diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
new file mode 100644
index 0000000..06a1c8c
--- /dev/null
+++ b/services/audioflinger/Threads.h
@@ -0,0 +1,801 @@
+/*
+**
+** 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
+    };
+
+    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.
+                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; }
+
+    // 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);
+                // 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();
+                void        acquireWakeLock_l();
+                void        releaseWakeLock();
+                void        releaseWakeLock_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);
+
+    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;
+                uint32_t                mSampleRate;
+                size_t                  mFrameCount;       // output HAL, direct output, record
+                size_t                  mNormalFrameCount; // normal mixer and effects
+                audio_channel_mask_t    mChannelMask;
+                uint16_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<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;
+};
+
+// --- 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
+        // standby mode does not have an enum value
+        // suspend by audio policy manager is orthogonal to mixer state
+    };
+
+    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     void        threadLoop_write();
+    virtual     void        threadLoop_standby();
+    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;
+
+    // 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,
+                                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;
+                        void     invalidateTracks(audio_stream_type_t streamType);
+
+
+protected:
+    int16_t*                        mMixBuffer;
+
+    // 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<>
+
+    // 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);
+    void        destroyTrack_l(const sp<Track>& track);
+    void        removeTrack_l(const sp<Track>& track);
+
+    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;
+
+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;
+    // For dumpsys
+    sp<NBAIO_Sink>          mTeeSink;
+    sp<NBAIO_Source>        mTeeSource;
+    uint32_t                mScreenState;   // cached copy of gScreenState
+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
+
+};
+
+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     void        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();
+
+private:
+    // volumes last sent to audio HAL with stream->set_volume()
+    float mLeftVolFloat;
+    float mRightVolFloat;
+
+    // prepareTracks_l() tells threadLoop_mix() the name of the single active track
+    sp<Track>               mActiveTrack;
+public:
+    virtual     bool        hasFastMixer() const { return false; }
+};
+
+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     void        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 device,
+                    const sp<NBAIO_Sink>& teeSink);
+            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,
+                    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_l(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);
+
+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;
+            AudioResampler                      *mResampler;
+            int32_t                             *mRsmpOutBuffer;
+            int16_t                             *mRsmpInBuffer;
+            size_t                              mRsmpInIndex;
+            size_t                              mInputBytes;
+            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;
+};