diff --git a/media/libaudioprocessing/AudioResampler.cpp b/media/libaudioprocessing/AudioResampler.cpp
new file mode 100644
index 0000000..c761b38
--- /dev/null
+++ b/media/libaudioprocessing/AudioResampler.cpp
@@ -0,0 +1,787 @@
+/*
+ * 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.
+ */
+
+#define LOG_TAG "AudioResampler"
+//#define LOG_NDEBUG 0
+
+#include <pthread.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/types.h>
+
+#include <cutils/properties.h>
+#include <log/log.h>
+
+#include <audio_utils/primitives.h>
+#include <media/AudioResampler.h>
+#include "AudioResamplerSinc.h"
+#include "AudioResamplerCubic.h"
+#include "AudioResamplerDyn.h"
+
+#ifdef __arm__
+    // bug 13102576
+    //#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioResamplerOrder1 : public AudioResampler {
+public:
+    AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
+        AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
+    }
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+private:
+    // number of bits used in interpolation multiply - 15 bits avoids overflow
+    static const int kNumInterpBits = 15;
+
+    // bits to shift the phase fraction down to avoid overflow
+    static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
+
+    void init() {}
+    size_t resampleMono16(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+    size_t resampleStereo16(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+    void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement);
+    void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement);
+#endif  // ASM_ARM_RESAMP1
+
+    static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
+        return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
+    }
+    static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
+        *frac += inc;
+        *index += (size_t)(*frac >> kNumPhaseBits);
+        *frac &= kPhaseMask;
+    }
+    int mX0L;
+    int mX0R;
+};
+
+/*static*/
+const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
+
+bool AudioResampler::qualityIsSupported(src_quality quality)
+{
+    switch (quality) {
+    case DEFAULT_QUALITY:
+    case LOW_QUALITY:
+    case MED_QUALITY:
+    case HIGH_QUALITY:
+    case VERY_HIGH_QUALITY:
+    case DYN_LOW_QUALITY:
+    case DYN_MED_QUALITY:
+    case DYN_HIGH_QUALITY:
+        return true;
+    default:
+        return false;
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+static pthread_once_t once_control = PTHREAD_ONCE_INIT;
+static AudioResampler::src_quality defaultQuality = AudioResampler::DEFAULT_QUALITY;
+
+void AudioResampler::init_routine()
+{
+    char value[PROPERTY_VALUE_MAX];
+    if (property_get("af.resampler.quality", value, NULL) > 0) {
+        char *endptr;
+        unsigned long l = strtoul(value, &endptr, 0);
+        if (*endptr == '\0') {
+            defaultQuality = (src_quality) l;
+            ALOGD("forcing AudioResampler quality to %d", defaultQuality);
+            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
+                defaultQuality = DEFAULT_QUALITY;
+            }
+        }
+    }
+}
+
+uint32_t AudioResampler::qualityMHz(src_quality quality)
+{
+    switch (quality) {
+    default:
+    case DEFAULT_QUALITY:
+    case LOW_QUALITY:
+        return 3;
+    case MED_QUALITY:
+        return 6;
+    case HIGH_QUALITY:
+        return 20;
+    case VERY_HIGH_QUALITY:
+        return 34;
+    case DYN_LOW_QUALITY:
+        return 4;
+    case DYN_MED_QUALITY:
+        return 6;
+    case DYN_HIGH_QUALITY:
+        return 12;
+    }
+}
+
+static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, should be tunable
+static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
+static uint32_t currentMHz = 0;
+
+AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount,
+        int32_t sampleRate, src_quality quality) {
+
+    bool atFinalQuality;
+    if (quality == DEFAULT_QUALITY) {
+        // read the resampler default quality property the first time it is needed
+        int ok = pthread_once(&once_control, init_routine);
+        if (ok != 0) {
+            ALOGE("%s pthread_once failed: %d", __func__, ok);
+        }
+        quality = defaultQuality;
+        atFinalQuality = false;
+    } else {
+        atFinalQuality = true;
+    }
+
+    /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+     * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+     * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+     * due to estimated CPU load of having too many active resamplers
+     * (the code below the if).
+     */
+    if (quality == DEFAULT_QUALITY) {
+        quality = DYN_MED_QUALITY;
+    }
+
+    // naive implementation of CPU load throttling doesn't account for whether resampler is active
+    pthread_mutex_lock(&mutex);
+    for (;;) {
+        uint32_t deltaMHz = qualityMHz(quality);
+        uint32_t newMHz = currentMHz + deltaMHz;
+        if ((qualityIsSupported(quality) && newMHz <= maxMHz) || atFinalQuality) {
+            ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
+                    currentMHz, newMHz, deltaMHz, quality);
+            currentMHz = newMHz;
+            break;
+        }
+        // not enough CPU available for proposed quality level, so try next lowest level
+        switch (quality) {
+        default:
+        case LOW_QUALITY:
+            atFinalQuality = true;
+            break;
+        case MED_QUALITY:
+            quality = LOW_QUALITY;
+            break;
+        case HIGH_QUALITY:
+            quality = MED_QUALITY;
+            break;
+        case VERY_HIGH_QUALITY:
+            quality = HIGH_QUALITY;
+            break;
+        case DYN_LOW_QUALITY:
+            atFinalQuality = true;
+            break;
+        case DYN_MED_QUALITY:
+            quality = DYN_LOW_QUALITY;
+            break;
+        case DYN_HIGH_QUALITY:
+            quality = DYN_MED_QUALITY;
+            break;
+        }
+    }
+    pthread_mutex_unlock(&mutex);
+
+    AudioResampler* resampler;
+
+    switch (quality) {
+    default:
+    case LOW_QUALITY:
+        ALOGV("Create linear Resampler");
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
+        break;
+    case MED_QUALITY:
+        ALOGV("Create cubic Resampler");
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerCubic(inChannelCount, sampleRate);
+        break;
+    case HIGH_QUALITY:
+        ALOGV("Create HIGH_QUALITY sinc Resampler");
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerSinc(inChannelCount, sampleRate);
+        break;
+    case VERY_HIGH_QUALITY:
+        ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);
+        break;
+    case DYN_LOW_QUALITY:
+    case DYN_MED_QUALITY:
+    case DYN_HIGH_QUALITY:
+        ALOGV("Create dynamic Resampler = %d", quality);
+        if (format == AUDIO_FORMAT_PCM_FLOAT) {
+            resampler = new AudioResamplerDyn<float, float, float>(inChannelCount,
+                    sampleRate, quality);
+        } else {
+            LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+            if (quality == DYN_HIGH_QUALITY) {
+                resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount,
+                        sampleRate, quality);
+            } else {
+                resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount,
+                        sampleRate, quality);
+            }
+        }
+        break;
+    }
+
+    // initialize resampler
+    resampler->init();
+    return resampler;
+}
+
+AudioResampler::AudioResampler(int inChannelCount,
+        int32_t sampleRate, src_quality quality) :
+        mChannelCount(inChannelCount),
+        mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
+        mPhaseFraction(0),
+        mQuality(quality) {
+
+    const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8;
+    if (inChannelCount < 1
+            || inChannelCount > maxChannels) {
+        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
+                quality, inChannelCount);
+    }
+    if (sampleRate <= 0) {
+        LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
+    }
+
+    // initialize common members
+    mVolume[0] = mVolume[1] = 0;
+    mBuffer.frameCount = 0;
+}
+
+AudioResampler::~AudioResampler() {
+    pthread_mutex_lock(&mutex);
+    src_quality quality = getQuality();
+    uint32_t deltaMHz = qualityMHz(quality);
+    int32_t newMHz = currentMHz - deltaMHz;
+    ALOGV("resampler load %u -> %d MHz due to delta -%u MHz from quality %d",
+            currentMHz, newMHz, deltaMHz, quality);
+    LOG_ALWAYS_FATAL_IF(newMHz < 0, "negative resampler load %d MHz", newMHz);
+    currentMHz = newMHz;
+    pthread_mutex_unlock(&mutex);
+}
+
+void AudioResampler::setSampleRate(int32_t inSampleRate) {
+    mInSampleRate = inSampleRate;
+    mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
+}
+
+void AudioResampler::setVolume(float left, float right) {
+    // TODO: Implement anti-zipper filter
+    // convert to U4.12 for internal integer use (round down)
+    // integer volume values are clamped to 0 to UNITY_GAIN.
+    mVolume[0] = u4_12_from_float(clampFloatVol(left));
+    mVolume[1] = u4_12_from_float(clampFloatVol(right));
+}
+
+void AudioResampler::reset() {
+    mInputIndex = 0;
+    mPhaseFraction = 0;
+    mBuffer.frameCount = 0;
+}
+
+// ----------------------------------------------------------------------------
+
+size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
+        AudioBufferProvider* provider) {
+
+    // should never happen, but we overflow if it does
+    // ALOG_ASSERT(outFrameCount < 32767);
+
+    // select the appropriate resampler
+    switch (mChannelCount) {
+    case 1:
+        return resampleMono16(out, outFrameCount, provider);
+    case 2:
+        return resampleStereo16(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
+    }
+}
+
+size_t AudioResamplerOrder1::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;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
+
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
+    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
+
+    while (outputIndex < outputSampleCount) {
+
+        // buffer is empty, fetch a new one
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
+            provider->getNextBuffer(&mBuffer);
+            if (mBuffer.raw == NULL) {
+                goto resampleStereo16_exit;
+            }
+
+            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
+            if (mBuffer.frameCount > inputIndex) break;
+
+            inputIndex -= mBuffer.frameCount;
+            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
+            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
+            provider->releaseBuffer(&mBuffer);
+            // mBuffer.frameCount == 0 now so we reload a new buffer
+        }
+
+        int16_t *in = mBuffer.i16;
+
+        // handle boundary case
+        while (inputIndex == 0) {
+            // ALOGE("boundary case");
+            out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
+            out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
+            Advance(&inputIndex, &phaseFraction, phaseIncrement);
+            if (outputIndex == outputSampleCount) {
+                break;
+            }
+        }
+
+        // process input samples
+        // ALOGE("general case");
+
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+        if (inputIndex + 2 < mBuffer.frameCount) {
+            int32_t* maxOutPt;
+            int32_t maxInIdx;
+
+            maxOutPt = out + (outputSampleCount - 2);   // 2 because 2 frames per loop
+            maxInIdx = mBuffer.frameCount - 2;
+            AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+                    phaseFraction, phaseIncrement);
+        }
+#endif  // ASM_ARM_RESAMP1
+
+        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
+            out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
+                    in[inputIndex*2], phaseFraction);
+            out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
+                    in[inputIndex*2+1], phaseFraction);
+            Advance(&inputIndex, &phaseFraction, phaseIncrement);
+        }
+
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
+
+        // if done with buffer, save samples
+        if (inputIndex >= mBuffer.frameCount) {
+            inputIndex -= mBuffer.frameCount;
+
+            // ALOGE("buffer done, new input index %d", inputIndex);
+
+            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
+            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
+            provider->releaseBuffer(&mBuffer);
+
+            // verify that the releaseBuffer resets the buffer frameCount
+            // ALOG_ASSERT(mBuffer.frameCount == 0);
+        }
+    }
+
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
+
+resampleStereo16_exit:
+    // save state
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+    return outputIndex / 2 /* channels for stereo */;
+}
+
+size_t AudioResamplerOrder1::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;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
+
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
+    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
+    while (outputIndex < outputSampleCount) {
+        // buffer is empty, fetch a new one
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
+            provider->getNextBuffer(&mBuffer);
+            if (mBuffer.raw == NULL) {
+                mInputIndex = inputIndex;
+                mPhaseFraction = phaseFraction;
+                goto resampleMono16_exit;
+            }
+            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
+            if (mBuffer.frameCount >  inputIndex) break;
+
+            inputIndex -= mBuffer.frameCount;
+            mX0L = mBuffer.i16[mBuffer.frameCount-1];
+            provider->releaseBuffer(&mBuffer);
+            // mBuffer.frameCount == 0 now so we reload a new buffer
+        }
+        int16_t *in = mBuffer.i16;
+
+        // handle boundary case
+        while (inputIndex == 0) {
+            // ALOGE("boundary case");
+            int32_t sample = Interp(mX0L, in[0], phaseFraction);
+            out[outputIndex++] += vl * sample;
+            out[outputIndex++] += vr * sample;
+            Advance(&inputIndex, &phaseFraction, phaseIncrement);
+            if (outputIndex == outputSampleCount) {
+                break;
+            }
+        }
+
+        // process input samples
+        // ALOGE("general case");
+
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+        if (inputIndex + 2 < mBuffer.frameCount) {
+            int32_t* maxOutPt;
+            int32_t maxInIdx;
+
+            maxOutPt = out + (outputSampleCount - 2);
+            maxInIdx = (int32_t)mBuffer.frameCount - 2;
+                AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+                        phaseFraction, phaseIncrement);
+        }
+#endif  // ASM_ARM_RESAMP1
+
+        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
+            int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
+                    phaseFraction);
+            out[outputIndex++] += vl * sample;
+            out[outputIndex++] += vr * sample;
+            Advance(&inputIndex, &phaseFraction, phaseIncrement);
+        }
+
+
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
+
+        // if done with buffer, save samples
+        if (inputIndex >= mBuffer.frameCount) {
+            inputIndex -= mBuffer.frameCount;
+
+            // ALOGE("buffer done, new input index %d", inputIndex);
+
+            mX0L = mBuffer.i16[mBuffer.frameCount-1];
+            provider->releaseBuffer(&mBuffer);
+
+            // verify that the releaseBuffer resets the buffer frameCount
+            // ALOG_ASSERT(mBuffer.frameCount == 0);
+        }
+    }
+
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
+
+resampleMono16_exit:
+    // save state
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+    return outputIndex;
+}
+
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+
+/*******************************************************************
+*
+*   AsmMono16Loop
+*   asm optimized monotonic loop version; one loop is 2 frames
+*   Input:
+*       in : pointer on input samples
+*       maxOutPt : pointer on first not filled
+*       maxInIdx : index on first not used
+*       outputIndex : pointer on current output index
+*       out : pointer on output buffer
+*       inputIndex : pointer on current input index
+*       vl, vr : left and right gain
+*       phaseFraction : pointer on current phase fraction
+*       phaseIncrement
+*   Ouput:
+*       outputIndex :
+*       out : updated buffer
+*       inputIndex : index of next to use
+*       phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+__attribute__((noinline))
+void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
+#define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
+
+    asm(
+        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
+        // get parameters
+        "   ldr r6, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
+        "   ldr r6, [r6]\n"                         // phaseFraction
+        "   ldr r7, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
+        "   ldr r7, [r7]\n"                         // inputIndex
+        "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
+        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
+        "   ldr r0, [r0]\n"                         // outputIndex
+        "   add r8, r8, r0, asl #2\n"               // curOut
+        "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
+        "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
+        "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
+
+        // r0 pin, x0, Samp
+
+        // r1 in
+        // r2 maxOutPt
+        // r3 maxInIdx
+
+        // r4 x1, i1, i3, Out1
+        // r5 out0
+
+        // r6 frac
+        // r7 inputIndex
+        // r8 curOut
+
+        // r9 inc
+        // r10 vl
+        // r11 vr
+
+        // r12
+        // r13 sp
+        // r14
+
+        // the following loop works on 2 frames
+
+        "1:\n"
+        "   cmp r8, r2\n"                   // curOut - maxCurOut
+        "   bcs 2f\n"
+
+#define MO_ONE_FRAME \
+    "   add r0, r1, r7, asl #1\n"       /* in + inputIndex */\
+    "   ldrsh r4, [r0]\n"               /* in[inputIndex] */\
+    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
+    "   ldrsh r0, [r0, #-2]\n"          /* in[inputIndex-1] */\
+    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
+    "   sub r4, r4, r0\n"               /* in[inputIndex] - in[inputIndex-1] */\
+    "   mov r4, r4, lsl #2\n"           /* <<2 */\
+    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
+    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
+    "   add r0, r0, r4\n"               /* x0 - (..) */\
+    "   mla r5, r0, r10, r5\n"          /* vl*interp + out[] */\
+    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
+    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
+    "   mla r4, r0, r11, r4\n"          /* vr*interp + out[] */\
+    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */\
+    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */
+
+        MO_ONE_FRAME    // frame 1
+        MO_ONE_FRAME    // frame 2
+
+        "   cmp r7, r3\n"                   // inputIndex - maxInIdx
+        "   bcc 1b\n"
+        "2:\n"
+
+        "   bic r6, r6, #0xC0000000\n"             // phaseFraction & ...
+        // save modified values
+        "   ldr r0, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
+        "   str r6, [r0]\n"                         // phaseFraction
+        "   ldr r0, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
+        "   str r7, [r0]\n"                         // inputIndex
+        "   ldr r0, [sp, #" MO_PARAM5 " + 4]\n"     // out
+        "   sub r8, r0\n"                           // curOut - out
+        "   asr r8, #2\n"                           // new outputIndex
+        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
+        "   str r8, [r0]\n"                         // save outputIndex
+
+        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
+    );
+}
+
+/*******************************************************************
+*
+*   AsmStereo16Loop
+*   asm optimized stereo loop version; one loop is 2 frames
+*   Input:
+*       in : pointer on input samples
+*       maxOutPt : pointer on first not filled
+*       maxInIdx : index on first not used
+*       outputIndex : pointer on current output index
+*       out : pointer on output buffer
+*       inputIndex : pointer on current input index
+*       vl, vr : left and right gain
+*       phaseFraction : pointer on current phase fraction
+*       phaseIncrement
+*   Ouput:
+*       outputIndex :
+*       out : updated buffer
+*       inputIndex : index of next to use
+*       phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+__attribute__((noinline))
+void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
+#define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
+    asm(
+        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
+        // get parameters
+        "   ldr r6, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
+        "   ldr r6, [r6]\n"                         // phaseFraction
+        "   ldr r7, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
+        "   ldr r7, [r7]\n"                         // inputIndex
+        "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
+        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
+        "   ldr r0, [r0]\n"                         // outputIndex
+        "   add r8, r8, r0, asl #2\n"               // curOut
+        "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
+        "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
+        "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr
+
+        // r0 pin, x0, Samp
+
+        // r1 in
+        // r2 maxOutPt
+        // r3 maxInIdx
+
+        // r4 x1, i1, i3, out1
+        // r5 out0
+
+        // r6 frac
+        // r7 inputIndex
+        // r8 curOut
+
+        // r9 inc
+        // r10 vl
+        // r11 vr
+
+        // r12 temporary
+        // r13 sp
+        // r14
+
+        "3:\n"
+        "   cmp r8, r2\n"                   // curOut - maxCurOut
+        "   bcs 4f\n"
+
+#define ST_ONE_FRAME \
+    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
+\
+    "   add r0, r1, r7, asl #2\n"       /* in + 2*inputIndex */\
+\
+    "   ldrsh r4, [r0]\n"               /* in[2*inputIndex] */\
+    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
+    "   ldrsh r12, [r0, #-4]\n"         /* in[2*inputIndex-2] */\
+    "   sub r4, r4, r12\n"              /* in[2*InputIndex] - in[2*InputIndex-2] */\
+    "   mov r4, r4, lsl #2\n"           /* <<2 */\
+    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
+    "   add r12, r12, r4\n"             /* x0 - (..) */\
+    "   mla r5, r12, r10, r5\n"         /* vl*interp + out[] */\
+    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
+    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
+\
+    "   ldrsh r12, [r0, #+2]\n"         /* in[2*inputIndex+1] */\
+    "   ldrsh r0, [r0, #-2]\n"          /* in[2*inputIndex-1] */\
+    "   sub r12, r12, r0\n"             /* in[2*InputIndex] - in[2*InputIndex-2] */\
+    "   mov r12, r12, lsl #2\n"         /* <<2 */\
+    "   smulwt r12, r12, r6\n"          /* (x1-x0)*.. */\
+    "   add r12, r0, r12\n"             /* x0 - (..) */\
+    "   mla r4, r12, r11, r4\n"         /* vr*interp + out[] */\
+    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */\
+\
+    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
+    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */
+
+    ST_ONE_FRAME    // frame 1
+    ST_ONE_FRAME    // frame 1
+
+        "   cmp r7, r3\n"                       // inputIndex - maxInIdx
+        "   bcc 3b\n"
+        "4:\n"
+
+        "   bic r6, r6, #0xC0000000\n"              // phaseFraction & ...
+        // save modified values
+        "   ldr r0, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
+        "   str r6, [r0]\n"                         // phaseFraction
+        "   ldr r0, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
+        "   str r7, [r0]\n"                         // inputIndex
+        "   ldr r0, [sp, #" ST_PARAM5 " + 4]\n"     // out
+        "   sub r8, r0\n"                           // curOut - out
+        "   asr r8, #2\n"                           // new outputIndex
+        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
+        "   str r8, [r0]\n"                         // save outputIndex
+
+        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
+    );
+}
+
+#endif  // ASM_ARM_RESAMP1
+
+
+// ----------------------------------------------------------------------------
+
+} // namespace android