diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b11c812..09160cc 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -128,8 +128,10 @@
     // - BAD_VALUE: invalid parameter
     // NOTE: this feature is not supported on all hardware platforms and it is
     // necessary to check returned status before using the returned values.
-    static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getRenderPosition(audio_io_handle_t output,
+                                      size_t *halFrames,
+                                      size_t *dspFrames,
+                                      audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
 
     // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
     static size_t getInputFramesLost(audio_io_handle_t ioHandle);
@@ -197,7 +199,8 @@
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
+                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                        const audio_offload_info_t *offloadInfo = NULL);
     static status_t startOutput(audio_io_handle_t output,
                                 audio_stream_type_t stream,
                                 int session = 0);
@@ -245,6 +248,10 @@
     static uint32_t getPrimaryOutputSamplingRate();
     static size_t getPrimaryOutputFrameCount();
 
+    // Check if hw offload is possible for given format, stream type, sample rate,
+    // bit rate, duration, video and streaming or offload property is enabled
+    static bool isOffloadSupported(const audio_offload_info_t& info);
+
     // ----------------------------------------------------------------------------
 
 private:
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e9bb76a..6727601 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -60,6 +60,8 @@
                                     // Not currently used by android.media.AudioTrack.
         EVENT_NEW_IAUDIOTRACK = 6,  // IAudioTrack was re-created, either due to re-routing and
                                     // voluntary invalidation by mediaserver, or mediaserver crash.
+        EVENT_STREAM_END = 7,       // Sent after all the buffers queued in AF and HW are played
+                                    // back (after stop is called)
     };
 
     /* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -175,7 +177,8 @@
                                     void* user           = NULL,
                                     int notificationFrames = 0,
                                     int sessionId        = 0,
-                                    transfer_type transferType = TRANSFER_DEFAULT);
+                                    transfer_type transferType = TRANSFER_DEFAULT,
+                                    const audio_offload_info_t *offloadInfo = NULL);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -198,7 +201,8 @@
                                     void* user          = NULL,
                                     int notificationFrames = 0,
                                     int sessionId       = 0,
-                                    transfer_type transferType = TRANSFER_DEFAULT);
+                                    transfer_type transferType = TRANSFER_DEFAULT,
+                                    const audio_offload_info_t *offloadInfo = NULL);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
@@ -233,7 +237,8 @@
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
                             int sessionId       = 0,
-                            transfer_type transferType = TRANSFER_DEFAULT);
+                            transfer_type transferType = TRANSFER_DEFAULT,
+                            const audio_offload_info_t *offloadInfo = NULL);
 
     /* Result of constructing the AudioTrack. This must be checked
      * before using any AudioTrack API (except for set()), because using
@@ -521,6 +526,15 @@
                                      struct timespec *elapsed = NULL, size_t *nonContig = NULL);
 public:
 
+//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
+//            enum {
+//            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
+//            TEAR_DOWN       = 0x80000002,
+//            STOPPED = 1,
+//            STREAM_END_WAIT,
+//            STREAM_END
+//        };
+
     /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
     // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
             void        releaseBuffer(Buffer* audioBuffer);
@@ -550,6 +564,15 @@
      */
             uint32_t    getUnderrunFrames() const;
 
+    /* Get the flags */
+            audio_output_flags_t getFlags() const { return mFlags; }
+
+    /* Set parameters - only possible when using direct output */
+            status_t    setParameters(const String8& keyValuePairs);
+
+    /* Get parameters */
+            String8     getParameters(const String8& keys);
+
 protected:
     /* copying audio tracks is not allowed */
                         AudioTrack(const AudioTrack& other);
@@ -590,8 +613,11 @@
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
             nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
+            status_t processStreamEnd(int32_t waitCount);
+
 
             // caller must hold lock on mLock for all _l methods
+
             status_t createTrack_l(audio_stream_type_t streamType,
                                  uint32_t sampleRate,
                                  audio_format_t format,
@@ -607,6 +633,8 @@
             void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
             audio_io_handle_t getOutput_l();
 
+            status_t getPosition_l(uint32_t *position);
+
             // FIXME enum is faster than strcmp() for parameter 'from'
             status_t restoreTrack_l(const char *from);
 
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9c3067e..f8a9f2b 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -49,6 +49,7 @@
         TRACK_DEFAULT = 0,  // client requests a default AudioTrack
         TRACK_TIMED   = 1,  // client requests a TimedAudioTrack
         TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord
+        TRACK_OFFLOAD = 4,  // client requests offload to hw codec
     };
     typedef uint32_t track_flags_t;
 
@@ -137,7 +138,8 @@
                                          audio_format_t *pFormat,
                                          audio_channel_mask_t *pChannelMask,
                                          uint32_t *pLatencyMs,
-                                         audio_output_flags_t flags) = 0;
+                                         audio_output_flags_t flags,
+                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
                                     audio_io_handle_t output2) = 0;
     virtual status_t closeOutput(audio_io_handle_t output) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index b5ad4ef..09b9ea6 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -53,7 +53,8 @@
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = 0,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                        const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  int session = 0) = 0;
@@ -95,6 +96,9 @@
     virtual status_t queryDefaultPreProcessing(int audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count) = 0;
+   // Check if offload is possible for given format, stream type, sample rate,
+    // bit rate, duration, video and streaming or offload property is enabled
+    virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
 };
 
 
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 144be0e..1014403 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -25,6 +25,7 @@
 #include <binder/IInterface.h>
 #include <binder/IMemory.h>
 #include <utils/LinearTransform.h>
+#include <utils/String8.h>
 
 namespace android {
 
@@ -82,6 +83,9 @@
        or Tungsten time. The values for target are defined in AudioTrack.h */
     virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
                                               int target) = 0;
+
+    /* Send parameters to the audio hardware */
+    virtual status_t    setParameters(const String8& keyValuePairs) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 9a75f81..61f7dc7 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -74,9 +74,18 @@
     // AudioSink: abstraction layer for audio output
     class AudioSink : public RefBase {
     public:
+        enum cb_event_t {
+            CB_EVENT_FILL_BUFFER,   // Request to write more data to buffer.
+            CB_EVENT_STREAM_END,    // Sent after all the buffers queued in AF and HW are played
+                                    // back (after stop is called)
+            CB_EVENT_TEAR_DOWN      // The AudioTrack was invalidated due to use case change:
+                                    // Need to re-evaluate offloading options
+        };
+
         // Callback returns the number of bytes actually written to the buffer.
         typedef size_t (*AudioCallback)(
-                AudioSink *audioSink, void *buffer, size_t size, void *cookie);
+                AudioSink *audioSink, void *buffer, size_t size, void *cookie,
+                        cb_event_t event);
 
         virtual             ~AudioSink() {}
         virtual bool        ready() const = 0; // audio output is open and ready
@@ -99,7 +108,8 @@
                 int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
                 AudioCallback cb = NULL,
                 void *cookie = NULL,
-                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                const audio_offload_info_t *offloadInfo = NULL) = 0;
 
         virtual void        start() = 0;
         virtual ssize_t     write(const void* buffer, size_t size) = 0;
@@ -110,6 +120,9 @@
 
         virtual status_t    setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
         virtual bool        needsTrailingPadding() { return true; }
+
+        virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR; };
+        virtual String8     getParameters(const String8& keys) { return String8::empty(); };
     };
 
                         MediaPlayerBase() : mCookie(0), mNotify(0) {}
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 3bf046d..ec9f2df 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -36,8 +36,13 @@
         SEEK_COMPLETE
     };
 
+    enum {
+        ALLOW_DEEP_BUFFERING = 0x01,
+        USE_OFFLOAD = 0x02
+    };
+
     AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink,
-                bool allowDeepBuffering = false,
+                uint32_t flags = 0,
                 AwesomePlayer *audioObserver = NULL);
 
     virtual ~AudioPlayer();
@@ -67,6 +72,8 @@
 
     status_t setPlaybackRatePermille(int32_t ratePermille);
 
+    void notifyAudioEOS();
+
 private:
     friend class VideoEditorAudioPlayer;
     sp<MediaSource> mSource;
@@ -107,7 +114,8 @@
 
     static size_t AudioSinkCallback(
             MediaPlayerBase::AudioSink *audioSink,
-            void *data, size_t size, void *me);
+            void *data, size_t size, void *me,
+            MediaPlayerBase::AudioSink::cb_event_t event);
 
     size_t fillBuffer(void *data, size_t size);
 
@@ -116,6 +124,7 @@
     void reset();
 
     uint32_t getNumFramesPendingPlayout() const;
+    int64_t getOutputPlayPositionUs_l() const;
 
     AudioPlayer(const AudioPlayer &);
     AudioPlayer &operator=(const AudioPlayer &);
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index 73940d3..c24f612 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -22,6 +22,8 @@
 #include <stdint.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
+#include <system/audio.h>
+#include <media/MediaPlayerInterface.h>
 
 namespace android {
 
@@ -48,6 +50,15 @@
 
 AString MakeUserAgent();
 
+// Convert a MIME type to a AudioSystem::audio_format
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
+
+// Send information from MetaData to the HAL via AudioSink
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
+
+// Check whether the stream defined by meta can be offloaded to hardware
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming);
+
 }  // namespace android
 
 #endif  // UTILS_H_
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ef5bb8d..b41684a 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -44,6 +44,10 @@
 #define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping
 #define CBLK_OVERRUN   0x100 // set by server immediately on input overrun, cleared by client
 #define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer()
+#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
+
+//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
 
 struct AudioTrackSharedStreaming {
     // similar to NBAIO MonoPipe