diff --git a/libs/audioflinger/AudioResamplerSinc.h b/libs/audioflinger/AudioResamplerSinc.h
new file mode 100644
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--- /dev/null
+++ b/libs/audioflinger/AudioResamplerSinc.h
@@ -0,0 +1,88 @@
+/*
+ * 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 <cutils/log.h>
+
+#include "AudioResampler.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioResamplerSinc : public AudioResampler {
+public:
+    AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
+
+    ~AudioResamplerSinc();
+
+    virtual void resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+private:
+    void init();
+
+    template<int CHANNELS>
+    void resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+
+    template<int CHANNELS>
+    inline void filterCoefficient(
+            int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples);
+
+    template<int CHANNELS>
+    inline void interpolate(
+            int32_t& l, int32_t& r,
+            int32_t const* coefs, int16_t lerp, int16_t const* samples);
+
+    template<int CHANNELS>
+    inline void read(int16_t*& impulse, uint32_t& phaseFraction,
+            int16_t const* in, size_t inputIndex);
+
+    int16_t *mState;
+    int16_t *mImpulse;
+    int16_t *mRingFull;
+
+    int32_t const * mFirCoefs;
+    static const int32_t mFirCoefsDown[];
+    static const int32_t mFirCoefsUp[];
+
+    // ----------------------------------------------------------------------------
+    static const int32_t RESAMPLE_FIR_NUM_COEF       = 8;
+    static const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 4;
+
+    // we have 16 coefs samples per zero-crossing
+    static const int coefsBits = RESAMPLE_FIR_LERP_INT_BITS;        // 4
+    static const int cShift = kNumPhaseBits - coefsBits;            // 26
+    static const uint32_t cMask  = ((1<<coefsBits)-1) << cShift;    // 0xf<<26 = 3c00 0000
+
+    // and we use 15 bits to interpolate between these samples
+    // this cannot change because the mul below rely on it.
+    static const int pLerpBits = 15;
+    static const int pShift = kNumPhaseBits - coefsBits - pLerpBits;    // 11
+    static const uint32_t pMask  = ((1<<pLerpBits)-1) << pShift;    // 0x7fff << 11
+
+    // number of zero-crossing on each side
+    static const unsigned int halfNumCoefs = RESAMPLE_FIR_NUM_COEF;
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/