diff --git a/include/media/BufferProviders.h b/include/media/BufferProviders.h
index 68b3f23..d5899ea 100644
--- a/include/media/BufferProviders.h
+++ b/include/media/BufferProviders.h
@@ -21,6 +21,7 @@
 #include <sys/types.h>
 
 #include <media/AudioBufferProvider.h>
+#include <media/AudioResamplerPublic.h>
 #include <system/audio.h>
 #include <system/audio_effect.h>
 #include <utils/StrongPointer.h>
diff --git a/include/media/RecordBufferConverter.h b/include/media/RecordBufferConverter.h
new file mode 100644
index 0000000..2abc45e
--- /dev/null
+++ b/include/media/RecordBufferConverter.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2017 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_RECORD_BUFFER_CONVERTER_H
+#define ANDROID_RECORD_BUFFER_CONVERTER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <media/AudioBufferProvider.h>
+#include <system/audio.h>
+
+class AudioResampler;
+class PassthruBufferProvider;
+
+namespace android {
+
+/* The RecordBufferConverter is used for format, channel, and sample rate
+ * conversion for a RecordTrack.
+ *
+ * RecordBufferConverter uses the convert() method rather than exposing a
+ * buffer provider interface; this is to save a memory copy.
+ *
+ * There are legacy conversion requirements for this converter, specifically
+ * due to mono handling, so be careful about modifying.
+ *
+ * Original source audioflinger/Threads.{h,cpp}
+ */
+class RecordBufferConverter
+{
+public:
+    RecordBufferConverter(
+            audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+            uint32_t srcSampleRate,
+            audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+            uint32_t dstSampleRate);
+
+    ~RecordBufferConverter();
+
+    /* Converts input data from an AudioBufferProvider by format, channelMask,
+     * and sampleRate to a destination buffer.
+     *
+     * Parameters
+     *      dst:  buffer to place the converted data.
+     * provider:  buffer provider to obtain source data.
+     *   frames:  number of frames to convert
+     *
+     * Returns the number of frames converted.
+     */
+    size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
+
+    // returns NO_ERROR if constructor was successful
+    status_t initCheck() const {
+        // mSrcChannelMask set on successful updateParameters
+        return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
+    }
+
+    // allows dynamic reconfigure of all parameters
+    status_t updateParameters(
+            audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+            uint32_t srcSampleRate,
+            audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+            uint32_t dstSampleRate);
+
+    // called to reset resampler buffers on record track discontinuity
+    void reset();
+
+private:
+    // format conversion when not using resampler
+    void convertNoResampler(void *dst, const void *src, size_t frames);
+
+    // format conversion when using resampler; modifies src in-place
+    void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames);
+
+    // user provided information
+    audio_channel_mask_t mSrcChannelMask;
+    audio_format_t       mSrcFormat;
+    uint32_t             mSrcSampleRate;
+    audio_channel_mask_t mDstChannelMask;
+    audio_format_t       mDstFormat;
+    uint32_t             mDstSampleRate;
+
+    // derived information
+    uint32_t             mSrcChannelCount;
+    uint32_t             mDstChannelCount;
+    size_t               mDstFrameSize;
+
+    // format conversion buffer
+    void                *mBuf;
+    size_t               mBufFrames;
+    size_t               mBufFrameSize;
+
+    // resampler info
+    AudioResampler      *mResampler;
+
+    bool                 mIsLegacyDownmix;  // legacy stereo to mono conversion needed
+    bool                 mIsLegacyUpmix;    // legacy mono to stereo conversion needed
+    bool                 mRequiresFloat;    // data processing requires float (e.g. resampler)
+    PassthruBufferProvider *mInputConverterProvider;    // converts input to float
+    int8_t               mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_RECORD_BUFFER_CONVERTER_H
diff --git a/media/libaudioprocessing/Android.mk b/media/libaudioprocessing/Android.mk
index d47d158..b7ea99e 100644
--- a/media/libaudioprocessing/Android.mk
+++ b/media/libaudioprocessing/Android.mk
@@ -9,6 +9,7 @@
     AudioResamplerSinc.cpp.arm \
     AudioResamplerDyn.cpp.arm \
     BufferProviders.cpp \
+    RecordBufferConverter.cpp \
 
 LOCAL_C_INCLUDES := \
     $(TOP) \
diff --git a/media/libaudioprocessing/RecordBufferConverter.cpp b/media/libaudioprocessing/RecordBufferConverter.cpp
new file mode 100644
index 0000000..54151f5
--- /dev/null
+++ b/media/libaudioprocessing/RecordBufferConverter.cpp
@@ -0,0 +1,294 @@
+/*
+ * Copyright (C) 2017 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 "RecordBufferConverter"
+//#define LOG_NDEBUG 0
+
+#include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
+#include <media/AudioMixer.h>  // for UNITY_GAIN_FLOAT
+#include <media/AudioResampler.h>
+#include <media/BufferProviders.h>
+#include <media/RecordBufferConverter.h>
+#include <utils/Log.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
+#endif
+
+template <typename T>
+static inline T max(const T& a, const T& b)
+{
+    return a > b ? a : b;
+}
+
+namespace android {
+
+RecordBufferConverter::RecordBufferConverter(
+        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+        uint32_t srcSampleRate,
+        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+        uint32_t dstSampleRate) :
+            mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
+            // mSrcFormat
+            // mSrcSampleRate
+            // mDstChannelMask
+            // mDstFormat
+            // mDstSampleRate
+            // mSrcChannelCount
+            // mDstChannelCount
+            // mDstFrameSize
+            mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
+            mResampler(NULL),
+            mIsLegacyDownmix(false),
+            mIsLegacyUpmix(false),
+            mRequiresFloat(false),
+            mInputConverterProvider(NULL)
+{
+    (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
+            dstChannelMask, dstFormat, dstSampleRate);
+}
+
+RecordBufferConverter::~RecordBufferConverter() {
+    free(mBuf);
+    delete mResampler;
+    delete mInputConverterProvider;
+}
+
+void RecordBufferConverter::reset() {
+    if (mResampler != NULL) {
+        mResampler->reset();
+    }
+}
+
+size_t RecordBufferConverter::convert(void *dst,
+        AudioBufferProvider *provider, size_t frames)
+{
+    if (mInputConverterProvider != NULL) {
+        mInputConverterProvider->setBufferProvider(provider);
+        provider = mInputConverterProvider;
+    }
+
+    if (mResampler == NULL) {
+        ALOGV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+                mSrcSampleRate, mSrcFormat, mDstFormat);
+
+        AudioBufferProvider::Buffer buffer;
+        for (size_t i = frames; i > 0; ) {
+            buffer.frameCount = i;
+            status_t status = provider->getNextBuffer(&buffer);
+            if (status != OK || buffer.frameCount == 0) {
+                frames -= i; // cannot fill request.
+                break;
+            }
+            // format convert to destination buffer
+            convertNoResampler(dst, buffer.raw, buffer.frameCount);
+
+            dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
+            i -= buffer.frameCount;
+            provider->releaseBuffer(&buffer);
+        }
+    } else {
+         ALOGV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+                 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
+
+         // reallocate buffer if needed
+         if (mBufFrameSize != 0 && mBufFrames < frames) {
+             free(mBuf);
+             mBufFrames = frames;
+             (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
+         }
+        // resampler accumulates, but we only have one source track
+        memset(mBuf, 0, frames * mBufFrameSize);
+        frames = mResampler->resample((int32_t*)mBuf, frames, provider);
+        // format convert to destination buffer
+        convertResampler(dst, mBuf, frames);
+    }
+    return frames;
+}
+
+status_t RecordBufferConverter::updateParameters(
+        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+        uint32_t srcSampleRate,
+        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+        uint32_t dstSampleRate)
+{
+    // quick evaluation if there is any change.
+    if (mSrcFormat == srcFormat
+            && mSrcChannelMask == srcChannelMask
+            && mSrcSampleRate == srcSampleRate
+            && mDstFormat == dstFormat
+            && mDstChannelMask == dstChannelMask
+            && mDstSampleRate == dstSampleRate) {
+        return NO_ERROR;
+    }
+
+    ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
+            "  srcFormat:%#x dstFormat:%#x  srcRate:%u dstRate:%u",
+            srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
+    const bool valid =
+            audio_is_input_channel(srcChannelMask)
+            && audio_is_input_channel(dstChannelMask)
+            && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
+            && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
+            && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
+            ; // no upsampling checks for now
+    if (!valid) {
+        return BAD_VALUE;
+    }
+
+    mSrcFormat = srcFormat;
+    mSrcChannelMask = srcChannelMask;
+    mSrcSampleRate = srcSampleRate;
+    mDstFormat = dstFormat;
+    mDstChannelMask = dstChannelMask;
+    mDstSampleRate = dstSampleRate;
+
+    // compute derived parameters
+    mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
+    mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
+    mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
+
+    // do we need to resample?
+    delete mResampler;
+    mResampler = NULL;
+    if (mSrcSampleRate != mDstSampleRate) {
+        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT,
+                mSrcChannelCount, mDstSampleRate);
+        mResampler->setSampleRate(mSrcSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
+    }
+
+    // are we running legacy channel conversion modes?
+    mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO
+                            || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK)
+                   && mDstChannelMask == AUDIO_CHANNEL_IN_MONO;
+    mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO
+                   && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO
+                            || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK);
+
+    // do we need to process in float?
+    mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix;
+
+    // do we need a staging buffer to convert for destination (we can still optimize this)?
+    // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity
+    if (mResampler != NULL) {
+        mBufFrameSize = max(mSrcChannelCount, (uint32_t)FCC_2)
+                * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
+    } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float
+        mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
+    } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) {
+        mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
+    } else {
+        mBufFrameSize = 0;
+    }
+    mBufFrames = 0; // force the buffer to be resized.
+
+    // do we need an input converter buffer provider to give us float?
+    delete mInputConverterProvider;
+    mInputConverterProvider = NULL;
+    if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) {
+        mInputConverterProvider = new ReformatBufferProvider(
+                audio_channel_count_from_in_mask(mSrcChannelMask),
+                mSrcFormat,
+                AUDIO_FORMAT_PCM_FLOAT,
+                256 /* provider buffer frame count */);
+    }
+
+    // do we need a remixer to do channel mask conversion
+    if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) {
+        (void) memcpy_by_index_array_initialization_from_channel_mask(
+                mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask);
+    }
+    return NO_ERROR;
+}
+
+void RecordBufferConverter::convertNoResampler(
+        void *dst, const void *src, size_t frames)
+{
+    // src is native type unless there is legacy upmix or downmix, whereupon it is float.
+    if (mBufFrameSize != 0 && mBufFrames < frames) {
+        free(mBuf);
+        mBufFrames = frames;
+        (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
+    }
+    // do we need to do legacy upmix and downmix?
+    if (mIsLegacyUpmix || mIsLegacyDownmix) {
+        void *dstBuf = mBuf != NULL ? mBuf : dst;
+        if (mIsLegacyUpmix) {
+            upmix_to_stereo_float_from_mono_float((float *)dstBuf,
+                    (const float *)src, frames);
+        } else /*mIsLegacyDownmix */ {
+            downmix_to_mono_float_from_stereo_float((float *)dstBuf,
+                    (const float *)src, frames);
+        }
+        if (mBuf != NULL) {
+            memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT,
+                    frames * mDstChannelCount);
+        }
+        return;
+    }
+    // do we need to do channel mask conversion?
+    if (mSrcChannelMask != mDstChannelMask) {
+        void *dstBuf = mBuf != NULL ? mBuf : dst;
+        memcpy_by_index_array(dstBuf, mDstChannelCount,
+                src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames);
+        if (dstBuf == dst) {
+            return; // format is the same
+        }
+    }
+    // convert to destination buffer
+    const void *convertBuf = mBuf != NULL ? mBuf : src;
+    memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat,
+            frames * mDstChannelCount);
+}
+
+void RecordBufferConverter::convertResampler(
+        void *dst, /*not-a-const*/ void *src, size_t frames)
+{
+    // src buffer format is ALWAYS float when entering this routine
+    if (mIsLegacyUpmix) {
+        ; // mono to stereo already handled by resampler
+    } else if (mIsLegacyDownmix
+            || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) {
+        // the resampler outputs stereo for mono input channel (a feature?)
+        // must convert to mono
+        downmix_to_mono_float_from_stereo_float((float *)src,
+                (const float *)src, frames);
+    } else if (mSrcChannelMask != mDstChannelMask) {
+        // convert to mono channel again for channel mask conversion (could be skipped
+        // with further optimization).
+        if (mSrcChannelCount == 1) {
+            downmix_to_mono_float_from_stereo_float((float *)src,
+                (const float *)src, frames);
+        }
+        // convert to destination format (in place, OK as float is larger than other types)
+        if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) {
+            memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
+                    frames * mSrcChannelCount);
+        }
+        // channel convert and save to dst
+        memcpy_by_index_array(dst, mDstChannelCount,
+                src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames);
+        return;
+    }
+    // convert to destination format and save to dst
+    memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
+            frames * mDstChannelCount);
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/media/libaudioprocessing/tests/resampler_tests.cpp b/media/libaudioprocessing/tests/resampler_tests.cpp
index 8d5e016..a23c000 100644
--- a/media/libaudioprocessing/tests/resampler_tests.cpp
+++ b/media/libaudioprocessing/tests/resampler_tests.cpp
@@ -32,8 +32,8 @@
 #include <utility>
 #include <vector>
 
-#include <android/log.h>
 #include <gtest/gtest.h>
+#include <log/log.h>
 #include <media/AudioBufferProvider.h>
 
 #include <media/AudioResampler.h>
diff --git a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
index 19d408d..9d29cf1 100644
--- a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
+++ b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
@@ -19,12 +19,13 @@
 
 #include <assert.h>
 #include <math.h>
-#include <new>
 #include <stdlib.h>
 #include <string.h>
 #include <time.h>
 
-#include <android/log.h>
+#include <new>
+
+#include <log/log.h>
 
 #include <audio_effects/effect_loudnessenhancer.h>
 #include "dsp/core/dynamic_range_compression.h"
diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp
index f5e11a6..db4d009 100644
--- a/media/libeffects/testlibs/EffectEqualizer.cpp
+++ b/media/libeffects/testlibs/EffectEqualizer.cpp
@@ -22,9 +22,10 @@
 #include <assert.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <new>
 
-#include <android/log.h>
+#include <log/log.h>
 
 #include "AudioEqualizer.h"
 #include "AudioBiquadFilter.h"
diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c
index 08bf9ae..fce9bed 100644
--- a/media/libeffects/testlibs/EffectReverb.c
+++ b/media/libeffects/testlibs/EffectReverb.c
@@ -21,7 +21,7 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include <android/log.h>
+#include <log/log.h>
 
 #include "EffectReverb.h"
 #include "EffectsMath.h"
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index d1d1077..031b473 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -618,6 +618,12 @@
           break;
       }
 
+      case kWhatGetTrackInfo:
+      {
+          onGetTrackInfo(msg);
+          break;
+      }
+
       case kWhatSelectTrack:
       {
           onSelectTrack(msg);
@@ -868,6 +874,34 @@
 }
 
 sp<AMessage> NuPlayer::GenericSource::getTrackInfo(size_t trackIndex) const {
+    sp<AMessage> msg = new AMessage(kWhatGetTrackInfo, this);
+    msg->setSize("trackIndex", trackIndex);
+
+    sp<AMessage> response;
+    sp<RefBase> format;
+    status_t err = msg->postAndAwaitResponse(&response);
+    if (err == OK && response != NULL) {
+        CHECK(response->findObject("format", &format));
+        return static_cast<AMessage*>(format.get());
+    } else {
+        return NULL;
+    }
+}
+
+void NuPlayer::GenericSource::onGetTrackInfo(const sp<AMessage>& msg) const {
+    size_t trackIndex;
+    CHECK(msg->findSize("trackIndex", &trackIndex));
+
+    sp<AMessage> response = new AMessage;
+    sp<AMessage> format = doGetTrackInfo(trackIndex);
+    response->setObject("format", format);
+
+    sp<AReplyToken> replyID;
+    CHECK(msg->senderAwaitsResponse(&replyID));
+    response->postReply(replyID);
+}
+
+sp<AMessage> NuPlayer::GenericSource::doGetTrackInfo(size_t trackIndex) const {
     size_t trackCount = mSources.size();
     if (trackIndex >= trackCount) {
         return NULL;
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index a14056f..81e19bc 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -245,6 +245,9 @@
     void onGetFormatMeta(const sp<AMessage>& msg) const;
     sp<MetaData> doGetFormatMeta(bool audio) const;
 
+    void onGetTrackInfo(const sp<AMessage>& msg) const;
+    sp<AMessage> doGetTrackInfo(size_t trackIndex) const;
+
     void onGetSelectedTrack(const sp<AMessage>& msg) const;
     ssize_t doGetSelectedTrack(media_track_type type) const;
 
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index 045e044..1b0db33 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -22,8 +22,8 @@
 
 #include "AMessage.h"
 
-#include <android/log.h>
 #include <binder/Parcel.h>
+#include <log/log.h>
 
 #include "AAtomizer.h"
 #include "ABuffer.h"
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 35eceb2..e97d1ed 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -80,6 +80,7 @@
 class EffectsFactoryHalInterface;
 class FastMixer;
 class PassthruBufferProvider;
+class RecordBufferConverter;
 class ServerProxy;
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e025316..a8c761f 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -29,6 +29,7 @@
 #include <cutils/properties.h>
 #include <media/AudioParameter.h>
 #include <media/AudioResamplerPublic.h>
+#include <media/RecordBufferConverter.h>
 #include <media/TypeConverter.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
@@ -6936,252 +6937,6 @@
     buffer->frameCount = 0;
 }
 
-AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
-        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
-        uint32_t srcSampleRate,
-        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
-        uint32_t dstSampleRate) :
-            mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
-            // mSrcFormat
-            // mSrcSampleRate
-            // mDstChannelMask
-            // mDstFormat
-            // mDstSampleRate
-            // mSrcChannelCount
-            // mDstChannelCount
-            // mDstFrameSize
-            mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
-            mResampler(NULL),
-            mIsLegacyDownmix(false),
-            mIsLegacyUpmix(false),
-            mRequiresFloat(false),
-            mInputConverterProvider(NULL)
-{
-    (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
-            dstChannelMask, dstFormat, dstSampleRate);
-}
-
-AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
-    free(mBuf);
-    delete mResampler;
-    delete mInputConverterProvider;
-}
-
-size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
-        AudioBufferProvider *provider, size_t frames)
-{
-    if (mInputConverterProvider != NULL) {
-        mInputConverterProvider->setBufferProvider(provider);
-        provider = mInputConverterProvider;
-    }
-
-    if (mResampler == NULL) {
-        ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
-                mSrcSampleRate, mSrcFormat, mDstFormat);
-
-        AudioBufferProvider::Buffer buffer;
-        for (size_t i = frames; i > 0; ) {
-            buffer.frameCount = i;
-            status_t status = provider->getNextBuffer(&buffer);
-            if (status != OK || buffer.frameCount == 0) {
-                frames -= i; // cannot fill request.
-                break;
-            }
-            // format convert to destination buffer
-            convertNoResampler(dst, buffer.raw, buffer.frameCount);
-
-            dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
-            i -= buffer.frameCount;
-            provider->releaseBuffer(&buffer);
-        }
-    } else {
-         ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
-                 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
-
-         // reallocate buffer if needed
-         if (mBufFrameSize != 0 && mBufFrames < frames) {
-             free(mBuf);
-             mBufFrames = frames;
-             (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
-         }
-        // resampler accumulates, but we only have one source track
-        memset(mBuf, 0, frames * mBufFrameSize);
-        frames = mResampler->resample((int32_t*)mBuf, frames, provider);
-        // format convert to destination buffer
-        convertResampler(dst, mBuf, frames);
-    }
-    return frames;
-}
-
-status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
-        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
-        uint32_t srcSampleRate,
-        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
-        uint32_t dstSampleRate)
-{
-    // quick evaluation if there is any change.
-    if (mSrcFormat == srcFormat
-            && mSrcChannelMask == srcChannelMask
-            && mSrcSampleRate == srcSampleRate
-            && mDstFormat == dstFormat
-            && mDstChannelMask == dstChannelMask
-            && mDstSampleRate == dstSampleRate) {
-        return NO_ERROR;
-    }
-
-    ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
-            "  srcFormat:%#x dstFormat:%#x  srcRate:%u dstRate:%u",
-            srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
-    const bool valid =
-            audio_is_input_channel(srcChannelMask)
-            && audio_is_input_channel(dstChannelMask)
-            && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
-            && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
-            && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
-            ; // no upsampling checks for now
-    if (!valid) {
-        return BAD_VALUE;
-    }
-
-    mSrcFormat = srcFormat;
-    mSrcChannelMask = srcChannelMask;
-    mSrcSampleRate = srcSampleRate;
-    mDstFormat = dstFormat;
-    mDstChannelMask = dstChannelMask;
-    mDstSampleRate = dstSampleRate;
-
-    // compute derived parameters
-    mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
-    mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
-    mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
-
-    // do we need to resample?
-    delete mResampler;
-    mResampler = NULL;
-    if (mSrcSampleRate != mDstSampleRate) {
-        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT,
-                mSrcChannelCount, mDstSampleRate);
-        mResampler->setSampleRate(mSrcSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
-    }
-
-    // are we running legacy channel conversion modes?
-    mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO
-                            || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK)
-                   && mDstChannelMask == AUDIO_CHANNEL_IN_MONO;
-    mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO
-                   && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO
-                            || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK);
-
-    // do we need to process in float?
-    mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix;
-
-    // do we need a staging buffer to convert for destination (we can still optimize this)?
-    // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity
-    if (mResampler != NULL) {
-        mBufFrameSize = max(mSrcChannelCount, FCC_2)
-                * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
-    } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float
-        mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
-    } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) {
-        mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
-    } else {
-        mBufFrameSize = 0;
-    }
-    mBufFrames = 0; // force the buffer to be resized.
-
-    // do we need an input converter buffer provider to give us float?
-    delete mInputConverterProvider;
-    mInputConverterProvider = NULL;
-    if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) {
-        mInputConverterProvider = new ReformatBufferProvider(
-                audio_channel_count_from_in_mask(mSrcChannelMask),
-                mSrcFormat,
-                AUDIO_FORMAT_PCM_FLOAT,
-                256 /* provider buffer frame count */);
-    }
-
-    // do we need a remixer to do channel mask conversion
-    if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) {
-        (void) memcpy_by_index_array_initialization_from_channel_mask(
-                mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask);
-    }
-    return NO_ERROR;
-}
-
-void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler(
-        void *dst, const void *src, size_t frames)
-{
-    // src is native type unless there is legacy upmix or downmix, whereupon it is float.
-    if (mBufFrameSize != 0 && mBufFrames < frames) {
-        free(mBuf);
-        mBufFrames = frames;
-        (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
-    }
-    // do we need to do legacy upmix and downmix?
-    if (mIsLegacyUpmix || mIsLegacyDownmix) {
-        void *dstBuf = mBuf != NULL ? mBuf : dst;
-        if (mIsLegacyUpmix) {
-            upmix_to_stereo_float_from_mono_float((float *)dstBuf,
-                    (const float *)src, frames);
-        } else /*mIsLegacyDownmix */ {
-            downmix_to_mono_float_from_stereo_float((float *)dstBuf,
-                    (const float *)src, frames);
-        }
-        if (mBuf != NULL) {
-            memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT,
-                    frames * mDstChannelCount);
-        }
-        return;
-    }
-    // do we need to do channel mask conversion?
-    if (mSrcChannelMask != mDstChannelMask) {
-        void *dstBuf = mBuf != NULL ? mBuf : dst;
-        memcpy_by_index_array(dstBuf, mDstChannelCount,
-                src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames);
-        if (dstBuf == dst) {
-            return; // format is the same
-        }
-    }
-    // convert to destination buffer
-    const void *convertBuf = mBuf != NULL ? mBuf : src;
-    memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat,
-            frames * mDstChannelCount);
-}
-
-void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler(
-        void *dst, /*not-a-const*/ void *src, size_t frames)
-{
-    // src buffer format is ALWAYS float when entering this routine
-    if (mIsLegacyUpmix) {
-        ; // mono to stereo already handled by resampler
-    } else if (mIsLegacyDownmix
-            || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) {
-        // the resampler outputs stereo for mono input channel (a feature?)
-        // must convert to mono
-        downmix_to_mono_float_from_stereo_float((float *)src,
-                (const float *)src, frames);
-    } else if (mSrcChannelMask != mDstChannelMask) {
-        // convert to mono channel again for channel mask conversion (could be skipped
-        // with further optimization).
-        if (mSrcChannelCount == 1) {
-            downmix_to_mono_float_from_stereo_float((float *)src,
-                (const float *)src, frames);
-        }
-        // convert to destination format (in place, OK as float is larger than other types)
-        if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) {
-            memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
-                    frames * mSrcChannelCount);
-        }
-        // channel convert and save to dst
-        memcpy_by_index_array(dst, mDstChannelCount,
-                src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames);
-        return;
-    }
-    // convert to destination format and save to dst
-    memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
-            frames * mDstChannelCount);
-}
 
 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                         status_t& status)
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index e43f001..3fb0b07 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1323,92 +1323,6 @@
                                             // rolling counter that is never cleared
     };
 
-    /* The RecordBufferConverter is used for format, channel, and sample rate
-     * conversion for a RecordTrack.
-     *
-     * TODO: Self contained, so move to a separate file later.
-     *
-     * RecordBufferConverter uses the convert() method rather than exposing a
-     * buffer provider interface; this is to save a memory copy.
-     */
-    class RecordBufferConverter
-    {
-    public:
-        RecordBufferConverter(
-                audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
-                uint32_t srcSampleRate,
-                audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
-                uint32_t dstSampleRate);
-
-        ~RecordBufferConverter();
-
-        /* Converts input data from an AudioBufferProvider by format, channelMask,
-         * and sampleRate to a destination buffer.
-         *
-         * Parameters
-         *      dst:  buffer to place the converted data.
-         * provider:  buffer provider to obtain source data.
-         *   frames:  number of frames to convert
-         *
-         * Returns the number of frames converted.
-         */
-        size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
-
-        // returns NO_ERROR if constructor was successful
-        status_t initCheck() const {
-            // mSrcChannelMask set on successful updateParameters
-            return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
-        }
-
-        // allows dynamic reconfigure of all parameters
-        status_t updateParameters(
-                audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
-                uint32_t srcSampleRate,
-                audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
-                uint32_t dstSampleRate);
-
-        // called to reset resampler buffers on record track discontinuity
-        void reset() {
-            if (mResampler != NULL) {
-                mResampler->reset();
-            }
-        }
-
-    private:
-        // format conversion when not using resampler
-        void convertNoResampler(void *dst, const void *src, size_t frames);
-
-        // format conversion when using resampler; modifies src in-place
-        void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames);
-
-        // user provided information
-        audio_channel_mask_t mSrcChannelMask;
-        audio_format_t       mSrcFormat;
-        uint32_t             mSrcSampleRate;
-        audio_channel_mask_t mDstChannelMask;
-        audio_format_t       mDstFormat;
-        uint32_t             mDstSampleRate;
-
-        // derived information
-        uint32_t             mSrcChannelCount;
-        uint32_t             mDstChannelCount;
-        size_t               mDstFrameSize;
-
-        // format conversion buffer
-        void                *mBuf;
-        size_t               mBufFrames;
-        size_t               mBufFrameSize;
-
-        // resampler info
-        AudioResampler      *mResampler;
-
-        bool                 mIsLegacyDownmix;  // legacy stereo to mono conversion needed
-        bool                 mIsLegacyUpmix;    // legacy mono to stereo conversion needed
-        bool                 mRequiresFloat;    // data processing requires float (e.g. resampler)
-        PassthruBufferProvider *mInputConverterProvider;    // converts input to float
-        int8_t               mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion
-    };
-
 #include "RecordTracks.h"
 
             RecordThread(const sp<AudioFlinger>& audioFlinger,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 48e09c7..f2dd884 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -32,6 +32,7 @@
 
 #include <media/nbaio/Pipe.h>
 #include <media/nbaio/PipeReader.h>
+#include <media/RecordBufferConverter.h>
 #include <audio_utils/minifloat.h>
 
 // ----------------------------------------------------------------------------
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index 1d6787a..c2981a1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -21,7 +21,7 @@
 #include "AudioGain.h"
 #include "TypeConverter.h"
 
-#include <android/log.h>
+#include <log/log.h>
 #include <utils/String8.h>
 
 namespace android {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
index f19b43c..dbdcca7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -23,7 +23,7 @@
 #include "AudioGain.h"
 #include "TypeConverter.h"
 
-#include <android/log.h>
+#include <log/log.h>
 #include <utils/String8.h>
 
 namespace android {
diff --git a/services/mediacodec/minijail/minijail.cpp b/services/mediacodec/minijail/minijail.cpp
index 7926380..463f161 100644
--- a/services/mediacodec/minijail/minijail.cpp
+++ b/services/mediacodec/minijail/minijail.cpp
@@ -19,7 +19,8 @@
 
 #include <unistd.h>
 
-#include <android/log.h>
+#include <log/log.h>
+
 #include <libminijail.h>
 
 #include "minijail.h"
diff --git a/services/mediaextractor/minijail/minijail.cpp b/services/mediaextractor/minijail/minijail.cpp
index 8291633..c44d00d 100644
--- a/services/mediaextractor/minijail/minijail.cpp
+++ b/services/mediaextractor/minijail/minijail.cpp
@@ -19,7 +19,8 @@
 
 #include <unistd.h>
 
-#include <android/log.h>
+#include <log/log.h>
+
 #include <libminijail.h>
 
 #include "minijail.h"