diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index c2ef68c..8eb1656 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -75,7 +75,8 @@
 
 AudioRecord::AudioRecord()
     : mStatus(NO_INIT), mSessionId(0),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
+      mProxy(NULL)
 {
 }
 
@@ -90,7 +91,9 @@
         int notificationFrames,
         int sessionId)
     : mStatus(NO_INIT), mSessionId(0),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mProxy(NULL)
 {
     mStatus = set(inputSource, sampleRate, format, channelMask,
             frameCount, cbf, user, notificationFrames, sessionId);
@@ -112,6 +115,7 @@
         IPCThreadState::self()->flushCommands();
         AudioSystem::releaseAudioSessionId(mSessionId);
     }
+    delete mProxy;
 }
 
 status_t AudioRecord::set(
@@ -149,6 +153,8 @@
     if (sampleRate == 0) {
         sampleRate = DEFAULT_SAMPLE_RATE;
     }
+    mSampleRate = sampleRate;
+
     // these below should probably come from the audioFlinger too...
     if (format == AUDIO_FORMAT_DEFAULT) {
         format = AUDIO_FORMAT_PCM_16_BIT;
@@ -166,6 +172,12 @@
     uint32_t channelCount = popcount(channelMask);
     mChannelCount = channelCount;
 
+    if (audio_is_linear_pcm(mFormat)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+    }
+
     if (sessionId == 0 ) {
         mSessionId = AudioSystem::newAudioSessionId();
     } else {
@@ -218,12 +230,6 @@
     // Update buffer size in case it has been limited by AudioFlinger during track creation
     mFrameCount = mCblk->frameCount_;
 
-    if (audio_is_linear_pcm(mFormat)) {
-        mFrameSize = channelCount * audio_bytes_per_sample(format);
-    } else {
-        mFrameSize = sizeof(uint8_t);
-    }
-
     mActive = false;
     mCbf = cbf;
     mNotificationFrames = notificationFrames;
@@ -360,7 +366,7 @@
 
 uint32_t AudioRecord::getSampleRate() const
 {
-    return mCblk->sampleRate;
+    return mSampleRate;
 }
 
 status_t AudioRecord::setMarkerPosition(uint32_t marker)
@@ -473,11 +479,18 @@
     mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
     cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
     cblk->waitTimeMs = 0;
+
+    // update proxy
+    delete mProxy;
+    mProxy = new AudioRecordClientProxy(cblk, mBuffers, frameCount, mFrameSize);
+
     return NO_ERROR;
 }
 
 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
 {
+    ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+
     AutoMutex lock(mLock);
     bool active;
     status_t result = NO_ERROR;
@@ -488,7 +501,7 @@
     audioBuffer->frameCount  = 0;
     audioBuffer->size        = 0;
 
-    uint32_t framesReady = cblk->framesReadyIn();
+    size_t framesReady = mProxy->framesReady();
 
     if (framesReady == 0) {
         cblk->lock.lock();
@@ -551,7 +564,7 @@
             }
             // read the server count again
         start_loop_here:
-            framesReady = cblk->framesReadyIn();
+            framesReady = mProxy->framesReady();
         }
         cblk->lock.unlock();
     }
@@ -573,15 +586,17 @@
 
     audioBuffer->frameCount  = framesReq;
     audioBuffer->size        = framesReq * mFrameSize;
-    audioBuffer->raw         = cblk->buffer(mBuffers, mFrameSize, u);
+    audioBuffer->raw         = mProxy->buffer(u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
 }
 
 void AudioRecord::releaseBuffer(Buffer* audioBuffer)
 {
+    ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+
     AutoMutex lock(mLock);
-    mCblk->stepUserIn(audioBuffer->frameCount, mFrameCount);
+    (void) mProxy->stepUser(audioBuffer->frameCount);
 }
 
 audio_io_handle_t AudioRecord::getInput() const
@@ -594,7 +609,7 @@
 audio_io_handle_t AudioRecord::getInput_l()
 {
     mInput = AudioSystem::getInput(mInputSource,
-                                mCblk->sampleRate,
+                                mSampleRate,
                                 mFormat,
                                 mChannelMask,
                                 mSessionId);
@@ -745,7 +760,7 @@
 
 
     // Manage overrun callback
-    if (active && (cblk->framesAvailableIn(mFrameCount) == 0)) {
+    if (active && (mProxy->framesAvailable() == 0)) {
         // The value of active is stale, but we are almost sure to be active here because
         // otherwise we would have exited when obtainBuffer returned STOPPED earlier.
         ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
@@ -781,7 +796,7 @@
     // if the new IAudioRecord is created, openRecord_l() will modify the
     // following member variables: mAudioRecord, mCblkMemory and mCblk.
     // It will also delete the strong references on previous IAudioRecord and IMemory
-    result = openRecord_l(cblk->sampleRate, mFormat, mFrameCount, getInput_l());
+    result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l());
     if (result == NO_ERROR) {
         newCblk = mCblk;
         // callback thread or sync event hasn't changed
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 1d87ff8..86a5579 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -97,7 +97,8 @@
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mProxy(NULL)
 {
 }
 
@@ -115,7 +116,8 @@
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mProxy(NULL)
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
@@ -136,7 +138,8 @@
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mProxy(NULL)
 {
     if (sharedBuffer == 0) {
         ALOGE("sharedBuffer must be non-0");
@@ -166,6 +169,7 @@
         IPCThreadState::self()->flushCommands();
         AudioSystem::releaseAudioSessionId(mSessionId);
     }
+    delete mProxy;
 }
 
 status_t AudioTrack::set(
@@ -212,6 +216,7 @@
         }
         sampleRate = afSampleRate;
     }
+    mSampleRate = sampleRate;
 
     // these below should probably come from the audioFlinger too...
     if (format == AUDIO_FORMAT_DEFAULT) {
@@ -252,6 +257,14 @@
     uint32_t channelCount = popcount(channelMask);
     mChannelCount = channelCount;
 
+    if (audio_is_linear_pcm(format)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+        mFrameSizeAF = channelCount * sizeof(int16_t);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+        mFrameSizeAF = sizeof(uint8_t);
+    }
+
     audio_io_handle_t output = AudioSystem::getOutput(
                                     streamType,
                                     sampleRate, format, channelMask,
@@ -300,14 +313,6 @@
     mStreamType = streamType;
     mFormat = format;
 
-    if (audio_is_linear_pcm(format)) {
-        mFrameSize = channelCount * audio_bytes_per_sample(format);
-        mFrameSizeAF = channelCount * sizeof(int16_t);
-    } else {
-        mFrameSize = sizeof(uint8_t);
-        mFrameSizeAF = sizeof(uint8_t);
-    }
-
     mSharedBuffer = sharedBuffer;
     mActive = false;
     mUserData = user;
@@ -460,6 +465,11 @@
 
 status_t AudioTrack::setVolume(float left, float right)
 {
+    if (mStatus != NO_ERROR) {
+        return mStatus;
+    }
+    ALOG_ASSERT(mProxy != NULL);
+
     if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
         return BAD_VALUE;
     }
@@ -468,7 +478,7 @@
     mVolume[LEFT] = left;
     mVolume[RIGHT] = right;
 
-    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
+    mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
 
     return NO_ERROR;
 }
@@ -481,14 +491,19 @@
 status_t AudioTrack::setAuxEffectSendLevel(float level)
 {
     ALOGV("setAuxEffectSendLevel(%f)", level);
+
+    if (mStatus != NO_ERROR) {
+        return mStatus;
+    }
+    ALOG_ASSERT(mProxy != NULL);
+
     if (level < 0.0f || level > 1.0f) {
         return BAD_VALUE;
     }
     AutoMutex lock(mLock);
 
     mSendLevel = level;
-
-    mCblk->setSendLevel(level);
+    mProxy->setSendLevel(level);
 
     return NO_ERROR;
 }
@@ -517,7 +532,9 @@
     }
 
     AutoMutex lock(mLock);
-    mCblk->sampleRate = rate;
+    mSampleRate = rate;
+    mProxy->setSampleRate(rate);
+
     return NO_ERROR;
 }
 
@@ -528,7 +545,7 @@
     }
 
     AutoMutex lock(mLock);
-    return mCblk->sampleRate;
+    return mSampleRate;
 }
 
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -665,6 +682,11 @@
 
 status_t AudioTrack::reload()
 {
+    if (mStatus != NO_ERROR) {
+        return mStatus;
+    }
+    ALOG_ASSERT(mProxy != NULL);
+
     if (mSharedBuffer == 0 || mIsTimed) {
         return INVALID_OPERATION;
     }
@@ -677,8 +699,7 @@
 
     flush_l();
 
-    audio_track_cblk_t* cblk = mCblk;
-    cblk->stepUserOut(mFrameCount, mFrameCount);
+    (void) mProxy->stepUser(mFrameCount);
 
     return NO_ERROR;
 }
@@ -693,7 +714,7 @@
 audio_io_handle_t AudioTrack::getOutput_l()
 {
     return AudioSystem::getOutput(mStreamType,
-            mCblk->sampleRate, mFormat, mChannelMask, mFlags);
+            mSampleRate, mFormat, mChannelMask, mFlags);
 }
 
 status_t AudioTrack::attachAuxEffect(int effectId)
@@ -890,13 +911,8 @@
         mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
     } else {
         mBuffers = sharedBuffer->pointer();
-        // Force buffer full condition as data is already present in shared memory
-        cblk->stepUserOut(frameCount, frameCount);
     }
 
-    cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
-            uint16_t(mVolume[LEFT] * 0x1000));
-    cblk->setSendLevel(mSendLevel);
     mAudioTrack->attachAuxEffect(mAuxEffectId);
     cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
     cblk->waitTimeMs = 0;
@@ -909,11 +925,26 @@
     if (frameCount > mReqFrameCount) {
         mReqFrameCount = frameCount;
     }
+
+    // update proxy
+    delete mProxy;
+    mProxy = new AudioTrackClientProxy(cblk, mBuffers, frameCount, mFrameSizeAF);
+    mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
+            uint16_t(mVolume[LEFT] * 0x1000));
+    mProxy->setSendLevel(mSendLevel);
+    mProxy->setSampleRate(mSampleRate);
+    if (sharedBuffer != 0) {
+        // Force buffer full condition as data is already present in shared memory
+        mProxy->stepUser(frameCount);
+    }
+
     return NO_ERROR;
 }
 
 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
 {
+    ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+
     AutoMutex lock(mLock);
     bool active;
     status_t result = NO_ERROR;
@@ -924,7 +955,7 @@
     audioBuffer->frameCount  = 0;
     audioBuffer->size = 0;
 
-    uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
+    size_t framesAvail = mProxy->framesAvailable();
 
     cblk->lock.lock();
     if (cblk->flags & CBLK_INVALID) {
@@ -1000,7 +1031,7 @@
             }
             // read the server count again
         start_loop_here:
-            framesAvail = cblk->framesAvailableOut_l(mFrameCount);
+            framesAvail = mProxy->framesAvailable_l();
         }
         cblk->lock.unlock();
     }
@@ -1020,16 +1051,18 @@
 
     audioBuffer->frameCount = framesReq;
     audioBuffer->size = framesReq * mFrameSizeAF;
-    audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u);
+    audioBuffer->raw = mProxy->buffer(u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
 }
 
 void AudioTrack::releaseBuffer(Buffer* audioBuffer)
 {
+    ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+
     AutoMutex lock(mLock);
     audio_track_cblk_t* cblk = mCblk;
-    cblk->stepUserOut(audioBuffer->frameCount, mFrameCount);
+    (void) mProxy->stepUser(audioBuffer->frameCount);
     if (audioBuffer->frameCount > 0) {
         // restart track if it was disabled by audioflinger due to previous underrun
         if (mActive && (cblk->flags & CBLK_DISABLED)) {
@@ -1199,7 +1232,7 @@
     // so all cblk references might still refer to old shared memory, but that should be benign
 
     // Manage underrun callback
-    if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) {
+    if (active && (mProxy->framesAvailable() == mFrameCount)) {
         ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
         if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
             mCbf(EVENT_UNDERRUN, mUserData, 0);
@@ -1346,7 +1379,7 @@
     // following member variables: mAudioTrack, mCblkMemory and mCblk.
     // It will also delete the strong references on previous IAudioTrack and IMemory
     result = createTrack_l(mStreamType,
-                           cblk->sampleRate,
+                           mSampleRate,
                            mFormat,
                            mReqFrameCount,  // so that frame count never goes down
                            mFlags,
@@ -1365,12 +1398,12 @@
         // restore loop: this is not guaranteed to succeed if new frame count is not
         // compatible with loop length
         setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
+        size_t frames = 0;
         if (!fromStart) {
             newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
             // Make sure that a client relying on callback events indicating underrun or
             // the actual amount of audio frames played (e.g SoundPool) receives them.
             if (mSharedBuffer == 0) {
-                uint32_t frames = 0;
                 if (user > server) {
                     frames = ((user - server) > mFrameCount) ?
                             mFrameCount : (user - server);
@@ -1378,13 +1411,15 @@
                 }
                 // restart playback even if buffer is not completely filled.
                 android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
-                // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
-                // the client
-                newCblk->stepUserOut(frames, mFrameCount);
             }
         }
         if (mSharedBuffer != 0) {
-            newCblk->stepUserOut(mFrameCount, mFrameCount);
+            frames = mFrameCount;
+        }
+        if (frames > 0) {
+            // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
+            // the client
+            mProxy->stepUser(frames);
         }
         if (mActive) {
             result = mAudioTrack->start();
@@ -1416,7 +1451,6 @@
     char buffer[SIZE];
     String8 result;
 
-    audio_track_cblk_t* cblk = mCblk;
     result.append(" AudioTrack::dump\n");
     snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
             mVolume[0], mVolume[1]);
@@ -1424,8 +1458,7 @@
     snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
             mChannelCount, mFrameCount);
     result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%u), status(%d)\n",
-            (cblk == 0) ? 0 : cblk->sampleRate, mStatus);
+    snprintf(buffer, 255, "  sample rate(%u), status(%d)\n", mSampleRate, mStatus);
     result.append(buffer);
     snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
     result.append(buffer);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index bee13c8..13d47c9 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
     : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
     userBase(0), serverBase(0), frameCount_(0),
     loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
-    mSendLevel(0), flags(0)
+    mSampleRate(0), mSendLevel(0), flags(0)
 {
 }