/*
**
** 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.
*/
#ifndef INCLUDING_FROM_AUDIOFLINGER_H
#error This header file should only be included from AudioFlinger.h
#endif
class ThreadBase : public Thread {
public:
#include "TrackBase.h"
enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD, // Thread class is RecordThread
OFFLOAD // Thread class is OffloadThread
};
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
void dumpBase(int fd, const Vector<String16>& args);
void dumpEffectChains(int fd, const Vector<String16>& args);
void clearPowerManager();
// base for record and playback
enum {
CFG_EVENT_IO,
CFG_EVENT_PRIO
};
class ConfigEvent {
public:
ConfigEvent(int type) : mType(type) {}
virtual ~ConfigEvent() {}
int type() const { return mType; }
virtual void dump(char *buffer, size_t size) = 0;
private:
const int mType;
};
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(int event, int param) :
ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
virtual ~IoConfigEvent() {}
int event() const { return mEvent; }
int param() const { return mParam; }
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam);
}
private:
const int mEvent;
const int mParam;
};
class PrioConfigEvent : public ConfigEvent {
public:
PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) :
ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {}
virtual ~PrioConfigEvent() {}
pid_t pid() const { return mPid; }
pid_t tid() const { return mTid; }
int32_t prio() const { return mPrio; }
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio);
}
private:
const pid_t mPid;
const pid_t mTid;
const int32_t mPrio;
};
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
virtual ~PMDeathRecipient() {}
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
private:
PMDeathRecipient(const PMDeathRecipient&);
PMDeathRecipient& operator = (const PMDeathRecipient&);
wp<ThreadBase> mThread;
};
virtual status_t initCheck() const = 0;
// static externally-visible
type_t type() const { return mType; }
audio_io_handle_t id() const { return mId;}
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
uint32_t channelCount() const { return mChannelCount; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mFormat; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
size_t frameSize() const { return mFrameSize; }
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys) = 0;
virtual void audioConfigChanged_l(int event, int param = 0) = 0;
void sendIoConfigEvent(int event, int param = 0);
void sendIoConfigEvent_l(int event, int param = 0);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
void processConfigEvents();
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
audio_devices_t outDevice() const { return mOutDevice; }
audio_devices_t inDevice() const { return mInDevice; }
virtual audio_stream_t* stream() const = 0;
sp<EffectHandle> createEffect_l(
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
int sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
EffectHandle *handle,
bool unpinIfLast);
// return values for hasAudioSession (bit field)
enum effect_state {
EFFECT_SESSION = 0x1, // the audio session corresponds to at least one
// effect
TRACK_SESSION = 0x2 // the audio session corresponds to at least one
// track
};
// get effect chain corresponding to session Id.
sp<EffectChain> getEffectChain(int sessionId);
// same as getEffectChain() but must be called with ThreadBase mutex locked
sp<EffectChain> getEffectChain_l(int sessionId) const;
// add an effect chain to the chain list (mEffectChains)
virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
// remove an effect chain from the chain list (mEffectChains)
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
// lock all effect chains Mutexes. Must be called before releasing the
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
// unlock effect chains after process
void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
// get a copy of mEffectChains vector
Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; };
// set audio mode to all effect chains
void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId);
sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
// add and effect module. Also creates the effect chain is none exists for
// the effects audio session
status_t addEffect_l(const sp< EffectModule>& effect);
// remove and effect module. Also removes the effect chain is this was the last
// effect
void removeEffect_l(const sp< EffectModule>& effect);
// detach all tracks connected to an auxiliary effect
virtual void detachAuxEffect_l(int effectId) {}
// returns either EFFECT_SESSION if effects on this audio session exist in one
// chain, or TRACK_SESSION if tracks on this audio session exist, or both
virtual uint32_t hasAudioSession(int sessionId) const = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
// suspend or restore effect according to the type of effect passed. a NULL
// type pointer means suspend all effects in the session
void setEffectSuspended(const effect_uuid_t *type,
bool suspend,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
bool enabled,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0;
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
mutable Mutex mLock;
protected:
// entry describing an effect being suspended in mSuspendedSessions keyed vector
class SuspendedSessionDesc : public RefBase {
public:
SuspendedSessionDesc() : mRefCount(0) {}
int mRefCount; // number of active suspend requests
effect_uuid_t mType; // effect type UUID
};
void acquireWakeLock(int uid = -1);
void acquireWakeLock_l(int uid = -1);
void releaseWakeLock();
void releaseWakeLock_l();
void updateWakeLockUids(const SortedVector<int> &uids);
void updateWakeLockUids_l(const SortedVector<int> &uids);
void getPowerManager_l();
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
int sessionId);
// updated mSuspendedSessions when an effect suspended or restored
void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
int sessionId);
// check if some effects must be suspended when an effect chain is added
void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
String16 getWakeLockTag();
virtual void preExit() { }
friend class AudioFlinger; // for mEffectChains
const type_t mType;
// Used by parameters, config events, addTrack_l, exit
Condition mWaitWorkCV;
const sp<AudioFlinger> mAudioFlinger;
// updated by PlaybackThread::readOutputParameters() or
// RecordThread::readInputParameters()
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
// Parameter sequence by client: binder thread calling setParameters():
// 1. Lock mLock
// 2. Append to mNewParameters
// 3. mWaitWorkCV.signal
// 4. mParamCond.waitRelative with timeout
// 5. read mParamStatus
// 6. mWaitWorkCV.signal
// 7. Unlock
//
// Parameter sequence by server: threadLoop calling checkForNewParameters_l():
// 1. Lock mLock
// 2. If there is an entry in mNewParameters proceed ...
// 2. Read first entry in mNewParameters
// 3. Process
// 4. Remove first entry from mNewParameters
// 5. Set mParamStatus
// 6. mParamCond.signal
// 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus)
// 8. Unlock
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
// vector owns each ConfigEvent *, so must delete after removing
Vector<ConfigEvent *> mConfigEvents;
// These fields are written and read by thread itself without lock or barrier,
// and read by other threads without lock or barrier via standby() , outDevice()
// and inDevice().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
audio_source_t mAudioSource; // (see audio.h, audio_source_t)
const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
static const int kNameLength = 16; // prctl(PR_SET_NAME) limit
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > >
mSuspendedSessions;
static const size_t kLogSize = 4 * 1024;
sp<NBLog::Writer> mNBLogWriter;
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
#include "PlaybackTracks.h"
enum mixer_state {
MIXER_IDLE, // no active tracks
MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready
MIXER_TRACKS_READY, // at least one active track, and at least one track has data
MIXER_DRAIN_TRACK, // drain currently playing track
MIXER_DRAIN_ALL, // fully drain the hardware
// standby mode does not have an enum value
// suspend by audio policy manager is orthogonal to mixer state
};
// retry count before removing active track in case of underrun on offloaded thread:
// we need to make sure that AudioTrack client has enough time to send large buffers
//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled
// for offloaded tracks
static const int8_t kMaxTrackRetriesOffload = 20;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, type_t type);
virtual ~PlaybackThread();
void dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
virtual bool threadLoop();
// RefBase
virtual void onFirstRef();
protected:
// Code snippets that were lifted up out of threadLoop()
virtual void threadLoop_mix() = 0;
virtual void threadLoop_sleepTime() = 0;
virtual ssize_t threadLoop_write();
virtual void threadLoop_drain();
virtual void threadLoop_standby();
virtual void threadLoop_exit();
virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
// prepareTracks_l reads and writes mActiveTracks, and returns
// the pending set of tracks to remove via Vector 'tracksToRemove'. The caller
// is responsible for clearing or destroying this Vector later on, when it
// is safe to do so. That will drop the final ref count and destroy the tracks.
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
void removeTracks_l(const Vector< sp<Track> >& tracksToRemove);
void writeCallback();
void resetWriteBlocked(uint32_t sequence);
void drainCallback();
void resetDraining(uint32_t sequence);
static int asyncCallback(stream_callback_event_t event, void *param, void *cookie);
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
// ThreadBase virtuals
virtual void preExit();
public:
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const;
// same, but lock must already be held
uint32_t latency_l() const;
void setMasterVolume(float value);
void setMasterMute(bool muted);
void setStreamVolume(audio_stream_type_t stream, float value);
void setStreamMute(audio_stream_type_t stream, bool muted);
float streamVolume(audio_stream_type_t stream) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::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,
pid_t tid,
int uid,
status_t *status);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual audio_stream_t* stream() const;
// a very large number of suspend() will eventually wraparound, but unlikely
void suspend() { (void) android_atomic_inc(&mSuspended); }
void restore()
{
// if restore() is done without suspend(), get back into
// range so that the next suspend() will operate correctly
if (android_atomic_dec(&mSuspended) <= 0) {
android_atomic_release_store(0, &mSuspended);
}
}
bool isSuspended() const
{ return android_atomic_acquire_load(&mSuspended) > 0; }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
status_t getRenderPosition(size_t *halFrames, size_t *dspFrames);
int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
int EffectId);
status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track,
int EffectId);
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
virtual uint32_t hasAudioSession(int sessionId) const;
virtual uint32_t getStrategyForSession_l(int sessionId);
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
// called with AudioFlinger lock held
void invalidateTracks(audio_stream_type_t streamType);
virtual size_t frameCount() const { return mNormalFrameCount; }
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const { return mFrameCount; }
status_t getTimestamp_l(AudioTimestamp& timestamp);
protected:
// updated by readOutputParameters()
size_t mNormalFrameCount; // normal mixer and effects
int16_t* mMixBuffer; // frame size aligned mix buffer
int8_t* mAllocMixBuffer; // mixer buffer allocation address
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
// concurrent use of both of them, so Audio Policy Service suspends one of the threads to
// workaround that restriction.
// 'volatile' means accessed via atomic operations and no lock.
volatile int32_t mSuspended;
// FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples
// mFramesWritten would be better, or 64-bit even better
size_t mBytesWritten;
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
// copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
protected:
SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<>
SortedVector<int> mWakeLockUids;
int mActiveTracksGeneration;
wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks
// Allocate a track name for a given channel mask.
// Returns name >= 0 if successful, -1 on failure.
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
virtual void deleteTrackName_l(int name) = 0;
// Time to sleep between cycles when:
virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED
virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE
virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us
// No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
// No sleep in standby mode; waits on a condition
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
void checkSilentMode_l();
// Non-trivial for DUPLICATING only
virtual void saveOutputTracks() { }
virtual void clearOutputTracks() { }
// Cache various calculated values, at threadLoop() entry and after a parameter change
virtual void cacheParameters_l();
virtual uint32_t correctLatency_l(uint32_t latency) const;
private:
friend class AudioFlinger; // for numerous
PlaybackThread(const Client&);
PlaybackThread& operator = (const PlaybackThread&);
status_t addTrack_l(const sp<Track>& track);
bool destroyTrack_l(const sp<Track>& track);
void removeTrack_l(const sp<Track>& track);
void broadcast_l();
void readOutputParameters();
virtual void dumpInternals(int fd, const Vector<String16>& args);
void dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
// mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by
// DuplicatingThread
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1];
AudioStreamOut *mOutput;
float mMasterVolume;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
bool mInWrite;
// FIXME rename these former local variables of threadLoop to standard "m" names
nsecs_t standbyTime;
size_t mixBufferSize;
// cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
uint32_t activeSleepTime;
uint32_t idleSleepTime;
uint32_t sleepTime;
// mixer status returned by prepareTracks_l()
mixer_state mMixerStatus; // current cycle
// previous cycle when in prepareTracks_l()
mixer_state mMixerStatusIgnoringFastTracks;
// FIXME or a separate ready state per track
// FIXME move these declarations into the specific sub-class that needs them
// MIXER only
uint32_t sleepTimeShift;
// same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
nsecs_t standbyDelay;
// MIXER only
nsecs_t maxPeriod;
// DUPLICATING only
uint32_t writeFrames;
size_t mBytesRemaining;
size_t mCurrentWriteLength;
bool mUseAsyncWrite;
// mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is
// incremented each time a write(), a flush() or a standby() occurs.
// Bit 0 is set when a write blocks and indicates a callback is expected.
// Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence
// callbacks are ignored.
uint32_t mWriteAckSequence;
// mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is
// incremented each time a drain is requested or a flush() or standby() occurs.
// Bit 0 is set when the drain() command is called at the HAL and indicates a callback is
// expected.
// Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence
// callbacks are ignored.
uint32_t mDrainSequence;
// A condition that must be evaluated by prepareTrack_l() has changed and we must not wait
// for async write callback in the thread loop before evaluating it
bool mSignalPending;
sp<AsyncCallbackThread> mCallbackThread;
private:
// The HAL output sink is treated as non-blocking, but current implementation is blocking
sp<NBAIO_Sink> mOutputSink;
// If a fast mixer is present, the blocking pipe sink, otherwise clear
sp<NBAIO_Sink> mPipeSink;
// The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
sp<NBAIO_Sink> mNormalSink;
#ifdef TEE_SINK
// For dumpsys
sp<NBAIO_Sink> mTeeSink;
sp<NBAIO_Source> mTeeSource;
#endif
uint32_t mScreenState; // cached copy of gScreenState
static const size_t kFastMixerLogSize = 4 * 1024;
sp<NBLog::Writer> mFastMixerNBLogWriter;
public:
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
virtual void flushOutput_l();
private:
// timestamp latch:
// D input is written by threadLoop_write while mutex is unlocked, and read while locked
// Q output is written while locked, and read while locked
struct {
AudioTimestamp mTimestamp;
uint32_t mUnpresentedFrames;
} mLatchD, mLatchQ;
bool mLatchDValid; // true means mLatchD is valid, and clock it into latch at next opportunity
bool mLatchQValid; // true means mLatchQ is valid
};
class MixerThread : public PlaybackThread {
public:
MixerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool checkForNewParameters_l();
virtual void dumpInternals(int fd, const Vector<String16>& args);
protected:
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
virtual ssize_t threadLoop_write();
virtual void threadLoop_standby();
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
virtual uint32_t correctLatency_l(uint32_t latency) const;
AudioMixer* mAudioMixer; // normal mixer
private:
// one-time initialization, no locks required
FastMixer* mFastMixer; // non-NULL if there is also a fast mixer
sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
FastMixerDumpState mFastMixerDumpState;
#ifdef STATE_QUEUE_DUMP
StateQueueObserverDump mStateQueueObserverDump;
StateQueueMutatorDump mStateQueueMutatorDump;
#endif
AudioWatchdogDump mAudioWatchdogDump;
// accessible only within the threadLoop(), no locks required
// mFastMixer->sq() // for mutating and pushing state
int32_t mFastMixerFutex; // for cold idle
public:
virtual bool hasFastMixer() const { return mFastMixer != NULL; }
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks);
return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
}
};
class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device);
virtual ~DirectOutputThread();
// Thread virtuals
virtual bool checkForNewParameters_l();
protected:
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
// volumes last sent to audio HAL with stream->set_volume()
float mLeftVolFloat;
float mRightVolFloat;
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device, ThreadBase::type_t type);
void processVolume_l(Track *track, bool lastTrack);
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
sp<Track> mActiveTrack;
public:
virtual bool hasFastMixer() const { return false; }
};
class OffloadThread : public DirectOutputThread {
public:
OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device);
virtual ~OffloadThread() {};
protected:
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_exit();
virtual void flushOutput_l();
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
private:
void flushHw_l();
private:
bool mHwPaused;
bool mFlushPending;
size_t mPausedWriteLength; // length in bytes of write interrupted by pause
size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume
wp<Track> mPreviousTrack; // used to detect track switch
};
class AsyncCallbackThread : public Thread {
public:
AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
virtual ~AsyncCallbackThread();
// Thread virtuals
virtual bool threadLoop();
// RefBase
virtual void onFirstRef();
void exit();
void setWriteBlocked(uint32_t sequence);
void resetWriteBlocked();
void setDraining(uint32_t sequence);
void resetDraining();
private:
const wp<PlaybackThread> mPlaybackThread;
// mWriteAckSequence corresponds to the last write sequence passed by the offload thread via
// setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used
// to indicate that the callback has been received via resetWriteBlocked()
uint32_t mWriteAckSequence;
// mDrainSequence corresponds to the last drain sequence passed by the offload thread via
// setDraining(). The sequence is shifted one bit to the left and the lsb is used
// to indicate that the callback has been received via resetDraining()
uint32_t mDrainSequence;
Condition mWaitWorkCV;
Mutex mLock;
};
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
audio_io_handle_t id);
virtual ~DuplicatingThread();
// Thread virtuals
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() const { return mWaitTimeMs; }
protected:
virtual uint32_t activeSleepTimeUs() const;
private:
bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
protected:
// threadLoop snippets
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual ssize_t threadLoop_write();
virtual void threadLoop_standby();
virtual void cacheParameters_l();
private:
// called from threadLoop, addOutputTrack, removeOutputTrack
virtual void updateWaitTime_l();
protected:
virtual void saveOutputTracks();
virtual void clearOutputTracks();
private:
uint32_t mWaitTimeMs;
SortedVector < sp<OutputTrack> > outputTracks;
SortedVector < sp<OutputTrack> > mOutputTracks;
public:
virtual bool hasFastMixer() const { return false; }
};
// record thread
class RecordThread : public ThreadBase, public AudioBufferProvider
// derives from AudioBufferProvider interface for use by resampler
{
public:
#include "RecordTracks.h"
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_io_handle_t id,
audio_devices_t outDevice,
audio_devices_t inDevice
#ifdef TEE_SINK
, const sp<NBAIO_Sink>& teeSink
#endif
);
virtual ~RecordThread();
// no addTrack_l ?
void destroyTrack_l(const sp<RecordTrack>& track);
void removeTrack_l(const sp<RecordTrack>& track);
void dumpInternals(int fd, const Vector<String16>& args);
void dumpTracks(int fd, const Vector<String16>& args);
// Thread virtuals
virtual bool threadLoop();
virtual status_t readyToRun();
// RefBase
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
int triggerSession);
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
bool stop(RecordTrack* recordTrack);
void dump(int fd, const Vector<String16>& args);
AudioStreamIn* clearInput();
virtual audio_stream_t* stream() const;
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
virtual unsigned int getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
virtual uint32_t hasAudioSession(int sessionId) const;
// Return the set of unique session IDs across all tracks.
// The keys are the session IDs, and the associated values are meaningless.
// FIXME replace by Set [and implement Bag/Multiset for other uses].
KeyedVector<int, bool> sessionIds() const;
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
static void syncStartEventCallback(const wp<SyncEvent>& event);
void handleSyncStartEvent(const sp<SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
bool hasFastRecorder() const { return false; }
private:
void clearSyncStartEvent();
// Enter standby if not already in standby, and set mStandby flag
void standby();
// Call the HAL standby method unconditionally, and don't change mStandby flag
void inputStandBy();
AudioStreamIn *mInput;
SortedVector < sp<RecordTrack> > mTracks;
// mActiveTrack has dual roles: it indicates the current active track, and
// is used together with mStartStopCond to indicate start()/stop() progress
sp<RecordTrack> mActiveTrack;
Condition mStartStopCond;
// updated by RecordThread::readInputParameters()
AudioResampler *mResampler;
// interleaved stereo pairs of fixed-point signed Q19.12
int32_t *mRsmpOutBuffer;
int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount]
size_t mRsmpInIndex;
size_t mBufferSize; // stream buffer size for read()
const uint32_t mReqChannelCount;
const uint32_t mReqSampleRate;
ssize_t mBytesRead;
// sync event triggering actual audio capture. Frames read before this event will
// be dropped and therefore not read by the application.
sp<SyncEvent> mSyncStartEvent;
// number of captured frames to drop after the start sync event has been received.
// when < 0, maximum frames to drop before starting capture even if sync event is
// not received
ssize_t mFramestoDrop;
// For dumpsys
const sp<NBAIO_Sink> mTeeSink;
};