diff --git a/media/libaudioprocessing/AudioResamplerDyn.h b/media/libaudioprocessing/AudioResamplerDyn.h
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+/*
+ * Copyright (C) 2013 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_RESAMPLER_DYN_H
+#define ANDROID_AUDIO_RESAMPLER_DYN_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <android/log.h>
+
+#include <media/AudioResampler.h>
+
+namespace android {
+
+/* AudioResamplerDyn
+ *
+ * This class template is used for floating point and integer resamplers.
+ *
+ * Type variables:
+ * TC = filter coefficient type (one of int16_t, int32_t, or float)
+ * TI = input data type (one of int16_t or float)
+ * TO = output data type (one of int32_t or float)
+ *
+ * For integer input data types TI, the coefficient type TC is either int16_t or int32_t.
+ * For float input data types TI, the coefficient type TC is float.
+ */
+
+template<typename TC, typename TI, typename TO>
+class AudioResamplerDyn: public AudioResampler {
+public:
+    AudioResamplerDyn(int inChannelCount,
+            int32_t sampleRate, src_quality quality);
+
+    virtual ~AudioResamplerDyn();
+
+    virtual void init();
+
+    virtual void setSampleRate(int32_t inSampleRate);
+
+    virtual void setVolume(float left, float right);
+
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+
+private:
+
+    class Constants { // stores the filter constants.
+    public:
+        Constants() :
+            mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefs(NULL)
+        {}
+        void set(int L, int halfNumCoefs,
+                int inSampleRate, int outSampleRate);
+
+                 int mL;            // interpolation phases in the filter.
+                 int mShift;        // right shift to get polyphase index
+        unsigned int mHalfNumCoefs; // filter half #coefs
+           const TC* mFirCoefs;     // polyphase filter bank
+    };
+
+    class InBuffer { // buffer management for input type TI
+    public:
+        InBuffer();
+        ~InBuffer();
+        void init();
+
+        void resize(int CHANNELS, int halfNumCoefs);
+
+        // used for direct management of the mImpulse pointer
+        inline TI* getImpulse() {
+            return mImpulse;
+        }
+
+        inline void setImpulse(TI *impulse) {
+            mImpulse = impulse;
+        }
+
+        template<int CHANNELS>
+        inline void readAgain(TI*& impulse, const int halfNumCoefs,
+                const TI* const in, const size_t inputIndex);
+
+        template<int CHANNELS>
+        inline void readAdvance(TI*& impulse, const int halfNumCoefs,
+                const TI* const in, const size_t inputIndex);
+
+        void reset();
+
+    private:
+        // tuning parameter guidelines: 2 <= multiple <= 8
+        static const int kStateSizeMultipleOfFilterLength = 4;
+
+        // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
+           TI* mState;      // base pointer for the input buffer storage
+           TI* mImpulse;    // current location of the impulse response (centered)
+           TI* mRingFull;   // mState <= mImpulse < mRingFull
+        size_t mStateCount; // size of state in units of TI.
+    };
+
+    void createKaiserFir(Constants &c, double stopBandAtten,
+            int inSampleRate, int outSampleRate, double tbwCheat);
+
+    template<int CHANNELS, bool LOCKED, int STRIDE>
+    size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+
+    // define a pointer to member function type for resample
+    typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+            size_t outFrameCount, AudioBufferProvider* provider);
+
+    // data - the contiguous storage and layout of these is important.
+           InBuffer mInBuffer;
+          Constants mConstants;        // current set of coefficient parameters
+    TO __attribute__ ((aligned (8))) mVolumeSimd[2]; // must be aligned or NEON may crash
+     resample_ABP_t mResampleFunc;     // called function for resampling
+            int32_t mFilterSampleRate; // designed filter sample rate.
+        src_quality mFilterQuality;    // designed filter quality.
+              void* mCoefBuffer;       // if a filter is created, this is not null
+};
+
+} // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/