Tracks.cpp revision ced6e74215937182fe2f9f6b0867f7c28ccd02c1
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 <sys/syscall.h> 25#include <utils/Log.h> 26 27#include <private/media/AudioTrackShared.h> 28 29#include <common_time/cc_helper.h> 30#include <common_time/local_clock.h> 31 32#include "AudioMixer.h" 33#include "AudioFlinger.h" 34#include "ServiceUtilities.h" 35 36#include <media/nbaio/Pipe.h> 37#include <media/nbaio/PipeReader.h> 38#include <audio_utils/minifloat.h> 39 40// ---------------------------------------------------------------------------- 41 42// Note: the following macro is used for extremely verbose logging message. In 43// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 44// 0; but one side effect of this is to turn all LOGV's as well. Some messages 45// are so verbose that we want to suppress them even when we have ALOG_ASSERT 46// turned on. Do not uncomment the #def below unless you really know what you 47// are doing and want to see all of the extremely verbose messages. 48//#define VERY_VERY_VERBOSE_LOGGING 49#ifdef VERY_VERY_VERBOSE_LOGGING 50#define ALOGVV ALOGV 51#else 52#define ALOGVV(a...) do { } while(0) 53#endif 54 55namespace android { 56 57// ---------------------------------------------------------------------------- 58// TrackBase 59// ---------------------------------------------------------------------------- 60 61static volatile int32_t nextTrackId = 55; 62 63// TrackBase constructor must be called with AudioFlinger::mLock held 64AudioFlinger::ThreadBase::TrackBase::TrackBase( 65 ThreadBase *thread, 66 const sp<Client>& client, 67 uint32_t sampleRate, 68 audio_format_t format, 69 audio_channel_mask_t channelMask, 70 size_t frameCount, 71 const sp<IMemory>& sharedBuffer, 72 int sessionId, 73 int clientUid, 74 IAudioFlinger::track_flags_t flags, 75 bool isOut, 76 bool useReadOnlyHeap) 77 : RefBase(), 78 mThread(thread), 79 mClient(client), 80 mCblk(NULL), 81 // mBuffer 82 mState(IDLE), 83 mSampleRate(sampleRate), 84 mFormat(format), 85 mChannelMask(channelMask), 86 mChannelCount(isOut ? 87 audio_channel_count_from_out_mask(channelMask) : 88 audio_channel_count_from_in_mask(channelMask)), 89 mFrameSize(audio_is_linear_pcm(format) ? 90 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 91 mFrameCount(frameCount), 92 mSessionId(sessionId), 93 mFlags(flags), 94 mIsOut(isOut), 95 mServerProxy(NULL), 96 mId(android_atomic_inc(&nextTrackId)), 97 mTerminated(false) 98{ 99 // if the caller is us, trust the specified uid 100 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 101 int newclientUid = IPCThreadState::self()->getCallingUid(); 102 if (clientUid != -1 && clientUid != newclientUid) { 103 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 104 } 105 clientUid = newclientUid; 106 } 107 // clientUid contains the uid of the app that is responsible for this track, so we can blame 108 // battery usage on it. 109 mUid = clientUid; 110 111 // client == 0 implies sharedBuffer == 0 112 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 113 114 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 115 sharedBuffer->size()); 116 117 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 118 size_t size = sizeof(audio_track_cblk_t); 119 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 120 if (sharedBuffer == 0 && !useReadOnlyHeap) { 121 size += bufferSize; 122 } 123 124 if (client != 0) { 125 mCblkMemory = client->heap()->allocate(size); 126 if (mCblkMemory == 0 || 127 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 128 ALOGE("not enough memory for AudioTrack size=%u", size); 129 client->heap()->dump("AudioTrack"); 130 mCblkMemory.clear(); 131 return; 132 } 133 } else { 134 // this syntax avoids calling the audio_track_cblk_t constructor twice 135 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 136 // assume mCblk != NULL 137 } 138 139 // construct the shared structure in-place. 140 if (mCblk != NULL) { 141 new(mCblk) audio_track_cblk_t(); 142 if (useReadOnlyHeap) { 143 const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); 144 if (roHeap == 0 || 145 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || 146 (mBuffer = mBufferMemory->pointer()) == NULL) { 147 ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); 148 if (roHeap != 0) { 149 roHeap->dump("buffer"); 150 } 151 mCblkMemory.clear(); 152 mBufferMemory.clear(); 153 return; 154 } 155 memset(mBuffer, 0, bufferSize); 156 } else { 157 // clear all buffers 158 if (sharedBuffer == 0) { 159 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 160 memset(mBuffer, 0, bufferSize); 161 } else { 162 mBuffer = sharedBuffer->pointer(); 163#if 0 164 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 165#endif 166 } 167 } 168 169#ifdef TEE_SINK 170 if (mTeeSinkTrackEnabled) { 171 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 172 if (Format_isValid(pipeFormat)) { 173 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 174 size_t numCounterOffers = 0; 175 const NBAIO_Format offers[1] = {pipeFormat}; 176 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 177 ALOG_ASSERT(index == 0); 178 PipeReader *pipeReader = new PipeReader(*pipe); 179 numCounterOffers = 0; 180 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 181 ALOG_ASSERT(index == 0); 182 mTeeSink = pipe; 183 mTeeSource = pipeReader; 184 } 185 } 186#endif 187 188 } 189} 190 191AudioFlinger::ThreadBase::TrackBase::~TrackBase() 192{ 193#ifdef TEE_SINK 194 dumpTee(-1, mTeeSource, mId); 195#endif 196 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 197 delete mServerProxy; 198 if (mCblk != NULL) { 199 if (mClient == 0) { 200 delete mCblk; 201 } else { 202 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 203 } 204 } 205 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 206 if (mClient != 0) { 207 // Client destructor must run with AudioFlinger client mutex locked 208 Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); 209 // If the client's reference count drops to zero, the associated destructor 210 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 211 // relying on the automatic clear() at end of scope. 212 mClient.clear(); 213 } 214} 215 216// AudioBufferProvider interface 217// getNextBuffer() = 0; 218// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 219void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 220{ 221#ifdef TEE_SINK 222 if (mTeeSink != 0) { 223 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 224 } 225#endif 226 227 ServerProxy::Buffer buf; 228 buf.mFrameCount = buffer->frameCount; 229 buf.mRaw = buffer->raw; 230 buffer->frameCount = 0; 231 buffer->raw = NULL; 232 mServerProxy->releaseBuffer(&buf); 233} 234 235status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 236{ 237 mSyncEvents.add(event); 238 return NO_ERROR; 239} 240 241// ---------------------------------------------------------------------------- 242// Playback 243// ---------------------------------------------------------------------------- 244 245AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 246 : BnAudioTrack(), 247 mTrack(track) 248{ 249} 250 251AudioFlinger::TrackHandle::~TrackHandle() { 252 // just stop the track on deletion, associated resources 253 // will be freed from the main thread once all pending buffers have 254 // been played. Unless it's not in the active track list, in which 255 // case we free everything now... 256 mTrack->destroy(); 257} 258 259sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 260 return mTrack->getCblk(); 261} 262 263status_t AudioFlinger::TrackHandle::start() { 264 return mTrack->start(); 265} 266 267void AudioFlinger::TrackHandle::stop() { 268 mTrack->stop(); 269} 270 271void AudioFlinger::TrackHandle::flush() { 272 mTrack->flush(); 273} 274 275void AudioFlinger::TrackHandle::pause() { 276 mTrack->pause(); 277} 278 279status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 280{ 281 return mTrack->attachAuxEffect(EffectId); 282} 283 284status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 285 sp<IMemory>* buffer) { 286 if (!mTrack->isTimedTrack()) 287 return INVALID_OPERATION; 288 289 PlaybackThread::TimedTrack* tt = 290 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 291 return tt->allocateTimedBuffer(size, buffer); 292} 293 294status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 295 int64_t pts) { 296 if (!mTrack->isTimedTrack()) 297 return INVALID_OPERATION; 298 299 if (buffer == 0 || buffer->pointer() == NULL) { 300 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 301 return BAD_VALUE; 302 } 303 304 PlaybackThread::TimedTrack* tt = 305 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 306 return tt->queueTimedBuffer(buffer, pts); 307} 308 309status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 310 const LinearTransform& xform, int target) { 311 312 if (!mTrack->isTimedTrack()) 313 return INVALID_OPERATION; 314 315 PlaybackThread::TimedTrack* tt = 316 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 317 return tt->setMediaTimeTransform( 318 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 319} 320 321status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 322 return mTrack->setParameters(keyValuePairs); 323} 324 325status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 326{ 327 return mTrack->getTimestamp(timestamp); 328} 329 330 331void AudioFlinger::TrackHandle::signal() 332{ 333 return mTrack->signal(); 334} 335 336status_t AudioFlinger::TrackHandle::onTransact( 337 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 338{ 339 return BnAudioTrack::onTransact(code, data, reply, flags); 340} 341 342// ---------------------------------------------------------------------------- 343 344// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 345AudioFlinger::PlaybackThread::Track::Track( 346 PlaybackThread *thread, 347 const sp<Client>& client, 348 audio_stream_type_t streamType, 349 uint32_t sampleRate, 350 audio_format_t format, 351 audio_channel_mask_t channelMask, 352 size_t frameCount, 353 const sp<IMemory>& sharedBuffer, 354 int sessionId, 355 int uid, 356 IAudioFlinger::track_flags_t flags) 357 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 358 sessionId, uid, flags, true /*isOut*/), 359 mFillingUpStatus(FS_INVALID), 360 // mRetryCount initialized later when needed 361 mSharedBuffer(sharedBuffer), 362 mStreamType(streamType), 363 mName(-1), // see note below 364 mMainBuffer(thread->mixBuffer()), 365 mAuxBuffer(NULL), 366 mAuxEffectId(0), mHasVolumeController(false), 367 mPresentationCompleteFrames(0), 368 mFastIndex(-1), 369 mCachedVolume(1.0), 370 mIsInvalid(false), 371 mAudioTrackServerProxy(NULL), 372 mResumeToStopping(false), 373 mFlushHwPending(false), 374 mPreviousValid(false), 375 mPreviousFramesWritten(0) 376 // mPreviousTimestamp 377{ 378 if (mCblk == NULL) { 379 return; 380 } 381 382 if (sharedBuffer == 0) { 383 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 384 mFrameSize); 385 } else { 386 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 387 mFrameSize); 388 } 389 mServerProxy = mAudioTrackServerProxy; 390 391 mName = thread->getTrackName_l(channelMask, sessionId); 392 if (mName < 0) { 393 ALOGE("no more track names available"); 394 return; 395 } 396 // only allocate a fast track index if we were able to allocate a normal track name 397 if (flags & IAudioFlinger::TRACK_FAST) { 398 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 399 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 400 int i = __builtin_ctz(thread->mFastTrackAvailMask); 401 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 402 // FIXME This is too eager. We allocate a fast track index before the 403 // fast track becomes active. Since fast tracks are a scarce resource, 404 // this means we are potentially denying other more important fast tracks from 405 // being created. It would be better to allocate the index dynamically. 406 mFastIndex = i; 407 // Read the initial underruns because this field is never cleared by the fast mixer 408 mObservedUnderruns = thread->getFastTrackUnderruns(i); 409 thread->mFastTrackAvailMask &= ~(1 << i); 410 } 411} 412 413AudioFlinger::PlaybackThread::Track::~Track() 414{ 415 ALOGV("PlaybackThread::Track destructor"); 416 417 // The destructor would clear mSharedBuffer, 418 // but it will not push the decremented reference count, 419 // leaving the client's IMemory dangling indefinitely. 420 // This prevents that leak. 421 if (mSharedBuffer != 0) { 422 mSharedBuffer.clear(); 423 // flush the binder command buffer 424 IPCThreadState::self()->flushCommands(); 425 } 426} 427 428status_t AudioFlinger::PlaybackThread::Track::initCheck() const 429{ 430 status_t status = TrackBase::initCheck(); 431 if (status == NO_ERROR && mName < 0) { 432 status = NO_MEMORY; 433 } 434 return status; 435} 436 437void AudioFlinger::PlaybackThread::Track::destroy() 438{ 439 // NOTE: destroyTrack_l() can remove a strong reference to this Track 440 // by removing it from mTracks vector, so there is a risk that this Tracks's 441 // destructor is called. As the destructor needs to lock mLock, 442 // we must acquire a strong reference on this Track before locking mLock 443 // here so that the destructor is called only when exiting this function. 444 // On the other hand, as long as Track::destroy() is only called by 445 // TrackHandle destructor, the TrackHandle still holds a strong ref on 446 // this Track with its member mTrack. 447 sp<Track> keep(this); 448 { // scope for mLock 449 sp<ThreadBase> thread = mThread.promote(); 450 if (thread != 0) { 451 Mutex::Autolock _l(thread->mLock); 452 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 453 bool wasActive = playbackThread->destroyTrack_l(this); 454 if (!isOutputTrack() && !wasActive) { 455 AudioSystem::releaseOutput(thread->id()); 456 } 457 } 458 } 459} 460 461/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 462{ 463 result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " 464 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 465} 466 467void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) 468{ 469 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 470 if (isFastTrack()) { 471 sprintf(buffer, " F %2d", mFastIndex); 472 } else if (mName >= AudioMixer::TRACK0) { 473 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 474 } else { 475 sprintf(buffer, " none"); 476 } 477 track_state state = mState; 478 char stateChar; 479 if (isTerminated()) { 480 stateChar = 'T'; 481 } else { 482 switch (state) { 483 case IDLE: 484 stateChar = 'I'; 485 break; 486 case STOPPING_1: 487 stateChar = 's'; 488 break; 489 case STOPPING_2: 490 stateChar = '5'; 491 break; 492 case STOPPED: 493 stateChar = 'S'; 494 break; 495 case RESUMING: 496 stateChar = 'R'; 497 break; 498 case ACTIVE: 499 stateChar = 'A'; 500 break; 501 case PAUSING: 502 stateChar = 'p'; 503 break; 504 case PAUSED: 505 stateChar = 'P'; 506 break; 507 case FLUSHED: 508 stateChar = 'F'; 509 break; 510 default: 511 stateChar = '?'; 512 break; 513 } 514 } 515 char nowInUnderrun; 516 switch (mObservedUnderruns.mBitFields.mMostRecent) { 517 case UNDERRUN_FULL: 518 nowInUnderrun = ' '; 519 break; 520 case UNDERRUN_PARTIAL: 521 nowInUnderrun = '<'; 522 break; 523 case UNDERRUN_EMPTY: 524 nowInUnderrun = '*'; 525 break; 526 default: 527 nowInUnderrun = '?'; 528 break; 529 } 530 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 531 "%08X %p %p 0x%03X %9u%c\n", 532 active ? "yes" : "no", 533 (mClient == 0) ? getpid_cached : mClient->pid(), 534 mStreamType, 535 mFormat, 536 mChannelMask, 537 mSessionId, 538 mFrameCount, 539 stateChar, 540 mFillingUpStatus, 541 mAudioTrackServerProxy->getSampleRate(), 542 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 543 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 544 mCblk->mServer, 545 mMainBuffer, 546 mAuxBuffer, 547 mCblk->mFlags, 548 mAudioTrackServerProxy->getUnderrunFrames(), 549 nowInUnderrun); 550} 551 552uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 553 return mAudioTrackServerProxy->getSampleRate(); 554} 555 556// AudioBufferProvider interface 557status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 558 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 559{ 560 ServerProxy::Buffer buf; 561 size_t desiredFrames = buffer->frameCount; 562 buf.mFrameCount = desiredFrames; 563 status_t status = mServerProxy->obtainBuffer(&buf); 564 buffer->frameCount = buf.mFrameCount; 565 buffer->raw = buf.mRaw; 566 if (buf.mFrameCount == 0) { 567 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 568 } 569 return status; 570} 571 572// releaseBuffer() is not overridden 573 574// ExtendedAudioBufferProvider interface 575 576// Note that framesReady() takes a mutex on the control block using tryLock(). 577// This could result in priority inversion if framesReady() is called by the normal mixer, 578// as the normal mixer thread runs at lower 579// priority than the client's callback thread: there is a short window within framesReady() 580// during which the normal mixer could be preempted, and the client callback would block. 581// Another problem can occur if framesReady() is called by the fast mixer: 582// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 583// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 584size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 585 return mAudioTrackServerProxy->framesReady(); 586} 587 588size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 589{ 590 return mAudioTrackServerProxy->framesReleased(); 591} 592 593// Don't call for fast tracks; the framesReady() could result in priority inversion 594bool AudioFlinger::PlaybackThread::Track::isReady() const { 595 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 596 return true; 597 } 598 599 if (isStopping()) { 600 if (framesReady() > 0) { 601 mFillingUpStatus = FS_FILLED; 602 } 603 return true; 604 } 605 606 if (framesReady() >= mFrameCount || 607 (mCblk->mFlags & CBLK_FORCEREADY)) { 608 mFillingUpStatus = FS_FILLED; 609 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 610 return true; 611 } 612 return false; 613} 614 615status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 616 int triggerSession __unused) 617{ 618 status_t status = NO_ERROR; 619 ALOGV("start(%d), calling pid %d session %d", 620 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 621 622 sp<ThreadBase> thread = mThread.promote(); 623 if (thread != 0) { 624 if (isOffloaded()) { 625 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 626 Mutex::Autolock _lth(thread->mLock); 627 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 628 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 629 (ec != 0 && ec->isNonOffloadableEnabled())) { 630 invalidate(); 631 return PERMISSION_DENIED; 632 } 633 } 634 Mutex::Autolock _lth(thread->mLock); 635 track_state state = mState; 636 // here the track could be either new, or restarted 637 // in both cases "unstop" the track 638 639 // initial state-stopping. next state-pausing. 640 // What if resume is called ? 641 642 if (state == PAUSED || state == PAUSING) { 643 if (mResumeToStopping) { 644 // happened we need to resume to STOPPING_1 645 mState = TrackBase::STOPPING_1; 646 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 647 } else { 648 mState = TrackBase::RESUMING; 649 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 650 } 651 } else { 652 mState = TrackBase::ACTIVE; 653 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 654 } 655 656 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 657 status = playbackThread->addTrack_l(this); 658 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 659 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 660 // restore previous state if start was rejected by policy manager 661 if (status == PERMISSION_DENIED) { 662 mState = state; 663 } 664 } 665 // track was already in the active list, not a problem 666 if (status == ALREADY_EXISTS) { 667 status = NO_ERROR; 668 } else { 669 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 670 // It is usually unsafe to access the server proxy from a binder thread. 671 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 672 // isn't looking at this track yet: we still hold the normal mixer thread lock, 673 // and for fast tracks the track is not yet in the fast mixer thread's active set. 674 ServerProxy::Buffer buffer; 675 buffer.mFrameCount = 1; 676 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 677 } 678 } else { 679 status = BAD_VALUE; 680 } 681 return status; 682} 683 684void AudioFlinger::PlaybackThread::Track::stop() 685{ 686 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 687 sp<ThreadBase> thread = mThread.promote(); 688 if (thread != 0) { 689 Mutex::Autolock _l(thread->mLock); 690 track_state state = mState; 691 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 692 // If the track is not active (PAUSED and buffers full), flush buffers 693 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 694 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 695 reset(); 696 mState = STOPPED; 697 } else if (!isFastTrack() && !isOffloaded()) { 698 mState = STOPPED; 699 } else { 700 // For fast tracks prepareTracks_l() will set state to STOPPING_2 701 // presentation is complete 702 // For an offloaded track this starts a drain and state will 703 // move to STOPPING_2 when drain completes and then STOPPED 704 mState = STOPPING_1; 705 } 706 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 707 playbackThread); 708 } 709 } 710} 711 712void AudioFlinger::PlaybackThread::Track::pause() 713{ 714 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 715 sp<ThreadBase> thread = mThread.promote(); 716 if (thread != 0) { 717 Mutex::Autolock _l(thread->mLock); 718 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 719 switch (mState) { 720 case STOPPING_1: 721 case STOPPING_2: 722 if (!isOffloaded()) { 723 /* nothing to do if track is not offloaded */ 724 break; 725 } 726 727 // Offloaded track was draining, we need to carry on draining when resumed 728 mResumeToStopping = true; 729 // fall through... 730 case ACTIVE: 731 case RESUMING: 732 mState = PAUSING; 733 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 734 playbackThread->broadcast_l(); 735 break; 736 737 default: 738 break; 739 } 740 } 741} 742 743void AudioFlinger::PlaybackThread::Track::flush() 744{ 745 ALOGV("flush(%d)", mName); 746 sp<ThreadBase> thread = mThread.promote(); 747 if (thread != 0) { 748 Mutex::Autolock _l(thread->mLock); 749 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 750 751 if (isOffloaded()) { 752 // If offloaded we allow flush during any state except terminated 753 // and keep the track active to avoid problems if user is seeking 754 // rapidly and underlying hardware has a significant delay handling 755 // a pause 756 if (isTerminated()) { 757 return; 758 } 759 760 ALOGV("flush: offload flush"); 761 reset(); 762 763 if (mState == STOPPING_1 || mState == STOPPING_2) { 764 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 765 mState = ACTIVE; 766 } 767 768 if (mState == ACTIVE) { 769 ALOGV("flush called in active state, resetting buffer time out retry count"); 770 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 771 } 772 773 mFlushHwPending = true; 774 mResumeToStopping = false; 775 } else { 776 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 777 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 778 return; 779 } 780 // No point remaining in PAUSED state after a flush => go to 781 // FLUSHED state 782 mState = FLUSHED; 783 // do not reset the track if it is still in the process of being stopped or paused. 784 // this will be done by prepareTracks_l() when the track is stopped. 785 // prepareTracks_l() will see mState == FLUSHED, then 786 // remove from active track list, reset(), and trigger presentation complete 787 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 788 reset(); 789 } 790 } 791 // Prevent flush being lost if the track is flushed and then resumed 792 // before mixer thread can run. This is important when offloading 793 // because the hardware buffer could hold a large amount of audio 794 playbackThread->broadcast_l(); 795 } 796} 797 798// must be called with thread lock held 799void AudioFlinger::PlaybackThread::Track::flushAck() 800{ 801 if (!isOffloaded()) 802 return; 803 804 mFlushHwPending = false; 805} 806 807void AudioFlinger::PlaybackThread::Track::reset() 808{ 809 // Do not reset twice to avoid discarding data written just after a flush and before 810 // the audioflinger thread detects the track is stopped. 811 if (!mResetDone) { 812 // Force underrun condition to avoid false underrun callback until first data is 813 // written to buffer 814 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 815 mFillingUpStatus = FS_FILLING; 816 mResetDone = true; 817 if (mState == FLUSHED) { 818 mState = IDLE; 819 } 820 } 821} 822 823status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 824{ 825 sp<ThreadBase> thread = mThread.promote(); 826 if (thread == 0) { 827 ALOGE("thread is dead"); 828 return FAILED_TRANSACTION; 829 } else if ((thread->type() == ThreadBase::DIRECT) || 830 (thread->type() == ThreadBase::OFFLOAD)) { 831 return thread->setParameters(keyValuePairs); 832 } else { 833 return PERMISSION_DENIED; 834 } 835} 836 837status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 838{ 839 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 840 if (isFastTrack()) { 841 // FIXME no lock held to set mPreviousValid = false 842 return INVALID_OPERATION; 843 } 844 sp<ThreadBase> thread = mThread.promote(); 845 if (thread == 0) { 846 // FIXME no lock held to set mPreviousValid = false 847 return INVALID_OPERATION; 848 } 849 Mutex::Autolock _l(thread->mLock); 850 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 851 if (!isOffloaded()) { 852 if (!playbackThread->mLatchQValid) { 853 mPreviousValid = false; 854 return INVALID_OPERATION; 855 } 856 uint32_t unpresentedFrames = 857 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 858 playbackThread->mSampleRate; 859 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 860 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 861 if (framesWritten < unpresentedFrames) { 862 mPreviousValid = false; 863 return INVALID_OPERATION; 864 } 865 mPreviousFramesWritten = framesWritten; 866 uint32_t position = framesWritten - unpresentedFrames; 867 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 868 if (checkPreviousTimestamp) { 869 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 870 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 871 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 872 ALOGW("Time is going backwards"); 873 } 874 // position can bobble slightly as an artifact; this hides the bobble 875 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 876 if ((position <= mPreviousTimestamp.mPosition) || 877 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 878 position = mPreviousTimestamp.mPosition; 879 time = mPreviousTimestamp.mTime; 880 } 881 } 882 timestamp.mPosition = position; 883 timestamp.mTime = time; 884 mPreviousTimestamp = timestamp; 885 mPreviousValid = true; 886 return NO_ERROR; 887 } 888 889 return playbackThread->getTimestamp_l(timestamp); 890} 891 892status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 893{ 894 status_t status = DEAD_OBJECT; 895 sp<ThreadBase> thread = mThread.promote(); 896 if (thread != 0) { 897 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 898 sp<AudioFlinger> af = mClient->audioFlinger(); 899 900 Mutex::Autolock _l(af->mLock); 901 902 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 903 904 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 905 Mutex::Autolock _dl(playbackThread->mLock); 906 Mutex::Autolock _sl(srcThread->mLock); 907 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 908 if (chain == 0) { 909 return INVALID_OPERATION; 910 } 911 912 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 913 if (effect == 0) { 914 return INVALID_OPERATION; 915 } 916 srcThread->removeEffect_l(effect); 917 status = playbackThread->addEffect_l(effect); 918 if (status != NO_ERROR) { 919 srcThread->addEffect_l(effect); 920 return INVALID_OPERATION; 921 } 922 // removeEffect_l() has stopped the effect if it was active so it must be restarted 923 if (effect->state() == EffectModule::ACTIVE || 924 effect->state() == EffectModule::STOPPING) { 925 effect->start(); 926 } 927 928 sp<EffectChain> dstChain = effect->chain().promote(); 929 if (dstChain == 0) { 930 srcThread->addEffect_l(effect); 931 return INVALID_OPERATION; 932 } 933 AudioSystem::unregisterEffect(effect->id()); 934 AudioSystem::registerEffect(&effect->desc(), 935 srcThread->id(), 936 dstChain->strategy(), 937 AUDIO_SESSION_OUTPUT_MIX, 938 effect->id()); 939 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 940 } 941 status = playbackThread->attachAuxEffect(this, EffectId); 942 } 943 return status; 944} 945 946void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 947{ 948 mAuxEffectId = EffectId; 949 mAuxBuffer = buffer; 950} 951 952bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 953 size_t audioHalFrames) 954{ 955 // a track is considered presented when the total number of frames written to audio HAL 956 // corresponds to the number of frames written when presentationComplete() is called for the 957 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 958 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 959 // to detect when all frames have been played. In this case framesWritten isn't 960 // useful because it doesn't always reflect whether there is data in the h/w 961 // buffers, particularly if a track has been paused and resumed during draining 962 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 963 mPresentationCompleteFrames, framesWritten); 964 if (mPresentationCompleteFrames == 0) { 965 mPresentationCompleteFrames = framesWritten + audioHalFrames; 966 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 967 mPresentationCompleteFrames, audioHalFrames); 968 } 969 970 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 971 ALOGV("presentationComplete() session %d complete: framesWritten %d", 972 mSessionId, framesWritten); 973 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 974 mAudioTrackServerProxy->setStreamEndDone(); 975 return true; 976 } 977 return false; 978} 979 980void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 981{ 982 for (size_t i = 0; i < mSyncEvents.size(); i++) { 983 if (mSyncEvents[i]->type() == type) { 984 mSyncEvents[i]->trigger(); 985 mSyncEvents.removeAt(i); 986 i--; 987 } 988 } 989} 990 991// implement VolumeBufferProvider interface 992 993gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 994{ 995 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 996 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 997 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 998 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 999 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1000 // track volumes come from shared memory, so can't be trusted and must be clamped 1001 if (vl > GAIN_FLOAT_UNITY) { 1002 vl = GAIN_FLOAT_UNITY; 1003 } 1004 if (vr > GAIN_FLOAT_UNITY) { 1005 vr = GAIN_FLOAT_UNITY; 1006 } 1007 // now apply the cached master volume and stream type volume; 1008 // this is trusted but lacks any synchronization or barrier so may be stale 1009 float v = mCachedVolume; 1010 vl *= v; 1011 vr *= v; 1012 // re-combine into packed minifloat 1013 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1014 // FIXME look at mute, pause, and stop flags 1015 return vlr; 1016} 1017 1018status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1019{ 1020 if (isTerminated() || mState == PAUSED || 1021 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1022 (mState == STOPPED)))) { 1023 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1024 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1025 event->cancel(); 1026 return INVALID_OPERATION; 1027 } 1028 (void) TrackBase::setSyncEvent(event); 1029 return NO_ERROR; 1030} 1031 1032void AudioFlinger::PlaybackThread::Track::invalidate() 1033{ 1034 // FIXME should use proxy, and needs work 1035 audio_track_cblk_t* cblk = mCblk; 1036 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1037 android_atomic_release_store(0x40000000, &cblk->mFutex); 1038 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1039 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1040 mIsInvalid = true; 1041} 1042 1043void AudioFlinger::PlaybackThread::Track::signal() 1044{ 1045 sp<ThreadBase> thread = mThread.promote(); 1046 if (thread != 0) { 1047 PlaybackThread *t = (PlaybackThread *)thread.get(); 1048 Mutex::Autolock _l(t->mLock); 1049 t->broadcast_l(); 1050 } 1051} 1052 1053//To be called with thread lock held 1054bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1055 1056 if (mState == RESUMING) 1057 return true; 1058 /* Resume is pending if track was stopping before pause was called */ 1059 if (mState == STOPPING_1 && 1060 mResumeToStopping) 1061 return true; 1062 1063 return false; 1064} 1065 1066//To be called with thread lock held 1067void AudioFlinger::PlaybackThread::Track::resumeAck() { 1068 1069 1070 if (mState == RESUMING) 1071 mState = ACTIVE; 1072 1073 // Other possibility of pending resume is stopping_1 state 1074 // Do not update the state from stopping as this prevents 1075 // drain being called. 1076 if (mState == STOPPING_1) { 1077 mResumeToStopping = false; 1078 } 1079} 1080// ---------------------------------------------------------------------------- 1081 1082sp<AudioFlinger::PlaybackThread::TimedTrack> 1083AudioFlinger::PlaybackThread::TimedTrack::create( 1084 PlaybackThread *thread, 1085 const sp<Client>& client, 1086 audio_stream_type_t streamType, 1087 uint32_t sampleRate, 1088 audio_format_t format, 1089 audio_channel_mask_t channelMask, 1090 size_t frameCount, 1091 const sp<IMemory>& sharedBuffer, 1092 int sessionId, 1093 int uid) 1094{ 1095 if (!client->reserveTimedTrack()) 1096 return 0; 1097 1098 return new TimedTrack( 1099 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1100 sharedBuffer, sessionId, uid); 1101} 1102 1103AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1104 PlaybackThread *thread, 1105 const sp<Client>& client, 1106 audio_stream_type_t streamType, 1107 uint32_t sampleRate, 1108 audio_format_t format, 1109 audio_channel_mask_t channelMask, 1110 size_t frameCount, 1111 const sp<IMemory>& sharedBuffer, 1112 int sessionId, 1113 int uid) 1114 : Track(thread, client, streamType, sampleRate, format, channelMask, 1115 frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED), 1116 mQueueHeadInFlight(false), 1117 mTrimQueueHeadOnRelease(false), 1118 mFramesPendingInQueue(0), 1119 mTimedSilenceBuffer(NULL), 1120 mTimedSilenceBufferSize(0), 1121 mTimedAudioOutputOnTime(false), 1122 mMediaTimeTransformValid(false) 1123{ 1124 LocalClock lc; 1125 mLocalTimeFreq = lc.getLocalFreq(); 1126 1127 mLocalTimeToSampleTransform.a_zero = 0; 1128 mLocalTimeToSampleTransform.b_zero = 0; 1129 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1130 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1131 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1132 &mLocalTimeToSampleTransform.a_to_b_denom); 1133 1134 mMediaTimeToSampleTransform.a_zero = 0; 1135 mMediaTimeToSampleTransform.b_zero = 0; 1136 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1137 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1138 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1139 &mMediaTimeToSampleTransform.a_to_b_denom); 1140} 1141 1142AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1143 mClient->releaseTimedTrack(); 1144 delete [] mTimedSilenceBuffer; 1145} 1146 1147status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1148 size_t size, sp<IMemory>* buffer) { 1149 1150 Mutex::Autolock _l(mTimedBufferQueueLock); 1151 1152 trimTimedBufferQueue_l(); 1153 1154 // lazily initialize the shared memory heap for timed buffers 1155 if (mTimedMemoryDealer == NULL) { 1156 const int kTimedBufferHeapSize = 512 << 10; 1157 1158 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1159 "AudioFlingerTimed"); 1160 if (mTimedMemoryDealer == NULL) { 1161 return NO_MEMORY; 1162 } 1163 } 1164 1165 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1166 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1167 return NO_MEMORY; 1168 } 1169 1170 *buffer = newBuffer; 1171 return NO_ERROR; 1172} 1173 1174// caller must hold mTimedBufferQueueLock 1175void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1176 int64_t mediaTimeNow; 1177 { 1178 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1179 if (!mMediaTimeTransformValid) 1180 return; 1181 1182 int64_t targetTimeNow; 1183 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1184 ? mCCHelper.getCommonTime(&targetTimeNow) 1185 : mCCHelper.getLocalTime(&targetTimeNow); 1186 1187 if (OK != res) 1188 return; 1189 1190 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1191 &mediaTimeNow)) { 1192 return; 1193 } 1194 } 1195 1196 size_t trimEnd; 1197 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1198 int64_t bufEnd; 1199 1200 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1201 // We have a next buffer. Just use its PTS as the PTS of the frame 1202 // following the last frame in this buffer. If the stream is sparse 1203 // (ie, there are deliberate gaps left in the stream which should be 1204 // filled with silence by the TimedAudioTrack), then this can result 1205 // in one extra buffer being left un-trimmed when it could have 1206 // been. In general, this is not typical, and we would rather 1207 // optimized away the TS calculation below for the more common case 1208 // where PTSes are contiguous. 1209 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1210 } else { 1211 // We have no next buffer. Compute the PTS of the frame following 1212 // the last frame in this buffer by computing the duration of of 1213 // this frame in media time units and adding it to the PTS of the 1214 // buffer. 1215 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1216 / mFrameSize; 1217 1218 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1219 &bufEnd)) { 1220 ALOGE("Failed to convert frame count of %lld to media time" 1221 " duration" " (scale factor %d/%u) in %s", 1222 frameCount, 1223 mMediaTimeToSampleTransform.a_to_b_numer, 1224 mMediaTimeToSampleTransform.a_to_b_denom, 1225 __PRETTY_FUNCTION__); 1226 break; 1227 } 1228 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1229 } 1230 1231 if (bufEnd > mediaTimeNow) 1232 break; 1233 1234 // Is the buffer we want to use in the middle of a mix operation right 1235 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1236 // from the mixer which should be coming back shortly. 1237 if (!trimEnd && mQueueHeadInFlight) { 1238 mTrimQueueHeadOnRelease = true; 1239 } 1240 } 1241 1242 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1243 if (trimStart < trimEnd) { 1244 // Update the bookkeeping for framesReady() 1245 for (size_t i = trimStart; i < trimEnd; ++i) { 1246 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1247 } 1248 1249 // Now actually remove the buffers from the queue. 1250 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1251 } 1252} 1253 1254void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1255 const char* logTag) { 1256 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1257 "%s called (reason \"%s\"), but timed buffer queue has no" 1258 " elements to trim.", __FUNCTION__, logTag); 1259 1260 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1261 mTimedBufferQueue.removeAt(0); 1262} 1263 1264void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1265 const TimedBuffer& buf, 1266 const char* logTag __unused) { 1267 uint32_t bufBytes = buf.buffer()->size(); 1268 uint32_t consumedAlready = buf.position(); 1269 1270 ALOG_ASSERT(consumedAlready <= bufBytes, 1271 "Bad bookkeeping while updating frames pending. Timed buffer is" 1272 " only %u bytes long, but claims to have consumed %u" 1273 " bytes. (update reason: \"%s\")", 1274 bufBytes, consumedAlready, logTag); 1275 1276 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1277 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1278 "Bad bookkeeping while updating frames pending. Should have at" 1279 " least %u queued frames, but we think we have only %u. (update" 1280 " reason: \"%s\")", 1281 bufFrames, mFramesPendingInQueue, logTag); 1282 1283 mFramesPendingInQueue -= bufFrames; 1284} 1285 1286status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1287 const sp<IMemory>& buffer, int64_t pts) { 1288 1289 { 1290 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1291 if (!mMediaTimeTransformValid) 1292 return INVALID_OPERATION; 1293 } 1294 1295 Mutex::Autolock _l(mTimedBufferQueueLock); 1296 1297 uint32_t bufFrames = buffer->size() / mFrameSize; 1298 mFramesPendingInQueue += bufFrames; 1299 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1300 1301 return NO_ERROR; 1302} 1303 1304status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1305 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1306 1307 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1308 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1309 target); 1310 1311 if (!(target == TimedAudioTrack::LOCAL_TIME || 1312 target == TimedAudioTrack::COMMON_TIME)) { 1313 return BAD_VALUE; 1314 } 1315 1316 Mutex::Autolock lock(mMediaTimeTransformLock); 1317 mMediaTimeTransform = xform; 1318 mMediaTimeTransformTarget = target; 1319 mMediaTimeTransformValid = true; 1320 1321 return NO_ERROR; 1322} 1323 1324#define min(a, b) ((a) < (b) ? (a) : (b)) 1325 1326// implementation of getNextBuffer for tracks whose buffers have timestamps 1327status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1328 AudioBufferProvider::Buffer* buffer, int64_t pts) 1329{ 1330 if (pts == AudioBufferProvider::kInvalidPTS) { 1331 buffer->raw = NULL; 1332 buffer->frameCount = 0; 1333 mTimedAudioOutputOnTime = false; 1334 return INVALID_OPERATION; 1335 } 1336 1337 Mutex::Autolock _l(mTimedBufferQueueLock); 1338 1339 ALOG_ASSERT(!mQueueHeadInFlight, 1340 "getNextBuffer called without releaseBuffer!"); 1341 1342 while (true) { 1343 1344 // if we have no timed buffers, then fail 1345 if (mTimedBufferQueue.isEmpty()) { 1346 buffer->raw = NULL; 1347 buffer->frameCount = 0; 1348 return NOT_ENOUGH_DATA; 1349 } 1350 1351 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1352 1353 // calculate the PTS of the head of the timed buffer queue expressed in 1354 // local time 1355 int64_t headLocalPTS; 1356 { 1357 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1358 1359 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1360 1361 if (mMediaTimeTransform.a_to_b_denom == 0) { 1362 // the transform represents a pause, so yield silence 1363 timedYieldSilence_l(buffer->frameCount, buffer); 1364 return NO_ERROR; 1365 } 1366 1367 int64_t transformedPTS; 1368 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1369 &transformedPTS)) { 1370 // the transform failed. this shouldn't happen, but if it does 1371 // then just drop this buffer 1372 ALOGW("timedGetNextBuffer transform failed"); 1373 buffer->raw = NULL; 1374 buffer->frameCount = 0; 1375 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1376 return NO_ERROR; 1377 } 1378 1379 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1380 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1381 &headLocalPTS)) { 1382 buffer->raw = NULL; 1383 buffer->frameCount = 0; 1384 return INVALID_OPERATION; 1385 } 1386 } else { 1387 headLocalPTS = transformedPTS; 1388 } 1389 } 1390 1391 uint32_t sr = sampleRate(); 1392 1393 // adjust the head buffer's PTS to reflect the portion of the head buffer 1394 // that has already been consumed 1395 int64_t effectivePTS = headLocalPTS + 1396 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1397 1398 // Calculate the delta in samples between the head of the input buffer 1399 // queue and the start of the next output buffer that will be written. 1400 // If the transformation fails because of over or underflow, it means 1401 // that the sample's position in the output stream is so far out of 1402 // whack that it should just be dropped. 1403 int64_t sampleDelta; 1404 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1405 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1406 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1407 " mix"); 1408 continue; 1409 } 1410 if (!mLocalTimeToSampleTransform.doForwardTransform( 1411 (effectivePTS - pts) << 32, &sampleDelta)) { 1412 ALOGV("*** too late during sample rate transform: dropped buffer"); 1413 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1414 continue; 1415 } 1416 1417 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1418 " sampleDelta=[%d.%08x]", 1419 head.pts(), head.position(), pts, 1420 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1421 + (sampleDelta >> 32)), 1422 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1423 1424 // if the delta between the ideal placement for the next input sample and 1425 // the current output position is within this threshold, then we will 1426 // concatenate the next input samples to the previous output 1427 const int64_t kSampleContinuityThreshold = 1428 (static_cast<int64_t>(sr) << 32) / 250; 1429 1430 // if this is the first buffer of audio that we're emitting from this track 1431 // then it should be almost exactly on time. 1432 const int64_t kSampleStartupThreshold = 1LL << 32; 1433 1434 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1435 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1436 // the next input is close enough to being on time, so concatenate it 1437 // with the last output 1438 timedYieldSamples_l(buffer); 1439 1440 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1441 head.position(), buffer->frameCount); 1442 return NO_ERROR; 1443 } 1444 1445 // Looks like our output is not on time. Reset our on timed status. 1446 // Next time we mix samples from our input queue, then should be within 1447 // the StartupThreshold. 1448 mTimedAudioOutputOnTime = false; 1449 if (sampleDelta > 0) { 1450 // the gap between the current output position and the proper start of 1451 // the next input sample is too big, so fill it with silence 1452 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1453 1454 timedYieldSilence_l(framesUntilNextInput, buffer); 1455 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1456 return NO_ERROR; 1457 } else { 1458 // the next input sample is late 1459 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1460 size_t onTimeSamplePosition = 1461 head.position() + lateFrames * mFrameSize; 1462 1463 if (onTimeSamplePosition > head.buffer()->size()) { 1464 // all the remaining samples in the head are too late, so 1465 // drop it and move on 1466 ALOGV("*** too late: dropped buffer"); 1467 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1468 continue; 1469 } else { 1470 // skip over the late samples 1471 head.setPosition(onTimeSamplePosition); 1472 1473 // yield the available samples 1474 timedYieldSamples_l(buffer); 1475 1476 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1477 return NO_ERROR; 1478 } 1479 } 1480 } 1481} 1482 1483// Yield samples from the timed buffer queue head up to the given output 1484// buffer's capacity. 1485// 1486// Caller must hold mTimedBufferQueueLock 1487void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1488 AudioBufferProvider::Buffer* buffer) { 1489 1490 const TimedBuffer& head = mTimedBufferQueue[0]; 1491 1492 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1493 head.position()); 1494 1495 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1496 mFrameSize); 1497 size_t framesRequested = buffer->frameCount; 1498 buffer->frameCount = min(framesLeftInHead, framesRequested); 1499 1500 mQueueHeadInFlight = true; 1501 mTimedAudioOutputOnTime = true; 1502} 1503 1504// Yield samples of silence up to the given output buffer's capacity 1505// 1506// Caller must hold mTimedBufferQueueLock 1507void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1508 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1509 1510 // lazily allocate a buffer filled with silence 1511 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1512 delete [] mTimedSilenceBuffer; 1513 mTimedSilenceBufferSize = numFrames * mFrameSize; 1514 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1515 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1516 } 1517 1518 buffer->raw = mTimedSilenceBuffer; 1519 size_t framesRequested = buffer->frameCount; 1520 buffer->frameCount = min(numFrames, framesRequested); 1521 1522 mTimedAudioOutputOnTime = false; 1523} 1524 1525// AudioBufferProvider interface 1526void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1527 AudioBufferProvider::Buffer* buffer) { 1528 1529 Mutex::Autolock _l(mTimedBufferQueueLock); 1530 1531 // If the buffer which was just released is part of the buffer at the head 1532 // of the queue, be sure to update the amt of the buffer which has been 1533 // consumed. If the buffer being returned is not part of the head of the 1534 // queue, its either because the buffer is part of the silence buffer, or 1535 // because the head of the timed queue was trimmed after the mixer called 1536 // getNextBuffer but before the mixer called releaseBuffer. 1537 if (buffer->raw == mTimedSilenceBuffer) { 1538 ALOG_ASSERT(!mQueueHeadInFlight, 1539 "Queue head in flight during release of silence buffer!"); 1540 goto done; 1541 } 1542 1543 ALOG_ASSERT(mQueueHeadInFlight, 1544 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1545 " head in flight."); 1546 1547 if (mTimedBufferQueue.size()) { 1548 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1549 1550 void* start = head.buffer()->pointer(); 1551 void* end = reinterpret_cast<void*>( 1552 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1553 + head.buffer()->size()); 1554 1555 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1556 "released buffer not within the head of the timed buffer" 1557 " queue; qHead = [%p, %p], released buffer = %p", 1558 start, end, buffer->raw); 1559 1560 head.setPosition(head.position() + 1561 (buffer->frameCount * mFrameSize)); 1562 mQueueHeadInFlight = false; 1563 1564 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1565 "Bad bookkeeping during releaseBuffer! Should have at" 1566 " least %u queued frames, but we think we have only %u", 1567 buffer->frameCount, mFramesPendingInQueue); 1568 1569 mFramesPendingInQueue -= buffer->frameCount; 1570 1571 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1572 || mTrimQueueHeadOnRelease) { 1573 trimTimedBufferQueueHead_l("releaseBuffer"); 1574 mTrimQueueHeadOnRelease = false; 1575 } 1576 } else { 1577 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1578 " buffers in the timed buffer queue"); 1579 } 1580 1581done: 1582 buffer->raw = 0; 1583 buffer->frameCount = 0; 1584} 1585 1586size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1587 Mutex::Autolock _l(mTimedBufferQueueLock); 1588 return mFramesPendingInQueue; 1589} 1590 1591AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1592 : mPTS(0), mPosition(0) {} 1593 1594AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1595 const sp<IMemory>& buffer, int64_t pts) 1596 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1597 1598 1599// ---------------------------------------------------------------------------- 1600 1601AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1602 PlaybackThread *playbackThread, 1603 DuplicatingThread *sourceThread, 1604 uint32_t sampleRate, 1605 audio_format_t format, 1606 audio_channel_mask_t channelMask, 1607 size_t frameCount, 1608 int uid) 1609 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1610 NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT), 1611 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1612{ 1613 1614 if (mCblk != NULL) { 1615 mOutBuffer.frameCount = 0; 1616 playbackThread->mTracks.add(this); 1617 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1618 "frameCount %u, mChannelMask 0x%08x", 1619 mCblk, mBuffer, 1620 frameCount, mChannelMask); 1621 // since client and server are in the same process, 1622 // the buffer has the same virtual address on both sides 1623 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1624 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1625 mClientProxy->setSendLevel(0.0); 1626 mClientProxy->setSampleRate(sampleRate); 1627 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1628 true /*clientInServer*/); 1629 } else { 1630 ALOGW("Error creating output track on thread %p", playbackThread); 1631 } 1632} 1633 1634AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1635{ 1636 clearBufferQueue(); 1637 delete mClientProxy; 1638 // superclass destructor will now delete the server proxy and shared memory both refer to 1639} 1640 1641status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1642 int triggerSession) 1643{ 1644 status_t status = Track::start(event, triggerSession); 1645 if (status != NO_ERROR) { 1646 return status; 1647 } 1648 1649 mActive = true; 1650 mRetryCount = 127; 1651 return status; 1652} 1653 1654void AudioFlinger::PlaybackThread::OutputTrack::stop() 1655{ 1656 Track::stop(); 1657 clearBufferQueue(); 1658 mOutBuffer.frameCount = 0; 1659 mActive = false; 1660} 1661 1662bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1663{ 1664 Buffer *pInBuffer; 1665 Buffer inBuffer; 1666 uint32_t channelCount = mChannelCount; 1667 bool outputBufferFull = false; 1668 inBuffer.frameCount = frames; 1669 inBuffer.i16 = data; 1670 1671 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1672 1673 if (!mActive && frames != 0) { 1674 start(); 1675 sp<ThreadBase> thread = mThread.promote(); 1676 if (thread != 0) { 1677 MixerThread *mixerThread = (MixerThread *)thread.get(); 1678 if (mFrameCount > frames) { 1679 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1680 uint32_t startFrames = (mFrameCount - frames); 1681 pInBuffer = new Buffer; 1682 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1683 pInBuffer->frameCount = startFrames; 1684 pInBuffer->i16 = pInBuffer->mBuffer; 1685 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1686 mBufferQueue.add(pInBuffer); 1687 } else { 1688 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1689 } 1690 } 1691 } 1692 } 1693 1694 while (waitTimeLeftMs) { 1695 // First write pending buffers, then new data 1696 if (mBufferQueue.size()) { 1697 pInBuffer = mBufferQueue.itemAt(0); 1698 } else { 1699 pInBuffer = &inBuffer; 1700 } 1701 1702 if (pInBuffer->frameCount == 0) { 1703 break; 1704 } 1705 1706 if (mOutBuffer.frameCount == 0) { 1707 mOutBuffer.frameCount = pInBuffer->frameCount; 1708 nsecs_t startTime = systemTime(); 1709 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1710 if (status != NO_ERROR) { 1711 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1712 mThread.unsafe_get(), status); 1713 outputBufferFull = true; 1714 break; 1715 } 1716 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1717 if (waitTimeLeftMs >= waitTimeMs) { 1718 waitTimeLeftMs -= waitTimeMs; 1719 } else { 1720 waitTimeLeftMs = 0; 1721 } 1722 } 1723 1724 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1725 pInBuffer->frameCount; 1726 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1727 Proxy::Buffer buf; 1728 buf.mFrameCount = outFrames; 1729 buf.mRaw = NULL; 1730 mClientProxy->releaseBuffer(&buf); 1731 pInBuffer->frameCount -= outFrames; 1732 pInBuffer->i16 += outFrames * channelCount; 1733 mOutBuffer.frameCount -= outFrames; 1734 mOutBuffer.i16 += outFrames * channelCount; 1735 1736 if (pInBuffer->frameCount == 0) { 1737 if (mBufferQueue.size()) { 1738 mBufferQueue.removeAt(0); 1739 delete [] pInBuffer->mBuffer; 1740 delete pInBuffer; 1741 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1742 mThread.unsafe_get(), mBufferQueue.size()); 1743 } else { 1744 break; 1745 } 1746 } 1747 } 1748 1749 // If we could not write all frames, allocate a buffer and queue it for next time. 1750 if (inBuffer.frameCount) { 1751 sp<ThreadBase> thread = mThread.promote(); 1752 if (thread != 0 && !thread->standby()) { 1753 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1754 pInBuffer = new Buffer; 1755 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1756 pInBuffer->frameCount = inBuffer.frameCount; 1757 pInBuffer->i16 = pInBuffer->mBuffer; 1758 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1759 sizeof(int16_t)); 1760 mBufferQueue.add(pInBuffer); 1761 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1762 mThread.unsafe_get(), mBufferQueue.size()); 1763 } else { 1764 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1765 mThread.unsafe_get(), this); 1766 } 1767 } 1768 } 1769 1770 // Calling write() with a 0 length buffer, means that no more data will be written: 1771 // If no more buffers are pending, fill output track buffer to make sure it is started 1772 // by output mixer. 1773 if (frames == 0 && mBufferQueue.size() == 0) { 1774 // FIXME borken, replace by getting framesReady() from proxy 1775 size_t user = 0; // was mCblk->user 1776 if (user < mFrameCount) { 1777 frames = mFrameCount - user; 1778 pInBuffer = new Buffer; 1779 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1780 pInBuffer->frameCount = frames; 1781 pInBuffer->i16 = pInBuffer->mBuffer; 1782 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1783 mBufferQueue.add(pInBuffer); 1784 } else if (mActive) { 1785 stop(); 1786 } 1787 } 1788 1789 return outputBufferFull; 1790} 1791 1792status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1793 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1794{ 1795 ClientProxy::Buffer buf; 1796 buf.mFrameCount = buffer->frameCount; 1797 struct timespec timeout; 1798 timeout.tv_sec = waitTimeMs / 1000; 1799 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1800 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1801 buffer->frameCount = buf.mFrameCount; 1802 buffer->raw = buf.mRaw; 1803 return status; 1804} 1805 1806void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1807{ 1808 size_t size = mBufferQueue.size(); 1809 1810 for (size_t i = 0; i < size; i++) { 1811 Buffer *pBuffer = mBufferQueue.itemAt(i); 1812 delete [] pBuffer->mBuffer; 1813 delete pBuffer; 1814 } 1815 mBufferQueue.clear(); 1816} 1817 1818 1819// ---------------------------------------------------------------------------- 1820// Record 1821// ---------------------------------------------------------------------------- 1822 1823AudioFlinger::RecordHandle::RecordHandle( 1824 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1825 : BnAudioRecord(), 1826 mRecordTrack(recordTrack) 1827{ 1828} 1829 1830AudioFlinger::RecordHandle::~RecordHandle() { 1831 stop_nonvirtual(); 1832 mRecordTrack->destroy(); 1833} 1834 1835status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1836 int triggerSession) { 1837 ALOGV("RecordHandle::start()"); 1838 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1839} 1840 1841void AudioFlinger::RecordHandle::stop() { 1842 stop_nonvirtual(); 1843} 1844 1845void AudioFlinger::RecordHandle::stop_nonvirtual() { 1846 ALOGV("RecordHandle::stop()"); 1847 mRecordTrack->stop(); 1848} 1849 1850status_t AudioFlinger::RecordHandle::onTransact( 1851 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1852{ 1853 return BnAudioRecord::onTransact(code, data, reply, flags); 1854} 1855 1856// ---------------------------------------------------------------------------- 1857 1858// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1859AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1860 RecordThread *thread, 1861 const sp<Client>& client, 1862 uint32_t sampleRate, 1863 audio_format_t format, 1864 audio_channel_mask_t channelMask, 1865 size_t frameCount, 1866 int sessionId, 1867 int uid, 1868 IAudioFlinger::track_flags_t flags) 1869 : TrackBase(thread, client, sampleRate, format, 1870 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, 1871 flags, false /*isOut*/, 1872 (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/), 1873 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1874 // See real initialization of mRsmpInFront at RecordThread::start() 1875 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1876{ 1877 if (mCblk == NULL) { 1878 return; 1879 } 1880 1881 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1882 1883 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 1884 // FIXME I don't understand either of the channel count checks 1885 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 1886 channelCount <= FCC_2) { 1887 // sink SR 1888 mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate); 1889 // source SR 1890 mResampler->setSampleRate(thread->mSampleRate); 1891 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 1892 mResamplerBufferProvider = new ResamplerBufferProvider(this); 1893 } 1894} 1895 1896AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1897{ 1898 ALOGV("%s", __func__); 1899 delete mResampler; 1900 delete[] mRsmpOutBuffer; 1901 delete mResamplerBufferProvider; 1902} 1903 1904// AudioBufferProvider interface 1905status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1906 int64_t pts __unused) 1907{ 1908 ServerProxy::Buffer buf; 1909 buf.mFrameCount = buffer->frameCount; 1910 status_t status = mServerProxy->obtainBuffer(&buf); 1911 buffer->frameCount = buf.mFrameCount; 1912 buffer->raw = buf.mRaw; 1913 if (buf.mFrameCount == 0) { 1914 // FIXME also wake futex so that overrun is noticed more quickly 1915 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1916 } 1917 return status; 1918} 1919 1920status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1921 int triggerSession) 1922{ 1923 sp<ThreadBase> thread = mThread.promote(); 1924 if (thread != 0) { 1925 RecordThread *recordThread = (RecordThread *)thread.get(); 1926 return recordThread->start(this, event, triggerSession); 1927 } else { 1928 return BAD_VALUE; 1929 } 1930} 1931 1932void AudioFlinger::RecordThread::RecordTrack::stop() 1933{ 1934 sp<ThreadBase> thread = mThread.promote(); 1935 if (thread != 0) { 1936 RecordThread *recordThread = (RecordThread *)thread.get(); 1937 if (recordThread->stop(this)) { 1938 AudioSystem::stopInput(recordThread->id()); 1939 } 1940 } 1941} 1942 1943void AudioFlinger::RecordThread::RecordTrack::destroy() 1944{ 1945 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1946 sp<RecordTrack> keep(this); 1947 { 1948 sp<ThreadBase> thread = mThread.promote(); 1949 if (thread != 0) { 1950 if (mState == ACTIVE || mState == RESUMING) { 1951 AudioSystem::stopInput(thread->id()); 1952 } 1953 AudioSystem::releaseInput(thread->id()); 1954 Mutex::Autolock _l(thread->mLock); 1955 RecordThread *recordThread = (RecordThread *) thread.get(); 1956 recordThread->destroyTrack_l(this); 1957 } 1958 } 1959} 1960 1961void AudioFlinger::RecordThread::RecordTrack::invalidate() 1962{ 1963 // FIXME should use proxy, and needs work 1964 audio_track_cblk_t* cblk = mCblk; 1965 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1966 android_atomic_release_store(0x40000000, &cblk->mFutex); 1967 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1968 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1969} 1970 1971 1972/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1973{ 1974 result.append(" Active Client Fmt Chn mask Session S Server fCount Resampling\n"); 1975} 1976 1977void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 1978{ 1979 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n", 1980 active ? "yes" : "no", 1981 (mClient == 0) ? getpid_cached : mClient->pid(), 1982 mFormat, 1983 mChannelMask, 1984 mSessionId, 1985 mState, 1986 mCblk->mServer, 1987 mFrameCount, 1988 mResampler != NULL); 1989 1990} 1991 1992void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 1993{ 1994 if (event == mSyncStartEvent) { 1995 ssize_t framesToDrop = 0; 1996 sp<ThreadBase> threadBase = mThread.promote(); 1997 if (threadBase != 0) { 1998 // TODO: use actual buffer filling status instead of 2 buffers when info is available 1999 // from audio HAL 2000 framesToDrop = threadBase->mFrameCount * 2; 2001 } 2002 mFramesToDrop = framesToDrop; 2003 } 2004} 2005 2006void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2007{ 2008 if (mSyncStartEvent != 0) { 2009 mSyncStartEvent->cancel(); 2010 mSyncStartEvent.clear(); 2011 } 2012 mFramesToDrop = 0; 2013} 2014 2015}; // namespace android 2016