diff --git a/libs/audioflinger/AudioMixer.h b/libs/audioflinger/AudioMixer.h
new file mode 100644
index 0000000..72ca28a
--- /dev/null
+++ b/libs/audioflinger/AudioMixer.h
@@ -0,0 +1,192 @@
+/* //device/include/server/AudioFlinger/AudioMixer.h
+**
+** Copyright 2007, 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 ANDROID_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include "AudioBufferProvider.h"
+#include "AudioResampler.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
+#define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+                            AudioMixer(size_t frameCount, uint32_t sampleRate);
+
+                            ~AudioMixer();
+
+    static const uint32_t MAX_NUM_TRACKS = 32;
+    static const uint32_t MAX_NUM_CHANNELS = 2;
+
+    static const uint16_t UNITY_GAIN = 0x1000;
+
+    enum { // names
+
+        // track units (32 units)
+        TRACK0          = 0x1000,
+
+        // enable/disable
+        MIXING          = 0x2000,
+
+        // setParameter targets
+        TRACK           = 0x3000,
+        RESAMPLE        = 0x3001,
+        RAMP_VOLUME     = 0x3002, // ramp to new volume
+        VOLUME          = 0x3003, // don't ramp
+
+        // set Parameter names
+        // for target TRACK
+        CHANNEL_COUNT   = 0x4000,
+        FORMAT          = 0x4001,
+        // for TARGET RESAMPLE
+        SAMPLE_RATE     = 0x4100,
+        // for TARGET VOLUME (8 channels max)
+        VOLUME0         = 0x4200,
+        VOLUME1         = 0x4201,
+    };
+
+
+    int         getTrackName();
+    void        deleteTrackName(int name);
+
+    status_t    enable(int name);
+    status_t    disable(int name);
+
+    status_t    setActiveTrack(int track);
+    status_t    setParameter(int target, int name, int value);
+
+    status_t    setBufferProvider(AudioBufferProvider* bufferProvider);
+    void        process(void* output);
+
+    uint32_t    trackNames() const { return mTrackNames; }
+
+private:
+
+    enum {
+        NEEDS_CHANNEL_COUNT__MASK   = 0x00000003,
+        NEEDS_FORMAT__MASK          = 0x000000F0,
+        NEEDS_MUTE__MASK            = 0x00000100,
+        NEEDS_RESAMPLE__MASK        = 0x00001000,
+    };
+
+    enum {
+        NEEDS_CHANNEL_1             = 0x00000000,
+        NEEDS_CHANNEL_2             = 0x00000001,
+
+        NEEDS_FORMAT_16             = 0x00000010,
+
+        NEEDS_MUTE_DISABLED         = 0x00000000,
+        NEEDS_MUTE_ENABLED          = 0x00000100,
+
+        NEEDS_RESAMPLE_DISABLED     = 0x00000000,
+        NEEDS_RESAMPLE_ENABLED      = 0x00001000,
+    };
+
+    static inline int32_t applyVolume(int32_t in, int32_t v) {
+        return in * v;
+    }
+
+
+    struct state_t;
+
+    typedef void (*mix_t)(state_t* state, void* output);
+
+    static const int BLOCKSIZE = 16; // 4 cache lines
+
+    struct track_t {
+        uint32_t    needs;
+
+        union {
+        int16_t     volume[2];      // [0]3.12 fixed point
+        int32_t     volumeRL;
+        };
+
+        int32_t     prevVolume[2];
+
+        int32_t     volumeInc[2];
+
+        uint16_t    frameCount;
+
+        uint8_t     channelCount : 4;
+        uint8_t     enabled      : 1;
+        uint8_t     reserved0    : 3;
+        uint8_t     format;
+
+        AudioBufferProvider*                bufferProvider;
+        mutable AudioBufferProvider::Buffer buffer;
+
+        void (*hook)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp);
+        void const* in;             // current location in buffer
+
+        AudioResampler*     resampler;
+        uint32_t            sampleRate;
+
+        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
+        bool        doesResample() const;
+        void        adjustVolumeRamp();
+    };
+
+    // pad to 32-bytes to fill cache line
+    struct state_t {
+        uint32_t        enabledTracks;
+        uint32_t        needsChanged;
+        size_t          frameCount;
+        mix_t           hook;
+        int32_t         *outputTemp;
+        int32_t         *resampleTemp;
+        int32_t         reserved[2];
+        track_t         tracks[32]; __attribute__((aligned(32)));
+    };
+
+    int             mActiveTrack;
+    uint32_t        mTrackNames;
+    const uint32_t  mSampleRate;
+
+    state_t         mState __attribute__((aligned(32)));
+
+    void invalidateState(uint32_t mask);
+
+    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp);
+    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
+
+    static void process__validate(state_t* state, void* output);
+    static void process__nop(state_t* state, void* output);
+    static void process__genericNoResampling(state_t* state, void* output);
+    static void process__genericResampling(state_t* state, void* output);
+    static void process__OneTrack16BitsStereoNoResampling(state_t* state, void* output);
+    static void process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output);
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H