diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
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+++ b/include/media/AudioMixer.h
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+/*
+**
+** 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 <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/BufferProviders.h>
+#include <media/nbaio/NBLog.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+#include <utils/threads.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+                            AudioMixer(size_t frameCount, uint32_t sampleRate,
+                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
+
+    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
+
+
+    // This mixer has a hard-coded upper limit of 32 active track inputs.
+    // Adding support for > 32 tracks would require more than simply changing this value.
+    static const uint32_t MAX_NUM_TRACKS = 32;
+    // maximum number of channels supported by the mixer
+
+    // This mixer has a hard-coded upper limit of 8 channels for output.
+    static const uint32_t MAX_NUM_CHANNELS = 8;
+    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
+    // maximum number of channels supported for the content
+    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
+    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+    enum { // names
+
+        // track names (MAX_NUM_TRACKS units)
+        TRACK0          = 0x1000,
+
+        // 0x2000 is unused
+
+        // setParameter targets
+        TRACK           = 0x3000,
+        RESAMPLE        = 0x3001,
+        RAMP_VOLUME     = 0x3002, // ramp to new volume
+        VOLUME          = 0x3003, // don't ramp
+        TIMESTRETCH     = 0x3004,
+
+        // set Parameter names
+        // for target TRACK
+        CHANNEL_MASK    = 0x4000,
+        FORMAT          = 0x4001,
+        MAIN_BUFFER     = 0x4002,
+        AUX_BUFFER      = 0x4003,
+        DOWNMIX_TYPE    = 0X4004,
+        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+        // for target RESAMPLE
+        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
+                                  // parameter 'value' is the new sample rate in Hz.
+                                  // Only creates a sample rate converter the first time that
+                                  // the track sample rate is different from the mix sample rate.
+                                  // If the new sample rate is the same as the mix sample rate,
+                                  // and a sample rate converter already exists,
+                                  // then the sample rate converter remains present but is a no-op.
+        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
+                                  // This clears out the resampler's input buffer.
+        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
+                                  // the track is restored to the mix sample rate.
+        // for target RAMP_VOLUME and VOLUME (8 channels max)
+        // FIXME use float for these 3 to improve the dynamic range
+        VOLUME0         = 0x4200,
+        VOLUME1         = 0x4201,
+        AUXLEVEL        = 0x4210,
+        // for target TIMESTRETCH
+        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
+                                  // parameter 'value' is a pointer to the new playback rate.
+    };
+
+
+    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
+
+    // Allocate a track name.  Returns new track name if successful, -1 on failure.
+    // The failure could be because of an invalid channelMask or format, or that
+    // the track capacity of the mixer is exceeded.
+    int         getTrackName(audio_channel_mask_t channelMask,
+                             audio_format_t format, int sessionId);
+
+    // Free an allocated track by name
+    void        deleteTrackName(int name);
+
+    // Enable or disable an allocated track by name
+    void        enable(int name);
+    void        disable(int name);
+
+    void        setParameter(int name, int target, int param, void *value);
+
+    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+    void        process();
+
+    uint32_t    trackNames() const { return mTrackNames; }
+
+    size_t      getUnreleasedFrames(int name) const;
+
+    static inline bool isValidPcmTrackFormat(audio_format_t format) {
+        switch (format) {
+        case AUDIO_FORMAT_PCM_8_BIT:
+        case AUDIO_FORMAT_PCM_16_BIT:
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+        case AUDIO_FORMAT_PCM_32_BIT:
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return true;
+        default:
+            return false;
+        }
+    }
+
+private:
+
+    enum {
+        // FIXME this representation permits up to 8 channels
+        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
+    };
+
+    enum {
+        NEEDS_CHANNEL_1             = 0x00000000,   // mono
+        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
+
+        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+        NEEDS_MUTE                  = 0x00000100,
+        NEEDS_RESAMPLE              = 0x00001000,
+        NEEDS_AUX                   = 0x00010000,
+    };
+
+    struct state_t;
+    struct track_t;
+
+    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
+                           int32_t* aux);
+    static const int BLOCKSIZE = 16; // 4 cache lines
+
+    struct track_t {
+        uint32_t    needs;
+
+        // TODO: Eventually remove legacy integer volume settings
+        union {
+        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+        int32_t     volumeRL;
+        };
+
+        int32_t     prevVolume[MAX_NUM_VOLUMES];
+
+        // 16-byte boundary
+
+        int32_t     volumeInc[MAX_NUM_VOLUMES];
+        int32_t     auxInc;
+        int32_t     prevAuxLevel;
+
+        // 16-byte boundary
+
+        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+        uint16_t    frameCount;
+
+        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+        uint8_t     unused_padding; // formerly format, was always 16
+        uint16_t    enabled;        // actually bool
+        audio_channel_mask_t channelMask;
+
+        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
+        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
+        AudioBufferProvider*                bufferProvider;
+
+        // 16-byte boundary
+
+        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+        hook_t      hook;
+        const void* in;             // current location in buffer
+
+        // 16-byte boundary
+
+        AudioResampler*     resampler;
+        uint32_t            sampleRate;
+        int32_t*           mainBuffer;
+        int32_t*           auxBuffer;
+
+        // 16-byte boundary
+
+        /* Buffer providers are constructed to translate the track input data as needed.
+         *
+         * TODO: perhaps make a single PlaybackConverterProvider class to move
+         * all pre-mixer track buffer conversions outside the AudioMixer class.
+         *
+         * 1) mInputBufferProvider: The AudioTrack buffer provider.
+         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+         *    requires reformat. For example, it may convert floating point input to
+         *    PCM_16_bit if that's required by the downmixer.
+         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+         *    the number of channels required by the mixer sink.
+         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+         *    the downmixer requirements to the mixer engine input requirements.
+         * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
+         */
+        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
+        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
+        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
+        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
+        PassthruBufferProvider*  mTimestretchBufferProvider;
+
+        int32_t     sessionId;
+
+        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        audio_format_t mFormat;          // input track format
+        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+                                         // each track must be converted to this format.
+        audio_format_t mDownmixRequiresFormat;  // required downmixer format
+                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+                                                // AUDIO_FORMAT_INVALID if no required format
+
+        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
+        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
+
+        float          mAuxLevel;                     // floating point set aux level
+        float          mPrevAuxLevel;                 // floating point prev aux level
+        float          mAuxInc;                       // floating point aux increment
+
+        audio_channel_mask_t mMixerChannelMask;
+        uint32_t             mMixerChannelCount;
+
+        AudioPlaybackRate    mPlaybackRate;
+
+        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+        bool        doesResample() const { return resampler != NULL; }
+        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
+        void        adjustVolumeRamp(bool aux, bool useFloat = false);
+        size_t      getUnreleasedFrames() const { return resampler != NULL ?
+                                                    resampler->getUnreleasedFrames() : 0; };
+
+        status_t    prepareForDownmix();
+        void        unprepareForDownmix();
+        status_t    prepareForReformat();
+        void        unprepareForReformat();
+        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
+        void        reconfigureBufferProviders();
+    };
+
+    typedef void (*process_hook_t)(state_t* state);
+
+    // pad to 32-bytes to fill cache line
+    struct state_t {
+        uint32_t        enabledTracks;
+        uint32_t        needsChanged;
+        size_t          frameCount;
+        process_hook_t  hook;   // one of process__*, never NULL
+        int32_t         *outputTemp;
+        int32_t         *resampleTemp;
+        NBLog::Writer*  mLog;
+        int32_t         reserved[1];
+        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
+        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
+    };
+
+    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
+    uint32_t        mTrackNames;
+
+    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
+    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
+    const uint32_t  mConfiguredNames;
+
+    const uint32_t  mSampleRate;
+
+    NBLog::Writer   mDummyLog;
+public:
+    void            setLog(NBLog::Writer* log);
+private:
+    state_t         mState __attribute__((aligned(32)));
+
+    // Call after changing either the enabled status of a track, or parameters of an enabled track.
+    // OK to call more often than that, but unnecessary.
+    void invalidateState(uint32_t mask);
+
+    bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
+    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
+    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
+    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+            int32_t* aux);
+    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+            int32_t* aux);
+
+    static void process__validate(state_t* state);
+    static void process__nop(state_t* state);
+    static void process__genericNoResampling(state_t* state);
+    static void process__genericResampling(state_t* state);
+    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
+
+    static pthread_once_t   sOnceControl;
+    static void             sInitRoutine();
+
+    /* multi-format volume mixing function (calls template functions
+     * in AudioMixerOps.h).  The template parameters are as follows:
+     *
+     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+     *   USEFLOATVOL (set to true if float volume is used)
+     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+     *   TO: int32_t (Q4.27) or float
+     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+     *   TA: int32_t (Q4.27)
+     */
+    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+        typename TO, typename TI, typename TA>
+    static void volumeMix(TO *out, size_t outFrames,
+            const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
+
+    // multi-format process hooks
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void process_NoResampleOneTrack(state_t* state);
+
+    // multi-format track hooks
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void track__Resample(track_t* t, TO* out, size_t frameCount,
+            TO* temp __unused, TA* aux);
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+            TO* temp __unused, TA* aux);
+
+    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+            void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+    // hook types
+    enum {
+        PROCESSTYPE_NORESAMPLEONETRACK,
+    };
+    enum {
+        TRACKTYPE_NOP,
+        TRACKTYPE_RESAMPLE,
+        TRACKTYPE_NORESAMPLE,
+        TRACKTYPE_NORESAMPLEMONO,
+    };
+
+    // functions for determining the proper process and track hooks.
+    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+    static hook_t getTrackHook(int trackType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H