Tracks.cpp revision 2d3ca68363f723fbe269d3ce52dab4985dfc7154
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 (Format_isValid(pipeFormat)) { 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 return; 355 } 356 357 if (sharedBuffer == 0) { 358 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 359 mFrameSize); 360 } else { 361 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 362 mFrameSize); 363 } 364 mServerProxy = mAudioTrackServerProxy; 365 366 mName = thread->getTrackName_l(channelMask, sessionId); 367 if (mName < 0) { 368 ALOGE("no more track names available"); 369 return; 370 } 371 // only allocate a fast track index if we were able to allocate a normal track name 372 if (flags & IAudioFlinger::TRACK_FAST) { 373 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 374 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 375 int i = __builtin_ctz(thread->mFastTrackAvailMask); 376 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 377 // FIXME This is too eager. We allocate a fast track index before the 378 // fast track becomes active. Since fast tracks are a scarce resource, 379 // this means we are potentially denying other more important fast tracks from 380 // being created. It would be better to allocate the index dynamically. 381 mFastIndex = i; 382 // Read the initial underruns because this field is never cleared by the fast mixer 383 mObservedUnderruns = thread->getFastTrackUnderruns(i); 384 thread->mFastTrackAvailMask &= ~(1 << i); 385 } 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 Active 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, bool active) 443{ 444 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 445 if (isFastTrack()) { 446 sprintf(buffer, " F %2d", mFastIndex); 447 } else if (mName >= AudioMixer::TRACK0) { 448 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 449 } else { 450 sprintf(buffer, " none"); 451 } 452 track_state state = mState; 453 char stateChar; 454 if (isTerminated()) { 455 stateChar = 'T'; 456 } else { 457 switch (state) { 458 case IDLE: 459 stateChar = 'I'; 460 break; 461 case STOPPING_1: 462 stateChar = 's'; 463 break; 464 case STOPPING_2: 465 stateChar = '5'; 466 break; 467 case STOPPED: 468 stateChar = 'S'; 469 break; 470 case RESUMING: 471 stateChar = 'R'; 472 break; 473 case ACTIVE: 474 stateChar = 'A'; 475 break; 476 case PAUSING: 477 stateChar = 'p'; 478 break; 479 case PAUSED: 480 stateChar = 'P'; 481 break; 482 case FLUSHED: 483 stateChar = 'F'; 484 break; 485 default: 486 stateChar = '?'; 487 break; 488 } 489 } 490 char nowInUnderrun; 491 switch (mObservedUnderruns.mBitFields.mMostRecent) { 492 case UNDERRUN_FULL: 493 nowInUnderrun = ' '; 494 break; 495 case UNDERRUN_PARTIAL: 496 nowInUnderrun = '<'; 497 break; 498 case UNDERRUN_EMPTY: 499 nowInUnderrun = '*'; 500 break; 501 default: 502 nowInUnderrun = '?'; 503 break; 504 } 505 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 506 "%08X %p %p 0x%03X %9u%c\n", 507 active ? "yes" : "no", 508 (mClient == 0) ? getpid_cached : mClient->pid(), 509 mStreamType, 510 mFormat, 511 mChannelMask, 512 mSessionId, 513 mFrameCount, 514 stateChar, 515 mFillingUpStatus, 516 mAudioTrackServerProxy->getSampleRate(), 517 20.0 * log10((vlr & 0xFFFF) / 4096.0), 518 20.0 * log10((vlr >> 16) / 4096.0), 519 mCblk->mServer, 520 mMainBuffer, 521 mAuxBuffer, 522 mCblk->mFlags, 523 mAudioTrackServerProxy->getUnderrunFrames(), 524 nowInUnderrun); 525} 526 527uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 528 return mAudioTrackServerProxy->getSampleRate(); 529} 530 531// AudioBufferProvider interface 532status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 533 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 534{ 535 ServerProxy::Buffer buf; 536 size_t desiredFrames = buffer->frameCount; 537 buf.mFrameCount = desiredFrames; 538 status_t status = mServerProxy->obtainBuffer(&buf); 539 buffer->frameCount = buf.mFrameCount; 540 buffer->raw = buf.mRaw; 541 if (buf.mFrameCount == 0) { 542 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 543 } 544 return status; 545} 546 547// releaseBuffer() is not overridden 548 549// ExtendedAudioBufferProvider interface 550 551// Note that framesReady() takes a mutex on the control block using tryLock(). 552// This could result in priority inversion if framesReady() is called by the normal mixer, 553// as the normal mixer thread runs at lower 554// priority than the client's callback thread: there is a short window within framesReady() 555// during which the normal mixer could be preempted, and the client callback would block. 556// Another problem can occur if framesReady() is called by the fast mixer: 557// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 558// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 559size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 560 return mAudioTrackServerProxy->framesReady(); 561} 562 563size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 564{ 565 return mAudioTrackServerProxy->framesReleased(); 566} 567 568// Don't call for fast tracks; the framesReady() could result in priority inversion 569bool AudioFlinger::PlaybackThread::Track::isReady() const { 570 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 571 return true; 572 } 573 574 if (isStopping()) { 575 if (framesReady() > 0) { 576 mFillingUpStatus = FS_FILLED; 577 } 578 return true; 579 } 580 581 if (framesReady() >= mFrameCount || 582 (mCblk->mFlags & CBLK_FORCEREADY)) { 583 mFillingUpStatus = FS_FILLED; 584 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 585 return true; 586 } 587 return false; 588} 589 590status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 591 int triggerSession __unused) 592{ 593 status_t status = NO_ERROR; 594 ALOGV("start(%d), calling pid %d session %d", 595 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 596 597 sp<ThreadBase> thread = mThread.promote(); 598 if (thread != 0) { 599 if (isOffloaded()) { 600 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 601 Mutex::Autolock _lth(thread->mLock); 602 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 603 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 604 (ec != 0 && ec->isNonOffloadableEnabled())) { 605 invalidate(); 606 return PERMISSION_DENIED; 607 } 608 } 609 Mutex::Autolock _lth(thread->mLock); 610 track_state state = mState; 611 // here the track could be either new, or restarted 612 // in both cases "unstop" the track 613 614 // initial state-stopping. next state-pausing. 615 // What if resume is called ? 616 617 if (state == PAUSED || state == PAUSING) { 618 if (mResumeToStopping) { 619 // happened we need to resume to STOPPING_1 620 mState = TrackBase::STOPPING_1; 621 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 622 } else { 623 mState = TrackBase::RESUMING; 624 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 625 } 626 } else { 627 mState = TrackBase::ACTIVE; 628 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 629 } 630 631 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 632 status = playbackThread->addTrack_l(this); 633 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 634 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 635 // restore previous state if start was rejected by policy manager 636 if (status == PERMISSION_DENIED) { 637 mState = state; 638 } 639 } 640 // track was already in the active list, not a problem 641 if (status == ALREADY_EXISTS) { 642 status = NO_ERROR; 643 } else { 644 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 645 // It is usually unsafe to access the server proxy from a binder thread. 646 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 647 // isn't looking at this track yet: we still hold the normal mixer thread lock, 648 // and for fast tracks the track is not yet in the fast mixer thread's active set. 649 ServerProxy::Buffer buffer; 650 buffer.mFrameCount = 1; 651 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 652 } 653 } else { 654 status = BAD_VALUE; 655 } 656 return status; 657} 658 659void AudioFlinger::PlaybackThread::Track::stop() 660{ 661 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 662 sp<ThreadBase> thread = mThread.promote(); 663 if (thread != 0) { 664 Mutex::Autolock _l(thread->mLock); 665 track_state state = mState; 666 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 667 // If the track is not active (PAUSED and buffers full), flush buffers 668 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 669 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 670 reset(); 671 mState = STOPPED; 672 } else if (!isFastTrack() && !isOffloaded()) { 673 mState = STOPPED; 674 } else { 675 // For fast tracks prepareTracks_l() will set state to STOPPING_2 676 // presentation is complete 677 // For an offloaded track this starts a drain and state will 678 // move to STOPPING_2 when drain completes and then STOPPED 679 mState = STOPPING_1; 680 } 681 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 682 playbackThread); 683 } 684 } 685} 686 687void AudioFlinger::PlaybackThread::Track::pause() 688{ 689 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 690 sp<ThreadBase> thread = mThread.promote(); 691 if (thread != 0) { 692 Mutex::Autolock _l(thread->mLock); 693 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 694 switch (mState) { 695 case STOPPING_1: 696 case STOPPING_2: 697 if (!isOffloaded()) { 698 /* nothing to do if track is not offloaded */ 699 break; 700 } 701 702 // Offloaded track was draining, we need to carry on draining when resumed 703 mResumeToStopping = true; 704 // fall through... 705 case ACTIVE: 706 case RESUMING: 707 mState = PAUSING; 708 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 709 playbackThread->broadcast_l(); 710 break; 711 712 default: 713 break; 714 } 715 } 716} 717 718void AudioFlinger::PlaybackThread::Track::flush() 719{ 720 ALOGV("flush(%d)", mName); 721 sp<ThreadBase> thread = mThread.promote(); 722 if (thread != 0) { 723 Mutex::Autolock _l(thread->mLock); 724 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 725 726 if (isOffloaded()) { 727 // If offloaded we allow flush during any state except terminated 728 // and keep the track active to avoid problems if user is seeking 729 // rapidly and underlying hardware has a significant delay handling 730 // a pause 731 if (isTerminated()) { 732 return; 733 } 734 735 ALOGV("flush: offload flush"); 736 reset(); 737 738 if (mState == STOPPING_1 || mState == STOPPING_2) { 739 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 740 mState = ACTIVE; 741 } 742 743 if (mState == ACTIVE) { 744 ALOGV("flush called in active state, resetting buffer time out retry count"); 745 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 746 } 747 748 mFlushHwPending = true; 749 mResumeToStopping = false; 750 } else { 751 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 752 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 753 return; 754 } 755 // No point remaining in PAUSED state after a flush => go to 756 // FLUSHED state 757 mState = FLUSHED; 758 // do not reset the track if it is still in the process of being stopped or paused. 759 // this will be done by prepareTracks_l() when the track is stopped. 760 // prepareTracks_l() will see mState == FLUSHED, then 761 // remove from active track list, reset(), and trigger presentation complete 762 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 763 reset(); 764 } 765 } 766 // Prevent flush being lost if the track is flushed and then resumed 767 // before mixer thread can run. This is important when offloading 768 // because the hardware buffer could hold a large amount of audio 769 playbackThread->broadcast_l(); 770 } 771} 772 773// must be called with thread lock held 774void AudioFlinger::PlaybackThread::Track::flushAck() 775{ 776 if (!isOffloaded()) 777 return; 778 779 mFlushHwPending = false; 780} 781 782void AudioFlinger::PlaybackThread::Track::reset() 783{ 784 // Do not reset twice to avoid discarding data written just after a flush and before 785 // the audioflinger thread detects the track is stopped. 786 if (!mResetDone) { 787 // Force underrun condition to avoid false underrun callback until first data is 788 // written to buffer 789 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 790 mFillingUpStatus = FS_FILLING; 791 mResetDone = true; 792 if (mState == FLUSHED) { 793 mState = IDLE; 794 } 795 } 796} 797 798status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 799{ 800 sp<ThreadBase> thread = mThread.promote(); 801 if (thread == 0) { 802 ALOGE("thread is dead"); 803 return FAILED_TRANSACTION; 804 } else if ((thread->type() == ThreadBase::DIRECT) || 805 (thread->type() == ThreadBase::OFFLOAD)) { 806 return thread->setParameters(keyValuePairs); 807 } else { 808 return PERMISSION_DENIED; 809 } 810} 811 812status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 813{ 814 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 815 if (isFastTrack()) { 816 return INVALID_OPERATION; 817 } 818 sp<ThreadBase> thread = mThread.promote(); 819 if (thread == 0) { 820 return INVALID_OPERATION; 821 } 822 Mutex::Autolock _l(thread->mLock); 823 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 824 if (!isOffloaded()) { 825 if (!playbackThread->mLatchQValid) { 826 return INVALID_OPERATION; 827 } 828 uint32_t unpresentedFrames = 829 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 830 playbackThread->mSampleRate; 831 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 832 if (framesWritten < unpresentedFrames) { 833 return INVALID_OPERATION; 834 } 835 timestamp.mPosition = framesWritten - unpresentedFrames; 836 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 837 return NO_ERROR; 838 } 839 840 return playbackThread->getTimestamp_l(timestamp); 841} 842 843status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 844{ 845 status_t status = DEAD_OBJECT; 846 sp<ThreadBase> thread = mThread.promote(); 847 if (thread != 0) { 848 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 849 sp<AudioFlinger> af = mClient->audioFlinger(); 850 851 Mutex::Autolock _l(af->mLock); 852 853 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 854 855 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 856 Mutex::Autolock _dl(playbackThread->mLock); 857 Mutex::Autolock _sl(srcThread->mLock); 858 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 859 if (chain == 0) { 860 return INVALID_OPERATION; 861 } 862 863 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 864 if (effect == 0) { 865 return INVALID_OPERATION; 866 } 867 srcThread->removeEffect_l(effect); 868 status = playbackThread->addEffect_l(effect); 869 if (status != NO_ERROR) { 870 srcThread->addEffect_l(effect); 871 return INVALID_OPERATION; 872 } 873 // removeEffect_l() has stopped the effect if it was active so it must be restarted 874 if (effect->state() == EffectModule::ACTIVE || 875 effect->state() == EffectModule::STOPPING) { 876 effect->start(); 877 } 878 879 sp<EffectChain> dstChain = effect->chain().promote(); 880 if (dstChain == 0) { 881 srcThread->addEffect_l(effect); 882 return INVALID_OPERATION; 883 } 884 AudioSystem::unregisterEffect(effect->id()); 885 AudioSystem::registerEffect(&effect->desc(), 886 srcThread->id(), 887 dstChain->strategy(), 888 AUDIO_SESSION_OUTPUT_MIX, 889 effect->id()); 890 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 891 } 892 status = playbackThread->attachAuxEffect(this, EffectId); 893 } 894 return status; 895} 896 897void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 898{ 899 mAuxEffectId = EffectId; 900 mAuxBuffer = buffer; 901} 902 903bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 904 size_t audioHalFrames) 905{ 906 // a track is considered presented when the total number of frames written to audio HAL 907 // corresponds to the number of frames written when presentationComplete() is called for the 908 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 909 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 910 // to detect when all frames have been played. In this case framesWritten isn't 911 // useful because it doesn't always reflect whether there is data in the h/w 912 // buffers, particularly if a track has been paused and resumed during draining 913 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 914 mPresentationCompleteFrames, framesWritten); 915 if (mPresentationCompleteFrames == 0) { 916 mPresentationCompleteFrames = framesWritten + audioHalFrames; 917 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 918 mPresentationCompleteFrames, audioHalFrames); 919 } 920 921 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 922 ALOGV("presentationComplete() session %d complete: framesWritten %d", 923 mSessionId, framesWritten); 924 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 925 mAudioTrackServerProxy->setStreamEndDone(); 926 return true; 927 } 928 return false; 929} 930 931void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 932{ 933 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 934 if (mSyncEvents[i]->type() == type) { 935 mSyncEvents[i]->trigger(); 936 mSyncEvents.removeAt(i); 937 i--; 938 } 939 } 940} 941 942// implement VolumeBufferProvider interface 943 944uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 945{ 946 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 947 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 948 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 949 uint32_t vl = vlr & 0xFFFF; 950 uint32_t vr = vlr >> 16; 951 // track volumes come from shared memory, so can't be trusted and must be clamped 952 if (vl > MAX_GAIN_INT) { 953 vl = MAX_GAIN_INT; 954 } 955 if (vr > MAX_GAIN_INT) { 956 vr = MAX_GAIN_INT; 957 } 958 // now apply the cached master volume and stream type volume; 959 // this is trusted but lacks any synchronization or barrier so may be stale 960 float v = mCachedVolume; 961 vl *= v; 962 vr *= v; 963 // re-combine into U4.16 964 vlr = (vr << 16) | (vl & 0xFFFF); 965 // FIXME look at mute, pause, and stop flags 966 return vlr; 967} 968 969status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 970{ 971 if (isTerminated() || mState == PAUSED || 972 ((framesReady() == 0) && ((mSharedBuffer != 0) || 973 (mState == STOPPED)))) { 974 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 975 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 976 event->cancel(); 977 return INVALID_OPERATION; 978 } 979 (void) TrackBase::setSyncEvent(event); 980 return NO_ERROR; 981} 982 983void AudioFlinger::PlaybackThread::Track::invalidate() 984{ 985 // FIXME should use proxy, and needs work 986 audio_track_cblk_t* cblk = mCblk; 987 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 988 android_atomic_release_store(0x40000000, &cblk->mFutex); 989 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 990 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 991 mIsInvalid = true; 992} 993 994void AudioFlinger::PlaybackThread::Track::signal() 995{ 996 sp<ThreadBase> thread = mThread.promote(); 997 if (thread != 0) { 998 PlaybackThread *t = (PlaybackThread *)thread.get(); 999 Mutex::Autolock _l(t->mLock); 1000 t->broadcast_l(); 1001 } 1002} 1003 1004//To be called with thread lock held 1005bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1006 1007 if (mState == RESUMING) 1008 return true; 1009 /* Resume is pending if track was stopping before pause was called */ 1010 if (mState == STOPPING_1 && 1011 mResumeToStopping) 1012 return true; 1013 1014 return false; 1015} 1016 1017//To be called with thread lock held 1018void AudioFlinger::PlaybackThread::Track::resumeAck() { 1019 1020 1021 if (mState == RESUMING) 1022 mState = ACTIVE; 1023 1024 // Other possibility of pending resume is stopping_1 state 1025 // Do not update the state from stopping as this prevents 1026 // drain being called. 1027 if (mState == STOPPING_1) { 1028 mResumeToStopping = false; 1029 } 1030} 1031// ---------------------------------------------------------------------------- 1032 1033sp<AudioFlinger::PlaybackThread::TimedTrack> 1034AudioFlinger::PlaybackThread::TimedTrack::create( 1035 PlaybackThread *thread, 1036 const sp<Client>& client, 1037 audio_stream_type_t streamType, 1038 uint32_t sampleRate, 1039 audio_format_t format, 1040 audio_channel_mask_t channelMask, 1041 size_t frameCount, 1042 const sp<IMemory>& sharedBuffer, 1043 int sessionId, 1044 int uid) 1045{ 1046 if (!client->reserveTimedTrack()) 1047 return 0; 1048 1049 return new TimedTrack( 1050 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1051 sharedBuffer, sessionId, uid); 1052} 1053 1054AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1055 PlaybackThread *thread, 1056 const sp<Client>& client, 1057 audio_stream_type_t streamType, 1058 uint32_t sampleRate, 1059 audio_format_t format, 1060 audio_channel_mask_t channelMask, 1061 size_t frameCount, 1062 const sp<IMemory>& sharedBuffer, 1063 int sessionId, 1064 int uid) 1065 : Track(thread, client, streamType, sampleRate, format, channelMask, 1066 frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED), 1067 mQueueHeadInFlight(false), 1068 mTrimQueueHeadOnRelease(false), 1069 mFramesPendingInQueue(0), 1070 mTimedSilenceBuffer(NULL), 1071 mTimedSilenceBufferSize(0), 1072 mTimedAudioOutputOnTime(false), 1073 mMediaTimeTransformValid(false) 1074{ 1075 LocalClock lc; 1076 mLocalTimeFreq = lc.getLocalFreq(); 1077 1078 mLocalTimeToSampleTransform.a_zero = 0; 1079 mLocalTimeToSampleTransform.b_zero = 0; 1080 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1081 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1082 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1083 &mLocalTimeToSampleTransform.a_to_b_denom); 1084 1085 mMediaTimeToSampleTransform.a_zero = 0; 1086 mMediaTimeToSampleTransform.b_zero = 0; 1087 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1088 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1089 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1090 &mMediaTimeToSampleTransform.a_to_b_denom); 1091} 1092 1093AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1094 mClient->releaseTimedTrack(); 1095 delete [] mTimedSilenceBuffer; 1096} 1097 1098status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1099 size_t size, sp<IMemory>* buffer) { 1100 1101 Mutex::Autolock _l(mTimedBufferQueueLock); 1102 1103 trimTimedBufferQueue_l(); 1104 1105 // lazily initialize the shared memory heap for timed buffers 1106 if (mTimedMemoryDealer == NULL) { 1107 const int kTimedBufferHeapSize = 512 << 10; 1108 1109 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1110 "AudioFlingerTimed"); 1111 if (mTimedMemoryDealer == NULL) { 1112 return NO_MEMORY; 1113 } 1114 } 1115 1116 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1117 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1118 return NO_MEMORY; 1119 } 1120 1121 *buffer = newBuffer; 1122 return NO_ERROR; 1123} 1124 1125// caller must hold mTimedBufferQueueLock 1126void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1127 int64_t mediaTimeNow; 1128 { 1129 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1130 if (!mMediaTimeTransformValid) 1131 return; 1132 1133 int64_t targetTimeNow; 1134 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1135 ? mCCHelper.getCommonTime(&targetTimeNow) 1136 : mCCHelper.getLocalTime(&targetTimeNow); 1137 1138 if (OK != res) 1139 return; 1140 1141 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1142 &mediaTimeNow)) { 1143 return; 1144 } 1145 } 1146 1147 size_t trimEnd; 1148 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1149 int64_t bufEnd; 1150 1151 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1152 // We have a next buffer. Just use its PTS as the PTS of the frame 1153 // following the last frame in this buffer. If the stream is sparse 1154 // (ie, there are deliberate gaps left in the stream which should be 1155 // filled with silence by the TimedAudioTrack), then this can result 1156 // in one extra buffer being left un-trimmed when it could have 1157 // been. In general, this is not typical, and we would rather 1158 // optimized away the TS calculation below for the more common case 1159 // where PTSes are contiguous. 1160 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1161 } else { 1162 // We have no next buffer. Compute the PTS of the frame following 1163 // the last frame in this buffer by computing the duration of of 1164 // this frame in media time units and adding it to the PTS of the 1165 // buffer. 1166 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1167 / mFrameSize; 1168 1169 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1170 &bufEnd)) { 1171 ALOGE("Failed to convert frame count of %lld to media time" 1172 " duration" " (scale factor %d/%u) in %s", 1173 frameCount, 1174 mMediaTimeToSampleTransform.a_to_b_numer, 1175 mMediaTimeToSampleTransform.a_to_b_denom, 1176 __PRETTY_FUNCTION__); 1177 break; 1178 } 1179 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1180 } 1181 1182 if (bufEnd > mediaTimeNow) 1183 break; 1184 1185 // Is the buffer we want to use in the middle of a mix operation right 1186 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1187 // from the mixer which should be coming back shortly. 1188 if (!trimEnd && mQueueHeadInFlight) { 1189 mTrimQueueHeadOnRelease = true; 1190 } 1191 } 1192 1193 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1194 if (trimStart < trimEnd) { 1195 // Update the bookkeeping for framesReady() 1196 for (size_t i = trimStart; i < trimEnd; ++i) { 1197 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1198 } 1199 1200 // Now actually remove the buffers from the queue. 1201 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1202 } 1203} 1204 1205void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1206 const char* logTag) { 1207 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1208 "%s called (reason \"%s\"), but timed buffer queue has no" 1209 " elements to trim.", __FUNCTION__, logTag); 1210 1211 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1212 mTimedBufferQueue.removeAt(0); 1213} 1214 1215void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1216 const TimedBuffer& buf, 1217 const char* logTag __unused) { 1218 uint32_t bufBytes = buf.buffer()->size(); 1219 uint32_t consumedAlready = buf.position(); 1220 1221 ALOG_ASSERT(consumedAlready <= bufBytes, 1222 "Bad bookkeeping while updating frames pending. Timed buffer is" 1223 " only %u bytes long, but claims to have consumed %u" 1224 " bytes. (update reason: \"%s\")", 1225 bufBytes, consumedAlready, logTag); 1226 1227 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1228 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1229 "Bad bookkeeping while updating frames pending. Should have at" 1230 " least %u queued frames, but we think we have only %u. (update" 1231 " reason: \"%s\")", 1232 bufFrames, mFramesPendingInQueue, logTag); 1233 1234 mFramesPendingInQueue -= bufFrames; 1235} 1236 1237status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1238 const sp<IMemory>& buffer, int64_t pts) { 1239 1240 { 1241 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1242 if (!mMediaTimeTransformValid) 1243 return INVALID_OPERATION; 1244 } 1245 1246 Mutex::Autolock _l(mTimedBufferQueueLock); 1247 1248 uint32_t bufFrames = buffer->size() / mFrameSize; 1249 mFramesPendingInQueue += bufFrames; 1250 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1251 1252 return NO_ERROR; 1253} 1254 1255status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1256 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1257 1258 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1259 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1260 target); 1261 1262 if (!(target == TimedAudioTrack::LOCAL_TIME || 1263 target == TimedAudioTrack::COMMON_TIME)) { 1264 return BAD_VALUE; 1265 } 1266 1267 Mutex::Autolock lock(mMediaTimeTransformLock); 1268 mMediaTimeTransform = xform; 1269 mMediaTimeTransformTarget = target; 1270 mMediaTimeTransformValid = true; 1271 1272 return NO_ERROR; 1273} 1274 1275#define min(a, b) ((a) < (b) ? (a) : (b)) 1276 1277// implementation of getNextBuffer for tracks whose buffers have timestamps 1278status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1279 AudioBufferProvider::Buffer* buffer, int64_t pts) 1280{ 1281 if (pts == AudioBufferProvider::kInvalidPTS) { 1282 buffer->raw = NULL; 1283 buffer->frameCount = 0; 1284 mTimedAudioOutputOnTime = false; 1285 return INVALID_OPERATION; 1286 } 1287 1288 Mutex::Autolock _l(mTimedBufferQueueLock); 1289 1290 ALOG_ASSERT(!mQueueHeadInFlight, 1291 "getNextBuffer called without releaseBuffer!"); 1292 1293 while (true) { 1294 1295 // if we have no timed buffers, then fail 1296 if (mTimedBufferQueue.isEmpty()) { 1297 buffer->raw = NULL; 1298 buffer->frameCount = 0; 1299 return NOT_ENOUGH_DATA; 1300 } 1301 1302 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1303 1304 // calculate the PTS of the head of the timed buffer queue expressed in 1305 // local time 1306 int64_t headLocalPTS; 1307 { 1308 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1309 1310 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1311 1312 if (mMediaTimeTransform.a_to_b_denom == 0) { 1313 // the transform represents a pause, so yield silence 1314 timedYieldSilence_l(buffer->frameCount, buffer); 1315 return NO_ERROR; 1316 } 1317 1318 int64_t transformedPTS; 1319 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1320 &transformedPTS)) { 1321 // the transform failed. this shouldn't happen, but if it does 1322 // then just drop this buffer 1323 ALOGW("timedGetNextBuffer transform failed"); 1324 buffer->raw = NULL; 1325 buffer->frameCount = 0; 1326 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1327 return NO_ERROR; 1328 } 1329 1330 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1331 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1332 &headLocalPTS)) { 1333 buffer->raw = NULL; 1334 buffer->frameCount = 0; 1335 return INVALID_OPERATION; 1336 } 1337 } else { 1338 headLocalPTS = transformedPTS; 1339 } 1340 } 1341 1342 uint32_t sr = sampleRate(); 1343 1344 // adjust the head buffer's PTS to reflect the portion of the head buffer 1345 // that has already been consumed 1346 int64_t effectivePTS = headLocalPTS + 1347 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1348 1349 // Calculate the delta in samples between the head of the input buffer 1350 // queue and the start of the next output buffer that will be written. 1351 // If the transformation fails because of over or underflow, it means 1352 // that the sample's position in the output stream is so far out of 1353 // whack that it should just be dropped. 1354 int64_t sampleDelta; 1355 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1356 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1357 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1358 " mix"); 1359 continue; 1360 } 1361 if (!mLocalTimeToSampleTransform.doForwardTransform( 1362 (effectivePTS - pts) << 32, &sampleDelta)) { 1363 ALOGV("*** too late during sample rate transform: dropped buffer"); 1364 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1365 continue; 1366 } 1367 1368 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1369 " sampleDelta=[%d.%08x]", 1370 head.pts(), head.position(), pts, 1371 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1372 + (sampleDelta >> 32)), 1373 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1374 1375 // if the delta between the ideal placement for the next input sample and 1376 // the current output position is within this threshold, then we will 1377 // concatenate the next input samples to the previous output 1378 const int64_t kSampleContinuityThreshold = 1379 (static_cast<int64_t>(sr) << 32) / 250; 1380 1381 // if this is the first buffer of audio that we're emitting from this track 1382 // then it should be almost exactly on time. 1383 const int64_t kSampleStartupThreshold = 1LL << 32; 1384 1385 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1386 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1387 // the next input is close enough to being on time, so concatenate it 1388 // with the last output 1389 timedYieldSamples_l(buffer); 1390 1391 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1392 head.position(), buffer->frameCount); 1393 return NO_ERROR; 1394 } 1395 1396 // Looks like our output is not on time. Reset our on timed status. 1397 // Next time we mix samples from our input queue, then should be within 1398 // the StartupThreshold. 1399 mTimedAudioOutputOnTime = false; 1400 if (sampleDelta > 0) { 1401 // the gap between the current output position and the proper start of 1402 // the next input sample is too big, so fill it with silence 1403 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1404 1405 timedYieldSilence_l(framesUntilNextInput, buffer); 1406 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1407 return NO_ERROR; 1408 } else { 1409 // the next input sample is late 1410 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1411 size_t onTimeSamplePosition = 1412 head.position() + lateFrames * mFrameSize; 1413 1414 if (onTimeSamplePosition > head.buffer()->size()) { 1415 // all the remaining samples in the head are too late, so 1416 // drop it and move on 1417 ALOGV("*** too late: dropped buffer"); 1418 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1419 continue; 1420 } else { 1421 // skip over the late samples 1422 head.setPosition(onTimeSamplePosition); 1423 1424 // yield the available samples 1425 timedYieldSamples_l(buffer); 1426 1427 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1428 return NO_ERROR; 1429 } 1430 } 1431 } 1432} 1433 1434// Yield samples from the timed buffer queue head up to the given output 1435// buffer's capacity. 1436// 1437// Caller must hold mTimedBufferQueueLock 1438void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1439 AudioBufferProvider::Buffer* buffer) { 1440 1441 const TimedBuffer& head = mTimedBufferQueue[0]; 1442 1443 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1444 head.position()); 1445 1446 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1447 mFrameSize); 1448 size_t framesRequested = buffer->frameCount; 1449 buffer->frameCount = min(framesLeftInHead, framesRequested); 1450 1451 mQueueHeadInFlight = true; 1452 mTimedAudioOutputOnTime = true; 1453} 1454 1455// Yield samples of silence up to the given output buffer's capacity 1456// 1457// Caller must hold mTimedBufferQueueLock 1458void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1459 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1460 1461 // lazily allocate a buffer filled with silence 1462 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1463 delete [] mTimedSilenceBuffer; 1464 mTimedSilenceBufferSize = numFrames * mFrameSize; 1465 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1466 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1467 } 1468 1469 buffer->raw = mTimedSilenceBuffer; 1470 size_t framesRequested = buffer->frameCount; 1471 buffer->frameCount = min(numFrames, framesRequested); 1472 1473 mTimedAudioOutputOnTime = false; 1474} 1475 1476// AudioBufferProvider interface 1477void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1478 AudioBufferProvider::Buffer* buffer) { 1479 1480 Mutex::Autolock _l(mTimedBufferQueueLock); 1481 1482 // If the buffer which was just released is part of the buffer at the head 1483 // of the queue, be sure to update the amt of the buffer which has been 1484 // consumed. If the buffer being returned is not part of the head of the 1485 // queue, its either because the buffer is part of the silence buffer, or 1486 // because the head of the timed queue was trimmed after the mixer called 1487 // getNextBuffer but before the mixer called releaseBuffer. 1488 if (buffer->raw == mTimedSilenceBuffer) { 1489 ALOG_ASSERT(!mQueueHeadInFlight, 1490 "Queue head in flight during release of silence buffer!"); 1491 goto done; 1492 } 1493 1494 ALOG_ASSERT(mQueueHeadInFlight, 1495 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1496 " head in flight."); 1497 1498 if (mTimedBufferQueue.size()) { 1499 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1500 1501 void* start = head.buffer()->pointer(); 1502 void* end = reinterpret_cast<void*>( 1503 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1504 + head.buffer()->size()); 1505 1506 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1507 "released buffer not within the head of the timed buffer" 1508 " queue; qHead = [%p, %p], released buffer = %p", 1509 start, end, buffer->raw); 1510 1511 head.setPosition(head.position() + 1512 (buffer->frameCount * mFrameSize)); 1513 mQueueHeadInFlight = false; 1514 1515 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1516 "Bad bookkeeping during releaseBuffer! Should have at" 1517 " least %u queued frames, but we think we have only %u", 1518 buffer->frameCount, mFramesPendingInQueue); 1519 1520 mFramesPendingInQueue -= buffer->frameCount; 1521 1522 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1523 || mTrimQueueHeadOnRelease) { 1524 trimTimedBufferQueueHead_l("releaseBuffer"); 1525 mTrimQueueHeadOnRelease = false; 1526 } 1527 } else { 1528 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1529 " buffers in the timed buffer queue"); 1530 } 1531 1532done: 1533 buffer->raw = 0; 1534 buffer->frameCount = 0; 1535} 1536 1537size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1538 Mutex::Autolock _l(mTimedBufferQueueLock); 1539 return mFramesPendingInQueue; 1540} 1541 1542AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1543 : mPTS(0), mPosition(0) {} 1544 1545AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1546 const sp<IMemory>& buffer, int64_t pts) 1547 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1548 1549 1550// ---------------------------------------------------------------------------- 1551 1552AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1553 PlaybackThread *playbackThread, 1554 DuplicatingThread *sourceThread, 1555 uint32_t sampleRate, 1556 audio_format_t format, 1557 audio_channel_mask_t channelMask, 1558 size_t frameCount, 1559 int uid) 1560 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1561 NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT), 1562 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1563{ 1564 1565 if (mCblk != NULL) { 1566 mOutBuffer.frameCount = 0; 1567 playbackThread->mTracks.add(this); 1568 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1569 "frameCount %u, mChannelMask 0x%08x", 1570 mCblk, mBuffer, 1571 frameCount, mChannelMask); 1572 // since client and server are in the same process, 1573 // the buffer has the same virtual address on both sides 1574 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1575 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1576 mClientProxy->setSendLevel(0.0); 1577 mClientProxy->setSampleRate(sampleRate); 1578 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1579 true /*clientInServer*/); 1580 } else { 1581 ALOGW("Error creating output track on thread %p", playbackThread); 1582 } 1583} 1584 1585AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1586{ 1587 clearBufferQueue(); 1588 delete mClientProxy; 1589 // superclass destructor will now delete the server proxy and shared memory both refer to 1590} 1591 1592status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1593 int triggerSession) 1594{ 1595 status_t status = Track::start(event, triggerSession); 1596 if (status != NO_ERROR) { 1597 return status; 1598 } 1599 1600 mActive = true; 1601 mRetryCount = 127; 1602 return status; 1603} 1604 1605void AudioFlinger::PlaybackThread::OutputTrack::stop() 1606{ 1607 Track::stop(); 1608 clearBufferQueue(); 1609 mOutBuffer.frameCount = 0; 1610 mActive = false; 1611} 1612 1613bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1614{ 1615 Buffer *pInBuffer; 1616 Buffer inBuffer; 1617 uint32_t channelCount = mChannelCount; 1618 bool outputBufferFull = false; 1619 inBuffer.frameCount = frames; 1620 inBuffer.i16 = data; 1621 1622 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1623 1624 if (!mActive && frames != 0) { 1625 start(); 1626 sp<ThreadBase> thread = mThread.promote(); 1627 if (thread != 0) { 1628 MixerThread *mixerThread = (MixerThread *)thread.get(); 1629 if (mFrameCount > frames) { 1630 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1631 uint32_t startFrames = (mFrameCount - frames); 1632 pInBuffer = new Buffer; 1633 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1634 pInBuffer->frameCount = startFrames; 1635 pInBuffer->i16 = pInBuffer->mBuffer; 1636 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1637 mBufferQueue.add(pInBuffer); 1638 } else { 1639 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1640 } 1641 } 1642 } 1643 } 1644 1645 while (waitTimeLeftMs) { 1646 // First write pending buffers, then new data 1647 if (mBufferQueue.size()) { 1648 pInBuffer = mBufferQueue.itemAt(0); 1649 } else { 1650 pInBuffer = &inBuffer; 1651 } 1652 1653 if (pInBuffer->frameCount == 0) { 1654 break; 1655 } 1656 1657 if (mOutBuffer.frameCount == 0) { 1658 mOutBuffer.frameCount = pInBuffer->frameCount; 1659 nsecs_t startTime = systemTime(); 1660 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1661 if (status != NO_ERROR) { 1662 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1663 mThread.unsafe_get(), status); 1664 outputBufferFull = true; 1665 break; 1666 } 1667 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1668 if (waitTimeLeftMs >= waitTimeMs) { 1669 waitTimeLeftMs -= waitTimeMs; 1670 } else { 1671 waitTimeLeftMs = 0; 1672 } 1673 } 1674 1675 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1676 pInBuffer->frameCount; 1677 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1678 Proxy::Buffer buf; 1679 buf.mFrameCount = outFrames; 1680 buf.mRaw = NULL; 1681 mClientProxy->releaseBuffer(&buf); 1682 pInBuffer->frameCount -= outFrames; 1683 pInBuffer->i16 += outFrames * channelCount; 1684 mOutBuffer.frameCount -= outFrames; 1685 mOutBuffer.i16 += outFrames * channelCount; 1686 1687 if (pInBuffer->frameCount == 0) { 1688 if (mBufferQueue.size()) { 1689 mBufferQueue.removeAt(0); 1690 delete [] pInBuffer->mBuffer; 1691 delete pInBuffer; 1692 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1693 mThread.unsafe_get(), mBufferQueue.size()); 1694 } else { 1695 break; 1696 } 1697 } 1698 } 1699 1700 // If we could not write all frames, allocate a buffer and queue it for next time. 1701 if (inBuffer.frameCount) { 1702 sp<ThreadBase> thread = mThread.promote(); 1703 if (thread != 0 && !thread->standby()) { 1704 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1705 pInBuffer = new Buffer; 1706 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1707 pInBuffer->frameCount = inBuffer.frameCount; 1708 pInBuffer->i16 = pInBuffer->mBuffer; 1709 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1710 sizeof(int16_t)); 1711 mBufferQueue.add(pInBuffer); 1712 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1713 mThread.unsafe_get(), mBufferQueue.size()); 1714 } else { 1715 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1716 mThread.unsafe_get(), this); 1717 } 1718 } 1719 } 1720 1721 // Calling write() with a 0 length buffer, means that no more data will be written: 1722 // If no more buffers are pending, fill output track buffer to make sure it is started 1723 // by output mixer. 1724 if (frames == 0 && mBufferQueue.size() == 0) { 1725 // FIXME borken, replace by getting framesReady() from proxy 1726 size_t user = 0; // was mCblk->user 1727 if (user < mFrameCount) { 1728 frames = mFrameCount - user; 1729 pInBuffer = new Buffer; 1730 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1731 pInBuffer->frameCount = frames; 1732 pInBuffer->i16 = pInBuffer->mBuffer; 1733 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1734 mBufferQueue.add(pInBuffer); 1735 } else if (mActive) { 1736 stop(); 1737 } 1738 } 1739 1740 return outputBufferFull; 1741} 1742 1743status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1744 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1745{ 1746 ClientProxy::Buffer buf; 1747 buf.mFrameCount = buffer->frameCount; 1748 struct timespec timeout; 1749 timeout.tv_sec = waitTimeMs / 1000; 1750 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1751 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1752 buffer->frameCount = buf.mFrameCount; 1753 buffer->raw = buf.mRaw; 1754 return status; 1755} 1756 1757void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1758{ 1759 size_t size = mBufferQueue.size(); 1760 1761 for (size_t i = 0; i < size; i++) { 1762 Buffer *pBuffer = mBufferQueue.itemAt(i); 1763 delete [] pBuffer->mBuffer; 1764 delete pBuffer; 1765 } 1766 mBufferQueue.clear(); 1767} 1768 1769 1770// ---------------------------------------------------------------------------- 1771// Record 1772// ---------------------------------------------------------------------------- 1773 1774AudioFlinger::RecordHandle::RecordHandle( 1775 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1776 : BnAudioRecord(), 1777 mRecordTrack(recordTrack) 1778{ 1779} 1780 1781AudioFlinger::RecordHandle::~RecordHandle() { 1782 stop_nonvirtual(); 1783 mRecordTrack->destroy(); 1784} 1785 1786sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1787 return mRecordTrack->getCblk(); 1788} 1789 1790status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1791 int triggerSession) { 1792 ALOGV("RecordHandle::start()"); 1793 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1794} 1795 1796void AudioFlinger::RecordHandle::stop() { 1797 stop_nonvirtual(); 1798} 1799 1800void AudioFlinger::RecordHandle::stop_nonvirtual() { 1801 ALOGV("RecordHandle::stop()"); 1802 mRecordTrack->stop(); 1803} 1804 1805status_t AudioFlinger::RecordHandle::onTransact( 1806 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1807{ 1808 return BnAudioRecord::onTransact(code, data, reply, flags); 1809} 1810 1811// ---------------------------------------------------------------------------- 1812 1813// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1814AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1815 RecordThread *thread, 1816 const sp<Client>& client, 1817 uint32_t sampleRate, 1818 audio_format_t format, 1819 audio_channel_mask_t channelMask, 1820 size_t frameCount, 1821 int sessionId, 1822 int uid) 1823 : TrackBase(thread, client, sampleRate, format, 1824 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/), 1825 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1826 // See real initialization of mRsmpInFront at RecordThread::start() 1827 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1828{ 1829 if (mCblk == NULL) { 1830 return; 1831 } 1832 1833 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1834 1835 uint32_t channelCount = popcount(channelMask); 1836 // FIXME I don't understand either of the channel count checks 1837 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 1838 channelCount <= FCC_2) { 1839 // sink SR 1840 mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate); 1841 // source SR 1842 mResampler->setSampleRate(thread->mSampleRate); 1843 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 1844 mResamplerBufferProvider = new ResamplerBufferProvider(this); 1845 } 1846} 1847 1848AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1849{ 1850 ALOGV("%s", __func__); 1851 delete mResampler; 1852 delete[] mRsmpOutBuffer; 1853 delete mResamplerBufferProvider; 1854} 1855 1856// AudioBufferProvider interface 1857status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1858 int64_t pts __unused) 1859{ 1860 ServerProxy::Buffer buf; 1861 buf.mFrameCount = buffer->frameCount; 1862 status_t status = mServerProxy->obtainBuffer(&buf); 1863 buffer->frameCount = buf.mFrameCount; 1864 buffer->raw = buf.mRaw; 1865 if (buf.mFrameCount == 0) { 1866 // FIXME also wake futex so that overrun is noticed more quickly 1867 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1868 } 1869 return status; 1870} 1871 1872status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1873 int triggerSession) 1874{ 1875 sp<ThreadBase> thread = mThread.promote(); 1876 if (thread != 0) { 1877 RecordThread *recordThread = (RecordThread *)thread.get(); 1878 return recordThread->start(this, event, triggerSession); 1879 } else { 1880 return BAD_VALUE; 1881 } 1882} 1883 1884void AudioFlinger::RecordThread::RecordTrack::stop() 1885{ 1886 sp<ThreadBase> thread = mThread.promote(); 1887 if (thread != 0) { 1888 RecordThread *recordThread = (RecordThread *)thread.get(); 1889 if (recordThread->stop(this)) { 1890 AudioSystem::stopInput(recordThread->id()); 1891 } 1892 } 1893} 1894 1895void AudioFlinger::RecordThread::RecordTrack::destroy() 1896{ 1897 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1898 sp<RecordTrack> keep(this); 1899 { 1900 sp<ThreadBase> thread = mThread.promote(); 1901 if (thread != 0) { 1902 if (mState == ACTIVE || mState == RESUMING) { 1903 AudioSystem::stopInput(thread->id()); 1904 } 1905 AudioSystem::releaseInput(thread->id()); 1906 Mutex::Autolock _l(thread->mLock); 1907 RecordThread *recordThread = (RecordThread *) thread.get(); 1908 recordThread->destroyTrack_l(this); 1909 } 1910 } 1911} 1912 1913void AudioFlinger::RecordThread::RecordTrack::invalidate() 1914{ 1915 // FIXME should use proxy, and needs work 1916 audio_track_cblk_t* cblk = mCblk; 1917 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1918 android_atomic_release_store(0x40000000, &cblk->mFutex); 1919 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1920 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1921} 1922 1923 1924/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1925{ 1926 result.append(" Active Client Fmt Chn mask Session S Server fCount Resampling\n"); 1927} 1928 1929void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 1930{ 1931 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n", 1932 active ? "yes" : "no", 1933 (mClient == 0) ? getpid_cached : mClient->pid(), 1934 mFormat, 1935 mChannelMask, 1936 mSessionId, 1937 mState, 1938 mCblk->mServer, 1939 mFrameCount, 1940 mResampler != NULL); 1941 1942} 1943 1944void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 1945{ 1946 if (event == mSyncStartEvent) { 1947 ssize_t framesToDrop = 0; 1948 sp<ThreadBase> threadBase = mThread.promote(); 1949 if (threadBase != 0) { 1950 // TODO: use actual buffer filling status instead of 2 buffers when info is available 1951 // from audio HAL 1952 framesToDrop = threadBase->mFrameCount * 2; 1953 } 1954 mFramesToDrop = framesToDrop; 1955 } 1956} 1957 1958void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 1959{ 1960 if (mSyncStartEvent != 0) { 1961 mSyncStartEvent->cancel(); 1962 mSyncStartEvent.clear(); 1963 } 1964 mFramesToDrop = 0; 1965} 1966 1967}; // namespace android 1968