diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 84df353..a3ee6c2 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -567,7 +567,7 @@
     void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
 
     // update proxy
-    mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
+    mProxy = new AudioRecordClientProxy(cblk, buffers, mCblk->frameCount_, mFrameSize);
     mProxy->setEpoch(epoch);
     mProxy->setMinimum(mNotificationFramesAct);
 
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 596a069..06a7beb 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -2301,6 +2301,7 @@
              * Decoding time: decodingTimeUs
              * Composition time offset = composition time - decoding time
              */
+            int64_t tmpCttsOffsetTimeUs;
             int64_t decodingTimeUs;
             CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
 #ifdef QCOM_HARDWARE
@@ -2310,7 +2311,9 @@
             decodingTimeUs -= previousPausedDurationUs;
             cttsOffsetTimeUs =
                     timestampUs - decodingTimeUs;
-            CHECK_GE(kMaxCttsOffsetTimeUs, decodingTimeUs - timestampUs);
+            tmpCttsOffsetTimeUs = kMaxCttsOffsetTimeUs;
+            ExtendedUtils::HFR::reCalculateTimeStamp(mMeta, tmpCttsOffsetTimeUs);
+            CHECK_GE(tmpCttsOffsetTimeUs, decodingTimeUs - timestampUs);
             timestampUs = decodingTimeUs;
             ALOGV("decoding time: %lld and ctts offset time: %lld",
                 timestampUs, cttsOffsetTimeUs);
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 124205c..09c9b3e 100755
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -36,10 +36,6 @@
 #include <media/stagefright/ExtendedCodec.h>
 #endif
 
-#ifdef ENABLE_AV_ENHANCEMENTS
-#include "QCMetaData.h"
-#endif
-
 namespace android {
 
 uint16_t U16_AT(const uint8_t *ptr) {
@@ -135,30 +131,6 @@
         if (meta->findInt32(kKeyIsADTS, &isADTS)) {
             msg->setInt32("is-adts", true);
         }
-
-#ifdef ENABLE_AV_ENHANCEMENTS
-        uint32_t type;
-        const void *data;
-        size_t size;
-
-        if (meta->findData(kKeyAacCodecSpecificData, &type, &data, &size)) {
-            if (size > 0 && data != NULL) {
-                sp<ABuffer> buffer = new ABuffer(size);
-                if (buffer != NULL) {
-                    memcpy(buffer->data(), data, size);
-                    buffer->meta()->setInt32("csd", true);
-                    buffer->meta()->setInt64("timeUs", 0);
-                    msg->setBuffer("csd-0", buffer);
-                }
-                else {
-                    ALOGE("kKeyAacCodecSpecificData ABuffer Allocation failed");
-                }
-            }
-            else {
-                ALOGE("Not a valid data pointer or size == 0");
-            }
-       }
-#endif
     }
 
     int32_t maxInputSize;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 28715d6..a00e42e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -191,6 +191,9 @@
     Mutex::Autolock _l(mLock);
 
     /* TODO: move all this work into an Init() function */
+    mLPASessionId = -2; // -2 is invalid session ID
+    mIsEffectConfigChanged = false;
+    mLPAEffectChain = NULL;
     char val_str[PROPERTY_VALUE_MAX] = { 0 };
     if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
         uint32_t int_val;
@@ -1127,6 +1130,8 @@
 #ifdef SRS_PROCESSING
                 ALOGV("setParameters:: routing change to device %d", device);
                 POSTPRO_PATCH_ICS_OUTPROC_MIX_ROUTE(desc->trackRefPtr, param, device);
+                if(desc->flag & AUDIO_OUTPUT_FLAG_TUNNEL)
+                    audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
 #endif
                 if(mLPAEffectChain != NULL){
                     mLPAEffectChain->setDevice_l(device);
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index de21985..4cb6df5 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -1033,7 +1033,9 @@
             } else {
                 data2->mKeyValuePairs = param2.toString();
             }
-            command->mTime = command2->mTime;
+            if (!strcmp(data2->mKeyValuePairs.string(), data->mKeyValuePairs.string())){
+                command->mTime = command2->mTime;
+            }
             // force delayMs to non 0 so that code below does not request to wait for
             // command status as the command is now delayed
             delayMs = 1;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index aa9097b..7559d4c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3718,7 +3718,8 @@
                 // TODO: implement behavior for compressed audio
                 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
                 size_t framesWritten = mBytesWritten / mFrameSize;
-                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames) ||
+                        track->isTerminated()) {
                     if (track->isStopped()) {
                         track->reset();
                     }