Merge commit 'b2059ff384eee8ffb70a7ec8fc5570405201c734' into HEAD

58 files changed, 1855 insertions, 409 deletions

diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 37626a4..79682b8 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -51,8 +51,14 @@ struct CameraTraits<Camera>
     typedef CameraListener        TCamListener;
     typedef ICamera               TCamUser;
     typedef ICameraClient         TCamCallbacks;
+    typedef status_t (ICameraService::*TCamConnectService)(const sp<ICameraClient>&,
+                                                           int, const String16&, int,
+                                                           /*out*/
+                                                           sp<ICamera>&);
+    static TCamConnectService     fnConnectService;
 };
 
+
 class Camera :
     public CameraBase<Camera>,
     public BnCameraClient
@@ -75,9 +81,9 @@ public:
             status_t    unlock();
 
             // pass the buffered IGraphicBufferProducer to the camera service
-            status_t    setPreviewTexture(const sp<IGraphicBufferProducer>& bufferProducer);
+            status_t    setPreviewTarget(const sp<IGraphicBufferProducer>& bufferProducer);
 
-            // start preview mode, must call setPreviewDisplay first
+            // start preview mode, must call setPreviewTarget first
             status_t    startPreview();
 
             // stop preview mode
@@ -86,7 +92,7 @@ public:
             // get preview state
             bool        previewEnabled();
 
-            // start recording mode, must call setPreviewDisplay first
+            // start recording mode, must call setPreviewTarget first
             status_t    startRecording();
 
             // stop recording mode
@@ -121,7 +127,15 @@ public:
 
             void        setListener(const sp<CameraListener>& listener);
             void        setRecordingProxyListener(const sp<ICameraRecordingProxyListener>& listener);
+
+            // Configure preview callbacks to app. Only one of the older
+            // callbacks or the callback surface can be active at the same time;
+            // enabling one will disable the other if active. Flags can be
+            // disabled by calling it with CAMERA_FRAME_CALLBACK_FLAG_NOOP, and
+            // Target by calling it with a NULL interface.
             void        setPreviewCallbackFlags(int preview_callback_flag);
+            status_t    setPreviewCallbackTarget(
+                    const sp<IGraphicBufferProducer>& callbackProducer);
 
             sp<ICameraRecordingProxy> getRecordingProxy();
 
diff --git a/include/camera/CameraBase.h b/include/camera/CameraBase.h
index 9b08c0f..1b93157 100644
--- a/include/camera/CameraBase.h
+++ b/include/camera/CameraBase.h
@@ -54,9 +54,10 @@ template <typename TCam, typename TCamTraits = CameraTraits<TCam> >
 class CameraBase : public IBinder::DeathRecipient
 {
 public:
-    typedef typename TCamTraits::TCamListener    TCamListener;
-    typedef typename TCamTraits::TCamUser        TCamUser;
-    typedef typename TCamTraits::TCamCallbacks   TCamCallbacks;
+    typedef typename TCamTraits::TCamListener       TCamListener;
+    typedef typename TCamTraits::TCamUser           TCamUser;
+    typedef typename TCamTraits::TCamCallbacks      TCamCallbacks;
+    typedef typename TCamTraits::TCamConnectService TCamConnectService;
 
     static sp<TCam>      connect(int cameraId,
                                  const String16& clientPackageName,
diff --git a/include/camera/CameraMetadata.h b/include/camera/CameraMetadata.h
index 8eeb2e7..fe2bd19 100644
--- a/include/camera/CameraMetadata.h
+++ b/include/camera/CameraMetadata.h
@@ -22,6 +22,7 @@
 #include <utils/Vector.h>
 
 namespace android {
+class Parcel;
 
 /**
  * A convenience wrapper around the C-based camera_metadata_t library.
@@ -159,6 +160,12 @@ class CameraMetadata {
     status_t erase(uint32_t tag);
 
     /**
+     * Swap the underlying camera metadata between this and the other
+     * metadata object.
+     */
+    void swap(CameraMetadata &other);
+
+    /**
      * Dump contents into FD for debugging. The verbosity levels are
      * 0: Tag entry information only, no data values
      * 1: Level 0 plus at most 16 data values per entry
@@ -169,6 +176,31 @@ class CameraMetadata {
      */
     void dump(int fd, int verbosity = 1, int indentation = 0) const;
 
+    /**
+     * Serialization over Binder
+     */
+
+    // Metadata object is unchanged when reading from parcel fails.
+    status_t readFromParcel(Parcel *parcel);
+    status_t writeToParcel(Parcel *parcel) const;
+
+    /**
+      * Caller becomes the owner of the new metadata
+      * 'const Parcel' doesnt prevent us from calling the read functions.
+      *  which is interesting since it changes the internal state
+      *
+      * NULL can be returned when no metadata was sent, OR if there was an issue
+      * unpacking the serialized data (i.e. bad parcel or invalid structure).
+      */
+    static status_t readFromParcel(const Parcel &parcel,
+                                   camera_metadata_t** out);
+    /**
+      * Caller retains ownership of metadata
+      * - Write 2 (int32 + blob) args in the current position
+      */
+    static status_t writeToParcel(Parcel &parcel,
+                                  const camera_metadata_t* metadata);
+
   private:
     camera_metadata_t *mBuffer;
     bool               mLocked;
diff --git a/include/camera/ICamera.h b/include/camera/ICamera.h
index 2236c1f..b025735 100644
--- a/include/camera/ICamera.h
+++ b/include/camera/ICamera.h
@@ -32,6 +32,9 @@ class Surface;
 
 class ICamera: public IInterface
 {
+    /**
+     * Keep up-to-date with ICamera.aidl in frameworks/base
+     */
 public:
     DECLARE_META_INTERFACE(Camera);
 
@@ -47,14 +50,21 @@ public:
     virtual status_t        unlock() = 0;
 
     // pass the buffered IGraphicBufferProducer to the camera service
-    virtual status_t        setPreviewTexture(
+    virtual status_t        setPreviewTarget(
             const sp<IGraphicBufferProducer>& bufferProducer) = 0;
 
     // set the preview callback flag to affect how the received frames from
-    // preview are handled.
+    // preview are handled. Enabling preview callback flags disables any active
+    // preview callback surface set by setPreviewCallbackTarget().
     virtual void            setPreviewCallbackFlag(int flag) = 0;
-
-    // start preview mode, must call setPreviewDisplay first
+    // set a buffer interface to use for client-received preview frames instead
+    // of preview callback buffers. Passing a valid interface here disables any
+    // active preview callbacks set by setPreviewCallbackFlag(). Passing NULL
+    // disables the use of the callback target.
+    virtual status_t        setPreviewCallbackTarget(
+            const sp<IGraphicBufferProducer>& callbackProducer) = 0;
+
+    // start preview mode, must call setPreviewTarget first
     virtual status_t        startPreview() = 0;
 
     // stop preview mode
diff --git a/include/camera/ICameraClient.h b/include/camera/ICameraClient.h
index b30aa7a..1584dba 100644
--- a/include/camera/ICameraClient.h
+++ b/include/camera/ICameraClient.h
@@ -28,6 +28,9 @@ namespace android {
 
 class ICameraClient: public IInterface
 {
+    /**
+     * Keep up-to-date with ICameraClient.aidl in frameworks/base
+     */
 public:
     DECLARE_META_INTERFACE(CameraClient);
 
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index aaf6eb3..f342122 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -28,17 +28,25 @@ class ICameraClient;
 class IProCameraUser;
 class IProCameraCallbacks;
 class ICameraServiceListener;
+class ICameraDeviceUser;
+class ICameraDeviceCallbacks;
+class CameraMetadata;
 
 class ICameraService : public IInterface
 {
 public:
+    /**
+     * Keep up-to-date with ICameraService.aidl in frameworks/base
+     */
     enum {
         GET_NUMBER_OF_CAMERAS = IBinder::FIRST_CALL_TRANSACTION,
         GET_CAMERA_INFO,
         CONNECT,
         CONNECT_PRO,
+        CONNECT_DEVICE,
         ADD_LISTENER,
         REMOVE_LISTENER,
+        GET_CAMERA_CHARACTERISTICS,
     };
 
     enum {
@@ -52,6 +60,9 @@ public:
     virtual status_t getCameraInfo(int cameraId,
                                           struct CameraInfo* cameraInfo) = 0;
 
+    virtual status_t getCameraCharacteristics(int cameraId,
+                                              CameraMetadata* cameraInfo) = 0;
+
     // Returns 'OK' if operation succeeded
     // - Errors: ALREADY_EXISTS if the listener was already added
     virtual status_t addListener(const sp<ICameraServiceListener>& listener)
@@ -65,15 +76,27 @@ public:
      * clientUid == USE_CALLING_UID, then the calling UID is used instead. Only
      * trusted callers can set a clientUid other than USE_CALLING_UID.
      */
-    virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient,
+    virtual status_t connect(const sp<ICameraClient>& cameraClient,
+            int cameraId,
+            const String16& clientPackageName,
+            int clientUid,
+            /*out*/
+            sp<ICamera>& device) = 0;
+
+    virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb,
             int cameraId,
             const String16& clientPackageName,
-            int clientUid) = 0;
+            int clientUid,
+            /*out*/
+            sp<IProCameraUser>& device) = 0;
 
-    virtual sp<IProCameraUser> connect(const sp<IProCameraCallbacks>& cameraCb,
+    virtual status_t connectDevice(
+            const sp<ICameraDeviceCallbacks>& cameraCb,
             int cameraId,
             const String16& clientPackageName,
-            int clientUid) = 0;
+            int clientUid,
+            /*out*/
+            sp<ICameraDeviceUser>& device) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/camera/ICameraServiceListener.h b/include/camera/ICameraServiceListener.h
index f2a11c2..0a0e43a 100644
--- a/include/camera/ICameraServiceListener.h
+++ b/include/camera/ICameraServiceListener.h
@@ -26,6 +26,9 @@ namespace android {
 
 class ICameraServiceListener : public IInterface
 {
+    /**
+     * Keep up-to-date with ICameraServiceListener.aidl in frameworks/base
+     */
 public:
 
     /**
diff --git a/include/camera/IProCameraCallbacks.h b/include/camera/IProCameraCallbacks.h
index 563ec17..e8abb89 100644
--- a/include/camera/IProCameraCallbacks.h
+++ b/include/camera/IProCameraCallbacks.h
@@ -30,6 +30,9 @@ namespace android {
 
 class IProCameraCallbacks : public IInterface
 {
+    /**
+     * Keep up-to-date with IProCameraCallbacks.aidl in frameworks/base
+     */
 public:
     DECLARE_META_INTERFACE(ProCameraCallbacks);
 
@@ -48,7 +51,7 @@ public:
     /** Missing by design: implementation is client-side in ProCamera.cpp **/
     // virtual void onBufferReceived(int streamId,
     //                               const CpuConsumer::LockedBufer& buf);
-    virtual void            onResultReceived(int32_t frameId,
+    virtual void            onResultReceived(int32_t requestId,
                                              camera_metadata* result) = 0;
 };
 
diff --git a/include/camera/IProCameraUser.h b/include/camera/IProCameraUser.h
index 45b818c..2ccc4d2 100644
--- a/include/camera/IProCameraUser.h
+++ b/include/camera/IProCameraUser.h
@@ -34,6 +34,9 @@ class Surface;
 
 class IProCameraUser: public IInterface
 {
+    /**
+     * Keep up-to-date with IProCameraUser.aidl in frameworks/base
+     */
 public:
     DECLARE_META_INTERFACE(ProCameraUser);
 
diff --git a/include/camera/ProCamera.h b/include/camera/ProCamera.h
index 3d1652f..83a3028 100644
--- a/include/camera/ProCamera.h
+++ b/include/camera/ProCamera.h
@@ -25,6 +25,7 @@
 #include <camera/IProCameraUser.h>
 #include <camera/Camera.h>
 #include <camera/CameraMetadata.h>
+#include <camera/ICameraService.h>
 #include <gui/CpuConsumer.h>
 
 #include <gui/Surface.h>
@@ -87,8 +88,14 @@ struct CameraTraits<ProCamera>
     typedef ProCameraListener     TCamListener;
     typedef IProCameraUser        TCamUser;
     typedef IProCameraCallbacks   TCamCallbacks;
+    typedef status_t (ICameraService::*TCamConnectService)(const sp<IProCameraCallbacks>&,
+                                                           int, const String16&, int,
+                                                           /*out*/
+                                                           sp<IProCameraUser>&);
+    static TCamConnectService     fnConnectService;
 };
 
+
 class ProCamera :
     public CameraBase<ProCamera>,
     public BnProCameraCallbacks
@@ -245,7 +252,7 @@ protected:
     virtual void        onLockStatusChanged(
                                 IProCameraCallbacks::LockStatus newLockStatus);
 
-    virtual void        onResultReceived(int32_t frameId,
+    virtual void        onResultReceived(int32_t requestId,
                                          camera_metadata* result);
 private:
     ProCamera(int cameraId);
diff --git a/include/camera/camera2/CaptureRequest.h b/include/camera/camera2/CaptureRequest.h
new file mode 100644
index 0000000..e56d61f
--- /dev/null
+++ b/include/camera/camera2/CaptureRequest.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_PHOTOGRAPHY_CAPTUREREQUEST_H
+#define ANDROID_HARDWARE_PHOTOGRAPHY_CAPTUREREQUEST_H
+
+#include <utils/RefBase.h>
+#include <utils/Vector.h>
+#include <camera/CameraMetadata.h>
+
+namespace android {
+
+class Surface;
+
+struct CaptureRequest : public virtual RefBase {
+public:
+
+    CameraMetadata          mMetadata;
+    Vector<sp<Surface> >    mSurfaceList;
+
+    /**
+     * Keep impl up-to-date with CaptureRequest.java in frameworks/base
+     */
+    status_t                readFromParcel(Parcel* parcel);
+    status_t                writeToParcel(Parcel* parcel) const;
+};
+}; // namespace android
+
+#endif
diff --git a/include/camera/camera2/ICameraDeviceCallbacks.h b/include/camera/camera2/ICameraDeviceCallbacks.h
new file mode 100644
index 0000000..8dac4f2
--- /dev/null
+++ b/include/camera/camera2/ICameraDeviceCallbacks.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_PHOTOGRAPHY_CALLBACKS_H
+#define ANDROID_HARDWARE_PHOTOGRAPHY_CALLBACKS_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <binder/IMemory.h>
+#include <utils/Timers.h>
+#include <system/camera.h>
+
+namespace android {
+class CameraMetadata;
+
+class ICameraDeviceCallbacks : public IInterface
+{
+    /**
+     * Keep up-to-date with ICameraDeviceCallbacks.aidl in frameworks/base
+     */
+public:
+    DECLARE_META_INTERFACE(CameraDeviceCallbacks);
+
+    /**
+     * Error codes for CAMERA_MSG_ERROR
+     */
+    enum CameraErrorCode {
+        ERROR_CAMERA_DISCONNECTED = 0,
+        ERROR_CAMERA_DEVICE = 1,
+        ERROR_CAMERA_SERVICE = 2
+    };
+
+    // One way
+    virtual void            onDeviceError(CameraErrorCode errorCode) = 0;
+
+    // One way
+    virtual void            onDeviceIdle() = 0;
+
+    // One way
+    virtual void            onCaptureStarted(int32_t requestId,
+                                             int64_t timestamp) = 0;
+
+    // One way
+    virtual void            onResultReceived(int32_t requestId,
+                                             const CameraMetadata& result) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnCameraDeviceCallbacks : public BnInterface<ICameraDeviceCallbacks>
+{
+public:
+    virtual status_t    onTransact( uint32_t code,
+                                    const Parcel& data,
+                                    Parcel* reply,
+                                    uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif
diff --git a/include/camera/camera2/ICameraDeviceUser.h b/include/camera/camera2/ICameraDeviceUser.h
new file mode 100644
index 0000000..f71f302
--- /dev/null
+++ b/include/camera/camera2/ICameraDeviceUser.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_PHOTOGRAPHY_ICAMERADEVICEUSER_H
+#define ANDROID_HARDWARE_PHOTOGRAPHY_ICAMERADEVICEUSER_H
+
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+
+struct camera_metadata;
+
+namespace android {
+
+class ICameraDeviceUserClient;
+class IGraphicBufferProducer;
+class Surface;
+class CaptureRequest;
+class CameraMetadata;
+
+class ICameraDeviceUser : public IInterface
+{
+    /**
+     * Keep up-to-date with ICameraDeviceUser.aidl in frameworks/base
+     */
+public:
+    DECLARE_META_INTERFACE(CameraDeviceUser);
+
+    virtual void            disconnect() = 0;
+
+    /**
+     * Request Handling
+     **/
+
+    virtual int             submitRequest(sp<CaptureRequest> request,
+                                          bool streaming = false) = 0;
+    virtual status_t        cancelRequest(int requestId) = 0;
+
+    virtual status_t        deleteStream(int streamId) = 0;
+    virtual status_t        createStream(
+            int width, int height, int format,
+            const sp<IGraphicBufferProducer>& bufferProducer) = 0;
+
+    // Create a request object from a template.
+    virtual status_t        createDefaultRequest(int templateId,
+                                                 /*out*/
+                                                 CameraMetadata* request) = 0;
+    // Get static camera metadata
+    virtual status_t        getCameraInfo(/*out*/
+                                          CameraMetadata* info) = 0;
+
+    // Wait until all the submitted requests have finished processing
+    virtual status_t        waitUntilIdle() =  0;
+
+    // Flush all pending and in-progress work as quickly as possible.
+    virtual status_t        flush() = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnCameraDeviceUser: public BnInterface<ICameraDeviceUser>
+{
+public:
+    virtual status_t    onTransact( uint32_t code,
+                                    const Parcel& data,
+                                    Parcel* reply,
+                                    uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif
diff --git a/include/cpustats/CentralTendencyStatistics.h b/include/cpustats/CentralTendencyStatistics.h
new file mode 100644
index 0000000..21b6981
--- /dev/null
+++ b/include/cpustats/CentralTendencyStatistics.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef _CENTRAL_TENDENCY_STATISTICS_H
+#define _CENTRAL_TENDENCY_STATISTICS_H
+
+#include <math.h>
+
+// Not multithread safe
+class CentralTendencyStatistics {
+
+public:
+
+    CentralTendencyStatistics() :
+            mMean(NAN), mMedian(NAN), mMinimum(INFINITY), mMaximum(-INFINITY), mN(0), mM2(0),
+            mVariance(NAN), mVarianceKnownForN(0), mStddev(NAN), mStddevKnownForN(0) { }
+
+    ~CentralTendencyStatistics() { }
+
+    // add x to the set of samples
+    void sample(double x);
+
+    // return the arithmetic mean of all samples so far
+    double mean() const { return mMean; }
+
+    // return the minimum of all samples so far
+    double minimum() const { return mMinimum; }
+
+    // return the maximum of all samples so far
+    double maximum() const { return mMaximum; }
+
+    // return the variance of all samples so far
+    double variance() const;
+
+    // return the standard deviation of all samples so far
+    double stddev() const;
+
+    // return the number of samples added so far
+    unsigned n() const { return mN; }
+
+    // reset the set of samples to be empty
+    void reset();
+
+private:
+    double mMean;
+    double mMedian;
+    double mMinimum;
+    double mMaximum;
+    unsigned mN;    // number of samples so far
+    double mM2;
+
+    // cached variance, and n at time of caching
+    mutable double mVariance;
+    mutable unsigned mVarianceKnownForN;
+
+    // cached standard deviation, and n at time of caching
+    mutable double mStddev;
+    mutable unsigned mStddevKnownForN;
+
+};
+
+#endif // _CENTRAL_TENDENCY_STATISTICS_H
diff --git a/include/cpustats/README.txt b/include/cpustats/README.txt
new file mode 100644
index 0000000..14439f0
--- /dev/null
+++ b/include/cpustats/README.txt
@@ -0,0 +1,6 @@
+This is a static library of CPU usage statistics, originally written
+for audio but most are not actually specific to audio.
+
+Requirements to be here:
+ * should be related to CPU usage statistics
+ * should be portable to host; avoid Android OS dependencies without a conditional
diff --git a/include/cpustats/ThreadCpuUsage.h b/include/cpustats/ThreadCpuUsage.h
new file mode 100644
index 0000000..9756844
--- /dev/null
+++ b/include/cpustats/ThreadCpuUsage.h
@@ -0,0 +1,140 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef _THREAD_CPU_USAGE_H
+#define _THREAD_CPU_USAGE_H
+
+#include <fcntl.h>
+#include <pthread.h>
+
+namespace android {
+
+// Track CPU usage for the current thread.
+// Units are in per-thread CPU ns, as reported by
+// clock_gettime(CLOCK_THREAD_CPUTIME_ID).  Simple usage: for cyclic
+// threads where you want to measure the execution time of the whole
+// cycle, just call sampleAndEnable() at the start of each cycle.
+// For acyclic threads, or for cyclic threads where you want to measure/track
+// only part of each cycle, call enable(), disable(), and/or setEnabled()
+// to demarcate the region(s) of interest, and then call sample() periodically.
+// This class is not thread-safe for concurrent calls from multiple threads;
+// the methods of this class may only be called by the current thread
+// which constructed the object.
+
+class ThreadCpuUsage
+{
+
+public:
+    ThreadCpuUsage() :
+        mIsEnabled(false),
+        mWasEverEnabled(false),
+        mAccumulator(0),
+        // mPreviousTs
+        // mMonotonicTs
+        mMonotonicKnown(false)
+        {
+            (void) pthread_once(&sOnceControl, &init);
+            for (int i = 0; i < sKernelMax; ++i) {
+                mCurrentkHz[i] = (uint32_t) ~0;   // unknown
+            }
+        }
+
+    ~ThreadCpuUsage() { }
+
+    // Return whether currently tracking CPU usage by current thread
+    bool isEnabled() const  { return mIsEnabled; }
+
+    // Enable tracking of CPU usage by current thread;
+    // any CPU used from this point forward will be tracked.
+    // Returns the previous enabled status.
+    bool enable()       { return setEnabled(true); }
+
+    // Disable tracking of CPU usage by current thread;
+    // any CPU used from this point forward will be ignored.
+    // Returns the previous enabled status.
+    bool disable()      { return setEnabled(false); }
+
+    // Set the enabled status and return the previous enabled status.
+    // This method is intended to be used for safe nested enable/disabling.
+    bool setEnabled(bool isEnabled);
+
+    // Add a sample point, and also enable tracking if needed.
+    // If tracking has never been enabled, then this call enables tracking but
+    // does _not_ add a sample -- it is not possible to add a sample the
+    // first time because there is no previous point to subtract from.
+    // Otherwise, if tracking is enabled,
+    // then adds a sample for tracked CPU ns since the previous
+    // sample, or since the first call to sampleAndEnable(), enable(), or
+    // setEnabled(true).  If there was a previous sample but tracking is
+    // now disabled, then adds a sample for the tracked CPU ns accumulated
+    // up until the most recent disable(), resets this accumulator, and then
+    // enables tracking.  Calling this method rather than enable() followed
+    // by sample() avoids a race condition for the first sample.
+    // Returns true if the sample 'ns' is valid, or false if invalid.
+    // Note that 'ns' is an output parameter passed by reference.
+    // The caller does not need to initialize this variable.
+    // The units are CPU nanoseconds consumed by current thread.
+    bool sampleAndEnable(double& ns);
+
+    // Add a sample point, but do not
+    // change the tracking enabled status.  If tracking has either never been
+    // enabled, or has never been enabled since the last sample, then log a warning
+    // and don't add sample.  Otherwise, adds a sample for tracked CPU ns since
+    // the previous sample or since the first call to sampleAndEnable(),
+    // enable(), or setEnabled(true) if no previous sample.
+    // Returns true if the sample is valid, or false if invalid.
+    // Note that 'ns' is an output parameter passed by reference.
+    // The caller does not need to initialize this variable.
+    // The units are CPU nanoseconds consumed by current thread.
+    bool sample(double& ns);
+
+    // Return the elapsed delta wall clock ns since initial enable or reset,
+    // as reported by clock_gettime(CLOCK_MONOTONIC).
+    long long elapsed() const;
+
+    // Reset elapsed wall clock.  Has no effect on tracking or accumulator.
+    void resetElapsed();
+
+    // Return current clock frequency for specified CPU, in kHz.
+    // You can get your CPU number using sched_getcpu(2).  Note that, unless CPU affinity
+    // has been configured appropriately, the CPU number can change.
+    // Also note that, unless the CPU governor has been configured appropriately,
+    // the CPU frequency can change.  And even if the CPU frequency is locked down
+    // to a particular value, that the frequency might still be adjusted
+    // to prevent thermal overload.  Therefore you should poll for your thread's
+    // current CPU number and clock frequency periodically.
+    uint32_t getCpukHz(int cpuNum);
+
+private:
+    bool mIsEnabled;                // whether tracking is currently enabled
+    bool mWasEverEnabled;           // whether tracking was ever enabled
+    long long mAccumulator;         // accumulated thread CPU time since last sample, in ns
+    struct timespec mPreviousTs;    // most recent thread CPU time, valid only if mIsEnabled is true
+    struct timespec mMonotonicTs;   // most recent monotonic time
+    bool mMonotonicKnown;           // whether mMonotonicTs has been set
+
+    static const int MAX_CPU = 8;
+    static int sScalingFds[MAX_CPU];// file descriptor per CPU for reading scaling_cur_freq
+    uint32_t mCurrentkHz[MAX_CPU];  // current CPU frequency in kHz, not static to avoid a race
+    static pthread_once_t sOnceControl;
+    static int sKernelMax;          // like MAX_CPU, but determined at runtime == cpu/kernel_max + 1
+    static void init();             // called once at first ThreadCpuUsage construction
+    static pthread_mutex_t sMutex;  // protects sScalingFds[] after initialization
+};
+
+}   // namespace android
+
+#endif //  _THREAD_CPU_USAGE_H
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index 43e4de7..ef392f0 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -26,6 +26,8 @@ class AudioBufferProvider
 {
 public:
 
+    // FIXME merge with AudioTrackShared::Buffer, AudioTrack::Buffer, and AudioRecord::Buffer
+    //       and rename getNextBuffer() to obtainBuffer()
     struct Buffer {
         Buffer() : raw(NULL), frameCount(0) { }
         union {
@@ -44,6 +46,19 @@ public:
     // pts is the local time when the next sample yielded by getNextBuffer
     // will be rendered.
     // Pass kInvalidPTS if the PTS is unknown or not applicable.
+    // On entry:
+    //  buffer              != NULL
+    //  buffer->raw         unused
+    //  buffer->frameCount  maximum number of desired frames
+    // On successful return:
+    //  status              NO_ERROR
+    //  buffer->raw         non-NULL pointer to buffer->frameCount contiguous available frames
+    //  buffer->frameCount  number of contiguous available frames at buffer->raw,
+    //                      0 < buffer->frameCount <= entry value
+    // On error return:
+    //  status              != NO_ERROR
+    //  buffer->raw         NULL
+    //  buffer->frameCount  0
     virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
 
     virtual void releaseBuffer(Buffer* buffer) = 0;
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 38c6548..052064d 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -14,31 +14,27 @@
  * limitations under the License.
  */
 
-#ifndef AUDIORECORD_H_
-#define AUDIORECORD_H_
+#ifndef ANDROID_AUDIORECORD_H
+#define ANDROID_AUDIORECORD_H
 
-#include <binder/IMemory.h>
 #include <cutils/sched_policy.h>
 #include <media/AudioSystem.h>
 #include <media/IAudioRecord.h>
-#include <system/audio.h>
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
 #include <utils/threads.h>
 
 namespace android {
 
+// ----------------------------------------------------------------------------
+
 class audio_track_cblk_t;
 class AudioRecordClientProxy;
 
 // ----------------------------------------------------------------------------
 
-class AudioRecord : virtual public RefBase
+class AudioRecord : public RefBase
 {
 public:
 
-    static const int DEFAULT_SAMPLE_RATE = 8000;
-
     /* Events used by AudioRecord callback function (callback_t).
      * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
      */
@@ -49,6 +45,8 @@ public:
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 3,          // Record head is at a new position
                                     // (See setPositionUpdatePeriod()).
+        EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
+                                    // voluntary invalidation by mediaserver, or mediaserver crash.
     };
 
     /* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -58,11 +56,17 @@ public:
     class Buffer
     {
     public:
+        // FIXME use m prefix
         size_t      frameCount;     // number of sample frames corresponding to size;
                                     // on input it is the number of frames available,
                                     // on output is the number of frames actually drained
+                                    // (currently ignored, but will make the primary field in future)
+
+        size_t      size;           // input/output in bytes == frameCount * frameSize
+                                    // FIXME this is redundant with respect to frameCount,
+                                    // and TRANSFER_OBTAIN mode is broken for 8-bit data
+                                    // since we don't define the frame format
 
-        size_t      size;           // total size in bytes == frameCount * frameSize
         union {
             void*       raw;
             short*      i16;        // signed 16-bit
@@ -84,6 +88,7 @@ public:
      *          - EVENT_OVERRUN: unused.
      *          - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
      *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
+     *          - EVENT_NEW_IAUDIORECORD: unused.
      */
 
     typedef void (*callback_t)(int event, void* user, void *info);
@@ -101,94 +106,112 @@ public:
                                       audio_format_t format,
                                       audio_channel_mask_t channelMask);
 
+    /* How data is transferred from AudioRecord
+     */
+    enum transfer_type {
+        TRANSFER_DEFAULT,   // not specified explicitly; determine from other parameters
+        TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
+        TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+        TRANSFER_SYNC,      // synchronous read()
+    };
+
     /* Constructs an uninitialized AudioRecord. No connection with
-     * AudioFlinger takes place.
+     * AudioFlinger takes place.  Use set() after this.
      */
                         AudioRecord();
 
     /* Creates an AudioRecord object and registers it with AudioFlinger.
      * Once created, the track needs to be started before it can be used.
-     * Unspecified values are set to the audio hardware's current
-     * values.
+     * Unspecified values are set to appropriate default values.
      *
      * Parameters:
      *
-     * inputSource:        Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).
-     * sampleRate:         Track sampling rate in Hz.
+     * inputSource:        Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT).
+     * sampleRate:         Data sink sampling rate in Hz.
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelMask:        Channel mask.
+     * channelMask:        Channel mask, such that audio_is_input_channel(channelMask) is true.
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the
      *                     latency of the track.  The actual size selected by the AudioRecord could
      *                     be larger if the requested size is not compatible with current audio HAL
      *                     latency.  Zero means to use a default value.
      * cbf:                Callback function. If not null, this function is called periodically
-     *                     to consume new PCM data.
+     *                     to consume new PCM data and inform of marker, position updates, etc.
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames are ready in record track output buffer.
      * sessionId:          Not yet supported.
+     * transferType:       How data is transferred from AudioRecord.
+     * flags:              See comments on audio_input_flags_t in <system/audio.h>
+     * threadCanCallJava:  Not present in parameter list, and so is fixed at false.
      */
 
                         AudioRecord(audio_source_t inputSource,
-                                    uint32_t sampleRate = 0,
-                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                    audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
+                                    uint32_t sampleRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
                                     int frameCount      = 0,
                                     callback_t cbf = NULL,
                                     void* user = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId = 0);
-
+                                    int sessionId = 0,
+                                    transfer_type transferType = TRANSFER_DEFAULT,
+                                    audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
     /* Terminates the AudioRecord and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioRecord.
      */
-                        ~AudioRecord();
-
+protected:
+                        virtual ~AudioRecord();
+public:
 
-    /* Initialize an uninitialized AudioRecord.
+    /* Initialize an AudioRecord that was created using the AudioRecord() constructor.
+     * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters.
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
-     *  - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use
+     *  - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
      *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - PERMISSION_DENIED: recording is not allowed for the requesting process
+     *
+     * Parameters not listed in the AudioRecord constructors above:
+     *
+     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
      */
-            status_t    set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT,
-                            uint32_t sampleRate = 0,
-                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                            audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
+            status_t    set(audio_source_t inputSource,
+                            uint32_t sampleRate,
+                            audio_format_t format,
+                            audio_channel_mask_t channelMask,
                             int frameCount      = 0,
                             callback_t cbf = NULL,
                             void* user = NULL,
                             int notificationFrames = 0,
                             bool threadCanCallJava = false,
-                            int sessionId = 0);
-
+                            int sessionId = 0,
+                            transfer_type transferType = TRANSFER_DEFAULT,
+                            audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
     /* Result of constructing the AudioRecord. This must be checked
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
      * See set() method above for possible return codes.
      */
-            status_t    initCheck() const;
+            status_t    initCheck() const   { return mStatus; }
 
     /* Returns this track's estimated latency in milliseconds.
      * This includes the latency due to AudioRecord buffer size,
      * and audio hardware driver.
      */
-            uint32_t     latency() const;
+            uint32_t    latency() const     { return mLatency; }
 
    /* getters, see constructor and set() */
 
-            audio_format_t format() const;
-            uint32_t    channelCount() const;
-            size_t      frameCount() const;
-            size_t      frameSize() const { return mFrameSize; }
-            audio_source_t inputSource() const;
-
+            audio_format_t format() const   { return mFormat; }
+            uint32_t    channelCount() const    { return mChannelCount; }
+            size_t      frameCount() const  { return mFrameCount; }
+            size_t      frameSize() const   { return mFrameSize; }
+            audio_source_t inputSource() const  { return mInputSource; }
 
     /* After it's created the track is not active. Call start() to
      * make it active. If set, the callback will start being called.
@@ -198,26 +221,29 @@ public:
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
                               int triggerSession = 0);
 
-    /* Stop a track. If set, the callback will cease being called and
-     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
-     * and will drain buffers until the pool is exhausted.
+    /* Stop a track. If set, 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.
      */
             void        stop();
             bool        stopped() const;
 
-    /* Get sample rate for this record track in Hz.
+    /* Return the sink sample rate for this record track in Hz.
+     * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock.
      */
-            uint32_t    getSampleRate() const;
+            uint32_t    getSampleRate() const   { return mSampleRate; }
 
     /* Sets marker position. When record reaches the number of frames specified,
      * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
      * with marker == 0 cancels marker notification callback.
+     * To set a marker at a position which would compute as 0,
+     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
      * Parameters:
      *
-     * marker:   marker position expressed in frames.
+     * marker:   marker position expressed in wrapping (overflow) frame units,
+     *           like the return value of getPosition().
      *
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful operation
@@ -226,13 +252,13 @@ public:
             status_t    setMarkerPosition(uint32_t marker);
             status_t    getMarkerPosition(uint32_t *marker) const;
 
-
     /* Sets position update period. Every time the number of frames specified has been recorded,
      * a callback with event type EVENT_NEW_POS is called.
      * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
      * callback.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
+     * Extremely small values may be rounded up to a value the implementation can support.
      *
      * Parameters:
      *
@@ -245,13 +271,13 @@ public:
             status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
             status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;
 
-
-    /* Gets record head position. The position is the total number of frames
-     * recorded since record start.
+    /* Return the total number of frames recorded since recording started.
+     * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
+     * It is reset to zero by stop().
      *
      * Parameters:
      *
-     *  position:  Address where to return record head position within AudioRecord buffer.
+     *  position:  Address where to return record head position.
      *
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful operation
@@ -276,38 +302,74 @@ public:
      *
      * Returned value:
      *  AudioRecord session ID.
+     *
+     * No lock needed because session ID doesn't change after first set().
      */
-            int    getSessionId() const;
-
-    /* Obtains a buffer of "frameCount" frames. The buffer must be
-     * drained entirely, and then released with releaseBuffer().
-     * If the track is stopped, obtainBuffer() returns
-     * STOPPED instead of NO_ERROR as long as there are buffers available,
-     * at which point NO_MORE_BUFFERS is returned.
+            int    getSessionId() const { return mSessionId; }
+
+    /* Obtains a buffer of up to "audioBuffer->frameCount" full frames.
+     * After draining these frames of data, the caller should release them with releaseBuffer().
+     * If the track buffer is not empty, obtainBuffer() returns as many contiguous
+     * full frames as are available immediately.
+     * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK
+     * regardless of the value of waitCount.
+     * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a
+     * maximum timeout based on waitCount; see chart below.
      * Buffers will be returned until the pool
      * is exhausted, at which point obtainBuffer() will either block
-     * or return WOULD_BLOCK depending on the value of the "blocking"
+     * or return WOULD_BLOCK depending on the value of the "waitCount"
      * parameter.
      *
+     * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
+     * which should use read() or callback EVENT_MORE_DATA instead.
+     *
      * Interpretation of waitCount:
      *  +n  limits wait time to n * WAIT_PERIOD_MS,
      *  -1  causes an (almost) infinite wait time,
      *   0  non-blocking.
+     *
+     * Buffer fields
+     * On entry:
+     *  frameCount  number of frames requested
+     * After error return:
+     *  frameCount  0
+     *  size        0
+     *  raw         undefined
+     * After successful return:
+     *  frameCount  actual number of frames available, <= number requested
+     *  size        actual number of bytes available
+     *  raw         pointer to the buffer
      */
 
-        enum {
-            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
-            STOPPED = 1
-        };
+    /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
+            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+                                __attribute__((__deprecated__));
 
-            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+private:
+    /* If nonContig is non-NULL, it is an output parameter that will be set to the number of
+     * additional non-contiguous frames that are available immediately.
+     * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
+     * in case the requested amount of frames is in two or more non-contiguous regions.
+     * FIXME requested and elapsed are both relative times.  Consider changing to absolute time.
+     */
+            status_t    obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+                                     struct timespec *elapsed = NULL, size_t *nonContig = NULL);
+public:
 
-    /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */
+    /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */
+    // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
             void        releaseBuffer(Buffer* audioBuffer);
 
-
     /* As a convenience we provide a read() interface to the audio buffer.
-     * This is implemented on top of obtainBuffer/releaseBuffer.
+     * Input parameter 'size' is in byte units.
+     * This is implemented on top of obtainBuffer/releaseBuffer. For best
+     * performance use callbacks. Returns actual number of bytes read >= 0,
+     * or one of the following negative status codes:
+     *      INVALID_OPERATION   AudioRecord is configured for streaming mode
+     *      BAD_VALUE           size is invalid
+     *      WOULD_BLOCK         when obtainBuffer() returns same, or
+     *                          AudioRecord was stopped during the read
+     *      or any other error code returned by IAudioRecord::start() or restoreRecord_l().
      */
             ssize_t     read(void* buffer, size_t size);
 
@@ -338,66 +400,113 @@ private:
                 void        resume();   // allow thread to execute, if not requested to exit
 
     private:
+                void        pauseInternal(nsecs_t ns = 0LL);
+                                        // like pause(), but only used internally within thread
+
         friend class AudioRecord;
         virtual bool        threadLoop();
-        AudioRecord& mReceiver;
+        AudioRecord&        mReceiver;
         virtual ~AudioRecordThread();
         Mutex               mMyLock;    // Thread::mLock is private
         Condition           mMyCond;    // Thread::mThreadExitedCondition is private
-        bool                mPaused;    // whether thread is currently paused
+        bool                mPaused;    // whether thread is requested to pause at next loop entry
+        bool                mPausedInt; // whether thread internally requests pause
+        nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
     };
 
             // body of AudioRecordThread::threadLoop()
-            bool processAudioBuffer(const sp<AudioRecordThread>& thread);
+            // returns the maximum amount of time before we would like to run again, where:
+            //      0           immediately
+            //      > 0         no later than this many nanoseconds from now
+            //      NS_WHENEVER still active but no particular deadline
+            //      NS_INACTIVE inactive so don't run again until re-started
+            //      NS_NEVER    never again
+            static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
+            nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+
+            // caller must hold lock on mLock for all _l methods
+            status_t openRecord_l(size_t epoch);
 
-            status_t openRecord_l(uint32_t sampleRate,
-                                audio_format_t format,
-                                size_t frameCount,
-                                audio_io_handle_t input);
-            audio_io_handle_t getInput_l();
-            status_t restoreRecord_l(audio_track_cblk_t*& cblk);
+            // FIXME enum is faster than strcmp() for parameter 'from'
+            status_t restoreRecord_l(const char *from);
 
     sp<AudioRecordThread>   mAudioRecordThread;
     mutable Mutex           mLock;
 
-    bool                    mActive;            // protected by mLock
+    // Current client state:  false = stopped, true = active.  Protected by mLock.  If more states
+    // are added, consider changing this to enum State { ... } mState as in AudioTrack.
+    bool                    mActive;
 
     // for client callback handler
     callback_t              mCbf;               // callback handler for events, or NULL
     void*                   mUserData;
 
     // for notification APIs
-    uint32_t                mNotificationFrames;
-    uint32_t                mRemainingFrames;
-    uint32_t                mMarkerPosition;    // in frames
+    uint32_t                mNotificationFramesReq; // requested number of frames between each
+                                                    // notification callback
+    uint32_t                mNotificationFramesAct; // actual number of frames between each
+                                                    // notification callback
+    bool                    mRefreshRemaining;  // processAudioBuffer() should refresh next 2
+
+    // These are private to processAudioBuffer(), and are not protected by a lock
+    uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
+    bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
+    int                     mObservedSequence;      // last observed value of mSequence
+
+    uint32_t                mMarkerPosition;    // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;       // in frames
-    uint32_t                mUpdatePeriod;      // in ms
+    uint32_t                mUpdatePeriod;      // in frames, zero means no EVENT_NEW_POS
+
+    status_t                mStatus;
 
     // constant after constructor or set()
     uint32_t                mSampleRate;
     size_t                  mFrameCount;
     audio_format_t          mFormat;
-    uint8_t                 mChannelCount;
+    uint32_t                mChannelCount;
     size_t                  mFrameSize;         // app-level frame size == AudioFlinger frame size
     audio_source_t          mInputSource;
-    status_t                mStatus;
-    uint32_t                mLatency;
+    uint32_t                mLatency;           // in ms
     audio_channel_mask_t    mChannelMask;
-    audio_io_handle_t       mInput;                     // returned by AudioSystem::getInput()
+    audio_input_flags_t     mFlags;
     int                     mSessionId;
+    transfer_type           mTransfer;
+
+    audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
 
     // may be changed if IAudioRecord object is re-created
     sp<IAudioRecord>        mAudioRecord;
     sp<IMemory>             mCblkMemory;
-    audio_track_cblk_t*     mCblk;
-    void*                   mBuffers;           // starting address of buffers in shared memory
+    audio_track_cblk_t*     mCblk;              // re-load after mLock.unlock()
 
-    int                     mPreviousPriority;          // before start()
+    int                     mPreviousPriority;  // before start()
     SchedPolicy             mPreviousSchedulingGroup;
-    AudioRecordClientProxy* mProxy;
+    bool                    mAwaitBoost;    // thread should wait for priority boost before running
+
+    // The proxy should only be referenced while a lock is held because the proxy isn't
+    // multi-thread safe.
+    // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock,
+    // provided that the caller also holds an extra reference to the proxy and shared memory to keep
+    // them around in case they are replaced during the obtainBuffer().
+    sp<AudioRecordClientProxy> mProxy;
+
+    bool                    mInOverrun;         // whether recorder is currently in overrun state
+
+private:
+    class DeathNotifier : public IBinder::DeathRecipient {
+    public:
+        DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { }
+    protected:
+        virtual void        binderDied(const wp<IBinder>& who);
+    private:
+        const wp<AudioRecord> mAudioRecord;
+    };
+
+    sp<DeathNotifier>       mDeathNotifier;
+    uint32_t                mSequence;              // incremented for each new IAudioRecord attempt
 };
 
 }; // namespace android
 
-#endif /*AUDIORECORD_H_*/
+#endif // ANDROID_AUDIORECORD_H
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b11c812..225ef76 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -17,20 +17,18 @@
 #ifndef ANDROID_AUDIOSYSTEM_H_
 #define ANDROID_AUDIOSYSTEM_H_
 
-#include <utils/RefBase.h>
-#include <utils/threads.h>
-#include <media/IAudioFlinger.h>
-
+#include <hardware/audio_effect.h>
+#include <media/IAudioFlingerClient.h>
 #include <system/audio.h>
 #include <system/audio_policy.h>
-
-/* XXX: Should be include by all the users instead */
-#include <media/AudioParameter.h>
+#include <utils/Errors.h>
+#include <utils/Mutex.h>
 
 namespace android {
 
 typedef void (*audio_error_callback)(status_t err);
 
+class IAudioFlinger;
 class IAudioPolicyService;
 class String8;
 
@@ -128,8 +126,10 @@ public:
     // - BAD_VALUE: invalid parameter
     // NOTE: this feature is not supported on all hardware platforms and it is
     // necessary to check returned status before using the returned values.
-    static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getRenderPosition(audio_io_handle_t output,
+                                      size_t *halFrames,
+                                      size_t *dspFrames,
+                                      audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
 
     // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
     static size_t getInputFramesLost(audio_io_handle_t ioHandle);
@@ -155,11 +155,11 @@ public:
     class OutputDescriptor {
     public:
         OutputDescriptor()
-        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0)  {}
+        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)  {}
 
         uint32_t samplingRate;
-        int32_t format;
-        int32_t channels;
+        audio_format_t format;
+        audio_channel_mask_t channelMask;
         size_t frameCount;
         uint32_t latency;
     };
@@ -197,7 +197,8 @@ public:
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
+                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                        const audio_offload_info_t *offloadInfo = NULL);
     static status_t startOutput(audio_io_handle_t output,
                                 audio_stream_type_t stream,
                                 int session = 0);
@@ -245,6 +246,15 @@ public:
     static uint32_t getPrimaryOutputSamplingRate();
     static size_t getPrimaryOutputFrameCount();
 
+    static status_t setLowRamDevice(bool isLowRamDevice);
+
+    // Check if hw offload is possible for given format, stream type, sample rate,
+    // bit rate, duration, video and streaming or offload property is enabled
+    static bool isOffloadSupported(const audio_offload_info_t& info);
+
+    // check presence of audio flinger service.
+    // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
+    static status_t checkAudioFlinger();
     // ----------------------------------------------------------------------------
 
 private:
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
new file mode 100644
index 0000000..c29c7e5
--- /dev/null
+++ b/include/media/AudioTimestamp.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_TIMESTAMP_H
+#define ANDROID_AUDIO_TIMESTAMP_H
+
+#include <time.h>
+
+class AudioTimestamp {
+public:
+    AudioTimestamp() : mPosition(0) {
+        mTime.tv_sec = 0;
+        mTime.tv_nsec = 0;
+    }
+    // FIXME change type to match android.media.AudioTrack
+    uint32_t        mPosition; // a frame position in AudioTrack::getPosition() units
+    struct timespec mTime;     // corresponding CLOCK_MONOTONIC when frame is expected to present
+};
+
+#endif  // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 64f82bb..f2f9c22 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -17,18 +17,10 @@
 #ifndef ANDROID_AUDIOTRACK_H
 #define ANDROID_AUDIOTRACK_H
 
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <media/IAudioFlinger.h>
-#include <media/IAudioTrack.h>
-#include <media/AudioSystem.h>
-
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <binder/IInterface.h>
-#include <binder/IMemory.h>
 #include <cutils/sched_policy.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTimestamp.h>
+#include <media/IAudioTrack.h>
 #include <utils/threads.h>
 
 namespace android {
@@ -37,10 +29,11 @@ namespace android {
 
 class audio_track_cblk_t;
 class AudioTrackClientProxy;
+class StaticAudioTrackClientProxy;
 
 // ----------------------------------------------------------------------------
 
-class AudioTrack : virtual public RefBase
+class AudioTrack : public RefBase
 {
 public:
     enum channel_index {
@@ -49,7 +42,7 @@ public:
         RIGHT  = 1
     };
 
-    /* Events used by AudioTrack callback function (audio_track_cblk_t).
+    /* Events used by AudioTrack callback function (callback_t).
      * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
      */
     enum event_type {
@@ -64,7 +57,15 @@ public:
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 4,          // Playback head is at a new position
                                     // (See setPositionUpdatePeriod()).
-        EVENT_BUFFER_END = 5        // Playback head is at the end of the buffer.
+        EVENT_BUFFER_END = 5,       // Playback head is at the end of the buffer.
+                                    // Not currently used by android.media.AudioTrack.
+        EVENT_NEW_IAUDIOTRACK = 6,  // IAudioTrack was re-created, either due to re-routing and
+                                    // voluntary invalidation by mediaserver, or mediaserver crash.
+        EVENT_STREAM_END = 7,       // Sent after all the buffers queued in AF and HW are played
+                                    // back (after stop is called)
+        EVENT_NEW_TIMESTAMP = 8,    // Delivered periodically and when there's a significant change
+                                    // in the mapping from frame position to presentation time.
+                                    // See AudioTimestamp for the information included with event.
     };
 
     /* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -74,19 +75,25 @@ public:
     class Buffer
     {
     public:
+        // FIXME use m prefix
         size_t      frameCount;   // number of sample frames corresponding to size;
                                   // on input it is the number of frames desired,
                                   // on output is the number of frames actually filled
+                                  // (currently ignored, but will make the primary field in future)
+
+        size_t      size;         // input/output in bytes == frameCount * frameSize
+                                  // on output is the number of bytes actually filled
+                                  // FIXME this is redundant with respect to frameCount,
+                                  // and TRANSFER_OBTAIN mode is broken for 8-bit data
+                                  // since we don't define the frame format
 
-        size_t      size;         // input/output in byte units
         union {
             void*       raw;
-            short*      i16;    // signed 16-bit
-            int8_t*     i8;     // unsigned 8-bit, offset by 0x80
+            short*      i16;      // signed 16-bit
+            int8_t*     i8;       // unsigned 8-bit, offset by 0x80
         };
     };
 
-
     /* As a convenience, if a callback is supplied, a handler thread
      * is automatically created with the appropriate priority. This thread
      * invokes the callback when a new buffer becomes available or various conditions occur.
@@ -100,9 +107,12 @@ public:
      *            written.
      *          - EVENT_UNDERRUN: unused.
      *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
-     *          - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
-     *          - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
+     *          - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
+     *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
      *          - EVENT_BUFFER_END: unused.
+     *          - EVENT_NEW_IAUDIOTRACK: unused.
+     *          - EVENT_STREAM_END: unused.
+     *          - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
      */
 
     typedef void (*callback_t)(int event, void* user, void *info);
@@ -112,11 +122,22 @@ public:
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
+     *  - BAD_VALUE: unsupported configuration
      */
 
-     static status_t getMinFrameCount(size_t* frameCount,
-                                      audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
-                                      uint32_t sampleRate = 0);
+    static status_t getMinFrameCount(size_t* frameCount,
+                                     audio_stream_type_t streamType,
+                                     uint32_t sampleRate);
+
+    /* How data is transferred to AudioTrack
+     */
+    enum transfer_type {
+        TRANSFER_DEFAULT,   // not specified explicitly; determine from the other parameters
+        TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
+        TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+        TRANSFER_SYNC,      // synchronous write()
+        TRANSFER_SHARED,    // shared memory
+    };
 
     /* Constructs an uninitialized AudioTrack. No connection with
      * AudioFlinger takes place.  Use set() after this.
@@ -128,13 +149,13 @@ public:
      * Unspecified values are set to appropriate default values.
      * With this constructor, the track is configured for streaming mode.
      * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA.
-     * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is deprecated.
+     * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed.
      *
      * Parameters:
      *
      * streamType:         Select the type of audio stream this track is attached to
      *                     (e.g. AUDIO_STREAM_MUSIC).
-     * sampleRate:         Track sampling rate in Hz.
+     * sampleRate:         Data source sampling rate in Hz.
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
      * channelMask:        Channel mask.
@@ -149,21 +170,24 @@ public:
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames have been consumed from track input buffer.
+     *                     This is expressed in units of frames at the initial source sample rate.
      * sessionId:          Specific session ID, or zero to use default.
-     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
-     *                     If not present in parameter list, then fixed at false.
+     * transferType:       How data is transferred to AudioTrack.
+     * threadCanCallJava:  Not present in parameter list, and so is fixed at false.
      */
 
                         AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate  = 0,
-                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                    audio_channel_mask_t channelMask = 0,
+                                    uint32_t sampleRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t,
                                     int frameCount       = 0,
                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                     callback_t cbf       = NULL,
                                     void* user           = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId        = 0);
+                                    int sessionId        = 0,
+                                    transfer_type transferType = TRANSFER_DEFAULT,
+                                    const audio_offload_info_t *offloadInfo = NULL);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -174,38 +198,47 @@ public:
      * The write() method is not supported in this case.
      * It is recommended to pass a callback function to be notified of playback end by an
      * EVENT_UNDERRUN event.
-     * FIXME EVENT_MORE_DATA still occurs; it must be ignored.
      */
 
                         AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate = 0,
-                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                    audio_channel_mask_t channelMask = 0,
-                                    const sp<IMemory>& sharedBuffer = 0,
+                                    uint32_t sampleRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    const sp<IMemory>& sharedBuffer,
                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                     callback_t cbf      = NULL,
                                     void* user          = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId       = 0);
+                                    int sessionId       = 0,
+                                    transfer_type transferType = TRANSFER_DEFAULT,
+                                    const audio_offload_info_t *offloadInfo = NULL);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
      */
-                        ~AudioTrack();
+protected:
+                        virtual ~AudioTrack();
+public:
 
-    /* Initialize an uninitialized AudioTrack.
+    /* Initialize an AudioTrack that was created using the AudioTrack() constructor.
+     * Don't call set() more than once, or after the AudioTrack() constructors that take parameters.
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful initialization
      *  - INVALID_OPERATION: AudioTrack is already initialized
      *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
+     * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack.
      * If sharedBuffer is non-0, the frameCount parameter is ignored and
      * replaced by the shared buffer's total allocated size in frame units.
+     *
+     * Parameters not listed in the AudioTrack constructors above:
+     *
+     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
      */
-            status_t    set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
-                            uint32_t sampleRate = 0,
-                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                            audio_channel_mask_t channelMask = 0,
+            status_t    set(audio_stream_type_t streamType,
+                            uint32_t sampleRate,
+                            audio_format_t format,
+                            audio_channel_mask_t channelMask,
                             int frameCount      = 0,
                             audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                             callback_t cbf      = NULL,
@@ -213,9 +246,11 @@ public:
                             int notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
-                            int sessionId       = 0);
+                            int sessionId       = 0,
+                            transfer_type transferType = TRANSFER_DEFAULT,
+                            const audio_offload_info_t *offloadInfo = NULL);
 
-    /* Result of constructing the AudioTrack. This must be checked
+    /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
      * an uninitialized AudioTrack produces undefined results.
      * See set() method above for possible return codes.
@@ -233,14 +268,15 @@ public:
             audio_stream_type_t streamType() const { return mStreamType; }
             audio_format_t format() const   { return mFormat; }
 
-    /* Return frame size in bytes, which for linear PCM is channelCount * (bit depth per channel / 8).
+    /* Return frame size in bytes, which for linear PCM is
+     * channelCount * (bit depth per channel / 8).
      * channelCount is determined from channelMask, and bit depth comes from format.
      * For non-linear formats, the frame size is typically 1 byte.
      */
-            uint32_t    channelCount() const { return mChannelCount; }
+            size_t      frameSize() const   { return mFrameSize; }
 
+            uint32_t    channelCount() const { return mChannelCount; }
             uint32_t    frameCount() const  { return mFrameCount; }
-            size_t      frameSize() const   { return mFrameSize; }
 
     /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
@@ -249,14 +285,13 @@ public:
      * make it active. If set, the callback will start being called.
      * If the track was previously paused, volume is ramped up over the first mix buffer.
      */
-            void        start();
+            status_t        start();
 
     /* Stop a track.
      * In static buffer mode, the track is stopped immediately.
-     * In streaming mode, the callback will cease being called and
-     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
-     * and will fill up buffers until the pool is exhausted.
-     * The stop does not occur immediately: any data remaining in the buffer
+     * In streaming mode, the callback will cease being called.  Note that obtainBuffer() still
+     * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK.
+     * In streaming mode the stop does not occur immediately: any data remaining in the buffer
      * is first drained, mixed, and output, and only then is the track marked as stopped.
      */
             void        stop();
@@ -270,7 +305,7 @@ public:
             void        flush();
 
     /* Pause a track. After pause, the callback will cease being called and
-     * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
+     * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works
      * and will fill up buffers until the pool is exhausted.
      * Volume is ramped down over the next mix buffer following the pause request,
      * and then the track is marked as paused.  It can be resumed with ramp up by start().
@@ -294,11 +329,11 @@ public:
             status_t    setAuxEffectSendLevel(float level);
             void        getAuxEffectSendLevel(float* level) const;
 
-    /* Set sample rate for this track in Hz, mostly used for games' sound effects
+    /* Set source sample rate for this track in Hz, mostly used for games' sound effects
      */
             status_t    setSampleRate(uint32_t sampleRate);
 
-    /* Return current sample rate in Hz, or 0 if unknown */
+    /* Return current source sample rate in Hz, or 0 if unknown */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -306,20 +341,24 @@ public:
      *
      * Parameters:
      *
-     * loopStart:   loop start expressed as the number of PCM frames played since AudioTrack start.
-     * loopEnd:     loop end expressed as the number of PCM frames played since AudioTrack start.
+     * loopStart:   loop start in frames relative to start of buffer.
+     * loopEnd:     loop end in frames relative to start of buffer.
      * loopCount:   number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
-     *              pending or active loop. loopCount = -1 means infinite looping.
+     *              pending or active loop. loopCount == -1 means infinite looping.
      *
      * For proper operation the following condition must be respected:
-     *          (loopEnd-loopStart) <= framecount()
+     *      loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount().
+     *
+     * If the loop period (loopEnd - loopStart) is too small for the implementation to support,
+     * setLoop() will return BAD_VALUE.  loopCount must be >= -1.
+     *
      */
             status_t    setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
 
     /* Sets marker position. When playback reaches the number of frames specified, a callback with
      * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
      * notification callback.  To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as -1 or 1.
+     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
      * If the AudioTrack has been opened with no callback function associated, the operation will
      * fail.
      *
@@ -354,18 +393,14 @@ public:
             status_t    setPositionUpdatePeriod(uint32_t updatePeriod);
             status_t    getPositionUpdatePeriod(uint32_t *updatePeriod) const;
 
-    /* Sets playback head position within AudioTrack buffer. The new position is specified
-     * in number of frames.
-     * This method must be called with the AudioTrack in paused or stopped state.
-     * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
-     * Therefore using this method makes sense only when playing a "static" audio buffer
-     * as opposed to streaming.
-     * The getPosition() method on the other hand returns the total number of frames played since
-     * playback start.
+    /* Sets playback head position.
+     * Only supported for static buffer mode.
      *
      * Parameters:
      *
-     * position:  New playback head position within AudioTrack buffer.
+     * position:  New playback head position in frames relative to start of buffer.
+     *            0 <= position <= frameCount().  Note that end of buffer is permitted,
+     *            but will result in an immediate underrun if started.
      *
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful operation
@@ -378,8 +413,22 @@ public:
     /* Return the total number of frames played since playback start.
      * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
      * It is reset to zero by flush(), reload(), and stop().
+     *
+     * Parameters:
+     *
+     *  position:  Address where to return play head position.
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *  - NO_ERROR: successful operation
+     *  - BAD_VALUE:  position is NULL
      */
-            status_t    getPosition(uint32_t *position);
+            status_t    getPosition(uint32_t *position) const;
+
+    /* For static buffer mode only, this returns the current playback position in frames
+     * relative to start of buffer.  It is analogous to the position units used by
+     * setLoop() and setPosition().  After underrun, the position will be at end of buffer.
+     */
+            status_t    getBufferPosition(uint32_t *position);
 
     /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
      * rewriting the buffer before restarting playback after a stop.
@@ -426,15 +475,19 @@ public:
      */
             status_t    attachAuxEffect(int effectId);
 
-    /* Obtains a buffer of "frameCount" frames. The buffer must be
-     * filled entirely, and then released with releaseBuffer().
-     * If the track is stopped, obtainBuffer() returns
-     * STOPPED instead of NO_ERROR as long as there are buffers available,
-     * at which point NO_MORE_BUFFERS is returned.
+    /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
+     * After filling these slots with data, the caller should release them with releaseBuffer().
+     * If the track buffer is not full, obtainBuffer() returns as many contiguous
+     * [empty slots for] frames as are available immediately.
+     * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK
+     * regardless of the value of waitCount.
+     * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a
+     * maximum timeout based on waitCount; see chart below.
      * Buffers will be returned until the pool
      * is exhausted, at which point obtainBuffer() will either block
-     * or return WOULD_BLOCK depending on the value of the "blocking"
+     * or return WOULD_BLOCK depending on the value of the "waitCount"
      * parameter.
+     * Each sample is 16-bit signed PCM.
      *
      * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
      * which should use write() or callback EVENT_MORE_DATA instead.
@@ -457,33 +510,76 @@ public:
      *  raw         pointer to the buffer
      */
 
-        enum {
-            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
-            STOPPED = 1
-        };
+    /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
+            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+                                __attribute__((__deprecated__));
 
-            status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+private:
+    /* If nonContig is non-NULL, it is an output parameter that will be set to the number of
+     * additional non-contiguous frames that are available immediately.
+     * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
+     * in case the requested amount of frames is in two or more non-contiguous regions.
+     * FIXME requested and elapsed are both relative times.  Consider changing to absolute time.
+     */
+            status_t    obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+                                     struct timespec *elapsed = NULL, size_t *nonContig = NULL);
+public:
 
-    /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
+//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
+//            enum {
+//            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
+//            TEAR_DOWN       = 0x80000002,
+//            STOPPED = 1,
+//            STREAM_END_WAIT,
+//            STREAM_END
+//        };
+
+    /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
+    // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
             void        releaseBuffer(Buffer* audioBuffer);
 
     /* As a convenience we provide a write() interface to the audio buffer.
+     * Input parameter 'size' is in byte units.
      * This is implemented on top of obtainBuffer/releaseBuffer. For best
      * performance use callbacks. Returns actual number of bytes written >= 0,
      * or one of the following negative status codes:
-     *      INVALID_OPERATION   AudioTrack is configured for shared buffer mode
+     *      INVALID_OPERATION   AudioTrack is configured for static buffer or streaming mode
      *      BAD_VALUE           size is invalid
-     *      STOPPED             AudioTrack was stopped during the write
-     *      NO_MORE_BUFFERS     when obtainBuffer() returns same
+     *      WOULD_BLOCK         when obtainBuffer() returns same, or
+     *                          AudioTrack was stopped during the write
      *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
-     * Not supported for static buffer mode.
      */
             ssize_t     write(const void* buffer, size_t size);
 
     /*
      * Dumps the state of an audio track.
      */
-            status_t dump(int fd, const Vector<String16>& args) const;
+            status_t    dump(int fd, const Vector<String16>& args) const;
+
+    /*
+     * Return the total number of frames which AudioFlinger desired but were unavailable,
+     * and thus which resulted in an underrun.  Reset to zero by stop().
+     */
+            uint32_t    getUnderrunFrames() const;
+
+    /* Get the flags */
+            audio_output_flags_t getFlags() const { return mFlags; }
+
+    /* Set parameters - only possible when using direct output */
+            status_t    setParameters(const String8& keyValuePairs);
+
+    /* Get parameters */
+            String8     getParameters(const String8& keys);
+
+    /* Poll for a timestamp on demand.
+     * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs,
+     * or if you need to get the most recent timestamp outside of the event callback handler.
+     * Caution: calling this method too often may be inefficient;
+     * if you need a high resolution mapping between frame position and presentation time,
+     * consider implementing that at application level, based on the low resolution timestamps.
+     * Returns NO_ERROR if timestamp is valid.
+     */
+            status_t    getTimestamp(AudioTimestamp& timestamp);
 
 protected:
     /* copying audio tracks is not allowed */
@@ -504,39 +600,62 @@ protected:
                 void        resume();   // allow thread to execute, if not requested to exit
 
     private:
+                void        pauseInternal(nsecs_t ns = 0LL);
+                                        // like pause(), but only used internally within thread
+
         friend class AudioTrack;
         virtual bool        threadLoop();
-        AudioTrack& mReceiver;
-        ~AudioTrackThread();
+        AudioTrack&         mReceiver;
+        virtual ~AudioTrackThread();
         Mutex               mMyLock;    // Thread::mLock is private
         Condition           mMyCond;    // Thread::mThreadExitedCondition is private
-        bool                mPaused;    // whether thread is currently paused
+        bool                mPaused;    // whether thread is requested to pause at next loop entry
+        bool                mPausedInt; // whether thread internally requests pause
+        nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
+        bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
     };
 
             // body of AudioTrackThread::threadLoop()
-            bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+            // returns the maximum amount of time before we would like to run again, where:
+            //      0           immediately
+            //      > 0         no later than this many nanoseconds from now
+            //      NS_WHENEVER still active but no particular deadline
+            //      NS_INACTIVE inactive so don't run again until re-started
+            //      NS_NEVER    never again
+            static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
+            nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
+            status_t processStreamEnd(int32_t waitCount);
+
 
             // caller must hold lock on mLock for all _l methods
+
             status_t createTrack_l(audio_stream_type_t streamType,
                                  uint32_t sampleRate,
                                  audio_format_t format,
                                  size_t frameCount,
                                  audio_output_flags_t flags,
                                  const sp<IMemory>& sharedBuffer,
-                                 audio_io_handle_t output);
+                                 audio_io_handle_t output,
+                                 size_t epoch);
 
-            // can only be called when !mActive
+            // can only be called when mState != STATE_ACTIVE
             void flush_l();
 
-            status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
+            void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
             audio_io_handle_t getOutput_l();
-            status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
-            bool stopped_l() const { return !mActive; }
 
+            // FIXME enum is faster than strcmp() for parameter 'from'
+            status_t restoreTrack_l(const char *from);
+
+            bool     isOffloaded() const
+                { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
+
+    // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
-    sp<AudioTrackThread>    mAudioTrackThread;
+    audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
 
+    sp<AudioTrackThread>    mAudioTrackThread;
     float                   mVolume[2];
     float                   mSendLevel;
     uint32_t                mSampleRate;
@@ -544,62 +663,94 @@ protected:
     size_t                  mReqFrameCount;         // frame count to request the next time a new
                                                     // IAudioTrack is needed
 
-    audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
-
-            // Starting address of buffers in shared memory.  If there is a shared buffer, mBuffers
-            // is the value of pointer() for the shared buffer, otherwise mBuffers points
-            // immediately after the control block.  This address is for the mapping within client
-            // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.
-    void*                   mBuffers;
 
+    // constant after constructor or set()
     audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
     audio_stream_type_t     mStreamType;
     uint32_t                mChannelCount;
     audio_channel_mask_t    mChannelMask;
+    transfer_type           mTransfer;
 
-                // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.
-                // For 8-bit PCM data, mFrameSizeAF is
-                // twice as large because data is expanded to 16-bit before being stored in buffer.
+    // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.  For 8-bit PCM data, it's
+    // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
     size_t                  mFrameSize;             // app-level frame size
     size_t                  mFrameSizeAF;           // AudioFlinger frame size
 
     status_t                mStatus;
-    uint32_t                mLatency;
 
-    bool                    mActive;                // protected by mLock
+    // can change dynamically when IAudioTrack invalidated
+    uint32_t                mLatency;               // in ms
 
+    // Indicates the current track state.  Protected by mLock.
+    enum State {
+        STATE_ACTIVE,
+        STATE_STOPPED,
+        STATE_PAUSED,
+        STATE_PAUSED_STOPPING,
+        STATE_FLUSHED,
+        STATE_STOPPING,
+    }                       mState;
+
+    // for client callback handler
     callback_t              mCbf;                   // callback handler for events, or NULL
-    void*                   mUserData;              // for client callback handler
+    void*                   mUserData;
 
     // for notification APIs
     uint32_t                mNotificationFramesReq; // requested number of frames between each
-                                                    // notification callback
+                                                    // notification callback,
+                                                    // at initial source sample rate
     uint32_t                mNotificationFramesAct; // actual number of frames between each
-                                                    // notification callback
+                                                    // notification callback,
+                                                    // at initial source sample rate
+    bool                    mRefreshRemaining;      // processAudioBuffer() should refresh next 2
+
+    // These are private to processAudioBuffer(), and are not protected by a lock
+    uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
+    bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
+    uint32_t                mObservedSequence;      // last observed value of mSequence
+
     sp<IMemory>             mSharedBuffer;
-    int                     mLoopCount;
-    uint32_t                mRemainingFrames;
+    uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;           // in frames
-    uint32_t                mUpdatePeriod;          // in frames
+    uint32_t                mUpdatePeriod;          // in frames, zero means no EVENT_NEW_POS
 
-    bool                    mFlushed; // FIXME will be made obsolete by making flush() synchronous
     audio_output_flags_t    mFlags;
     int                     mSessionId;
     int                     mAuxEffectId;
 
-    // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock:
-    //      1. mLock
-    //      2. mCblk->lock
-    // It is OK to lock only mCblk->lock.
     mutable Mutex           mLock;
 
     bool                    mIsTimed;
     int                     mPreviousPriority;          // before start()
     SchedPolicy             mPreviousSchedulingGroup;
-    AudioTrackClientProxy*  mProxy;
     bool                    mAwaitBoost;    // thread should wait for priority boost before running
+
+    // The proxy should only be referenced while a lock is held because the proxy isn't
+    // multi-thread safe, especially the SingleStateQueue part of the proxy.
+    // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock,
+    // provided that the caller also holds an extra reference to the proxy and shared memory to keep
+    // them around in case they are replaced during the obtainBuffer().
+    sp<StaticAudioTrackClientProxy> mStaticProxy;   // for type safety only
+    sp<AudioTrackClientProxy>       mProxy;         // primary owner of the memory
+
+    bool                    mInUnderrun;            // whether track is currently in underrun state
+    String8                 mName;                  // server's name for this IAudioTrack
+
+private:
+    class DeathNotifier : public IBinder::DeathRecipient {
+    public:
+        DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { }
+    protected:
+        virtual void        binderDied(const wp<IBinder>& who);
+    private:
+        const wp<AudioTrack> mAudioTrack;
+    };
+
+    sp<DeathNotifier>       mDeathNotifier;
+    uint32_t                mSequence;              // incremented for each new IAudioTrack attempt
+    audio_io_handle_t       mOutput;                // cached output io handle
 };
 
 class TimedAudioTrack : public AudioTrack
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1ed7b0..b1143b9 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,6 +171,30 @@ int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *p
 ////////////////////////////////////////////////////////////////////////////////
 int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
 
+////////////////////////////////////////////////////////////////////////////////
+//
+//    Function:       EffectGetSubEffects
+//
+//    Description:    Returns the descriptors of the sub effects of the effect
+//                    whose uuid is pointed to by first argument.
+//
+//    Input:
+//          pEffectUuid:    pointer to the effect uuid.
+//          size:           size of the buffer pointed by pDescriptor.
+//
+//    Input/Output:
+//          pDescriptor:    address where to return the sub effect descriptors.
+//
+//    Output:
+//        returned value:    0          successful operation.
+//                          -ENODEV     factory failed to initialize
+//                          -EINVAL     invalid pEffectUuid or pDescriptor
+//                          -ENOENT     no effect with this uuid found
+//        *pDescriptor:     updated with the sub effect descriptors.
+//
+////////////////////////////////////////////////////////////////////////////////
+int EffectGetSubEffects(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptors, size_t size);
+
 #if __cplusplus
 }  // extern "C"
 #endif
diff --git a/include/media/ExtendedAudioBufferProvider.h b/include/media/ExtendedAudioBufferProvider.h
index 00c4444..2539ed3 100644
--- a/include/media/ExtendedAudioBufferProvider.h
+++ b/include/media/ExtendedAudioBufferProvider.h
@@ -18,12 +18,20 @@
 #define ANDROID_EXTENDED_AUDIO_BUFFER_PROVIDER_H
 
 #include <media/AudioBufferProvider.h>
+#include <media/AudioTimestamp.h>
 
 namespace android {
 
 class ExtendedAudioBufferProvider : public AudioBufferProvider {
 public:
     virtual size_t  framesReady() const = 0;  // see description at AudioFlinger.h
+
+    // Return the total number of frames that have been obtained and released
+    virtual size_t  framesReleased() const { return 0; }
+
+    // Invoked by buffer consumer when a new timestamp is available.
+    // Default implementation ignores the timestamp.
+    virtual void    onTimestamp(const AudioTimestamp& timestamp) { }
 };
 
 }   // namespace android
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9c3067e..eaf7780 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -49,9 +49,13 @@ public:
         TRACK_DEFAULT = 0,  // client requests a default AudioTrack
         TRACK_TIMED   = 1,  // client requests a TimedAudioTrack
         TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord
+        TRACK_OFFLOAD = 4,  // client requests offload to hw codec
     };
     typedef uint32_t track_flags_t;
 
+    // invariant on exit for all APIs that return an sp<>:
+    //   (return value != 0) == (*status == NO_ERROR)
+
     /* create an audio track and registers it with AudioFlinger.
      * return null if the track cannot be created.
      */
@@ -66,6 +70,10 @@ public:
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
+                                // input: ignored
+                                // output: server's description of IAudioTrack for display in logs.
+                                // Don't attempt to parse, as the format could change.
+                                String8& name,
                                 status_t *status) = 0;
 
     virtual sp<IAudioRecord> openRecord(
@@ -74,7 +82,7 @@ public:
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
-                                track_flags_t flags,
+                                track_flags_t *flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
                                 status_t *status) = 0;
@@ -124,7 +132,9 @@ public:
     virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys)
                                     const = 0;
 
-    // register a current process for audio output change notifications
+    // Register an object to receive audio input/output change and track notifications.
+    // For a given calling pid, AudioFlinger disregards any registrations after the first.
+    // Thus the IAudioFlingerClient must be a singleton per process.
     virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
 
     // retrieve the audio recording buffer size
@@ -137,7 +147,8 @@ public:
                                          audio_format_t *pFormat,
                                          audio_channel_mask_t *pChannelMask,
                                          uint32_t *pLatencyMs,
-                                         audio_output_flags_t flags) = 0;
+                                         audio_output_flags_t flags,
+                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
                                     audio_io_handle_t output2) = 0;
     virtual status_t closeOutput(audio_io_handle_t output) = 0;
@@ -193,6 +204,10 @@ public:
     virtual uint32_t getPrimaryOutputSamplingRate() = 0;
     virtual size_t getPrimaryOutputFrameCount() = 0;
 
+    // Intended for AudioService to inform AudioFlinger of device's low RAM attribute,
+    // and should be called at most once.  For a definition of what "low RAM" means, see
+    // android.app.ActivityManager.isLowRamDevice().
+    virtual status_t setLowRamDevice(bool isLowRamDevice) = 0;
 };
 
 
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index b5ad4ef..09b9ea6 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -53,7 +53,8 @@ public:
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = 0,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                        const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  int session = 0) = 0;
@@ -95,6 +96,9 @@ public:
     virtual status_t queryDefaultPreProcessing(int audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count) = 0;
+   // Check if offload is possible for given format, stream type, sample rate,
+    // bit rate, duration, video and streaming or offload property is enabled
+    virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
 };
 
 
diff --git a/include/media/IAudioRecord.h b/include/media/IAudioRecord.h
index d6e3141..eccc2ca 100644
--- a/include/media/IAudioRecord.h
+++ b/include/media/IAudioRecord.h
@@ -34,6 +34,9 @@ class IAudioRecord : public IInterface
 public:
     DECLARE_META_INTERFACE(AudioRecord);
 
+    /* get this tracks control block */
+    virtual sp<IMemory> getCblk() const = 0;
+
     /* After it's created the track is not active. Call start() to
      * make it active.
      */
@@ -44,9 +47,6 @@ public:
      * will be processed, unless flush() is called.
      */
     virtual void        stop() = 0;
-
-    /* get this tracks control block */
-    virtual sp<IMemory> getCblk() const = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 144be0e..5c8a484 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -25,6 +25,8 @@
 #include <binder/IInterface.h>
 #include <binder/IMemory.h>
 #include <utils/LinearTransform.h>
+#include <utils/String8.h>
+#include <media/AudioTimestamp.h>
 
 namespace android {
 
@@ -82,6 +84,15 @@ public:
        or Tungsten time. The values for target are defined in AudioTrack.h */
     virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
                                               int target) = 0;
+
+    /* Send parameters to the audio hardware */
+    virtual status_t    setParameters(const String8& keyValuePairs) = 0;
+
+    /* Return NO_ERROR if timestamp is valid */
+    virtual status_t    getTimestamp(AudioTimestamp& timestamp) = 0;
+
+    /* Signal the playback thread for a change in control block */
+    virtual void        signal() = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/IDrm.h b/include/media/IDrm.h
index d630c40..5ef26af 100644
--- a/include/media/IDrm.h
+++ b/include/media/IDrm.h
@@ -32,7 +32,7 @@ struct IDrm : public IInterface {
 
     virtual status_t initCheck() const = 0;
 
-    virtual bool isCryptoSchemeSupported(const uint8_t uuid[16]) = 0;
+    virtual bool isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType) = 0;
 
     virtual status_t createPlugin(const uint8_t uuid[16]) = 0;
 
diff --git a/include/media/IHDCP.h b/include/media/IHDCP.h
index 6d27b18..352561e 100644
--- a/include/media/IHDCP.h
+++ b/include/media/IHDCP.h
@@ -17,6 +17,7 @@
 #include <binder/IInterface.h>
 #include <media/hardware/HDCPAPI.h>
 #include <media/stagefright/foundation/ABase.h>
+#include <ui/GraphicBuffer.h>
 
 namespace android {
 
@@ -45,6 +46,17 @@ struct IHDCP : public IInterface {
     // Request to shutdown the active HDCP session.
     virtual status_t shutdownAsync() = 0;
 
+    // Returns the capability bitmask of this HDCP session.
+    // Possible return values (please refer to HDCAPAPI.h):
+    //   HDCP_CAPS_ENCRYPT: mandatory, meaning the HDCP module can encrypt
+    //   from an input byte-array buffer to an output byte-array buffer
+    //   HDCP_CAPS_ENCRYPT_NATIVE: the HDCP module supports encryption from
+    //   a native buffer to an output byte-array buffer. The format of the
+    //   input native buffer is specific to vendor's encoder implementation.
+    //   It is the same format as that used by the encoder when
+    //   "storeMetaDataInBuffers" extension is enabled on its output port.
+    virtual uint32_t getCaps() = 0;
+
     // ENCRYPTION only:
     // Encrypt data according to the HDCP spec. "size" bytes of data are
     // available at "inData" (virtual address), "size" may not be a multiple
@@ -59,6 +71,20 @@ struct IHDCP : public IInterface {
             const void *inData, size_t size, uint32_t streamCTR,
             uint64_t *outInputCTR, void *outData) = 0;
 
+    // Encrypt data according to the HDCP spec. "size" bytes of data starting
+    // at location "offset" are available in "buffer" (buffer handle). "size"
+    // may not be a multiple of 128 bits (16 bytes). An equal number of
+    // encrypted bytes should be written to the buffer at "outData" (virtual
+    // address). This operation is to be synchronous, i.e. this call does not
+    // return until outData contains size bytes of encrypted data.
+    // streamCTR will be assigned by the caller (to 0 for the first PES stream,
+    // 1 for the second and so on)
+    // inputCTR _will_be_maintained_by_the_callee_ for each PES stream.
+    virtual status_t encryptNative(
+            const sp<GraphicBuffer> &graphicBuffer,
+            size_t offset, size_t size, uint32_t streamCTR,
+            uint64_t *outInputCTR, void *outData) = 0;
+
     // DECRYPTION only:
     // Decrypt data according to the HDCP spec.
     // "size" bytes of encrypted data are available at "inData"
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index fef7af2..2998b37 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -49,8 +49,12 @@ public:
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
     virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
 
-    virtual sp<IMemory>         decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
-    virtual sp<IMemory>         decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
+    virtual status_t         decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
+                                    audio_format_t* pFormat,
+                                    const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+    virtual status_t         decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
+                                    int* pNumChannels, audio_format_t* pFormat,
+                                    const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
     virtual sp<IOMX>            getOMX() = 0;
     virtual sp<ICrypto>         makeCrypto() = 0;
     virtual sp<IDrm>            makeDrm() = 0;
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 0b1d1e4..9c8451c 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -83,6 +83,10 @@ public:
     virtual status_t storeMetaDataInBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable) = 0;
 
+    virtual status_t prepareForAdaptivePlayback(
+            node_id node, OMX_U32 portIndex, OMX_BOOL enable,
+            OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight) = 0;
+
     virtual status_t enableGraphicBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable) = 0;
 
@@ -97,6 +101,10 @@ public:
             node_id node, OMX_U32 port_index,
             const sp<GraphicBuffer> &graphicBuffer, buffer_id *buffer) = 0;
 
+    virtual status_t updateGraphicBufferInMeta(
+            node_id node, OMX_U32 port_index,
+            const sp<GraphicBuffer> &graphicBuffer, buffer_id buffer) = 0;
+
     virtual status_t createInputSurface(
             node_id node, OMX_U32 port_index,
             sp<IGraphicBufferProducer> *bufferProducer) = 0;
@@ -130,6 +138,17 @@ public:
             node_id node,
             const char *parameter_name,
             OMX_INDEXTYPE *index) = 0;
+
+    enum InternalOptionType {
+        INTERNAL_OPTION_SUSPEND,  // data is a bool
+        INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY,  // data is an int64_t
+    };
+    virtual status_t setInternalOption(
+            node_id node,
+            OMX_U32 port_index,
+            InternalOptionType type,
+            const void *data,
+            size_t size) = 0;
 };
 
 struct omx_message {
diff --git a/include/media/IRemoteDisplayClient.h b/include/media/IRemoteDisplayClient.h
index 7b0fa9e..0e6d55d 100644
--- a/include/media/IRemoteDisplayClient.h
+++ b/include/media/IRemoteDisplayClient.h
@@ -49,7 +49,7 @@ public:
     // Provides a surface texture that the client should use to stream buffers to
     // the remote display.
     virtual void onDisplayConnected(const sp<IGraphicBufferProducer>& bufferProducer,
-            uint32_t width, uint32_t height, uint32_t flags) = 0; // one-way
+            uint32_t width, uint32_t height, uint32_t flags, uint32_t session) = 0; // one-way
 
     // Indicates that the remote display has been disconnected normally.
     // This method should only be called once the client has called 'dispose()'
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
index 0616bf0..388f767 100644
--- a/include/media/JetPlayer.h
+++ b/include/media/JetPlayer.h
@@ -88,7 +88,7 @@ private:
     EAS_DATA_HANDLE     mEasData;
     EAS_FILE_LOCATOR    mEasJetFileLoc;
     EAS_PCM*            mAudioBuffer;// EAS renders the MIDI data into this buffer,
-    AudioTrack*         mAudioTrack; // and we play it in this audio track
+    sp<AudioTrack>      mAudioTrack; // and we play it in this audio track
     int                 mTrackBufferSize;
     S_JET_STATUS        mJetStatus;
     S_JET_STATUS        mPreviousJetStatus;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 9a75f81..3b151ef 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -74,9 +74,18 @@ public:
     // AudioSink: abstraction layer for audio output
     class AudioSink : public RefBase {
     public:
+        enum cb_event_t {
+            CB_EVENT_FILL_BUFFER,   // Request to write more data to buffer.
+            CB_EVENT_STREAM_END,    // Sent after all the buffers queued in AF and HW are played
+                                    // back (after stop is called)
+            CB_EVENT_TEAR_DOWN      // The AudioTrack was invalidated due to use case change:
+                                    // Need to re-evaluate offloading options
+        };
+
         // Callback returns the number of bytes actually written to the buffer.
         typedef size_t (*AudioCallback)(
-                AudioSink *audioSink, void *buffer, size_t size, void *cookie);
+                AudioSink *audioSink, void *buffer, size_t size, void *cookie,
+                        cb_event_t event);
 
         virtual             ~AudioSink() {}
         virtual bool        ready() const = 0; // audio output is open and ready
@@ -99,9 +108,10 @@ public:
                 int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
                 AudioCallback cb = NULL,
                 void *cookie = NULL,
-                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                const audio_offload_info_t *offloadInfo = NULL) = 0;
 
-        virtual void        start() = 0;
+        virtual status_t    start() = 0;
         virtual ssize_t     write(const void* buffer, size_t size) = 0;
         virtual void        stop() = 0;
         virtual void        flush() = 0;
@@ -110,6 +120,9 @@ public:
 
         virtual status_t    setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
         virtual bool        needsTrailingPadding() { return true; }
+
+        virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR; };
+        virtual String8     getParameters(const String8& keys) { return String8::empty(); };
     };
 
                         MediaPlayerBase() : mCookie(0), mNotify(0) {}
diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h
index 7bf3069..2dd78cc 100644
--- a/include/media/SoundPool.h
+++ b/include/media/SoundPool.h
@@ -22,6 +22,8 @@
 #include <utils/Vector.h>
 #include <utils/KeyedVector.h>
 #include <media/AudioTrack.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/MemoryBase.h>
 
 namespace android {
 
@@ -85,6 +87,7 @@ private:
     int64_t             mLength;
     char*               mUrl;
     sp<IMemory>         mData;
+    sp<MemoryHeapBase>  mHeap;
 };
 
 // stores pending events for stolen channels
@@ -118,7 +121,7 @@ protected:
 class SoundChannel : public SoundEvent {
 public:
     enum state { IDLE, RESUMING, STOPPING, PAUSED, PLAYING };
-    SoundChannel() : mAudioTrack(NULL), mState(IDLE), mNumChannels(1),
+    SoundChannel() : mState(IDLE), mNumChannels(1),
             mPos(0), mToggle(0), mAutoPaused(false) {}
     ~SoundChannel();
     void init(SoundPool* soundPool);
@@ -148,7 +151,7 @@ private:
     bool doStop_l();
 
     SoundPool*          mSoundPool;
-    AudioTrack*         mAudioTrack;
+    sp<AudioTrack>      mAudioTrack;
     SoundEvent          mNextEvent;
     Mutex               mLock;
     int                 mState;
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 2183fbe..98c4332 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -160,7 +160,7 @@ public:
     bool isInited() { return (mState == TONE_IDLE)?false:true;}
 
     // returns the audio session this ToneGenerator belongs to or 0 if an error occured.
-    int getSessionId() { return (mpAudioTrack == NULL) ? 0 : mpAudioTrack->getSessionId(); }
+    int getSessionId() { return (mpAudioTrack == 0) ? 0 : mpAudioTrack->getSessionId(); }
 
 private:
 
@@ -264,7 +264,7 @@ private:
     unsigned short mLoopCounter; // Current tone loopback count
 
     uint32_t mSamplingRate;  // AudioFlinger Sampling rate
-    AudioTrack *mpAudioTrack;  // Pointer to audio track used for playback
+    sp<AudioTrack> mpAudioTrack;  // Pointer to audio track used for playback
     Mutex mLock;  // Mutex to control concurent access to ToneGenerator object from audio callback and application API
     Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
     Condition mWaitCbkCond; // condition enabling interface to wait for audio callback completion after a change is requested
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index aa58905..6167dd6 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -19,7 +19,7 @@
 
 #include <media/AudioEffect.h>
 #include <audio_effects/effect_visualizer.h>
-#include <string.h>
+#include <utils/Thread.h>
 
 /**
  * The Visualizer class enables application to retrieve part of the currently playing audio for
@@ -114,6 +114,14 @@ public:
     status_t setScalingMode(uint32_t mode);
     uint32_t getScalingMode() { return mScalingMode; }
 
+    // set which measurements are done on the audio buffers processed by the effect.
+    // valid measurements (mask): MEASUREMENT_MODE_PEAK_RMS
+    status_t setMeasurementMode(uint32_t mode);
+    uint32_t getMeasurementMode() { return mMeasurementMode; }
+
+    // return a set of int32_t measurements
+    status_t getIntMeasurements(uint32_t type, uint32_t number, int32_t *measurements);
+
     // return a capture in PCM 8 bit unsigned format. The size of the capture is equal to
     // getCaptureSize()
     status_t getWaveForm(uint8_t *waveform);
@@ -156,6 +164,7 @@ private:
     uint32_t mCaptureSize;
     uint32_t mSampleRate;
     uint32_t mScalingMode;
+    uint32_t mMeasurementMode;
     capture_cbk_t mCaptureCallBack;
     void *mCaptureCbkUser;
     sp<CaptureThread> mCaptureThread;
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 14381c7..4c05fc3 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -42,9 +42,14 @@ enum media_event_type {
     MEDIA_BUFFERING_UPDATE  = 3,
     MEDIA_SEEK_COMPLETE     = 4,
     MEDIA_SET_VIDEO_SIZE    = 5,
+    MEDIA_STARTED           = 6,
+    MEDIA_PAUSED            = 7,
+    MEDIA_STOPPED           = 8,
+    MEDIA_SKIPPED           = 9,
     MEDIA_TIMED_TEXT        = 99,
     MEDIA_ERROR             = 100,
     MEDIA_INFO              = 200,
+    MEDIA_SUBTITLE_DATA     = 201,
 };
 
 // Generic error codes for the media player framework.  Errors are fatal, the
@@ -173,6 +178,7 @@ enum media_track_type {
     MEDIA_TRACK_TYPE_VIDEO = 1,
     MEDIA_TRACK_TYPE_AUDIO = 2,
     MEDIA_TRACK_TYPE_TIMEDTEXT = 3,
+    MEDIA_TRACK_TYPE_SUBTITLE = 4,
 };
 
 // ----------------------------------------------------------------------------
@@ -218,8 +224,12 @@ public:
             bool            isLooping();
             status_t        setVolume(float leftVolume, float rightVolume);
             void            notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
-    static  sp<IMemory>     decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
-    static  sp<IMemory>     decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
+    static  status_t        decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
+                                   audio_format_t* pFormat,
+                                   const sp<IMemoryHeap>& heap, size_t *pSize);
+    static  status_t        decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
+                                   int* pNumChannels, audio_format_t* pFormat,
+                                   const sp<IMemoryHeap>& heap, size_t *pSize);
             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);
diff --git a/include/media/nbaio/AudioStreamOutSink.h b/include/media/nbaio/AudioStreamOutSink.h
index 5976b18..7948d40 100644
--- a/include/media/nbaio/AudioStreamOutSink.h
+++ b/include/media/nbaio/AudioStreamOutSink.h
@@ -52,6 +52,8 @@ public:
     // implementation of GNWT (if any)
     virtual status_t getNextWriteTimestamp(int64_t *timestamp);
 
+    virtual status_t getTimestamp(AudioTimestamp& timestamp);
+
     // NBAIO_Sink end
 
 #if 0   // until necessary
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index 5fcfe9e..d3802fe 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -20,9 +20,12 @@
 #include <time.h>
 #include <utils/LinearTransform.h>
 #include "NBAIO.h"
+#include <media/SingleStateQueue.h>
 
 namespace android {
 
+typedef SingleStateQueue<AudioTimestamp> AudioTimestampSingleStateQueue;
+
 // MonoPipe is similar to Pipe except:
 //  - supports only a single reader, called MonoPipeReader
 //  - write() cannot overrun; instead it will return a short actual count if insufficient space
@@ -88,6 +91,9 @@ public:
             // Return true if the write side of a pipe is currently shutdown.
             bool    isShutdown();
 
+            // Return NO_ERROR if there is a timestamp available
+            status_t getTimestamp(AudioTimestamp& timestamp);
+
 private:
     // A pair of methods and a helper variable which allows the reader and the
     // writer to update and observe the values of mFront and mNextRdPTS in an
@@ -127,6 +133,10 @@ private:
     LinearTransform mSamplesToLocalTime;
 
     bool            mIsShutdown;    // whether shutdown(true) was called, no barriers are needed
+
+    AudioTimestampSingleStateQueue::Shared      mTimestampShared;
+    AudioTimestampSingleStateQueue::Mutator     mTimestampMutator;
+    AudioTimestampSingleStateQueue::Observer    mTimestampObserver;
 };
 
 }   // namespace android
diff --git a/include/media/nbaio/MonoPipeReader.h b/include/media/nbaio/MonoPipeReader.h
index 0e1c992..78fe867 100644
--- a/include/media/nbaio/MonoPipeReader.h
+++ b/include/media/nbaio/MonoPipeReader.h
@@ -49,6 +49,8 @@ public:
 
     virtual ssize_t read(void *buffer, size_t count, int64_t readPTS);
 
+    virtual void    onTimestamp(const AudioTimestamp& timestamp);
+
     // NBAIO_Source end
 
 #if 0   // until necessary
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index f5d6eb5..1da0c73 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -28,6 +28,7 @@
 #include <stdlib.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
+#include <media/AudioTimestamp.h>
 
 namespace android {
 
@@ -213,6 +214,11 @@ public:
     //  <other> Something unexpected happened internally.  Check the logs and start debugging.
     virtual status_t getNextWriteTimestamp(int64_t *ts) { return INVALID_OPERATION; }
 
+    // Returns NO_ERROR if a timestamp is available.  The timestamp includes the total number
+    // of frames presented to an external observer, together with the value of CLOCK_MONOTONIC
+    // as of this presentation count.
+    virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
+
 protected:
     NBAIO_Sink(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
     virtual ~NBAIO_Sink() { }
@@ -300,6 +306,10 @@ public:
     virtual ssize_t readVia(readVia_t via, size_t total, void *user,
                             int64_t readPTS, size_t block = 0);
 
+    // Invoked asynchronously by corresponding sink when a new timestamp is available.
+    // Default implementation ignores the timestamp.
+    virtual void    onTimestamp(const AudioTimestamp& timestamp) { }
+
 protected:
     NBAIO_Source(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
     virtual ~NBAIO_Source() { }
diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
index 107ba66..6d59ea7 100644
--- a/include/media/nbaio/NBLog.h
+++ b/include/media/nbaio/NBLog.h
@@ -90,6 +90,8 @@ public:
     virtual ~Timeline();
 #endif
 
+    // Input parameter 'size' is the desired size of the timeline in byte units.
+    // Returns the size rounded up to a power-of-2, plus the constant size overhead for indices.
     static size_t sharedSize(size_t size);
 
 #if 0
@@ -110,8 +112,12 @@ private:
 class Writer : public RefBase {
 public:
     Writer();                   // dummy nop implementation without shared memory
+
+    // Input parameter 'size' is the desired size of the timeline in byte units.
+    // The size of the shared memory must be at least Timeline::sharedSize(size).
     Writer(size_t size, void *shared);
     Writer(size_t size, const sp<IMemory>& iMemory);
+
     virtual ~Writer() { }
 
     virtual void    log(const char *string);
@@ -165,8 +171,12 @@ private:
 
 class Reader : public RefBase {
 public:
+
+    // Input parameter 'size' is the desired size of the timeline in byte units.
+    // The size of the shared memory must be at least Timeline::sharedSize(size).
     Reader(size_t size, const void *shared);
     Reader(size_t size, const sp<IMemory>& iMemory);
+
     virtual ~Reader() { }
 
     void    dump(int fd, size_t indent = 0);
diff --git a/include/media/nbaio/SourceAudioBufferProvider.h b/include/media/nbaio/SourceAudioBufferProvider.h
index c08331b..cdfb6fe 100644
--- a/include/media/nbaio/SourceAudioBufferProvider.h
+++ b/include/media/nbaio/SourceAudioBufferProvider.h
@@ -36,6 +36,8 @@ public:
 
     // ExtendedAudioBufferProvider interface
     virtual size_t   framesReady() const;
+    virtual size_t   framesReleased() const;
+    virtual void     onTimestamp(const AudioTimestamp& timestamp);
 
 private:
     const sp<NBAIO_Source> mSource;     // the wrapped source
@@ -45,6 +47,7 @@ private:
     size_t              mOffset;    // frame offset within mAllocated of valid data
     size_t              mRemaining; // frame count within mAllocated of valid data
     size_t              mGetCount;  // buffer.frameCount of the most recent getNextBuffer
+    uint32_t            mFramesReleased;    // counter of the total number of frames released
 };
 
 }   // namespace android
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index df25d7b..a8ffd4a 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -124,7 +124,8 @@ private:
     };
 
     enum {
-        kFlagIsSecure   = 1,
+        kFlagIsSecure                                 = 1,
+        kFlagPushBlankBuffersToNativeWindowOnShutdown = 2,
     };
 
     struct BufferInfo {
@@ -138,6 +139,7 @@ private:
 
         IOMX::buffer_id mBufferID;
         Status mStatus;
+        unsigned mDequeuedAt;
 
         sp<ABuffer> mData;
         sp<GraphicBuffer> mGraphicBuffer;
@@ -182,7 +184,7 @@ private:
 
     bool mSentFormat;
     bool mIsEncoder;
-
+    bool mUseMetadataOnEncoderOutput;
     bool mShutdownInProgress;
 
     // If "mKeepComponentAllocated" we only transition back to Loaded state
@@ -194,12 +196,22 @@ private:
 
     bool mChannelMaskPresent;
     int32_t mChannelMask;
+    unsigned mDequeueCounter;
+    bool mStoreMetaDataInOutputBuffers;
+    int32_t mMetaDataBuffersToSubmit;
+
+    int64_t mRepeatFrameDelayUs;
 
     status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
     status_t allocateBuffersOnPort(OMX_U32 portIndex);
     status_t freeBuffersOnPort(OMX_U32 portIndex);
     status_t freeBuffer(OMX_U32 portIndex, size_t i);
 
+    status_t configureOutputBuffersFromNativeWindow(
+            OMX_U32 *nBufferCount, OMX_U32 *nBufferSize,
+            OMX_U32 *nMinUndequeuedBuffers);
+    status_t allocateOutputMetaDataBuffers();
+    status_t submitOutputMetaDataBuffer();
     status_t allocateOutputBuffersFromNativeWindow();
     status_t cancelBufferToNativeWindow(BufferInfo *info);
     status_t freeOutputBuffersNotOwnedByComponent();
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 1dc408f..912a43c 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -36,8 +36,16 @@ public:
         SEEK_COMPLETE
     };
 
+    enum {
+        ALLOW_DEEP_BUFFERING = 0x01,
+        USE_OFFLOAD = 0x02,
+        HAS_VIDEO   = 0x1000,
+        IS_STREAMING = 0x2000
+
+    };
+
     AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink,
-                bool allowDeepBuffering = false,
+                uint32_t flags = 0,
                 AwesomePlayer *audioObserver = NULL);
 
     virtual ~AudioPlayer();
@@ -51,7 +59,7 @@ public:
     status_t start(bool sourceAlreadyStarted = false);
 
     void pause(bool playPendingSamples = false);
-    void resume();
+    status_t resume();
 
     // Returns the timestamp of the last buffer played (in us).
     int64_t getMediaTimeUs();
@@ -67,10 +75,12 @@ public:
 
     status_t setPlaybackRatePermille(int32_t ratePermille);
 
+    void notifyAudioEOS();
+
 private:
     friend class VideoEditorAudioPlayer;
     sp<MediaSource> mSource;
-    AudioTrack *mAudioTrack;
+    sp<AudioTrack> mAudioTrack;
 
     MediaBuffer *mInputBuffer;
 
@@ -97,17 +107,20 @@ private:
     MediaBuffer *mFirstBuffer;
 
     sp<MediaPlayerBase::AudioSink> mAudioSink;
-    bool mAllowDeepBuffering;       // allow audio deep audio buffers. Helps with low power audio
-                                    // playback but implies high latency
     AwesomePlayer *mObserver;
     int64_t mPinnedTimeUs;
 
+    bool mPlaying;
+    int64_t mStartPosUs;
+    const uint32_t mCreateFlags;
+
     static void AudioCallback(int event, void *user, void *info);
     void AudioCallback(int event, void *info);
 
     static size_t AudioSinkCallback(
             MediaPlayerBase::AudioSink *audioSink,
-            void *data, size_t size, void *me);
+            void *data, size_t size, void *me,
+            MediaPlayerBase::AudioSink::cb_event_t event);
 
     size_t fillBuffer(void *data, size_t size);
 
@@ -116,6 +129,10 @@ private:
     void reset();
 
     uint32_t getNumFramesPendingPlayout() const;
+    int64_t getOutputPlayPositionUs_l() const;
+
+    bool allowDeepBuffering() const { return (mCreateFlags & ALLOW_DEEP_BUFFERING) != 0; }
+    bool useOffload() const { return (mCreateFlags & USE_OFFLOAD) != 0; }
 
     AudioPlayer(const AudioPlayer &);
     AudioPlayer &operator=(const AudioPlayer &);
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 99f3c3b..4c9aaad 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -73,7 +73,7 @@ private:
     Condition mFrameAvailableCondition;
     Condition mFrameEncodingCompletionCondition;
 
-    AudioRecord *mRecord;
+    sp<AudioRecord> mRecord;
     status_t mInitCheck;
     bool mStarted;
     int32_t mSampleRate;
diff --git a/include/media/stagefright/MediaCodecList.h b/include/media/stagefright/MediaCodecList.h
index dfb845b..590623b 100644
--- a/include/media/stagefright/MediaCodecList.h
+++ b/include/media/stagefright/MediaCodecList.h
@@ -50,7 +50,8 @@ struct MediaCodecList {
     status_t getCodecCapabilities(
             size_t index, const char *type,
             Vector<ProfileLevel> *profileLevels,
-            Vector<uint32_t> *colorFormats) const;
+            Vector<uint32_t> *colorFormats,
+            uint32_t *flags) const;
 
 private:
     enum Section {
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 81de6e4..85693d4 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -22,7 +22,8 @@ namespace android {
 
 extern const char *MEDIA_MIMETYPE_IMAGE_JPEG;
 
-extern const char *MEDIA_MIMETYPE_VIDEO_VPX;
+extern const char *MEDIA_MIMETYPE_VIDEO_VP8;
+extern const char *MEDIA_MIMETYPE_VIDEO_VP9;
 extern const char *MEDIA_MIMETYPE_VIDEO_AVC;
 extern const char *MEDIA_MIMETYPE_VIDEO_MPEG4;
 extern const char *MEDIA_MIMETYPE_VIDEO_H263;
diff --git a/include/media/stagefright/MediaErrors.h b/include/media/stagefright/MediaErrors.h
index ee5e4e2..686f286 100644
--- a/include/media/stagefright/MediaErrors.h
+++ b/include/media/stagefright/MediaErrors.h
@@ -56,14 +56,11 @@ enum {
     ERROR_DRM_TAMPER_DETECTED               = DRM_ERROR_BASE - 7,
     ERROR_DRM_NOT_PROVISIONED               = DRM_ERROR_BASE - 8,
     ERROR_DRM_DEVICE_REVOKED                = DRM_ERROR_BASE - 9,
+    ERROR_DRM_RESOURCE_BUSY                 = DRM_ERROR_BASE - 10,
 
     ERROR_DRM_VENDOR_MAX                    = DRM_ERROR_BASE - 500,
     ERROR_DRM_VENDOR_MIN                    = DRM_ERROR_BASE - 999,
 
-    // Deprecated
-    ERROR_DRM_WV_VENDOR_MAX                 = ERROR_DRM_VENDOR_MAX,
-    ERROR_DRM_WV_VENDOR_MIN                 = ERROR_DRM_VENDOR_MIN,
-
     // Heartbeat Error Codes
     HEARTBEAT_ERROR_BASE = -3000,
     ERROR_HEARTBEAT_TERMINATE_REQUESTED                     = HEARTBEAT_ERROR_BASE,
diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h
index c1fdbad..ff6a66e 100644
--- a/include/media/stagefright/MediaMuxer.h
+++ b/include/media/stagefright/MediaMuxer.h
@@ -79,6 +79,16 @@ public:
     status_t setOrientationHint(int degrees);
 
     /**
+     * Set the location.
+     * @param latitude The latitude in degree x 1000. Its value must be in the range
+     * [-900000, 900000].
+     * @param longitude The longitude in degree x 1000. Its value must be in the range
+     * [-1800000, 1800000].
+     * @return OK if no error.
+     */
+    status_t setLocation(int latitude, int longitude);
+
+    /**
      * Stop muxing.
      * This method is a blocking call. Depending on how
      * much data is bufferred internally, the time needed for stopping
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 583c3b3..daaf20f 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -361,9 +361,14 @@ private:
 };
 
 struct CodecCapabilities {
+    enum {
+        kFlagSupportsAdaptivePlayback = 1 << 0,
+    };
+
     String8 mComponentName;
     Vector<CodecProfileLevel> mProfileLevels;
     Vector<OMX_U32> mColorFormats;
+    uint32_t mFlags;
 };
 
 // Return a vector of componentNames with supported profile/level pairs
diff --git a/include/media/stagefright/SurfaceMediaSource.h b/include/media/stagefright/SurfaceMediaSource.h
index 5f21da9..db5f947 100644
--- a/include/media/stagefright/SurfaceMediaSource.h
+++ b/include/media/stagefright/SurfaceMediaSource.h
@@ -56,7 +56,7 @@ class GraphicBuffer;
 
 class SurfaceMediaSource : public MediaSource,
                                 public MediaBufferObserver,
-                                protected BufferQueue::ConsumerListener {
+                                protected ConsumerListener {
 public:
     enum { MIN_UNDEQUEUED_BUFFERS = 4};
 
@@ -146,9 +146,13 @@ private:
     // this consumer
     sp<BufferQueue> mBufferQueue;
 
-    // mBufferSlot caches GraphicBuffers from the buffer queue
-    sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS];
+    struct SlotData {
+        sp<GraphicBuffer> mGraphicBuffer;
+        uint64_t mFrameNumber;
+    };
 
+    // mSlots caches GraphicBuffers and frameNumbers from the buffer queue
+    SlotData mSlots[BufferQueue::NUM_BUFFER_SLOTS];
 
     // The permenent width and height of SMS buffers
     int mWidth;
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index 73940d3..c24f612 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -22,6 +22,8 @@
 #include <stdint.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
+#include <system/audio.h>
+#include <media/MediaPlayerInterface.h>
 
 namespace android {
 
@@ -48,6 +50,15 @@ void convertMessageToMetaData(
 
 AString MakeUserAgent();
 
+// Convert a MIME type to a AudioSystem::audio_format
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
+
+// Send information from MetaData to the HAL via AudioSink
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
+
+// Check whether the stream defined by meta can be offloaded to hardware
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming);
+
 }  // namespace android
 
 #endif  // UTILS_H_
diff --git a/include/media/stagefright/foundation/ALooperRoster.h b/include/media/stagefright/foundation/ALooperRoster.h
index 2e5fd73..940fc55 100644
--- a/include/media/stagefright/foundation/ALooperRoster.h
+++ b/include/media/stagefright/foundation/ALooperRoster.h
@@ -30,6 +30,7 @@ struct ALooperRoster {
             const sp<ALooper> looper, const sp<AHandler> &handler);
 
     void unregisterHandler(ALooper::handler_id handlerID);
+    void unregisterStaleHandlers();
 
     status_t postMessage(const sp<AMessage> &msg, int64_t delayUs = 0);
     void deliverMessage(const sp<AMessage> &msg);
diff --git a/include/media/stagefright/foundation/ANetworkSession.h b/include/media/stagefright/foundation/ANetworkSession.h
new file mode 100644
index 0000000..fd3ebaa
--- /dev/null
+++ b/include/media/stagefright/foundation/ANetworkSession.h
@@ -0,0 +1,135 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef A_NETWORK_SESSION_H_
+
+#define A_NETWORK_SESSION_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/Thread.h>
+
+#include <netinet/in.h>
+
+namespace android {
+
+struct AMessage;
+
+// Helper class to manage a number of live sockets (datagram and stream-based)
+// on a single thread. Clients are notified about activity through AMessages.
+struct ANetworkSession : public RefBase {
+    ANetworkSession();
+
+    status_t start();
+    status_t stop();
+
+    status_t createRTSPClient(
+            const char *host, unsigned port, const sp<AMessage> &notify,
+            int32_t *sessionID);
+
+    status_t createRTSPServer(
+            const struct in_addr &addr, unsigned port,
+            const sp<AMessage> &notify, int32_t *sessionID);
+
+    status_t createUDPSession(
+            unsigned localPort, const sp<AMessage> &notify, int32_t *sessionID);
+
+    status_t createUDPSession(
+            unsigned localPort,
+            const char *remoteHost,
+            unsigned remotePort,
+            const sp<AMessage> &notify,
+            int32_t *sessionID);
+
+    status_t connectUDPSession(
+            int32_t sessionID, const char *remoteHost, unsigned remotePort);
+
+    // passive
+    status_t createTCPDatagramSession(
+            const struct in_addr &addr, unsigned port,
+            const sp<AMessage> &notify, int32_t *sessionID);
+
+    // active
+    status_t createTCPDatagramSession(
+            unsigned localPort,
+            const char *remoteHost,
+            unsigned remotePort,
+            const sp<AMessage> &notify,
+            int32_t *sessionID);
+
+    status_t destroySession(int32_t sessionID);
+
+    status_t sendRequest(
+            int32_t sessionID, const void *data, ssize_t size = -1,
+            bool timeValid = false, int64_t timeUs = -1ll);
+
+    status_t switchToWebSocketMode(int32_t sessionID);
+
+    enum NotificationReason {
+        kWhatError,
+        kWhatConnected,
+        kWhatClientConnected,
+        kWhatData,
+        kWhatDatagram,
+        kWhatBinaryData,
+        kWhatWebSocketMessage,
+        kWhatNetworkStall,
+    };
+
+protected:
+    virtual ~ANetworkSession();
+
+private:
+    struct NetworkThread;
+    struct Session;
+
+    Mutex mLock;
+    sp<Thread> mThread;
+
+    int32_t mNextSessionID;
+
+    int mPipeFd[2];
+
+    KeyedVector<int32_t, sp<Session> > mSessions;
+
+    enum Mode {
+        kModeCreateUDPSession,
+        kModeCreateTCPDatagramSessionPassive,
+        kModeCreateTCPDatagramSessionActive,
+        kModeCreateRTSPServer,
+        kModeCreateRTSPClient,
+    };
+    status_t createClientOrServer(
+            Mode mode,
+            const struct in_addr *addr,
+            unsigned port,
+            const char *remoteHost,
+            unsigned remotePort,
+            const sp<AMessage> &notify,
+            int32_t *sessionID);
+
+    void threadLoop();
+    void interrupt();
+
+    static status_t MakeSocketNonBlocking(int s);
+
+    DISALLOW_EVIL_CONSTRUCTORS(ANetworkSession);
+};
+
+}  // namespace android
+
+#endif  // A_NETWORK_SESSION_H_
diff --git a/include/media/stagefright/foundation/ParsedMessage.h b/include/media/stagefright/foundation/ParsedMessage.h
new file mode 100644
index 0000000..9d43a93
--- /dev/null
+++ b/include/media/stagefright/foundation/ParsedMessage.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AString.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+// Encapsulates an "HTTP/RTSP style" response, i.e. a status line,
+// key/value pairs making up the headers and an optional body/content.
+struct ParsedMessage : public RefBase {
+    static sp<ParsedMessage> Parse(
+            const char *data, size_t size, bool noMoreData, size_t *length);
+
+    bool findString(const char *name, AString *value) const;
+    bool findInt32(const char *name, int32_t *value) const;
+
+    const char *getContent() const;
+
+    bool getRequestField(size_t index, AString *field) const;
+    bool getStatusCode(int32_t *statusCode) const;
+
+    AString debugString() const;
+
+    static bool GetAttribute(const char *s, const char *key, AString *value);
+
+    static bool GetInt32Attribute(
+            const char *s, const char *key, int32_t *value);
+
+
+protected:
+    virtual ~ParsedMessage();
+
+private:
+    KeyedVector<AString, AString> mDict;
+    AString mContent;
+
+    ParsedMessage();
+
+    ssize_t parse(const char *data, size_t size, bool noMoreData);
+
+    DISALLOW_EVIL_CONSTRUCTORS(ParsedMessage);
+};
+
+}  // namespace android
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 41e20f8..395f164 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -22,32 +22,51 @@
 
 #include <utils/threads.h>
 #include <utils/Log.h>
+#include <utils/RefBase.h>
+#include <media/nbaio/roundup.h>
+#include <media/SingleStateQueue.h>
+#include <private/media/StaticAudioTrackState.h>
 
 namespace android {
 
 // ----------------------------------------------------------------------------
 
-// Maximum cumulated timeout milliseconds before restarting audioflinger thread
-#define MAX_STARTUP_TIMEOUT_MS  3000    // Longer timeout period at startup to cope with A2DP
-                                        // init time
-#define MAX_RUN_TIMEOUT_MS      1000
-#define WAIT_PERIOD_MS          10
-
-#define CBLK_UNDERRUN   0x01 // set: underrun (out) or overrrun (in), clear: no underrun or overrun
+// for audio_track_cblk_t::mFlags
+#define CBLK_UNDERRUN   0x01 // set by server immediately on output underrun, cleared by client
 #define CBLK_FORCEREADY 0x02 // set: track is considered ready immediately by AudioFlinger,
                              // clear: track is ready when buffer full
 #define CBLK_INVALID    0x04 // track buffer invalidated by AudioFlinger, need to re-create
-#define CBLK_DISABLED   0x08 // track disabled by AudioFlinger due to underrun, need to re-start
+#define CBLK_DISABLED   0x08 // output track disabled by AudioFlinger due to underrun,
+                             // need to re-start.  Unlike CBLK_UNDERRUN, this is not set
+                             // immediately, but only after a long string of underruns.
+// 0x10 unused
+#define CBLK_LOOP_CYCLE 0x20 // set by server each time a loop cycle other than final one completes
+#define CBLK_LOOP_FINAL 0x40 // set by server when the final loop cycle completes
+#define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping
+#define CBLK_OVERRUN   0x100 // set by server immediately on input overrun, cleared by client
+#define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer()
+#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
+
+//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
 
 struct AudioTrackSharedStreaming {
     // similar to NBAIO MonoPipe
-    volatile int32_t mFront;
-    volatile int32_t mRear;
+    // in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
+    volatile int32_t mFront;    // read by server
+    volatile int32_t mRear;     // write by client
+    volatile int32_t mFlush;    // incremented by client to indicate a request to flush;
+                                // server notices and discards all data between mFront and mRear
+    volatile uint32_t mUnderrunFrames;  // server increments for each unavailable but desired frame
 };
 
-// future
+typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue;
+
 struct AudioTrackSharedStatic {
-    int mReserved;
+    StaticAudioTrackSingleStateQueue::Shared
+                    mSingleStateQueue;
+    size_t          mBufferPosition;    // updated asynchronously by server,
+                                        // "for entertainment purposes only"
 };
 
 // ----------------------------------------------------------------------------
@@ -55,65 +74,63 @@ struct AudioTrackSharedStatic {
 // Important: do not add any virtual methods, including ~
 struct audio_track_cblk_t
 {
+                // Since the control block is always located in shared memory, this constructor
+                // is only used for placement new().  It is never used for regular new() or stack.
+                            audio_track_cblk_t();
+                /*virtual*/ ~audio_track_cblk_t() { }
+
                 friend class Proxy;
+                friend class ClientProxy;
                 friend class AudioTrackClientProxy;
                 friend class AudioRecordClientProxy;
                 friend class ServerProxy;
+                friend class AudioTrackServerProxy;
+                friend class AudioRecordServerProxy;
 
     // The data members are grouped so that members accessed frequently and in the same context
     // are in the same line of data cache.
-                Mutex       lock;           // sizeof(int)
-                Condition   cv;             // sizeof(int)
 
-                // next 4 are offsets within "buffers"
-    volatile    uint32_t    user;
-    volatile    uint32_t    server;
-                uint32_t    userBase;
-                uint32_t    serverBase;
-
-                int         mPad1;          // unused, but preserves cache line alignment
+                uint32_t    mServer;    // Number of filled frames consumed by server (mIsOut),
+                                        // or filled frames provided by server (!mIsOut).
+                                        // It is updated asynchronously by server without a barrier.
+                                        // The value should be used "for entertainment purposes only",
+                                        // which means don't make important decisions based on it.
 
                 size_t      frameCount_;    // used during creation to pass actual track buffer size
                                             // from AudioFlinger to client, and not referenced again
-                                            // FIXME remove here and replace by createTrack() in/out parameter
+                                            // FIXME remove here and replace by createTrack() in/out
+                                            // parameter
                                             // renamed to "_" to detect incorrect use
 
-                // Cache line boundary (32 bytes)
+    volatile    int32_t     mFutex;     // event flag: down (P) by client,
+                                        // up (V) by server or binderDied() or interrupt()
+#define CBLK_FUTEX_WAKE 1               // if event flag bit is set, then a deferred wake is pending
 
-                uint32_t    loopStart;
-                uint32_t    loopEnd;        // read-only for server, read/write for client
-                int         loopCount;      // read/write for client
+private:
+
+                size_t      mMinimum;       // server wakes up client if available >= mMinimum
 
                 // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
                 // Left channel is in [0:15], right channel is in [16:31].
                 // Always read and write the combined pair atomically.
                 // For AudioTrack only, not used by AudioRecord.
-private:
                 uint32_t    mVolumeLR;
 
                 uint32_t    mSampleRate;    // AudioTrack only: client's requested sample rate in Hz
                                             // or 0 == default. Write-only client, read-only server.
 
-                uint8_t     mPad2;           // unused
-
-public:
-                // read-only for client, server writes once at initialization and is then read-only
-                uint8_t     mName;           // normal tracks: track name, fast tracks: track index
-
-                // used by client only
-                uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting
-                                             // audioflinger
-
-                uint16_t    waitTimeMs;      // Cumulated wait time, used by client only
-private:
                 // client write-only, server read-only
                 uint16_t    mSendLevel;      // Fixed point U4.12 so 0x1000 means 1.0
+
+                uint16_t    mPad2;           // unused
+
 public:
-    volatile    int32_t     flags;
+
+    volatile    int32_t     mFlags;         // combinations of CBLK_*
 
                 // Cache line boundary (32 bytes)
 
-#if 0
+public:
                 union {
                     AudioTrackSharedStreaming   mStreaming;
                     AudioTrackSharedStatic      mStatic;
@@ -121,25 +138,6 @@ public:
                 } u;
 
                 // Cache line boundary (32 bytes)
-#endif
-
-                // Since the control block is always located in shared memory, this constructor
-                // is only used for placement new().  It is never used for regular new() or stack.
-                            audio_track_cblk_t();
-
-private:
-                // if there is a shared buffer, "buffers" is the value of pointer() for the shared
-                // buffer, otherwise "buffers" points immediately after the control block
-                void*       buffer(void *buffers, uint32_t frameSize, size_t offset) const;
-
-                bool        tryLock();
-
-                // isOut == true means AudioTrack, isOut == false means AudioRecord
-                bool        stepServer(size_t stepCount, size_t frameCount, bool isOut);
-                uint32_t    stepUser(size_t stepCount, size_t frameCount, bool isOut);
-                uint32_t    framesAvailable(size_t frameCount, bool isOut);
-                uint32_t    framesAvailable_l(size_t frameCount, bool isOut);
-                uint32_t    framesReady(bool isOut);
 };
 
 // ----------------------------------------------------------------------------
@@ -147,29 +145,32 @@ private:
 // Proxy for shared memory control block, to isolate callers from needing to know the details.
 // There is exactly one ClientProxy and one ServerProxy per shared memory control block.
 // The proxies are located in normal memory, and are not multi-thread safe within a given side.
-class Proxy {
+class Proxy : public RefBase {
 protected:
-    Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
-        : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize) { }
+    Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut,
+            bool clientInServer);
     virtual ~Proxy() { }
 
 public:
-    void*   buffer(size_t offset) const {
-        return mCblk->buffer(mBuffers, mFrameSize, offset);
-    }
+    struct Buffer {
+        size_t  mFrameCount;            // number of frames available in this buffer
+        void*   mRaw;                   // pointer to first frame
+        size_t  mNonContig;             // number of additional non-contiguous frames available
+    };
 
 protected:
     // These refer to shared memory, and are virtual addresses with respect to the current process.
     // They may have different virtual addresses within the other process.
-    audio_track_cblk_t* const   mCblk;          // the control block
-    void* const                 mBuffers;       // starting address of buffers
-
-    const size_t                mFrameCount;    // not necessarily a power of 2
-    const size_t                mFrameSize;     // in bytes
-#if 0
-    const size_t                mFrameCountP2;  // mFrameCount rounded to power of 2, streaming mode
-#endif
-
+    audio_track_cblk_t* const   mCblk;  // the control block
+    void* const     mBuffers;           // starting address of buffers
+
+    const size_t    mFrameCount;        // not necessarily a power of 2
+    const size_t    mFrameSize;         // in bytes
+    const size_t    mFrameCountP2;      // mFrameCount rounded to power of 2, streaming mode
+    const bool      mIsOut;             // true for AudioTrack, false for AudioRecord
+    const bool      mClientInServer;    // true for OutputTrack, false for AudioTrack & AudioRecord
+    bool            mIsShutdown;        // latch set to true when shared memory corruption detected
+    size_t          mUnreleased;        // unreleased frames remaining from most recent obtainBuffer
 };
 
 // ----------------------------------------------------------------------------
@@ -177,9 +178,88 @@ protected:
 // Proxy seen by AudioTrack client and AudioRecord client
 class ClientProxy : public Proxy {
 protected:
-    ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
-        : Proxy(cblk, buffers, frameCount, frameSize) { }
+    ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
+            bool isOut, bool clientInServer);
     virtual ~ClientProxy() { }
+
+public:
+    static const struct timespec kForever;
+    static const struct timespec kNonBlocking;
+
+    // Obtain a buffer with filled frames (reading) or empty frames (writing).
+    // It is permitted to call obtainBuffer() multiple times in succession, without any intervening
+    // calls to releaseBuffer().  In that case, the final obtainBuffer() is the one that effectively
+    // sets or extends the unreleased frame count.
+    // On entry:
+    //  buffer->mFrameCount should be initialized to maximum number of desired frames,
+    //      which must be > 0.
+    //  buffer->mNonContig is unused.
+    //  buffer->mRaw is unused.
+    //  requested is the requested timeout in local monotonic delta time units:
+    //      NULL or &kNonBlocking means non-blocking (zero timeout).
+    //      &kForever means block forever (infinite timeout).
+    //      Other values mean a specific timeout in local monotonic delta time units.
+    //  elapsed is a pointer to a location that will hold the total local monotonic time that
+    //      elapsed while blocked, or NULL if not needed.
+    // On exit:
+    //  buffer->mFrameCount has the actual number of contiguous available frames,
+    //      which is always 0 when the return status != NO_ERROR.
+    //  buffer->mNonContig is the number of additional non-contiguous available frames.
+    //  buffer->mRaw is a pointer to the first available frame,
+    //      or NULL when buffer->mFrameCount == 0.
+    // The return status is one of:
+    //  NO_ERROR    Success, buffer->mFrameCount > 0.
+    //  WOULD_BLOCK Non-blocking mode and no frames are available.
+    //  TIMED_OUT   Timeout occurred before any frames became available.
+    //              This can happen even for infinite timeout, due to a spurious wakeup.
+    //              In this case, the caller should investigate and then re-try as appropriate.
+    //  DEAD_OBJECT Server has died or invalidated, caller should destroy this proxy and re-create.
+    //  -EINTR      Call has been interrupted.  Look around to see why, and then perhaps try again.
+    //  NO_INIT     Shared memory is corrupt.
+    // Assertion failure on entry, if buffer == NULL or buffer->mFrameCount == 0.
+    status_t    obtainBuffer(Buffer* buffer, const struct timespec *requested = NULL,
+            struct timespec *elapsed = NULL);
+
+    // Release (some of) the frames last obtained.
+    // On entry, buffer->mFrameCount should have the number of frames to release,
+    // which must (cumulatively) be <= the number of frames last obtained but not yet released.
+    // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer().
+    // It is permitted to call releaseBuffer() multiple times to release the frames in chunks.
+    // On exit:
+    //  buffer->mFrameCount is zero.
+    //  buffer->mRaw is NULL.
+    void        releaseBuffer(Buffer* buffer);
+
+    // Call after detecting server's death
+    void        binderDied();
+
+    // Call to force an obtainBuffer() to return quickly with -EINTR
+    void        interrupt();
+
+    size_t      getPosition() {
+        return mEpoch + mCblk->mServer;
+    }
+
+    void        setEpoch(size_t epoch) {
+        mEpoch = epoch;
+    }
+
+    void        setMinimum(size_t minimum) {
+        mCblk->mMinimum = minimum;
+    }
+
+    // Return the number of frames that would need to be obtained and released
+    // in order for the client to be aligned at start of buffer
+    virtual size_t  getMisalignment();
+
+    size_t      getEpoch() const {
+        return mEpoch;
+    }
+
+    size_t      getFramesFilled();
+
+private:
+    size_t      mEpoch;
 };
 
 // ----------------------------------------------------------------------------
@@ -187,8 +267,10 @@ protected:
 // Proxy used by AudioTrack client, which also includes AudioFlinger::PlaybackThread::OutputTrack
 class AudioTrackClientProxy : public ClientProxy {
 public:
-    AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
-        : ClientProxy(cblk, buffers, frameCount, frameSize) { }
+    AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize, bool clientInServer = false)
+        : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
+          clientInServer) { }
     virtual ~AudioTrackClientProxy() { }
 
     // No barriers on the following operations, so the ordering of loads/stores
@@ -208,27 +290,42 @@ public:
         mCblk->mSampleRate = sampleRate;
     }
 
-    // called by:
-    //   PlaybackThread::OutputTrack::write
-    //   AudioTrack::createTrack_l
-    //   AudioTrack::releaseBuffer
-    //   AudioTrack::reload
-    //   AudioTrack::restoreTrack_l (2 places)
-    size_t      stepUser(size_t stepCount) {
-        return mCblk->stepUser(stepCount, mFrameCount, true /*isOut*/);
+    virtual void flush();
+
+    virtual uint32_t    getUnderrunFrames() const {
+        return mCblk->u.mStreaming.mUnderrunFrames;
     }
 
-    // called by AudioTrack::obtainBuffer and AudioTrack::processBuffer
-    size_t      framesAvailable() {
-        return mCblk->framesAvailable(mFrameCount, true /*isOut*/);
+    bool        clearStreamEndDone();   // and return previous value
+
+    bool        getStreamEndDone() const;
+
+    status_t    waitStreamEndDone(const struct timespec *requested);
+};
+
+class StaticAudioTrackClientProxy : public AudioTrackClientProxy {
+public:
+    StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize);
+    virtual ~StaticAudioTrackClientProxy() { }
+
+    virtual void    flush();
+
+#define MIN_LOOP    16  // minimum length of each loop iteration in frames
+            void    setLoop(size_t loopStart, size_t loopEnd, int loopCount);
+            size_t  getBufferPosition();
+
+    virtual size_t  getMisalignment() {
+        return 0;
     }
 
-    // called by AudioTrack::obtainBuffer and PlaybackThread::OutputTrack::obtainBuffer
-    // FIXME remove this API since it assumes a lock that should be invisible to caller
-    size_t      framesAvailable_l() {
-        return mCblk->framesAvailable_l(mFrameCount, true /*isOut*/);
+    virtual uint32_t    getUnderrunFrames() const {
+        return 0;
     }
 
+private:
+    StaticAudioTrackSingleStateQueue::Mutator   mMutator;
+    size_t          mBufferPosition;    // so that getBufferPosition() appears to be synchronous
 };
 
 // ----------------------------------------------------------------------------
@@ -236,60 +333,133 @@ public:
 // Proxy used by AudioRecord client
 class AudioRecordClientProxy : public ClientProxy {
 public:
-    AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
-        : ClientProxy(cblk, buffers, frameCount, frameSize) { }
+    AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize)
+        : ClientProxy(cblk, buffers, frameCount, frameSize,
+            false /*isOut*/, false /*clientInServer*/) { }
     ~AudioRecordClientProxy() { }
-
-    // called by AudioRecord::releaseBuffer
-    size_t      stepUser(size_t stepCount) {
-        return mCblk->stepUser(stepCount, mFrameCount, false /*isOut*/);
-    }
-
-    // called by AudioRecord::processBuffer
-    size_t      framesAvailable() {
-        return mCblk->framesAvailable(mFrameCount, false /*isOut*/);
-    }
-
-    // called by AudioRecord::obtainBuffer
-    size_t      framesReady() {
-        return mCblk->framesReady(false /*isOut*/);
-    }
-
 };
 
 // ----------------------------------------------------------------------------
 
 // Proxy used by AudioFlinger server
 class ServerProxy : public Proxy {
+protected:
+    ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
+            bool isOut, bool clientInServer);
 public:
-    ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut)
-        : Proxy(cblk, buffers, frameCount, frameSize), mIsOut(isOut) { }
     virtual ~ServerProxy() { }
 
-    // for AudioTrack and AudioRecord
-    bool        step(size_t stepCount) { return mCblk->stepServer(stepCount, mFrameCount, mIsOut); }
+    // Obtain a buffer with filled frames (writing) or empty frames (reading).
+    // It is permitted to call obtainBuffer() multiple times in succession, without any intervening
+    // calls to releaseBuffer().  In that case, the final obtainBuffer() is the one that effectively
+    // sets or extends the unreleased frame count.
+    // Always non-blocking.
+    // On entry:
+    //  buffer->mFrameCount should be initialized to maximum number of desired frames,
+    //      which must be > 0.
+    //  buffer->mNonContig is unused.
+    //  buffer->mRaw is unused.
+    // On exit:
+    //  buffer->mFrameCount has the actual number of contiguous available frames,
+    //      which is always 0 when the return status != NO_ERROR.
+    //  buffer->mNonContig is the number of additional non-contiguous available frames.
+    //  buffer->mRaw is a pointer to the first available frame,
+    //      or NULL when buffer->mFrameCount == 0.
+    // The return status is one of:
+    //  NO_ERROR    Success, buffer->mFrameCount > 0.
+    //  WOULD_BLOCK No frames are available.
+    //  NO_INIT     Shared memory is corrupt.
+    virtual status_t    obtainBuffer(Buffer* buffer);
+
+    // Release (some of) the frames last obtained.
+    // On entry, buffer->mFrameCount should have the number of frames to release,
+    // which must (cumulatively) be <= the number of frames last obtained but not yet released.
+    // It is permitted to call releaseBuffer() multiple times to release the frames in chunks.
+    // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer().
+    // On exit:
+    //  buffer->mFrameCount is zero.
+    //  buffer->mRaw is NULL.
+    virtual void        releaseBuffer(Buffer* buffer);
 
+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
+};
+
+// Proxy used by AudioFlinger for servicing AudioTrack
+class AudioTrackServerProxy : public ServerProxy {
+public:
+    AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize, bool clientInServer = false)
+        : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) { }
+protected:
+    virtual ~AudioTrackServerProxy() { }
+
+public:
     // return value of these methods must be validated by the caller
     uint32_t    getSampleRate() const { return mCblk->mSampleRate; }
     uint16_t    getSendLevel_U4_12() const { return mCblk->mSendLevel; }
     uint32_t    getVolumeLR() const { return mCblk->mVolumeLR; }
 
-    // for AudioTrack only
-    size_t      framesReady() {
-        ALOG_ASSERT(mIsOut);
-        return mCblk->framesReady(true);
-    }
+    // estimated total number of filled frames available to server to read,
+    // which may include non-contiguous frames
+    virtual size_t      framesReady();
 
-    // for AudioRecord only, called by RecordThread::RecordTrack::getNextBuffer
-    // FIXME remove this API since it assumes a lock that should be invisible to caller
-    size_t      framesAvailableIn_l() {
-        ALOG_ASSERT(!mIsOut);
-        return mCblk->framesAvailable_l(mFrameCount, false);
-    }
+    // Currently AudioFlinger will call framesReady() for a fast track from two threads:
+    // FastMixer thread, and normal mixer thread.  This is dangerous, as the proxy is intended
+    // to be called from at most one thread of server, and one thread of client.
+    // As a temporary workaround, this method informs the proxy implementation that it
+    // should avoid doing a state queue poll from within framesReady().
+    // FIXME Change AudioFlinger to not call framesReady() from normal mixer thread.
+    virtual void        framesReadyIsCalledByMultipleThreads() { }
+
+    bool     setStreamEndDone();    // and return previous value
+
+    // Add to the tally of underrun frames, and inform client of underrun
+    virtual void        tallyUnderrunFrames(uint32_t frameCount);
+
+    // Return the total number of frames which AudioFlinger desired but were unavailable,
+    // and thus which resulted in an underrun.
+    virtual uint32_t    getUnderrunFrames() const { return mCblk->u.mStreaming.mUnderrunFrames; }
+
+    // Return the total number of frames that AudioFlinger has obtained and released
+    virtual size_t      framesReleased() const { return mCblk->mServer; }
+};
+
+class StaticAudioTrackServerProxy : public AudioTrackServerProxy {
+public:
+    StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize);
+protected:
+    virtual ~StaticAudioTrackServerProxy() { }
+
+public:
+    virtual size_t      framesReady();
+    virtual void        framesReadyIsCalledByMultipleThreads();
+    virtual status_t    obtainBuffer(Buffer* buffer);
+    virtual void        releaseBuffer(Buffer* buffer);
+    virtual void        tallyUnderrunFrames(uint32_t frameCount);
+    virtual uint32_t    getUnderrunFrames() const { return 0; }
 
 private:
-    const bool  mIsOut;     // true for AudioTrack, false for AudioRecord
+    ssize_t             pollPosition(); // poll for state queue update, and return current position
+    StaticAudioTrackSingleStateQueue::Observer  mObserver;
+    size_t              mPosition;  // server's current play position in frames, relative to 0
+    size_t              mEnd;       // cached value computed from mState, safe for asynchronous read
+    bool                mFramesReadyIsCalledByMultipleThreads;
+    StaticAudioTrackState   mState;
+};
 
+// Proxy used by AudioFlinger for servicing AudioRecord
+class AudioRecordServerProxy : public ServerProxy {
+public:
+    AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+            size_t frameSize)
+        : ServerProxy(cblk, buffers, frameCount, frameSize, false /*isOut*/,
+            false /*clientInServer*/) { }
+protected:
+    virtual ~AudioRecordServerProxy() { }
 };
 
 // ----------------------------------------------------------------------------