diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 5af6c10..6af1962 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -139,7 +139,7 @@
      *               of descriptors to return.
      *               *count is limited to kMaxPreProcessing on return.
      */
-    static status_t queryDefaultPreProcessing(int audioSession,
+    static status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count);
 
@@ -237,7 +237,7 @@
                   int32_t priority = 0,
                   effect_callback_t cbf = NULL,
                   void* user = NULL,
-                  int sessionId = AUDIO_SESSION_OUTPUT_MIX,
+                  audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
                   audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
                   );
 
@@ -250,7 +250,7 @@
                     int32_t priority = 0,
                     effect_callback_t cbf = NULL,
                     void* user = NULL,
-                    int sessionId = AUDIO_SESSION_OUTPUT_MIX,
+                    audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
                     audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
                     );
 
@@ -272,7 +272,7 @@
                             int32_t priority = 0,
                             effect_callback_t cbf = NULL,
                             void* user = NULL,
-                            int sessionId = AUDIO_SESSION_OUTPUT_MIX,
+                            audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
                             audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
                             );
 
@@ -405,7 +405,7 @@
 
 protected:
      bool                    mEnabled;           // enable state
-     int32_t                 mSessionId;         // audio session ID
+     audio_session_t         mSessionId;         // audio session ID
      int32_t                 mPriority;          // priority for effect control
      status_t                mStatus;            // effect status
      effect_callback_t       mCbf;               // callback function for status, control and
diff --git a/include/media/AudioPolicy.h b/include/media/AudioPolicy.h
index c769a62..8528c7a 100644
--- a/include/media/AudioPolicy.h
+++ b/include/media/AudioPolicy.h
@@ -51,6 +51,7 @@
 
 #define MIX_ROUTE_FLAG_RENDER 0x1
 #define MIX_ROUTE_FLAG_LOOP_BACK (0x1 << 1)
+#define MIX_ROUTE_FLAG_ALL (MIX_ROUTE_FLAG_RENDER | MIX_ROUTE_FLAG_LOOP_BACK)
 
 #define MAX_MIXES_PER_POLICY 10
 #define MAX_CRITERIA_PER_MIX 20
@@ -81,7 +82,7 @@
     AudioMix(Vector<AudioMixMatchCriterion> criteria, uint32_t mixType, audio_config_t format,
              uint32_t routeFlags, String8 registrationId, uint32_t flags) :
         mCriteria(criteria), mMixType(mixType), mFormat(format),
-        mRouteFlags(routeFlags), mRegistrationId(registrationId), mCbFlags(flags){}
+        mRouteFlags(routeFlags), mDeviceAddress(registrationId), mCbFlags(flags){}
 
     status_t readFromParcel(Parcel *parcel);
     status_t writeToParcel(Parcel *parcel) const;
@@ -90,7 +91,8 @@
     uint32_t        mMixType;
     audio_config_t  mFormat;
     uint32_t        mRouteFlags;
-    String8         mRegistrationId;
+    audio_devices_t mDeviceType;
+    String8         mDeviceAddress;
     uint32_t        mCbFlags; // flags indicating which callbacks to use, see kCbFlag*
 };
 
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index b034c8e..2fa1a4e 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -177,7 +177,7 @@
                                     callback_t cbf = NULL,
                                     void* user = NULL,
                                     uint32_t notificationFrames = 0,
-                                    int sessionId = AUDIO_SESSION_ALLOCATE,
+                                    audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
                                     int uid = -1,
@@ -215,7 +215,7 @@
                             void* user = NULL,
                             uint32_t notificationFrames = 0,
                             bool threadCanCallJava = false,
-                            int sessionId = AUDIO_SESSION_ALLOCATE,
+                            audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
                             int uid = -1,
@@ -249,7 +249,7 @@
      * the specified event occurs on the specified trigger session.
      */
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
-                              int triggerSession = 0);
+                              audio_session_t triggerSession = AUDIO_SESSION_NONE);
 
     /* Stop a track.  The callback will cease being called.  Note that obtainBuffer() still
      * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
@@ -352,7 +352,7 @@
      *
      * No lock needed because session ID doesn't change after first set().
      */
-            int    getSessionId() const { return mSessionId; }
+            audio_session_t getSessionId() const { return mSessionId; }
 
     /* Public API for TRANSFER_OBTAIN mode.
      * Obtains a buffer of up to "audioBuffer->frameCount" full frames.
@@ -603,7 +603,7 @@
                                                     // held to read or write those bits reliably.
     audio_input_flags_t     mOrigFlags;             // as specified in constructor or set(), const
 
-    int                     mSessionId;
+    audio_session_t         mSessionId;
     transfer_type           mTransfer;
 
     // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 91e2bbd..c9eac2e 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -31,7 +31,7 @@
 
 typedef void (*audio_error_callback)(status_t err);
 typedef void (*dynamic_policy_callback)(int event, String8 regId, int val);
-typedef void (*record_config_callback)(int event, int session, int source,
+typedef void (*record_config_callback)(int event, audio_session_t session, int source,
                 const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig,
                 audio_patch_handle_t patchHandle);
 
@@ -155,8 +155,8 @@
     //       or an unspecified existing unique ID.
     static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use);
 
-    static void acquireAudioSessionId(int audioSession, pid_t pid);
-    static void releaseAudioSessionId(int audioSession, pid_t pid);
+    static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid);
+    static void releaseAudioSessionId(audio_session_t audioSession, pid_t pid);
 
     // Get the HW synchronization source used for an audio session.
     // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs
@@ -260,7 +260,7 @@
     static status_t registerEffect(const effect_descriptor_t *desc,
                                     audio_io_handle_t io,
                                     uint32_t strategy,
-                                    int session,
+                                    audio_session_t session,
                                     int id);
     static status_t unregisterEffect(int id);
     static status_t setEffectEnabled(int id, bool enabled);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 4de55a2..9e0e98b 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -213,7 +213,7 @@
                                     callback_t cbf       = NULL,
                                     void* user           = NULL,
                                     uint32_t notificationFrames = 0,
-                                    int sessionId        = AUDIO_SESSION_ALLOCATE,
+                                    audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
                                     int uid = -1,
@@ -242,7 +242,7 @@
                                     callback_t cbf      = NULL,
                                     void* user          = NULL,
                                     uint32_t notificationFrames = 0,
-                                    int sessionId       = AUDIO_SESSION_ALLOCATE,
+                                    audio_session_t sessionId   = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
                                     int uid = -1,
@@ -287,7 +287,7 @@
                             uint32_t notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
-                            int sessionId       = AUDIO_SESSION_ALLOCATE,
+                            audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             const audio_offload_info_t *offloadInfo = NULL,
                             int uid = -1,
@@ -583,7 +583,7 @@
      * Returned value:
      *  AudioTrack session ID.
      */
-            int    getSessionId() const { return mSessionId; }
+            audio_session_t getSessionId() const { return mSessionId; }
 
     /* Attach track auxiliary output to specified effect. Use effectId = 0
      * to detach track from effect.
@@ -959,7 +959,7 @@
 
     bool                    mDoNotReconnect;
 
-    int                     mSessionId;
+    audio_session_t         mSessionId;
     int                     mAuxEffectId;
 
     mutable Mutex           mLock;
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 0d9bc3b..e48aa1c 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -74,7 +74,7 @@
                                 // However on failure, the client is responsible for release.
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 int clientUid,
                                 status_t *status) = 0;
 
@@ -91,7 +91,7 @@
                                 track_flags_t *flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int clientUid,
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 size_t *notificationFrames,
                                 sp<IMemory>& cblk,
                                 sp<IMemory>& buffers,   // return value 0 means it follows cblk
@@ -184,8 +184,8 @@
 
     virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use) = 0;
 
-    virtual void acquireAudioSessionId(int audioSession, pid_t pid) = 0;
-    virtual void releaseAudioSessionId(int audioSession, pid_t pid) = 0;
+    virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid) = 0;
+    virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid) = 0;
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
 
@@ -200,13 +200,13 @@
                                     int32_t priority,
                                     // AudioFlinger doesn't take over handle reference from client
                                     audio_io_handle_t output,
-                                    int sessionId,
+                                    audio_session_t sessionId,
                                     const String16& callingPackage,
                                     status_t *status,
                                     int *id,
                                     int *enabled) = 0;
 
-    virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+    virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
                                     audio_io_handle_t dstOutput) = 0;
 
     virtual audio_module_handle_t loadHwModule(const char *name) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index ceca71a..80437dc 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -59,16 +59,16 @@
                                         audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual status_t getOutputForAttr(const audio_attributes_t *attr,
-                                        audio_io_handle_t *output,
-                                        audio_session_t session,
-                                        audio_stream_type_t *stream,
-                                        uid_t uid,
-                                        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,
-                                        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
-                                        const audio_offload_info_t *offloadInfo = NULL) = 0;
+                                      audio_io_handle_t *output,
+                                      audio_session_t session,
+                                      audio_stream_type_t *stream,
+                                      uid_t uid,
+                                      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,
+                                      audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
+                                      const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  audio_session_t session) = 0;
@@ -108,7 +108,7 @@
     virtual status_t registerEffect(const effect_descriptor_t *desc,
                                     audio_io_handle_t io,
                                     uint32_t strategy,
-                                    int session,
+                                    audio_session_t session,
                                     int id) = 0;
     virtual status_t unregisterEffect(int id) = 0;
     virtual status_t setEffectEnabled(int id, bool enabled) = 0;
@@ -116,7 +116,7 @@
     virtual bool     isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0)
                              const = 0;
     virtual bool     isSourceActive(audio_source_t source) const = 0;
-    virtual status_t queryDefaultPreProcessing(int audioSession,
+    virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count) = 0;
    // Check if offload is possible for given format, stream type, sample rate,
diff --git a/include/media/IAudioRecord.h b/include/media/IAudioRecord.h
index 2003985..7768176 100644
--- a/include/media/IAudioRecord.h
+++ b/include/media/IAudioRecord.h
@@ -24,6 +24,7 @@
 #include <utils/Errors.h>
 #include <binder/IInterface.h>
 #include <binder/IMemory.h>
+#include <system/audio.h>
 
 namespace android {
 
@@ -37,7 +38,8 @@
     /* After it's created the track is not active. Call start() to
      * make it active.
      */
-    virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) = 0;
+    virtual status_t    start(int /*AudioSystem::sync_event_t*/ event,
+                              audio_session_t triggerSession) = 0;
 
     /* Stop a track. If set, the callback will cease being called and
      * obtainBuffer will return an error. Buffers that are already released
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 99ca6f0..e5d3cda 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -49,8 +49,8 @@
 
     virtual sp<IMediaRecorder> createMediaRecorder(const String16 &opPackageName) = 0;
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
-    virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0)
-            = 0;
+    virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client,
+            audio_session_t audioSessionId = AUDIO_SESSION_ALLOCATE) = 0;
     virtual sp<IOMX>            getOMX() = 0;
     virtual sp<ICrypto>         makeCrypto() = 0;
     virtual sp<IDrm>            makeDrm() = 0;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 9e5056f..54862d1 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -101,7 +101,7 @@
         virtual status_t    getTimestamp(AudioTimestamp &ts) const = 0;
         virtual int64_t     getPlayedOutDurationUs(int64_t nowUs) const = 0;
         virtual status_t    getFramesWritten(uint32_t *frameswritten) const = 0;
-        virtual int         getSessionId() const = 0;
+        virtual audio_session_t getSessionId() const = 0;
         virtual audio_stream_type_t getAudioStreamType() const = 0;
         virtual uint32_t    getSampleRate() const = 0;
 
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index 186e018..ec0dad5 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -69,7 +69,7 @@
                                    int32_t priority = 0,
                                    effect_callback_t cbf = NULL,
                                    void* user = NULL,
-                                   int sessionId = 0);
+                                   audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX);
 
                         ~Visualizer();
 
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 00df523..cec9d99 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -246,8 +246,8 @@
             status_t        invoke(const Parcel& request, Parcel *reply);
             status_t        setMetadataFilter(const Parcel& filter);
             status_t        getMetadata(bool update_only, bool apply_filter, Parcel *metadata);
-            status_t        setAudioSessionId(int sessionId);
-            int             getAudioSessionId();
+            status_t        setAudioSessionId(audio_session_t sessionId);
+            audio_session_t getAudioSessionId();
             status_t        setAuxEffectSendLevel(float level);
             status_t        attachAuxEffect(int effectId);
             status_t        setParameter(int key, const Parcel& request);
@@ -284,7 +284,7 @@
     float                       mRightVolume;
     int                         mVideoWidth;
     int                         mVideoHeight;
-    int                         mAudioSessionId;
+    audio_session_t             mAudioSessionId;
     float                       mSendLevel;
     struct sockaddr_in          mRetransmitEndpoint;
     bool                        mRetransmitEndpointValid;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 2dfb850..92cf6bb 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -177,6 +177,10 @@
                 // server write-only, client read
                 ExtendedTimestampQueue::Shared mExtendedTimestampQueue;
 
+                // This is set by AudioTrack.setBufferSizeInFrames().
+                // A write will not fill the buffer above this limit.
+    volatile    uint32_t   mBufferSizeInFrames;  // effective size of the buffer
+
 public:
 
     volatile    int32_t     mFlags;         // combinations of CBLK_*
@@ -314,9 +318,9 @@
         return mEpoch;
     }
 
-    size_t      getBufferSizeInFrames() const { return mBufferSizeInFrames; }
-    // See documentation for AudioTrack.setBufferSizeInFrames()
-    size_t      setBufferSizeInFrames(size_t requestedSize);
+    uint32_t      getBufferSizeInFrames() const { return mBufferSizeInFrames; }
+    // See documentation for AudioTrack::setBufferSizeInFrames()
+    uint32_t      setBufferSizeInFrames(uint32_t requestedSize);
 
     status_t    getTimestamp(ExtendedTimestamp *timestamp) {
         if (timestamp == nullptr) {
@@ -331,12 +335,10 @@
         mTimestamp.clear();
     }
 
-protected:
-    // This is set by AudioTrack.setBufferSizeInFrames().
-    // A write will not fill the buffer above this limit.
-    size_t      mBufferSizeInFrames;      // effective size of the buffer
-
 private:
+    // This is a copy of mCblk->mBufferSizeInFrames
+    uint32_t   mBufferSizeInFrames;  // effective size of the buffer
+
     Modulo<uint32_t> mEpoch;
 
     // The shared buffer contents referred to by the timestamp observer
@@ -520,6 +522,11 @@
         mTimestampMutator.push(timestamp);
     }
 
+    // Get dynamic buffer size from the shared control block.
+    uint32_t            getBufferSizeInFrames() const {
+        return android_atomic_acquire_load((int32_t *)&mCblk->mBufferSizeInFrames);
+    }
+
 protected:
     size_t      mAvailToClient; // estimated frames available to client prior to releaseBuffer()
     int32_t     mFlush;         // our copy of cblk->u.mStreaming.mFlush, for streaming output only
diff --git a/media/img_utils/src/NOTICE b/media/img_utils/src/NOTICE
new file mode 100644
index 0000000..90cea57
--- /dev/null
+++ b/media/img_utils/src/NOTICE
@@ -0,0 +1,2 @@
+This product includes DNG technology under license by Adobe Systems
+Incorporated.
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index ff82544..590952f 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -47,7 +47,7 @@
                 int32_t priority,
                 effect_callback_t cbf,
                 void* user,
-                int sessionId,
+                audio_session_t sessionId,
                 audio_io_handle_t io
                 )
     : mStatus(NO_INIT), mOpPackageName(opPackageName)
@@ -61,7 +61,7 @@
                 int32_t priority,
                 effect_callback_t cbf,
                 void* user,
-                int sessionId,
+                audio_session_t sessionId,
                 audio_io_handle_t io
                 )
     : mStatus(NO_INIT), mOpPackageName(opPackageName)
@@ -93,7 +93,7 @@
                 int32_t priority,
                 effect_callback_t cbf,
                 void* user,
-                int sessionId,
+                audio_session_t sessionId,
                 audio_io_handle_t io)
 {
     sp<IEffect> iEffect;
@@ -433,7 +433,7 @@
 }
 
 
-status_t AudioEffect::queryDefaultPreProcessing(int audioSession,
+status_t AudioEffect::queryDefaultPreProcessing(audio_session_t audioSession,
                                           effect_descriptor_t *descriptors,
                                           uint32_t *count)
 {
diff --git a/media/libmedia/AudioPolicy.cpp b/media/libmedia/AudioPolicy.cpp
index ea22b6c..d1f7525 100644
--- a/media/libmedia/AudioPolicy.cpp
+++ b/media/libmedia/AudioPolicy.cpp
@@ -67,7 +67,8 @@
     mFormat.channel_mask = (audio_channel_mask_t)parcel->readInt32();
     mFormat.format = (audio_format_t)parcel->readInt32();
     mRouteFlags = parcel->readInt32();
-    mRegistrationId = parcel->readString8();
+    mDeviceType = (audio_devices_t) parcel->readInt32();
+    mDeviceAddress = parcel->readString8();
     mCbFlags = (uint32_t)parcel->readInt32();
     size_t size = (size_t)parcel->readInt32();
     if (size > MAX_CRITERIA_PER_MIX) {
@@ -89,7 +90,8 @@
     parcel->writeInt32(mFormat.channel_mask);
     parcel->writeInt32(mFormat.format);
     parcel->writeInt32(mRouteFlags);
-    parcel->writeString8(mRegistrationId);
+    parcel->writeInt32(mDeviceType);
+    parcel->writeString8(mDeviceAddress);
     parcel->writeInt32(mCbFlags);
     size_t size = mCriteria.size();
     if (size > MAX_CRITERIA_PER_MIX) {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 9eb5ec7..79ce75e 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -82,7 +82,7 @@
         callback_t cbf,
         void* user,
         uint32_t notificationFrames,
-        int sessionId,
+        audio_session_t sessionId,
         transfer_type transferType,
         audio_input_flags_t flags,
         int uid,
@@ -139,7 +139,7 @@
         void* user,
         uint32_t notificationFrames,
         bool threadCanCallJava,
-        int sessionId,
+        audio_session_t sessionId,
         transfer_type transferType,
         audio_input_flags_t flags,
         int uid,
@@ -227,7 +227,7 @@
     // mNotificationFramesAct is initialized in openRecord_l
 
     if (sessionId == AUDIO_SESSION_ALLOCATE) {
-        mSessionId = AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
+        mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
     } else {
         mSessionId = sessionId;
     }
@@ -288,7 +288,7 @@
 
 // -------------------------------------------------------------------------
 
-status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
+status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
 {
     ALOGV("start, sync event %d trigger session %d", event, triggerSession);
 
@@ -522,8 +522,16 @@
     // After fast request is denied, we will request again if IAudioRecord is re-created.
 
     status_t status;
+
+    // Not a conventional loop, but a retry loop for at most two iterations total.
+    // Try first maybe with FAST flag then try again without FAST flag if that fails.
+    // Exits loop normally via a return at the bottom, or with error via a break.
+    // The sp<> references will be dropped when re-entering scope.
+    // The lack of indentation is deliberate, to reduce code churn and ease merges.
+    for (;;) {
+
     status = AudioSystem::getInputForAttr(&mAttributes, &input,
-                                        (audio_session_t)mSessionId,
+                                        mSessionId,
                                         // FIXME compare to AudioTrack
                                         IPCThreadState::self()->getCallingUid(),
                                         mSampleRate, mFormat, mChannelMask,
@@ -535,7 +543,7 @@
               mSessionId, mAttributes.source, mSampleRate, mFormat, mChannelMask, mFlags);
         return BAD_VALUE;
     }
-    {
+
     // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
     // we must release it ourselves if anything goes wrong.
 
@@ -544,7 +552,7 @@
     status = AudioSystem::getFrameCount(input, &afFrameCount);
     if (status != NO_ERROR) {
         ALOGE("getFrameCount(input=%d) status %d", input, status);
-        goto release;
+        break;
     }
 #endif
 
@@ -552,7 +560,7 @@
     status = AudioSystem::getSamplingRate(input, &afSampleRate);
     if (status != NO_ERROR) {
         ALOGE("getSamplingRate(input=%d) status %d", input, status);
-        goto release;
+        break;
     }
     if (mSampleRate == 0) {
         mSampleRate = afSampleRate;
@@ -572,7 +580,10 @@
             ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, "
                 "track %u Hz, input %u Hz",
                 mTransfer, mSampleRate, afSampleRate);
-            mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
+            mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
+                    AUDIO_INPUT_FLAG_RAW));
+            AudioSystem::releaseInput(input, mSessionId);
+            continue;   // retry
         }
     }
 
@@ -593,7 +604,7 @@
 
     size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
                                 // but we will still need the original value also
-    int originalSessionId = mSessionId;
+    audio_session_t originalSessionId = mSessionId;
 
     sp<IMemory> iMem;           // for cblk
     sp<IMemory> bufferMem;
@@ -616,13 +627,26 @@
 
     if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create record track, status: %d", status);
-        goto release;
+        break;
     }
     ALOG_ASSERT(record != 0);
 
     // AudioFlinger now owns the reference to the I/O handle,
     // so we are no longer responsible for releasing it.
 
+    mAwaitBoost = false;
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        if (trackFlags & IAudioFlinger::TRACK_FAST) {
+            ALOGI("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
+            mAwaitBoost = true;
+        } else {
+            ALOGW("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
+            mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
+                    AUDIO_INPUT_FLAG_RAW));
+            continue;   // retry
+        }
+    }
+
     if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
@@ -665,17 +689,6 @@
     }
     frameCount = temp;
 
-    mAwaitBoost = false;
-    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
-        if (trackFlags & IAudioFlinger::TRACK_FAST) {
-            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
-            mAwaitBoost = true;
-        } else {
-            ALOGW("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
-            mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
-        }
-    }
-
     // Make sure that application is notified with sufficient margin before overrun.
     // The computation is done on server side.
     if (mNotificationFramesReq > 0 && notificationFrames != mNotificationFramesReq) {
@@ -708,10 +721,13 @@
     }
 
     return NO_ERROR;
+
+    // End of retry loop.
+    // The lack of indentation is deliberate, to reduce code churn and ease merges.
     }
 
-release:
-    AudioSystem::releaseInput(input, (audio_session_t)mSessionId);
+// Arrive here on error, via a break
+    AudioSystem::releaseInput(input, mSessionId);
     if (status == NO_ERROR) {
         status = NO_INIT;
     }
@@ -1153,7 +1169,7 @@
         if (mActive) {
             // callback thread or sync event hasn't changed
             // FIXME this fails if we have a new AudioFlinger instance
-            result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
+            result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE);
         }
         mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset.
     }
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index c55a90b..3a5dee6 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -399,7 +399,7 @@
     return af->newAudioUniqueId(use);
 }
 
-void AudioSystem::acquireAudioSessionId(int audioSession, pid_t pid)
+void AudioSystem::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
@@ -407,7 +407,7 @@
     }
 }
 
-void AudioSystem::releaseAudioSessionId(int audioSession, pid_t pid)
+void AudioSystem::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
@@ -902,7 +902,7 @@
 status_t AudioSystem::registerEffect(const effect_descriptor_t *desc,
                                 audio_io_handle_t io,
                                 uint32_t strategy,
-                                int session,
+                                audio_session_t session,
                                 int id)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index ab18c27..ef0ccc2 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -184,7 +184,7 @@
         callback_t cbf,
         void* user,
         uint32_t notificationFrames,
-        int sessionId,
+        audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
         int uid,
@@ -213,7 +213,7 @@
         callback_t cbf,
         void* user,
         uint32_t notificationFrames,
-        int sessionId,
+        audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
         int uid,
@@ -272,7 +272,7 @@
         uint32_t notificationFrames,
         const sp<IMemory>& sharedBuffer,
         bool threadCanCallJava,
-        int sessionId,
+        audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
         int uid,
@@ -438,7 +438,7 @@
     mNotificationFramesReq = notificationFrames;
     mNotificationFramesAct = 0;
     if (sessionId == AUDIO_SESSION_ALLOCATE) {
-        mSessionId = AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
+        mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
     } else {
         mSessionId = sessionId;
     }
@@ -860,7 +860,7 @@
     if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
         return NO_INIT;
     }
-    return mProxy->getBufferSizeInFrames();
+    return (ssize_t) mProxy->getBufferSizeInFrames();
 }
 
 ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
@@ -873,10 +873,7 @@
     if (!audio_is_linear_pcm(mFormat)) {
         return INVALID_OPERATION;
     }
-    // TODO also need to inform the server side (through mAudioTrack) that
-    // the buffer count is reduced, otherwise the track may never start
-    // because the server thinks it is never filled.
-    return mProxy->setBufferSizeInFrames(bufferSizeInFrames);
+    return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
 }
 
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -1158,7 +1155,7 @@
 
     status_t status;
     status = AudioSystem::getOutputForAttr(attr, &output,
-                                           (audio_session_t)mSessionId, &streamType, mClientUid,
+                                           mSessionId, &streamType, mClientUid,
                                            mSampleRate, mFormat, mChannelMask,
                                            mFlags, mSelectedDeviceId, mOffloadInfo);
 
@@ -1291,7 +1288,7 @@
 
     size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
                                 // but we will still need the original value also
-    int originalSessionId = mSessionId;
+    audio_session_t originalSessionId = mSessionId;
     sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
                                                       mSampleRate,
                                                       mFormat,
@@ -1451,7 +1448,7 @@
     }
 
 release:
-    AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId);
+    AudioSystem::releaseOutput(output, streamType, mSessionId);
     if (status == NO_ERROR) {
         status = NO_INIT;
     }
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 6b6865b..e5d6fc3 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -46,8 +46,10 @@
 }
 
 audio_track_cblk_t::audio_track_cblk_t()
-    : mServer(0), mFutex(0), mMinimum(0),
-    mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0), mFlags(0)
+    : mServer(0), mFutex(0), mMinimum(0)
+    , mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0)
+    , mBufferSizeInFrames(0)
+    , mFlags(0)
 {
     memset(&u, 0, sizeof(u));
 }
@@ -67,10 +69,10 @@
 ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
         size_t frameSize, bool isOut, bool clientInServer)
     : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer)
-    , mBufferSizeInFrames(frameCount)
     , mEpoch(0)
     , mTimestampObserver(&cblk->mExtendedTimestampQueue)
 {
+    setBufferSizeInFrames(frameCount);
 }
 
 const struct timespec ClientProxy::kForever = {INT_MAX /*tv_sec*/, 0 /*tv_nsec*/};
@@ -84,6 +86,26 @@
 // order of minutes.
 #define MAX_SEC    5
 
+uint32_t ClientProxy::setBufferSizeInFrames(uint32_t size)
+{
+    // TODO set minimum to 2X the fast mixer buffer size.
+    // The minimum should be  greater than zero and less than the size
+    // at which underruns will occur.
+    const uint32_t minimum = 128 * 2; // arbitrary
+    const uint32_t maximum = frameCount();
+    uint32_t clippedSize = size;
+    if (clippedSize < minimum) {
+        clippedSize = minimum;
+    } else if (clippedSize > maximum) {
+        clippedSize = maximum;
+    }
+    // for server to read
+    android_atomic_release_store(clippedSize, (int32_t *)&mCblk->mBufferSizeInFrames);
+    // for client to read
+    mBufferSizeInFrames = clippedSize;
+    return clippedSize;
+}
+
 status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested,
         struct timespec *elapsed)
 {
@@ -179,7 +201,7 @@
         // The calculation for avail can go negative if the buffer size
         // is suddenly dropped below the amount already in the buffer.
         // So use a signed calculation to prevent a numeric overflow abort.
-        ssize_t adjustableSize = (ssize_t) mBufferSizeInFrames;
+        ssize_t adjustableSize = (ssize_t) getBufferSizeInFrames();
         ssize_t avail =  (mIsOut) ? adjustableSize - filled : filled;
         if (avail < 0) {
             avail = 0;
@@ -362,20 +384,6 @@
             (mFrameCountP2 - 1);
 }
 
-size_t ClientProxy::setBufferSizeInFrames(size_t size)
-{
-    // TODO set minimum to 2X the fast mixer buffer size.
-    size_t minimum = 128 * 2; // arbitrary
-    size_t maximum = frameCount();
-    if (size < minimum) {
-        size = minimum;
-    } else if (size > maximum) {
-        size = maximum;
-    }
-    mBufferSizeInFrames = size;
-    return size;
-}
-
 // ---------------------------------------------------------------------------
 
 void AudioTrackClientProxy::flush()
@@ -607,6 +615,7 @@
       mAvailToClient(0), mFlush(0), mReleased(0)
     , mTimestampMutator(&cblk->mExtendedTimestampQueue)
 {
+    cblk->mBufferSizeInFrames = frameCount;
 }
 
 status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 6da4524..042eac5 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -104,7 +104,7 @@
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 int clientUid,
                                 status_t *status)
     {
@@ -128,7 +128,7 @@
         }
         data.writeInt32((int32_t) output);
         data.writeInt32((int32_t) tid);
-        int lSessionId = AUDIO_SESSION_ALLOCATE;
+        audio_session_t lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
@@ -146,7 +146,7 @@
             if (flags != NULL) {
                 *flags = lFlags;
             }
-            lSessionId = reply.readInt32();
+            lSessionId = (audio_session_t) reply.readInt32();
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
@@ -180,7 +180,7 @@
                                 track_flags_t *flags,
                                 pid_t tid,
                                 int clientUid,
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 size_t *notificationFrames,
                                 sp<IMemory>& cblk,
                                 sp<IMemory>& buffers,
@@ -200,7 +200,7 @@
         data.writeInt32(lFlags);
         data.writeInt32((int32_t) tid);
         data.writeInt32((int32_t) clientUid);
-        int lSessionId = AUDIO_SESSION_ALLOCATE;
+        audio_session_t lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
@@ -220,7 +220,7 @@
             if (flags != NULL) {
                 *flags = lFlags;
             }
-            lSessionId = reply.readInt32();
+            lSessionId = (audio_session_t) reply.readInt32();
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
@@ -625,7 +625,7 @@
         return id;
     }
 
-    virtual void acquireAudioSessionId(int audioSession, int pid)
+    virtual void acquireAudioSessionId(audio_session_t audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -634,7 +634,7 @@
         remote()->transact(ACQUIRE_AUDIO_SESSION_ID, data, &reply);
     }
 
-    virtual void releaseAudioSessionId(int audioSession, int pid)
+    virtual void releaseAudioSessionId(audio_session_t audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -707,7 +707,7 @@
                                     const sp<IEffectClient>& client,
                                     int32_t priority,
                                     audio_io_handle_t output,
-                                    int sessionId,
+                                    audio_session_t sessionId,
                                     const String16& opPackageName,
                                     status_t *status,
                                     int *id,
@@ -754,7 +754,7 @@
         return effect;
     }
 
-    virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+    virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
             audio_io_handle_t dstOutput)
     {
         Parcel data, reply;
@@ -936,7 +936,7 @@
             }
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
             pid_t tid = (pid_t) data.readInt32();
-            int sessionId = data.readInt32();
+            audio_session_t sessionId = (audio_session_t) data.readInt32();
             int clientUid = data.readInt32();
             status_t status = NO_ERROR;
             sp<IAudioTrack> track;
@@ -969,7 +969,7 @@
             track_flags_t flags = (track_flags_t) data.readInt32();
             pid_t tid = (pid_t) data.readInt32();
             int clientUid = data.readInt32();
-            int sessionId = data.readInt32();
+            audio_session_t sessionId = (audio_session_t) data.readInt32();
             size_t notificationFrames = data.readInt64();
             sp<IMemory> cblk;
             sp<IMemory> buffers;
@@ -1214,14 +1214,14 @@
         } break;
         case ACQUIRE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int audioSession = data.readInt32();
+            audio_session_t audioSession = (audio_session_t) data.readInt32();
             int pid = data.readInt32();
             acquireAudioSessionId(audioSession, pid);
             return NO_ERROR;
         } break;
         case RELEASE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int audioSession = data.readInt32();
+            audio_session_t audioSession = (audio_session_t) data.readInt32();
             int pid = data.readInt32();
             releaseAudioSessionId(audioSession, pid);
             return NO_ERROR;
@@ -1267,7 +1267,7 @@
             sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
             int32_t priority = data.readInt32();
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            int sessionId = data.readInt32();
+            audio_session_t sessionId = (audio_session_t) data.readInt32();
             const String16 opPackageName = data.readString16();
             status_t status = NO_ERROR;
             int id = 0;
@@ -1284,7 +1284,7 @@
         } break;
         case MOVE_EFFECTS: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int session = data.readInt32();
+            audio_session_t session = (audio_session_t) data.readInt32();
             audio_io_handle_t srcOutput = (audio_io_handle_t) data.readInt32();
             audio_io_handle_t dstOutput = (audio_io_handle_t) data.readInt32();
             reply->writeInt32(moveEffects(session, srcOutput, dstOutput));
@@ -1411,7 +1411,7 @@
         } break;
         case GET_AUDIO_HW_SYNC: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32(getAudioHwSyncForSession((audio_session_t)data.readInt32()));
+            reply->writeInt32(getAudioHwSyncForSession((audio_session_t) data.readInt32()));
             return NO_ERROR;
         } break;
         case SYSTEM_READY: {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index c95d4c4..16e8f11 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -247,7 +247,7 @@
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
         data.writeInt32(output);
         data.writeInt32((int32_t) stream);
-        data.writeInt32((int32_t)session);
+        data.writeInt32((int32_t) session);
         remote()->transact(START_OUTPUT, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
@@ -260,7 +260,7 @@
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
         data.writeInt32(output);
         data.writeInt32((int32_t) stream);
-        data.writeInt32((int32_t)session);
+        data.writeInt32((int32_t) session);
         remote()->transact(STOP_OUTPUT, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
@@ -420,7 +420,7 @@
     virtual status_t registerEffect(const effect_descriptor_t *desc,
                                         audio_io_handle_t io,
                                         uint32_t strategy,
-                                        int session,
+                                        audio_session_t session,
                                         int id)
     {
         Parcel data, reply;
@@ -482,7 +482,7 @@
         return reply.readInt32();
     }
 
-    virtual status_t queryDefaultPreProcessing(int audioSession,
+    virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                                effect_descriptor_t *descriptors,
                                                uint32_t *count)
     {
@@ -1065,7 +1065,7 @@
             data.read(&desc, sizeof(effect_descriptor_t));
             audio_io_handle_t io = data.readInt32();
             uint32_t strategy = data.readInt32();
-            int session = data.readInt32();
+            audio_session_t session = (audio_session_t) data.readInt32();
             int id = data.readInt32();
             reply->writeInt32(static_cast <int32_t>(registerEffect(&desc,
                                                                    io,
@@ -1115,7 +1115,7 @@
 
         case QUERY_DEFAULT_PRE_PROCESSING: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            int audioSession = data.readInt32();
+            audio_session_t audioSession = (audio_session_t) data.readInt32();
             uint32_t count = data.readInt32();
             if (count > AudioEffect::kMaxPreProcessing) {
                 count = AudioEffect::kMaxPreProcessing;
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 9d80753..ae66436 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -42,7 +42,7 @@
     {
     }
 
-    virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession)
+    virtual status_t start(int /*AudioSystem::sync_event_t*/ event, audio_session_t triggerSession)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
@@ -77,7 +77,7 @@
         case START: {
             CHECK_INTERFACE(IAudioRecord, data, reply);
             int /*AudioSystem::sync_event_t*/ event = data.readInt32();
-            int triggerSession = data.readInt32();
+            audio_session_t triggerSession = (audio_session_t) data.readInt32();
             reply->writeInt32(start(event, triggerSession));
             return NO_ERROR;
         } break;
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index afc94ab..27b9edd 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -68,7 +68,7 @@
     }
 
     virtual sp<IMediaPlayer> create(
-            const sp<IMediaPlayerClient>& client, int audioSessionId) {
+            const sp<IMediaPlayerClient>& client, audio_session_t audioSessionId) {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
         data.writeStrongBinder(IInterface::asBinder(client));
@@ -161,7 +161,7 @@
             CHECK_INTERFACE(IMediaPlayerService, data, reply);
             sp<IMediaPlayerClient> client =
                 interface_cast<IMediaPlayerClient>(data.readStrongBinder());
-            int audioSessionId = data.readInt32();
+            audio_session_t audioSessionId = (audio_session_t) data.readInt32();
             sp<IMediaPlayer> player = create(client, audioSessionId);
             reply->writeStrongBinder(IInterface::asBinder(player));
             return NO_ERROR;
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index f5c1b1f..31e310b 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -38,7 +38,7 @@
          int32_t priority,
          effect_callback_t cbf,
          void* user,
-         int sessionId)
+         audio_session_t sessionId)
     :   AudioEffect(SL_IID_VISUALIZATION, opPackageName, NULL, priority, cbf, user, sessionId),
         mCaptureRate(CAPTURE_RATE_DEF),
         mCaptureSize(CAPTURE_SIZE_DEF),
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 7e502af..53b6df6 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -63,7 +63,7 @@
     mLeftVolume = mRightVolume = 1.0;
     mVideoWidth = mVideoHeight = 0;
     mLockThreadId = 0;
-    mAudioSessionId = AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
+    mAudioSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
     AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);
     mSendLevel = 0;
     mRetransmitEndpointValid = false;
@@ -648,7 +648,7 @@
     return OK;
 }
 
-status_t MediaPlayer::setAudioSessionId(int sessionId)
+status_t MediaPlayer::setAudioSessionId(audio_session_t sessionId)
 {
     ALOGV("MediaPlayer::setAudioSessionId(%d)", sessionId);
     Mutex::Autolock _l(mLock);
@@ -667,7 +667,7 @@
     return NO_ERROR;
 }
 
-int MediaPlayer::getAudioSessionId()
+audio_session_t MediaPlayer::getAudioSessionId()
 {
     Mutex::Autolock _l(mLock);
     return mAudioSessionId;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index bb24403..0025660 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -321,7 +321,7 @@
 }
 
 sp<IMediaPlayer> MediaPlayerService::create(const sp<IMediaPlayerClient>& client,
-        int audioSessionId)
+        audio_session_t audioSessionId)
 {
     pid_t pid = IPCThreadState::self()->getCallingPid();
     int32_t connId = android_atomic_inc(&mNextConnId);
@@ -556,7 +556,7 @@
 MediaPlayerService::Client::Client(
         const sp<MediaPlayerService>& service, pid_t pid,
         int32_t connId, const sp<IMediaPlayerClient>& client,
-        int audioSessionId, uid_t uid)
+        audio_session_t audioSessionId, uid_t uid)
 {
     ALOGV("Client(%d) constructor", connId);
     mPid = pid;
@@ -1332,7 +1332,7 @@
 
 #undef LOG_TAG
 #define LOG_TAG "AudioSink"
-MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid,
+MediaPlayerService::AudioOutput::AudioOutput(audio_session_t sessionId, int uid, int pid,
         const audio_attributes_t* attr)
     : mCallback(NULL),
       mCallbackCookie(NULL),
@@ -2111,7 +2111,7 @@
     data->unlock();
 }
 
-int MediaPlayerService::AudioOutput::getSessionId() const
+audio_session_t MediaPlayerService::AudioOutput::getSessionId() const
 {
     Mutex::Autolock lock(mLock);
     return mSessionId;
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index bd98ef1..1dd2ddd 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -75,7 +75,7 @@
         class CallbackData;
 
      public:
-                                AudioOutput(int sessionId, int uid, int pid,
+                                AudioOutput(audio_session_t sessionId, int uid, int pid,
                                         const audio_attributes_t * attr);
         virtual                 ~AudioOutput();
 
@@ -90,7 +90,7 @@
         virtual status_t        getTimestamp(AudioTimestamp &ts) const;
         virtual int64_t         getPlayedOutDurationUs(int64_t nowUs) const;
         virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
-        virtual int             getSessionId() const;
+        virtual audio_session_t getSessionId() const;
         virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
@@ -150,7 +150,7 @@
         uint32_t                mSampleRateHz; // sample rate of the content, as set in open()
         float                   mMsecsPerFrame;
         size_t                  mFrameSize;
-        int                     mSessionId;
+        audio_session_t         mSessionId;
         int                     mUid;
         int                     mPid;
         float                   mSendLevel;
@@ -214,7 +214,8 @@
     void    removeMediaRecorderClient(wp<MediaRecorderClient> client);
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever();
 
-    virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client, int audioSessionId);
+    virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client,
+                                       audio_session_t audioSessionId);
 
     virtual sp<IMediaCodecList> getCodecList() const;
     virtual sp<IOMX>            getOMX();
@@ -332,7 +333,7 @@
                 pid_t           pid() const { return mPid; }
         virtual status_t        dump(int fd, const Vector<String16>& args);
 
-                int             getAudioSessionId() { return mAudioSessionId; }
+                audio_session_t getAudioSessionId() { return mAudioSessionId; }
 
     private:
         friend class MediaPlayerService;
@@ -340,7 +341,7 @@
                                         pid_t pid,
                                         int32_t connId,
                                         const sp<IMediaPlayerClient>& client,
-                                        int audioSessionId,
+                                        audio_session_t audioSessionId,
                                         uid_t uid);
                                 Client();
         virtual                 ~Client();
@@ -375,7 +376,7 @@
                     status_t                    mStatus;
                     bool                        mLoop;
                     int32_t                     mConnId;
-                    int                         mAudioSessionId;
+                    audio_session_t             mAudioSessionId;
                     audio_attributes_t *        mAudioAttributes;
                     uid_t                       mUID;
                     sp<ANativeWindow>           mConnectedWindow;
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 1b248db..78eb3b0 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -1761,12 +1761,16 @@
 
     // 30 ms buffer to avoid timestamp overlap
     mTotalPausedDurationUs += (systemTime() / 1000) - mPauseStartTimeUs - 30000;
+    double timeOffset = -mTotalPausedDurationUs;
+    if (mCaptureFpsEnable) {
+        timeOffset *= mCaptureFps / mFrameRate;
+    }
     if (mAudioEncoderSource != NULL) {
-        mAudioEncoderSource->setInputBufferTimeOffset(-mTotalPausedDurationUs);
+        mAudioEncoderSource->setInputBufferTimeOffset((int64_t)timeOffset);
         mAudioEncoderSource->start();
     }
     if (mVideoEncoderSource != NULL) {
-        mVideoEncoderSource->setInputBufferTimeOffset(-mTotalPausedDurationUs);
+        mVideoEncoderSource->setInputBufferTimeOffset((int64_t)timeOffset);
         mVideoEncoderSource->start();
     }
     mPauseStartTimeUs = 0;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index e0fe7e3..520f599 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2555,6 +2555,7 @@
             : OMX_AUDIO_AACStreamFormatMP4FF;
 
     OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE presentation;
+    InitOMXParams(&presentation);
     presentation.nMaxOutputChannels = maxOutputChannelCount;
     presentation.nDrcCut = drc.drcCut;
     presentation.nDrcBoost = drc.drcBoost;
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 59f839c..cb974ae 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -758,7 +758,7 @@
     mStartTimeUs = 0;
     mNumInputBuffers = 0;
     mEncoderFormat = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
-    mEncoderDataSpace = HAL_DATASPACE_BT709;
+    mEncoderDataSpace = HAL_DATASPACE_V0_BT709;
 
     if (meta) {
         int64_t startTimeUs;
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 7c03886..58bfa67 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -2302,6 +2302,7 @@
     const bool hasMultipleTracks = (mOwner->numTracks() > 1);
     int64_t chunkTimestampUs = 0;
     int32_t nChunks = 0;
+    int32_t nActualFrames = 0;        // frames containing non-CSD data (non-0 length)
     int32_t nZeroLengthFrames = 0;
     int64_t lastTimestampUs = 0;      // Previous sample time stamp
     int64_t lastDurationUs = 0;       // Between the previous two samples
@@ -2354,24 +2355,31 @@
         int32_t isCodecConfig;
         if (buffer->meta_data()->findInt32(kKeyIsCodecConfig, &isCodecConfig)
                 && isCodecConfig) {
-            CHECK(!mGotAllCodecSpecificData);
-            mMeta = mSource->getFormat(); // get output format after format change
+            // if config format (at track addition) already had CSD, keep that
+            // UNLESS we have not received any frames yet.
+            // TODO: for now the entire CSD has to come in one frame for encoders, even though
+            // they need to be spread out for decoders.
+            if (mGotAllCodecSpecificData && nActualFrames > 0) {
+                ALOGI("ignoring additional CSD for video track after first frame");
+            } else {
+                mMeta = mSource->getFormat(); // get output format after format change
 
-            if (mIsAvc) {
-                status_t err = makeAVCCodecSpecificData(
-                        (const uint8_t *)buffer->data()
-                            + buffer->range_offset(),
-                        buffer->range_length());
-                CHECK_EQ((status_t)OK, err);
-            } else if (mIsHevc) {
-                status_t err = makeHEVCCodecSpecificData(
-                        (const uint8_t *)buffer->data()
-                            + buffer->range_offset(),
-                        buffer->range_length());
-                CHECK_EQ((status_t)OK, err);
-            } else if (mIsMPEG4) {
-                copyCodecSpecificData((const uint8_t *)buffer->data() + buffer->range_offset(),
-                        buffer->range_length());
+                if (mIsAvc) {
+                    status_t err = makeAVCCodecSpecificData(
+                            (const uint8_t *)buffer->data()
+                                + buffer->range_offset(),
+                            buffer->range_length());
+                    CHECK_EQ((status_t)OK, err);
+                } else if (mIsHevc) {
+                    status_t err = makeHEVCCodecSpecificData(
+                            (const uint8_t *)buffer->data()
+                                + buffer->range_offset(),
+                            buffer->range_length());
+                    CHECK_EQ((status_t)OK, err);
+                } else if (mIsMPEG4) {
+                    copyCodecSpecificData((const uint8_t *)buffer->data() + buffer->range_offset(),
+                            buffer->range_length());
+                }
             }
 
             buffer->release();
@@ -2381,6 +2389,8 @@
             continue;
         }
 
+        ++nActualFrames;
+
         // Make a deep copy of the MediaBuffer and Metadata and release
         // the original as soon as we can
         MediaBuffer *copy = new MediaBuffer(buffer->range_length());
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index b5f7b12..0a052d2 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -42,7 +42,7 @@
 
 const int32_t kDefaultSwVideoEncoderFormat = HAL_PIXEL_FORMAT_YCbCr_420_888;
 const int32_t kDefaultHwVideoEncoderFormat = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
-const int32_t kDefaultVideoEncoderDataSpace = HAL_DATASPACE_BT709;
+const int32_t kDefaultVideoEncoderDataSpace = HAL_DATASPACE_V0_BT709;
 
 const int kStopTimeoutUs = 300000; // allow 1 sec for shutting down encoder
 
diff --git a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
index db3e058..8165f69 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
@@ -51,11 +51,11 @@
 #define vo_shr_r(var1, var2)        ((var1+((Word16)(1L<<(var2-1))))>>var2)
 #define vo_sub(a,b)         (a - b)
 #define vo_L_deposit_h(a)       ((Word32)((a) << 16))
-#define vo_round(a)         ((a + 0x00008000) >> 16)
+#define vo_round(a)         ((((a) >> 15) + 1) >> 1)
 #define vo_extract_l(a)         ((Word16)(a))
 #define vo_L_add(a,b)           (a + b)
 #define vo_L_sub(a,b)           (a - b)
-#define vo_mult_r(a,b)          ((( a * b ) + 0x4000 ) >> 15 )
+#define vo_mult_r(a,b)          (((( a * b ) >> 14) + 1 ) >> 1 )
 #define vo_negate(a)                (-a)
 #define vo_L_shr_r(L_var1, var2)        ((L_var1+((Word32)(1L<<(var2-1))))>>var2)
 
diff --git a/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c b/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
index 8bf15ea..4d877f1 100644
--- a/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
+++ b/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
@@ -267,13 +267,13 @@
 
     for (i = 0; i < L_SUBFR/4; i++)
     {
-        s = (k_cn* (*p0++))+(k_dn * (*p1++));
+        s = L_add((k_cn* (*p0++)), (k_dn * (*p1++)));
         *p2++ = s >> 7;
-        s = (k_cn* (*p0++))+(k_dn * (*p1++));
+        s = L_add((k_cn* (*p0++)), (k_dn * (*p1++)));
         *p2++ = s >> 7;
-        s = (k_cn* (*p0++))+(k_dn * (*p1++));
+        s = L_add((k_cn* (*p0++)), (k_dn * (*p1++)));
         *p2++ = s >> 7;
-        s = (k_cn* (*p0++))+(k_dn * (*p1++));
+        s = L_add((k_cn* (*p0++)), (k_dn * (*p1++)));
         *p2++ = s >> 7;
     }
 
@@ -342,7 +342,7 @@
     {
         *h++ = 0;
         *h_inv++ = 0;
-        L_tmp += (H[i] * H[i]) << 1;
+        L_tmp = L_add(L_tmp, (H[i] * H[i]) << 1);
     }
     /* scale h[] down (/2) when energy of h[] is high with many pulses used */
     val = extract_h(L_tmp);
@@ -386,16 +386,16 @@
     cor = 0x00008000L;                             /* for rounding */
     for (i = 0; i < NB_POS; i++)
     {
-        cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h1)));
         ptr_h1++;
         *p3-- = extract_h(cor);
-        cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h1)));
         ptr_h1++;
         *p2-- = extract_h(cor);
-        cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h1)));
         ptr_h1++;
         *p1-- = extract_h(cor);
-        cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h1)));
         ptr_h1++;
         *p0-- = extract_h(cor);
     }
@@ -425,19 +425,19 @@
 
         for (i = k + 1; i < NB_POS; i++)
         {
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p3 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p2 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p1 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p0 = extract_h(cor);
@@ -447,15 +447,15 @@
             p1 -= (NB_POS + 1);
             p0 -= (NB_POS + 1);
         }
-        cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
         ptr_h1++;
         ptr_h2++;
         *p3 = extract_h(cor);
-        cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
         ptr_h1++;
         ptr_h2++;
         *p2 = extract_h(cor);
-        cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
         ptr_h1++;
         ptr_h2++;
         *p1 = extract_h(cor);
@@ -482,19 +482,19 @@
 
         for (i = k + 1; i < NB_POS; i++)
         {
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p3 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p2 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p1 = extract_h(cor);
-            cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+            cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
             ptr_h1++;
             ptr_h2++;
             *p0 = extract_h(cor);
@@ -504,7 +504,7 @@
             p1 -= (NB_POS + 1);
             p0 -= (NB_POS + 1);
         }
-        cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+        cor = L_add(cor, vo_L_mult((*ptr_h1), (*ptr_h2)));
         ptr_h1++;
         ptr_h2++;
         *p3 = extract_h(cor);
@@ -698,7 +698,7 @@
         }
         /* memorise the best codevector */
         ps = vo_mult(ps, ps);
-        s = vo_L_msu(vo_L_mult(alpk, ps), psk, alp);
+        s = L_sub(vo_L_mult(alpk, ps), vo_L_mult(psk, alp));
         if (s > 0)
         {
             psk = ps;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 5980b58..d07ca85 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -552,7 +552,7 @@
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output,
         pid_t tid,
-        int *sessionId,
+        audio_session_t *sessionId,
         int clientUid,
         status_t *status)
 {
@@ -560,7 +560,7 @@
     sp<TrackHandle> trackHandle;
     sp<Client> client;
     status_t lStatus;
-    int lSessionId;
+    audio_session_t lSessionId;
 
     // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
     // but if someone uses binder directly they could bypass that and cause us to crash
@@ -631,7 +631,7 @@
             }
         } else {
             // if no audio session id is provided, create one here
-            lSessionId = nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
+            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
@@ -667,7 +667,7 @@
             }
         }
 
-        setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId);
+        setAudioHwSyncForSession_l(thread, lSessionId);
     }
 
     if (lStatus != NO_ERROR) {
@@ -1075,10 +1075,10 @@
                     audio_devices_t device = thread->inDevice();
                     bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
                     // collect all of the thread's session IDs
-                    KeyedVector<int, bool> ids = thread->sessionIds();
+                    KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
                     // suspend effects associated with those session IDs
                     for (size_t j = 0; j < ids.size(); ++j) {
-                        int sessionId = ids.keyAt(j);
+                        audio_session_t sessionId = ids.keyAt(j);
                         thread->setEffectSuspended(FX_IID_AEC,
                                                    suspend,
                                                    sessionId);
@@ -1344,7 +1344,8 @@
 }
 
 // getEffectThread_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
+sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
+        int EffectId)
 {
     sp<PlaybackThread> thread;
 
@@ -1423,7 +1424,7 @@
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         int clientUid,
-        int *sessionId,
+        audio_session_t *sessionId,
         size_t *notificationFrames,
         sp<IMemory>& cblk,
         sp<IMemory>& buffers,
@@ -1433,7 +1434,7 @@
     sp<RecordHandle> recordHandle;
     sp<Client> client;
     status_t lStatus;
-    int lSessionId;
+    audio_session_t lSessionId;
 
     cblk.clear();
     buffers.clear();
@@ -1493,7 +1494,7 @@
             lSessionId = *sessionId;
         } else {
             // if no audio session id is provided, create one here
-            lSessionId = nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
+            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
@@ -1508,7 +1509,7 @@
         if (lStatus == NO_ERROR) {
             // Check if one effect chain was awaiting for an AudioRecord to be created on this
             // session and move it to this thread.
-            sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId);
+            sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId);
             if (chain != 0) {
                 Mutex::Autolock _l(thread->mLock);
                 thread->addEffectChain_l(chain);
@@ -2270,7 +2271,7 @@
     return nextUniqueId(use);
 }
 
-void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
+void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
@@ -2304,7 +2305,7 @@
     ALOGV(" added new entry for %d", audioSession);
 }
 
-void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
+void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
@@ -2462,8 +2463,8 @@
 }
 
 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
-                                    int triggerSession,
-                                    int listenerSession,
+                                    audio_session_t triggerSession,
+                                    audio_session_t listenerSession,
                                     sync_event_callback_t callBack,
                                     wp<RefBase> cookie)
 {
@@ -2523,7 +2524,7 @@
         const sp<IEffectClient>& effectClient,
         int32_t priority,
         audio_io_handle_t io,
-        int sessionId,
+        audio_session_t sessionId,
         const String16& opPackageName,
         status_t *status,
         int *id,
@@ -2684,7 +2685,7 @@
         } else {
             // Check if one effect chain was awaiting for an effect to be created on this
             // session and used it instead of creating a new one.
-            sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId);
+            sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
             if (chain != 0) {
                 Mutex::Autolock _l(thread->mLock);
                 thread->addEffectChain_l(chain);
@@ -2711,7 +2712,7 @@
     return handle;
 }
 
-status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
+status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
         audio_io_handle_t dstOutput)
 {
     ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
@@ -2738,7 +2739,7 @@
 }
 
 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held
-status_t AudioFlinger::moveEffectChain_l(int sessionId,
+status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
                                    AudioFlinger::PlaybackThread *srcThread,
                                    AudioFlinger::PlaybackThread *dstThread,
                                    bool reRegister)
@@ -2864,7 +2865,7 @@
 
 status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
 {
-    audio_session_t session = (audio_session_t)chain->sessionId();
+    audio_session_t session = chain->sessionId();
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
     ALOGV("putOrphanEffectChain_l session %d index %d", session, index);
     if (index >= 0) {
@@ -2890,7 +2891,7 @@
 bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
 {
     Mutex::Autolock _l(mLock);
-    audio_session_t session = (audio_session_t)effect->sessionId();
+    audio_session_t session = effect->sessionId();
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
     ALOGV("updateOrphanEffectChains session %d index %d", session, index);
     if (index >= 0) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fb1ece4..f11fd1c 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -108,7 +108,7 @@
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 int clientUid,
                                 status_t *status /*non-NULL*/);
 
@@ -122,7 +122,7 @@
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
                                 int clientUid,
-                                int *sessionId,
+                                audio_session_t *sessionId,
                                 size_t *notificationFrames,
                                 sp<IMemory>& cblk,
                                 sp<IMemory>& buffers,
@@ -198,9 +198,9 @@
 
     virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use);
 
-    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
+    virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid);
 
-    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
+    virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid);
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const;
 
@@ -214,13 +214,13 @@
                         const sp<IEffectClient>& effectClient,
                         int32_t priority,
                         audio_io_handle_t io,
-                        int sessionId,
+                        audio_session_t sessionId,
                         const String16& opPackageName,
                         status_t *status /*non-NULL*/,
                         int *id,
                         int *enabled);
 
-    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
+    virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
                         audio_io_handle_t dstOutput);
 
     virtual audio_module_handle_t loadHwModule(const char *name);
@@ -283,8 +283,8 @@
     class SyncEvent : public RefBase {
     public:
         SyncEvent(AudioSystem::sync_event_t type,
-                  int triggerSession,
-                  int listenerSession,
+                  audio_session_t triggerSession,
+                  audio_session_t listenerSession,
                   sync_event_callback_t callBack,
                   wp<RefBase> cookie)
         : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
@@ -297,22 +297,22 @@
         bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
         void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
         AudioSystem::sync_event_t type() const { return mType; }
-        int triggerSession() const { return mTriggerSession; }
-        int listenerSession() const { return mListenerSession; }
+        audio_session_t triggerSession() const { return mTriggerSession; }
+        audio_session_t listenerSession() const { return mListenerSession; }
         wp<RefBase> cookie() const { return mCookie; }
 
     private:
           const AudioSystem::sync_event_t mType;
-          const int mTriggerSession;
-          const int mListenerSession;
+          const audio_session_t mTriggerSession;
+          const audio_session_t mListenerSession;
           sync_event_callback_t mCallback;
           const wp<RefBase> mCookie;
           mutable Mutex mLock;
     };
 
     sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
-                                        int triggerSession,
-                                        int listenerSession,
+                                        audio_session_t triggerSession,
+                                        audio_session_t listenerSession,
                                         sync_event_callback_t callBack,
                                         wp<RefBase> cookie);
 
@@ -505,7 +505,8 @@
     public:
         RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
         virtual             ~RecordHandle();
-        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
+        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event,
+                audio_session_t triggerSession);
         virtual void        stop();
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
@@ -560,7 +561,7 @@
               //       or by returning a non-unique ID.
               audio_unique_id_t nextUniqueId(audio_unique_id_use_t use);
 
-              status_t moveEffectChain_l(int sessionId,
+              status_t moveEffectChain_l(audio_session_t sessionId,
                                      PlaybackThread *srcThread,
                                      PlaybackThread *dstThread,
                                      bool reRegister);
@@ -569,7 +570,7 @@
               PlaybackThread *primaryPlaybackThread_l() const;
               audio_devices_t primaryOutputDevice_l() const;
 
-              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
+              sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId);
 
 
                 void        removeClient_l(pid_t pid);
@@ -612,9 +613,9 @@
 
     // for mAudioSessionRefs only
     struct AudioSessionRef {
-        AudioSessionRef(int sessionid, pid_t pid) :
+        AudioSessionRef(audio_session_t sessionid, pid_t pid) :
             mSessionid(sessionid), mPid(pid), mCnt(1) {}
-        const int   mSessionid;
+        const audio_session_t mSessionid;
         const pid_t mPid;
         int         mCnt;
     };
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 949c91d..060ffe9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -59,7 +59,7 @@
                                         const wp<AudioFlinger::EffectChain>& chain,
                                         effect_descriptor_t *desc,
                                         int id,
-                                        int sessionId)
+                                        audio_session_t sessionId)
     : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX),
       mThread(thread), mChain(chain), mId(id), mSessionId(sessionId),
       mDescriptor(*desc),
@@ -1341,7 +1341,7 @@
 #define LOG_TAG "AudioFlinger::EffectChain"
 
 AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
-                                        int sessionId)
+                                        audio_session_t sessionId)
     : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
       mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
       mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX), mForceVolume(false)
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 6f93f81..bc9bc94 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -45,7 +45,7 @@
                     const wp<AudioFlinger::EffectChain>& chain,
                     effect_descriptor_t *desc,
                     int id,
-                    int sessionId);
+                    audio_session_t sessionId);
     virtual ~EffectModule();
 
     enum effect_state {
@@ -76,7 +76,7 @@
     uint32_t status() {
         return mStatus;
     }
-    int sessionId() const {
+    audio_session_t sessionId() const {
         return mSessionId;
     }
     status_t    setEnabled(bool enabled);
@@ -141,7 +141,7 @@
     wp<ThreadBase>      mThread;    // parent thread
     wp<EffectChain>     mChain;     // parent effect chain
     const int           mId;        // this instance unique ID
-    const int           mSessionId; // audio session ID
+    const audio_session_t mSessionId; // audio session ID
     const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
     effect_config_t     mConfig;    // input and output audio configuration
     effect_handle_t  mEffectInterface; // Effect module C API
@@ -235,15 +235,17 @@
 
 // the EffectChain class represents a group of effects associated to one audio session.
 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
-// The EffecChain with session ID 0 contains global effects applied to the output mix.
+// The EffectChain with session ID AUDIO_SESSION_OUTPUT_MIX contains global effects applied
+// to the output mix.
 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to
 // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the
-// order corresponding in the effect process order. When attached to a track (session ID != 0),
+// order corresponding in the effect process order. When attached to a track (session ID !=
+// AUDIO_SESSION_OUTPUT_MIX),
 // it also provide it's own input buffer used by the track as accumulation buffer.
 class EffectChain : public RefBase {
 public:
-    EffectChain(const wp<ThreadBase>& wThread, int sessionId);
-    EffectChain(ThreadBase *thread, int sessionId);
+    EffectChain(const wp<ThreadBase>& wThread, audio_session_t sessionId);
+    EffectChain(ThreadBase *thread, audio_session_t sessionId);
     virtual ~EffectChain();
 
     // special key used for an entry in mSuspendedEffects keyed vector
@@ -266,8 +268,8 @@
     status_t addEffect_l(const sp<EffectModule>& handle);
     size_t removeEffect_l(const sp<EffectModule>& handle);
 
-    int sessionId() const { return mSessionId; }
-    void setSessionId(int sessionId) { mSessionId = sessionId; }
+    audio_session_t sessionId() const { return mSessionId; }
+    void setSessionId(audio_session_t sessionId) { mSessionId = sessionId; }
 
     sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
     sp<EffectModule> getEffectFromId_l(int id);
@@ -362,7 +364,7 @@
     wp<ThreadBase> mThread;     // parent mixer thread
     Mutex mLock;                // mutex protecting effect list
     Vector< sp<EffectModule> > mEffects; // list of effect modules
-    int mSessionId;             // audio session ID
+    audio_session_t mSessionId; // audio session ID
     int16_t *mInBuffer;         // chain input buffer
     int16_t *mOutBuffer;        // chain output buffer
 
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 27149ed..a99becf 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -453,7 +453,7 @@
     patch->mPatchTrack->setPeerProxy(patch->mPatchRecord.get());
 
     // start capture and playback
-    patch->mPatchRecord->start(AudioSystem::SYNC_EVENT_NONE, 0);
+    patch->mPatchRecord->start(AudioSystem::SYNC_EVENT_NONE, AUDIO_SESSION_NONE);
     patch->mPatchTrack->start();
 
     return status;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 6e0c46d..270e27f 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -31,7 +31,7 @@
                                 size_t frameCount,
                                 void *buffer,
                                 const sp<IMemory>& sharedBuffer,
-                                int sessionId,
+                                audio_session_t sessionId,
                                 int uid,
                                 IAudioFlinger::track_flags_t flags,
                                 track_type type);
@@ -42,7 +42,7 @@
             void        dump(char* buffer, size_t size, bool active);
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             int triggerSession = 0);
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
     virtual void        stop();
             void        pause();
 
@@ -190,7 +190,7 @@
 
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             int triggerSession = 0);
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
     virtual void        stop();
             bool        write(void* data, uint32_t frames);
             bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
@@ -231,7 +231,7 @@
 
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             int triggerSession = 0);
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
 
     // AudioBufferProvider interface
     virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 6f84af1..13396a6 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -29,14 +29,14 @@
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
                                 void *buffer,
-                                int sessionId,
+                                audio_session_t sessionId,
                                 int uid,
                                 IAudioFlinger::track_flags_t flags,
                                 track_type type);
     virtual             ~RecordTrack();
     virtual status_t    initCheck() const;
 
-    virtual status_t    start(AudioSystem::sync_event_t event, int triggerSession);
+    virtual status_t    start(AudioSystem::sync_event_t event, audio_session_t triggerSession);
     virtual void        stop();
 
             void        destroy();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f206e96..e2932f1 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1134,14 +1134,14 @@
 }
 
 void AudioFlinger::ThreadBase::setEffectSuspended(
-        const effect_uuid_t *type, bool suspend, int sessionId)
+        const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
 {
     Mutex::Autolock _l(mLock);
     setEffectSuspended_l(type, suspend, sessionId);
 }
 
 void AudioFlinger::ThreadBase::setEffectSuspended_l(
-        const effect_uuid_t *type, bool suspend, int sessionId)
+        const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
 {
     sp<EffectChain> chain = getEffectChain_l(sessionId);
     if (chain != 0) {
@@ -1181,7 +1181,7 @@
 
 void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
                                                          bool suspend,
-                                                         int sessionId)
+                                                         audio_session_t sessionId)
 {
     ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
 
@@ -1242,7 +1242,7 @@
 
 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
                                                             bool enabled,
-                                                            int sessionId)
+                                                            audio_session_t sessionId)
 {
     Mutex::Autolock _l(mLock);
     checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
@@ -1250,7 +1250,7 @@
 
 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
                                                             bool enabled,
-                                                            int sessionId)
+                                                            audio_session_t sessionId)
 {
     if (mType != RECORD) {
         // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
@@ -1274,7 +1274,7 @@
         const sp<AudioFlinger::Client>& client,
         const sp<IEffectClient>& effectClient,
         int32_t priority,
-        int sessionId,
+        audio_session_t sessionId,
         effect_descriptor_t *desc,
         int *enabled,
         status_t *status)
@@ -1413,13 +1413,15 @@
     return handle;
 }
 
-sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
+        int effectId)
 {
     Mutex::Autolock _l(mLock);
     return getEffect_l(sessionId, effectId);
 }
 
-sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,
+        int effectId)
 {
     sp<EffectChain> chain = getEffectChain_l(sessionId);
     return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
@@ -1430,7 +1432,7 @@
 status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
 {
     // check for existing effect chain with the requested audio session
-    int sessionId = effect->sessionId();
+    audio_session_t sessionId = effect->sessionId();
     sp<EffectChain> chain = getEffectChain_l(sessionId);
     bool chainCreated = false;
 
@@ -1507,13 +1509,14 @@
     }
 }
 
-sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId)
 {
     Mutex::Autolock _l(mLock);
     return getEffectChain_l(sessionId);
 }
 
-sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId)
+        const
 {
     size_t size = mEffectChains.size();
     for (size_t i = 0; i < size; i++) {
@@ -1769,7 +1772,7 @@
         audio_channel_mask_t channelMask,
         size_t *pFrameCount,
         const sp<IMemory>& sharedBuffer,
-        int sessionId,
+        audio_session_t sessionId,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         int uid,
@@ -2039,14 +2042,14 @@
             TrackBase::track_state state = track->mState;
             mLock.unlock();
             status = AudioSystem::startOutput(mId, track->streamType(),
-                                              (audio_session_t)track->sessionId());
+                                              track->sessionId());
             mLock.lock();
             // abort track was stopped/paused while we released the lock
             if (state != track->mState) {
                 if (status == NO_ERROR) {
                     mLock.unlock();
                     AudioSystem::stopOutput(mId, track->streamType(),
-                                            (audio_session_t)track->sessionId());
+                                            track->sessionId());
                     mLock.lock();
                 }
                 return INVALID_OPERATION;
@@ -2421,7 +2424,7 @@
     }
 }
 
-uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
+uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const
 {
     Mutex::Autolock _l(mLock);
     uint32_t result = 0;
@@ -2440,7 +2443,7 @@
     return result;
 }
 
-uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
+uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId)
 {
     // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
     // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
@@ -2523,14 +2526,14 @@
             const sp<Track>& track = tracksToRemove.itemAt(i);
             if (track->isExternalTrack()) {
                 AudioSystem::stopOutput(mId, track->streamType(),
-                                        (audio_session_t)track->sessionId());
+                                        track->sessionId());
 #ifdef ADD_BATTERY_DATA
                 // to track the speaker usage
                 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
 #endif
                 if (track->isTerminated()) {
                     AudioSystem::releaseOutput(mId, track->streamType(),
-                                               (audio_session_t)track->sessionId());
+                                               track->sessionId());
                 }
             }
         }
@@ -2696,13 +2699,13 @@
 
 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
 {
-    int session = chain->sessionId();
+    audio_session_t session = chain->sessionId();
     int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
             ? mEffectBuffer : mSinkBuffer);
     bool ownsBuffer = false;
 
     ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
-    if (session > 0) {
+    if (session > AUDIO_SESSION_OUTPUT_MIX) {
         // Only one effect chain can be present in direct output thread and it uses
         // the sink buffer as input
         if (mType != DIRECT) {
@@ -2741,15 +2744,18 @@
     chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
             ? mEffectBuffer : mSinkBuffer));
     // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
-    // chains list in order to be processed last as it contains output stage effects
+    // chains list in order to be processed last as it contains output stage effects.
     // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
     // session AUDIO_SESSION_OUTPUT_STAGE to be processed
-    // after track specific effects and before output stage
+    // after track specific effects and before output stage.
     // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
-    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
+    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX.
     // Effect chain for other sessions are inserted at beginning of effect
     // chains list to be processed before output mix effects. Relative order between other
-    // sessions is not important
+    // sessions is not important.
+    static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
+            AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX,
+            "audio_session_t constants misdefined");
     size_t size = mEffectChains.size();
     size_t i = 0;
     for (i = 0; i < size; i++) {
@@ -2765,7 +2771,7 @@
 
 size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
 {
-    int session = chain->sessionId();
+    audio_session_t session = chain->sessionId();
 
     ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
 
@@ -4427,7 +4433,7 @@
 
 // getTrackName_l() must be called with ThreadBase::mLock held
 int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
-        audio_format_t format, int sessionId)
+        audio_format_t format, audio_session_t sessionId)
 {
     return mAudioMixer->getTrackName(channelMask, format, sessionId);
 }
@@ -5003,7 +5009,7 @@
 
 // getTrackName_l() must be called with ThreadBase::mLock held
 int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
-        audio_format_t format __unused, int sessionId __unused)
+        audio_format_t format __unused, audio_session_t sessionId __unused)
 {
     return 0;
 }
@@ -6201,7 +6207,8 @@
                                   (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
                                   activeTrack->sessionId(),
                                   (activeTrack->mSyncStartEvent != 0) ?
-                                          activeTrack->mSyncStartEvent->triggerSession() : 0);
+                                          activeTrack->mSyncStartEvent->triggerSession() :
+                                          AUDIO_SESSION_NONE);
                             activeTrack->clearSyncStartEvent();
                         }
                     }
@@ -6307,7 +6314,7 @@
         audio_format_t format,
         audio_channel_mask_t channelMask,
         size_t *pFrameCount,
-        int sessionId,
+        audio_session_t sessionId,
         size_t *notificationFrames,
         int uid,
         IAudioFlinger::track_flags_t *flags,
@@ -6426,7 +6433,7 @@
 
 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
                                            AudioSystem::sync_event_t event,
-                                           int triggerSession)
+                                           audio_session_t triggerSession)
 {
     ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
     sp<ThreadBase> strongMe = this;
@@ -6473,7 +6480,7 @@
         status_t status = NO_ERROR;
         if (recordTrack->isExternalTrack()) {
             mLock.unlock();
-            status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId());
+            status = AudioSystem::startInput(mId, recordTrack->sessionId());
             mLock.lock();
             // FIXME should verify that recordTrack is still in mActiveTracks
             if (status != NO_ERROR) {
@@ -6505,7 +6512,7 @@
 
 startError:
     if (recordTrack->isExternalTrack()) {
-        AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId());
+        AudioSystem::stopInput(mId, recordTrack->sessionId());
     }
     recordTrack->clearSyncStartEvent();
     // FIXME I wonder why we do not reset the state here?
@@ -6559,7 +6566,7 @@
         return BAD_VALUE;
     }
 
-    int eventSession = event->triggerSession();
+    audio_session_t eventSession = event->triggerSession();
     status_t ret = NAME_NOT_FOUND;
 
     Mutex::Autolock _l(mLock);
@@ -7225,7 +7232,7 @@
     return mInput->stream->get_input_frames_lost(mInput->stream);
 }
 
-uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
+uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const
 {
     Mutex::Autolock _l(mLock);
     uint32_t result = 0;
@@ -7243,13 +7250,13 @@
     return result;
 }
 
-KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
+KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
 {
-    KeyedVector<int, bool> ids;
+    KeyedVector<audio_session_t, bool> ids;
     Mutex::Autolock _l(mLock);
     for (size_t j = 0; j < mTracks.size(); ++j) {
         sp<RecordThread::RecordTrack> track = mTracks[j];
-        int sessionId = track->sessionId();
+        audio_session_t sessionId = track->sessionId();
         if (ids.indexOfKey(sessionId) < 0) {
             ids.add(sessionId, true);
         }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index d1613db..761fc71 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -288,7 +288,7 @@
                                     const sp<AudioFlinger::Client>& client,
                                     const sp<IEffectClient>& effectClient,
                                     int32_t priority,
-                                    int sessionId,
+                                    audio_session_t sessionId,
                                     effect_descriptor_t *desc,
                                     int *enabled,
                                     status_t *status /*non-NULL*/);
@@ -302,9 +302,9 @@
                 };
 
                 // get effect chain corresponding to session Id.
-                sp<EffectChain> getEffectChain(int sessionId);
+                sp<EffectChain> getEffectChain(audio_session_t sessionId);
                 // same as getEffectChain() but must be called with ThreadBase mutex locked
-                sp<EffectChain> getEffectChain_l(int sessionId) const;
+                sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const;
                 // add an effect chain to the chain list (mEffectChains)
     virtual     status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
                 // remove an effect chain from the chain list (mEffectChains)
@@ -321,8 +321,8 @@
                 // set audio mode to all effect chains
                 void setMode(audio_mode_t mode);
                 // get effect module with corresponding ID on specified audio session
-                sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId);
-                sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
+                sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId);
+                sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId);
                 // add and effect module. Also creates the effect chain is none exists for
                 // the effects audio session
                 status_t addEffect_l(const sp< EffectModule>& effect);
@@ -333,24 +333,27 @@
     virtual     void detachAuxEffect_l(int effectId __unused) {}
                 // returns either EFFECT_SESSION if effects on this audio session exist in one
                 // chain, or TRACK_SESSION if tracks on this audio session exist, or both
-                virtual uint32_t hasAudioSession(int sessionId) const = 0;
+                virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0;
                 // the value returned by default implementation is not important as the
                 // strategy is only meaningful for PlaybackThread which implements this method
-                virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
+                virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused)
+                        { return 0; }
 
                 // suspend or restore effect according to the type of effect passed. a NULL
                 // type pointer means suspend all effects in the session
                 void setEffectSuspended(const effect_uuid_t *type,
                                         bool suspend,
-                                        int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+                                        audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX);
                 // check if some effects must be suspended/restored when an effect is enabled
                 // or disabled
                 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
                                                  bool enabled,
-                                                 int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+                                                 audio_session_t sessionId =
+                                                        AUDIO_SESSION_OUTPUT_MIX);
                 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
                                                    bool enabled,
-                                                   int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+                                                   audio_session_t sessionId =
+                                                        AUDIO_SESSION_OUTPUT_MIX);
 
                 virtual status_t    setSyncEvent(const sp<SyncEvent>& event) = 0;
                 virtual bool        isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
@@ -389,11 +392,11 @@
                 void        getPowerManager_l();
                 void setEffectSuspended_l(const effect_uuid_t *type,
                                           bool suspend,
-                                          int sessionId);
+                                          audio_session_t sessionId);
                 // updated mSuspendedSessions when an effect suspended or restored
                 void        updateSuspendedSessions_l(const effect_uuid_t *type,
                                                       bool suspend,
-                                                      int sessionId);
+                                                      audio_session_t sessionId);
                 // check if some effects must be suspended when an effect chain is added
                 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
 
@@ -452,9 +455,9 @@
                 sp<IPowerManager>       mPowerManager;
                 sp<IBinder>             mWakeLockToken;
                 const sp<PMDeathRecipient> mDeathRecipient;
-                // list of suspended effects per session and per type. The first vector is
-                // keyed by session ID, the second by type UUID timeLow field
-                KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > >
+                // list of suspended effects per session and per type. The first (outer) vector is
+                // keyed by session ID, the second (inner) by type UUID timeLow field
+                KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > >
                                         mSuspendedSessions;
                 static const size_t     kLogSize = 4 * 1024;
                 sp<NBLog::Writer>       mNBLogWriter;
@@ -549,7 +552,7 @@
                                 audio_channel_mask_t channelMask,
                                 size_t *pFrameCount,
                                 const sp<IMemory>& sharedBuffer,
-                                int sessionId,
+                                audio_session_t sessionId,
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
                                 int uid,
@@ -589,8 +592,8 @@
 
                 virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
                 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
-                virtual uint32_t hasAudioSession(int sessionId) const;
-                virtual uint32_t getStrategyForSession_l(int sessionId);
+                virtual uint32_t hasAudioSession(audio_session_t sessionId) const;
+                virtual uint32_t getStrategyForSession_l(audio_session_t sessionId);
 
 
                 virtual status_t setSyncEvent(const sp<SyncEvent>& event);
@@ -704,7 +707,7 @@
     // Allocate a track name for a given channel mask.
     //   Returns name >= 0 if successful, -1 on failure.
     virtual int             getTrackName_l(audio_channel_mask_t channelMask,
-                                           audio_format_t format, int sessionId) = 0;
+                                           audio_format_t format, audio_session_t sessionId) = 0;
     virtual void            deleteTrackName_l(int name) = 0;
 
     // Time to sleep between cycles when:
@@ -859,7 +862,7 @@
 protected:
     virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
     virtual     int         getTrackName_l(audio_channel_mask_t channelMask,
-                                           audio_format_t format, int sessionId);
+                                           audio_format_t format, audio_session_t sessionId);
     virtual     void        deleteTrackName_l(int name);
     virtual     uint32_t    idleSleepTimeUs() const;
     virtual     uint32_t    suspendSleepTimeUs() const;
@@ -940,7 +943,7 @@
 
 protected:
     virtual     int         getTrackName_l(audio_channel_mask_t channelMask,
-                                           audio_format_t format, int sessionId);
+                                           audio_format_t format, audio_session_t sessionId);
     virtual     void        deleteTrackName_l(int name);
     virtual     uint32_t    activeSleepTimeUs() const;
     virtual     uint32_t    idleSleepTimeUs() const;
@@ -1243,7 +1246,7 @@
                     audio_format_t format,
                     audio_channel_mask_t channelMask,
                     size_t *pFrameCount,
-                    int sessionId,
+                    audio_session_t sessionId,
                     size_t *notificationFrames,
                     int uid,
                     IAudioFlinger::track_flags_t *flags,
@@ -1252,7 +1255,7 @@
 
             status_t    start(RecordTrack* recordTrack,
                               AudioSystem::sync_event_t event,
-                              int triggerSession);
+                              audio_session_t triggerSession);
 
             // ask the thread to stop the specified track, and
             // return true if the caller should then do it's part of the stopping process
@@ -1280,12 +1283,12 @@
 
     virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
     virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
-    virtual uint32_t hasAudioSession(int sessionId) const;
+    virtual uint32_t hasAudioSession(audio_session_t sessionId) const;
 
             // Return the set of unique session IDs across all tracks.
             // The keys are the session IDs, and the associated values are meaningless.
             // FIXME replace by Set [and implement Bag/Multiset for other uses].
-            KeyedVector<int, bool> sessionIds() const;
+            KeyedVector<audio_session_t, bool> sessionIds() const;
 
     virtual status_t setSyncEvent(const sp<SyncEvent>& event);
     virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 26067e3..67a5e58 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -61,7 +61,7 @@
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
                                 void *buffer,
-                                int sessionId,
+                                audio_session_t sessionId,
                                 int uid,
                                 IAudioFlinger::track_flags_t flags,
                                 bool isOut,
@@ -71,11 +71,11 @@
     virtual status_t    initCheck() const;
 
     virtual status_t    start(AudioSystem::sync_event_t event,
-                             int triggerSession) = 0;
+                             audio_session_t triggerSession) = 0;
     virtual void        stop() = 0;
             sp<IMemory> getCblk() const { return mCblkMemory; }
             audio_track_cblk_t* cblk() const { return mCblk; }
-            int         sessionId() const { return mSessionId; }
+            audio_session_t sessionId() const { return mSessionId; }
             int         uid() const { return mUid; }
     virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
 
@@ -153,7 +153,7 @@
     const size_t        mFrameCount;// size of track buffer given at createTrack() or
                                     // openRecord(), and then adjusted as needed
 
-    const int           mSessionId;
+    const audio_session_t mSessionId;
     int                 mUid;
     Vector < sp<SyncEvent> >mSyncEvents;
     const IAudioFlinger::track_flags_t mFlags;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 0c51e81..f575918 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -71,7 +71,7 @@
             audio_channel_mask_t channelMask,
             size_t frameCount,
             void *buffer,
-            int sessionId,
+            audio_session_t sessionId,
             int clientUid,
             IAudioFlinger::track_flags_t flags,
             bool isOut,
@@ -343,7 +343,7 @@
             size_t frameCount,
             void *buffer,
             const sp<IMemory>& sharedBuffer,
-            int sessionId,
+            audio_session_t sessionId,
             int uid,
             IAudioFlinger::track_flags_t flags,
             track_type type)
@@ -454,7 +454,7 @@
             wasActive = playbackThread->destroyTrack_l(this);
         }
         if (isExternalTrack() && !wasActive) {
-            AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, (audio_session_t)mSessionId);
+            AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, mSessionId);
         }
     }
 }
@@ -617,7 +617,7 @@
         return true;
     }
 
-    if (framesReady() >= mFrameCount ||
+    if (framesReady() >= mServerProxy->getBufferSizeInFrames() ||
             (mCblk->mFlags & CBLK_FORCEREADY)) {
         mFillingUpStatus = FS_FILLED;
         android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
@@ -627,7 +627,7 @@
 }
 
 status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
-                                                    int triggerSession __unused)
+                                                    audio_session_t triggerSession __unused)
 {
     status_t status = NO_ERROR;
     ALOGV("start(%d), calling pid %d session %d",
@@ -1128,7 +1128,8 @@
             int uid)
     :   Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
               sampleRate, format, channelMask, frameCount,
-              NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT),
+              NULL, 0, AUDIO_SESSION_NONE, uid, IAudioFlinger::TRACK_DEFAULT,
+              TYPE_OUTPUT),
     mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
 {
 
@@ -1159,7 +1160,7 @@
 }
 
 status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
-                                                          int triggerSession)
+                                                          audio_session_t triggerSession)
 {
     status_t status = Track::start(event, triggerSession);
     if (status != NO_ERROR) {
@@ -1326,7 +1327,7 @@
                                                      IAudioFlinger::track_flags_t flags)
     :   Track(playbackThread, NULL, streamType,
               sampleRate, format, channelMask, frameCount,
-              buffer, 0, 0, getuid(), flags, TYPE_PATCH),
+              buffer, 0, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
               mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
 {
     uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
@@ -1345,7 +1346,7 @@
 }
 
 status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
-                                                          int triggerSession)
+                                                          audio_session_t triggerSession)
 {
     status_t status = Track::start(event, triggerSession);
     if (status != NO_ERROR) {
@@ -1429,7 +1430,7 @@
 }
 
 status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
-        int triggerSession) {
+        audio_session_t triggerSession) {
     ALOGV("RecordHandle::start()");
     return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
 }
@@ -1460,7 +1461,7 @@
             audio_channel_mask_t channelMask,
             size_t frameCount,
             void *buffer,
-            int sessionId,
+            audio_session_t sessionId,
             int uid,
             IAudioFlinger::track_flags_t flags,
             track_type type)
@@ -1537,7 +1538,7 @@
 }
 
 status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
-                                                        int triggerSession)
+                                                        audio_session_t triggerSession)
 {
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
@@ -1554,7 +1555,7 @@
     if (thread != 0) {
         RecordThread *recordThread = (RecordThread *)thread.get();
         if (recordThread->stop(this) && isExternalTrack()) {
-            AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
+            AudioSystem::stopInput(mThreadIoHandle, mSessionId);
         }
     }
 }
@@ -1566,9 +1567,9 @@
     {
         if (isExternalTrack()) {
             if (mState == ACTIVE || mState == RESUMING) {
-                AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
+                AudioSystem::stopInput(mThreadIoHandle, mSessionId);
             }
-            AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId);
+            AudioSystem::releaseInput(mThreadIoHandle, mSessionId);
         }
         sp<ThreadBase> thread = mThread.promote();
         if (thread != 0) {
@@ -1660,7 +1661,7 @@
                                                      void *buffer,
                                                      IAudioFlinger::track_flags_t flags)
     :   RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount,
-                buffer, 0, getuid(), flags, TYPE_PATCH),
+                buffer, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
                 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
 {
     uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 1c41ce1..b1347f4 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -312,7 +312,7 @@
     virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
 
     // move effect to the specified output
-    virtual status_t moveEffects(int session,
+    virtual status_t moveEffects(audio_session_t session,
                                      audio_io_handle_t srcOutput,
                                      audio_io_handle_t dstOutput) = 0;
 
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 6968a74..f73548d 100755
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -42,9 +42,9 @@
  * A device mask for all audio input and output devices where matching inputs/outputs on device
  * type alone is not enough: the address must match too
  */
-#define APM_AUDIO_DEVICE_OUT_MATCH_ADDRESS_ALL (AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+#define APM_AUDIO_DEVICE_OUT_MATCH_ADDRESS_ALL (AUDIO_DEVICE_OUT_REMOTE_SUBMIX|AUDIO_DEVICE_OUT_BUS)
 
-#define APM_AUDIO_DEVICE_IN_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX)
+#define APM_AUDIO_DEVICE_IN_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_BUS)
 
 /**
  * Check if the state given correspond to an in call state.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index c952831..8f5ebef 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -54,7 +54,7 @@
 public:
     status_t getAudioPolicyMix(String8 address, sp<AudioPolicyMix> &policyMix) const;
 
-    status_t registerMix(String8 address, AudioMix mix);
+    status_t registerMix(String8 address, AudioMix mix, sp<SwAudioOutputDescriptor> desc);
 
     status_t unregisterMix(String8 address);
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 0fe1a84..c5fee50 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -315,14 +315,14 @@
     if ((oldGlobalRefCount == 0) && (mGlobalRefCount > 0)) {
         if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
         {
-            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mRegistrationId,
+            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
                     MIX_STATE_MIXING);
         }
 
     } else if ((oldGlobalRefCount > 0) && (mGlobalRefCount == 0)) {
         if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
         {
-            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mRegistrationId,
+            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
                     MIX_STATE_IDLE);
         }
     }
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 3735c05..4af3d54 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -51,7 +51,8 @@
     return &mMix;
 }
 
-status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix)
+status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix,
+                                               sp<SwAudioOutputDescriptor> desc)
 {
     ssize_t index = indexOfKey(address);
     if (index >= 0) {
@@ -61,6 +62,11 @@
     sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
     policyMix->setMix(mix);
     add(address, policyMix);
+
+    if (desc != 0) {
+        desc->mPolicyMix = policyMix->getMix();
+        policyMix->setOutput(desc);
+    }
     return NO_ERROR;
 }
 
@@ -128,7 +134,7 @@
                 // if there is an address match, prioritize that match
                 if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
                         strncmp(attributes.tags + strlen("addr="),
-                                mix->mRegistrationId.string(),
+                                mix->mDeviceAddress.string(),
                                 AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
                     hasAddrMatch = true;
                     break;
@@ -207,7 +213,7 @@
             if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
                     strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
                     strncmp(attributes.tags + strlen("addr="),
-                            mix->mRegistrationId.string(),
+                            mix->mDeviceAddress.string(),
                             AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
                 desc = policyMix->getOutput();
             }
@@ -260,7 +266,7 @@
     for (size_t i = 0; i < size(); i++) {
             sp<AudioPolicyMix> policyMix = valueAt(i);
             AudioMix *mix = policyMix->getMix();
-            ALOGV("\tmix %zu address=%s", i, mix->mRegistrationId.string());
+            ALOGV("\tmix %zu address=%s", i, mix->mDeviceAddress.string());
     }
 #endif
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
index ce9c170..da983c5 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -79,7 +79,7 @@
         // if input maps to a dynamic policy with an activity listener, notify of state change
         if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
         {
-            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mRegistrationId,
+            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
                     (event == RECORD_CONFIG_EVENT_START) ? MIX_STATE_MIXING : MIX_STATE_IDLE);
         }
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index c3b1529..b2b014a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -954,6 +954,13 @@
     }
 
 non_direct_output:
+
+    // A request for HW A/V sync cannot fallback to a mixed output because time
+    // stamps are embedded in audio data
+    if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+        return AUDIO_IO_HANDLE_NONE;
+    }
+
     // ignoring channel mask due to downmix capability in mixer
 
     // open a non direct output
@@ -1096,7 +1103,7 @@
             outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
             setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                     AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                    outputDesc->mPolicyMix->mRegistrationId,
+                    outputDesc->mPolicyMix->mDeviceAddress,
                     "remote-submix");
     }
 
@@ -1213,7 +1220,7 @@
                 outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
             setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                     AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                    outputDesc->mPolicyMix->mRegistrationId,
+                    outputDesc->mPolicyMix->mDeviceAddress,
                     "remote-submix");
         }
     }
@@ -1388,7 +1395,7 @@
             return BAD_VALUE;
         }
         if (policyMix != NULL) {
-            address = policyMix->mRegistrationId;
+            address = policyMix->mDeviceAddress;
             if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
                 // there is an external policy, but this input is attached to a mix of recorders,
                 // meaning it receives audio injected into the framework, so the recorder doesn't
@@ -1622,7 +1629,7 @@
         // if input maps to a dynamic policy with an activity listener, notify of state change
         if ((inputDesc->mPolicyMix != NULL)
                 && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
-            mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
+            mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
                     MIX_STATE_MIXING);
         }
 
@@ -1639,7 +1646,7 @@
             if (inputDesc->mPolicyMix == NULL) {
                 address = String8("0");
             } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
-                address = inputDesc->mPolicyMix->mRegistrationId;
+                address = inputDesc->mPolicyMix->mDeviceAddress;
             }
             if (address != "") {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
@@ -1686,7 +1693,7 @@
         // if input maps to a dynamic policy with an activity listener, notify of state change
         if ((inputDesc->mPolicyMix != NULL)
                 && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
-            mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
+            mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
                     MIX_STATE_IDLE);
         }
 
@@ -1697,7 +1704,7 @@
             if (inputDesc->mPolicyMix == NULL) {
                 address = String8("0");
             } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
-                address = inputDesc->mPolicyMix->mRegistrationId;
+                address = inputDesc->mPolicyMix->mDeviceAddress;
             }
             if (address != "") {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
@@ -2010,94 +2017,152 @@
 
 status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
 {
-    sp<HwModule> module;
-    for (size_t i = 0; i < mHwModules.size(); i++) {
-        if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
-                mHwModules[i]->mHandle != 0) {
-            module = mHwModules[i];
+    ALOGV("registerPolicyMixes() %zu mix(es)", mixes.size());
+    status_t res = NO_ERROR;
+
+    sp<HwModule> rSubmixModule;
+    // examine each mix's route type
+    for (size_t i = 0; i < mixes.size(); i++) {
+        // we only support MIX_ROUTE_FLAG_LOOP_BACK or MIX_ROUTE_FLAG_RENDER, not the combination
+        if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_ALL) == MIX_ROUTE_FLAG_ALL) {
+            res = INVALID_OPERATION;
             break;
         }
-    }
+        if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
+            // Loop back through "remote submix"
+            if (rSubmixModule == 0) {
+                for (size_t j = 0; i < mHwModules.size(); j++) {
+                    if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
+                            && mHwModules[j]->mHandle != 0) {
+                        rSubmixModule = mHwModules[j];
+                        break;
+                    }
+                }
+            }
 
-    if (module == 0) {
-        return INVALID_OPERATION;
-    }
+            ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK", i, mixes.size());
 
-    ALOGV("registerPolicyMixes() num mixes %zu", mixes.size());
+            if (rSubmixModule == 0) {
+                ALOGE(" Unable to find audio module for submix, aborting mix %zu registration", i);
+                res = INVALID_OPERATION;
+                break;
+            }
 
-    for (size_t i = 0; i < mixes.size(); i++) {
-        String8 address = mixes[i].mRegistrationId;
+            String8 address = mixes[i].mDeviceAddress;
 
-        if (mPolicyMixes.registerMix(address, mixes[i]) != NO_ERROR) {
-            continue;
-        }
-        audio_config_t outputConfig = mixes[i].mFormat;
-        audio_config_t inputConfig = mixes[i].mFormat;
-        // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
-        // stereo and let audio flinger do the channel conversion if needed.
-        outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
-        inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
-        module->addOutputProfile(address, &outputConfig,
-                                 AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
-        module->addInputProfile(address, &inputConfig,
-                                 AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
+            if (mPolicyMixes.registerMix(address, mixes[i], 0 /*output desc*/) != NO_ERROR) {
+                ALOGE(" Error regisering mix %zu for address %s", i, address.string());
+                res = INVALID_OPERATION;
+                break;
+            }
+            audio_config_t outputConfig = mixes[i].mFormat;
+            audio_config_t inputConfig = mixes[i].mFormat;
+            // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
+            // stereo and let audio flinger do the channel conversion if needed.
+            outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+            inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+            rSubmixModule->addOutputProfile(address, &outputConfig,
+                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
+            rSubmixModule->addInputProfile(address, &inputConfig,
+                    AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
 
-        if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
-            setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
-                                     AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                                     address.string(), "remote-submix");
-        } else {
-            setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                                     AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                                     address.string(), "remote-submix");
+            if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
+                setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                        AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                        address.string(), "remote-submix");
+            } else {
+                setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                        AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                        address.string(), "remote-submix");
+            }
+        } else if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
+            ALOGV("registerPolicyMixes() mix %zu of %zu is RENDER", i, mixes.size());
+            String8 address = mixes[i].mDeviceAddress;
+
+            audio_devices_t device = mixes[i].mDeviceType;
+
+            for (size_t j = 0 ; j < mOutputs.size() ; j++) {
+                sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(j);
+                sp<AudioPatch> patch = mAudioPatches.valueFor(desc->getPatchHandle());
+                if ((patch != 0) && (patch->mPatch.num_sinks != 0)
+                        && (patch->mPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE)
+                        && (patch->mPatch.sinks[0].ext.device.type == device)
+                        && (patch->mPatch.sinks[0].ext.device.address == address)) {
+
+                    if (mPolicyMixes.registerMix(address, mixes[i], desc) != NO_ERROR) {
+                        res = INVALID_OPERATION;
+                    }
+                    break;
+                }
+            }
+
+            if (res != NO_ERROR) {
+                ALOGE(" Error registering mix %zu for device 0x%X addr %s",
+                        i,device, address.string());
+                res = INVALID_OPERATION;
+                break;
+            }
         }
     }
-    return NO_ERROR;
+    if (res != NO_ERROR) {
+        unregisterPolicyMixes(mixes);
+    }
+    return res;
 }
 
 status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
 {
-    sp<HwModule> module;
-    for (size_t i = 0; i < mHwModules.size(); i++) {
-        if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
-                mHwModules[i]->mHandle != 0) {
-            module = mHwModules[i];
-            break;
-        }
-    }
-
-    if (module == 0) {
-        return INVALID_OPERATION;
-    }
-
     ALOGV("unregisterPolicyMixes() num mixes %zu", mixes.size());
-
+    status_t res = NO_ERROR;
+    sp<HwModule> rSubmixModule;
+    // examine each mix's route type
     for (size_t i = 0; i < mixes.size(); i++) {
-        String8 address = mixes[i].mRegistrationId;
+        if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
 
-        if (mPolicyMixes.unregisterMix(address) != NO_ERROR) {
-            continue;
-        }
+            if (rSubmixModule == 0) {
+                for (size_t j = 0; i < mHwModules.size(); j++) {
+                    if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
+                            && mHwModules[j]->mHandle != 0) {
+                        rSubmixModule = mHwModules[j];
+                        break;
+                    }
+                }
+            }
+            if (rSubmixModule == 0) {
+                res = INVALID_OPERATION;
+                continue;
+            }
 
-        if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
-                                             AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
-        {
-            setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
-                                     AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                     address.string(), "remote-submix");
-        }
+            String8 address = mixes[i].mDeviceAddress;
 
-        if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
-                                             AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
-        {
-            setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                                     AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                     address.string(), "remote-submix");
+            if (mPolicyMixes.unregisterMix(address) != NO_ERROR) {
+                res = INVALID_OPERATION;
+                continue;
+            }
+
+            if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
+                    AUDIO_POLICY_DEVICE_STATE_AVAILABLE)  {
+                setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                        AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+                        address.string(), "remote-submix");
+            }
+            if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
+                    AUDIO_POLICY_DEVICE_STATE_AVAILABLE)  {
+                setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                        AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+                        address.string(), "remote-submix");
+            }
+            rSubmixModule->removeOutputProfile(address);
+            rSubmixModule->removeInputProfile(address);
+
+        } if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
+            if (mPolicyMixes.unregisterMix(mixes[i].mDeviceAddress) != NO_ERROR) {
+                res = INVALID_OPERATION;
+                continue;
+            }
         }
-        module->removeOutputProfile(address);
-        module->removeInputProfile(address);
     }
-    return NO_ERROR;
+    return res;
 }
 
 
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index f1c0d1a..08f9cc1 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -171,7 +171,7 @@
     return mAudioPolicyService->setVoiceVolume(volume, delay_ms);
 }
 
-status_t AudioPolicyService::AudioPolicyClient::moveEffects(int session,
+status_t AudioPolicyService::AudioPolicyClient::moveEffects(audio_session_t session,
                         audio_io_handle_t src_output,
                         audio_io_handle_t dst_output)
 {
diff --git a/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
index a79f8ae..580d740 100644
--- a/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
@@ -252,7 +252,7 @@
     return af->invalidateStream(stream);
 }
 
-int aps_move_effects(void *service __unused, int session,
+int aps_move_effects(void *service __unused, audio_session_t session,
                                 audio_io_handle_t src_output,
                                 audio_io_handle_t dst_output)
 {
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index ce77814..b732b20 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -79,7 +79,7 @@
 
 status_t AudioPolicyEffects::addInputEffects(audio_io_handle_t input,
                              audio_source_t inputSource,
-                             int audioSession)
+                             audio_session_t audioSession)
 {
     status_t status = NO_ERROR;
 
@@ -152,7 +152,7 @@
     return status;
 }
 
-status_t AudioPolicyEffects::queryDefaultInputEffects(int audioSession,
+status_t AudioPolicyEffects::queryDefaultInputEffects(audio_session_t audioSession,
                                                       effect_descriptor_t *descriptors,
                                                       uint32_t *count)
 {
@@ -185,7 +185,7 @@
 }
 
 
-status_t AudioPolicyEffects::queryDefaultOutputSessionEffects(int audioSession,
+status_t AudioPolicyEffects::queryDefaultOutputSessionEffects(audio_session_t audioSession,
                          effect_descriptor_t *descriptors,
                          uint32_t *count)
 {
@@ -220,7 +220,7 @@
 
 status_t AudioPolicyEffects::addOutputSessionEffects(audio_io_handle_t output,
                          audio_stream_type_t stream,
-                         int audioSession)
+                         audio_session_t audioSession)
 {
     status_t status = NO_ERROR;
 
@@ -275,7 +275,7 @@
 
 status_t AudioPolicyEffects::releaseOutputSessionEffects(audio_io_handle_t output,
                          audio_stream_type_t stream,
-                         int audioSession)
+                         audio_session_t audioSession)
 {
     status_t status = NO_ERROR;
     (void) output; // argument not used for now
diff --git a/services/audiopolicy/service/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
index 266a45e..ee9bd50 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.h
+++ b/services/audiopolicy/service/AudioPolicyEffects.h
@@ -51,7 +51,7 @@
 
     // Return a list of effect descriptors for default input effects
     // associated with audioSession
-    status_t queryDefaultInputEffects(int audioSession,
+    status_t queryDefaultInputEffects(audio_session_t audioSession,
                              effect_descriptor_t *descriptors,
                              uint32_t *count);
 
@@ -59,7 +59,7 @@
     // Effects are attached depending on the audio_source_t
     status_t addInputEffects(audio_io_handle_t input,
                              audio_source_t inputSource,
-                             int audioSession);
+                             audio_session_t audioSession);
 
     // Add all input effects associated to this input
     status_t releaseInputEffects(audio_io_handle_t input);
@@ -67,7 +67,7 @@
 
     // Return a list of effect descriptors for default output effects
     // associated with audioSession
-    status_t queryDefaultOutputSessionEffects(int audioSession,
+    status_t queryDefaultOutputSessionEffects(audio_session_t audioSession,
                              effect_descriptor_t *descriptors,
                              uint32_t *count);
 
@@ -75,12 +75,12 @@
     // Effects are attached depending on the audio_stream_type_t
     status_t addOutputSessionEffects(audio_io_handle_t output,
                              audio_stream_type_t stream,
-                             int audioSession);
+                             audio_session_t audioSession);
 
     // release all output effects associated with this output stream and audiosession
     status_t releaseOutputSessionEffects(audio_io_handle_t output,
                              audio_stream_type_t stream,
-                             int audioSession);
+                             audio_session_t audioSession);
 
 private:
 
@@ -135,13 +135,13 @@
     // class to store voctor of AudioEffects
     class EffectVector {
     public:
-        EffectVector(int session) : mSessionId(session), mRefCount(0) {}
+        EffectVector(audio_session_t session) : mSessionId(session), mRefCount(0) {}
         /*virtual*/ ~EffectVector() {}
 
         // Enable or disable all effects in effect vector
         void setProcessorEnabled(bool enabled);
 
-        const int mSessionId;
+        const audio_session_t mSessionId;
         // AudioPolicyManager keeps mLock, no need for lock on reference count here
         int mRefCount;
         Vector< sp<AudioEffect> >mEffects;
@@ -188,7 +188,7 @@
     // Automatic output effects are organized per audio_stream_type_t
     KeyedVector< audio_stream_type_t, EffectDescVector* > mOutputStreams;
     // Automatic output effects are unique for audiosession ID
-    KeyedVector< int32_t, EffectVector* > mOutputSessions;
+    KeyedVector< audio_session_t, EffectVector* > mOutputSessions;
 };
 
 }; // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index fdd6dd2..92a1285 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -476,7 +476,7 @@
 status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc,
                                 audio_io_handle_t io,
                                 uint32_t strategy,
-                                int session,
+                                audio_session_t session,
                                 int id)
 {
     if (mAudioPolicyManager == NULL) {
@@ -537,7 +537,7 @@
     return mAudioPolicyManager->isSourceActive(source);
 }
 
-status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
+status_t AudioPolicyService::queryDefaultPreProcessing(audio_session_t audioSession,
                                                        effect_descriptor_t *descriptors,
                                                        uint32_t *count)
 {
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
index 08b2a3b..c830454 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
@@ -427,7 +427,7 @@
 status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc,
                                 audio_io_handle_t io,
                                 uint32_t strategy,
-                                int session,
+                                audio_session_t session,
                                 int id)
 {
     if (mpAudioPolicy == NULL) {
@@ -488,7 +488,7 @@
     return mpAudioPolicy->is_source_active(mpAudioPolicy, source);
 }
 
-status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
+status_t AudioPolicyService::queryDefaultPreProcessing(audio_session_t audioSession,
                                                        effect_descriptor_t *descriptors,
                                                        uint32_t *count)
 {
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 94e0701..a6cd50e 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -1182,7 +1182,7 @@
                                                   audio_channel_mask_t *pChannelMask);
 int aps_close_input(void *service __unused, audio_io_handle_t input);
 int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream);
-int aps_move_effects(void *service __unused, int session,
+int aps_move_effects(void *service __unused, audio_session_t session,
                                 audio_io_handle_t src_output,
                                 audio_io_handle_t dst_output);
 char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index cecd617..2710ac7 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -128,7 +128,7 @@
     virtual status_t registerEffect(const effect_descriptor_t *desc,
                                     audio_io_handle_t io,
                                     uint32_t strategy,
-                                    int session,
+                                    audio_session_t session,
                                     int id);
     virtual status_t unregisterEffect(int id);
     virtual status_t setEffectEnabled(int id, bool enabled);
@@ -136,7 +136,7 @@
     virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
     virtual bool isSourceActive(audio_source_t source) const;
 
-    virtual status_t queryDefaultPreProcessing(int audioSession,
+    virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count);
     virtual     status_t    onTransact(
@@ -498,7 +498,7 @@
         virtual status_t setVoiceVolume(float volume, int delayMs = 0);
 
         // move effect to the specified output
-        virtual status_t moveEffects(int session,
+        virtual status_t moveEffects(audio_session_t session,
                                          audio_io_handle_t srcOutput,
                                          audio_io_handle_t dstOutput);
 
diff --git a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
index 5f4fb22..b4b269a 100644
--- a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
@@ -155,7 +155,7 @@
                 callbackFormat, params.previewFormat);
         res = device->createStream(mCallbackWindow,
                 params.previewWidth, params.previewHeight, callbackFormat,
-                HAL_DATASPACE_JFIF, CAMERA3_STREAM_ROTATION_0, &mCallbackStreamId);
+                HAL_DATASPACE_V0_JFIF, CAMERA3_STREAM_ROTATION_0, &mCallbackStreamId);
         if (res != OK) {
             ALOGE("%s: Camera %d: Can't create output stream for callbacks: "
                     "%s (%d)", __FUNCTION__, mId,
diff --git a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
index 3923853..e97618c 100644
--- a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
@@ -145,7 +145,7 @@
         // Create stream for HAL production
         res = device->createStream(mCaptureWindow,
                 params.pictureWidth, params.pictureHeight,
-                HAL_PIXEL_FORMAT_BLOB, HAL_DATASPACE_JFIF,
+                HAL_PIXEL_FORMAT_BLOB, HAL_DATASPACE_V0_JFIF,
                 CAMERA3_STREAM_ROTATION_0, &mCaptureStreamId);
         if (res != OK) {
             ALOGE("%s: Camera %d: Can't create output stream for capture: "
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index d4022cd..5779176 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -872,7 +872,7 @@
 
     // Set up initial state for non-Camera.Parameters state variables
     videoFormat = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
-    videoDataSpace = HAL_DATASPACE_BT709;
+    videoDataSpace = HAL_DATASPACE_V0_BT709;
     videoBufferMode = hardware::ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV;
     playShutterSound = true;
     enableFaceDetect = false;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 05c5323..331f10d 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -410,6 +410,31 @@
     return measured;
 }
 
+/**
+ * Map Android N dataspace definitions back to Android M definitions, for
+ * use with HALv3.3 or older.
+ *
+ * Only map where correspondences exist, and otherwise preserve the value.
+ */
+android_dataspace Camera3Device::mapToLegacyDataspace(android_dataspace dataSpace) {
+    switch (dataSpace) {
+        case HAL_DATASPACE_V0_SRGB_LINEAR:
+            return HAL_DATASPACE_SRGB_LINEAR;
+        case HAL_DATASPACE_V0_SRGB:
+            return HAL_DATASPACE_SRGB;
+        case HAL_DATASPACE_V0_JFIF:
+            return HAL_DATASPACE_JFIF;
+        case HAL_DATASPACE_V0_BT601_625:
+            return HAL_DATASPACE_BT601_625;
+        case HAL_DATASPACE_V0_BT601_525:
+            return HAL_DATASPACE_BT601_525;
+        case HAL_DATASPACE_V0_BT709:
+            return HAL_DATASPACE_BT709;
+        default:
+            return dataSpace;
+    }
+}
+
 ssize_t Camera3Device::getJpegBufferSize(uint32_t width, uint32_t height) const {
     // Get max jpeg size (area-wise).
     Size maxJpegResolution = getMaxJpegResolution();
@@ -1006,6 +1031,10 @@
     if (mDeviceVersion <= CAMERA_DEVICE_API_VERSION_3_2) {
         streamSetId = CAMERA3_STREAM_SET_ID_INVALID;
     }
+    // Use legacy dataspace values for older HALs
+    if (mDeviceVersion <= CAMERA_DEVICE_API_VERSION_3_3) {
+        dataSpace = mapToLegacyDataspace(dataSpace);
+    }
     if (format == HAL_PIXEL_FORMAT_BLOB) {
         ssize_t blobBufferSize;
         if (dataSpace != HAL_DATASPACE_DEPTH) {
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 5b1c87e..ba092d0 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -404,6 +404,11 @@
      */
     static nsecs_t getMonoToBoottimeOffset();
 
+    /**
+     * Helper function to map between legacy and new dataspace enums
+     */
+    static android_dataspace mapToLegacyDataspace(android_dataspace dataSpace);
+
     struct RequestTrigger {
         // Metadata tag number, e.g. android.control.aePrecaptureTrigger
         uint32_t metadataTag;