diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index fb47448..b2e4518 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -422,7 +422,7 @@
             //      NS_INACTIVE inactive so don't run again until re-started
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
-            nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+            nsecs_t processAudioBuffer();
 
             // caller must hold lock on mLock for all _l methods
             status_t openRecord_l(size_t epoch);
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b96b8a1..ba29e26 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -79,8 +79,12 @@
     // set/get audio hardware parameters. The function accepts a list of parameters
     // key value pairs in the form: key1=value1;key2=value2;...
     // Some keys are reserved for standard parameters (See AudioParameter class).
+    // The versions with audio_io_handle_t are intended for internal media framework use only.
     static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
     static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
+    // The versions without audio_io_handle_t are intended for JNI.
+    static status_t setParameters(const String8& keyValuePairs);
+    static String8  getParameters(const String8& keys);
 
     static void setErrorCallback(audio_error_callback cb);
 
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e163f88..9f631cf 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -626,7 +626,7 @@
             //      NS_INACTIVE inactive so don't run again until re-started
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
-            nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
+            nsecs_t processAudioBuffer();
             status_t processStreamEnd(int32_t waitCount);
 
 
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index de3fc36..3a87474 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -134,6 +134,7 @@
     kKeyRequiresSecureBuffers = 'secu',  // bool (int32_t)
 
     kKeyIsADTS            = 'adts',  // bool (int32_t)
+    kKeyAACAOT            = 'aaot',  // int32_t
 
     // If a MediaBuffer's data represents (at least partially) encrypted
     // data, the following fields aid in decryption.
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 666fafa..6ec8ed2 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -692,7 +692,7 @@
 
 // -------------------------------------------------------------------------
 
-nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
+nsecs_t AudioRecord::processAudioBuffer()
 {
     mLock.lock();
     if (mAwaitBoost) {
@@ -954,7 +954,7 @@
 
 // =========================================================================
 
-void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
 {
     sp<AudioRecord> audioRecord = mAudioRecord.promote();
     if (audioRecord != 0) {
@@ -993,7 +993,7 @@
             return true;
         }
     }
-    nsecs_t ns =  mReceiver.processAudioBuffer(this);
+    nsecs_t ns =  mReceiver.processAudioBuffer();
     switch (ns) {
     case 0:
         return true;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 8033c2c..9a76111 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -190,6 +190,16 @@
     return result;
 }
 
+status_t AudioSystem::setParameters(const String8& keyValuePairs)
+{
+    return setParameters((audio_io_handle_t) 0, keyValuePairs);
+}
+
+String8 AudioSystem::getParameters(const String8& keys)
+{
+    return getParameters((audio_io_handle_t) 0, keys);
+}
+
 // convert volume steps to natural log scale
 
 // change this value to change volume scaling
@@ -419,7 +429,7 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused) {
     Mutex::Autolock _l(AudioSystem::gLock);
 
     AudioSystem::gAudioFlinger.clear();
@@ -804,7 +814,7 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused) {
     Mutex::Autolock _l(AudioSystem::gLock);
     AudioSystem::gAudioPolicyService.clear();
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index b48cb1f..ddd4612 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1341,7 +1341,7 @@
 
 // -------------------------------------------------------------------------
 
-nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
+nsecs_t AudioTrack::processAudioBuffer()
 {
     // Currently the AudioTrack thread is not created if there are no callbacks.
     // Would it ever make sense to run the thread, even without callbacks?
@@ -1767,7 +1767,7 @@
     }
 }
 
-status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
+status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
 {
 
     const size_t SIZE = 256;
@@ -1797,7 +1797,7 @@
 
 // =========================================================================
 
-void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
 {
     sp<AudioTrack> audioTrack = mAudioTrack.promote();
     if (audioTrack != 0) {
@@ -1841,7 +1841,7 @@
             return true;
         }
     }
-    nsecs_t ns = mReceiver.processAudioBuffer(this);
+    nsecs_t ns = mReceiver.processAudioBuffer();
     switch (ns) {
     case 0:
         return true;
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index caa7900..7a1e207 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -765,7 +765,7 @@
     return (ssize_t) position;
 }
 
-status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
+status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
 {
     if (mIsShutdown) {
         buffer->mFrameCount = 0;
@@ -847,7 +847,7 @@
     buffer->mNonContig = 0;
 }
 
-void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount)
+void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount __unused)
 {
     // Unlike AudioTrackServerProxy::tallyUnderrunFrames() used for streaming tracks,
     // we don't have a location to count underrun frames.  The underrun frame counter
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 9db5b1b..10b4934 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -75,7 +75,7 @@
 }
 
 void
-IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who) {
+IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
     ALOGW("media server died");
 
     // Need to do this with the lock held
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index b420c95..98acd1f 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -199,7 +199,7 @@
     return NULL;
 }
 
-int SoundPool::load(const char* path, int priority)
+int SoundPool::load(const char* path, int priority __unused)
 {
     ALOGV("load: path=%s, priority=%d", path, priority);
     Mutex::Autolock lock(&mLock);
@@ -209,7 +209,7 @@
     return sample->sampleID();
 }
 
-int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
+int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
 {
     ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
             fd, offset, length, priority);
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 110b94c..bad2494 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -157,7 +157,7 @@
     return mRetriever->extractAlbumArt();
 }
 
-void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who) {
+void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
     Mutex::Autolock lock(MediaMetadataRetriever::sServiceLock);
     MediaMetadataRetriever::sService.clear();
     ALOGW("MediaMetadataRetriever server died!");
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 9b36b6a..dc73980 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2419,6 +2419,11 @@
         return ERROR_MALFORMED;
     }
 
+    static uint32_t kSamplingRate[] = {
+        96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+        16000, 12000, 11025, 8000, 7350
+    };
+
     ABitReader br(csd, csd_size);
     uint32_t objectType = br.getBits(5);
 
@@ -2426,6 +2431,9 @@
         objectType = 32 + br.getBits(6);
     }
 
+    //keep AOT type
+    mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
+
     uint32_t freqIndex = br.getBits(4);
 
     int32_t sampleRate = 0;
@@ -2438,29 +2446,30 @@
         numChannels = br.getBits(4);
     } else {
         numChannels = br.getBits(4);
-        if (objectType == 5) {
-            // SBR specific config per 14496-3 table 1.13
-            freqIndex = br.getBits(4);
-            if (freqIndex == 15) {
-                if (csd_size < 8) {
-                    return ERROR_MALFORMED;
-                }
-                sampleRate = br.getBits(24);
-            }
+
+        if (freqIndex == 13 || freqIndex == 14) {
+            return ERROR_MALFORMED;
         }
 
-        if (sampleRate == 0) {
-            static uint32_t kSamplingRate[] = {
-                96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-                16000, 12000, 11025, 8000, 7350
-            };
+        sampleRate = kSamplingRate[freqIndex];
+    }
 
-            if (freqIndex == 13 || freqIndex == 14) {
+    if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+        uint32_t extFreqIndex = br.getBits(4);
+        int32_t extSampleRate;
+        if (extFreqIndex == 15) {
+            if (csd_size < 8) {
                 return ERROR_MALFORMED;
             }
-
-            sampleRate = kSamplingRate[freqIndex];
+            extSampleRate = br.getBits(24);
+        } else {
+            if (extFreqIndex == 13 || extFreqIndex == 14) {
+                return ERROR_MALFORMED;
+            }
+            extSampleRate = kSamplingRate[extFreqIndex];
         }
+        //TODO: save the extension sampling rate value in meta data =>
+        //      mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
     }
 
     if (numChannels == 0) {
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 9041c21..216a329 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -562,6 +562,17 @@
         return false;
     }
 
+    // check whether it is ELD/LD content -> no offloading
+    // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
+    // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+    int32_t aacaot = -1;
+    if (meta->findInt32(kKeyAACAOT, &aacaot)) {
+        if (aacaot == 23 || aacaot == 39 ) {
+            ALOGV("track of type '%s' is ELD/LD content", mime);
+            return false;
+        }
+    }
+
     int32_t srate = -1;
     if (!meta->findInt32(kKeySampleRate, &srate)) {
         ALOGV("track of type '%s' does not publish sample rate", mime);