diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
new file mode 100644
index 0000000..17de49b
--- /dev/null
+++ b/services/audioflinger/TrackBase.h
@@ -0,0 +1,139 @@
+/*
+**
+** 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
+
+// base for record and playback
+class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
+
+public:
+    enum track_state {
+        IDLE,
+        TERMINATED,
+        FLUSHED,
+        STOPPED,
+        // next 2 states are currently used for fast tracks only
+        STOPPING_1,     // waiting for first underrun
+        STOPPING_2,     // waiting for presentation complete
+        RESUMING,
+        ACTIVE,
+        PAUSING,
+        PAUSED
+    };
+
+                        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);
+    virtual             ~TrackBase();
+
+    virtual status_t    start(AudioSystem::sync_event_t event,
+                             int triggerSession) = 0;
+    virtual void        stop() = 0;
+            sp<IMemory> getCblk() const { return mCblkMemory; }
+            audio_track_cblk_t* cblk() const { return mCblk; }
+            int         sessionId() const { return mSessionId; }
+    virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
+
+protected:
+                        TrackBase(const TrackBase&);
+                        TrackBase& operator = (const TrackBase&);
+
+    // AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
+    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
+    // ExtendedAudioBufferProvider interface is only needed for Track,
+    // but putting it in TrackBase avoids the complexity of virtual inheritance
+    virtual size_t  framesReady() const { return SIZE_MAX; }
+
+    audio_format_t format() const {
+        return mFormat;
+    }
+
+    uint32_t channelCount() const { return mChannelCount; }
+
+    audio_channel_mask_t channelMask() const { return mChannelMask; }
+
+    uint32_t sampleRate() const; // FIXME inline after cblk sr moved
+
+    // Return a pointer to the start of a contiguous slice of the track buffer.
+    // Parameter 'offset' is the requested start position, expressed in
+    // monotonically increasing frame units relative to the track epoch.
+    // Parameter 'frames' is the requested length, also in frame units.
+    // Always returns non-NULL.  It is the caller's responsibility to
+    // verify that this will be successful; the result of calling this
+    // function with invalid 'offset' or 'frames' is undefined.
+    void* getBuffer(uint32_t offset, uint32_t frames) const;
+
+    bool isStopped() const {
+        return (mState == STOPPED || mState == FLUSHED);
+    }
+
+    // for fast tracks only
+    bool isStopping() const {
+        return mState == STOPPING_1 || mState == STOPPING_2;
+    }
+    bool isStopping_1() const {
+        return mState == STOPPING_1;
+    }
+    bool isStopping_2() const {
+        return mState == STOPPING_2;
+    }
+
+    bool isTerminated() const {
+        return mState == TERMINATED;
+    }
+
+    bool step();    // mStepCount is an implicit input
+    void reset();
+
+    virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack,
+                                    // this could be a track type if needed later
+
+    const wp<ThreadBase> mThread;
+    /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
+    sp<IMemory>         mCblkMemory;
+    audio_track_cblk_t* mCblk;
+    void*               mBuffer;    // start of track buffer, typically in shared memory
+    void*               mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize
+                                    //   is based on mChannelCount and 16-bit samples
+    uint32_t            mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of
+                                    // time of releaseBuffer() for later use by step()
+    // we don't really need a lock for these
+    track_state         mState;
+    const uint32_t      mSampleRate;    // initial sample rate only; for tracks which
+                        // support dynamic rates, the current value is in control block
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const uint8_t       mChannelCount;
+    const size_t        mFrameSize; // AudioFlinger's view of frame size in shared memory,
+                                    // where for AudioTrack (but not AudioRecord),
+                                    // 8-bit PCM samples are stored as 16-bit
+    const size_t        mFrameCount;// size of track buffer given at createTrack() or
+                                    // openRecord(), and then adjusted as needed
+
+    bool                mStepServerFailed;
+    const int           mSessionId;
+    Vector < sp<SyncEvent> >mSyncEvents;
+};