diff --git a/libs/audioflinger/AudioResamplerCubic.cpp b/libs/audioflinger/AudioResamplerCubic.cpp
new file mode 100644
index 0000000..4f437bf
--- /dev/null
+++ b/libs/audioflinger/AudioResamplerCubic.cpp
@@ -0,0 +1,178 @@
+/*
+ * 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 <stdint.h>
+#include <string.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+#include "AudioResamplerCubic.h"
+
+#define LOG_TAG "AudioSRC"
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+void AudioResamplerCubic::init() {
+    memset(&left, 0, sizeof(state));
+    memset(&right, 0, sizeof(state));
+}
+
+void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
+        AudioBufferProvider* provider) {
+
+    // should never happen, but we overflow if it does
+    // LOG_ASSERT(outFrameCount < 32767);
+
+    // select the appropriate resampler
+    switch (mChannelCount) {
+    case 1:
+        resampleMono16(out, outFrameCount, provider);
+        break;
+    case 2:
+        resampleStereo16(out, outFrameCount, provider);
+        break;
+    }
+}
+
+void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
+        AudioBufferProvider* provider) {
+
+    int32_t vl = mVolume[0];
+    int32_t vr = mVolume[1];
+
+    size_t inputIndex = mInputIndex;
+    uint32_t phaseFraction = mPhaseFraction;
+    uint32_t phaseIncrement = mPhaseIncrement;
+    size_t outputIndex = 0;
+    size_t outputSampleCount = outFrameCount * 2;
+    
+    // fetch first buffer
+    if (mBuffer.raw == NULL) {
+        provider->getNextBuffer(&mBuffer);
+        if (mBuffer.raw == NULL)
+            return;
+        // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
+    }
+    int16_t *in = mBuffer.i16;
+
+    while (outputIndex < outputSampleCount) {
+        int32_t sample;
+        int32_t x;
+
+        // calculate output sample
+        x = phaseFraction >> kPreInterpShift;
+        out[outputIndex++] += vl * interp(&left, x);
+        out[outputIndex++] += vr * interp(&right, x);
+        // out[outputIndex++] += vr * in[inputIndex*2];
+        
+        // increment phase
+        phaseFraction += phaseIncrement;
+        uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
+        phaseFraction &= kPhaseMask;
+
+        // time to fetch another sample
+        while (indexIncrement--) {
+
+            inputIndex++;
+            if (inputIndex == mBuffer.frameCount) {
+                inputIndex = 0;
+                provider->releaseBuffer(&mBuffer);
+                provider->getNextBuffer(&mBuffer);
+                if (mBuffer.raw == NULL)
+                    goto save_state;  // ugly, but efficient
+                in = mBuffer.i16;
+                // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+            }
+
+            // advance sample state
+            advance(&left, in[inputIndex*2]);
+            advance(&right, in[inputIndex*2+1]);
+        }
+    }
+
+save_state:
+    // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+}
+
+void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
+        AudioBufferProvider* provider) {
+
+    int32_t vl = mVolume[0];
+    int32_t vr = mVolume[1];
+
+    size_t inputIndex = mInputIndex;
+    uint32_t phaseFraction = mPhaseFraction;
+    uint32_t phaseIncrement = mPhaseIncrement;
+    size_t outputIndex = 0;
+    size_t outputSampleCount = outFrameCount * 2;
+    
+    // fetch first buffer
+    if (mBuffer.raw == NULL) {
+        provider->getNextBuffer(&mBuffer);
+        if (mBuffer.raw == NULL)
+            return;
+        // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+    }
+    int16_t *in = mBuffer.i16;
+
+    while (outputIndex < outputSampleCount) {
+        int32_t sample;
+        int32_t x;
+
+        // calculate output sample
+        x = phaseFraction >> kPreInterpShift;
+        sample = interp(&left, x);
+        out[outputIndex++] += vl * sample;
+        out[outputIndex++] += vr * sample;
+        
+        // increment phase
+        phaseFraction += phaseIncrement;
+        uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
+        phaseFraction &= kPhaseMask;
+
+        // time to fetch another sample
+        while (indexIncrement--) {
+
+            inputIndex++;
+            if (inputIndex == mBuffer.frameCount) {
+                inputIndex = 0;
+                provider->releaseBuffer(&mBuffer);
+                provider->getNextBuffer(&mBuffer);
+                if (mBuffer.raw == NULL)
+                    goto save_state;  // ugly, but efficient
+                // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
+                in = mBuffer.i16;
+            }
+
+            // advance sample state
+            advance(&left, in[inputIndex]);
+        }
+    }
+
+save_state:
+    // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+}
+
+// ----------------------------------------------------------------------------
+}
+; // namespace android
+