diff --git a/libs/audioflinger/AudioResamplerSinc.cpp b/libs/audioflinger/AudioResamplerSinc.cpp
new file mode 100644
index 0000000..e710d16
--- /dev/null
+++ b/libs/audioflinger/AudioResamplerSinc.cpp
@@ -0,0 +1,320 @@
+/*
+ * 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.
+ */
+
+#include <string.h>
+#include "AudioResamplerSinc.h"
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+
+/*
+ * These coeficients are computed with the "fir" utility found in
+ * tools/resampler_tools
+ * TODO: A good optimization would be to transpose this matrix, to take
+ * better advantage of the data-cache. 
+ */
+const int32_t AudioResamplerSinc::mFirCoefsUp[] = {
+		0x7fffffff, 0x7f15d078, 0x7c5e0da6, 0x77ecd867, 0x71e2e251, 0x6a6c304a, 0x61be7269, 0x58170412, 0x4db8ab05, 0x42e92ea6, 0x37eee214, 0x2d0e3bb1, 0x22879366, 0x18951e95, 0x0f693d0d, 0x072d2621, 
+	    0x00000000, 0xf9f66655, 0xf51a5fd7, 0xf16bbd84, 0xeee0d9ac, 0xed67a922, 0xece70de6, 0xed405897, 0xee50e505, 0xeff3be30, 0xf203370f, 0xf45a6741, 0xf6d67d53, 0xf957db66, 0xfbc2f647, 0xfe00f2b9, 
+	    0x00000000, 0x01b37218, 0x0313a0c6, 0x041d930d, 0x04d28057, 0x053731b0, 0x05534dff, 0x05309bfd, 0x04da440d, 0x045c1aee, 0x03c1fcdd, 0x03173ef5, 0x02663ae8, 0x01b7f736, 0x0113ec79, 0x007fe6a9, 
+	    0x00000000, 0xff96b229, 0xff44f99f, 0xff0a86be, 0xfee5f803, 0xfed518fd, 0xfed521fd, 0xfee2f4fd, 0xfefb54f8, 0xff1b159b, 0xff3f4203, 0xff6539e0, 0xff8ac502, 0xffae1ddd, 0xffcdf3f9, 0xffe96798, 
+	    0x00000000, 0x00119de6, 0x001e6b7e, 0x0026cb7a, 0x002b4830, 0x002c83d6, 0x002b2a82, 0x0027e67a, 0x002356f9, 0x001e098e, 0x001875e4, 0x0012fbbe, 0x000de2d1, 0x00095c10, 0x00058414, 0x00026636, 
+	    0x00000000, 0xfffe44a9, 0xfffd206d, 0xfffc7b7f, 0xfffc3c8f, 0xfffc4ac2, 0xfffc8f2b, 0xfffcf5c4, 0xfffd6df3, 0xfffdeab2, 0xfffe6275, 0xfffececf, 0xffff2c07, 0xffff788c, 0xffffb471, 0xffffe0f2, 
+	    0x00000000, 0x000013e6, 0x00001f03, 0x00002396, 0x00002399, 0x000020b6, 0x00001c3c, 0x00001722, 0x00001216, 0x00000d81, 0x0000099c, 0x0000067c, 0x00000419, 0x0000025f, 0x00000131, 0x00000070, 
+	    0x00000000, 0xffffffc7, 0xffffffb3, 0xffffffb3, 0xffffffbe, 0xffffffcd, 0xffffffdb, 0xffffffe7, 0xfffffff0, 0xfffffff7, 0xfffffffb, 0xfffffffe, 0xffffffff, 0x00000000, 0x00000000, 0x00000000, 
+	    0x00000000 // this one is needed for lerping the last coefficient
+};
+
+/*
+ * These coefficients are optimized for 48KHz -> 44.1KHz (stop-band at 22.050KHz)
+ * It's possible to use the above coefficient for any down-sampling
+ * at the expense of a slower processing loop (we can interpolate
+ * these coefficient from the above by "Stretching" them in time).
+ */
+const int32_t AudioResamplerSinc::mFirCoefsDown[] = {
+		0x7fffffff, 0x7f55e46d, 0x7d5b4c60, 0x7a1b4b98, 0x75a7fb14, 0x7019f0bd, 0x698f875a, 0x622bfd59, 0x5a167256, 0x5178cc54, 0x487e8e6c, 0x3f53aae8, 0x36235ad4, 0x2d17047b, 0x245539ab, 0x1c00d540, 
+	    0x14383e57, 0x0d14d5ca, 0x06aa910b, 0x0107c38b, 0xfc351654, 0xf835abae, 0xf5076b45, 0xf2a37202, 0xf0fe9faa, 0xf00a3bbd, 0xefb4aa81, 0xefea2b05, 0xf0959716, 0xf1a11e83, 0xf2f6f7a0, 0xf481fff4, 
+	    0xf62e48ce, 0xf7e98ca5, 0xf9a38b4c, 0xfb4e4bfa, 0xfcde456f, 0xfe4a6d30, 0xff8c2fdf, 0x009f5555, 0x0181d393, 0x0233940f, 0x02b62f06, 0x030ca07d, 0x033afa62, 0x03461725, 0x03334f83, 0x030835fa, 
+	    0x02ca59cc, 0x027f12d1, 0x022b570d, 0x01d39a49, 0x017bb78f, 0x0126e414, 0x00d7aaaf, 0x008feec7, 0x0050f584, 0x001b73e3, 0xffefa063, 0xffcd46ed, 0xffb3ddcd, 0xffa29aaa, 0xff988691, 0xff949066, 
+	    0xff959d24, 0xff9a959e, 0xffa27195, 0xffac4011, 0xffb72d2b, 0xffc28569, 0xffcdb706, 0xffd85171, 0xffe20364, 0xffea97e9, 0xfff1f2b2, 0xfff80c06, 0xfffcec92, 0x0000a955, 0x00035fd8, 0x000532cf, 
+	    0x00064735, 0x0006c1f9, 0x0006c62d, 0x000673ba, 0x0005e68f, 0x00053630, 0x000475a3, 0x0003b397, 0x0002fac1, 0x00025257, 0x0001be9e, 0x0001417a, 0x0000dafd, 0x000089eb, 0x00004c28, 0x00001f1d, 
+	    0x00000000, 0xffffec10, 0xffffe0be, 0xffffdbc5, 0xffffdb39, 0xffffdd8b, 0xffffe182, 0xffffe638, 0xffffeb0a, 0xffffef8f, 0xfffff38b, 0xfffff6e3, 0xfffff993, 0xfffffba6, 0xfffffd30, 0xfffffe4a, 
+	    0xffffff09, 0xffffff85, 0xffffffd1, 0xfffffffb, 0x0000000f, 0x00000016, 0x00000015, 0x00000012, 0x0000000d, 0x00000009, 0x00000006, 0x00000003, 0x00000002, 0x00000001, 0x00000000, 0x00000000, 
+	    0x00000000 // this one is needed for lerping the last coefficient
+};
+
+// ----------------------------------------------------------------------------
+
+static inline 
+int32_t mulRL(int left, int32_t in, uint32_t vRL)
+{
+#if defined(__arm__) && !defined(__thumb__)
+    int32_t out;
+    if (left) {
+        asm( "smultb %[out], %[in], %[vRL] \n"
+             : [out]"=r"(out)
+             : [in]"%r"(in), [vRL]"r"(vRL)
+             : );
+    } else {
+        asm( "smultt %[out], %[in], %[vRL] \n"
+             : [out]"=r"(out)
+             : [in]"%r"(in), [vRL]"r"(vRL)
+             : );
+    }
+    return out;
+#else
+    if (left) {
+        return int16_t(in>>16) * int16_t(vRL&0xFFFF);
+    } else {
+        return int16_t(in>>16) * int16_t(vRL>>16);
+    }
+#endif
+}
+
+static inline 
+int32_t mulAdd(int16_t in, int32_t v, int32_t a)
+{
+#if defined(__arm__) && !defined(__thumb__)
+    int32_t out;
+    asm( "smlawb %[out], %[v], %[in], %[a] \n"
+         : [out]"=r"(out)
+         : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+         : );
+    return out;
+#else    
+    return a + ((in * int32_t(v))>>16);
+#endif
+}
+
+static inline 
+int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a)
+{
+#if defined(__arm__) && !defined(__thumb__)
+    int32_t out;
+    if (left) {
+        asm( "smlawb %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    } else {
+        asm( "smlawt %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    }
+    return out;
+#else
+    if (left) {
+        return a + ((int16_t(inRL&0xFFFF) * int32_t(v))>>16);
+    } else {
+        return a + ((int16_t(inRL>>16) * int32_t(v))>>16);
+    }
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioResamplerSinc::AudioResamplerSinc(int bitDepth,
+		int inChannelCount, int32_t sampleRate)
+	: AudioResampler(bitDepth, inChannelCount, sampleRate),
+	mState(0)
+{
+	/* 
+	 * Layout of the state buffer for 32 tap:
+	 * 
+	 * "present" sample            beginning of 2nd buffer
+	 *                 v                v
+	 *  0              01               2              23              3
+	 *  0              F0               0              F0              F
+	 * [pppppppppppppppInnnnnnnnnnnnnnnnpppppppppppppppInnnnnnnnnnnnnnnn]
+	 *                 ^               ^ head
+	 * 
+	 * p = past samples, convoluted with the (p)ositive side of sinc()
+	 * n = future samples, convoluted with the (n)egative side of sinc()
+	 * r = extra space for implementing the ring buffer
+	 * 
+	 */
+
+	const size_t numCoefs = 2*halfNumCoefs;
+	const size_t stateSize = numCoefs * inChannelCount * 2;
+	mState = new int16_t[stateSize];
+	memset(mState, 0, sizeof(int16_t)*stateSize);
+	mImpulse = mState + (halfNumCoefs-1)*inChannelCount;
+	mRingFull = mImpulse + (numCoefs+1)*inChannelCount;
+}
+
+AudioResamplerSinc::~AudioResamplerSinc()
+{
+	delete [] mState;
+}
+
+void AudioResamplerSinc::init() {
+}
+
+void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider)
+{
+	mFirCoefs = (mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown;
+
+	// select the appropriate resampler
+    switch (mChannelCount) {
+    case 1:
+        resample<1>(out, outFrameCount, provider);
+        break;
+    case 2:
+        resample<2>(out, outFrameCount, provider);
+        break;
+    }
+}
+
+
+template<int CHANNELS>
+void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+        AudioBufferProvider* provider)
+{
+    int16_t* impulse = mImpulse;
+    uint32_t vRL = mVolumeRL;
+    size_t inputIndex = mInputIndex;
+    uint32_t phaseFraction = mPhaseFraction;
+    uint32_t phaseIncrement = mPhaseIncrement;
+    size_t outputIndex = 0;
+    size_t outputSampleCount = outFrameCount * 2;
+
+    AudioBufferProvider::Buffer& buffer(mBuffer);
+    while (outputIndex < outputSampleCount) {
+        // buffer is empty, fetch a new one
+        if (buffer.raw == NULL) {
+            provider->getNextBuffer(&buffer);
+            if (buffer.raw == NULL)
+                break;
+    		const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
+        	if (phaseIndex) {
+        		read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+            }
+        }
+        int16_t *in = buffer.i16;
+    	const size_t frameCount = buffer.frameCount;
+
+    	// Always read-in the first samples from the input buffer
+    	int16_t* head = impulse + halfNumCoefs*CHANNELS;
+		head[0] = in[inputIndex*CHANNELS + 0];
+		if (CHANNELS == 2)
+			head[1] = in[inputIndex*CHANNELS + 1];
+
+        // handle boundary case
+    	int32_t l, r;
+        while (outputIndex < outputSampleCount) {
+        	filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse);
+    		out[outputIndex++] = mulRL(1, l, vRL);
+    		out[outputIndex++] = mulRL(0, r, vRL);
+
+        	phaseFraction += phaseIncrement;
+    		const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
+        	if (phaseIndex) {
+        		inputIndex += phaseIndex;
+        		if (inputIndex >= frameCount)
+        			break;
+        		read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
+        	}
+        }
+
+        // if done with buffer, save samples
+        if (inputIndex >= frameCount) {
+            inputIndex -= frameCount;
+            provider->releaseBuffer(&buffer);
+        }
+    }
+
+    mImpulse = impulse;
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+}
+
+template<int CHANNELS>
+void AudioResamplerSinc::read(
+		int16_t*& impulse, uint32_t& phaseFraction,
+		int16_t const* in, size_t inputIndex)
+{
+	// read new samples into the ring buffer
+	while (phaseFraction >> kNumPhaseBits) {
+		const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
+		impulse += CHANNELS;
+		phaseFraction -= 1LU<<kNumPhaseBits;
+		if (impulse >= mRingFull) {
+			const size_t stateSize = (halfNumCoefs*2)*CHANNELS;
+			memcpy(mState, mState+stateSize, sizeof(int16_t)*stateSize);
+			impulse -= stateSize;
+		}
+		int16_t* head = impulse + halfNumCoefs*CHANNELS;
+		head[0] = in[inputIndex*CHANNELS + 0];
+		if (CHANNELS == 2)
+			head[1] = in[inputIndex*CHANNELS + 1];
+	}
+}
+
+template<int CHANNELS>
+void AudioResamplerSinc::filterCoefficient(
+		int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples)
+{	
+	// compute the index of the coefficient on the positive side and
+	// negative side
+	uint32_t indexP = (phase & cMask) >> cShift;
+	uint16_t lerpP  = (phase & pMask) >> pShift;
+	uint32_t indexN = (-phase & cMask) >> cShift;
+	uint16_t lerpN  = (-phase & pMask) >> pShift;
+	
+	l = 0;
+	r = 0;
+	int32_t const* coefs = mFirCoefs;
+	int16_t const *sP = samples;
+	int16_t const *sN = samples+CHANNELS;
+	for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) {
+		interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
+		interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
+		sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits;
+        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
+        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
+		sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits;
+        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
+        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
+		sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits;
+        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
+        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
+		sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits;
+	}
+}
+
+template<int CHANNELS>
+void AudioResamplerSinc::interpolate(
+        int32_t& l, int32_t& r,
+		int32_t const* coefs, int16_t lerp, int16_t const* samples)
+{
+	int32_t c0 = coefs[0];
+	int32_t c1 = coefs[1];
+	int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0);
+	if (CHANNELS == 2) {
+		uint32_t rl = *reinterpret_cast<uint32_t const*>(samples);
+		l = mulAddRL(1, rl, sinc, l);
+		r = mulAddRL(0, rl, sinc, r);
+	} else {
+		r = l = mulAdd(samples[0], sinc, l);
+	}
+}
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+