AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
new file mode 100644
index 0000000..2c6ba8b
--- /dev/null
+++ b/services/audioflinger/Tracks.cpp
@@ -0,0 +1,1789 @@
+/*
+**
+** Copyright 2012, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message. In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well. Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on. Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// TrackBase
+// ----------------------------------------------------------------------------
+
+// TrackBase constructor must be called with AudioFlinger::mLock held
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+ ThreadBase *thread,
+ const sp<Client>& client,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : RefBase(),
+ mThread(thread),
+ mClient(client),
+ mCblk(NULL),
+ // mBuffer
+ // mBufferEnd
+ mStepCount(0),
+ mState(IDLE),
+ mSampleRate(sampleRate),
+ mFormat(format),
+ mChannelMask(channelMask),
+ mChannelCount(popcount(channelMask)),
+ mFrameSize(audio_is_linear_pcm(format) ?
+ mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
+ mFrameCount(frameCount),
+ mStepServerFailed(false),
+ mSessionId(sessionId)
+{
+ // client == 0 implies sharedBuffer == 0
+ ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
+
+ ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
+ sharedBuffer->size());
+
+ // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+ size_t size = sizeof(audio_track_cblk_t);
+ size_t bufferSize = frameCount * mFrameSize;
+ if (sharedBuffer == 0) {
+ size += bufferSize;
+ }
+
+ if (client != 0) {
+ mCblkMemory = client->heap()->allocate(size);
+ if (mCblkMemory != 0) {
+ mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
+ // can't assume mCblk != NULL
+ } else {
+ ALOGE("not enough memory for AudioTrack size=%u", size);
+ client->heap()->dump("AudioTrack");
+ return;
+ }
+ } else {
+ mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+ // assume mCblk != NULL
+ }
+
+ // construct the shared structure in-place.
+ if (mCblk != NULL) {
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount_ = frameCount;
+ mCblk->sampleRate = sampleRate;
+// uncomment the following lines to quickly test 32-bit wraparound
+// mCblk->user = 0xffff0000;
+// mCblk->server = 0xffff0000;
+// mCblk->userBase = 0xffff0000;
+// mCblk->serverBase = 0xffff0000;
+ if (sharedBuffer == 0) {
+ mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffer, 0, bufferSize);
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer (other flags are cleared)
+ mCblk->flags = CBLK_UNDERRUN;
+ } else {
+ mBuffer = sharedBuffer->pointer();
+ }
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ }
+}
+
+AudioFlinger::ThreadBase::TrackBase::~TrackBase()
+{
+ if (mCblk != NULL) {
+ if (mClient == 0) {
+ delete mCblk;
+ } else {
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ }
+ }
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ if (mClient != 0) {
+ // Client destructor must run with AudioFlinger mutex locked
+ Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+ // If the client's reference count drops to zero, the associated destructor
+ // must run with AudioFlinger lock held. Thus the explicit clear() rather than
+ // relying on the automatic clear() at end of scope.
+ mClient.clear();
+ }
+}
+
+// AudioBufferProvider interface
+// getNextBuffer() = 0;
+// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
+void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ buffer->raw = NULL;
+ mStepCount = buffer->frameCount;
+ // FIXME See note at getNextBuffer()
+ (void) step(); // ignore return value of step()
+ buffer->frameCount = 0;
+}
+
+bool AudioFlinger::ThreadBase::TrackBase::step() {
+ bool result;
+ audio_track_cblk_t* cblk = this->cblk();
+
+ result = cblk->stepServer(mStepCount, mFrameCount, isOut());
+ if (!result) {
+ ALOGV("stepServer failed acquiring cblk mutex");
+ mStepServerFailed = true;
+ }
+ return result;
+}
+
+void AudioFlinger::ThreadBase::TrackBase::reset() {
+ audio_track_cblk_t* cblk = this->cblk();
+
+ cblk->user = 0;
+ cblk->server = 0;
+ cblk->userBase = 0;
+ cblk->serverBase = 0;
+ mStepServerFailed = false;
+ ALOGV("TrackBase::reset");
+}
+
+uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
+ return mCblk->sampleRate;
+}
+
+void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+ audio_track_cblk_t* cblk = this->cblk();
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
+ int8_t *bufferEnd = bufferStart + frames * mFrameSize;
+
+ // Check validity of returned pointer in case the track control block would have been corrupted.
+ ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
+ "TrackBase::getBuffer buffer out of range:\n"
+ " start: %p, end %p , mBuffer %p mBufferEnd %p\n"
+ " server %u, serverBase %u, user %u, userBase %u, frameSize %u",
+ bufferStart, bufferEnd, mBuffer, mBufferEnd,
+ cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
+
+ return bufferStart;
+}
+
+status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
+{
+ mSyncEvents.add(event);
+ return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// Playback
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
+ : BnAudioTrack(),
+ mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+ // just stop the track on deletion, associated resources
+ // will be freed from the main thread once all pending buffers have
+ // been played. Unless it's not in the active track list, in which
+ // case we free everything now...
+ mTrack->destroy();
+}
+
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+ return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+ return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+ mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+ mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+ mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+ mTrack->pause();
+}
+
+status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
+{
+ return mTrack->attachAuxEffect(EffectId);
+}
+
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+ const LinearTransform& xform, int target) {
+
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->setMediaTimeTransform(
+ xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
+status_t AudioFlinger::TrackHandle::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId,
+ IAudioFlinger::track_flags_t flags)
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
+ sessionId),
+ mMute(false),
+ mFillingUpStatus(FS_INVALID),
+ // mRetryCount initialized later when needed
+ mSharedBuffer(sharedBuffer),
+ mStreamType(streamType),
+ mName(-1), // see note below
+ mMainBuffer(thread->mixBuffer()),
+ mAuxBuffer(NULL),
+ mAuxEffectId(0), mHasVolumeController(false),
+ mPresentationCompleteFrames(0),
+ mFlags(flags),
+ mFastIndex(-1),
+ mUnderrunCount(0),
+ mCachedVolume(1.0)
+{
+ if (mCblk != NULL) {
+ // to avoid leaking a track name, do not allocate one unless there is an mCblk
+ mName = thread->getTrackName_l(channelMask, sessionId);
+ mCblk->mName = mName;
+ if (mName < 0) {
+ ALOGE("no more track names available");
+ return;
+ }
+ // only allocate a fast track index if we were able to allocate a normal track name
+ if (flags & IAudioFlinger::TRACK_FAST) {
+ ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
+ int i = __builtin_ctz(thread->mFastTrackAvailMask);
+ ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
+ // FIXME This is too eager. We allocate a fast track index before the
+ // fast track becomes active. Since fast tracks are a scarce resource,
+ // this means we are potentially denying other more important fast tracks from
+ // being created. It would be better to allocate the index dynamically.
+ mFastIndex = i;
+ mCblk->mName = i;
+ // Read the initial underruns because this field is never cleared by the fast mixer
+ mObservedUnderruns = thread->getFastTrackUnderruns(i);
+ thread->mFastTrackAvailMask &= ~(1 << i);
+ }
+ }
+ ALOGV("Track constructor name %d, calling pid %d", mName,
+ IPCThreadState::self()->getCallingPid());
+}
+
+AudioFlinger::PlaybackThread::Track::~Track()
+{
+ ALOGV("PlaybackThread::Track destructor");
+}
+
+void AudioFlinger::PlaybackThread::Track::destroy()
+{
+ // NOTE: destroyTrack_l() can remove a strong reference to this Track
+ // by removing it from mTracks vector, so there is a risk that this Tracks's
+ // destructor is called. As the destructor needs to lock mLock,
+ // we must acquire a strong reference on this Track before locking mLock
+ // here so that the destructor is called only when exiting this function.
+ // On the other hand, as long as Track::destroy() is only called by
+ // TrackHandle destructor, the TrackHandle still holds a strong ref on
+ // this Track with its member mTrack.
+ sp<Track> keep(this);
+ { // scope for mLock
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ if (!isOutputTrack()) {
+ if (mState == ACTIVE || mState == RESUMING) {
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+
+#ifdef ADD_BATTERY_DATA
+ // to track the speaker usage
+ addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+ }
+ AudioSystem::releaseOutput(thread->id());
+ }
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->destroyTrack_l(this);
+ }
+ }
+}
+
+/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
+{
+ result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate "
+ "L dB R dB Server User Main buf Aux Buf Flags Underruns\n");
+}
+
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+{
+ uint32_t vlr = mCblk->getVolumeLR();
+ if (isFastTrack()) {
+ sprintf(buffer, " F %2d", mFastIndex);
+ } else {
+ sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
+ }
+ track_state state = mState;
+ char stateChar;
+ switch (state) {
+ case IDLE:
+ stateChar = 'I';
+ break;
+ case TERMINATED:
+ stateChar = 'T';
+ break;
+ case STOPPING_1:
+ stateChar = 's';
+ break;
+ case STOPPING_2:
+ stateChar = '5';
+ break;
+ case STOPPED:
+ stateChar = 'S';
+ break;
+ case RESUMING:
+ stateChar = 'R';
+ break;
+ case ACTIVE:
+ stateChar = 'A';
+ break;
+ case PAUSING:
+ stateChar = 'p';
+ break;
+ case PAUSED:
+ stateChar = 'P';
+ break;
+ case FLUSHED:
+ stateChar = 'F';
+ break;
+ default:
+ stateChar = '?';
+ break;
+ }
+ char nowInUnderrun;
+ switch (mObservedUnderruns.mBitFields.mMostRecent) {
+ case UNDERRUN_FULL:
+ nowInUnderrun = ' ';
+ break;
+ case UNDERRUN_PARTIAL:
+ nowInUnderrun = '<';
+ break;
+ case UNDERRUN_EMPTY:
+ nowInUnderrun = '*';
+ break;
+ default:
+ nowInUnderrun = '?';
+ break;
+ }
+ snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g "
+ "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
+ (mClient == 0) ? getpid_cached : mClient->pid(),
+ mStreamType,
+ mFormat,
+ mChannelMask,
+ mSessionId,
+ mStepCount,
+ mFrameCount,
+ stateChar,
+ mMute,
+ mFillingUpStatus,
+ mCblk->sampleRate,
+ 20.0 * log10((vlr & 0xFFFF) / 4096.0),
+ 20.0 * log10((vlr >> 16) / 4096.0),
+ mCblk->server,
+ mCblk->user,
+ (int)mMainBuffer,
+ (int)mAuxBuffer,
+ mCblk->flags,
+ mUnderrunCount,
+ nowInUnderrun);
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ audio_track_cblk_t* cblk = this->cblk();
+ uint32_t framesReady;
+ uint32_t framesReq = buffer->frameCount;
+
+ // Check if last stepServer failed, try to step now
+ if (mStepServerFailed) {
+ // FIXME When called by fast mixer, this takes a mutex with tryLock().
+ // Since the fast mixer is higher priority than client callback thread,
+ // it does not result in priority inversion for client.
+ // But a non-blocking solution would be preferable to avoid
+ // fast mixer being unable to tryLock(), and
+ // to avoid the extra context switches if the client wakes up,
+ // discovers the mutex is locked, then has to wait for fast mixer to unlock.
+ if (!step()) goto getNextBuffer_exit;
+ ALOGV("stepServer recovered");
+ mStepServerFailed = false;
+ }
+
+ // FIXME Same as above
+ framesReady = cblk->framesReadyOut();
+
+ if (CC_LIKELY(framesReady)) {
+ uint32_t s = cblk->server;
+ uint32_t bufferEnd = cblk->serverBase + mFrameCount;
+
+ bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
+ if (framesReq > framesReady) {
+ framesReq = framesReady;
+ }
+ if (framesReq > bufferEnd - s) {
+ framesReq = bufferEnd - s;
+ }
+
+ buffer->raw = getBuffer(s, framesReq);
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+ return NOT_ENOUGH_DATA;
+}
+
+// Note that framesReady() takes a mutex on the control block using tryLock().
+// This could result in priority inversion if framesReady() is called by the normal mixer,
+// as the normal mixer thread runs at lower
+// priority than the client's callback thread: there is a short window within framesReady()
+// during which the normal mixer could be preempted, and the client callback would block.
+// Another problem can occur if framesReady() is called by the fast mixer:
+// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
+// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
+size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
+ return mCblk->framesReadyOut();
+}
+
+// Don't call for fast tracks; the framesReady() could result in priority inversion
+bool AudioFlinger::PlaybackThread::Track::isReady() const {
+ if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+ return true;
+ }
+
+ if (framesReady() >= mFrameCount ||
+ (mCblk->flags & CBLK_FORCEREADY)) {
+ mFillingUpStatus = FS_FILLED;
+ android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
+ int triggerSession)
+{
+ status_t status = NO_ERROR;
+ ALOGV("start(%d), calling pid %d session %d",
+ mName, IPCThreadState::self()->getCallingPid(), mSessionId);
+
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ track_state state = mState;
+ // here the track could be either new, or restarted
+ // in both cases "unstop" the track
+ if (mState == PAUSED) {
+ mState = TrackBase::RESUMING;
+ ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
+ } else {
+ mState = TrackBase::ACTIVE;
+ ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
+ }
+
+ if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
+ thread->mLock.unlock();
+ status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
+ thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+ // to track the speaker usage
+ if (status == NO_ERROR) {
+ addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
+ }
+#endif
+ }
+ if (status == NO_ERROR) {
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->addTrack_l(this);
+ } else {
+ mState = state;
+ triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+ }
+ } else {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::stop()
+{
+ ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ track_state state = mState;
+ if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
+ // If the track is not active (PAUSED and buffers full), flush buffers
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ mState = STOPPED;
+ } else if (!isFastTrack()) {
+ mState = STOPPED;
+ } else {
+ // prepareTracks_l() will set state to STOPPING_2 after next underrun,
+ // and then to STOPPED and reset() when presentation is complete
+ mState = STOPPING_1;
+ }
+ ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
+ playbackThread);
+ }
+ if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
+ thread->mLock.unlock();
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+ thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+ // to track the speaker usage
+ addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::pause()
+{
+ ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState == ACTIVE || mState == RESUMING) {
+ mState = PAUSING;
+ ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
+ if (!isOutputTrack()) {
+ thread->mLock.unlock();
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+ thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+ // to track the speaker usage
+ addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+ }
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::flush()
+{
+ ALOGV("flush(%d)", mName);
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
+ mState != PAUSING && mState != IDLE && mState != FLUSHED) {
+ return;
+ }
+ // No point remaining in PAUSED state after a flush => go to
+ // FLUSHED state
+ mState = FLUSHED;
+ // do not reset the track if it is still in the process of being stopped or paused.
+ // this will be done by prepareTracks_l() when the track is stopped.
+ // prepareTracks_l() will see mState == FLUSHED, then
+ // remove from active track list, reset(), and trigger presentation complete
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::reset()
+{
+ // Do not reset twice to avoid discarding data written just after a flush and before
+ // the audioflinger thread detects the track is stopped.
+ if (!mResetDone) {
+ TrackBase::reset();
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer
+ android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
+ android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
+ mFillingUpStatus = FS_FILLING;
+ mResetDone = true;
+ if (mState == FLUSHED) {
+ mState = IDLE;
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
+{
+ mMute = muted;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
+{
+ status_t status = DEAD_OBJECT;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ sp<AudioFlinger> af = mClient->audioFlinger();
+
+ Mutex::Autolock _l(af->mLock);
+
+ sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+
+ if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
+ Mutex::Autolock _dl(playbackThread->mLock);
+ Mutex::Autolock _sl(srcThread->mLock);
+ sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+ if (chain == 0) {
+ return INVALID_OPERATION;
+ }
+
+ sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
+ if (effect == 0) {
+ return INVALID_OPERATION;
+ }
+ srcThread->removeEffect_l(effect);
+ playbackThread->addEffect_l(effect);
+ // removeEffect_l() has stopped the effect if it was active so it must be restarted
+ if (effect->state() == EffectModule::ACTIVE ||
+ effect->state() == EffectModule::STOPPING) {
+ effect->start();
+ }
+
+ sp<EffectChain> dstChain = effect->chain().promote();
+ if (dstChain == 0) {
+ srcThread->addEffect_l(effect);
+ return INVALID_OPERATION;
+ }
+ AudioSystem::unregisterEffect(effect->id());
+ AudioSystem::registerEffect(&effect->desc(),
+ srcThread->id(),
+ dstChain->strategy(),
+ AUDIO_SESSION_OUTPUT_MIX,
+ effect->id());
+ }
+ status = playbackThread->attachAuxEffect(this, EffectId);
+ }
+ return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
+{
+ mAuxEffectId = EffectId;
+ mAuxBuffer = buffer;
+}
+
+bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
+ size_t audioHalFrames)
+{
+ // a track is considered presented when the total number of frames written to audio HAL
+ // corresponds to the number of frames written when presentationComplete() is called for the
+ // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
+ if (mPresentationCompleteFrames == 0) {
+ mPresentationCompleteFrames = framesWritten + audioHalFrames;
+ ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
+ mPresentationCompleteFrames, audioHalFrames);
+ }
+ if (framesWritten >= mPresentationCompleteFrames) {
+ ALOGV("presentationComplete() session %d complete: framesWritten %d",
+ mSessionId, framesWritten);
+ triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+ return true;
+ }
+ return false;
+}
+
+void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
+{
+ for (int i = 0; i < (int)mSyncEvents.size(); i++) {
+ if (mSyncEvents[i]->type() == type) {
+ mSyncEvents[i]->trigger();
+ mSyncEvents.removeAt(i);
+ i--;
+ }
+ }
+}
+
+// implement VolumeBufferProvider interface
+
+uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+{
+ // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
+ ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
+ uint32_t vlr = mCblk->getVolumeLR();
+ uint32_t vl = vlr & 0xFFFF;
+ uint32_t vr = vlr >> 16;
+ // track volumes come from shared memory, so can't be trusted and must be clamped
+ if (vl > MAX_GAIN_INT) {
+ vl = MAX_GAIN_INT;
+ }
+ if (vr > MAX_GAIN_INT) {
+ vr = MAX_GAIN_INT;
+ }
+ // now apply the cached master volume and stream type volume;
+ // this is trusted but lacks any synchronization or barrier so may be stale
+ float v = mCachedVolume;
+ vl *= v;
+ vr *= v;
+ // re-combine into U4.16
+ vlr = (vr << 16) | (vl & 0xFFFF);
+ // FIXME look at mute, pause, and stop flags
+ return vlr;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
+{
+ if (mState == TERMINATED || mState == PAUSED ||
+ ((framesReady() == 0) && ((mSharedBuffer != 0) ||
+ (mState == STOPPED)))) {
+ ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
+ mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
+ event->cancel();
+ return INVALID_OPERATION;
+ }
+ (void) TrackBase::setSyncEvent(event);
+ return NO_ERROR;
+}
+
+bool AudioFlinger::PlaybackThread::Track::isOut() const
+{
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId) {
+ if (!client->reserveTimedTrack())
+ return 0;
+
+ return new TimedTrack(
+ thread, client, streamType, sampleRate, format, channelMask, frameCount,
+ sharedBuffer, sessionId);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : Track(thread, client, streamType, sampleRate, format, channelMask,
+ frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
+ mQueueHeadInFlight(false),
+ mTrimQueueHeadOnRelease(false),
+ mFramesPendingInQueue(0),
+ mTimedSilenceBuffer(NULL),
+ mTimedSilenceBufferSize(0),
+ mTimedAudioOutputOnTime(false),
+ mMediaTimeTransformValid(false)
+{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
+ mLocalTimeToSampleTransform.a_zero = 0;
+ mLocalTimeToSampleTransform.b_zero = 0;
+ mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+ mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+ LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+ &mLocalTimeToSampleTransform.a_to_b_denom);
+
+ mMediaTimeToSampleTransform.a_zero = 0;
+ mMediaTimeToSampleTransform.b_zero = 0;
+ mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
+ mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
+ LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
+ &mMediaTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+ mClient->releaseTimedTrack();
+ delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+ size_t size, sp<IMemory>* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ trimTimedBufferQueue_l();
+
+ // lazily initialize the shared memory heap for timed buffers
+ if (mTimedMemoryDealer == NULL) {
+ const int kTimedBufferHeapSize = 512 << 10;
+
+ mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+ "AudioFlingerTimed");
+ if (mTimedMemoryDealer == NULL)
+ return NO_MEMORY;
+ }
+
+ sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL) {
+ newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL)
+ return NO_MEMORY;
+ }
+
+ *buffer = newBuffer;
+ return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+ int64_t mediaTimeNow;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return;
+
+ int64_t targetTimeNow;
+ status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+ ? mCCHelper.getCommonTime(&targetTimeNow)
+ : mCCHelper.getLocalTime(&targetTimeNow);
+
+ if (OK != res)
+ return;
+
+ if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+ &mediaTimeNow)) {
+ return;
+ }
+ }
+
+ size_t trimEnd;
+ for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
+ int64_t bufEnd;
+
+ if ((trimEnd + 1) < mTimedBufferQueue.size()) {
+ // We have a next buffer. Just use its PTS as the PTS of the frame
+ // following the last frame in this buffer. If the stream is sparse
+ // (ie, there are deliberate gaps left in the stream which should be
+ // filled with silence by the TimedAudioTrack), then this can result
+ // in one extra buffer being left un-trimmed when it could have
+ // been. In general, this is not typical, and we would rather
+ // optimized away the TS calculation below for the more common case
+ // where PTSes are contiguous.
+ bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
+ } else {
+ // We have no next buffer. Compute the PTS of the frame following
+ // the last frame in this buffer by computing the duration of of
+ // this frame in media time units and adding it to the PTS of the
+ // buffer.
+ int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
+ / mFrameSize;
+
+ if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
+ &bufEnd)) {
+ ALOGE("Failed to convert frame count of %lld to media time"
+ " duration" " (scale factor %d/%u) in %s",
+ frameCount,
+ mMediaTimeToSampleTransform.a_to_b_numer,
+ mMediaTimeToSampleTransform.a_to_b_denom,
+ __PRETTY_FUNCTION__);
+ break;
+ }
+ bufEnd += mTimedBufferQueue[trimEnd].pts();
+ }
+
+ if (bufEnd > mediaTimeNow)
+ break;
+
+ // Is the buffer we want to use in the middle of a mix operation right
+ // now? If so, don't actually trim it. Just wait for the releaseBuffer
+ // from the mixer which should be coming back shortly.
+ if (!trimEnd && mQueueHeadInFlight) {
+ mTrimQueueHeadOnRelease = true;
+ }
+ }
+
+ size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
+ if (trimStart < trimEnd) {
+ // Update the bookkeeping for framesReady()
+ for (size_t i = trimStart; i < trimEnd; ++i) {
+ updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
+ }
+
+ // Now actually remove the buffers from the queue.
+ mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
+ }
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
+ const char* logTag) {
+ ALOG_ASSERT(mTimedBufferQueue.size() > 0,
+ "%s called (reason \"%s\"), but timed buffer queue has no"
+ " elements to trim.", __FUNCTION__, logTag);
+
+ updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
+ mTimedBufferQueue.removeAt(0);
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
+ const TimedBuffer& buf,
+ const char* logTag) {
+ uint32_t bufBytes = buf.buffer()->size();
+ uint32_t consumedAlready = buf.position();
+
+ ALOG_ASSERT(consumedAlready <= bufBytes,
+ "Bad bookkeeping while updating frames pending. Timed buffer is"
+ " only %u bytes long, but claims to have consumed %u"
+ " bytes. (update reason: \"%s\")",
+ bufBytes, consumedAlready, logTag);
+
+ uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
+ ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
+ "Bad bookkeeping while updating frames pending. Should have at"
+ " least %u queued frames, but we think we have only %u. (update"
+ " reason: \"%s\")",
+ bufFrames, mFramesPendingInQueue, logTag);
+
+ mFramesPendingInQueue -= bufFrames;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts) {
+
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ uint32_t bufFrames = buffer->size() / mFrameSize;
+ mFramesPendingInQueue += bufFrames;
+ mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+ const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+ ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
+ xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+ target);
+
+ if (!(target == TimedAudioTrack::LOCAL_TIME ||
+ target == TimedAudioTrack::COMMON_TIME)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMediaTimeTransformLock);
+ mMediaTimeTransform = xform;
+ mMediaTimeTransformTarget = target;
+ mMediaTimeTransformValid = true;
+
+ return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ if (pts == AudioBufferProvider::kInvalidPTS) {
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ mTimedAudioOutputOnTime = false;
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ ALOG_ASSERT(!mQueueHeadInFlight,
+ "getNextBuffer called without releaseBuffer!");
+
+ while (true) {
+
+ // if we have no timed buffers, then fail
+ if (mTimedBufferQueue.isEmpty()) {
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ // calculate the PTS of the head of the timed buffer queue expressed in
+ // local time
+ int64_t headLocalPTS;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+ ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
+
+ if (mMediaTimeTransform.a_to_b_denom == 0) {
+ // the transform represents a pause, so yield silence
+ timedYieldSilence_l(buffer->frameCount, buffer);
+ return NO_ERROR;
+ }
+
+ int64_t transformedPTS;
+ if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+ &transformedPTS)) {
+ // the transform failed. this shouldn't happen, but if it does
+ // then just drop this buffer
+ ALOGW("timedGetNextBuffer transform failed");
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ trimTimedBufferQueueHead_l("getNextBuffer; no transform");
+ return NO_ERROR;
+ }
+
+ if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+ if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
+ &headLocalPTS)) {
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+ } else {
+ headLocalPTS = transformedPTS;
+ }
+ }
+
+ // adjust the head buffer's PTS to reflect the portion of the head buffer
+ // that has already been consumed
+ int64_t effectivePTS = headLocalPTS +
+ ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
+
+ // Calculate the delta in samples between the head of the input buffer
+ // queue and the start of the next output buffer that will be written.
+ // If the transformation fails because of over or underflow, it means
+ // that the sample's position in the output stream is so far out of
+ // whack that it should just be dropped.
+ int64_t sampleDelta;
+ if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+ ALOGV("*** head buffer is too far from PTS: dropped buffer");
+ trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
+ " mix");
+ continue;
+ }
+ if (!mLocalTimeToSampleTransform.doForwardTransform(
+ (effectivePTS - pts) << 32, &sampleDelta)) {
+ ALOGV("*** too late during sample rate transform: dropped buffer");
+ trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
+ continue;
+ }
+
+ ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
+ " sampleDelta=[%d.%08x]",
+ head.pts(), head.position(), pts,
+ static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
+ + (sampleDelta >> 32)),
+ static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+ // if the delta between the ideal placement for the next input sample and
+ // the current output position is within this threshold, then we will
+ // concatenate the next input samples to the previous output
+ const int64_t kSampleContinuityThreshold =
+ (static_cast<int64_t>(sampleRate()) << 32) / 250;
+
+ // if this is the first buffer of audio that we're emitting from this track
+ // then it should be almost exactly on time.
+ const int64_t kSampleStartupThreshold = 1LL << 32;
+
+ if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+ (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+ // the next input is close enough to being on time, so concatenate it
+ // with the last output
+ timedYieldSamples_l(buffer);
+
+ ALOGVV("*** on time: head.pos=%d frameCount=%u",
+ head.position(), buffer->frameCount);
+ return NO_ERROR;
+ }
+
+ // Looks like our output is not on time. Reset our on timed status.
+ // Next time we mix samples from our input queue, then should be within
+ // the StartupThreshold.
+ mTimedAudioOutputOnTime = false;
+ if (sampleDelta > 0) {
+ // the gap between the current output position and the proper start of
+ // the next input sample is too big, so fill it with silence
+ uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+ timedYieldSilence_l(framesUntilNextInput, buffer);
+ ALOGV("*** silence: frameCount=%u", buffer->frameCount);
+ return NO_ERROR;
+ } else {
+ // the next input sample is late
+ uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+ size_t onTimeSamplePosition =
+ head.position() + lateFrames * mFrameSize;
+
+ if (onTimeSamplePosition > head.buffer()->size()) {
+ // all the remaining samples in the head are too late, so
+ // drop it and move on
+ ALOGV("*** too late: dropped buffer");
+ trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
+ continue;
+ } else {
+ // skip over the late samples
+ head.setPosition(onTimeSamplePosition);
+
+ // yield the available samples
+ timedYieldSamples_l(buffer);
+
+ ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ }
+ }
+ }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
+ AudioBufferProvider::Buffer* buffer) {
+
+ const TimedBuffer& head = mTimedBufferQueue[0];
+
+ buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+ head.position());
+
+ uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+ mFrameSize);
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+ mQueueHeadInFlight = true;
+ mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
+ uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+ // lazily allocate a buffer filled with silence
+ if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
+ delete [] mTimedSilenceBuffer;
+ mTimedSilenceBufferSize = numFrames * mFrameSize;
+ mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+ memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+ }
+
+ buffer->raw = mTimedSilenceBuffer;
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(numFrames, framesRequested);
+
+ mTimedAudioOutputOnTime = false;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+ AudioBufferProvider::Buffer* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ // If the buffer which was just released is part of the buffer at the head
+ // of the queue, be sure to update the amt of the buffer which has been
+ // consumed. If the buffer being returned is not part of the head of the
+ // queue, its either because the buffer is part of the silence buffer, or
+ // because the head of the timed queue was trimmed after the mixer called
+ // getNextBuffer but before the mixer called releaseBuffer.
+ if (buffer->raw == mTimedSilenceBuffer) {
+ ALOG_ASSERT(!mQueueHeadInFlight,
+ "Queue head in flight during release of silence buffer!");
+ goto done;
+ }
+
+ ALOG_ASSERT(mQueueHeadInFlight,
+ "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
+ " head in flight.");
+
+ if (mTimedBufferQueue.size()) {
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ void* start = head.buffer()->pointer();
+ void* end = reinterpret_cast<void*>(
+ reinterpret_cast<uint8_t*>(head.buffer()->pointer())
+ + head.buffer()->size());
+
+ ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
+ "released buffer not within the head of the timed buffer"
+ " queue; qHead = [%p, %p], released buffer = %p",
+ start, end, buffer->raw);
+
+ head.setPosition(head.position() +
+ (buffer->frameCount * mFrameSize));
+ mQueueHeadInFlight = false;
+
+ ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
+ "Bad bookkeeping during releaseBuffer! Should have at"
+ " least %u queued frames, but we think we have only %u",
+ buffer->frameCount, mFramesPendingInQueue);
+
+ mFramesPendingInQueue -= buffer->frameCount;
+
+ if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
+ || mTrimQueueHeadOnRelease) {
+ trimTimedBufferQueueHead_l("releaseBuffer");
+ mTrimQueueHeadOnRelease = false;
+ }
+ } else {
+ LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
+ " buffers in the timed buffer queue");
+ }
+
+done:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+}
+
+size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+ return mFramesPendingInQueue;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+ : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts)
+ : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+ PlaybackThread *playbackThread,
+ DuplicatingThread *sourceThread,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount)
+ : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
+ NULL, 0, IAudioFlinger::TRACK_DEFAULT),
+ mActive(false), mSourceThread(sourceThread), mBuffers(NULL)
+{
+
+ if (mCblk != NULL) {
+ mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ mOutBuffer.frameCount = 0;
+ playbackThread->mTracks.add(this);
+ ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \
+ "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p",
+ mCblk, mBuffer, mBuffers,
+ mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
+ } else {
+ ALOGW("Error creating output track on thread %p", playbackThread);
+ }
+}
+
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
+{
+ clearBufferQueue();
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
+ int triggerSession)
+{
+ status_t status = Track::start(event, triggerSession);
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ mActive = true;
+ mRetryCount = 127;
+ return status;
+}
+
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
+{
+ Track::stop();
+ clearBufferQueue();
+ mOutBuffer.frameCount = 0;
+ mActive = false;
+}
+
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+{
+ Buffer *pInBuffer;
+ Buffer inBuffer;
+ uint32_t channelCount = mChannelCount;
+ bool outputBufferFull = false;
+ inBuffer.frameCount = frames;
+ inBuffer.i16 = data;
+
+ uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
+
+ if (!mActive && frames != 0) {
+ start();
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ MixerThread *mixerThread = (MixerThread *)thread.get();
+ if (mFrameCount > frames) {
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+ uint32_t startFrames = (mFrameCount - frames);
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+ pInBuffer->frameCount = startFrames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
+ }
+ }
+ }
+ }
+
+ while (waitTimeLeftMs) {
+ // First write pending buffers, then new data
+ if (mBufferQueue.size()) {
+ pInBuffer = mBufferQueue.itemAt(0);
+ } else {
+ pInBuffer = &inBuffer;
+ }
+
+ if (pInBuffer->frameCount == 0) {
+ break;
+ }
+
+ if (mOutBuffer.frameCount == 0) {
+ mOutBuffer.frameCount = pInBuffer->frameCount;
+ nsecs_t startTime = systemTime();
+ if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
+ ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
+ mThread.unsafe_get());
+ outputBufferFull = true;
+ break;
+ }
+ uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+ if (waitTimeLeftMs >= waitTimeMs) {
+ waitTimeLeftMs -= waitTimeMs;
+ } else {
+ waitTimeLeftMs = 0;
+ }
+ }
+
+ uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
+ pInBuffer->frameCount;
+ memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+ mCblk->stepUserOut(outFrames, mFrameCount);
+ pInBuffer->frameCount -= outFrames;
+ pInBuffer->i16 += outFrames * channelCount;
+ mOutBuffer.frameCount -= outFrames;
+ mOutBuffer.i16 += outFrames * channelCount;
+
+ if (pInBuffer->frameCount == 0) {
+ if (mBufferQueue.size()) {
+ mBufferQueue.removeAt(0);
+ delete [] pInBuffer->mBuffer;
+ delete pInBuffer;
+ ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
+ mThread.unsafe_get(), mBufferQueue.size());
+ } else {
+ break;
+ }
+ }
+ }
+
+ // If we could not write all frames, allocate a buffer and queue it for next time.
+ if (inBuffer.frameCount) {
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0 && !thread->standby()) {
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+ pInBuffer->frameCount = inBuffer.frameCount;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
+ sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
+ mThread.unsafe_get(), mBufferQueue.size());
+ } else {
+ ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
+ mThread.unsafe_get(), this);
+ }
+ }
+ }
+
+ // Calling write() with a 0 length buffer, means that no more data will be written:
+ // If no more buffers are pending, fill output track buffer to make sure it is started
+ // by output mixer.
+ if (frames == 0 && mBufferQueue.size() == 0) {
+ if (mCblk->user < mFrameCount) {
+ frames = mFrameCount - mCblk->user;
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[frames * channelCount];
+ pInBuffer->frameCount = frames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else if (mActive) {
+ stop();
+ }
+ }
+
+ return outputBufferFull;
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
+ AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
+{
+ int active;
+ status_t result;
+ audio_track_cblk_t* cblk = mCblk;
+ uint32_t framesReq = buffer->frameCount;
+
+ ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+ buffer->frameCount = 0;
+
+ uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
+
+
+ if (framesAvail == 0) {
+ Mutex::Autolock _l(cblk->lock);
+ goto start_loop_here;
+ while (framesAvail == 0) {
+ active = mActive;
+ if (CC_UNLIKELY(!active)) {
+ ALOGV("Not active and NO_MORE_BUFFERS");
+ return NO_MORE_BUFFERS;
+ }
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ if (result != NO_ERROR) {
+ return NO_MORE_BUFFERS;
+ }
+ // read the server count again
+ start_loop_here:
+ framesAvail = cblk->framesAvailableOut_l(mFrameCount);
+ }
+ }
+
+// if (framesAvail < framesReq) {
+// return NO_MORE_BUFFERS;
+// }
+
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+
+ uint32_t u = cblk->user;
+ uint32_t bufferEnd = cblk->userBase + mFrameCount;
+
+ if (framesReq > bufferEnd - u) {
+ framesReq = bufferEnd - u;
+ }
+
+ buffer->frameCount = framesReq;
+ buffer->raw = cblk->buffer(mBuffers, mFrameSize, u);
+ return NO_ERROR;
+}
+
+
+void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
+{
+ size_t size = mBufferQueue.size();
+
+ for (size_t i = 0; i < size; i++) {
+ Buffer *pBuffer = mBufferQueue.itemAt(i);
+ delete [] pBuffer->mBuffer;
+ delete pBuffer;
+ }
+ mBufferQueue.clear();
+}
+
+
+// ----------------------------------------------------------------------------
+// Record
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(
+ const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+ : BnAudioRecord(),
+ mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {
+ stop_nonvirtual();
+ mRecordTrack->destroy();
+}
+
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+ return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
+ int triggerSession) {
+ ALOGV("RecordHandle::start()");
+ return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
+}
+
+void AudioFlinger::RecordHandle::stop() {
+ stop_nonvirtual();
+}
+
+void AudioFlinger::RecordHandle::stop_nonvirtual() {
+ ALOGV("RecordHandle::stop()");
+ mRecordTrack->stop();
+}
+
+status_t AudioFlinger::RecordHandle::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+// RecordTrack constructor must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+ RecordThread *thread,
+ const sp<Client>& client,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ int sessionId)
+ : TrackBase(thread, client, sampleRate, format,
+ channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
+ mOverflow(false)
+{
+ ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+}
+
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
+{
+ ALOGV("%s", __func__);
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts)
+{
+ audio_track_cblk_t* cblk = this->cblk();
+ uint32_t framesAvail;
+ uint32_t framesReq = buffer->frameCount;
+
+ // Check if last stepServer failed, try to step now
+ if (mStepServerFailed) {
+ if (!step()) {
+ goto getNextBuffer_exit;
+ }
+ ALOGV("stepServer recovered");
+ mStepServerFailed = false;
+ }
+
+ // FIXME lock is not actually held, so overrun is possible
+ framesAvail = cblk->framesAvailableIn_l(mFrameCount);
+
+ if (CC_LIKELY(framesAvail)) {
+ uint32_t s = cblk->server;
+ uint32_t bufferEnd = cblk->serverBase + mFrameCount;
+
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+ if (framesReq > bufferEnd - s) {
+ framesReq = bufferEnd - s;
+ }
+
+ buffer->raw = getBuffer(s, framesReq);
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
+ int triggerSession)
+{
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ return recordThread->start(this, event, triggerSession);
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::stop()
+{
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ recordThread->mLock.lock();
+ bool doStop = recordThread->stop_l(this);
+ if (doStop) {
+ TrackBase::reset();
+ // Force overrun condition to avoid false overrun callback until first data is
+ // read from buffer
+ android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
+ }
+ recordThread->mLock.unlock();
+ if (doStop) {
+ AudioSystem::stopInput(recordThread->id());
+ }
+ }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::destroy()
+{
+ // see comments at AudioFlinger::PlaybackThread::Track::destroy()
+ sp<RecordTrack> keep(this);
+ {
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ if (mState == ACTIVE || mState == RESUMING) {
+ AudioSystem::stopInput(thread->id());
+ }
+ AudioSystem::releaseInput(thread->id());
+ Mutex::Autolock _l(thread->mLock);
+ RecordThread *recordThread = (RecordThread *) thread.get();
+ recordThread->destroyTrack_l(this);
+ }
+ }
+}
+
+
+/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
+{
+ result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n");
+}
+
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n",
+ (mClient == 0) ? getpid_cached : mClient->pid(),
+ mFormat,
+ mChannelMask,
+ mSessionId,
+ mStepCount,
+ mState,
+ mCblk->sampleRate,
+ mCblk->server,
+ mCblk->user,
+ mFrameCount);
+}
+
+bool AudioFlinger::RecordThread::RecordTrack::isOut() const
+{
+ return false;
+}
+
+}; // namespace android