buf.mRaw = NULL; 1641 mClientProxy->releaseBuffer(&buf); 1642 pInBuffer->frameCount -= outFrames; 1643 pInBuffer->i16 += outFrames * channelCount; 1644 mOutBuffer.frameCount -= outFrames; 1645 mOutBuffer.i16 += outFrames * channelCount; 1646 1647 if (pInBuffer->frameCount == 0) { 1648 if (mBufferQueue.size()) { 1649 mBufferQueue.removeAt(0); 1650 delete [] pInBuffer->mBuffer; 1651 delete pInBuffer; 1652 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1653 mThread.unsafe_get(), mBufferQueue.size()); 1654 } else { 1655 break; 1656 } 1657 } 1658 } 1659 1660 // If we could not write all frames, allocate a buffer and queue it for next time. 1661 if (inBuffer.frameCount) { 1662 sp<ThreadBase> thread = mThread.promote(); 1663 if (thread != 0 && !thread->standby()) { 1664 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1665 pInBuffer = new Buffer; 1666 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1667 pInBuffer->frameCount = inBuffer.frameCount; 1668 pInBuffer->i16 = pInBuffer->mBuffer; 1669 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1670 sizeof(int16_t)); 1671 mBufferQueue.add(pInBuffer); 1672 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1673 mThread.unsafe_get(), mBufferQueue.size()); 1674 } else { 1675 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1676 mThread.unsafe_get(), this); 1677 } 1678 } 1679 } 1680 1681 // Calling write() with a 0 length buffer, means that no more data will be written: 1682 // If no more buffers are pending, fill output track buffer to make sure it is started 1683 // by output mixer. 1684 if (frames == 0 && mBufferQueue.size() == 0) { 1685 // FIXME borken, replace by getting framesReady() from proxy 1686 size_t user = 0; // was mCblk->user 1687 if (user < mFrameCount) { 1688 frames = mFrameCount - user; 1689 pInBuffer = new Buffer; 1690 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1691 pInBuffer->frameCount = frames; 1692 pInBuffer->i16 = pInBuffer->mBuffer; 1693 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1694 mBufferQueue.add(pInBuffer); 1695 } else if (mActive) { 1696 stop(); 1697 } 1698 } 1699 1700 return outputBufferFull; 1701} 1702 1703status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1704 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1705{ 1706 ClientProxy::Buffer buf; 1707 buf.mFrameCount = buffer->frameCount; 1708 struct timespec timeout; 1709 timeout.tv_sec = waitTimeMs / 1000; 1710 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1711 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1712 buffer->frameCount = buf.mFrameCount; 1713 buffer->raw = buf.mRaw; 1714 return status; 1715} 1716 1717void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1718{ 1719 size_t size = mBufferQueue.size(); 1720 1721 for (size_t i = 0; i < size; i++) { 1722 Buffer *pBuffer = mBufferQueue.itemAt(i); 1723 delete [] pBuffer->mBuffer; 1724 delete pBuffer; 1725 } 1726 mBufferQueue.clear(); 1727} 1728 1729 1730// ---------------------------------------------------------------------------- 1731// Record 1732// ---------------------------------------------------------------------------- 1733 1734AudioFlinger::RecordHandle::RecordHandle( 1735 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1736 : BnAudioRecord(), 1737 mRecordTrack(recordTrack) 1738{ 1739} 1740 1741AudioFlinger::RecordHandle::~RecordHandle() { 1742 stop_nonvirtual(); 1743 mRecordTrack->destroy(); 1744} 1745 1746sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1747 return mRecordTrack->getCblk(); 1748} 1749 1750status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1751 int triggerSession) { 1752 ALOGV("RecordHandle::start()"); 1753 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1754} 1755 1756void AudioFlinger::RecordHandle::stop() { 1757 stop_nonvirtual(); 1758} 1759 1760void AudioFlinger::RecordHandle::stop_nonvirtual() { 1761 ALOGV("RecordHandle::stop()"); 1762 mRecordTrack->stop(); 1763} 1764 1765status_t AudioFlinger::RecordHandle::onTransact( 1766 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1767{ 1768 return BnAudioRecord::onTransact(code, data, reply, flags); 1769} 1770 1771// ---------------------------------------------------------------------------- 1772 1773// RecordTrack constructor must be called with AudioFlinger::mLock held 1774AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1775 RecordThread *thread, 1776 const sp<Client>& client, 1777 uint32_t sampleRate, 1778 audio_format_t format, 1779 audio_channel_mask_t channelMask, 1780 size_t frameCount, 1781 int sessionId, 1782 int uid) 1783 : TrackBase(thread, client, sampleRate, format, 1784 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/), 1785 mOverflow(false) 1786{ 1787 ALOGV("RecordTrack constructor"); 1788 if (mCblk != NULL) { 1789 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1790 } 1791} 1792 1793AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1794{ 1795 ALOGV("%s", __func__); 1796} 1797 1798// AudioBufferProvider interface 1799status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1800 int64_t pts __unused) 1801{ 1802 ServerProxy::Buffer buf; 1803 buf.mFrameCount = buffer->frameCount; 1804 status_t status = mServerProxy->obtainBuffer(&buf); 1805 buffer->frameCount = buf.mFrameCount; 1806 buffer->raw = buf.mRaw; 1807 if (buf.mFrameCount == 0) { 1808 // FIXME also wake futex so that overrun is noticed more quickly 1809 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1810 } 1811 return status; 1812} 1813 1814status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1815 int triggerSession) 1816{ 1817 sp<ThreadBase> thread = mThread.promote(); 1818 if (thread != 0) { 1819 RecordThread *recordThread = (RecordThread *)thread.get(); 1820 return recordThread->start(this, event, triggerSession); 1821 } else { 1822 return BAD_VALUE; 1823 } 1824} 1825 1826void AudioFlinger::RecordThread::RecordTrack::stop() 1827{ 1828 sp<ThreadBase> thread = mThread.promote(); 1829 if (thread != 0) { 1830 RecordThread *recordThread = (RecordThread *)thread.get(); 1831 if (recordThread->stop(this)) { 1832 AudioSystem::stopInput(recordThread->id()); 1833 } 1834 } 1835} 1836 1837void AudioFlinger::RecordThread::RecordTrack::destroy() 1838{ 1839 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1840 sp<RecordTrack> keep(this); 1841 { 1842 sp<ThreadBase> thread = mThread.promote(); 1843 if (thread != 0) { 1844 if (mState == ACTIVE || mState == RESUMING) { 1845 AudioSystem::stopInput(thread->id()); 1846 } 1847 AudioSystem::releaseInput(thread->id()); 1848 Mutex::Autolock _l(thread->mLock); 1849 RecordThread *recordThread = (RecordThread *) thread.get(); 1850 recordThread->destroyTrack_l(this); 1851 } 1852 } 1853} 1854 1855void AudioFlinger::RecordThread::RecordTrack::invalidate() 1856{ 1857 // FIXME should use proxy, and needs work 1858 audio_track_cblk_t* cblk = mCblk; 1859 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1860 android_atomic_release_store(0x40000000, &cblk->mFutex); 1861 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1862 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1863} 1864 1865 1866/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1867{ 1868 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1869} 1870 1871void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1872{ 1873 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1874 (mClient == 0) ? getpid_cached : mClient->pid(), 1875 mFormat, 1876 mChannelMask, 1877 mSessionId, 1878 mState, 1879 mCblk->mServer, 1880 mFrameCount); 1881} 1882 1883}; // namespace android 1884