diff --git a/media/libaudioprocessing/BufferProviders.cpp b/media/libaudioprocessing/BufferProviders.cpp
new file mode 100644
index 0000000..11ec367
--- /dev/null
+++ b/media/libaudioprocessing/BufferProviders.cpp
@@ -0,0 +1,590 @@
+/*
+ * Copyright (C) 2015 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 "BufferProvider"
+//#define LOG_NDEBUG 0
+
+#include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
+#include <external/sonic/sonic.h>
+#include <media/audiohal/EffectBufferHalInterface.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/BufferProviders.h>
+#include <system/audio_effects/effect_downmix.h>
+#include <utils/Log.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
+
+CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
+        size_t outputFrameSize, size_t bufferFrameCount) :
+        mInputFrameSize(inputFrameSize),
+        mOutputFrameSize(outputFrameSize),
+        mLocalBufferFrameCount(bufferFrameCount),
+        mLocalBufferData(NULL),
+        mConsumed(0)
+{
+    ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
+            inputFrameSize, outputFrameSize, bufferFrameCount);
+    LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
+            "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
+            inputFrameSize, outputFrameSize);
+    if (mLocalBufferFrameCount) {
+        (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
+    }
+    mBuffer.frameCount = 0;
+}
+
+CopyBufferProvider::~CopyBufferProvider()
+{
+    ALOGV("~CopyBufferProvider(%p)", this);
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    free(mLocalBufferData);
+}
+
+status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer)
+{
+    //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu))",
+    //        this, pBuffer, pBuffer->frameCount);
+    if (mLocalBufferFrameCount == 0) {
+        status_t res = mTrackBufferProvider->getNextBuffer(pBuffer);
+        if (res == OK) {
+            copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
+        }
+        return res;
+    }
+    if (mBuffer.frameCount == 0) {
+        mBuffer.frameCount = pBuffer->frameCount;
+        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
+        // At one time an upstream buffer provider had
+        // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
+        //
+        // By API spec, if res != OK, then mBuffer.frameCount == 0.
+        // but there may be improper implementations.
+        ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
+        if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
+            pBuffer->raw = NULL;
+            pBuffer->frameCount = 0;
+            return res;
+        }
+        mConsumed = 0;
+    }
+    ALOG_ASSERT(mConsumed < mBuffer.frameCount);
+    size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
+    count = min(count, pBuffer->frameCount);
+    pBuffer->raw = mLocalBufferData;
+    pBuffer->frameCount = count;
+    copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
+            pBuffer->frameCount);
+    return OK;
+}
+
+void CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
+{
+    //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
+    //        this, pBuffer, pBuffer->frameCount);
+    if (mLocalBufferFrameCount == 0) {
+        mTrackBufferProvider->releaseBuffer(pBuffer);
+        return;
+    }
+    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+    mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
+    if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+        ALOG_ASSERT(mBuffer.frameCount == 0);
+    }
+    pBuffer->raw = NULL;
+    pBuffer->frameCount = 0;
+}
+
+void CopyBufferProvider::reset()
+{
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    mConsumed = 0;
+}
+
+DownmixerBufferProvider::DownmixerBufferProvider(
+        audio_channel_mask_t inputChannelMask,
+        audio_channel_mask_t outputChannelMask, audio_format_t format,
+        uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
+        CopyBufferProvider(
+            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
+            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
+            bufferFrameCount)  // set bufferFrameCount to 0 to do in-place
+{
+    ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)",
+            this, inputChannelMask, outputChannelMask, format,
+            sampleRate, sessionId);
+    if (!sIsMultichannelCapable) {
+        ALOGE("DownmixerBufferProvider() error: not multichannel capable");
+        return;
+    }
+    mEffectsFactory = EffectsFactoryHalInterface::create();
+    if (mEffectsFactory == 0) {
+        ALOGE("DownmixerBufferProvider() error: could not obtain the effects factory");
+        return;
+    }
+    if (mEffectsFactory->createEffect(&sDwnmFxDesc.uuid,
+                                      sessionId,
+                                      SESSION_ID_INVALID_AND_IGNORED,
+                                      &mDownmixInterface) != 0) {
+         ALOGE("DownmixerBufferProvider() error creating downmixer effect");
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+     // channel input configuration will be overridden per-track
+     mDownmixConfig.inputCfg.channels = inputChannelMask;   // FIXME: Should be bits
+     mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
+     mDownmixConfig.inputCfg.format = format;
+     mDownmixConfig.outputCfg.format = format;
+     mDownmixConfig.inputCfg.samplingRate = sampleRate;
+     mDownmixConfig.outputCfg.samplingRate = sampleRate;
+     mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+     mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
+     // input and output buffer provider, and frame count will not be used as the downmix effect
+     // process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
+     mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
+             EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
+     mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
+
+     status_t status;
+     status = EffectBufferHalInterface::mirror(
+             nullptr,
+             audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
+             &mInBuffer);
+     if (status != 0) {
+         ALOGE("DownmixerBufferProvider() error %d while creating input buffer", status);
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+     status = EffectBufferHalInterface::mirror(
+             nullptr,
+             audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
+             &mOutBuffer);
+     if (status != 0) {
+         ALOGE("DownmixerBufferProvider() error %d while creating output buffer", status);
+         mInBuffer.clear();
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+
+     int cmdStatus;
+     uint32_t replySize = sizeof(int);
+
+     // Configure downmixer
+     status = mDownmixInterface->command(
+             EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
+             &mDownmixConfig /*pCmdData*/,
+             &replySize, &cmdStatus /*pReplyData*/);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
+                 status, cmdStatus);
+         mOutBuffer.clear();
+         mInBuffer.clear();
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+
+     // Enable downmixer
+     replySize = sizeof(int);
+     status = mDownmixInterface->command(
+             EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
+             &replySize, &cmdStatus /*pReplyData*/);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
+                 status, cmdStatus);
+         mOutBuffer.clear();
+         mInBuffer.clear();
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+
+     // Set downmix type
+     // parameter size rounded for padding on 32bit boundary
+     const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
+     const int downmixParamSize =
+             sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
+     effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
+     param->psize = sizeof(downmix_params_t);
+     const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
+     memcpy(param->data, &downmixParam, param->psize);
+     const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
+     param->vsize = sizeof(downmix_type_t);
+     memcpy(param->data + psizePadded, &downmixType, param->vsize);
+     replySize = sizeof(int);
+     status = mDownmixInterface->command(
+             EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
+             param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
+     free(param);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
+                 status, cmdStatus);
+         mOutBuffer.clear();
+         mInBuffer.clear();
+         mDownmixInterface.clear();
+         mEffectsFactory.clear();
+         return;
+     }
+     ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
+}
+
+DownmixerBufferProvider::~DownmixerBufferProvider()
+{
+    ALOGV("~DownmixerBufferProvider (%p)", this);
+    if (mDownmixInterface != 0) {
+        mDownmixInterface->close();
+    }
+}
+
+void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    mInBuffer->setExternalData(const_cast<void*>(src));
+    mInBuffer->setFrameCount(frames);
+    mInBuffer->update();
+    mOutBuffer->setExternalData(dst);
+    mOutBuffer->setFrameCount(frames);
+    mOutBuffer->update();
+    // may be in-place if src == dst.
+    status_t res = mDownmixInterface->process();
+    if (res == OK) {
+        mOutBuffer->commit();
+    } else {
+        ALOGE("DownmixBufferProvider error %d", res);
+    }
+}
+
+/* call once in a pthread_once handler. */
+/*static*/ status_t DownmixerBufferProvider::init()
+{
+    // find multichannel downmix effect if we have to play multichannel content
+    sp<EffectsFactoryHalInterface> effectsFactory = EffectsFactoryHalInterface::create();
+    if (effectsFactory == 0) {
+        ALOGE("AudioMixer() error: could not obtain the effects factory");
+        return NO_INIT;
+    }
+    uint32_t numEffects = 0;
+    int ret = effectsFactory->queryNumberEffects(&numEffects);
+    if (ret != 0) {
+        ALOGE("AudioMixer() error %d querying number of effects", ret);
+        return NO_INIT;
+    }
+    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
+
+    for (uint32_t i = 0 ; i < numEffects ; i++) {
+        if (effectsFactory->getDescriptor(i, &sDwnmFxDesc) == 0) {
+            ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
+            if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
+                ALOGI("found effect \"%s\" from %s",
+                        sDwnmFxDesc.name, sDwnmFxDesc.implementor);
+                sIsMultichannelCapable = true;
+                break;
+            }
+        }
+    }
+    ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
+    return NO_INIT;
+}
+
+/*static*/ bool DownmixerBufferProvider::sIsMultichannelCapable = false;
+/*static*/ effect_descriptor_t DownmixerBufferProvider::sDwnmFxDesc;
+
+RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
+        audio_channel_mask_t outputChannelMask, audio_format_t format,
+        size_t bufferFrameCount) :
+        CopyBufferProvider(
+                audio_bytes_per_sample(format)
+                    * audio_channel_count_from_out_mask(inputChannelMask),
+                audio_bytes_per_sample(format)
+                    * audio_channel_count_from_out_mask(outputChannelMask),
+                bufferFrameCount),
+        mFormat(format),
+        mSampleSize(audio_bytes_per_sample(format)),
+        mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
+        mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
+{
+    ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
+            this, format, inputChannelMask, outputChannelMask,
+            mInputChannels, mOutputChannels);
+    (void) memcpy_by_index_array_initialization_from_channel_mask(
+            mIdxAry, ARRAY_SIZE(mIdxAry), outputChannelMask, inputChannelMask);
+}
+
+void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    memcpy_by_index_array(dst, mOutputChannels,
+            src, mInputChannels, mIdxAry, mSampleSize, frames);
+}
+
+ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount,
+        audio_format_t inputFormat, audio_format_t outputFormat,
+        size_t bufferFrameCount) :
+        CopyBufferProvider(
+                channelCount * audio_bytes_per_sample(inputFormat),
+                channelCount * audio_bytes_per_sample(outputFormat),
+                bufferFrameCount),
+        mChannelCount(channelCount),
+        mInputFormat(inputFormat),
+        mOutputFormat(outputFormat)
+{
+    ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)",
+            this, channelCount, inputFormat, outputFormat);
+}
+
+void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
+}
+
+TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
+        audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
+        mChannelCount(channelCount),
+        mFormat(format),
+        mSampleRate(sampleRate),
+        mFrameSize(channelCount * audio_bytes_per_sample(format)),
+        mLocalBufferFrameCount(0),
+        mLocalBufferData(NULL),
+        mRemaining(0),
+        mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
+        mFallbackFailErrorShown(false),
+        mAudioPlaybackRateValid(false)
+{
+    LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
+            "TimestretchBufferProvider can't allocate Sonic stream");
+
+    setPlaybackRate(playbackRate);
+    ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f %d %d)",
+            this, channelCount, format, sampleRate, playbackRate.mSpeed,
+            playbackRate.mPitch, playbackRate.mStretchMode, playbackRate.mFallbackMode);
+    mBuffer.frameCount = 0;
+}
+
+TimestretchBufferProvider::~TimestretchBufferProvider()
+{
+    ALOGV("~TimestretchBufferProvider(%p)", this);
+    sonicDestroyStream(mSonicStream);
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    free(mLocalBufferData);
+}
+
+status_t TimestretchBufferProvider::getNextBuffer(
+        AudioBufferProvider::Buffer *pBuffer)
+{
+    ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu))",
+            this, pBuffer, pBuffer->frameCount);
+
+    // BYPASS
+    //return mTrackBufferProvider->getNextBuffer(pBuffer);
+
+    // check if previously processed data is sufficient.
+    if (pBuffer->frameCount <= mRemaining) {
+        ALOGV("previous sufficient");
+        pBuffer->raw = mLocalBufferData;
+        return OK;
+    }
+
+    // do we need to resize our buffer?
+    if (pBuffer->frameCount > mLocalBufferFrameCount) {
+        void *newmem;
+        if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) {
+            if (mRemaining != 0) {
+                memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize);
+            }
+            free(mLocalBufferData);
+            mLocalBufferData = newmem;
+            mLocalBufferFrameCount = pBuffer->frameCount;
+        }
+    }
+
+    // need to fetch more data
+    const size_t outputDesired = pBuffer->frameCount - mRemaining;
+    size_t dstAvailable;
+    do {
+        mBuffer.frameCount = mPlaybackRate.mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
+                ? outputDesired : outputDesired * mPlaybackRate.mSpeed + 1;
+
+        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
+
+        ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
+        if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
+            ALOGV("upstream provider cannot provide data");
+            if (mRemaining == 0) {
+                pBuffer->raw = NULL;
+                pBuffer->frameCount = 0;
+                return res;
+            } else { // return partial count
+                pBuffer->raw = mLocalBufferData;
+                pBuffer->frameCount = mRemaining;
+                return OK;
+            }
+        }
+
+        // time-stretch the data
+        dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired);
+        size_t srcAvailable = mBuffer.frameCount;
+        processFrames((uint8_t*)mLocalBufferData + mRemaining * mFrameSize, &dstAvailable,
+                mBuffer.raw, &srcAvailable);
+
+        // release all data consumed
+        mBuffer.frameCount = srcAvailable;
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    } while (dstAvailable == 0); // try until we get output data or upstream provider fails.
+
+    // update buffer vars with the actual data processed and return with buffer
+    mRemaining += dstAvailable;
+
+    pBuffer->raw = mLocalBufferData;
+    pBuffer->frameCount = mRemaining;
+
+    return OK;
+}
+
+void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
+{
+    ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))",
+       this, pBuffer, pBuffer->frameCount);
+
+    // BYPASS
+    //return mTrackBufferProvider->releaseBuffer(pBuffer);
+
+    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+    if (pBuffer->frameCount < mRemaining) {
+        memcpy(mLocalBufferData,
+                (uint8_t*)mLocalBufferData + pBuffer->frameCount * mFrameSize,
+                (mRemaining - pBuffer->frameCount) * mFrameSize);
+        mRemaining -= pBuffer->frameCount;
+    } else if (pBuffer->frameCount == mRemaining) {
+        mRemaining = 0;
+    } else {
+        LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)",
+                pBuffer->frameCount, mRemaining);
+    }
+
+    pBuffer->raw = NULL;
+    pBuffer->frameCount = 0;
+}
+
+void TimestretchBufferProvider::reset()
+{
+    mRemaining = 0;
+}
+
+status_t TimestretchBufferProvider::setPlaybackRate(const AudioPlaybackRate &playbackRate)
+{
+    mPlaybackRate = playbackRate;
+    mFallbackFailErrorShown = false;
+    sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
+    //TODO: pitch is ignored for now
+    //TODO: optimize: if parameters are the same, don't do any extra computation.
+
+    mAudioPlaybackRateValid = isAudioPlaybackRateValid(mPlaybackRate);
+    return OK;
+}
+
+void TimestretchBufferProvider::processFrames(void *dstBuffer, size_t *dstFrames,
+        const void *srcBuffer, size_t *srcFrames)
+{
+    ALOGV("processFrames(%zu %zu)  remaining(%zu)", *dstFrames, *srcFrames, mRemaining);
+    // Note dstFrames is the required number of frames.
+
+    if (!mAudioPlaybackRateValid) {
+        //fallback mode
+        // Ensure consumption from src is as expected.
+        // TODO: add logic to track "very accurate" consumption related to speed, original sampling
+        // rate, actual frames processed.
+
+        const size_t targetSrc = *dstFrames * mPlaybackRate.mSpeed;
+        if (*srcFrames < targetSrc) { // limit dst frames to that possible
+            *dstFrames = *srcFrames / mPlaybackRate.mSpeed;
+        } else if (*srcFrames > targetSrc + 1) {
+            *srcFrames = targetSrc + 1;
+        }
+        if (*dstFrames > 0) {
+            switch(mPlaybackRate.mFallbackMode) {
+            case AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT:
+                if (*dstFrames <= *srcFrames) {
+                      size_t copySize = mFrameSize * *dstFrames;
+                      memcpy(dstBuffer, srcBuffer, copySize);
+                  } else {
+                      // cyclically repeat the source.
+                      for (size_t count = 0; count < *dstFrames; count += *srcFrames) {
+                          size_t remaining = min(*srcFrames, *dstFrames - count);
+                          memcpy((uint8_t*)dstBuffer + mFrameSize * count,
+                                  srcBuffer, mFrameSize * remaining);
+                      }
+                  }
+                break;
+            case AUDIO_TIMESTRETCH_FALLBACK_DEFAULT:
+            case AUDIO_TIMESTRETCH_FALLBACK_MUTE:
+                memset(dstBuffer,0, mFrameSize * *dstFrames);
+                break;
+            case AUDIO_TIMESTRETCH_FALLBACK_FAIL:
+            default:
+                if(!mFallbackFailErrorShown) {
+                    ALOGE("invalid parameters in TimestretchBufferProvider fallbackMode:%d",
+                            mPlaybackRate.mFallbackMode);
+                    mFallbackFailErrorShown = true;
+                }
+                break;
+            }
+        }
+    } else {
+        switch (mFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) {
+                ALOGE("sonicWriteFloatToStream cannot realloc");
+                *srcFrames = 0; // cannot consume all of srcBuffer
+            }
+            *dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) {
+                ALOGE("sonicWriteShortToStream cannot realloc");
+                *srcFrames = 0; // cannot consume all of srcBuffer
+            }
+            *dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames);
+            break;
+        default:
+            // could also be caught on construction
+            LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat);
+        }
+    }
+}
+// ----------------------------------------------------------------------------
+} // namespace android