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** 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
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** See the License for the specific language governing permissions and
** limitations under the License.
#error This header file should only be included from AudioFlinger.h
// playback track
class Track : public TrackBase, public VolumeProvider {
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,
void *buffer,
size_t bufferSize,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
uid_t uid,
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
virtual ~Track();
virtual status_t initCheck() const;
static void appendDumpHeader(String8& result);
void appendDump(String8& result, bool active);
virtual status_t start(AudioSystem::sync_event_t event =
audio_session_t triggerSession = AUDIO_SESSION_NONE);
virtual void stop();
void pause();
void flush();
void destroy();
int name() const { return mName; }
virtual uint32_t sampleRate() const;
audio_stream_type_t streamType() const {
return mStreamType;
bool isOffloaded() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
bool isDirect() const { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
bool isOffloadedOrDirect() const { return (mFlags
status_t setParameters(const String8& keyValuePairs);
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
int32_t *auxBuffer() const { return mAuxBuffer; }
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
int16_t *mainBuffer() const { return mMainBuffer; }
int auxEffectId() const { return mAuxEffectId; }
virtual status_t getTimestamp(AudioTimestamp& timestamp);
void signal();
// implement FastMixerState::VolumeProvider interface
virtual gain_minifloat_packed_t getVolumeLR();
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isFastTrack() const { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
// implement volume handling.
VolumeShaper::Status applyVolumeShaper(
const sp<VolumeShaper::Configuration>& configuration,
const sp<VolumeShaper::Operation>& operation);
sp<VolumeShaper::State> getVolumeShaperState(int id);
sp<VolumeHandler> getVolumeHandler() { return mVolumeHandler; }
// for numerous
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
friend class OffloadThread;
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
// releaseBuffer() not overridden
// ExtendedAudioBufferProvider interface
virtual size_t framesReady() const;
virtual int64_t framesReleased() const;
virtual void onTimestamp(const ExtendedTimestamp &timestamp);
bool isPausing() const { return mState == PAUSING; }
bool isPaused() const { return mState == PAUSED; }
bool isResuming() const { return mState == RESUMING; }
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
bool isFlushPending() const { return mFlushHwPending; }
void flushAck();
bool isResumePending();
void resumeAck();
void updateTrackFrameInfo(int64_t trackFramesReleased, int64_t sinkFramesWritten,
const ExtendedTimestamp &timeStamp);
sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
// framesWritten is cumulative, never reset, and is shared all tracks
// audioHalFrames is derived from output latency
// FIXME parameters not needed, could get them from the thread
bool presentationComplete(int64_t framesWritten, size_t audioHalFrames);
void signalClientFlag(int32_t flag);
void triggerEvents(AudioSystem::sync_event_t type);
virtual void invalidate();
void disable();
int fastIndex() const { return mFastIndex; }
// FILLED state is used for suppressing volume ramp at begin of playing
mutable uint8_t mFillingUpStatus;
int8_t mRetryCount;
// see comment at AudioFlinger::PlaybackThread::Track::~Track for why this can't be const
sp<IMemory> mSharedBuffer;
bool mResetDone;
const audio_stream_type_t mStreamType;
int mName; // track name on the normal mixer,
// allocated statically at track creation time,
// and is even allocated (though unused) for fast tracks
// FIXME don't allocate track name for fast tracks
int16_t *mMainBuffer;
int32_t *mAuxBuffer;
int mAuxEffectId;
bool mHasVolumeController;
size_t mPresentationCompleteFrames; // number of frames written to the
// audio HAL when this track will be fully rendered
// zero means not monitoring
// access these three variables only when holding thread lock.
LinearMap<int64_t> mFrameMap; // track frame to server frame mapping
ExtendedTimestamp mSinkTimestamp;
sp<VolumeHandler> mVolumeHandler; // handles multiple VolumeShaper configs and operations
// The following fields are only for fast tracks, and should be in a subclass
int mFastIndex; // index within FastMixerState::mFastTracks[];
// either mFastIndex == -1 if not isFastTrack()
// or 0 < mFastIndex < FastMixerState::kMaxFast because
// index 0 is reserved for normal mixer's submix;
// index is allocated statically at track creation time
// but the slot is only used if track is active
FastTrackUnderruns mObservedUnderruns; // Most recently observed value of
// mFastMixerDumpState.mTracks[mFastIndex].mUnderruns
volatile float mCachedVolume; // combined master volume and stream type volume;
// 'volatile' means accessed without lock or
// barrier, but is read/written atomically
sp<AudioTrackServerProxy> mAudioTrackServerProxy;
bool mResumeToStopping; // track was paused in stopping state.
bool mFlushHwPending; // track requests for thread flush
audio_output_flags_t mFlags;
}; // end of Track
// playback track, used by DuplicatingThread
class OutputTrack : public Track {
class Buffer : public AudioBufferProvider::Buffer {
void *mBuffer;
OutputTrack(PlaybackThread *thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
uid_t uid);
virtual ~OutputTrack();
virtual status_t start(AudioSystem::sync_event_t event =
audio_session_t triggerSession = AUDIO_SESSION_NONE);
virtual void stop();
bool write(void* data, uint32_t frames);
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
bool isActive() const { return mActive; }
const wp<ThreadBase>& thread() const { return mThread; }
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer,
uint32_t waitTimeMs);
void clearBufferQueue();
void restartIfDisabled();
// Maximum number of pending buffers allocated by OutputTrack::write()
static const uint8_t kMaxOverFlowBuffers = 10;
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
sp<AudioTrackClientProxy> mClientProxy;
}; // end of OutputTrack
// playback track, used by PatchPanel
class PatchTrack : public Track, public PatchProxyBufferProvider {
PatchTrack(PlaybackThread *playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_output_flags_t flags);
virtual ~PatchTrack();
virtual status_t start(AudioSystem::sync_event_t event =
audio_session_t triggerSession = AUDIO_SESSION_NONE);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// PatchProxyBufferProvider interface
virtual status_t obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *timeOut = NULL);
virtual void releaseBuffer(Proxy::Buffer* buffer);
void setPeerProxy(PatchProxyBufferProvider *proxy) { mPeerProxy = proxy; }
void restartIfDisabled();
sp<ClientProxy> mProxy;
PatchProxyBufferProvider* mPeerProxy;
struct timespec mPeerTimeout;
}; // end of PatchTrack