diff --git a/media/libaudioprocessing/AudioResamplerSinc.h b/media/libaudioprocessing/AudioResamplerSinc.h
new file mode 100644
index 0000000..f6dcf91
--- /dev/null
+++ b/media/libaudioprocessing/AudioResamplerSinc.h
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 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_RESAMPLER_SINC_H
+#define ANDROID_AUDIO_RESAMPLER_SINC_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <android/log.h>
+
+#include <media/AudioResampler.h>
+
+namespace android {
+
+
+typedef const int32_t * (*readCoefficientsFn)(bool upDownSample);
+typedef int32_t (*readResampleFirNumCoeffFn)();
+typedef int32_t (*readResampleFirLerpIntBitsFn)();
+
+// ----------------------------------------------------------------------------
+
+class AudioResamplerSinc : public AudioResampler {
+public:
+    AudioResamplerSinc(int inChannelCount, int32_t sampleRate,
+            src_quality quality = HIGH_QUALITY);
+
+    virtual ~AudioResamplerSinc();
+
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+private:
+    void init();
+
+    virtual void setVolume(float left, float right);
+
+    template<int CHANNELS>
+    size_t resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+
+    template<int CHANNELS>
+    inline void filterCoefficient(
+            int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL);
+
+    template<int CHANNELS>
+    inline void interpolate(
+            int32_t& l, int32_t& r,
+            const int32_t* coefs, size_t offset,
+            int32_t lerp, const int16_t* samples);
+
+    template<int CHANNELS>
+    inline void read(int16_t*& impulse, uint32_t& phaseFraction,
+            const int16_t* in, size_t inputIndex);
+
+    int16_t *mState;
+    int16_t *mImpulse;
+    int16_t *mRingFull;
+    int32_t mVolumeSIMD[2];
+
+    const int32_t * mFirCoefs;
+    static const uint32_t mFirCoefsDown[];
+    static const uint32_t mFirCoefsUp[];
+
+    // ----------------------------------------------------------------------------
+    static const int32_t RESAMPLE_FIR_NUM_COEF       = 8;
+    static const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 7;
+
+    struct Constants {
+        int coefsBits;
+        int cShift;
+        uint32_t cMask;
+        int pShift;
+        uint32_t pMask;
+        // number of zero-crossing on each side
+        unsigned int halfNumCoefs;
+    };
+
+    static Constants highQualityConstants;
+    static Constants veryHighQualityConstants;
+    const Constants *mConstants;    // points to appropriate set of coefficient parameters
+
+    static void init_routine();
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/