diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
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+++ b/services/audioflinger/AudioResampler.h
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+/*
+ * 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_H
+#define ANDROID_AUDIO_RESAMPLER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include "AudioBufferProvider.h"
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+class AudioResampler {
+public:
+    // Determines quality of SRC.
+    //  LOW_QUALITY: linear interpolator (1st order)
+    //  MED_QUALITY: cubic interpolator (3rd order)
+    //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
+    // NOTE: high quality SRC will only be supported for
+    // certain fixed rate conversions. Sample rate cannot be
+    // changed dynamically. 
+    enum src_quality {
+        DEFAULT=0,
+        LOW_QUALITY=1,
+        MED_QUALITY=2,
+        HIGH_QUALITY=3
+    };
+
+    static AudioResampler* create(int bitDepth, int inChannelCount,
+            int32_t sampleRate, int quality=DEFAULT);
+
+    virtual ~AudioResampler();
+
+    virtual void init() = 0;
+    virtual void setSampleRate(int32_t inSampleRate);
+    virtual void setVolume(int16_t left, int16_t right);
+
+    virtual void resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider) = 0;
+
+protected:
+    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
+    static const int kNumPhaseBits = 30;
+
+    // phase mask for fraction
+    static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
+
+    // multiplier to calculate fixed point phase increment
+    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
+
+    enum format {MONO_16_BIT, STEREO_16_BIT};
+    AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
+
+    // prevent copying
+    AudioResampler(const AudioResampler&);
+    AudioResampler& operator=(const AudioResampler&);
+
+    int32_t mBitDepth;
+    int32_t mChannelCount;
+    int32_t mSampleRate;
+    int32_t mInSampleRate;
+    AudioBufferProvider::Buffer mBuffer;
+    union {
+        int16_t mVolume[2];
+        uint32_t mVolumeRL;
+    };
+    int16_t mTargetVolume[2];
+    format mFormat;
+    size_t mInputIndex;
+    int32_t mPhaseIncrement;
+    uint32_t mPhaseFraction;
+};
+
+// ----------------------------------------------------------------------------
+}
+; // namespace android
+
+#endif // ANDROID_AUDIO_RESAMPLER_H