diff --git a/CleanSpec.mk b/CleanSpec.mk
new file mode 100644
index 0000000..e6d9ebf
--- /dev/null
+++ b/CleanSpec.mk
@@ -0,0 +1,52 @@
+# Copyright (C) 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.
+#
+
+# If you don't need to do a full clean build but would like to touch
+# a file or delete some intermediate files, add a clean step to the end
+# of the list.  These steps will only be run once, if they haven't been
+# run before.
+#
+# E.g.:
+#     $(call add-clean-step, touch -c external/sqlite/sqlite3.h)
+#     $(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/STATIC_LIBRARIES/libz_intermediates)
+#
+# Always use "touch -c" and "rm -f" or "rm -rf" to gracefully deal with
+# files that are missing or have been moved.
+#
+# Use $(PRODUCT_OUT) to get to the "out/target/product/blah/" directory.
+# Use $(OUT_DIR) to refer to the "out" directory.
+#
+# If you need to re-do something that's already mentioned, just copy
+# the command and add it to the bottom of the list.  E.g., if a change
+# that you made last week required touching a file and a change you
+# made today requires touching the same file, just copy the old
+# touch step and add it to the end of the list.
+#
+# ************************************************
+# NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST
+# ************************************************
+
+# For example:
+#$(call add-clean-step, rm -rf $(OUT_DIR)/target/common/obj/APPS/AndroidTests_intermediates)
+#$(call add-clean-step, rm -rf $(OUT_DIR)/target/common/obj/JAVA_LIBRARIES/core_intermediates)
+#$(call add-clean-step, find $(OUT_DIR) -type f -name "IGTalkSession*" -print0 | xargs -0 rm -f)
+#$(call add-clean-step, rm -rf $(PRODUCT_OUT)/data/*)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/SHARED_LIBRARIES/libmedia_native_intermediates)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/lib/libmedia_native.so)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/symbols/system/lib/libmedia_native.so)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libmedia_native.so)
+# ************************************************
+# NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST
+# ************************************************
diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp
index fd91bf2..d10f2e5 100644
--- a/camera/CameraParameters.cpp
+++ b/camera/CameraParameters.cpp
@@ -90,6 +90,7 @@
 const char CameraParameters::KEY_VIDEO_SNAPSHOT_SUPPORTED[] = "video-snapshot-supported";
 const char CameraParameters::KEY_VIDEO_STABILIZATION[] = "video-stabilization";
 const char CameraParameters::KEY_VIDEO_STABILIZATION_SUPPORTED[] = "video-stabilization-supported";
+const char CameraParameters::KEY_LIGHTFX[] = "light-fx";
 
 const char CameraParameters::TRUE[] = "true";
 const char CameraParameters::FALSE[] = "false";
@@ -167,6 +168,10 @@
 const char CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO[] = "continuous-video";
 const char CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE[] = "continuous-picture";
 
+// Values for light fx settings
+const char CameraParameters::LIGHTFX_LOWLIGHT[] = "low-light";
+const char CameraParameters::LIGHTFX_HDR[] = "high-dynamic-range";
+
 CameraParameters::CameraParameters()
                 : mMap()
 {
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 1247588..f60b1a4 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -8,7 +8,7 @@
 	SineSource.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-	libstagefright libmedia libmedia_native libutils libbinder libstagefright_foundation \
+	libstagefright libmedia libutils libbinder libstagefright_foundation \
         libjpeg libgui
 
 LOCAL_C_INCLUDES:= \
@@ -104,7 +104,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright liblog libutils libbinder libgui \
-        libstagefright_foundation libmedia libmedia_native libcutils
+        libstagefright_foundation libmedia libcutils
 
 LOCAL_C_INCLUDES:= \
 	frameworks/av/media/libstagefright \
@@ -127,7 +127,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright liblog libutils libbinder libstagefright_foundation \
-        libmedia libmedia_native libgui libcutils libui
+        libmedia libgui libcutils libui
 
 LOCAL_C_INCLUDES:= \
 	frameworks/av/media/libstagefright \
@@ -151,7 +151,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright liblog libutils libbinder libstagefright_foundation \
-        libmedia libmedia_native libgui libcutils libui
+        libmedia libgui libcutils libui
 
 LOCAL_C_INCLUDES:= \
 	frameworks/av/media/libstagefright \
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index 7636906..eb3296e 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -297,9 +297,11 @@
         AString mime;
         CHECK(format->findString("mime", &mime));
 
+        bool isVideo = !strncasecmp(mime.c_str(), "video/", 6);
+
         if (!haveAudio && !strncasecmp(mime.c_str(), "audio/", 6)) {
             haveAudio = true;
-        } else if (!haveVideo && !strncasecmp(mime.c_str(), "video/", 6)) {
+        } else if (!haveVideo && isVideo) {
             haveVideo = true;
         } else {
             continue;
@@ -320,7 +322,7 @@
 
         err = state->mCodec->configure(
                 format,
-                mNativeWindow->getSurfaceTextureClient(),
+                isVideo ? mNativeWindow->getSurfaceTextureClient() : NULL,
                 NULL /* crypto */,
                 0 /* flags */);
 
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index b92a8a0..1e0e7f8 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -589,7 +589,7 @@
 }
 
 static void usage(const char *me) {
-    fprintf(stderr, "usage: %s\n", me);
+    fprintf(stderr, "usage: %s [options] [input_filename]\n", me);
     fprintf(stderr, "       -h(elp)\n");
     fprintf(stderr, "       -a(udio)\n");
     fprintf(stderr, "       -n repetitions\n");
@@ -607,8 +607,8 @@
                     "(video only)\n");
     fprintf(stderr, "       -S allocate buffers from a surface\n");
     fprintf(stderr, "       -T allocate buffers from a surface texture\n");
-    fprintf(stderr, "       -d(ump) filename (raw stream data to a file)\n");
-    fprintf(stderr, "       -D(ump) filename (decoded PCM data to a file)\n");
+    fprintf(stderr, "       -d(ump) output_filename (raw stream data to a file)\n");
+    fprintf(stderr, "       -D(ump) output_filename (decoded PCM data to a file)\n");
 }
 
 static void dumpCodecProfiles(const sp<IOMX>& omx, bool queryDecoders) {
diff --git a/include/camera/CameraParameters.h b/include/camera/CameraParameters.h
index 5540d32..d521543 100644
--- a/include/camera/CameraParameters.h
+++ b/include/camera/CameraParameters.h
@@ -525,6 +525,10 @@
     // stream and record stabilized videos.
     static const char KEY_VIDEO_STABILIZATION_SUPPORTED[];
 
+    // Supported modes for special effects with light.
+    // Example values: "lowlight,hdr".
+    static const char KEY_LIGHTFX[];
+
     // Value for KEY_ZOOM_SUPPORTED or KEY_SMOOTH_ZOOM_SUPPORTED.
     static const char TRUE[];
     static const char FALSE[];
@@ -664,6 +668,12 @@
     // other modes.
     static const char FOCUS_MODE_CONTINUOUS_PICTURE[];
 
+    // Values for light special effects
+    // Low-light enhancement mode
+    static const char LIGHTFX_LOWLIGHT[];
+    // High-dynamic range mode
+    static const char LIGHTFX_HDR[];
+
 private:
     DefaultKeyedVector<String8,String8>    mMap;
 };
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 156c592..ae444c3 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -43,46 +43,43 @@
      */
     enum event_type {
         EVENT_MORE_DATA = 0,        // Request to read more data from PCM buffer.
-        EVENT_OVERRUN = 1,          // PCM buffer overrun occured.
+        EVENT_OVERRUN = 1,          // PCM buffer overrun occurred.
         EVENT_MARKER = 2,           // Record head is at the specified marker position
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 3,          // Record head is at a new position
                                     // (See setPositionUpdatePeriod()).
     };
 
-    /* Create Buffer on the stack and pass it to obtainBuffer()
-     * and releaseBuffer().
+    /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+     * and releaseBuffer().  See also callback_t for EVENT_MORE_DATA.
      */
 
     class Buffer
     {
     public:
-        enum {
-            MUTE    = 0x00000001
-        };
-        uint32_t    flags;
-        int         channelCount;
-        audio_format_t format;
-        size_t      frameCount;
+        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
+
         size_t      size;           // total size in bytes == frameCount * frameSize
         union {
             void*       raw;
-            short*      i16;
-            int8_t*     i8;
+            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 ready or an overrun condition occurs.
+     * invokes the callback when a new buffer becomes ready or various conditions occur.
      * Parameters:
      *
      * event:   type of event notified (see enum AudioRecord::event_type).
      * user:    Pointer to context for use by the callback receiver.
      * info:    Pointer to optional parameter according to event type:
      *          - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read
-     *          more bytes than indicated by 'size' field and update 'size' if less bytes are
-     *          read.
+     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
+     *            consumed.
      *          - 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.
@@ -98,7 +95,7 @@
      *  - BAD_VALUE: unsupported configuration
      */
 
-     static status_t getMinFrameCount(int* frameCount,
+     static status_t getMinFrameCount(size_t* frameCount,
                                       uint32_t sampleRate,
                                       audio_format_t format,
                                       audio_channel_mask_t channelMask);
@@ -108,7 +105,7 @@
      */
                         AudioRecord();
 
-    /* Creates an AudioRecord track and registers it with AudioFlinger.
+    /* 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.
@@ -120,10 +117,13 @@
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
      * channelMask:        Channel mask.
-     * frameCount:         Total size of track PCM buffer in frames. This defines the
-     *                     latency of the track.
+     * 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 provide new PCM data.
+     *                     to consume new PCM data.
      * 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.
@@ -154,7 +154,7 @@
      *  - 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
-     * */
+     */
             status_t    set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT,
                             uint32_t sampleRate = 0,
                             audio_format_t format = AUDIO_FORMAT_DEFAULT,
@@ -168,14 +168,14 @@
 
 
     /* Result of constructing the AudioRecord. This must be checked
-     * before using any AudioRecord API (except for set()), using
+     * 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;
 
-    /* Returns this track's latency in milliseconds.
-     * This includes the latency due to AudioRecord buffer size
+    /* 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;
@@ -183,15 +183,15 @@
    /* getters, see constructor and set() */
 
             audio_format_t format() const;
-            int         channelCount() const;
-            uint32_t    frameCount() const;
-            size_t      frameSize() const;
+            uint32_t    channelCount() const;
+            size_t      frameCount() const;
+            size_t      frameSize() const { return mFrameSize; }
             audio_source_t inputSource() const;
 
 
     /* After it's created the track is not active. Call start() to
      * make it active. If set, the callback will start being called.
-     * if event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until
+     * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until
      * the specified event occurs on the specified trigger session.
      */
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
@@ -199,12 +199,12 @@
 
     /* Stop a track. If set, 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.
+     * and will drain buffers until the pool is exhausted.
      */
             void        stop();
             bool        stopped() const;
 
-    /* get sample rate for this record track
+    /* Get sample rate for this record track in Hz.
      */
             uint32_t    getSampleRate() const;
 
@@ -258,7 +258,7 @@
      */
             status_t    getPosition(uint32_t *position) const;
 
-    /* returns a handle on the audio input used by this AudioRecord.
+    /* Returns a handle on the audio input used by this AudioRecord.
      *
      * Parameters:
      *  none.
@@ -268,7 +268,7 @@
      */
             audio_io_handle_t    getInput() const;
 
-    /* returns the audio session ID associated with this AudioRecord.
+    /* Returns the audio session ID associated with this AudioRecord.
      *
      * Parameters:
      *  none.
@@ -278,22 +278,30 @@
      */
             int    getSessionId() const;
 
-    /* obtains a buffer of "frameCount" frames. The buffer must be
-     * filled entirely. If the track is stopped, obtainBuffer() returns
+    /* 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.
-     * Buffers will be returned until the pool (buffercount())
+     * 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"
      * parameter.
+     *
+     * Interpretation of waitCount:
+     *  +n  limits wait time to n * WAIT_PERIOD_MS,
+     *  -1  causes an (almost) infinite wait time,
+     *   0  non-blocking.
      */
 
         enum {
-            NO_MORE_BUFFERS = 0x80000001,
+            NO_MORE_BUFFERS = 0x80000001,   // same name in AudioFlinger.h, ok to be different value
             STOPPED = 1
         };
 
             status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+
+    /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */
             void        releaseBuffer(Buffer* audioBuffer);
 
 
@@ -302,16 +310,16 @@
      */
             ssize_t     read(void* buffer, size_t size);
 
-    /* Return the amount of input frames lost in the audio driver since the last call of this
+    /* Return the number of input frames lost in the audio driver since the last call of this
      * function.  Audio driver is expected to reset the value to 0 and restart counting upon
      * returning the current value by this function call.  Such loss typically occurs when the
      * user space process is blocked longer than the capacity of audio driver buffers.
-     * Unit: the number of input audio frames
+     * Units: the number of input audio frames.
      */
             unsigned int  getInputFramesLost() const;
 
 private:
-    /* copying audio tracks is not allowed */
+    /* copying audio record objects is not allowed */
                         AudioRecord(const AudioRecord& other);
             AudioRecord& operator = (const AudioRecord& other);
 
@@ -343,8 +351,7 @@
 
             status_t openRecord_l(uint32_t sampleRate,
                                 audio_format_t format,
-                                audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 audio_io_handle_t input);
             audio_io_handle_t getInput_l();
             status_t restoreRecord_l(audio_track_cblk_t*& cblk);
@@ -355,7 +362,7 @@
     bool                    mActive;            // protected by mLock
 
     // for client callback handler
-    callback_t              mCbf;
+    callback_t              mCbf;               // callback handler for events, or NULL
     void*                   mUserData;
 
     // for notification APIs
@@ -367,9 +374,10 @@
     uint32_t                mUpdatePeriod;      // in ms
 
     // constant after constructor or set()
-    uint32_t                mFrameCount;
+    size_t                  mFrameCount;
     audio_format_t          mFormat;
     uint8_t                 mChannelCount;
+    size_t                  mFrameSize;         // app-level frame size == AudioFlinger frame size
     audio_source_t          mInputSource;
     status_t                mStatus;
     uint32_t                mLatency;
@@ -381,6 +389,7 @@
     sp<IAudioRecord>        mAudioRecord;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;
+    void*                   mBuffers;           // starting address of buffers in shared memory
 
     int                     mPreviousPriority;          // before start()
     SchedPolicy             mPreviousSchedulingGroup;
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 49e1afc..126ef12 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -87,29 +87,26 @@
     static float linearToLog(int volume);
     static int logToLinear(float volume);
 
-    static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
-    static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
-    static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getOutputSamplingRate(uint32_t* samplingRate,
+            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getOutputFrameCount(size_t* frameCount,
+            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getOutputLatency(uint32_t* latency,
+            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
     static status_t getSamplingRate(audio_io_handle_t output,
                                           audio_stream_type_t streamType,
-                                          int* samplingRate);
+                                          uint32_t* samplingRate);
     // returns the number of frames per audio HAL write buffer. Corresponds to
     // audio_stream->get_buffer_size()/audio_stream_frame_size()
     static status_t getFrameCount(audio_io_handle_t output,
                                   audio_stream_type_t stream,
-                                  int* frameCount);
+                                  size_t* frameCount);
     // returns the audio output stream latency in ms. Corresponds to
     // audio_stream_out->get_latency()
     static status_t getLatency(audio_io_handle_t output,
                                audio_stream_type_t stream,
                                uint32_t* latency);
 
-    // DEPRECATED
-    static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);
-
-    // DEPRECATED
-    static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
-
     static bool routedToA2dpOutput(audio_stream_type_t streamType);
 
     static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
@@ -126,10 +123,11 @@
     // - 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(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+    static status_t getRenderPosition(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 unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
+    static size_t getInputFramesLost(audio_io_handle_t ioHandle);
 
     static int newAudioSessionId();
     static void acquireAudioSessionId(int audioSession);
@@ -147,8 +145,8 @@
         NUM_CONFIG_EVENTS
     };
 
-    // audio output descriptor used to cache output configurations in client process to avoid frequent calls
-    // through IAudioFlinger
+    // audio output descriptor used to cache output configurations in client process to avoid
+    // frequent calls through IAudioFlinger
     class OutputDescriptor {
     public:
         OutputDescriptor()
@@ -162,8 +160,8 @@
     };
 
     // Events used to synchronize actions between audio sessions.
-    // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until playback
-    // is complete on another audio session.
+    // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until
+    // playback is complete on another audio session.
     // See definitions in MediaSyncEvent.java
     enum sync_event_t {
         SYNC_EVENT_SAME = -1,             // used internally to indicate restart with same event
@@ -183,8 +181,10 @@
     //
     // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
     //
-    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address);
-    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address);
+    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
+                                                const char *device_address);
+    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+                                                                const char *device_address);
     static status_t setPhoneState(audio_mode_t state);
     static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
     static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
@@ -237,8 +237,8 @@
     static const sp<IAudioPolicyService>& get_audio_policy_service();
 
     // helpers for android.media.AudioManager.getProperty(), see description there for meaning
-    static int32_t getPrimaryOutputSamplingRate();
-    static int32_t getPrimaryOutputFrameCount();
+    static uint32_t getPrimaryOutputSamplingRate();
+    static size_t getPrimaryOutputFrameCount();
 
     // ----------------------------------------------------------------------------
 
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 34108b3..f1b77ab 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -49,13 +49,17 @@
     };
 
     /* Events used by AudioTrack callback function (audio_track_cblk_t).
+     * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
      */
     enum event_type {
         EVENT_MORE_DATA = 0,        // Request to write more data to PCM buffer.
-        EVENT_UNDERRUN = 1,         // PCM buffer underrun occured.
-        EVENT_LOOP_END = 2,         // Sample loop end was reached; playback restarted from loop start if loop count was not 0.
-        EVENT_MARKER = 3,           // Playback head is at the specified marker position (See setMarkerPosition()).
-        EVENT_NEW_POS = 4,          // Playback head is at a new position (See setPositionUpdatePeriod()).
+        EVENT_UNDERRUN = 1,         // PCM buffer underrun occurred.
+        EVENT_LOOP_END = 2,         // Sample loop end was reached; playback restarted from
+                                    // loop start if loop count was not 0.
+        EVENT_MARKER = 3,           // Playback head is at the specified marker position
+                                    // (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.
     };
 
@@ -66,14 +70,6 @@
     class Buffer
     {
     public:
-        enum {
-            MUTE    = 0x00000001
-        };
-        uint32_t    flags;        // 0 or MUTE
-        audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
-        // accessed directly by WebKit ANP callback
-        int         channelCount; // will be removed in the future, do not use
-
         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
@@ -114,7 +110,7 @@
      *  - NO_INIT: audio server or audio hardware not initialized
      */
 
-     static status_t getMinFrameCount(int* frameCount,
+     static status_t getMinFrameCount(size_t* frameCount,
                                       audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
                                       uint32_t sampleRate = 0);
 
@@ -123,7 +119,7 @@
      */
                         AudioTrack();
 
-    /* Creates an audio track and registers it with AudioFlinger.
+    /* Creates an AudioTrack 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.
@@ -137,12 +133,13 @@
      *                     16 bits per sample).
      * channelMask:        Channel mask.
      * 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 AudioTrack could be
      *                     larger if the requested size is not compatible with current audio HAL
      *                     latency.  Zero means to use a default value.
      * flags:              See comments on audio_output_flags_t in <system/audio.h>.
      * cbf:                Callback function. If not null, this function is called periodically
-     *                     to request new PCM data.
+     *                     to provide new PCM data.
      * 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.
@@ -162,18 +159,6 @@
                                     int notificationFrames = 0,
                                     int sessionId        = 0);
 
-                        // DEPRECATED
-                        explicit AudioTrack( int streamType,
-                                    uint32_t sampleRate  = 0,
-                                    int format = AUDIO_FORMAT_DEFAULT,
-                                    int channelMask      = 0,
-                                    int frameCount       = 0,
-                                    uint32_t flags       = (uint32_t) AUDIO_OUTPUT_FLAG_NONE,
-                                    callback_t cbf       = 0,
-                                    void* user           = 0,
-                                    int notificationFrames = 0,
-                                    int sessionId        = 0);
-
     /* Creates an audio track and registers it with AudioFlinger. With this constructor,
      * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
      * identified by the argument sharedBuffer. This prototype is for static buffer playback.
@@ -206,7 +191,7 @@
      *  - INVALID_OPERATION: AudioTrack is already initialized
      *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
-     * */
+     */
             status_t    set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
                             uint32_t sampleRate = 0,
                             audio_format_t format = AUDIO_FORMAT_DEFAULT,
@@ -238,13 +223,13 @@
 
             audio_stream_type_t streamType() const;
             audio_format_t format() const;
-            int         channelCount() const;
+            uint32_t    channelCount() const;
             uint32_t    frameCount() const;
 
     /* Return channelCount * (bit depth per channel / 8).
      * channelCount is determined from channelMask, and bit depth comes from format.
      */
-            size_t      frameSize() const;
+            size_t      frameSize() const { return mFrameSize; }
 
             sp<IMemory>& sharedBuffer();
 
@@ -280,9 +265,14 @@
 
     /* Set volume for this track, mostly used for games' sound effects
      * left and right volumes. Levels must be >= 0.0 and <= 1.0.
+     * This is the older API.  New applications should use setVolume(float) when possible.
      */
             status_t    setVolume(float left, float right);
-            void        getVolume(float* left, float* right) const;
+
+    /* Set volume for all channels.  This is the preferred API for new applications,
+     * especially for multi-channel content.
+     */
+            status_t    setVolume(float volume);
 
     /* Set the send level for this track. An auxiliary effect should be attached
      * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
@@ -290,9 +280,11 @@
             status_t    setAuxEffectSendLevel(float level);
             void        getAuxEffectSendLevel(float* level) const;
 
-    /* Set sample rate for this track, mostly used for games' sound effects
+    /* Set sample rate for this track in Hz, mostly used for games' sound effects
      */
-            status_t    setSampleRate(int sampleRate);
+            status_t    setSampleRate(uint32_t sampleRate);
+
+    /* Return current sample rate in Hz, or 0 if unknown */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -312,7 +304,8 @@
     /* 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.
-     * If the AudioTrack has been opened with no callback function associated, the operation will fail.
+     * If the AudioTrack has been opened with no callback function associated, the operation will
+     * fail.
      *
      * Parameters:
      *
@@ -330,7 +323,8 @@
      * a callback with event type EVENT_NEW_POS is called.
      * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
      * callback.
-     * If the AudioTrack has been opened with no callback function associated, the operation will fail.
+     * If the AudioTrack has been opened with no callback function associated, the operation will
+     * fail.
      *
      * Parameters:
      *
@@ -359,7 +353,8 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful operation
      *  - INVALID_OPERATION: the AudioTrack is not stopped.
-     *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
+     *  - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack
+     *               buffer
      */
             status_t    setPosition(uint32_t position);
             status_t    getPosition(uint32_t *position);
@@ -413,7 +408,7 @@
      * 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.
-     * Buffers will be returned until the pool (buffercount())
+     * 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"
      * parameter.
@@ -422,6 +417,18 @@
      *  +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 {
@@ -482,11 +489,11 @@
             // body of AudioTrackThread::threadLoop()
             bool processAudioBuffer(const sp<AudioTrackThread>& thread);
 
+            // 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,
-                                 audio_channel_mask_t channelMask,
-                                 int frameCount,
+                                 size_t frameCount,
                                  audio_output_flags_t flags,
                                  const sp<IMemory>& sharedBuffer,
                                  audio_io_handle_t output);
@@ -502,37 +509,63 @@
 
     float                   mVolume[2];
     float                   mSendLevel;
-    uint32_t                mFrameCount;
+    size_t                  mFrameCount;            // corresponds to current IAudioTrack
+    size_t                  mReqFrameCount;         // frame count to request the next time a new
+                                                    // IAudioTrack is needed
 
-    audio_track_cblk_t*     mCblk;
-    audio_format_t          mFormat;
+    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;
+
+    audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
     audio_stream_type_t     mStreamType;
     uint8_t                 mChannelCount;
     uint8_t                 mMuted;
     uint8_t                 mReserved;
     audio_channel_mask_t    mChannelMask;
+
+                // 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.
+    size_t                  mFrameSize;             // app-level frame size
+    size_t                  mFrameSizeAF;           // AudioFlinger frame size
+
     status_t                mStatus;
     uint32_t                mLatency;
 
     bool                    mActive;                // protected by mLock
 
     callback_t              mCbf;                   // callback handler for events, or NULL
-    void*                   mUserData;
-    uint32_t                mNotificationFramesReq; // requested number of frames between each notification callback
-    uint32_t                mNotificationFramesAct; // actual number of frames between each notification callback
+    void*                   mUserData;              // for client callback handler
+
+    // for notification APIs
+    uint32_t                mNotificationFramesReq; // requested number of frames between each
+                                                    // notification callback
+    uint32_t                mNotificationFramesAct; // actual number of frames between each
+                                                    // notification callback
     sp<IMemory>             mSharedBuffer;
     int                     mLoopCount;
     uint32_t                mRemainingFrames;
-    uint32_t                mMarkerPosition;
+    uint32_t                mMarkerPosition;        // in frames
     bool                    mMarkerReached;
-    uint32_t                mNewPosition;
-    uint32_t                mUpdatePeriod;
+    uint32_t                mNewPosition;           // in frames
+    uint32_t                mUpdatePeriod;          // in frames
+
     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;
-    status_t                mRestoreStatus;
+
     bool                    mIsTimed;
     int                     mPreviousPriority;          // before start()
     SchedPolicy             mPreviousSchedulingGroup;
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index 65c26f4..b1ed7b0 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -74,7 +74,8 @@
 //                          -ENOENT     no more effect available
 //                          -ENODEV     factory failed to initialize
 //                          -EINVAL     invalid pDescriptor
-//                          -ENOSYS     effect list has changed since last execution of EffectQueryNumberEffects()
+//                          -ENOSYS     effect list has changed since last execution of
+//                                      EffectQueryNumberEffects()
 //        *pDescriptor:     updated with the effect descriptor.
 //
 ////////////////////////////////////////////////////////////////////////////////
@@ -91,12 +92,12 @@
 //
 //    Input:
 //          pEffectUuid:    pointer to the effect uuid.
-//          sessionId:  audio session to which this effect instance will be attached. All effects created
-//              with the same session ID are connected in series and process the same signal stream.
-//              Knowing that two effects are part of the same effect chain can help the library implement
-//              some kind of optimizations.
-//          ioId:   identifies the output or input stream this effect is directed to at audio HAL. For future
-//              use especially with tunneled HW accelerated effects
+//          sessionId:  audio session to which this effect instance will be attached. All effects
+//              created with the same session ID are connected in series and process the same signal
+//              stream.  Knowing that two effects are part of the same effect chain can help the
+//              library implement some kind of optimizations.
+//          ioId:   identifies the output or input stream this effect is directed to at audio HAL.
+//              For future use especially with tunneled HW accelerated effects
 //
 //    Input/Output:
 //          pHandle:        address where to return the effect handle.
@@ -109,7 +110,8 @@
 //        *pHandle:         updated with the effect handle.
 //
 ////////////////////////////////////////////////////////////////////////////////
-int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
+int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
+        effect_handle_t *pHandle);
 
 ////////////////////////////////////////////////////////////////////////////////
 //
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 5170a87..9727143 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -61,8 +61,8 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
-                                track_flags_t flags,
+                                size_t frameCount,
+                                track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
@@ -75,7 +75,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -123,7 +123,8 @@
 
     virtual     status_t    setParameters(audio_io_handle_t ioHandle,
                                     const String8& keyValuePairs) = 0;
-    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0;
+    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys)
+                                    const = 0;
 
     // register a current process for audio output change notifications
     virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
@@ -156,10 +157,10 @@
 
     virtual status_t setVoiceVolume(float volume) = 0;
 
-    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
                                     audio_io_handle_t output) const = 0;
 
-    virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
 
     virtual int newAudioSessionId() = 0;
 
@@ -191,8 +192,8 @@
     // helpers for android.media.AudioManager.getProperty(), see description there for meaning
     // FIXME move these APIs to AudioPolicy to permit a more accurate implementation
     // that looks on primary device for a stream with fast flag, primary flag, or first one.
-    virtual int32_t getPrimaryOutputSamplingRate() = 0;
-    virtual int32_t getPrimaryOutputFrameCount() = 0;
+    virtual uint32_t getPrimaryOutputSamplingRate() = 0;
+    virtual size_t getPrimaryOutputFrameCount() = 0;
 
 };
 
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index cc2e069..f5b0604 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -44,9 +44,10 @@
                                               audio_policy_dev_state_t state,
                                               const char *device_address) = 0;
     virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
-                                                                          const char *device_address) = 0;
+                                                                  const char *device_address) = 0;
     virtual status_t setPhoneState(audio_mode_t state) = 0;
-    virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
+    virtual status_t setForceUse(audio_policy_force_use_t usage,
+                                    audio_policy_forced_cfg_t config) = 0;
     virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
     virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h
index 002b045..7bf3069 100644
--- a/include/media/SoundPool.h
+++ b/include/media/SoundPool.h
@@ -65,8 +65,10 @@
     sp<IMemory> getIMemory() { return mData; }
 
     // hack
-    void init(int numChannels, int sampleRate, audio_format_t format, size_t size, sp<IMemory> data ) {
-        mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; mData = data; }
+    void init(int numChannels, int sampleRate, audio_format_t format, size_t size,
+            sp<IMemory> data ) {
+        mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size;
+            mData = data; }
 
 private:
     void init();
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 29c8fd9..0529bcd 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -263,7 +263,7 @@
 
     unsigned short mLoopCounter; // Current tone loopback count
 
-    int mSamplingRate;  // AudioFlinger Sampling rate
+    uint32_t mSamplingRate;  // AudioFlinger Sampling rate
     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
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 81f42ed..f5d6eb5 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -45,17 +45,15 @@
 // Negotiation of format is based on the data provider and data sink, or the data consumer and
 // data source, exchanging prioritized arrays of offers and counter-offers until a single offer is
 // mutually agreed upon.  Each offer is an NBAIO_Format.  For simplicity and performance,
-// NBAIO_Format is an enum that ties together the most important combinations of the various
+// NBAIO_Format is a typedef that ties together the most important combinations of the various
 // attributes, rather than a struct with separate fields for format, sample rate, channel count,
 // interleave, packing, alignment, etc.  The reason is that NBAIO_Format tries to abstract out only
-// the combinations that are actually needed within AudioFligner.  If the list of combinations grows
+// the combinations that are actually needed within AudioFlinger.  If the list of combinations grows
 // too large, then this decision should be re-visited.
-enum NBAIO_Format {
-    Format_Invalid,
-    Format_SR44_1_C2_I16,   // 44.1 kHz PCM stereo interleaved 16-bit signed
-    Format_SR48_C2_I16,     // 48 kHz PCM stereo interleaved 16-bit signed
-    Format_SR44_1_C1_I16,   // 44.1 kHz PCM mono interleaved 16-bit signed
-    Format_SR48_C1_I16,     // 48 kHz PCM mono interleaved 16-bit signed
+// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
+typedef unsigned NBAIO_Format;
+enum {
+    Format_Invalid
 };
 
 // Return the frame size of an NBAIO_Format in bytes
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 457d5d7..81de6e4 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -42,6 +42,7 @@
 extern const char *MEDIA_MIMETYPE_AUDIO_RAW;
 extern const char *MEDIA_MIMETYPE_AUDIO_FLAC;
 extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
+extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM;
 
 extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
 extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5b133f3..48b6b21 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -27,33 +27,16 @@
 // ----------------------------------------------------------------------------
 
 // 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_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 RESTORE_TIMEOUT_MS      5000    // Maximum waiting time for a track to be restored
 
-#define CBLK_UNDERRUN_MSK       0x0001
-#define CBLK_UNDERRUN_ON        0x0001  // underrun (out) or overrrun (in) indication
-#define CBLK_UNDERRUN_OFF       0x0000  // no underrun
-#define CBLK_DIRECTION_MSK      0x0002
-#define CBLK_DIRECTION_OUT      0x0002  // this cblk is for an AudioTrack
-#define CBLK_DIRECTION_IN       0x0000  // this cblk is for an AudioRecord
-#define CBLK_FORCEREADY_MSK     0x0004
-#define CBLK_FORCEREADY_ON      0x0004  // track is considered ready immediately by AudioFlinger
-#define CBLK_FORCEREADY_OFF     0x0000  // track is ready when buffer full
-#define CBLK_INVALID_MSK        0x0008
-#define CBLK_INVALID_ON         0x0008  // track buffer is invalidated by AudioFlinger:
-#define CBLK_INVALID_OFF        0x0000  // must be re-created
-#define CBLK_DISABLED_MSK       0x0010
-#define CBLK_DISABLED_ON        0x0010  // track disabled by AudioFlinger due to underrun:
-#define CBLK_DISABLED_OFF       0x0000  // must be re-started
-#define CBLK_RESTORING_MSK      0x0020
-#define CBLK_RESTORING_ON       0x0020  // track is being restored after invalidation
-#define CBLK_RESTORING_OFF      0x0000  // by AudioFlinger
-#define CBLK_RESTORED_MSK       0x0040
-#define CBLK_RESTORED_ON        0x0040  // track has been restored after invalidation
-#define CBLK_RESTORED_OFF       0x0040  // by AudioFlinger
-#define CBLK_FAST               0x0080  // AudioFlinger successfully created a fast track
+#define CBLK_UNDERRUN   0x01 // set: underrun (out) or overrrun (in), clear: no underrun or overrun
+#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
 
 // Important: do not add any virtual methods, including ~
 struct audio_track_cblk_t
@@ -70,12 +53,14 @@
                 uint32_t    userBase;
                 uint32_t    serverBase;
 
-                // 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*       buffers;
-                uint32_t    frameCount;
+                int         mPad1;          // unused, but preserves cache line alignment
 
-                // Cache line boundary
+                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
+                                            // renamed to "_" to detect incorrect use
+
+                // Cache line boundary (32 bytes)
 
                 uint32_t    loopStart;
                 uint32_t    loopEnd;        // read-only for server, read/write for client
@@ -91,16 +76,14 @@
 
                 uint32_t    sampleRate;
 
-                // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
-                // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
-                // 16 bit because data is converted to 16 bit before being stored in buffer
+                uint8_t     mPad2;           // unused
 
                 // read-only for client, server writes once at initialization and is then read-only
-                uint8_t     frameSize;       // would normally be size_t, but 8 bits is plenty
                 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    bufferTimeoutMs; // Maximum cumulated timeout before restarting
+                                             // audioflinger
 
                 uint16_t    waitTimeMs;      // Cumulated wait time, used by client only
 private:
@@ -114,13 +97,29 @@
                 // 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();
-                uint32_t    stepUser(uint32_t frameCount);      // called by client only, where
-                // client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
-                bool        stepServer(uint32_t frameCount);    // called by server only
-                void*       buffer(uint32_t offset) const;
-                uint32_t    framesAvailable();
-                uint32_t    framesAvailable_l();
-                uint32_t    framesReady();                      // called by server only
+
+                // called by client only, where client includes regular
+                // AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
+                uint32_t    stepUserIn(size_t stepCount, size_t frameCount) { return stepUser(stepCount, frameCount, false); }
+                uint32_t    stepUserOut(size_t stepCount, size_t frameCount) { return stepUser(stepCount, frameCount, true); }
+
+                bool        stepServer(size_t stepCount, size_t frameCount, bool isOut);
+
+                // 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, uint32_t offset) const;
+
+                uint32_t    framesAvailableIn(size_t frameCount)
+                                { return framesAvailable(frameCount, false); }
+                uint32_t    framesAvailableOut(size_t frameCount)
+                                { return framesAvailable(frameCount, true); }
+                uint32_t    framesAvailableIn_l(size_t frameCount)
+                                { return framesAvailable_l(frameCount, false); }
+                uint32_t    framesAvailableOut_l(size_t frameCount)
+                                { return framesAvailable_l(frameCount, true); }
+                uint32_t    framesReadyIn() { return framesReady(false); }
+                uint32_t    framesReadyOut() { return framesReady(true); }
+
                 bool        tryLock();
 
                 // No barriers on the following operations, so the ordering of loads/stores
@@ -146,6 +145,12 @@
                     return mVolumeLR;
                 }
 
+private:
+                // isOut == true means AudioTrack, isOut == false means AudioRecord
+                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);
 };
 
 
diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk
index 0ed7e6c..778c5ac 100755
--- a/libvideoeditor/lvpp/Android.mk
+++ b/libvideoeditor/lvpp/Android.mk
@@ -54,7 +54,6 @@
     libGLESv2                 \
     libgui                    \
     libmedia                  \
-    libmedia_native           \
     libdrmframework           \
     libstagefright            \
     libstagefright_foundation \
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index fc9fb49..a47fc15 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -406,8 +406,8 @@
     }
     ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
     if (mTrack) close();
-    int afSampleRate;
-    int afFrameCount;
+    uint32_t afSampleRate;
+    size_t afFrameCount;
     int frameCount;
 
     if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) !=
diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk
index 95ca6fd..3052ad9 100644
--- a/media/libeffects/downmix/Android.mk
+++ b/media/libeffects/downmix/Android.mk
@@ -25,4 +25,6 @@
 
 LOCAL_PRELINK_MODULE := false
 
+LOCAL_CFLAGS += -fvisibility=hidden
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index 5bf052a..f17a6e8 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -58,13 +58,13 @@
         NULL /* no process_reverse function, no reference stream needed */
 };
 
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
     tag : AUDIO_EFFECT_LIBRARY_TAG,
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Downmix Library",
     implementor : "The Android Open Source Project",
-    query_num_effects : DownmixLib_QueryNumberEffects,
-    query_effect : DownmixLib_QueryEffect,
     create_effect : DownmixLib_Create,
     release_effect : DownmixLib_Release,
     get_descriptor : DownmixLib_GetDescriptor,
@@ -159,25 +159,6 @@
 
 /*--- Effect Library Interface Implementation ---*/
 
-int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects) {
-    ALOGV("DownmixLib_QueryNumberEffects()");
-    *pNumEffects = kNbEffects;
-    return 0;
-}
-
-int32_t DownmixLib_QueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) {
-    ALOGV("DownmixLib_QueryEffect() index=%d", index);
-    if (pDescriptor == NULL) {
-        return -EINVAL;
-    }
-    if (index >= (uint32_t)kNbEffects) {
-        return -EINVAL;
-    }
-    memcpy(pDescriptor, gDescriptors[index], sizeof(effect_descriptor_t));
-    return 0;
-}
-
-
 int32_t DownmixLib_Create(const effect_uuid_t *uuid,
         int32_t sessionId,
         int32_t ioId,
diff --git a/media/libeffects/downmix/EffectDownmix.h b/media/libeffects/downmix/EffectDownmix.h
index be3ca3f..cb6b957 100644
--- a/media/libeffects/downmix/EffectDownmix.h
+++ b/media/libeffects/downmix/EffectDownmix.h
@@ -65,9 +65,6 @@
  * Effect API
  *------------------------------------
 */
-int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects);
-int32_t DownmixLib_QueryEffect(uint32_t index,
-        effect_descriptor_t *pDescriptor);
 int32_t DownmixLib_Create(const effect_uuid_t *uuid,
         int32_t sessionId,
         int32_t ioId,
diff --git a/media/libeffects/lvm/lib/Android.mk b/media/libeffects/lvm/lib/Android.mk
index f49267e..bb56c75 100644
--- a/media/libeffects/lvm/lib/Android.mk
+++ b/media/libeffects/lvm/lib/Android.mk
@@ -105,8 +105,6 @@
 
 LOCAL_MODULE:= libmusicbundle
 
-
-
 LOCAL_C_INCLUDES += \
     $(LOCAL_PATH)/Eq/lib \
     $(LOCAL_PATH)/Eq/src \
@@ -121,8 +119,12 @@
     $(LOCAL_PATH)/StereoWidening/src \
     $(LOCAL_PATH)/StereoWidening/lib
 
+LOCAL_CFLAGS += -fvisibility=hidden
+
 include $(BUILD_STATIC_LIBRARY)
 
+
+
 # Reverb library
 include $(CLEAR_VARS)
 
@@ -168,12 +170,11 @@
 
 LOCAL_MODULE:= libreverb
 
-
-
 LOCAL_C_INCLUDES += \
     $(LOCAL_PATH)/Reverb/lib \
     $(LOCAL_PATH)/Reverb/src \
     $(LOCAL_PATH)/Common/lib \
     $(LOCAL_PATH)/Common/src
 
+LOCAL_CFLAGS += -fvisibility=hidden
 include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk
index 4313424..f1af389 100644
--- a/media/libeffects/lvm/wrapper/Android.mk
+++ b/media/libeffects/lvm/wrapper/Android.mk
@@ -9,28 +9,27 @@
 LOCAL_SRC_FILES:= \
 	Bundle/EffectBundle.cpp
 
+LOCAL_CFLAGS += -fvisibility=hidden
+
 LOCAL_MODULE:= libbundlewrapper
 
 LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
 
-
-
 LOCAL_STATIC_LIBRARIES += libmusicbundle
 
 LOCAL_SHARED_LIBRARIES := \
      libcutils \
      libdl
 
-
 LOCAL_C_INCLUDES += \
 	$(LOCAL_PATH)/Bundle \
 	$(LOCAL_PATH)/../lib/Common/lib/ \
 	$(LOCAL_PATH)/../lib/Bundle/lib/ \
 	$(call include-path-for, audio-effects)
 
-
 include $(BUILD_SHARED_LIBRARY)
 
+
 # reverb wrapper
 include $(CLEAR_VARS)
 
@@ -39,12 +38,12 @@
 LOCAL_SRC_FILES:= \
     Reverb/EffectReverb.cpp
 
+LOCAL_CFLAGS += -fvisibility=hidden
+
 LOCAL_MODULE:= libreverbwrapper
 
 LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
 
-
-
 LOCAL_STATIC_LIBRARIES += libreverb
 
 LOCAL_SHARED_LIBRARIES := \
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index d706c2d..94b9acf 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -158,42 +158,6 @@
 int Effect_setEnabled(EffectContext *pContext, bool enabled);
 
 /* Effect Library Interface Implementation */
-extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects){
-    ALOGV("\n\tEffectQueryNumberEffects start");
-    *pNumEffects = 4;
-    ALOGV("\tEffectQueryNumberEffects creating %d effects", *pNumEffects);
-    ALOGV("\tEffectQueryNumberEffects end\n");
-    return 0;
-}     /* end EffectQueryNumberEffects */
-
-extern "C" int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor){
-    ALOGV("\n\tEffectQueryEffect start");
-    ALOGV("\tEffectQueryEffect processing index %d", index);
-
-    if (pDescriptor == NULL){
-        ALOGV("\tLVM_ERROR : EffectQueryEffect was passed NULL pointer");
-        return -EINVAL;
-    }
-    if (index > 3){
-        ALOGV("\tLVM_ERROR : EffectQueryEffect index out of range %d", index);
-        return -ENOENT;
-    }
-    if(index == LVM_BASS_BOOST){
-        ALOGV("\tEffectQueryEffect processing LVM_BASS_BOOST");
-        *pDescriptor = gBassBoostDescriptor;
-    }else if(index == LVM_VIRTUALIZER){
-        ALOGV("\tEffectQueryEffect processing LVM_VIRTUALIZER");
-        *pDescriptor = gVirtualizerDescriptor;
-    } else if(index == LVM_EQUALIZER){
-        ALOGV("\tEffectQueryEffect processing LVM_EQUALIZER");
-        *pDescriptor = gEqualizerDescriptor;
-    } else if(index == LVM_VOLUME){
-        ALOGV("\tEffectQueryEffect processing LVM_VOLUME");
-        *pDescriptor = gVolumeDescriptor;
-    }
-    ALOGV("\tEffectQueryEffect end\n");
-    return 0;
-}     /* end EffectQueryEffect */
 
 extern "C" int EffectCreate(const effect_uuid_t *uuid,
                             int32_t             sessionId,
@@ -3299,13 +3263,13 @@
     NULL,
 };    /* end gLvmEffectInterface */
 
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
     tag : AUDIO_EFFECT_LIBRARY_TAG,
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Effect Bundle Library",
     implementor : "NXP Software Ltd.",
-    query_num_effects : android::EffectQueryNumberEffects,
-    query_effect : android::EffectQueryEffect,
     create_effect : android::EffectCreate,
     release_effect : android::EffectRelease,
     get_descriptor : android::EffectGetDescriptor,
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
old mode 100755
new mode 100644
index 941d651..87e2c85
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -186,30 +186,6 @@
 int Reverb_LoadPreset       (ReverbContext   *pContext);
 
 /* Effect Library Interface Implementation */
-extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects){
-    ALOGV("\n\tEffectQueryNumberEffects start");
-    *pNumEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
-    ALOGV("\tEffectQueryNumberEffects creating %d effects", *pNumEffects);
-    ALOGV("\tEffectQueryNumberEffects end\n");
-    return 0;
-}     /* end EffectQueryNumberEffects */
-
-extern "C" int EffectQueryEffect(uint32_t index,
-                                 effect_descriptor_t *pDescriptor){
-    ALOGV("\n\tEffectQueryEffect start");
-    ALOGV("\tEffectQueryEffect processing index %d", index);
-    if (pDescriptor == NULL){
-        ALOGV("\tLVM_ERROR : EffectQueryEffect was passed NULL pointer");
-        return -EINVAL;
-    }
-    if (index >= sizeof(gDescriptors) / sizeof(const effect_descriptor_t *)) {
-        ALOGV("\tLVM_ERROR : EffectQueryEffect index out of range %d", index);
-        return -ENOENT;
-    }
-    *pDescriptor = *gDescriptors[index];
-    ALOGV("\tEffectQueryEffect end\n");
-    return 0;
-}     /* end EffectQueryEffect */
 
 extern "C" int EffectCreate(const effect_uuid_t *uuid,
                             int32_t             sessionId,
@@ -2170,13 +2146,13 @@
     NULL,
 };    /* end gReverbInterface */
 
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
     tag : AUDIO_EFFECT_LIBRARY_TAG,
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Reverb Library",
     implementor : "NXP Software Ltd.",
-    query_num_effects : android::EffectQueryNumberEffects,
-    query_effect : android::EffectQueryEffect,
     create_effect : android::EffectCreate,
     release_effect : android::EffectRelease,
     get_descriptor : android::EffectGetDescriptor,
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
old mode 100755
new mode 100644
index c13b9d4..dfa1711
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -29,4 +29,6 @@
 LOCAL_SHARED_LIBRARIES += libdl
 endif
 
+LOCAL_CFLAGS += -fvisibility=hidden
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
old mode 100755
new mode 100644
index 597866a..25586e8
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -1818,30 +1818,6 @@
 // Effect Library Interface Implementation
 //------------------------------------------------------------------------------
 
-int PreProcessingLib_QueryNumberEffects(uint32_t *pNumEffects)
-{
-    if (PreProc_Init() != 0) {
-        return sInitStatus;
-    }
-    if (pNumEffects == NULL) {
-        return -EINVAL;
-    }
-    *pNumEffects = PREPROC_NUM_EFFECTS;
-    return sInitStatus;
-}
-
-int PreProcessingLib_QueryEffect(uint32_t index, effect_descriptor_t *pDescriptor)
-{
-    if (PreProc_Init() != 0) {
-        return sInitStatus;
-    }
-    if (index >= PREPROC_NUM_EFFECTS) {
-        return -EINVAL;
-    }
-    *pDescriptor = *sDescriptors[index];
-    return 0;
-}
-
 int PreProcessingLib_Create(const effect_uuid_t *uuid,
                             int32_t             sessionId,
                             int32_t             ioId,
@@ -1913,13 +1889,13 @@
     return 0;
 }
 
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
     tag : AUDIO_EFFECT_LIBRARY_TAG,
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Audio Preprocessing Library",
     implementor : "The Android Open Source Project",
-    query_num_effects : PreProcessingLib_QueryNumberEffects,
-    query_effect : PreProcessingLib_QueryEffect,
     create_effect : PreProcessingLib_Create,
     release_effect : PreProcessingLib_Release,
     get_descriptor : PreProcessingLib_GetDescriptor
diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp
index 90ebe1f..c35453b 100644
--- a/media/libeffects/testlibs/EffectEqualizer.cpp
+++ b/media/libeffects/testlibs/EffectEqualizer.cpp
@@ -123,23 +123,6 @@
 //--- Effect Library Interface Implementation
 //
 
-extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects) {
-    *pNumEffects = 1;
-    return 0;
-} /* end EffectQueryNumberEffects */
-
-extern "C" int EffectQueryEffect(uint32_t index,
-                                 effect_descriptor_t *pDescriptor) {
-    if (pDescriptor == NULL) {
-        return -EINVAL;
-    }
-    if (index > 0) {
-        return -EINVAL;
-    }
-    *pDescriptor = gEqualizerDescriptor;
-    return 0;
-} /* end EffectQueryNext */
-
 extern "C" int EffectCreate(const effect_uuid_t *uuid,
                             int32_t sessionId,
                             int32_t ioId,
@@ -771,8 +754,6 @@
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Test Equalizer Library",
     implementor : "The Android Open Source Project",
-    query_num_effects : android::EffectQueryNumberEffects,
-    query_effect : android::EffectQueryEffect,
     create_effect : android::EffectCreate,
     release_effect : android::EffectRelease,
     get_descriptor : android::EffectGetDescriptor,
diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c
index a87a834..c37f392 100644
--- a/media/libeffects/testlibs/EffectReverb.c
+++ b/media/libeffects/testlibs/EffectReverb.c
@@ -94,23 +94,6 @@
 
 /*--- Effect Library Interface Implementation ---*/
 
-int EffectQueryNumberEffects(uint32_t *pNumEffects) {
-    *pNumEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
-    return 0;
-}
-
-int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) {
-    if (pDescriptor == NULL) {
-        return -EINVAL;
-    }
-    if (index >= sizeof(gDescriptors) / sizeof(const effect_descriptor_t *)) {
-        return -EINVAL;
-    }
-    memcpy(pDescriptor, gDescriptors[index],
-            sizeof(effect_descriptor_t));
-    return 0;
-}
-
 int EffectCreate(const effect_uuid_t *uuid,
         int32_t sessionId,
         int32_t ioId,
@@ -2222,8 +2205,6 @@
     .version = EFFECT_LIBRARY_API_VERSION,
     .name = "Test Equalizer Library",
     .implementor = "The Android Open Source Project",
-    .query_num_effects = EffectQueryNumberEffects,
-    .query_effect = EffectQueryEffect,
     .create_effect = EffectCreate,
     .release_effect = EffectRelease,
     .get_descriptor = EffectGetDescriptor,
diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h
index 1fb14a7..e5248fe 100644
--- a/media/libeffects/testlibs/EffectReverb.h
+++ b/media/libeffects/testlibs/EffectReverb.h
@@ -300,9 +300,6 @@
  * Effect API
  *------------------------------------
 */
-int EffectQueryNumberEffects(uint32_t *pNumEffects);
-int EffectQueryEffect(uint32_t index,
-                      effect_descriptor_t *pDescriptor);
 int EffectCreate(const effect_uuid_t *effectUID,
                  int32_t sessionId,
                  int32_t ioId,
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index 76b5110..49cf4fa 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -6,7 +6,7 @@
 LOCAL_SRC_FILES:= \
 	EffectVisualizer.cpp
 
-LOCAL_CFLAGS+= -O2
+LOCAL_CFLAGS+= -O2 -fvisibility=hidden
 
 LOCAL_SHARED_LIBRARIES := \
 	libcutils \
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 44baf93..e7eccf1 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -177,23 +177,6 @@
 //--- Effect Library Interface Implementation
 //
 
-int VisualizerLib_QueryNumberEffects(uint32_t *pNumEffects) {
-    *pNumEffects = 1;
-    return 0;
-}
-
-int VisualizerLib_QueryEffect(uint32_t index,
-                              effect_descriptor_t *pDescriptor) {
-    if (pDescriptor == NULL) {
-        return -EINVAL;
-    }
-    if (index > 0) {
-        return -EINVAL;
-    }
-    *pDescriptor = gVisualizerDescriptor;
-    return 0;
-}
-
 int VisualizerLib_Create(const effect_uuid_t *uuid,
                          int32_t sessionId,
                          int32_t ioId,
@@ -574,14 +557,13 @@
         NULL,
 };
 
-
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
     tag : AUDIO_EFFECT_LIBRARY_TAG,
     version : EFFECT_LIBRARY_API_VERSION,
     name : "Visualizer Library",
     implementor : "The Android Open Source Project",
-    query_num_effects : VisualizerLib_QueryNumberEffects,
-    query_effect : VisualizerLib_QueryEffect,
     create_effect : VisualizerLib_Create,
     release_effect : VisualizerLib_Release,
     get_descriptor : VisualizerLib_GetDescriptor,
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 54666fb..f2b6441 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -54,7 +54,7 @@
 LOCAL_SHARED_LIBRARIES := \
 	libui libcutils libutils libbinder libsonivox libicuuc libexpat \
         libcamera_client libstagefright_foundation \
-        libgui libdl libaudioutils libmedia_native
+        libgui libdl libaudioutils
 
 LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
 
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 680604b..3317d57 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -152,7 +152,8 @@
     mCblk->buffer = (uint8_t *)mCblk + bufOffset;
 
     iEffect->asBinder()->linkToDeath(mIEffectClient);
-    ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled);
+    ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId,
+            mStatus, mEnabled);
 
     return mStatus;
 }
@@ -266,9 +267,11 @@
     uint32_t size = sizeof(int);
     uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
 
-    ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1);
+    ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data,
+            (param->psize == 8) ? *((int *)param->data + 1): -1);
 
-    return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size, &param->status);
+    return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size,
+            &param->status);
 }
 
 status_t AudioEffect::setParameterDeferred(effect_param_t *param)
@@ -321,11 +324,14 @@
         return BAD_VALUE;
     }
 
-    ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1);
+    ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data,
+            (param->psize == 8) ? *((int *)param->data + 1): -1);
 
-    uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
+    uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
+            param->vsize;
 
-    return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param, &psize, param);
+    return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param,
+            &psize, param);
 }
 
 
@@ -346,7 +352,8 @@
 
 void AudioEffect::controlStatusChanged(bool controlGranted)
 {
-    ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf, mUserData);
+    ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf,
+            mUserData);
     if (controlGranted) {
         if (mStatus == ALREADY_EXISTS) {
             mStatus = NO_ERROR;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 8ea6306..c2ef68c 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -36,7 +36,7 @@
 
 // static
 status_t AudioRecord::getMinFrameCount(
-        int* frameCount,
+        size_t* frameCount,
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask)
@@ -54,7 +54,7 @@
     }
 
     if (size == 0) {
-        ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+        ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
     }
@@ -63,7 +63,7 @@
     size <<= 1;
 
     if (audio_is_linear_pcm(format)) {
-        int channelCount = popcount(channelMask);
+        uint32_t channelCount = popcount(channelMask);
         size /= channelCount * audio_bytes_per_sample(format);
     }
 
@@ -119,15 +119,22 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        int frameCountInt,
         callback_t cbf,
         void* user,
         int notificationFrames,
         bool threadCanCallJava,
         int sessionId)
 {
+    // FIXME "int" here is legacy and will be replaced by size_t later
+    if (frameCountInt < 0) {
+        ALOGE("Invalid frame count %d", frameCountInt);
+        return BAD_VALUE;
+    }
+    size_t frameCount = frameCountInt;
 
-    ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount);
+    ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
+            frameCount);
 
     AutoMutex lock(mLock);
 
@@ -155,8 +162,9 @@
     if (!audio_is_input_channel(channelMask)) {
         return BAD_VALUE;
     }
-
-    int channelCount = popcount(channelMask);
+    mChannelMask = channelMask;
+    uint32_t channelCount = popcount(channelMask);
+    mChannelCount = channelCount;
 
     if (sessionId == 0 ) {
         mSessionId = AudioSystem::newAudioSessionId();
@@ -176,7 +184,7 @@
     }
 
     // validate framecount
-    int minFrameCount = 0;
+    size_t minFrameCount = 0;
     status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
     if (status != NO_ERROR) {
         return status;
@@ -194,8 +202,7 @@
     }
 
     // create the IAudioRecord
-    status = openRecord_l(sampleRate, format, channelMask,
-                        frameCount, input);
+    status = openRecord_l(sampleRate, format, frameCount, input);
     if (status != NO_ERROR) {
         return status;
     }
@@ -209,9 +216,14 @@
 
     mFormat = format;
     // Update buffer size in case it has been limited by AudioFlinger during track creation
-    mFrameCount = mCblk->frameCount;
-    mChannelCount = (uint8_t)channelCount;
-    mChannelMask = channelMask;
+    mFrameCount = mCblk->frameCount_;
+
+    if (audio_is_linear_pcm(mFormat)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+    }
+
     mActive = false;
     mCbf = cbf;
     mNotificationFrames = notificationFrames;
@@ -247,25 +259,16 @@
     return mFormat;
 }
 
-int AudioRecord::channelCount() const
+uint32_t AudioRecord::channelCount() const
 {
     return mChannelCount;
 }
 
-uint32_t AudioRecord::frameCount() const
+size_t AudioRecord::frameCount() const
 {
     return mFrameCount;
 }
 
-size_t AudioRecord::frameSize() const
-{
-    if (audio_is_linear_pcm(mFormat)) {
-        return channelCount()*audio_bytes_per_sample(mFormat);
-    } else {
-        return sizeof(uint8_t);
-    }
-}
-
 audio_source_t AudioRecord::inputSource() const
 {
     return mInputSource;
@@ -291,17 +294,19 @@
         mActive = true;
 
         cblk->lock.lock();
-        if (!(cblk->flags & CBLK_INVALID_MSK)) {
+        if (!(cblk->flags & CBLK_INVALID)) {
             cblk->lock.unlock();
             ALOGV("mAudioRecord->start()");
             ret = mAudioRecord->start(event, triggerSession);
             cblk->lock.lock();
             if (ret == DEAD_OBJECT) {
-                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
+                android_atomic_or(CBLK_INVALID, &cblk->flags);
             }
         }
-        if (cblk->flags & CBLK_INVALID_MSK) {
-            ret = restoreRecord_l(cblk);
+        if (cblk->flags & CBLK_INVALID) {
+            audio_track_cblk_t* temp = cblk;
+            ret = restoreRecord_l(temp);
+            cblk = temp;
         }
         cblk->lock.unlock();
         if (ret == NO_ERROR) {
@@ -425,13 +430,13 @@
 status_t AudioRecord::openRecord_l(
         uint32_t sampleRate,
         audio_format_t format,
-        audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         audio_io_handle_t input)
 {
     status_t status;
     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
     if (audioFlinger == 0) {
+        ALOGE("Could not get audioflinger");
         return NO_INIT;
     }
 
@@ -441,7 +446,7 @@
     int originalSessionId = mSessionId;
     sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
                                                        sampleRate, format,
-                                                       channelMask,
+                                                       mChannelMask,
                                                        frameCount,
                                                        IAudioFlinger::TRACK_DEFAULT,
                                                        tid,
@@ -454,20 +459,20 @@
         ALOGE("AudioFlinger could not create record track, status: %d", status);
         return status;
     }
-    sp<IMemory> cblk = record->getCblk();
-    if (cblk == 0) {
+    sp<IMemory> iMem = record->getCblk();
+    if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
     }
     mAudioRecord.clear();
     mAudioRecord = record;
     mCblkMemory.clear();
-    mCblkMemory = cblk;
-    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
-    mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
-    android_atomic_and(~CBLK_DIRECTION_MSK, &mCblk->flags);
-    mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
-    mCblk->waitTimeMs = 0;
+    mCblkMemory = iMem;
+    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+    mCblk = cblk;
+    mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
+    cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+    cblk->waitTimeMs = 0;
     return NO_ERROR;
 }
 
@@ -483,7 +488,7 @@
     audioBuffer->frameCount  = 0;
     audioBuffer->size        = 0;
 
-    uint32_t framesReady = cblk->framesReady();
+    uint32_t framesReady = cblk->framesReadyIn();
 
     if (framesReady == 0) {
         cblk->lock.lock();
@@ -498,17 +503,22 @@
                 cblk->lock.unlock();
                 return WOULD_BLOCK;
             }
-            if (!(cblk->flags & CBLK_INVALID_MSK)) {
+            if (!(cblk->flags & CBLK_INVALID)) {
                 mLock.unlock();
+                // this condition is in shared memory, so if IAudioRecord and control block
+                // are replaced due to mediaserver death or IAudioRecord invalidation then
+                // cv won't be signalled, but fortunately the timeout will limit the wait
                 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
                 cblk->lock.unlock();
                 mLock.lock();
                 if (!mActive) {
                     return status_t(STOPPED);
                 }
+                // IAudioRecord may have been re-created while mLock was unlocked
+                cblk = mCblk;
                 cblk->lock.lock();
             }
-            if (cblk->flags & CBLK_INVALID_MSK) {
+            if (cblk->flags & CBLK_INVALID) {
                 goto create_new_record;
             }
             if (CC_UNLIKELY(result != NO_ERROR)) {
@@ -521,9 +531,11 @@
                     result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
                     cblk->lock.lock();
                     if (result == DEAD_OBJECT) {
-                        android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
+                        android_atomic_or(CBLK_INVALID, &cblk->flags);
 create_new_record:
-                        result = AudioRecord::restoreRecord_l(cblk);
+                        audio_track_cblk_t* temp = cblk;
+                        result = AudioRecord::restoreRecord_l(temp);
+                        cblk = temp;
                     }
                     if (result != NO_ERROR) {
                         ALOGW("obtainBuffer create Track error %d", result);
@@ -539,7 +551,7 @@
             }
             // read the server count again
         start_loop_here:
-            framesReady = cblk->framesReady();
+            framesReady = cblk->framesReadyIn();
         }
         cblk->lock.unlock();
     }
@@ -553,18 +565,15 @@
     }
 
     uint32_t u = cblk->user;
-    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+    uint32_t bufferEnd = cblk->userBase + mFrameCount;
 
     if (framesReq > bufferEnd - u) {
         framesReq = bufferEnd - u;
     }
 
-    audioBuffer->flags       = 0;
-    audioBuffer->channelCount= mChannelCount;
-    audioBuffer->format      = mFormat;
     audioBuffer->frameCount  = framesReq;
-    audioBuffer->size        = framesReq*cblk->frameSize;
-    audioBuffer->raw         = (int8_t*)cblk->buffer(u);
+    audioBuffer->size        = framesReq * mFrameSize;
+    audioBuffer->raw         = cblk->buffer(mBuffers, mFrameSize, u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
 }
@@ -572,7 +581,7 @@
 void AudioRecord::releaseBuffer(Buffer* audioBuffer)
 {
     AutoMutex lock(mLock);
-    mCblk->stepUser(audioBuffer->frameCount);
+    mCblk->stepUserIn(audioBuffer->frameCount, mFrameCount);
 }
 
 audio_io_handle_t AudioRecord::getInput() const
@@ -631,10 +640,12 @@
         status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));
         if (err < 0) {
             // out of buffers, return #bytes written
-            if (err == status_t(NO_MORE_BUFFERS))
+            if (err == status_t(NO_MORE_BUFFERS)) {
                 break;
-            if (err == status_t(TIMED_OUT))
+            }
+            if (err == status_t(TIMED_OUT)) {
                 err = 0;
+            }
             return ssize_t(err);
         }
 
@@ -701,7 +712,8 @@
         status_t err = obtainBuffer(&audioBuffer, 1);
         if (err < NO_ERROR) {
             if (err != TIMED_OUT) {
-                ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
+                ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
+                        "Error obtaining an audio buffer, giving up.");
                 return false;
             }
             break;
@@ -733,11 +745,11 @@
 
 
     // Manage overrun callback
-    if (active && (cblk->framesAvailable() == 0)) {
+    if (active && (cblk->framesAvailableIn(mFrameCount) == 0)) {
         // The value of active is stale, but we are almost sure to be active here because
         // otherwise we would have exited when obtainBuffer returned STOPPED earlier.
         ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
-        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
+        if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
             mCbf(EVENT_OVERRUN, mUserData, NULL);
         }
     }
@@ -753,57 +765,40 @@
 // must be called with mLock and cblk.lock held. Callers must also hold strong references on
 // the IAudioRecord and IMemory in case they are recreated here.
 // If the IAudioRecord is successfully restored, the cblk pointer is updated
-status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& cblk)
+status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk)
 {
     status_t result;
 
-    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
-        ALOGW("dead IAudioRecord, creating a new one");
-        // signal old cblk condition so that other threads waiting for available buffers stop
-        // waiting now
-        cblk->cv.broadcast();
-        cblk->lock.unlock();
+    audio_track_cblk_t* cblk = refCblk;
+    audio_track_cblk_t* newCblk = cblk;
+    ALOGW("dead IAudioRecord, creating a new one");
 
-        // if the new IAudioRecord is created, openRecord_l() will modify the
-        // following member variables: mAudioRecord, mCblkMemory and mCblk.
-        // It will also delete the strong references on previous IAudioRecord and IMemory
-        result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask,
-                mFrameCount, getInput_l());
-        if (result == NO_ERROR) {
-            // callback thread or sync event hasn't changed
-            result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
-        }
-        if (result != NO_ERROR) {
-            mActive = false;
-        }
+    // signal old cblk condition so that other threads waiting for available buffers stop
+    // waiting now
+    cblk->cv.broadcast();
+    cblk->lock.unlock();
 
-        // signal old cblk condition for other threads waiting for restore completion
-        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
-        cblk->cv.broadcast();
-    } else {
-        if (!(cblk->flags & CBLK_RESTORED_MSK)) {
-            ALOGW("dead IAudioRecord, waiting for a new one to be created");
-            mLock.unlock();
-            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
-            cblk->lock.unlock();
-            mLock.lock();
-        } else {
-            ALOGW("dead IAudioRecord, already restored");
-            result = NO_ERROR;
-            cblk->lock.unlock();
-        }
-        if (result != NO_ERROR || !mActive) {
-            result = status_t(STOPPED);
-        }
+    // if the new IAudioRecord is created, openRecord_l() will modify the
+    // following member variables: mAudioRecord, mCblkMemory and mCblk.
+    // It will also delete the strong references on previous IAudioRecord and IMemory
+    result = openRecord_l(cblk->sampleRate, mFormat, mFrameCount, getInput_l());
+    if (result == NO_ERROR) {
+        newCblk = mCblk;
+        // callback thread or sync event hasn't changed
+        result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
     }
+    if (result != NO_ERROR) {
+        mActive = false;
+    }
+
     ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
-        result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
+        result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
 
     if (result == NO_ERROR) {
         // from now on we switch to the newly created cblk
-        cblk = mCblk;
+        refCblk = newCblk;
     }
-    cblk->lock.lock();
+    newCblk->lock.lock();
 
     ALOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result);
 
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 207f96f..028e4a3 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -205,12 +205,7 @@
     return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
 }
 
-// DEPRECATED
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
-    return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
 {
     audio_io_handle_t output;
 
@@ -228,7 +223,7 @@
 
 status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
                                       audio_stream_type_t streamType,
-                                      int* samplingRate)
+                                      uint32_t* samplingRate)
 {
     OutputDescriptor *outputDesc;
 
@@ -246,17 +241,13 @@
         gLock.unlock();
     }
 
-    ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate);
+    ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
+            *samplingRate);
 
     return NO_ERROR;
 }
 
-// DEPRECATED
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
-    return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_t streamType)
 {
     audio_io_handle_t output;
 
@@ -274,7 +265,7 @@
 
 status_t AudioSystem::getFrameCount(audio_io_handle_t output,
                                     audio_stream_type_t streamType,
-                                    int* frameCount)
+                                    size_t* frameCount)
 {
     OutputDescriptor *outputDesc;
 
@@ -290,7 +281,8 @@
         gLock.unlock();
     }
 
-    ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output, *frameCount);
+    ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output,
+            *frameCount);
 
     return NO_ERROR;
 }
@@ -369,7 +361,8 @@
     return af->setVoiceVolume(value);
 }
 
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream)
+status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
+        audio_stream_type_t stream)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
@@ -381,7 +374,7 @@
     return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
 }
 
-unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     unsigned int result = 0;
     if (af == 0) return result;
@@ -449,8 +442,10 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d latency %d",
-                outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %u "
+                "latency %d",
+                outputDesc->samplingRate, outputDesc->format, outputDesc->channels,
+                outputDesc->frameCount, outputDesc->latency);
         } break;
     case OUTPUT_CLOSED: {
         if (gOutputs.indexOfKey(ioHandle) < 0) {
@@ -471,7 +466,8 @@
         if (param2 == NULL) break;
         desc = (const OutputDescriptor *)param2;
 
-        ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x frameCount %d latency %d",
+        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x "
+                "frameCount %d latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channels, desc->frameCount, desc->latency);
         OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -510,7 +506,7 @@
 sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient;
 
 
-// establish binder interface to AudioFlinger service
+// establish binder interface to AudioPolicy service
 const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
 {
     gLock.lock();
@@ -744,14 +740,14 @@
     return NO_ERROR;
 }
 
-int32_t AudioSystem::getPrimaryOutputSamplingRate()
+uint32_t AudioSystem::getPrimaryOutputSamplingRate()
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
     return af->getPrimaryOutputSamplingRate();
 }
 
-int32_t AudioSystem::getPrimaryOutputFrameCount()
+size_t AudioSystem::getPrimaryOutputFrameCount()
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index aec8c4a..e40895a 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -50,11 +50,13 @@
 
 // static
 status_t AudioTrack::getMinFrameCount(
-        int* frameCount,
+        size_t* frameCount,
         audio_stream_type_t streamType,
         uint32_t sampleRate)
 {
-    if (frameCount == NULL) return BAD_VALUE;
+    if (frameCount == NULL) {
+        return BAD_VALUE;
+    }
 
     // default to 0 in case of error
     *frameCount = 0;
@@ -65,11 +67,11 @@
     //          audio_format_t format
     //          audio_channel_mask_t channelMask
     //          audio_output_flags_t flags
-    int afSampleRate;
+    uint32_t afSampleRate;
     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
     }
-    int afFrameCount;
+    size_t afFrameCount;
     if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
         return NO_INIT;
     }
@@ -120,28 +122,6 @@
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
 }
 
-// DEPRECATED
-AudioTrack::AudioTrack(
-        int streamType,
-        uint32_t sampleRate,
-        int format,
-        int channelMask,
-        int frameCount,
-        uint32_t flags,
-        callback_t cbf,
-        void* user,
-        int notificationFrames,
-        int sessionId)
-    : mStatus(NO_INIT),
-      mIsTimed(false),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
-{
-    mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format,
-            (audio_channel_mask_t) channelMask,
-            frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
-            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
-}
-
 AudioTrack::AudioTrack(
         audio_stream_type_t streamType,
         uint32_t sampleRate,
@@ -188,7 +168,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        int frameCountInt,
         audio_output_flags_t flags,
         callback_t cbf,
         void* user,
@@ -197,10 +177,17 @@
         bool threadCanCallJava,
         int sessionId)
 {
+    // FIXME "int" here is legacy and will be replaced by size_t later
+    if (frameCountInt < 0) {
+        ALOGE("Invalid frame count %d", frameCountInt);
+        return BAD_VALUE;
+    }
+    size_t frameCount = frameCountInt;
 
-    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
+            sharedBuffer->size());
 
-    ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
+    ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
 
     AutoMutex lock(mLock);
     if (mAudioTrack != 0) {
@@ -214,7 +201,7 @@
     }
 
     if (sampleRate == 0) {
-        int afSampleRate;
+        uint32_t afSampleRate;
         if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
             return NO_INIT;
         }
@@ -256,7 +243,9 @@
         ALOGE("Invalid channel mask %#x", channelMask);
         return BAD_VALUE;
     }
+    mChannelMask = channelMask;
     uint32_t channelCount = popcount(channelMask);
+    mChannelCount = channelCount;
 
     audio_io_handle_t output = AudioSystem::getOutput(
                                     streamType,
@@ -272,6 +261,7 @@
     mVolume[RIGHT] = 1.0f;
     mSendLevel = 0.0f;
     mFrameCount = frameCount;
+    mReqFrameCount = frameCount;
     mNotificationFramesReq = notificationFrames;
     mSessionId = sessionId;
     mAuxEffectId = 0;
@@ -287,7 +277,6 @@
     status_t status = createTrack_l(streamType,
                                   sampleRate,
                                   format,
-                                  channelMask,
                                   frameCount,
                                   flags,
                                   sharedBuffer,
@@ -305,8 +294,15 @@
 
     mStreamType = streamType;
     mFormat = format;
-    mChannelMask = channelMask;
-    mChannelCount = channelCount;
+
+    if (audio_is_linear_pcm(format)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+        mFrameSizeAF = channelCount * sizeof(int16_t);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+        mFrameSizeAF = sizeof(uint8_t);
+    }
+
     mSharedBuffer = sharedBuffer;
     mMuted = false;
     mActive = false;
@@ -318,7 +314,6 @@
     mUpdatePeriod = 0;
     mFlushed = false;
     AudioSystem::acquireAudioSessionId(mSessionId);
-    mRestoreStatus = NO_ERROR;
     return NO_ERROR;
 }
 
@@ -344,23 +339,14 @@
     return mFormat;
 }
 
-int AudioTrack::channelCount() const
+uint32_t AudioTrack::channelCount() const
 {
     return mChannelCount;
 }
 
-uint32_t AudioTrack::frameCount() const
+size_t AudioTrack::frameCount() const
 {
-    return mCblk->frameCount;
-}
-
-size_t AudioTrack::frameSize() const
-{
-    if (audio_is_linear_pcm(mFormat)) {
-        return channelCount()*audio_bytes_per_sample(mFormat);
-    } else {
-        return sizeof(uint8_t);
-    }
+    return mFrameCount;
 }
 
 sp<IMemory>& AudioTrack::sharedBuffer()
@@ -390,7 +376,7 @@
         cblk->lock.lock();
         cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
         cblk->waitTimeMs = 0;
-        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
+        android_atomic_and(~CBLK_DISABLED, &cblk->flags);
         if (t != 0) {
             t->resume();
         } else {
@@ -399,19 +385,21 @@
             androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
         }
 
-        ALOGV("start %p before lock cblk %p", this, mCblk);
+        ALOGV("start %p before lock cblk %p", this, cblk);
         status_t status = NO_ERROR;
-        if (!(cblk->flags & CBLK_INVALID_MSK)) {
+        if (!(cblk->flags & CBLK_INVALID)) {
             cblk->lock.unlock();
             ALOGV("mAudioTrack->start()");
             status = mAudioTrack->start();
             cblk->lock.lock();
             if (status == DEAD_OBJECT) {
-                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
+                android_atomic_or(CBLK_INVALID, &cblk->flags);
             }
         }
-        if (cblk->flags & CBLK_INVALID_MSK) {
-            status = restoreTrack_l(cblk, true);
+        if (cblk->flags & CBLK_INVALID) {
+            audio_track_cblk_t* temp = cblk;
+            status = restoreTrack_l(temp, true /*fromStart*/);
+            cblk = temp;
         }
         cblk->lock.unlock();
         if (status != NO_ERROR) {
@@ -528,14 +516,9 @@
     return NO_ERROR;
 }
 
-void AudioTrack::getVolume(float* left, float* right) const
+status_t AudioTrack::setVolume(float volume)
 {
-    if (left != NULL) {
-        *left  = mVolume[LEFT];
-    }
-    if (right != NULL) {
-        *right = mVolume[RIGHT];
-    }
+    return setVolume(volume, volume);
 }
 
 status_t AudioTrack::setAuxEffectSendLevel(float level)
@@ -560,9 +543,9 @@
     }
 }
 
-status_t AudioTrack::setSampleRate(int rate)
+status_t AudioTrack::setSampleRate(uint32_t rate)
 {
-    int afSamplingRate;
+    uint32_t afSamplingRate;
 
     if (mIsTimed) {
         return INVALID_OPERATION;
@@ -572,7 +555,9 @@
         return NO_INIT;
     }
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
+    if (rate == 0 || rate > afSamplingRate*2 ) {
+        return BAD_VALUE;
+    }
 
     AutoMutex lock(mLock);
     mCblk->sampleRate = rate;
@@ -582,7 +567,7 @@
 uint32_t AudioTrack::getSampleRate() const
 {
     if (mIsTimed) {
-        return INVALID_OPERATION;
+        return 0;
     }
 
     AutoMutex lock(mLock);
@@ -615,15 +600,17 @@
     }
 
     if (loopStart >= loopEnd ||
-        loopEnd - loopStart > cblk->frameCount ||
+        loopEnd - loopStart > mFrameCount ||
         cblk->server > loopStart) {
-        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
+        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, "
+              "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
         return BAD_VALUE;
     }
 
-    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
-        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
-            loopStart, loopEnd, cblk->frameCount);
+    if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
+        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, "
+            "framecount %d",
+            loopStart, loopEnd, mFrameCount);
         return BAD_VALUE;
     }
 
@@ -637,7 +624,9 @@
 
 status_t AudioTrack::setMarkerPosition(uint32_t marker)
 {
-    if (mCbf == NULL) return INVALID_OPERATION;
+    if (mCbf == NULL) {
+        return INVALID_OPERATION;
+    }
 
     mMarkerPosition = marker;
     mMarkerReached = false;
@@ -647,7 +636,9 @@
 
 status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
 {
-    if (marker == NULL) return BAD_VALUE;
+    if (marker == NULL) {
+        return BAD_VALUE;
+    }
 
     *marker = mMarkerPosition;
 
@@ -656,7 +647,9 @@
 
 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
 {
-    if (mCbf == NULL) return INVALID_OPERATION;
+    if (mCbf == NULL) {
+        return INVALID_OPERATION;
+    }
 
     uint32_t curPosition;
     getPosition(&curPosition);
@@ -668,7 +661,9 @@
 
 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
 {
-    if (updatePeriod == NULL) return BAD_VALUE;
+    if (updatePeriod == NULL) {
+        return BAD_VALUE;
+    }
 
     *updatePeriod = mUpdatePeriod;
 
@@ -677,25 +672,34 @@
 
 status_t AudioTrack::setPosition(uint32_t position)
 {
-    if (mIsTimed) return INVALID_OPERATION;
+    if (mIsTimed) {
+        return INVALID_OPERATION;
+    }
 
     AutoMutex lock(mLock);
 
-    if (!stopped_l()) return INVALID_OPERATION;
+    if (!stopped_l()) {
+        return INVALID_OPERATION;
+    }
 
-    Mutex::Autolock _l(mCblk->lock);
+    audio_track_cblk_t* cblk = mCblk;
+    Mutex::Autolock _l(cblk->lock);
 
-    if (position > mCblk->user) return BAD_VALUE;
+    if (position > cblk->user) {
+        return BAD_VALUE;
+    }
 
-    mCblk->server = position;
-    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
+    cblk->server = position;
+    android_atomic_or(CBLK_FORCEREADY, &cblk->flags);
 
     return NO_ERROR;
 }
 
 status_t AudioTrack::getPosition(uint32_t *position)
 {
-    if (position == NULL) return BAD_VALUE;
+    if (position == NULL) {
+        return BAD_VALUE;
+    }
     AutoMutex lock(mLock);
     *position = mFlushed ? 0 : mCblk->server;
 
@@ -706,11 +710,14 @@
 {
     AutoMutex lock(mLock);
 
-    if (!stopped_l()) return INVALID_OPERATION;
+    if (!stopped_l()) {
+        return INVALID_OPERATION;
+    }
 
     flush_l();
 
-    mCblk->stepUser(mCblk->frameCount);
+    audio_track_cblk_t* cblk = mCblk;
+    cblk->stepUserOut(mFrameCount, mFrameCount);
 
     return NO_ERROR;
 }
@@ -750,8 +757,7 @@
         audio_stream_type_t streamType,
         uint32_t sampleRate,
         audio_format_t format,
-        audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         audio_output_flags_t flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output)
@@ -791,7 +797,7 @@
             // Same comment as below about ignoring frameCount parameter for set()
             frameCount = sharedBuffer->size();
         } else if (frameCount == 0) {
-            int afFrameCount;
+            size_t afFrameCount;
             if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
                 return NO_INIT;
             }
@@ -800,17 +806,16 @@
 
     } else if (sharedBuffer != 0) {
 
-        // Ensure that buffer alignment matches channelCount
-        int channelCount = popcount(channelMask);
+        // Ensure that buffer alignment matches channel count
         // 8-bit data in shared memory is not currently supported by AudioFlinger
         size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
-        if (channelCount > 1) {
+        if (mChannelCount > 1) {
             // More than 2 channels does not require stronger alignment than stereo
             alignment <<= 1;
         }
-        if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
-            ALOGE("Invalid buffer alignment: address %p, channelCount %d",
-                    sharedBuffer->pointer(), channelCount);
+        if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+            ALOGE("Invalid buffer alignment: address %p, channel count %u",
+                    sharedBuffer->pointer(), mChannelCount);
             return BAD_VALUE;
         }
 
@@ -818,16 +823,16 @@
         // there's no frameCount parameter.
         // But when initializing a shared buffer AudioTrack via set(),
         // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+        frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
 
     } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
 
         // FIXME move these calculations and associated checks to server
-        int afSampleRate;
+        uint32_t afSampleRate;
         if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
             return NO_INIT;
         }
-        int afFrameCount;
+        size_t afFrameCount;
         if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
             return NO_INIT;
         }
@@ -836,8 +841,8 @@
         uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
         if (minBufCount < 2) minBufCount = 2;
 
-        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
+        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+        ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
                 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
 
@@ -849,7 +854,7 @@
         }
         // Make sure that application is notified with sufficient margin
         // before underrun
-        if (mNotificationFramesAct > (uint32_t)frameCount/2) {
+        if (mNotificationFramesAct > frameCount/2) {
             mNotificationFramesAct = frameCount/2;
         }
         if (frameCount < minFrameCount) {
@@ -879,10 +884,12 @@
     sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
                                                       streamType,
                                                       sampleRate,
-                                                      format,
-                                                      channelMask,
+                                                      // AudioFlinger only sees 16-bit PCM
+                                                      format == AUDIO_FORMAT_PCM_8_BIT ?
+                                                              AUDIO_FORMAT_PCM_16_BIT : format,
+                                                      mChannelMask,
                                                       frameCount,
-                                                      trackFlags,
+                                                      &trackFlags,
                                                       sharedBuffer,
                                                       output,
                                                       tid,
@@ -893,49 +900,58 @@
         ALOGE("AudioFlinger could not create track, status: %d", status);
         return status;
     }
-    sp<IMemory> cblk = track->getCblk();
-    if (cblk == 0) {
+    sp<IMemory> iMem = track->getCblk();
+    if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
     }
     mAudioTrack = track;
-    mCblkMemory = cblk;
-    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
-    // old has the previous value of mCblk->flags before the "or" operation
-    int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
+    mCblkMemory = iMem;
+    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+    mCblk = cblk;
+    size_t temp = cblk->frameCount_;
+    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
+        // In current design, AudioTrack client checks and ensures frame count validity before
+        // passing it to AudioFlinger so AudioFlinger should not return a different value except
+        // for fast track as it uses a special method of assigning frame count.
+        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+    }
+    frameCount = temp;
     if (flags & AUDIO_OUTPUT_FLAG_FAST) {
-        if (old & CBLK_FAST) {
-            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
+        if (trackFlags & IAudioFlinger::TRACK_FAST) {
+            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
         } else {
-            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
+            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
             // once denied, do not request again if IAudioTrack is re-created
             flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
             mFlags = flags;
         }
         if (sharedBuffer == 0) {
-            mNotificationFramesAct = mCblk->frameCount/2;
+            mNotificationFramesAct = frameCount/2;
         }
     }
     if (sharedBuffer == 0) {
-        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+        mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
     } else {
-        mCblk->buffers = sharedBuffer->pointer();
+        mBuffers = sharedBuffer->pointer();
         // Force buffer full condition as data is already present in shared memory
-        mCblk->stepUser(mCblk->frameCount);
+        cblk->stepUserOut(frameCount, frameCount);
     }
 
-    mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
-    mCblk->setSendLevel(mSendLevel);
+    cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
+            uint16_t(mVolume[LEFT] * 0x1000));
+    cblk->setSendLevel(mSendLevel);
     mAudioTrack->attachAuxEffect(mAuxEffectId);
-    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
-    mCblk->waitTimeMs = 0;
+    cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+    cblk->waitTimeMs = 0;
     mRemainingFrames = mNotificationFramesAct;
     // FIXME don't believe this lie
-    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
+    mLatency = afLatency + (1000*frameCount) / sampleRate;
+    mFrameCount = frameCount;
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
-    if (mCblk->frameCount > mFrameCount) {
-        mFrameCount = mCblk->frameCount;
+    if (frameCount > mReqFrameCount) {
+        mReqFrameCount = frameCount;
     }
     return NO_ERROR;
 }
@@ -952,10 +968,10 @@
     audioBuffer->frameCount  = 0;
     audioBuffer->size = 0;
 
-    uint32_t framesAvail = cblk->framesAvailable();
+    uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
 
     cblk->lock.lock();
-    if (cblk->flags & CBLK_INVALID_MSK) {
+    if (cblk->flags & CBLK_INVALID) {
         goto create_new_track;
     }
     cblk->lock.unlock();
@@ -974,18 +990,23 @@
                 cblk->lock.unlock();
                 return WOULD_BLOCK;
             }
-            if (!(cblk->flags & CBLK_INVALID_MSK)) {
+            if (!(cblk->flags & CBLK_INVALID)) {
                 mLock.unlock();
+                // this condition is in shared memory, so if IAudioTrack and control block
+                // are replaced due to mediaserver death or IAudioTrack invalidation then
+                // cv won't be signalled, but fortunately the timeout will limit the wait
                 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
                 cblk->lock.unlock();
                 mLock.lock();
                 if (!mActive) {
                     return status_t(STOPPED);
                 }
+                // IAudioTrack may have been re-created while mLock was unlocked
+                cblk = mCblk;
                 cblk->lock.lock();
             }
 
-            if (cblk->flags & CBLK_INVALID_MSK) {
+            if (cblk->flags & CBLK_INVALID) {
                 goto create_new_track;
             }
             if (CC_UNLIKELY(result != NO_ERROR)) {
@@ -994,16 +1015,18 @@
                     // timing out when a loop has been set and we have already written upto loop end
                     // is a normal condition: no need to wake AudioFlinger up.
                     if (cblk->user < cblk->loopEnd) {
-                        ALOGW(   "obtainBuffer timed out (is the CPU pegged?) %p name=%#x"
-                                "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server);
+                        ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, "
+                              "server=%08x", this, cblk->mName, cblk->user, cblk->server);
                         //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
                         cblk->lock.unlock();
                         result = mAudioTrack->start();
                         cblk->lock.lock();
                         if (result == DEAD_OBJECT) {
-                            android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
+                            android_atomic_or(CBLK_INVALID, &cblk->flags);
 create_new_track:
-                            result = restoreTrack_l(cblk, false);
+                            audio_track_cblk_t* temp = cblk;
+                            result = restoreTrack_l(temp, false /*fromStart*/);
+                            cblk = temp;
                         }
                         if (result != NO_ERROR) {
                             ALOGW("obtainBuffer create Track error %d", result);
@@ -1021,7 +1044,7 @@
             }
             // read the server count again
         start_loop_here:
-            framesAvail = cblk->framesAvailable_l();
+            framesAvail = cblk->framesAvailableOut_l(mFrameCount);
         }
         cblk->lock.unlock();
     }
@@ -1033,22 +1056,15 @@
     }
 
     uint32_t u = cblk->user;
-    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+    uint32_t bufferEnd = cblk->userBase + mFrameCount;
 
     if (framesReq > bufferEnd - u) {
         framesReq = bufferEnd - u;
     }
 
-    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
-    audioBuffer->channelCount = mChannelCount;
     audioBuffer->frameCount = framesReq;
-    audioBuffer->size = framesReq * cblk->frameSize;
-    if (audio_is_linear_pcm(mFormat)) {
-        audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
-    } else {
-        audioBuffer->format = mFormat;
-    }
-    audioBuffer->raw = (int8_t *)cblk->buffer(u);
+    audioBuffer->size = framesReq * mFrameSizeAF;
+    audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
 }
@@ -1056,12 +1072,13 @@
 void AudioTrack::releaseBuffer(Buffer* audioBuffer)
 {
     AutoMutex lock(mLock);
-    mCblk->stepUser(audioBuffer->frameCount);
+    audio_track_cblk_t* cblk = mCblk;
+    cblk->stepUserOut(audioBuffer->frameCount, mFrameCount);
     if (audioBuffer->frameCount > 0) {
         // restart track if it was disabled by audioflinger due to previous underrun
-        if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
-            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
-            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
+        if (mActive && (cblk->flags & CBLK_DISABLED)) {
+            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
+            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName);
             mAudioTrack->start();
         }
     }
@@ -1072,8 +1089,12 @@
 ssize_t AudioTrack::write(const void* buffer, size_t userSize)
 {
 
-    if (mSharedBuffer != 0) return INVALID_OPERATION;
-    if (mIsTimed) return INVALID_OPERATION;
+    if (mSharedBuffer != 0) {
+        return INVALID_OPERATION;
+    }
+    if (mIsTimed) {
+        return INVALID_OPERATION;
+    }
 
     if (ssize_t(userSize) < 0) {
         // Sanity-check: user is most-likely passing an error code, and it would
@@ -1096,6 +1117,9 @@
     sp<IMemory> iMem = mCblkMemory;
     mLock.unlock();
 
+    // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
+    // so all cblk references might still refer to old shared memory, but that should be benign
+
     ssize_t written = 0;
     const int8_t *src = (const int8_t *)buffer;
     Buffer audioBuffer;
@@ -1107,8 +1131,9 @@
         status_t err = obtainBuffer(&audioBuffer, -1);
         if (err < 0) {
             // out of buffers, return #bytes written
-            if (err == status_t(NO_MORE_BUFFERS))
+            if (err == status_t(NO_MORE_BUFFERS)) {
                 break;
+            }
             return ssize_t(err);
         }
 
@@ -1121,8 +1146,8 @@
         } else {
             toWrite = audioBuffer.size;
             memcpy(audioBuffer.i8, src, toWrite);
-            src += toWrite;
         }
+        src += toWrite;
         userSize -= toWrite;
         written += toWrite;
 
@@ -1140,27 +1165,37 @@
 
 status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
 {
+    AutoMutex lock(mLock);
     status_t result = UNKNOWN_ERROR;
 
+    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
+    // while we are accessing the cblk
+    sp<IAudioTrack> audioTrack = mAudioTrack;
+    sp<IMemory> iMem = mCblkMemory;
+
     // If the track is not invalid already, try to allocate a buffer.  alloc
     // fails indicating that the server is dead, flag the track as invalid so
     // we can attempt to restore in just a bit.
-    if (!(mCblk->flags & CBLK_INVALID_MSK)) {
+    audio_track_cblk_t* cblk = mCblk;
+    if (!(cblk->flags & CBLK_INVALID)) {
         result = mAudioTrack->allocateTimedBuffer(size, buffer);
         if (result == DEAD_OBJECT) {
-            android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
+            android_atomic_or(CBLK_INVALID, &cblk->flags);
         }
     }
 
     // If the track is invalid at this point, attempt to restore it. and try the
     // allocation one more time.
-    if (mCblk->flags & CBLK_INVALID_MSK) {
-        mCblk->lock.lock();
-        result = restoreTrack_l(mCblk, false);
-        mCblk->lock.unlock();
+    if (cblk->flags & CBLK_INVALID) {
+        cblk->lock.lock();
+        audio_track_cblk_t* temp = cblk;
+        result = restoreTrack_l(temp, false /*fromStart*/);
+        cblk = temp;
+        cblk->lock.unlock();
 
-        if (result == OK)
+        if (result == OK) {
             result = mAudioTrack->allocateTimedBuffer(size, buffer);
+        }
     }
 
     return result;
@@ -1172,10 +1207,11 @@
     status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
     {
         AutoMutex lock(mLock);
+        audio_track_cblk_t* cblk = mCblk;
         // restart track if it was disabled by audioflinger due to previous underrun
         if (buffer->size() != 0 && status == NO_ERROR &&
-                mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
-            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
+                mActive && (cblk->flags & CBLK_DISABLED)) {
+            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
             ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
             mAudioTrack->start();
         }
@@ -1206,15 +1242,20 @@
     bool active = mActive;
     mLock.unlock();
 
+    // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
+    // so all cblk references might still refer to old shared memory, but that should be benign
+
     // Manage underrun callback
-    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
+    if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) {
         ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
-        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
+        if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
             mCbf(EVENT_UNDERRUN, mUserData, 0);
-            if (cblk->server == cblk->frameCount) {
+            if (cblk->server == mFrameCount) {
                 mCbf(EVENT_BUFFER_END, mUserData, 0);
             }
-            if (mSharedBuffer != 0) return false;
+            if (mSharedBuffer != 0) {
+                return false;
+            }
         }
     }
 
@@ -1265,12 +1306,15 @@
         status_t err = obtainBuffer(&audioBuffer, waitCount);
         if (err < NO_ERROR) {
             if (err != TIMED_OUT) {
-                ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
+                ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
+                        "Error obtaining an audio buffer, giving up.");
                 return false;
             }
             break;
         }
-        if (err == status_t(STOPPED)) return false;
+        if (err == status_t(STOPPED)) {
+            return false;
+        }
 
         // Divide buffer size by 2 to take into account the expansion
         // due to 8 to 16 bit conversion: the callback must fill only half
@@ -1293,7 +1337,9 @@
             break;
         }
 
-        if (writtenSize > reqSize) writtenSize = reqSize;
+        if (writtenSize > reqSize) {
+            writtenSize = reqSize;
+        }
 
         if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
             // 8 to 16 bit conversion, note that source and destination are the same address
@@ -1302,10 +1348,10 @@
         }
 
         audioBuffer.size = writtenSize;
-        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
-        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
+        // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for
+        // 8 bit PCM data: in this case,  cblk->frameSize is based on a sample size of
         // 16 bit.
-        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
+        audioBuffer.frameCount = writtenSize / mFrameSizeAF;
 
         frames -= audioBuffer.frameCount;
 
@@ -1321,112 +1367,91 @@
     return true;
 }
 
-// must be called with mLock and cblk.lock held. Callers must also hold strong references on
+// must be called with mLock and refCblk.lock held. Callers must also hold strong references on
 // the IAudioTrack and IMemory in case they are recreated here.
-// If the IAudioTrack is successfully restored, the cblk pointer is updated
-status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
+// If the IAudioTrack is successfully restored, the refCblk pointer is updated
+// FIXME Don't depend on caller to hold strong references.
+status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart)
 {
     status_t result;
 
-    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
-        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
-            fromStart ? "start()" : "obtainBuffer()", gettid());
+    audio_track_cblk_t* cblk = refCblk;
+    audio_track_cblk_t* newCblk = cblk;
+    ALOGW("dead IAudioTrack, creating a new one from %s",
+        fromStart ? "start()" : "obtainBuffer()");
 
-        // signal old cblk condition so that other threads waiting for available buffers stop
-        // waiting now
-        cblk->cv.broadcast();
-        cblk->lock.unlock();
+    // signal old cblk condition so that other threads waiting for available buffers stop
+    // waiting now
+    cblk->cv.broadcast();
+    cblk->lock.unlock();
 
-        // refresh the audio configuration cache in this process to make sure we get new
-        // output parameters in getOutput_l() and createTrack_l()
-        AudioSystem::clearAudioConfigCache();
+    // refresh the audio configuration cache in this process to make sure we get new
+    // output parameters in getOutput_l() and createTrack_l()
+    AudioSystem::clearAudioConfigCache();
 
-        // if the new IAudioTrack is created, createTrack_l() will modify the
-        // following member variables: mAudioTrack, mCblkMemory and mCblk.
-        // It will also delete the strong references on previous IAudioTrack and IMemory
-        result = createTrack_l(mStreamType,
-                               cblk->sampleRate,
-                               mFormat,
-                               mChannelMask,
-                               mFrameCount,
-                               mFlags,
-                               mSharedBuffer,
-                               getOutput_l());
+    // if the new IAudioTrack is created, createTrack_l() will modify the
+    // following member variables: mAudioTrack, mCblkMemory and mCblk.
+    // It will also delete the strong references on previous IAudioTrack and IMemory
+    result = createTrack_l(mStreamType,
+                           cblk->sampleRate,
+                           mFormat,
+                           mReqFrameCount,  // so that frame count never goes down
+                           mFlags,
+                           mSharedBuffer,
+                           getOutput_l());
 
-        if (result == NO_ERROR) {
-            uint32_t user = cblk->user;
-            uint32_t server = cblk->server;
-            // restore write index and set other indexes to reflect empty buffer status
-            mCblk->user = user;
-            mCblk->server = user;
-            mCblk->userBase = user;
-            mCblk->serverBase = user;
-            // restore loop: this is not guaranteed to succeed if new frame count is not
-            // compatible with loop length
-            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
-            if (!fromStart) {
-                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
-                // Make sure that a client relying on callback events indicating underrun or
-                // the actual amount of audio frames played (e.g SoundPool) receives them.
-                if (mSharedBuffer == 0) {
-                    uint32_t frames = 0;
-                    if (user > server) {
-                        frames = ((user - server) > mCblk->frameCount) ?
-                                mCblk->frameCount : (user - server);
-                        memset(mCblk->buffers, 0, frames * mCblk->frameSize);
-                    }
-                    // restart playback even if buffer is not completely filled.
-                    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
-                    // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
-                    // the client
-                    mCblk->stepUser(frames);
+    if (result == NO_ERROR) {
+        uint32_t user = cblk->user;
+        uint32_t server = cblk->server;
+        // restore write index and set other indexes to reflect empty buffer status
+        newCblk = mCblk;
+        newCblk->user = user;
+        newCblk->server = user;
+        newCblk->userBase = user;
+        newCblk->serverBase = user;
+        // restore loop: this is not guaranteed to succeed if new frame count is not
+        // compatible with loop length
+        setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
+        if (!fromStart) {
+            newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+            // Make sure that a client relying on callback events indicating underrun or
+            // the actual amount of audio frames played (e.g SoundPool) receives them.
+            if (mSharedBuffer == 0) {
+                uint32_t frames = 0;
+                if (user > server) {
+                    frames = ((user - server) > mFrameCount) ?
+                            mFrameCount : (user - server);
+                    memset(mBuffers, 0, frames * mFrameSizeAF);
                 }
-            }
-            if (mSharedBuffer != 0) {
-                mCblk->stepUser(mCblk->frameCount);
-            }
-            if (mActive) {
-                result = mAudioTrack->start();
-                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
-            }
-            if (fromStart && result == NO_ERROR) {
-                mNewPosition = mCblk->server + mUpdatePeriod;
+                // restart playback even if buffer is not completely filled.
+                android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
+                // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
+                // the client
+                newCblk->stepUserOut(frames, mFrameCount);
             }
         }
-        if (result != NO_ERROR) {
-            android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
-            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
+        if (mSharedBuffer != 0) {
+            newCblk->stepUserOut(mFrameCount, mFrameCount);
         }
-        mRestoreStatus = result;
-        // signal old cblk condition for other threads waiting for restore completion
-        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
-        cblk->cv.broadcast();
-    } else {
-        if (!(cblk->flags & CBLK_RESTORED_MSK)) {
-            ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
-            mLock.unlock();
-            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
-            if (result == NO_ERROR) {
-                result = mRestoreStatus;
-            }
-            cblk->lock.unlock();
-            mLock.lock();
-        } else {
-            ALOGW("dead IAudioTrack, already restored TID %d", gettid());
-            result = mRestoreStatus;
-            cblk->lock.unlock();
+        if (mActive) {
+            result = mAudioTrack->start();
+            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
+        }
+        if (fromStart && result == NO_ERROR) {
+            mNewPosition = newCblk->server + mUpdatePeriod;
         }
     }
+    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
     ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
-        result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
+        result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
 
     if (result == NO_ERROR) {
         // from now on we switch to the newly created cblk
-        cblk = mCblk;
+        refCblk = newCblk;
     }
-    cblk->lock.lock();
+    newCblk->lock.lock();
 
-    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
+    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result);
 
     return result;
 }
@@ -1438,12 +1463,16 @@
     char buffer[SIZE];
     String8 result;
 
+    audio_track_cblk_t* cblk = mCblk;
     result.append(" AudioTrack::dump\n");
-    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
+    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
+            mVolume[0], mVolume[1]);
     result.append(buffer);
-    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, (mCblk == 0) ? 0 : mCblk->frameCount);
+    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
+            mChannelCount, mFrameCount);
     result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
+    snprintf(buffer, 255, "  sample rate(%u), status(%d), muted(%d)\n",
+            (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
     result.append(buffer);
     snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
     result.append(buffer);
@@ -1505,20 +1534,20 @@
 
 audio_track_cblk_t::audio_track_cblk_t()
     : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
-    userBase(0), serverBase(0), buffers(NULL), frameCount(0),
+    userBase(0), serverBase(0), frameCount_(0),
     loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
     mSendLevel(0), flags(0)
 {
 }
 
-uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
+uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut)
 {
-    ALOGV("stepuser %08x %08x %d", user, server, frameCount);
+    ALOGV("stepuser %08x %08x %d", user, server, stepCount);
 
     uint32_t u = user;
-    u += frameCount;
+    u += stepCount;
     // Ensure that user is never ahead of server for AudioRecord
-    if (flags & CBLK_DIRECTION_MSK) {
+    if (isOut) {
         // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
             bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -1528,30 +1557,29 @@
         u = server;
     }
 
-    uint32_t fc = this->frameCount;
-    if (u >= fc) {
+    if (u >= frameCount) {
         // common case, user didn't just wrap
-        if (u - fc >= userBase ) {
-            userBase += fc;
+        if (u - frameCount >= userBase ) {
+            userBase += frameCount;
         }
-    } else if (u >= userBase + fc) {
+    } else if (u >= userBase + frameCount) {
         // user just wrapped
-        userBase += fc;
+        userBase += frameCount;
     }
 
     user = u;
 
     // Clear flow control error condition as new data has been written/read to/from buffer.
-    if (flags & CBLK_UNDERRUN_MSK) {
-        android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
+    if (flags & CBLK_UNDERRUN) {
+        android_atomic_and(~CBLK_UNDERRUN, &flags);
     }
 
     return u;
 }
 
-bool audio_track_cblk_t::stepServer(uint32_t frameCount)
+bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut)
 {
-    ALOGV("stepserver %08x %08x %d", user, server, frameCount);
+    ALOGV("stepserver %08x %08x %d", user, server, stepCount);
 
     if (!tryLock()) {
         ALOGW("stepServer() could not lock cblk");
@@ -1561,8 +1589,8 @@
     uint32_t s = server;
     bool flushed = (s == user);
 
-    s += frameCount;
-    if (flags & CBLK_DIRECTION_MSK) {
+    s += stepCount;
+    if (isOut) {
         // Mark that we have read the first buffer so that next time stepUser() is called
         // we switch to normal obtainBuffer() timeout period
         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
@@ -1587,43 +1615,42 @@
         }
     }
 
-    uint32_t fc = this->frameCount;
-    if (s >= fc) {
+    if (s >= frameCount) {
         // common case, server didn't just wrap
-        if (s - fc >= serverBase ) {
-            serverBase += fc;
+        if (s - frameCount >= serverBase ) {
+            serverBase += frameCount;
         }
-    } else if (s >= serverBase + fc) {
+    } else if (s >= serverBase + frameCount) {
         // server just wrapped
-        serverBase += fc;
+        serverBase += frameCount;
     }
 
     server = s;
 
-    if (!(flags & CBLK_INVALID_MSK)) {
+    if (!(flags & CBLK_INVALID)) {
         cv.signal();
     }
     lock.unlock();
     return true;
 }
 
-void* audio_track_cblk_t::buffer(uint32_t offset) const
+void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const
 {
     return (int8_t *)buffers + (offset - userBase) * frameSize;
 }
 
-uint32_t audio_track_cblk_t::framesAvailable()
+uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut)
 {
     Mutex::Autolock _l(lock);
-    return framesAvailable_l();
+    return framesAvailable_l(frameCount, isOut);
 }
 
-uint32_t audio_track_cblk_t::framesAvailable_l()
+uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut)
 {
     uint32_t u = user;
     uint32_t s = server;
 
-    if (flags & CBLK_DIRECTION_MSK) {
+    if (isOut) {
         uint32_t limit = (s < loopStart) ? s : loopStart;
         return limit + frameCount - u;
     } else {
@@ -1631,12 +1658,12 @@
     }
 }
 
-uint32_t audio_track_cblk_t::framesReady()
+uint32_t audio_track_cblk_t::framesReady(bool isOut)
 {
     uint32_t u = user;
     uint32_t s = server;
 
-    if (flags & CBLK_DIRECTION_MSK) {
+    if (isOut) {
         if (u < loopEnd) {
             return u - s;
         } else {
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index ce8ffc4..a010bb6 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,8 +89,8 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
-                                track_flags_t flags,
+                                size_t frameCount,
+                                track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
@@ -106,7 +106,8 @@
         data.writeInt32(format);
         data.writeInt32(channelMask);
         data.writeInt32(frameCount);
-        data.writeInt32((int32_t) flags);
+        track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
+        data.writeInt32(lFlags);
         data.writeStrongBinder(sharedBuffer->asBinder());
         data.writeInt32((int32_t) output);
         data.writeInt32((int32_t) tid);
@@ -119,6 +120,10 @@
         if (lStatus != NO_ERROR) {
             ALOGE("createTrack error: %s", strerror(-lStatus));
         } else {
+            lFlags = reply.readInt32();
+            if (flags != NULL) {
+                *flags = lFlags;
+            }
             lSessionId = reply.readInt32();
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
@@ -138,7 +143,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -501,7 +506,7 @@
         return reply.readInt32();
     }
 
-    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
             audio_io_handle_t output) const
     {
         Parcel data, reply;
@@ -522,7 +527,7 @@
         return status;
     }
 
-    virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
+    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -690,7 +695,7 @@
         return (audio_module_handle_t) reply.readInt32();
     }
 
-    virtual int32_t getPrimaryOutputSamplingRate()
+    virtual uint32_t getPrimaryOutputSamplingRate()
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -698,7 +703,7 @@
         return reply.readInt32();
     }
 
-    virtual int32_t getPrimaryOutputFrameCount()
+    virtual size_t getPrimaryOutputFrameCount()
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -723,7 +728,7 @@
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
             audio_channel_mask_t channelMask = data.readInt32();
-            size_t bufferCount = data.readInt32();
+            size_t frameCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
@@ -732,7 +737,8 @@
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     (audio_stream_type_t) streamType, sampleRate, format,
-                    channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status);
+                    channelMask, frameCount, &flags, buffer, output, tid, &sessionId, &status);
+            reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
@@ -745,13 +751,13 @@
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
             audio_channel_mask_t channelMask = data.readInt32();
-            size_t bufferCount = data.readInt32();
+            size_t frameCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             pid_t tid = (pid_t) data.readInt32();
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(pid, input,
-                    sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status);
+                    sampleRate, format, channelMask, frameCount, flags, tid, &sessionId, &status);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(record->asBinder());
@@ -865,7 +871,8 @@
 
         case REGISTER_CLIENT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(data.readStrongBinder());
+            sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(
+                    data.readStrongBinder());
             registerClient(client);
             return NO_ERROR;
         } break;
@@ -965,8 +972,8 @@
         case GET_RENDER_POSITION: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            uint32_t halFrames;
-            uint32_t dspFrames;
+            size_t halFrames;
+            size_t dspFrames;
             status_t status = getRenderPosition(&halFrames, &dspFrames, output);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
@@ -1043,7 +1050,8 @@
             int id;
             int enabled;
 
-            sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, &status, &id, &enabled);
+            sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId,
+                    &status, &id, &enabled);
             reply->writeInt32(status);
             reply->writeInt32(id);
             reply->writeInt32(enabled);
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 4178b29..2d1e0f8 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -50,7 +50,8 @@
             ALOGV("ioConfigChanged stream %d", stream);
             data.writeInt32(stream);
         } else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) {
-            const AudioSystem::OutputDescriptor *desc = (const AudioSystem::OutputDescriptor *)param2;
+            const AudioSystem::OutputDescriptor *desc =
+                    (const AudioSystem::OutputDescriptor *)param2;
             data.writeInt32(desc->samplingRate);
             data.writeInt32(desc->format);
             data.writeInt32(desc->channels);
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 401437c..769deae 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -399,13 +399,15 @@
 
         case SET_PHONE_STATE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32())));
+            reply->writeInt32(static_cast <uint32_t>(setPhoneState(
+                    (audio_mode_t) data.readInt32())));
             return NO_ERROR;
         } break;
 
         case SET_FORCE_USE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32());
+            audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(
+                    data.readInt32());
             audio_policy_forced_cfg_t config =
                     static_cast <audio_policy_forced_cfg_t>(data.readInt32());
             reply->writeInt32(static_cast <uint32_t>(setForceUse(usage, config)));
@@ -414,7 +416,8 @@
 
         case GET_FORCE_USE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32());
+            audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(
+                    data.readInt32());
             reply->writeInt32(static_cast <uint32_t>(getForceUse(usage)));
             return NO_ERROR;
         } break;
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index abc8899..ee70ef7 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -489,7 +489,7 @@
         ::close(mFd);
     }
     mData.clear();
-    delete mUrl;
+    free(mUrl);
 }
 
 status_t Sample::doLoad()
@@ -568,8 +568,8 @@
         }
 
         // initialize track
-        int afFrameCount;
-        int afSampleRate;
+        size_t afFrameCount;
+        uint32_t afSampleRate;
         audio_stream_type_t streamType = mSoundPool->streamType();
         if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
             afFrameCount = kDefaultFrameCount;
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 253602d..42584fe 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -1036,7 +1036,7 @@
         goto initAudioTrack_exit;
     }
 
-    mpAudioTrack->setVolume(mVolume, mVolume);
+    mpAudioTrack->setVolume(mVolume);
 
     mState = TONE_INIT;
 
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index 8196e10..5b4071b 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -88,7 +88,8 @@
     return status;
 }
 
-status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, uint32_t rate)
+status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags,
+        uint32_t rate)
 {
     if (rate > CAPTURE_RATE_MAX) {
         return BAD_VALUE;
@@ -334,7 +335,8 @@
 
 //-------------------------------------------------------------------------
 
-Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate, bool bCanCallJava)
+Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate,
+        bool bCanCallJava)
     : Thread(bCanCallJava), mReceiver(receiver)
 {
     mSleepTimeUs = 1000000000 / captureRate;
diff --git a/media/libmedia_native/Android.mk b/media/libmedia_native/Android.mk
deleted file mode 100644
index 065a90f..0000000
--- a/media/libmedia_native/Android.mk
+++ /dev/null
@@ -1,11 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES :=
-
-LOCAL_MODULE:= libmedia_native
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 5b5ed71..48f48e4 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -28,7 +28,6 @@
     libdl                       \
     libgui                      \
     libmedia                    \
-    libmedia_native             \
     libsonivox                  \
     libstagefright              \
     libstagefright_foundation   \
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9bedff1..c3e5c40 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1387,8 +1387,8 @@
     }
     ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask,
             format, bufferCount, mSessionId);
-    int afSampleRate;
-    int afFrameCount;
+    uint32_t afSampleRate;
+    size_t afFrameCount;
     uint32_t frameCount;
 
     if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index ff27873..d3ec122 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -550,8 +550,6 @@
                     }
                 }
             } else if (what == Renderer::kWhatFlushComplete) {
-                CHECK_EQ(what, (int32_t)Renderer::kWhatFlushComplete);
-
                 int32_t audio;
                 CHECK(msg->findInt32("audio", &audio));
 
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index cf455bd..afaa5db 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -57,9 +57,7 @@
 }
 
 NuPlayer::RTSPSource::~RTSPSource() {
-    if (mLooper != NULL) {
-        mLooper->stop();
-    }
+   mLooper->stop();
 }
 
 void NuPlayer::RTSPSource::start() {
@@ -86,6 +84,9 @@
 }
 
 void NuPlayer::RTSPSource::stop() {
+    if (mLooper == NULL) {
+        return;
+    }
     sp<AMessage> msg = new AMessage(kWhatDisconnect, mReflector->id());
 
     sp<AMessage> dummy;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index 00d2017..e0d2c21 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -24,44 +24,55 @@
 
 size_t Format_frameSize(NBAIO_Format format)
 {
-    switch (format) {
-    case Format_SR44_1_C2_I16:
-    case Format_SR48_C2_I16:
-        return 2 * sizeof(short);
-    case Format_SR44_1_C1_I16:
-    case Format_SR48_C1_I16:
-        return 1 * sizeof(short);
-    case Format_Invalid:
-    default:
-        return 0;
-    }
+    return Format_channelCount(format) * sizeof(short);
 }
 
 size_t Format_frameBitShift(NBAIO_Format format)
 {
-    switch (format) {
-    case Format_SR44_1_C2_I16:
-    case Format_SR48_C2_I16:
-        return 2;   // 1 << 2 == 2 * sizeof(short)
-    case Format_SR44_1_C1_I16:
-    case Format_SR48_C1_I16:
-        return 1;   // 1 << 1 == 1 * sizeof(short)
-    case Format_Invalid:
-    default:
-        return 0;
-    }
+    // sizeof(short) == 2, so frame size == 1 << channels
+    return Format_channelCount(format);
 }
 
+enum {
+    Format_SR_8000,
+    Format_SR_11025,
+    Format_SR_16000,
+    Format_SR_22050,
+    Format_SR_24000,
+    Format_SR_32000,
+    Format_SR_44100,
+    Format_SR_48000,
+    Format_SR_Mask = 7
+};
+
+enum {
+    Format_C_1 = 0x08,
+    Format_C_2 = 0x10,
+    Format_C_Mask = 0x18
+};
+
 unsigned Format_sampleRate(NBAIO_Format format)
 {
-    switch (format) {
-    case Format_SR44_1_C1_I16:
-    case Format_SR44_1_C2_I16:
+    if (format == Format_Invalid) {
+        return 0;
+    }
+    switch (format & Format_SR_Mask) {
+    case Format_SR_8000:
+        return 8000;
+    case Format_SR_11025:
+        return 11025;
+    case Format_SR_16000:
+        return 16000;
+    case Format_SR_22050:
+        return 22050;
+    case Format_SR_24000:
+        return 24000;
+    case Format_SR_32000:
+        return 32000;
+    case Format_SR_44100:
         return 44100;
-    case Format_SR48_C1_I16:
-    case Format_SR48_C2_I16:
+    case Format_SR_48000:
         return 48000;
-    case Format_Invalid:
     default:
         return 0;
     }
@@ -69,14 +80,14 @@
 
 unsigned Format_channelCount(NBAIO_Format format)
 {
-    switch (format) {
-    case Format_SR44_1_C1_I16:
-    case Format_SR48_C1_I16:
+    if (format == Format_Invalid) {
+        return 0;
+    }
+    switch (format & Format_C_Mask) {
+    case Format_C_1:
         return 1;
-    case Format_SR44_1_C2_I16:
-    case Format_SR48_C2_I16:
+    case Format_C_2:
         return 2;
-    case Format_Invalid:
     default:
         return 0;
     }
@@ -84,11 +95,46 @@
 
 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
 {
-    if (sampleRate == 44100 && channelCount == 2) return Format_SR44_1_C2_I16;
-    if (sampleRate == 48000 && channelCount == 2) return Format_SR48_C2_I16;
-    if (sampleRate == 44100 && channelCount == 1) return Format_SR44_1_C1_I16;
-    if (sampleRate == 48000 && channelCount == 1) return Format_SR48_C1_I16;
-    return Format_Invalid;
+    NBAIO_Format format;
+    switch (sampleRate) {
+    case 8000:
+        format = Format_SR_8000;
+        break;
+    case 11025:
+        format = Format_SR_11025;
+        break;
+    case 16000:
+        format = Format_SR_16000;
+        break;
+    case 22050:
+        format = Format_SR_22050;
+        break;
+    case 24000:
+        format = Format_SR_24000;
+        break;
+    case 32000:
+        format = Format_SR_32000;
+        break;
+    case 44100:
+        format = Format_SR_44100;
+        break;
+    case 48000:
+        format = Format_SR_48000;
+        break;
+    default:
+        return Format_Invalid;
+    }
+    switch (channelCount) {
+    case 1:
+        format |= Format_C_1;
+        break;
+    case 2:
+        format |= Format_C_2;
+        break;
+    default:
+        return Format_Invalid;
+    }
+    return format;
 }
 
 // This is a default implementation; it is expected that subclasses will optimize this.
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 84b4962..a135222 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -803,6 +803,8 @@
             "audio_decoder.raw", "audio_encoder.raw" },
         { MEDIA_MIMETYPE_AUDIO_FLAC,
             "audio_decoder.flac", "audio_encoder.flac" },
+        { MEDIA_MIMETYPE_AUDIO_MSGSM,
+            "audio_decoder.gsm", "audio_encoder.gsm" },
     };
 
     static const size_t kNumMimeToRole =
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index faa0f31..a056706 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -78,7 +78,6 @@
         libicuuc \
         liblog \
         libmedia \
-        libmedia_native \
         libsonivox \
         libssl \
         libstagefright_omx \
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 861aebe..3cf4d5c 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -58,7 +58,7 @@
     ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
     CHECK(channelCount == 1 || channelCount == 2);
 
-    int minFrameCount;
+    size_t minFrameCount;
     status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
                                            sampleRate,
                                            AUDIO_FORMAT_PCM_16_BIT,
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
old mode 100755
new mode 100644
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
old mode 100755
new mode 100644
diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp
index e7b5903..5d8029c 100644
--- a/media/libstagefright/MediaDefs.cpp
+++ b/media/libstagefright/MediaDefs.cpp
@@ -40,6 +40,7 @@
 const char *MEDIA_MIMETYPE_AUDIO_RAW = "audio/raw";
 const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac";
 const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts";
+const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm";
 
 const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
 const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
old mode 100755
new mode 100644
index 70de174..22aefcc
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -1390,6 +1390,8 @@
             "audio_decoder.raw", "audio_encoder.raw" },
         { MEDIA_MIMETYPE_AUDIO_FLAC,
             "audio_decoder.flac", "audio_encoder.flac" },
+        { MEDIA_MIMETYPE_AUDIO_MSGSM,
+            "audio_decoder.gsm", "audio_encoder.gsm" },
     };
 
     static const size_t kNumMimeToRole =
diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp
old mode 100755
new mode 100644
diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index bccffd8..af8186c 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -42,7 +42,7 @@
         ".mpeg", ".ogg", ".mid", ".smf", ".imy", ".wma", ".aac",
         ".wav", ".amr", ".midi", ".xmf", ".rtttl", ".rtx", ".ota",
         ".mkv", ".mka", ".webm", ".ts", ".fl", ".flac", ".mxmf",
-        ".avi", ".mpeg", ".mpg", ".mpga"
+        ".avi", ".mpeg", ".mpg", ".awb", ".mpga"
     };
     static const size_t kNumValidExtensions =
         sizeof(kValidExtensions) / sizeof(kValidExtensions[0]);
diff --git a/media/libstagefright/ThrottledSource.cpp b/media/libstagefright/ThrottledSource.cpp
index 348a9d3..7496752 100644
--- a/media/libstagefright/ThrottledSource.cpp
+++ b/media/libstagefright/ThrottledSource.cpp
@@ -31,10 +31,6 @@
     CHECK(mBandwidthLimitBytesPerSecond > 0);
 }
 
-status_t ThrottledSource::initCheck() const {
-    return mSource->initCheck();
-}
-
 ssize_t ThrottledSource::readAt(off64_t offset, void *data, size_t size) {
     Mutex::Autolock autoLock(mLock);
 
@@ -62,17 +58,9 @@
     if (whenUs > nowUs) {
         usleep(whenUs - nowUs);
     }
-
     return n;
 }
 
-status_t ThrottledSource::getSize(off64_t *size) {
-    return mSource->getSize(size);
-}
-
-uint32_t ThrottledSource::flags() {
-    return mSource->flags();
-}
 
 }  // namespace android
 
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index a38400b..d32f4fb 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -38,6 +38,7 @@
     WAVE_FORMAT_PCM        = 0x0001,
     WAVE_FORMAT_ALAW       = 0x0006,
     WAVE_FORMAT_MULAW      = 0x0007,
+    WAVE_FORMAT_MSGSM      = 0x0031,
     WAVE_FORMAT_EXTENSIBLE = 0xFFFE
 };
 
@@ -178,6 +179,7 @@
             if (mWaveFormat != WAVE_FORMAT_PCM
                     && mWaveFormat != WAVE_FORMAT_ALAW
                     && mWaveFormat != WAVE_FORMAT_MULAW
+                    && mWaveFormat != WAVE_FORMAT_MSGSM
                     && mWaveFormat != WAVE_FORMAT_EXTENSIBLE) {
                 return ERROR_UNSUPPORTED;
             }
@@ -216,6 +218,10 @@
                     && mBitsPerSample != 24) {
                     return ERROR_UNSUPPORTED;
                 }
+            } else if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+                if (mBitsPerSample != 0) {
+                    return ERROR_UNSUPPORTED;
+                }
             } else {
                 CHECK(mWaveFormat == WAVE_FORMAT_MULAW
                         || mWaveFormat == WAVE_FORMAT_ALAW);
@@ -283,6 +289,10 @@
                         mTrackMeta->setCString(
                                 kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW);
                         break;
+                    case WAVE_FORMAT_MSGSM:
+                        mTrackMeta->setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MSGSM);
+                        break;
                     default:
                         CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW);
                         mTrackMeta->setCString(
@@ -294,11 +304,17 @@
                 mTrackMeta->setInt32(kKeyChannelMask, mChannelMask);
                 mTrackMeta->setInt32(kKeySampleRate, mSampleRate);
 
-                size_t bytesPerSample = mBitsPerSample >> 3;
-
-                int64_t durationUs =
-                    1000000LL * (mDataSize / (mNumChannels * bytesPerSample))
-                        / mSampleRate;
+                int64_t durationUs = 0;
+                if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+                    // 65 bytes decode to 320 8kHz samples
+                    durationUs =
+                        1000000LL * (mDataSize / 65 * 320) / 8000;
+                } else {
+                    size_t bytesPerSample = mBitsPerSample >> 3;
+                    durationUs =
+                        1000000LL * (mDataSize / (mNumChannels * bytesPerSample))
+                            / mSampleRate;
+                }
 
                 mTrackMeta->setInt64(kKeyDuration, durationUs);
 
@@ -388,7 +404,16 @@
     int64_t seekTimeUs;
     ReadOptions::SeekMode mode;
     if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) {
-        int64_t pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3);
+        int64_t pos = 0;
+
+        if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+            // 65 bytes decode to 320 8kHz samples
+            int64_t samplenumber = (seekTimeUs * mSampleRate) / 1000000;
+            int64_t framenumber = samplenumber / 320;
+            pos = framenumber * 65;
+        } else {
+            pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3);
+        }
         if (pos > mSize) {
             pos = mSize;
         }
@@ -412,6 +437,15 @@
         maxBytesToRead = maxBytesAvailable;
     }
 
+    if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+        // Microsoft packs 2 frames into 65 bytes, rather than using separate 33-byte frames,
+        // so read multiples of 65, and use smaller buffers to account for ~10:1 expansion ratio
+        if (maxBytesToRead > 1024) {
+            maxBytesToRead = 1024;
+        }
+        maxBytesToRead = (maxBytesToRead / 65) * 65;
+    }
+
     ssize_t n = mDataSource->readAt(
             mCurrentPos, buffer->data(),
             maxBytesToRead);
@@ -468,12 +502,17 @@
         }
     }
 
-    size_t bytesPerSample = mBitsPerSample >> 3;
+    int64_t timeStampUs = 0;
 
-    buffer->meta_data()->setInt64(
-            kKeyTime,
-            1000000LL * (mCurrentPos - mOffset)
-                / (mNumChannels * bytesPerSample) / mSampleRate);
+    if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+        timeStampUs = 1000000LL * (mCurrentPos - mOffset) * 320 / 65 / mSampleRate;
+    } else {
+        size_t bytesPerSample = mBitsPerSample >> 3;
+        timeStampUs = 1000000LL * (mCurrentPos - mOffset)
+                / (mNumChannels * bytesPerSample) / mSampleRate;
+    }
+
+    buffer->meta_data()->setInt64(kKeyTime, timeStampUs);
 
     buffer->meta_data()->setInt32(kKeyIsSyncFrame, 1);
     mCurrentPos += n;
diff --git a/media/libstagefright/codecs/gsm/Android.mk b/media/libstagefright/codecs/gsm/Android.mk
new file mode 100644
index 0000000..2e43120
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/Android.mk
@@ -0,0 +1,4 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/media/libstagefright/codecs/gsm/dec/Android.mk b/media/libstagefright/codecs/gsm/dec/Android.mk
new file mode 100644
index 0000000..9c0c6ae
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/dec/Android.mk
@@ -0,0 +1,21 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+        SoftGSM.cpp
+
+LOCAL_C_INCLUDES := \
+        frameworks/av/media/libstagefright/include \
+        frameworks/native/include/media/openmax \
+        external/libgsm/inc
+
+LOCAL_SHARED_LIBRARIES := \
+        libstagefright libstagefright_omx libstagefright_foundation libutils
+
+LOCAL_STATIC_LIBRARIES := \
+        libgsm
+
+LOCAL_MODULE := libstagefright_soft_gsmdec
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2 b/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2
diff --git a/media/libstagefright/codecs/gsm/dec/NOTICE b/media/libstagefright/codecs/gsm/dec/NOTICE
new file mode 100644
index 0000000..c5b1efa
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/dec/NOTICE
@@ -0,0 +1,190 @@
+
+   Copyright (c) 2005-2008, 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.
+
+   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.
+
+
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
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+      the terms of any separate license agreement you may have executed
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+
diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
new file mode 100644
index 0000000..00e0c85
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
@@ -0,0 +1,269 @@
+/*
+ * Copyright (C) 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftGSM"
+#include <utils/Log.h>
+
+#include "SoftGSM.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaDefs.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+    params->nSize = sizeof(T);
+    params->nVersion.s.nVersionMajor = 1;
+    params->nVersion.s.nVersionMinor = 0;
+    params->nVersion.s.nRevision = 0;
+    params->nVersion.s.nStep = 0;
+}
+
+SoftGSM::SoftGSM(
+        const char *name,
+        const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData,
+        OMX_COMPONENTTYPE **component)
+    : SimpleSoftOMXComponent(name, callbacks, appData, component),
+      mSignalledError(false) {
+
+    CHECK(!strcmp(name, "OMX.google.gsm.decoder"));
+
+    mGsm = gsm_create();
+    CHECK(mGsm);
+    int msopt = 1;
+    gsm_option(mGsm, GSM_OPT_WAV49, &msopt);
+
+    initPorts();
+}
+
+SoftGSM::~SoftGSM() {
+    gsm_destroy(mGsm);
+}
+
+void SoftGSM::initPorts() {
+    OMX_PARAM_PORTDEFINITIONTYPE def;
+    InitOMXParams(&def);
+
+    def.nPortIndex = 0;
+    def.eDir = OMX_DirInput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = sizeof(gsm_frame);
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 1;
+
+    def.format.audio.cMIMEType =
+        const_cast<char *>(MEDIA_MIMETYPE_AUDIO_MSGSM);
+
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingGSMFR;
+
+    addPort(def);
+
+    def.nPortIndex = 1;
+    def.eDir = OMX_DirOutput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = kMaxNumSamplesPerFrame * sizeof(int16_t);
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 2;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+    addPort(def);
+}
+
+OMX_ERRORTYPE SoftGSM::internalGetParameter(
+        OMX_INDEXTYPE index, OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex > 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eEndian = OMX_EndianBig;
+            pcmParams->bInterleaved = OMX_TRUE;
+            pcmParams->nBitPerSample = 16;
+            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+            pcmParams->nChannels = 1;
+            pcmParams->nSamplingRate = 8000;
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalGetParameter(index, params);
+    }
+}
+
+OMX_ERRORTYPE SoftGSM::internalSetParameter(
+        OMX_INDEXTYPE index, const OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 0 && pcmParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            if (pcmParams->nChannels != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            if (pcmParams->nSamplingRate != 8000) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamStandardComponentRole:
+        {
+            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+                (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+            if (strncmp((const char *)roleParams->cRole,
+                        "audio_decoder.gsm",
+                        OMX_MAX_STRINGNAME_SIZE - 1)) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalSetParameter(index, params);
+    }
+}
+
+void SoftGSM::onQueueFilled(OMX_U32 portIndex) {
+    if (mSignalledError) {
+        return;
+    }
+
+    List<BufferInfo *> &inQueue = getPortQueue(0);
+    List<BufferInfo *> &outQueue = getPortQueue(1);
+
+    while (!inQueue.empty() && !outQueue.empty()) {
+        BufferInfo *inInfo = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+            inQueue.erase(inQueue.begin());
+            inInfo->mOwnedByUs = false;
+            notifyEmptyBufferDone(inHeader);
+
+            outHeader->nFilledLen = 0;
+            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+
+            outQueue.erase(outQueue.begin());
+            outInfo->mOwnedByUs = false;
+            notifyFillBufferDone(outHeader);
+            return;
+        }
+
+        if (inHeader->nFilledLen > kMaxNumSamplesPerFrame) {
+            ALOGE("input buffer too large (%ld).", inHeader->nFilledLen);
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            mSignalledError = true;
+        }
+
+        if(((inHeader->nFilledLen / 65) * 65) != inHeader->nFilledLen) {
+            ALOGE("input buffer not multiple of 65 (%ld).", inHeader->nFilledLen);
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            mSignalledError = true;
+        }
+
+        uint8_t *inputptr = inHeader->pBuffer + inHeader->nOffset;
+
+        int n = mSignalledError ? 0 : DecodeGSM(mGsm,
+                  reinterpret_cast<int16_t *>(outHeader->pBuffer), inputptr, inHeader->nFilledLen);
+
+        outHeader->nTimeStamp = inHeader->nTimeStamp;
+        outHeader->nOffset = 0;
+        outHeader->nFilledLen = n * sizeof(int16_t);
+        outHeader->nFlags = 0;
+
+        inInfo->mOwnedByUs = false;
+        inQueue.erase(inQueue.begin());
+        inInfo = NULL;
+        notifyEmptyBufferDone(inHeader);
+        inHeader = NULL;
+
+        outInfo->mOwnedByUs = false;
+        outQueue.erase(outQueue.begin());
+        outInfo = NULL;
+        notifyFillBufferDone(outHeader);
+        outHeader = NULL;
+    }
+}
+
+
+// static
+int SoftGSM::DecodeGSM(gsm handle,
+        int16_t *out, uint8_t *in, size_t inSize) {
+
+    int ret = 0;
+    while (inSize > 0) {
+        gsm_decode(handle, in, out);
+        in += 33;
+        inSize -= 33;
+        out += 160;
+        ret += 160;
+        gsm_decode(handle, in, out);
+        in += 32;
+        inSize -= 32;
+        out += 160;
+        ret += 160;
+    }
+    return ret;
+}
+
+
+}  // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+        const char *name, const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+    return new android::SoftGSM(name, callbacks, appData, component);
+}
+
diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.h b/media/libstagefright/codecs/gsm/dec/SoftGSM.h
new file mode 100644
index 0000000..8ab6116
--- /dev/null
+++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 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 SOFT_GSM_H_
+
+#define SOFT_GSM_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+extern "C" {
+#include "gsm.h"
+}
+
+namespace android {
+
+struct SoftGSM : public SimpleSoftOMXComponent {
+    SoftGSM(const char *name,
+            const OMX_CALLBACKTYPE *callbacks,
+            OMX_PTR appData,
+            OMX_COMPONENTTYPE **component);
+
+protected:
+    virtual ~SoftGSM();
+
+    virtual OMX_ERRORTYPE internalGetParameter(
+            OMX_INDEXTYPE index, OMX_PTR params);
+
+    virtual OMX_ERRORTYPE internalSetParameter(
+            OMX_INDEXTYPE index, const OMX_PTR params);
+
+    virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+    enum {
+        kNumBuffers = 4,
+        kMaxNumSamplesPerFrame = 16384,
+    };
+
+    bool mSignalledError;
+    gsm mGsm;
+
+    void initPorts();
+
+    static int DecodeGSM(gsm handle, int16_t *out, uint8_t *in, size_t inSize);
+
+    DISALLOW_EVIL_CONSTRUCTORS(SoftGSM);
+};
+
+}  // namespace android
+
+#endif  // SOFT_GSM_H_
+
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index bf9ab3a..a400b4c 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -66,7 +66,7 @@
     def.eDir = OMX_DirInput;
     def.nBufferCountMin = kNumBuffers;
     def.nBufferCountActual = def.nBufferCountMin;
-    def.nBufferSize = 256 * 1024;
+    def.nBufferSize = 768 * 1024;
     def.bEnabled = OMX_TRUE;
     def.bPopulated = OMX_FALSE;
     def.eDomain = OMX_PortDomainVideo;
diff --git a/media/libstagefright/include/FragmentedMP4Parser.h b/media/libstagefright/include/FragmentedMP4Parser.h
index 0edafb9..dbe02b8 100644
--- a/media/libstagefright/include/FragmentedMP4Parser.h
+++ b/media/libstagefright/include/FragmentedMP4Parser.h
@@ -263,7 +263,7 @@
 
     void copyBuffer(
             sp<ABuffer> *dst,
-            size_t offset, uint64_t size, size_t extra = 0) const;
+            size_t offset, uint64_t size) const;
 
     DISALLOW_EVIL_CONSTRUCTORS(FragmentedMP4Parser);
 };
diff --git a/media/libstagefright/include/ThrottledSource.h b/media/libstagefright/include/ThrottledSource.h
index 7fe7c06..673268b 100644
--- a/media/libstagefright/include/ThrottledSource.h
+++ b/media/libstagefright/include/ThrottledSource.h
@@ -28,18 +28,44 @@
             const sp<DataSource> &source,
             int32_t bandwidthLimitBytesPerSecond);
 
-    virtual status_t initCheck() const;
-
+    // implementation of readAt() that sleeps to achieve the desired max throughput
     virtual ssize_t readAt(off64_t offset, void *data, size_t size);
 
-    virtual status_t getSize(off64_t *size);
-    virtual uint32_t flags();
+    // returns an empty string to prevent callers from using the Uri to construct a new datasource
+    virtual String8 getUri() {
+        return String8();
+    }
+
+    // following methods all call through to the wrapped DataSource's methods
+
+    status_t initCheck() const {
+        return mSource->initCheck();
+    }
+
+    virtual status_t getSize(off64_t *size) {
+        return mSource->getSize(size);
+    }
+
+    virtual uint32_t flags() {
+        return mSource->flags();
+    }
+
+    virtual status_t reconnectAtOffset(off64_t offset) {
+        return mSource->reconnectAtOffset(offset);
+    }
+
+    virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL) {
+        return mSource->DrmInitialization(mime);
+    }
+
+    virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) {
+        mSource->getDrmInfo(handle, client);
+    };
 
     virtual String8 getMIMEType() const {
         return mSource->getMIMEType();
     }
 
-
 private:
     Mutex mLock;
 
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index 8f7d12b..7fc7037 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -758,31 +758,69 @@
     esds = NULL;
 }
 
-void addVorbisCodecInfo(
+status_t addVorbisCodecInfo(
         const sp<MetaData> &meta,
         const void *_codecPrivate, size_t codecPrivateSize) {
-    // printf("vorbis private data follows:\n");
     // hexdump(_codecPrivate, codecPrivateSize);
 
-    CHECK(codecPrivateSize >= 3);
+    if (codecPrivateSize < 1) {
+        return ERROR_MALFORMED;
+    }
 
     const uint8_t *codecPrivate = (const uint8_t *)_codecPrivate;
-    CHECK(codecPrivate[0] == 0x02);
 
-    size_t len1 = codecPrivate[1];
-    size_t len2 = codecPrivate[2];
+    if (codecPrivate[0] != 0x02) {
+        return ERROR_MALFORMED;
+    }
 
-    CHECK(codecPrivateSize > 3 + len1 + len2);
+    // codecInfo starts with two lengths, len1 and len2, that are
+    // "Xiph-style-lacing encoded"...
 
-    CHECK(codecPrivate[3] == 0x01);
-    meta->setData(kKeyVorbisInfo, 0, &codecPrivate[3], len1);
+    size_t offset = 1;
+    size_t len1 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        len1 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    len1 += codecPrivate[offset++];
 
-    CHECK(codecPrivate[len1 + 3] == 0x03);
+    size_t len2 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        len2 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    len2 += codecPrivate[offset++];
 
-    CHECK(codecPrivate[len1 + len2 + 3] == 0x05);
+    if (codecPrivateSize < offset + len1 + len2) {
+        return ERROR_MALFORMED;
+    }
+
+    if (codecPrivate[offset] != 0x01) {
+        return ERROR_MALFORMED;
+    }
+    meta->setData(kKeyVorbisInfo, 0, &codecPrivate[offset], len1);
+
+    offset += len1;
+    if (codecPrivate[offset] != 0x03) {
+        return ERROR_MALFORMED;
+    }
+
+    offset += len2;
+    if (codecPrivate[offset] != 0x05) {
+        return ERROR_MALFORMED;
+    }
+
     meta->setData(
-            kKeyVorbisBooks, 0, &codecPrivate[len1 + len2 + 3],
-            codecPrivateSize - len1 - len2 - 3);
+            kKeyVorbisBooks, 0, &codecPrivate[offset],
+            codecPrivateSize - offset);
+
+    return OK;
 }
 
 void MatroskaExtractor::addTracks() {
@@ -809,6 +847,8 @@
 
         sp<MetaData> meta = new MetaData;
 
+        status_t err = OK;
+
         switch (track->GetType()) {
             case VIDEO_TRACK:
             {
@@ -855,7 +895,8 @@
                 } else if (!strcmp("A_VORBIS", codecID)) {
                     meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
 
-                    addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize);
+                    err = addVorbisCodecInfo(
+                            meta, codecPrivate, codecPrivateSize);
                 } else if (!strcmp("A_MPEG/L3", codecID)) {
                     meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
                 } else {
@@ -872,6 +913,11 @@
                 continue;
         }
 
+        if (err != OK) {
+            ALOGE("skipping track, codec specific data was malformed.");
+            continue;
+        }
+
         long long durationNs = mSegment->GetDuration();
         meta->setInt64(kKeyDuration, (durationNs + 500) / 1000);
 
diff --git a/media/libstagefright/mp4/FragmentedMP4Parser.cpp b/media/libstagefright/mp4/FragmentedMP4Parser.cpp
index 7fe4e63..54c3d63 100644
--- a/media/libstagefright/mp4/FragmentedMP4Parser.cpp
+++ b/media/libstagefright/mp4/FragmentedMP4Parser.cpp
@@ -1971,8 +1971,8 @@
 }
 
 void FragmentedMP4Parser::copyBuffer(
-        sp<ABuffer> *dst, size_t offset, uint64_t size, size_t extra) const {
-    sp<ABuffer> buf = new ABuffer(size + extra);
+        sp<ABuffer> *dst, size_t offset, uint64_t size) const {
+    sp<ABuffer> buf = new ABuffer(size);
     memcpy(buf->data(), mBuffer->data() + offset, size);
 
     *dst = buf;
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index 3747b3b..6e1c04d 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -53,6 +53,7 @@
     { "OMX.google.vpx.decoder", "vpxdec", "video_decoder.vpx" },
     { "OMX.google.raw.decoder", "rawdec", "audio_decoder.raw" },
     { "OMX.google.flac.encoder", "flacenc", "audio_encoder.flac" },
+    { "OMX.google.gsm.decoder", "gsmdec", "audio_decoder.gsm" },
 };
 
 static const size_t kNumComponents =
diff --git a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
index bc35aef..b913124 100644
--- a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
+++ b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
@@ -271,6 +271,7 @@
     if (mFirstFailedAttemptUs + 50000ll > ALooper::GetNowUs()) {
         // We're willing to wait a little while to get the right packet.
 
+#if 0
         if (!mRequestedRetransmission) {
             ALOGI("requesting retransmission of seqNo %d",
                   (mLastDequeuedExtSeqNo + 1) & 0xffff);
@@ -280,7 +281,9 @@
             notify->post();
 
             mRequestedRetransmission = true;
-        } else {
+        } else
+#endif
+        {
             ALOGI("still waiting for the correct packet to arrive.");
         }
 
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 2899953..c4050b8 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -15,6 +15,9 @@
 
 LOCAL_SRC_FILES:=               \
     AudioFlinger.cpp            \
+    Threads.cpp                 \
+    Tracks.cpp                  \
+    Effects.cpp                 \
     AudioMixer.cpp.arm          \
     AudioResampler.cpp.arm      \
     AudioPolicyService.cpp      \
@@ -31,7 +34,6 @@
     $(call include-path-for, audio-effects) \
     $(call include-path-for, audio-utils)
 
-# FIXME keep libmedia_native but remove libmedia after split
 LOCAL_SHARED_LIBRARIES := \
     libaudioutils \
     libcommon_time_client \
@@ -39,7 +41,6 @@
     libutils \
     libbinder \
     libmedia \
-    libmedia_native \
     libnbaio \
     libhardware \
     libhardware_legacy \
@@ -72,6 +73,10 @@
 # 47.5 seconds at 44.1 kHz, 8 megabytes
 # LOCAL_CFLAGS += -DTEE_SINK_FRAMES=0x200000
 
+# uncomment for dumpsys to write most recent audio input to .wav file
+# 47.5 seconds at 44.1 kHz, 8 megabytes
+# LOCAL_CFLAGS += -DTEE_SINK_INPUT_FRAMES=0x200000
+
 # uncomment to enable the audio watchdog
 # LOCAL_SRC_FILES += AudioWatchdog.cpp
 # LOCAL_CFLAGS += -DAUDIO_WATCHDOG
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1913b6f..514fcb1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -29,7 +29,6 @@
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include <binder/Parcel.h>
-#include <binder/IPCThreadState.h>
 #include <utils/String16.h>
 #include <utils/threads.h>
 #include <utils/Atomic.h>
@@ -38,15 +37,8 @@
 #include <cutils/properties.h>
 #include <cutils/compiler.h>
 
-#undef ADD_BATTERY_DATA
-
-#ifdef ADD_BATTERY_DATA
-#include <media/IMediaPlayerService.h>
-#include <media/IMediaDeathNotifier.h>
-#endif
-
-#include <private/media/AudioTrackShared.h>
-#include <private/media/AudioEffectShared.h>
+//#include <private/media/AudioTrackShared.h>
+//#include <private/media/AudioEffectShared.h>
 
 #include <system/audio.h>
 #include <hardware/audio.h>
@@ -64,26 +56,8 @@
 
 #include <powermanager/PowerManager.h>
 
-// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
-#ifdef DEBUG_CPU_USAGE
-#include <cpustats/CentralTendencyStatistics.h>
-#include <cpustats/ThreadCpuUsage.h>
-#endif
-
 #include <common_time/cc_helper.h>
-#include <common_time/local_clock.h>
-
-#include "FastMixer.h"
-
-// NBAIO implementations
-#include <media/nbaio/AudioStreamOutSink.h>
-#include <media/nbaio/MonoPipe.h>
-#include <media/nbaio/MonoPipeReader.h>
-#include <media/nbaio/Pipe.h>
-#include <media/nbaio/PipeReader.h>
-#include <media/nbaio/SourceAudioBufferProvider.h>
-
-#include "SchedulingPolicyService.h"
+//#include <common_time/local_clock.h>
 
 // ----------------------------------------------------------------------------
 
@@ -105,90 +79,13 @@
 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
 static const char kHardwareLockedString[] = "Hardware lock is taken\n";
 
-static const float MAX_GAIN = 4096.0f;
-static const uint32_t MAX_GAIN_INT = 0x1000;
-
-// retry counts for buffer fill timeout
-// 50 * ~20msecs = 1 second
-static const int8_t kMaxTrackRetries = 50;
-static const int8_t kMaxTrackStartupRetries = 50;
-// allow less retry attempts on direct output thread.
-// direct outputs can be a scarce resource in audio hardware and should
-// be released as quickly as possible.
-static const int8_t kMaxTrackRetriesDirect = 2;
-
-static const int kDumpLockRetries = 50;
-static const int kDumpLockSleepUs = 20000;
-
-// don't warn about blocked writes or record buffer overflows more often than this
-static const nsecs_t kWarningThrottleNs = seconds(5);
-
-// RecordThread loop sleep time upon application overrun or audio HAL read error
-static const int kRecordThreadSleepUs = 5000;
-
-// maximum time to wait for setParameters to complete
-static const nsecs_t kSetParametersTimeoutNs = seconds(2);
-
-// minimum sleep time for the mixer thread loop when tracks are active but in underrun
-static const uint32_t kMinThreadSleepTimeUs = 5000;
-// maximum divider applied to the active sleep time in the mixer thread loop
-static const uint32_t kMaxThreadSleepTimeShift = 2;
-
-// minimum normal mix buffer size, expressed in milliseconds rather than frames
-static const uint32_t kMinNormalMixBufferSizeMs = 20;
-// maximum normal mix buffer size
-static const uint32_t kMaxNormalMixBufferSizeMs = 24;
 
 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
 
-// Whether to use fast mixer
-static const enum {
-    FastMixer_Never,    // never initialize or use: for debugging only
-    FastMixer_Always,   // always initialize and use, even if not needed: for debugging only
-                        // normal mixer multiplier is 1
-    FastMixer_Static,   // initialize if needed, then use all the time if initialized,
-                        // multiplier is calculated based on min & max normal mixer buffer size
-    FastMixer_Dynamic,  // initialize if needed, then use dynamically depending on track load,
-                        // multiplier is calculated based on min & max normal mixer buffer size
-    // FIXME for FastMixer_Dynamic:
-    //  Supporting this option will require fixing HALs that can't handle large writes.
-    //  For example, one HAL implementation returns an error from a large write,
-    //  and another HAL implementation corrupts memory, possibly in the sample rate converter.
-    //  We could either fix the HAL implementations, or provide a wrapper that breaks
-    //  up large writes into smaller ones, and the wrapper would need to deal with scheduler.
-} kUseFastMixer = FastMixer_Static;
-
-static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off"
-                              // AudioFlinger::setParameters() updates, other threads read w/o lock
-
-// Priorities for requestPriority
-static const int kPriorityAudioApp = 2;
-static const int kPriorityFastMixer = 3;
-
-// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
-// for the track.  The client then sub-divides this into smaller buffers for its use.
-// Currently the client uses double-buffering by default, but doesn't tell us about that.
-// So for now we just assume that client is double-buffered.
-// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
-// N-buffering, so AudioFlinger could allocate the right amount of memory.
-// See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 2;
+uint32_t AudioFlinger::mScreenState;
 
 // ----------------------------------------------------------------------------
 
-#ifdef ADD_BATTERY_DATA
-// To collect the amplifier usage
-static void addBatteryData(uint32_t params) {
-    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
-    if (service == NULL) {
-        // it already logged
-        return;
-    }
-
-    service->addBatteryData(params);
-}
-#endif
-
 static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
 {
     const hw_module_t *mod;
@@ -364,7 +261,7 @@
     write(fd, result.string(), result.size());
 }
 
-static bool tryLock(Mutex& mutex)
+bool AudioFlinger::dumpTryLock(Mutex& mutex)
 {
     bool locked = false;
     for (int i = 0; i < kDumpLockRetries; ++i) {
@@ -383,7 +280,7 @@
         dumpPermissionDenial(fd, args);
     } else {
         // get state of hardware lock
-        bool hardwareLocked = tryLock(mHardwareLock);
+        bool hardwareLocked = dumpTryLock(mHardwareLock);
         if (!hardwareLocked) {
             String8 result(kHardwareLockedString);
             write(fd, result.string(), result.size());
@@ -391,7 +288,7 @@
             mHardwareLock.unlock();
         }
 
-        bool locked = tryLock(mLock);
+        bool locked = dumpTryLock(mLock);
 
         // failed to lock - AudioFlinger is probably deadlocked
         if (!locked) {
@@ -417,7 +314,15 @@
             audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
             dev->dump(dev, fd);
         }
-        if (locked) mLock.unlock();
+
+        // dump the serially shared record tee sink
+        if (mRecordTeeSource != 0) {
+            dumpTee(fd, mRecordTeeSource);
+        }
+
+        if (locked) {
+            mLock.unlock();
+        }
     }
     return NO_ERROR;
 }
@@ -444,8 +349,8 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
-        IAudioFlinger::track_flags_t flags,
+        size_t frameCount,
+        IAudioFlinger::track_flags_t *flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output,
         pid_t tid,
@@ -466,6 +371,14 @@
         goto Exit;
     }
 
+    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
+    // and we don't yet support 8.24 or 32-bit PCM
+    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
+        ALOGE("createTrack() invalid format %d", format);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
     {
         Mutex::Autolock _l(mLock);
         PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -856,8 +769,9 @@
 
 status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
 {
-    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
-            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
+    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
+            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
+
     // check calling permissions
     if (!settingsAllowed()) {
         return PERMISSION_DENIED;
@@ -906,8 +820,8 @@
         String8 screenState;
         if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
             bool isOff = screenState == "off";
-            if (isOff != (gScreenState & 1)) {
-                gScreenState = ((gScreenState & ~1) + 2) | isOff;
+            if (isOff != (AudioFlinger::mScreenState & 1)) {
+                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
             }
         }
         return final_result;
@@ -941,8 +855,8 @@
 
 String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
 {
-//    ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
-//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
+            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
 
     Mutex::Autolock _l(mLock);
 
@@ -1028,7 +942,7 @@
     return ret;
 }
 
-status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
         audio_io_handle_t output) const
 {
     status_t status;
@@ -1112,7 +1026,8 @@
 // removeClient_l() must be called with AudioFlinger::mLock held
 void AudioFlinger::removeClient_l(pid_t pid)
 {
-    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
+    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
+            IPCThreadState::self()->getCallingPid());
     mClients.removeItem(pid);
 }
 
@@ -1131,4596 +1046,7 @@
     return thread;
 }
 
-// ----------------------------------------------------------------------------
-
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
-        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
-    :   Thread(false /*canCallJava*/),
-        mType(type),
-        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
-        // mChannelMask
-        mChannelCount(0),
-        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
-        mParamStatus(NO_ERROR),
-        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
-        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
-        // mName will be set by concrete (non-virtual) subclass
-        mDeathRecipient(new PMDeathRecipient(this))
-{
-}
-
-AudioFlinger::ThreadBase::~ThreadBase()
-{
-    mParamCond.broadcast();
-    // do not lock the mutex in destructor
-    releaseWakeLock_l();
-    if (mPowerManager != 0) {
-        sp<IBinder> binder = mPowerManager->asBinder();
-        binder->unlinkToDeath(mDeathRecipient);
-    }
-}
-
-void AudioFlinger::ThreadBase::exit()
-{
-    ALOGV("ThreadBase::exit");
-    // do any cleanup required for exit to succeed
-    preExit();
-    {
-        // This lock prevents the following race in thread (uniprocessor for illustration):
-        //  if (!exitPending()) {
-        //      // context switch from here to exit()
-        //      // exit() calls requestExit(), what exitPending() observes
-        //      // exit() calls signal(), which is dropped since no waiters
-        //      // context switch back from exit() to here
-        //      mWaitWorkCV.wait(...);
-        //      // now thread is hung
-        //  }
-        AutoMutex lock(mLock);
-        requestExit();
-        mWaitWorkCV.broadcast();
-    }
-    // When Thread::requestExitAndWait is made virtual and this method is renamed to
-    // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
-    requestExitAndWait();
-}
-
-status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
-{
-    status_t status;
-
-    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
-    Mutex::Autolock _l(mLock);
-
-    mNewParameters.add(keyValuePairs);
-    mWaitWorkCV.signal();
-    // wait condition with timeout in case the thread loop has exited
-    // before the request could be processed
-    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
-        status = mParamStatus;
-        mWaitWorkCV.signal();
-    } else {
-        status = TIMED_OUT;
-    }
-    return status;
-}
-
-void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
-{
-    Mutex::Autolock _l(mLock);
-    sendIoConfigEvent_l(event, param);
-}
-
-// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
-{
-    IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
-    mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
-    ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
-    mWaitWorkCV.signal();
-}
-
-// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
-{
-    PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio);
-    mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent));
-    ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d",
-          mConfigEvents.size(), pid, tid, prio);
-    mWaitWorkCV.signal();
-}
-
-void AudioFlinger::ThreadBase::processConfigEvents()
-{
-    mLock.lock();
-    while (!mConfigEvents.isEmpty()) {
-        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
-        ConfigEvent *event = mConfigEvents[0];
-        mConfigEvents.removeAt(0);
-        // release mLock before locking AudioFlinger mLock: lock order is always
-        // AudioFlinger then ThreadBase to avoid cross deadlock
-        mLock.unlock();
-        switch(event->type()) {
-            case CFG_EVENT_PRIO: {
-                PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
-                int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
-                if (err != 0) {
-                    ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
-                          prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
-                }
-            } break;
-            case CFG_EVENT_IO: {
-                IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
-                mAudioFlinger->mLock.lock();
-                audioConfigChanged_l(ioEvent->event(), ioEvent->param());
-                mAudioFlinger->mLock.unlock();
-            } break;
-            default:
-                ALOGE("processConfigEvents() unknown event type %d", event->type());
-                break;
-        }
-        delete event;
-        mLock.lock();
-    }
-    mLock.unlock();
-}
-
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    bool locked = tryLock(mLock);
-    if (!locked) {
-        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
-        write(fd, buffer, strlen(buffer));
-    }
-
-    snprintf(buffer, SIZE, "io handle: %d\n", mId);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "TID: %d\n", getTid());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
-    result.append(buffer);
-
-    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
-    result.append(buffer);
-    result.append(" Index Command");
-    for (size_t i = 0; i < mNewParameters.size(); ++i) {
-        snprintf(buffer, SIZE, "\n %02d    ", i);
-        result.append(buffer);
-        result.append(mNewParameters[i]);
-    }
-
-    snprintf(buffer, SIZE, "\n\nPending config events: \n");
-    result.append(buffer);
-    for (size_t i = 0; i < mConfigEvents.size(); i++) {
-        mConfigEvents[i]->dump(buffer, SIZE);
-        result.append(buffer);
-    }
-    result.append("\n");
-
-    write(fd, result.string(), result.size());
-
-    if (locked) {
-        mLock.unlock();
-    }
-}
-
-void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
-    write(fd, buffer, strlen(buffer));
-
-    for (size_t i = 0; i < mEffectChains.size(); ++i) {
-        sp<EffectChain> chain = mEffectChains[i];
-        if (chain != 0) {
-            chain->dump(fd, args);
-        }
-    }
-}
-
-void AudioFlinger::ThreadBase::acquireWakeLock()
-{
-    Mutex::Autolock _l(mLock);
-    acquireWakeLock_l();
-}
-
-void AudioFlinger::ThreadBase::acquireWakeLock_l()
-{
-    if (mPowerManager == 0) {
-        // use checkService() to avoid blocking if power service is not up yet
-        sp<IBinder> binder =
-            defaultServiceManager()->checkService(String16("power"));
-        if (binder == 0) {
-            ALOGW("Thread %s cannot connect to the power manager service", mName);
-        } else {
-            mPowerManager = interface_cast<IPowerManager>(binder);
-            binder->linkToDeath(mDeathRecipient);
-        }
-    }
-    if (mPowerManager != 0) {
-        sp<IBinder> binder = new BBinder();
-        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
-                                                         binder,
-                                                         String16(mName));
-        if (status == NO_ERROR) {
-            mWakeLockToken = binder;
-        }
-        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
-    }
-}
-
-void AudioFlinger::ThreadBase::releaseWakeLock()
-{
-    Mutex::Autolock _l(mLock);
-    releaseWakeLock_l();
-}
-
-void AudioFlinger::ThreadBase::releaseWakeLock_l()
-{
-    if (mWakeLockToken != 0) {
-        ALOGV("releaseWakeLock_l() %s", mName);
-        if (mPowerManager != 0) {
-            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
-        }
-        mWakeLockToken.clear();
-    }
-}
-
-void AudioFlinger::ThreadBase::clearPowerManager()
-{
-    Mutex::Autolock _l(mLock);
-    releaseWakeLock_l();
-    mPowerManager.clear();
-}
-
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
-{
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        thread->clearPowerManager();
-    }
-    ALOGW("power manager service died !!!");
-}
-
-void AudioFlinger::ThreadBase::setEffectSuspended(
-        const effect_uuid_t *type, bool suspend, int sessionId)
-{
-    Mutex::Autolock _l(mLock);
-    setEffectSuspended_l(type, suspend, sessionId);
-}
-
-void AudioFlinger::ThreadBase::setEffectSuspended_l(
-        const effect_uuid_t *type, bool suspend, int sessionId)
-{
-    sp<EffectChain> chain = getEffectChain_l(sessionId);
-    if (chain != 0) {
-        if (type != NULL) {
-            chain->setEffectSuspended_l(type, suspend);
-        } else {
-            chain->setEffectSuspendedAll_l(suspend);
-        }
-    }
-
-    updateSuspendedSessions_l(type, suspend, sessionId);
-}
-
-void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
-{
-    ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
-    if (index < 0) {
-        return;
-    }
-
-    const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
-            mSuspendedSessions.valueAt(index);
-
-    for (size_t i = 0; i < sessionEffects.size(); i++) {
-        sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
-        for (int j = 0; j < desc->mRefCount; j++) {
-            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
-                chain->setEffectSuspendedAll_l(true);
-            } else {
-                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
-                    desc->mType.timeLow);
-                chain->setEffectSuspended_l(&desc->mType, true);
-            }
-        }
-    }
-}
-
-void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
-                                                         bool suspend,
-                                                         int sessionId)
-{
-    ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
-
-    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
-
-    if (suspend) {
-        if (index >= 0) {
-            sessionEffects = mSuspendedSessions.valueAt(index);
-        } else {
-            mSuspendedSessions.add(sessionId, sessionEffects);
-        }
-    } else {
-        if (index < 0) {
-            return;
-        }
-        sessionEffects = mSuspendedSessions.valueAt(index);
-    }
-
-
-    int key = EffectChain::kKeyForSuspendAll;
-    if (type != NULL) {
-        key = type->timeLow;
-    }
-    index = sessionEffects.indexOfKey(key);
-
-    sp<SuspendedSessionDesc> desc;
-    if (suspend) {
-        if (index >= 0) {
-            desc = sessionEffects.valueAt(index);
-        } else {
-            desc = new SuspendedSessionDesc();
-            if (type != NULL) {
-                desc->mType = *type;
-            }
-            sessionEffects.add(key, desc);
-            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
-        }
-        desc->mRefCount++;
-    } else {
-        if (index < 0) {
-            return;
-        }
-        desc = sessionEffects.valueAt(index);
-        if (--desc->mRefCount == 0) {
-            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
-            sessionEffects.removeItemsAt(index);
-            if (sessionEffects.isEmpty()) {
-                ALOGV("updateSuspendedSessions_l() restore removing session %d",
-                                 sessionId);
-                mSuspendedSessions.removeItem(sessionId);
-            }
-        }
-    }
-    if (!sessionEffects.isEmpty()) {
-        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
-    }
-}
-
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
-                                                            bool enabled,
-                                                            int sessionId)
-{
-    Mutex::Autolock _l(mLock);
-    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
-}
-
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
-                                                            bool enabled,
-                                                            int sessionId)
-{
-    if (mType != RECORD) {
-        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
-        // another session. This gives the priority to well behaved effect control panels
-        // and applications not using global effects.
-        // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
-        // global effects
-        if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
-            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
-        }
-    }
-
-    sp<EffectChain> chain = getEffectChain_l(sessionId);
-    if (chain != 0) {
-        chain->checkSuspendOnEffectEnabled(effect, enabled);
-    }
-}
-
-// ----------------------------------------------------------------------------
-
-AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
-                                             AudioStreamOut* output,
-                                             audio_io_handle_t id,
-                                             audio_devices_t device,
-                                             type_t type)
-    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
-        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
-        // mStreamTypes[] initialized in constructor body
-        mOutput(output),
-        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
-        mMixerStatus(MIXER_IDLE),
-        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
-        standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
-        mScreenState(gScreenState),
-        // index 0 is reserved for normal mixer's submix
-        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
-{
-    snprintf(mName, kNameLength, "AudioOut_%X", id);
-
-    // Assumes constructor is called by AudioFlinger with it's mLock held, but
-    // it would be safer to explicitly pass initial masterVolume/masterMute as
-    // parameter.
-    //
-    // If the HAL we are using has support for master volume or master mute,
-    // then do not attenuate or mute during mixing (just leave the volume at 1.0
-    // and the mute set to false).
-    mMasterVolume = audioFlinger->masterVolume_l();
-    mMasterMute = audioFlinger->masterMute_l();
-    if (mOutput && mOutput->audioHwDev) {
-        if (mOutput->audioHwDev->canSetMasterVolume()) {
-            mMasterVolume = 1.0;
-        }
-
-        if (mOutput->audioHwDev->canSetMasterMute()) {
-            mMasterMute = false;
-        }
-    }
-
-    readOutputParameters();
-
-    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
-    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
-    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
-            stream = (audio_stream_type_t) (stream + 1)) {
-        mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
-        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
-    }
-    // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
-    // because mAudioFlinger doesn't have one to copy from
-}
-
-AudioFlinger::PlaybackThread::~PlaybackThread()
-{
-    delete [] mMixBuffer;
-}
-
-void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
-{
-    dumpInternals(fd, args);
-    dumpTracks(fd, args);
-    dumpEffectChains(fd, args);
-}
-
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
-    for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
-        const stream_type_t *st = &mStreamTypes[i];
-        if (i > 0) {
-            result.appendFormat(", ");
-        }
-        result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
-        if (st->mute) {
-            result.append("M");
-        }
-    }
-    result.append("\n");
-    write(fd, result.string(), result.length());
-    result.clear();
-
-    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-
-    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
-        sp<Track> track = mActiveTracks[i].promote();
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-    write(fd, result.string(), result.size());
-
-    // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
-    FastTrackUnderruns underruns = getFastTrackUnderruns(0);
-    fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
-            underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
-}
-
-void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
-    result.append(buffer);
-    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
-
-    dumpBase(fd, args);
-}
-
-// Thread virtuals
-status_t AudioFlinger::PlaybackThread::readyToRun()
-{
-    status_t status = initCheck();
-    if (status == NO_ERROR) {
-        ALOGI("AudioFlinger's thread %p ready to run", this);
-    } else {
-        ALOGE("No working audio driver found.");
-    }
-    return status;
-}
-
-void AudioFlinger::PlaybackThread::onFirstRef()
-{
-    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
-}
-
-// ThreadBase virtuals
-void AudioFlinger::PlaybackThread::preExit()
-{
-    ALOGV("  preExit()");
-    // FIXME this is using hard-coded strings but in the future, this functionality will be
-    //       converted to use audio HAL extensions required to support tunneling
-    mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
-}
-
-// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
-        const sp<AudioFlinger::Client>& client,
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        int frameCount,
-        const sp<IMemory>& sharedBuffer,
-        int sessionId,
-        IAudioFlinger::track_flags_t flags,
-        pid_t tid,
-        status_t *status)
-{
-    sp<Track> track;
-    status_t lStatus;
-
-    bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0;
-
-    // client expresses a preference for FAST, but we get the final say
-    if (flags & IAudioFlinger::TRACK_FAST) {
-      if (
-            // not timed
-            (!isTimed) &&
-            // either of these use cases:
-            (
-              // use case 1: shared buffer with any frame count
-              (
-                (sharedBuffer != 0)
-              ) ||
-              // use case 2: callback handler and frame count is default or at least as large as HAL
-              (
-                (tid != -1) &&
-                ((frameCount == 0) ||
-                (frameCount >= (int) (mFrameCount * kFastTrackMultiplier)))
-              )
-            ) &&
-            // PCM data
-            audio_is_linear_pcm(format) &&
-            // mono or stereo
-            ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
-              (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
-#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
-            // hardware sample rate
-            (sampleRate == mSampleRate) &&
-#endif
-            // normal mixer has an associated fast mixer
-            hasFastMixer() &&
-            // there are sufficient fast track slots available
-            (mFastTrackAvailMask != 0)
-            // FIXME test that MixerThread for this fast track has a capable output HAL
-            // FIXME add a permission test also?
-        ) {
-        // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
-        if (frameCount == 0) {
-            frameCount = mFrameCount * kFastTrackMultiplier;
-        }
-        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
-                frameCount, mFrameCount);
-      } else {
-        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
-                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d "
-                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
-                isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
-                audio_is_linear_pcm(format),
-                channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
-        flags &= ~IAudioFlinger::TRACK_FAST;
-        // For compatibility with AudioTrack calculation, buffer depth is forced
-        // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
-        // This is probably too conservative, but legacy application code may depend on it.
-        // If you change this calculation, also review the start threshold which is related.
-        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
-        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
-        if (minBufCount < 2) {
-            minBufCount = 2;
-        }
-        int minFrameCount = mNormalFrameCount * minBufCount;
-        if (frameCount < minFrameCount) {
-            frameCount = minFrameCount;
-        }
-      }
-    }
-
-    if (mType == DIRECT) {
-        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
-            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
-                        "for output %p with format %d",
-                        sampleRate, format, channelMask, mOutput, mFormat);
-                lStatus = BAD_VALUE;
-                goto Exit;
-            }
-        }
-    } else {
-        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-        if (sampleRate > mSampleRate*2) {
-            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
-            lStatus = BAD_VALUE;
-            goto Exit;
-        }
-    }
-
-    lStatus = initCheck();
-    if (lStatus != NO_ERROR) {
-        ALOGE("Audio driver not initialized.");
-        goto Exit;
-    }
-
-    { // scope for mLock
-        Mutex::Autolock _l(mLock);
-
-        // all tracks in same audio session must share the same routing strategy otherwise
-        // conflicts will happen when tracks are moved from one output to another by audio policy
-        // manager
-        uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
-        for (size_t i = 0; i < mTracks.size(); ++i) {
-            sp<Track> t = mTracks[i];
-            if (t != 0 && !t->isOutputTrack()) {
-                uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
-                if (sessionId == t->sessionId() && strategy != actual) {
-                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
-                            strategy, actual);
-                    lStatus = BAD_VALUE;
-                    goto Exit;
-                }
-            }
-        }
-
-        if (!isTimed) {
-            track = new Track(this, client, streamType, sampleRate, format,
-                    channelMask, frameCount, sharedBuffer, sessionId, flags);
-        } else {
-            track = TimedTrack::create(this, client, streamType, sampleRate, format,
-                    channelMask, frameCount, sharedBuffer, sessionId);
-        }
-        if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
-            lStatus = NO_MEMORY;
-            goto Exit;
-        }
-        mTracks.add(track);
-
-        sp<EffectChain> chain = getEffectChain_l(sessionId);
-        if (chain != 0) {
-            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
-            track->setMainBuffer(chain->inBuffer());
-            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
-            chain->incTrackCnt();
-        }
-
-        if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
-            pid_t callingPid = IPCThreadState::self()->getCallingPid();
-            // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
-            // so ask activity manager to do this on our behalf
-            sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
-        }
-    }
-
-    lStatus = NO_ERROR;
-
-Exit:
-    if (status) {
-        *status = lStatus;
-    }
-    return track;
-}
-
-uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const
-{
-    if (mFastMixer != NULL) {
-        MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
-        latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
-    }
-    return latency;
-}
-
-uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const
-{
-    return latency;
-}
-
-uint32_t AudioFlinger::PlaybackThread::latency() const
-{
-    Mutex::Autolock _l(mLock);
-    return latency_l();
-}
-uint32_t AudioFlinger::PlaybackThread::latency_l() const
-{
-    if (initCheck() == NO_ERROR) {
-        return correctLatency(mOutput->stream->get_latency(mOutput->stream));
-    } else {
-        return 0;
-    }
-}
-
-void AudioFlinger::PlaybackThread::setMasterVolume(float value)
-{
-    Mutex::Autolock _l(mLock);
-    // Don't apply master volume in SW if our HAL can do it for us.
-    if (mOutput && mOutput->audioHwDev &&
-        mOutput->audioHwDev->canSetMasterVolume()) {
-        mMasterVolume = 1.0;
-    } else {
-        mMasterVolume = value;
-    }
-}
-
-void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
-{
-    Mutex::Autolock _l(mLock);
-    // Don't apply master mute in SW if our HAL can do it for us.
-    if (mOutput && mOutput->audioHwDev &&
-        mOutput->audioHwDev->canSetMasterMute()) {
-        mMasterMute = false;
-    } else {
-        mMasterMute = muted;
-    }
-}
-
-void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
-{
-    Mutex::Autolock _l(mLock);
-    mStreamTypes[stream].volume = value;
-}
-
-void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
-{
-    Mutex::Autolock _l(mLock);
-    mStreamTypes[stream].mute = muted;
-}
-
-float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
-{
-    Mutex::Autolock _l(mLock);
-    return mStreamTypes[stream].volume;
-}
-
-// addTrack_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
-{
-    status_t status = ALREADY_EXISTS;
-
-    // set retry count for buffer fill
-    track->mRetryCount = kMaxTrackStartupRetries;
-    if (mActiveTracks.indexOf(track) < 0) {
-        // the track is newly added, make sure it fills up all its
-        // buffers before playing. This is to ensure the client will
-        // effectively get the latency it requested.
-        track->mFillingUpStatus = Track::FS_FILLING;
-        track->mResetDone = false;
-        track->mPresentationCompleteFrames = 0;
-        mActiveTracks.add(track);
-        if (track->mainBuffer() != mMixBuffer) {
-            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
-            if (chain != 0) {
-                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
-                chain->incActiveTrackCnt();
-            }
-        }
-
-        status = NO_ERROR;
-    }
-
-    ALOGV("mWaitWorkCV.broadcast");
-    mWaitWorkCV.broadcast();
-
-    return status;
-}
-
-// destroyTrack_l() must be called with ThreadBase::mLock held
-void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
-{
-    track->mState = TrackBase::TERMINATED;
-    // active tracks are removed by threadLoop()
-    if (mActiveTracks.indexOf(track) < 0) {
-        removeTrack_l(track);
-    }
-}
-
-void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
-{
-    track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
-    mTracks.remove(track);
-    deleteTrackName_l(track->name());
-    // redundant as track is about to be destroyed, for dumpsys only
-    track->mName = -1;
-    if (track->isFastTrack()) {
-        int index = track->mFastIndex;
-        ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
-        ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
-        mFastTrackAvailMask |= 1 << index;
-        // redundant as track is about to be destroyed, for dumpsys only
-        track->mFastIndex = -1;
-    }
-    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
-    if (chain != 0) {
-        chain->decTrackCnt();
-    }
-}
-
-String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
-{
-    String8 out_s8 = String8("");
-    char *s;
-
-    Mutex::Autolock _l(mLock);
-    if (initCheck() != NO_ERROR) {
-        return out_s8;
-    }
-
-    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
-    out_s8 = String8(s);
-    free(s);
-    return out_s8;
-}
-
-// audioConfigChanged_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
-    AudioSystem::OutputDescriptor desc;
-    void *param2 = NULL;
-
-    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
-
-    switch (event) {
-    case AudioSystem::OUTPUT_OPENED:
-    case AudioSystem::OUTPUT_CONFIG_CHANGED:
-        desc.channels = mChannelMask;
-        desc.samplingRate = mSampleRate;
-        desc.format = mFormat;
-        desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t)
-        desc.latency = latency();
-        param2 = &desc;
-        break;
-
-    case AudioSystem::STREAM_CONFIG_CHANGED:
-        param2 = &param;
-    case AudioSystem::OUTPUT_CLOSED:
-    default:
-        break;
-    }
-    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
-}
-
-void AudioFlinger::PlaybackThread::readOutputParameters()
-{
-    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
-    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
-    mChannelCount = (uint16_t)popcount(mChannelMask);
-    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
-    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
-    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
-    if (mFrameCount & 15) {
-        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
-                mFrameCount);
-    }
-
-    // Calculate size of normal mix buffer relative to the HAL output buffer size
-    double multiplier = 1.0;
-    if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) {
-        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
-        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
-        // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
-        minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
-        maxNormalFrameCount = maxNormalFrameCount & ~15;
-        if (maxNormalFrameCount < minNormalFrameCount) {
-            maxNormalFrameCount = minNormalFrameCount;
-        }
-        multiplier = (double) minNormalFrameCount / (double) mFrameCount;
-        if (multiplier <= 1.0) {
-            multiplier = 1.0;
-        } else if (multiplier <= 2.0) {
-            if (2 * mFrameCount <= maxNormalFrameCount) {
-                multiplier = 2.0;
-            } else {
-                multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
-            }
-        } else {
-            // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC
-            // (it would be unusual for the normal mix buffer size to not be a multiple of fast
-            // track, but we sometimes have to do this to satisfy the maximum frame count constraint)
-            // FIXME this rounding up should not be done if no HAL SRC
-            uint32_t truncMult = (uint32_t) multiplier;
-            if ((truncMult & 1)) {
-                if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
-                    ++truncMult;
-                }
-            }
-            multiplier = (double) truncMult;
-        }
-    }
-    mNormalFrameCount = multiplier * mFrameCount;
-    // round up to nearest 16 frames to satisfy AudioMixer
-    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
-    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount);
-
-    delete[] mMixBuffer;
-    mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
-    memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
-
-    // force reconfiguration of effect chains and engines to take new buffer size and audio
-    // parameters into account
-    // Note that mLock is not held when readOutputParameters() is called from the constructor
-    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
-    // matter.
-    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
-    Vector< sp<EffectChain> > effectChains = mEffectChains;
-    for (size_t i = 0; i < effectChains.size(); i ++) {
-        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
-    }
-}
-
-
-status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
-{
-    if (halFrames == NULL || dspFrames == NULL) {
-        return BAD_VALUE;
-    }
-    Mutex::Autolock _l(mLock);
-    if (initCheck() != NO_ERROR) {
-        return INVALID_OPERATION;
-    }
-    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
-
-    if (isSuspended()) {
-        // return an estimation of rendered frames when the output is suspended
-        int32_t frames = mBytesWritten - latency_l();
-        if (frames < 0) {
-            frames = 0;
-        }
-        *dspFrames = (uint32_t)frames;
-        return NO_ERROR;
-    } else {
-        return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
-    }
-}
-
-uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
-{
-    Mutex::Autolock _l(mLock);
-    uint32_t result = 0;
-    if (getEffectChain_l(sessionId) != 0) {
-        result = EFFECT_SESSION;
-    }
-
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (sessionId == track->sessionId() &&
-                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
-            result |= TRACK_SESSION;
-            break;
-        }
-    }
-
-    return result;
-}
-
-uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
-{
-    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
-    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
-    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
-        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
-    }
-    for (size_t i = 0; i < mTracks.size(); i++) {
-        sp<Track> track = mTracks[i];
-        if (sessionId == track->sessionId() &&
-                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
-            return AudioSystem::getStrategyForStream(track->streamType());
-        }
-    }
-    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
-}
-
-
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
-{
-    Mutex::Autolock _l(mLock);
-    return mOutput;
-}
-
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
-{
-    Mutex::Autolock _l(mLock);
-    AudioStreamOut *output = mOutput;
-    mOutput = NULL;
-    // FIXME FastMixer might also have a raw ptr to mOutputSink;
-    //       must push a NULL and wait for ack
-    mOutputSink.clear();
-    mPipeSink.clear();
-    mNormalSink.clear();
-    return output;
-}
-
-// this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::PlaybackThread::stream() const
-{
-    if (mOutput == NULL) {
-        return NULL;
-    }
-    return &mOutput->stream->common;
-}
-
-uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
-{
-    return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
-}
-
-status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
-{
-    if (!isValidSyncEvent(event)) {
-        return BAD_VALUE;
-    }
-
-    Mutex::Autolock _l(mLock);
-
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (event->triggerSession() == track->sessionId()) {
-            (void) track->setSyncEvent(event);
-            return NO_ERROR;
-        }
-    }
-
-    return NAME_NOT_FOUND;
-}
-
-bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
-{
-    return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
-}
-
-void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
-{
-    size_t count = tracksToRemove.size();
-    if (CC_UNLIKELY(count)) {
-        for (size_t i = 0 ; i < count ; i++) {
-            const sp<Track>& track = tracksToRemove.itemAt(i);
-            if ((track->sharedBuffer() != 0) &&
-                    (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
-                AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
-            }
-        }
-    }
-
-}
-
-// ----------------------------------------------------------------------------
-
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-        audio_io_handle_t id, audio_devices_t device, type_t type)
-    :   PlaybackThread(audioFlinger, output, id, device, type),
-        // mAudioMixer below
-        // mFastMixer below
-        mFastMixerFutex(0)
-        // mOutputSink below
-        // mPipeSink below
-        // mNormalSink below
-{
-    ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
-    ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
-            "mFrameCount=%d, mNormalFrameCount=%d",
-            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
-            mNormalFrameCount);
-    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
-
-    // FIXME - Current mixer implementation only supports stereo output
-    if (mChannelCount != FCC_2) {
-        ALOGE("Invalid audio hardware channel count %d", mChannelCount);
-    }
-
-    // create an NBAIO sink for the HAL output stream, and negotiate
-    mOutputSink = new AudioStreamOutSink(output->stream);
-    size_t numCounterOffers = 0;
-    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
-    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
-    ALOG_ASSERT(index == 0);
-
-    // initialize fast mixer depending on configuration
-    bool initFastMixer;
-    switch (kUseFastMixer) {
-    case FastMixer_Never:
-        initFastMixer = false;
-        break;
-    case FastMixer_Always:
-        initFastMixer = true;
-        break;
-    case FastMixer_Static:
-    case FastMixer_Dynamic:
-        initFastMixer = mFrameCount < mNormalFrameCount;
-        break;
-    }
-    if (initFastMixer) {
-
-        // create a MonoPipe to connect our submix to FastMixer
-        NBAIO_Format format = mOutputSink->format();
-        // This pipe depth compensates for scheduling latency of the normal mixer thread.
-        // When it wakes up after a maximum latency, it runs a few cycles quickly before
-        // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
-        MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
-        const NBAIO_Format offers[1] = {format};
-        size_t numCounterOffers = 0;
-        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
-        ALOG_ASSERT(index == 0);
-        monoPipe->setAvgFrames((mScreenState & 1) ?
-                (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
-        mPipeSink = monoPipe;
-
-#ifdef TEE_SINK_FRAMES
-        // create a Pipe to archive a copy of FastMixer's output for dumpsys
-        Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
-        numCounterOffers = 0;
-        index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
-        ALOG_ASSERT(index == 0);
-        mTeeSink = teeSink;
-        PipeReader *teeSource = new PipeReader(*teeSink);
-        numCounterOffers = 0;
-        index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
-        ALOG_ASSERT(index == 0);
-        mTeeSource = teeSource;
-#endif
-
-        // create fast mixer and configure it initially with just one fast track for our submix
-        mFastMixer = new FastMixer();
-        FastMixerStateQueue *sq = mFastMixer->sq();
-#ifdef STATE_QUEUE_DUMP
-        sq->setObserverDump(&mStateQueueObserverDump);
-        sq->setMutatorDump(&mStateQueueMutatorDump);
-#endif
-        FastMixerState *state = sq->begin();
-        FastTrack *fastTrack = &state->mFastTracks[0];
-        // wrap the source side of the MonoPipe to make it an AudioBufferProvider
-        fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
-        fastTrack->mVolumeProvider = NULL;
-        fastTrack->mGeneration++;
-        state->mFastTracksGen++;
-        state->mTrackMask = 1;
-        // fast mixer will use the HAL output sink
-        state->mOutputSink = mOutputSink.get();
-        state->mOutputSinkGen++;
-        state->mFrameCount = mFrameCount;
-        state->mCommand = FastMixerState::COLD_IDLE;
-        // already done in constructor initialization list
-        //mFastMixerFutex = 0;
-        state->mColdFutexAddr = &mFastMixerFutex;
-        state->mColdGen++;
-        state->mDumpState = &mFastMixerDumpState;
-        state->mTeeSink = mTeeSink.get();
-        sq->end();
-        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
-
-        // start the fast mixer
-        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
-        pid_t tid = mFastMixer->getTid();
-        int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
-        if (err != 0) {
-            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
-                    kPriorityFastMixer, getpid_cached, tid, err);
-        }
-
-#ifdef AUDIO_WATCHDOG
-        // create and start the watchdog
-        mAudioWatchdog = new AudioWatchdog();
-        mAudioWatchdog->setDump(&mAudioWatchdogDump);
-        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
-        tid = mAudioWatchdog->getTid();
-        err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
-        if (err != 0) {
-            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
-                    kPriorityFastMixer, getpid_cached, tid, err);
-        }
-#endif
-
-    } else {
-        mFastMixer = NULL;
-    }
-
-    switch (kUseFastMixer) {
-    case FastMixer_Never:
-    case FastMixer_Dynamic:
-        mNormalSink = mOutputSink;
-        break;
-    case FastMixer_Always:
-        mNormalSink = mPipeSink;
-        break;
-    case FastMixer_Static:
-        mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
-        break;
-    }
-}
-
-AudioFlinger::MixerThread::~MixerThread()
-{
-    if (mFastMixer != NULL) {
-        FastMixerStateQueue *sq = mFastMixer->sq();
-        FastMixerState *state = sq->begin();
-        if (state->mCommand == FastMixerState::COLD_IDLE) {
-            int32_t old = android_atomic_inc(&mFastMixerFutex);
-            if (old == -1) {
-                __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
-            }
-        }
-        state->mCommand = FastMixerState::EXIT;
-        sq->end();
-        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
-        mFastMixer->join();
-        // Though the fast mixer thread has exited, it's state queue is still valid.
-        // We'll use that extract the final state which contains one remaining fast track
-        // corresponding to our sub-mix.
-        state = sq->begin();
-        ALOG_ASSERT(state->mTrackMask == 1);
-        FastTrack *fastTrack = &state->mFastTracks[0];
-        ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
-        delete fastTrack->mBufferProvider;
-        sq->end(false /*didModify*/);
-        delete mFastMixer;
-#ifdef AUDIO_WATCHDOG
-        if (mAudioWatchdog != 0) {
-            mAudioWatchdog->requestExit();
-            mAudioWatchdog->requestExitAndWait();
-            mAudioWatchdog.clear();
-        }
-#endif
-    }
-    delete mAudioMixer;
-}
-
-class CpuStats {
-public:
-    CpuStats();
-    void sample(const String8 &title);
-#ifdef DEBUG_CPU_USAGE
-private:
-    ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
-    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
-
-    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
-
-    int mCpuNum;                        // thread's current CPU number
-    int mCpukHz;                        // frequency of thread's current CPU in kHz
-#endif
-};
-
-CpuStats::CpuStats()
-#ifdef DEBUG_CPU_USAGE
-    : mCpuNum(-1), mCpukHz(-1)
-#endif
-{
-}
-
-void CpuStats::sample(const String8 &title) {
-#ifdef DEBUG_CPU_USAGE
-    // get current thread's delta CPU time in wall clock ns
-    double wcNs;
-    bool valid = mCpuUsage.sampleAndEnable(wcNs);
-
-    // record sample for wall clock statistics
-    if (valid) {
-        mWcStats.sample(wcNs);
-    }
-
-    // get the current CPU number
-    int cpuNum = sched_getcpu();
-
-    // get the current CPU frequency in kHz
-    int cpukHz = mCpuUsage.getCpukHz(cpuNum);
-
-    // check if either CPU number or frequency changed
-    if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
-        mCpuNum = cpuNum;
-        mCpukHz = cpukHz;
-        // ignore sample for purposes of cycles
-        valid = false;
-    }
-
-    // if no change in CPU number or frequency, then record sample for cycle statistics
-    if (valid && mCpukHz > 0) {
-        double cycles = wcNs * cpukHz * 0.000001;
-        mHzStats.sample(cycles);
-    }
-
-    unsigned n = mWcStats.n();
-    // mCpuUsage.elapsed() is expensive, so don't call it every loop
-    if ((n & 127) == 1) {
-        long long elapsed = mCpuUsage.elapsed();
-        if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
-            double perLoop = elapsed / (double) n;
-            double perLoop100 = perLoop * 0.01;
-            double perLoop1k = perLoop * 0.001;
-            double mean = mWcStats.mean();
-            double stddev = mWcStats.stddev();
-            double minimum = mWcStats.minimum();
-            double maximum = mWcStats.maximum();
-            double meanCycles = mHzStats.mean();
-            double stddevCycles = mHzStats.stddev();
-            double minCycles = mHzStats.minimum();
-            double maxCycles = mHzStats.maximum();
-            mCpuUsage.resetElapsed();
-            mWcStats.reset();
-            mHzStats.reset();
-            ALOGD("CPU usage for %s over past %.1f secs\n"
-                "  (%u mixer loops at %.1f mean ms per loop):\n"
-                "  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
-                "  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
-                "  MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
-                    title.string(),
-                    elapsed * .000000001, n, perLoop * .000001,
-                    mean * .001,
-                    stddev * .001,
-                    minimum * .001,
-                    maximum * .001,
-                    mean / perLoop100,
-                    stddev / perLoop100,
-                    minimum / perLoop100,
-                    maximum / perLoop100,
-                    meanCycles / perLoop1k,
-                    stddevCycles / perLoop1k,
-                    minCycles / perLoop1k,
-                    maxCycles / perLoop1k);
-
-        }
-    }
-#endif
-};
-
-void AudioFlinger::PlaybackThread::checkSilentMode_l()
-{
-    if (!mMasterMute) {
-        char value[PROPERTY_VALUE_MAX];
-        if (property_get("ro.audio.silent", value, "0") > 0) {
-            char *endptr;
-            unsigned long ul = strtoul(value, &endptr, 0);
-            if (*endptr == '\0' && ul != 0) {
-                ALOGD("Silence is golden");
-                // The setprop command will not allow a property to be changed after
-                // the first time it is set, so we don't have to worry about un-muting.
-                setMasterMute_l(true);
-            }
-        }
-    }
-}
-
-bool AudioFlinger::PlaybackThread::threadLoop()
-{
-    Vector< sp<Track> > tracksToRemove;
-
-    standbyTime = systemTime();
-
-    // MIXER
-    nsecs_t lastWarning = 0;
-
-    // DUPLICATING
-    // FIXME could this be made local to while loop?
-    writeFrames = 0;
-
-    cacheParameters_l();
-    sleepTime = idleSleepTime;
-
-    if (mType == MIXER) {
-        sleepTimeShift = 0;
-    }
-
-    CpuStats cpuStats;
-    const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
-
-    acquireWakeLock();
-
-    while (!exitPending())
-    {
-        cpuStats.sample(myName);
-
-        Vector< sp<EffectChain> > effectChains;
-
-        processConfigEvents();
-
-        { // scope for mLock
-
-            Mutex::Autolock _l(mLock);
-
-            if (checkForNewParameters_l()) {
-                cacheParameters_l();
-            }
-
-            saveOutputTracks();
-
-            // put audio hardware into standby after short delay
-            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
-                        isSuspended())) {
-                if (!mStandby) {
-
-                    threadLoop_standby();
-
-                    mStandby = true;
-                }
-
-                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
-                    // we're about to wait, flush the binder command buffer
-                    IPCThreadState::self()->flushCommands();
-
-                    clearOutputTracks();
-
-                    if (exitPending()) break;
-
-                    releaseWakeLock_l();
-                    // wait until we have something to do...
-                    ALOGV("%s going to sleep", myName.string());
-                    mWaitWorkCV.wait(mLock);
-                    ALOGV("%s waking up", myName.string());
-                    acquireWakeLock_l();
-
-                    mMixerStatus = MIXER_IDLE;
-                    mMixerStatusIgnoringFastTracks = MIXER_IDLE;
-                    mBytesWritten = 0;
-
-                    checkSilentMode_l();
-
-                    standbyTime = systemTime() + standbyDelay;
-                    sleepTime = idleSleepTime;
-                    if (mType == MIXER) {
-                        sleepTimeShift = 0;
-                    }
-
-                    continue;
-                }
-            }
-
-            // mMixerStatusIgnoringFastTracks is also updated internally
-            mMixerStatus = prepareTracks_l(&tracksToRemove);
-
-            // prevent any changes in effect chain list and in each effect chain
-            // during mixing and effect process as the audio buffers could be deleted
-            // or modified if an effect is created or deleted
-            lockEffectChains_l(effectChains);
-        }
-
-        if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
-            threadLoop_mix();
-        } else {
-            threadLoop_sleepTime();
-        }
-
-        if (isSuspended()) {
-            sleepTime = suspendSleepTimeUs();
-            mBytesWritten += mixBufferSize;
-        }
-
-        // only process effects if we're going to write
-        if (sleepTime == 0) {
-            for (size_t i = 0; i < effectChains.size(); i ++) {
-                effectChains[i]->process_l();
-            }
-        }
-
-        // enable changes in effect chain
-        unlockEffectChains(effectChains);
-
-        // sleepTime == 0 means we must write to audio hardware
-        if (sleepTime == 0) {
-
-            threadLoop_write();
-
-if (mType == MIXER) {
-            // write blocked detection
-            nsecs_t now = systemTime();
-            nsecs_t delta = now - mLastWriteTime;
-            if (!mStandby && delta > maxPeriod) {
-                mNumDelayedWrites++;
-                if ((now - lastWarning) > kWarningThrottleNs) {
-#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
-                    ScopedTrace st(ATRACE_TAG, "underrun");
-#endif
-                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
-                            ns2ms(delta), mNumDelayedWrites, this);
-                    lastWarning = now;
-                }
-            }
-}
-
-            mStandby = false;
-        } else {
-            usleep(sleepTime);
-        }
-
-        // Finally let go of removed track(s), without the lock held
-        // since we can't guarantee the destructors won't acquire that
-        // same lock.  This will also mutate and push a new fast mixer state.
-        threadLoop_removeTracks(tracksToRemove);
-        tracksToRemove.clear();
-
-        // FIXME I don't understand the need for this here;
-        //       it was in the original code but maybe the
-        //       assignment in saveOutputTracks() makes this unnecessary?
-        clearOutputTracks();
-
-        // Effect chains will be actually deleted here if they were removed from
-        // mEffectChains list during mixing or effects processing
-        effectChains.clear();
-
-        // FIXME Note that the above .clear() is no longer necessary since effectChains
-        // is now local to this block, but will keep it for now (at least until merge done).
-    }
-
-    // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
-    if (mType == MIXER || mType == DIRECT) {
-        // put output stream into standby mode
-        if (!mStandby) {
-            mOutput->stream->common.standby(&mOutput->stream->common);
-        }
-    }
-
-    releaseWakeLock();
-
-    ALOGV("Thread %p type %d exiting", this, mType);
-    return false;
-}
-
-void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
-{
-    PlaybackThread::threadLoop_removeTracks(tracksToRemove);
-}
-
-void AudioFlinger::MixerThread::threadLoop_write()
-{
-    // FIXME we should only do one push per cycle; confirm this is true
-    // Start the fast mixer if it's not already running
-    if (mFastMixer != NULL) {
-        FastMixerStateQueue *sq = mFastMixer->sq();
-        FastMixerState *state = sq->begin();
-        if (state->mCommand != FastMixerState::MIX_WRITE &&
-                (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
-            if (state->mCommand == FastMixerState::COLD_IDLE) {
-                int32_t old = android_atomic_inc(&mFastMixerFutex);
-                if (old == -1) {
-                    __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
-                }
-#ifdef AUDIO_WATCHDOG
-                if (mAudioWatchdog != 0) {
-                    mAudioWatchdog->resume();
-                }
-#endif
-            }
-            state->mCommand = FastMixerState::MIX_WRITE;
-            sq->end();
-            sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
-            if (kUseFastMixer == FastMixer_Dynamic) {
-                mNormalSink = mPipeSink;
-            }
-        } else {
-            sq->end(false /*didModify*/);
-        }
-    }
-    PlaybackThread::threadLoop_write();
-}
-
-// shared by MIXER and DIRECT, overridden by DUPLICATING
-void AudioFlinger::PlaybackThread::threadLoop_write()
-{
-    // FIXME rewrite to reduce number of system calls
-    mLastWriteTime = systemTime();
-    mInWrite = true;
-    int bytesWritten;
-
-    // If an NBAIO sink is present, use it to write the normal mixer's submix
-    if (mNormalSink != 0) {
-#define mBitShift 2 // FIXME
-        size_t count = mixBufferSize >> mBitShift;
-#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
-        Tracer::traceBegin(ATRACE_TAG, "write");
-#endif
-        // update the setpoint when gScreenState changes
-        uint32_t screenState = gScreenState;
-        if (screenState != mScreenState) {
-            mScreenState = screenState;
-            MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
-            if (pipe != NULL) {
-                pipe->setAvgFrames((mScreenState & 1) ?
-                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
-            }
-        }
-        ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
-#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
-        Tracer::traceEnd(ATRACE_TAG);
-#endif
-        if (framesWritten > 0) {
-            bytesWritten = framesWritten << mBitShift;
-        } else {
-            bytesWritten = framesWritten;
-        }
-    // otherwise use the HAL / AudioStreamOut directly
-    } else {
-        // Direct output thread.
-        bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
-    }
-
-    if (bytesWritten > 0) mBytesWritten += mixBufferSize;
-    mNumWrites++;
-    mInWrite = false;
-}
-
-void AudioFlinger::MixerThread::threadLoop_standby()
-{
-    // Idle the fast mixer if it's currently running
-    if (mFastMixer != NULL) {
-        FastMixerStateQueue *sq = mFastMixer->sq();
-        FastMixerState *state = sq->begin();
-        if (!(state->mCommand & FastMixerState::IDLE)) {
-            state->mCommand = FastMixerState::COLD_IDLE;
-            state->mColdFutexAddr = &mFastMixerFutex;
-            state->mColdGen++;
-            mFastMixerFutex = 0;
-            sq->end();
-            // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
-            sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
-            if (kUseFastMixer == FastMixer_Dynamic) {
-                mNormalSink = mOutputSink;
-            }
-#ifdef AUDIO_WATCHDOG
-            if (mAudioWatchdog != 0) {
-                mAudioWatchdog->pause();
-            }
-#endif
-        } else {
-            sq->end(false /*didModify*/);
-        }
-    }
-    PlaybackThread::threadLoop_standby();
-}
-
-// shared by MIXER and DIRECT, overridden by DUPLICATING
-void AudioFlinger::PlaybackThread::threadLoop_standby()
-{
-    ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
-    mOutput->stream->common.standby(&mOutput->stream->common);
-}
-
-void AudioFlinger::MixerThread::threadLoop_mix()
-{
-    // obtain the presentation timestamp of the next output buffer
-    int64_t pts;
-    status_t status = INVALID_OPERATION;
-
-    if (mNormalSink != 0) {
-        status = mNormalSink->getNextWriteTimestamp(&pts);
-    } else {
-        status = mOutputSink->getNextWriteTimestamp(&pts);
-    }
-
-    if (status != NO_ERROR) {
-        pts = AudioBufferProvider::kInvalidPTS;
-    }
-
-    // mix buffers...
-    mAudioMixer->process(pts);
-    // increase sleep time progressively when application underrun condition clears.
-    // Only increase sleep time if the mixer is ready for two consecutive times to avoid
-    // that a steady state of alternating ready/not ready conditions keeps the sleep time
-    // such that we would underrun the audio HAL.
-    if ((sleepTime == 0) && (sleepTimeShift > 0)) {
-        sleepTimeShift--;
-    }
-    sleepTime = 0;
-    standbyTime = systemTime() + standbyDelay;
-    //TODO: delay standby when effects have a tail
-}
-
-void AudioFlinger::MixerThread::threadLoop_sleepTime()
-{
-    // If no tracks are ready, sleep once for the duration of an output
-    // buffer size, then write 0s to the output
-    if (sleepTime == 0) {
-        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
-            sleepTime = activeSleepTime >> sleepTimeShift;
-            if (sleepTime < kMinThreadSleepTimeUs) {
-                sleepTime = kMinThreadSleepTimeUs;
-            }
-            // reduce sleep time in case of consecutive application underruns to avoid
-            // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
-            // duration we would end up writing less data than needed by the audio HAL if
-            // the condition persists.
-            if (sleepTimeShift < kMaxThreadSleepTimeShift) {
-                sleepTimeShift++;
-            }
-        } else {
-            sleepTime = idleSleepTime;
-        }
-    } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
-        memset (mMixBuffer, 0, mixBufferSize);
-        sleepTime = 0;
-        ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start");
-    }
-    // TODO add standby time extension fct of effect tail
-}
-
-// prepareTracks_l() must be called with ThreadBase::mLock held
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
-        Vector< sp<Track> > *tracksToRemove)
-{
-
-    mixer_state mixerStatus = MIXER_IDLE;
-    // find out which tracks need to be processed
-    size_t count = mActiveTracks.size();
-    size_t mixedTracks = 0;
-    size_t tracksWithEffect = 0;
-    // counts only _active_ fast tracks
-    size_t fastTracks = 0;
-    uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
-
-    float masterVolume = mMasterVolume;
-    bool masterMute = mMasterMute;
-
-    if (masterMute) {
-        masterVolume = 0;
-    }
-    // Delegate master volume control to effect in output mix effect chain if needed
-    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
-    if (chain != 0) {
-        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
-        chain->setVolume_l(&v, &v);
-        masterVolume = (float)((v + (1 << 23)) >> 24);
-        chain.clear();
-    }
-
-    // prepare a new state to push
-    FastMixerStateQueue *sq = NULL;
-    FastMixerState *state = NULL;
-    bool didModify = false;
-    FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
-    if (mFastMixer != NULL) {
-        sq = mFastMixer->sq();
-        state = sq->begin();
-    }
-
-    for (size_t i=0 ; i<count ; i++) {
-        sp<Track> t = mActiveTracks[i].promote();
-        if (t == 0) continue;
-
-        // this const just means the local variable doesn't change
-        Track* const track = t.get();
-
-        // process fast tracks
-        if (track->isFastTrack()) {
-
-            // It's theoretically possible (though unlikely) for a fast track to be created
-            // and then removed within the same normal mix cycle.  This is not a problem, as
-            // the track never becomes active so it's fast mixer slot is never touched.
-            // The converse, of removing an (active) track and then creating a new track
-            // at the identical fast mixer slot within the same normal mix cycle,
-            // is impossible because the slot isn't marked available until the end of each cycle.
-            int j = track->mFastIndex;
-            ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
-            ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
-            FastTrack *fastTrack = &state->mFastTracks[j];
-
-            // Determine whether the track is currently in underrun condition,
-            // and whether it had a recent underrun.
-            FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
-            FastTrackUnderruns underruns = ftDump->mUnderruns;
-            uint32_t recentFull = (underruns.mBitFields.mFull -
-                    track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
-            uint32_t recentPartial = (underruns.mBitFields.mPartial -
-                    track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
-            uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
-                    track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
-            uint32_t recentUnderruns = recentPartial + recentEmpty;
-            track->mObservedUnderruns = underruns;
-            // don't count underruns that occur while stopping or pausing
-            // or stopped which can occur when flush() is called while active
-            if (!(track->isStopping() || track->isPausing() || track->isStopped())) {
-                track->mUnderrunCount += recentUnderruns;
-            }
-
-            // This is similar to the state machine for normal tracks,
-            // with a few modifications for fast tracks.
-            bool isActive = true;
-            switch (track->mState) {
-            case TrackBase::STOPPING_1:
-                // track stays active in STOPPING_1 state until first underrun
-                if (recentUnderruns > 0) {
-                    track->mState = TrackBase::STOPPING_2;
-                }
-                break;
-            case TrackBase::PAUSING:
-                // ramp down is not yet implemented
-                track->setPaused();
-                break;
-            case TrackBase::RESUMING:
-                // ramp up is not yet implemented
-                track->mState = TrackBase::ACTIVE;
-                break;
-            case TrackBase::ACTIVE:
-                if (recentFull > 0 || recentPartial > 0) {
-                    // track has provided at least some frames recently: reset retry count
-                    track->mRetryCount = kMaxTrackRetries;
-                }
-                if (recentUnderruns == 0) {
-                    // no recent underruns: stay active
-                    break;
-                }
-                // there has recently been an underrun of some kind
-                if (track->sharedBuffer() == 0) {
-                    // were any of the recent underruns "empty" (no frames available)?
-                    if (recentEmpty == 0) {
-                        // no, then ignore the partial underruns as they are allowed indefinitely
-                        break;
-                    }
-                    // there has recently been an "empty" underrun: decrement the retry counter
-                    if (--(track->mRetryCount) > 0) {
-                        break;
-                    }
-                    // indicate to client process that the track was disabled because of underrun;
-                    // it will then automatically call start() when data is available
-                    android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags);
-                    // remove from active list, but state remains ACTIVE [confusing but true]
-                    isActive = false;
-                    break;
-                }
-                // fall through
-            case TrackBase::STOPPING_2:
-            case TrackBase::PAUSED:
-            case TrackBase::TERMINATED:
-            case TrackBase::STOPPED:
-            case TrackBase::FLUSHED:   // flush() while active
-                // Check for presentation complete if track is inactive
-                // We have consumed all the buffers of this track.
-                // This would be incomplete if we auto-paused on underrun
-                {
-                    size_t audioHALFrames =
-                            (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
-                    size_t framesWritten =
-                            mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
-                    if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
-                        // track stays in active list until presentation is complete
-                        break;
-                    }
-                }
-                if (track->isStopping_2()) {
-                    track->mState = TrackBase::STOPPED;
-                }
-                if (track->isStopped()) {
-                    // Can't reset directly, as fast mixer is still polling this track
-                    //   track->reset();
-                    // So instead mark this track as needing to be reset after push with ack
-                    resetMask |= 1 << i;
-                }
-                isActive = false;
-                break;
-            case TrackBase::IDLE:
-            default:
-                LOG_FATAL("unexpected track state %d", track->mState);
-            }
-
-            if (isActive) {
-                // was it previously inactive?
-                if (!(state->mTrackMask & (1 << j))) {
-                    ExtendedAudioBufferProvider *eabp = track;
-                    VolumeProvider *vp = track;
-                    fastTrack->mBufferProvider = eabp;
-                    fastTrack->mVolumeProvider = vp;
-                    fastTrack->mSampleRate = track->mSampleRate;
-                    fastTrack->mChannelMask = track->mChannelMask;
-                    fastTrack->mGeneration++;
-                    state->mTrackMask |= 1 << j;
-                    didModify = true;
-                    // no acknowledgement required for newly active tracks
-                }
-                // cache the combined master volume and stream type volume for fast mixer; this
-                // lacks any synchronization or barrier so VolumeProvider may read a stale value
-                track->mCachedVolume = track->isMuted() ?
-                        0 : masterVolume * mStreamTypes[track->streamType()].volume;
-                ++fastTracks;
-            } else {
-                // was it previously active?
-                if (state->mTrackMask & (1 << j)) {
-                    fastTrack->mBufferProvider = NULL;
-                    fastTrack->mGeneration++;
-                    state->mTrackMask &= ~(1 << j);
-                    didModify = true;
-                    // If any fast tracks were removed, we must wait for acknowledgement
-                    // because we're about to decrement the last sp<> on those tracks.
-                    block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
-                } else {
-                    LOG_FATAL("fast track %d should have been active", j);
-                }
-                tracksToRemove->add(track);
-                // Avoids a misleading display in dumpsys
-                track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
-            }
-            continue;
-        }
-
-        {   // local variable scope to avoid goto warning
-
-        audio_track_cblk_t* cblk = track->cblk();
-
-        // The first time a track is added we wait
-        // for all its buffers to be filled before processing it
-        int name = track->name();
-        // make sure that we have enough frames to mix one full buffer.
-        // enforce this condition only once to enable draining the buffer in case the client
-        // app does not call stop() and relies on underrun to stop:
-        // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
-        // during last round
-        uint32_t minFrames = 1;
-        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
-                (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
-            if (t->sampleRate() == (int)mSampleRate) {
-                minFrames = mNormalFrameCount;
-            } else {
-                // +1 for rounding and +1 for additional sample needed for interpolation
-                minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
-                // add frames already consumed but not yet released by the resampler
-                // because cblk->framesReady() will include these frames
-                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
-                // the minimum track buffer size is normally twice the number of frames necessary
-                // to fill one buffer and the resampler should not leave more than one buffer worth
-                // of unreleased frames after each pass, but just in case...
-                ALOG_ASSERT(minFrames <= cblk->frameCount);
-            }
-        }
-        if ((track->framesReady() >= minFrames) && track->isReady() &&
-                !track->isPaused() && !track->isTerminated())
-        {
-            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
-
-            mixedTracks++;
-
-            // track->mainBuffer() != mMixBuffer means there is an effect chain
-            // connected to the track
-            chain.clear();
-            if (track->mainBuffer() != mMixBuffer) {
-                chain = getEffectChain_l(track->sessionId());
-                // Delegate volume control to effect in track effect chain if needed
-                if (chain != 0) {
-                    tracksWithEffect++;
-                } else {
-                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
-                            name, track->sessionId());
-                }
-            }
-
-
-            int param = AudioMixer::VOLUME;
-            if (track->mFillingUpStatus == Track::FS_FILLED) {
-                // no ramp for the first volume setting
-                track->mFillingUpStatus = Track::FS_ACTIVE;
-                if (track->mState == TrackBase::RESUMING) {
-                    track->mState = TrackBase::ACTIVE;
-                    param = AudioMixer::RAMP_VOLUME;
-                }
-                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
-            } else if (cblk->server != 0) {
-                // If the track is stopped before the first frame was mixed,
-                // do not apply ramp
-                param = AudioMixer::RAMP_VOLUME;
-            }
-
-            // compute volume for this track
-            uint32_t vl, vr, va;
-            if (track->isMuted() || track->isPausing() ||
-                mStreamTypes[track->streamType()].mute) {
-                vl = vr = va = 0;
-                if (track->isPausing()) {
-                    track->setPaused();
-                }
-            } else {
-
-                // read original volumes with volume control
-                float typeVolume = mStreamTypes[track->streamType()].volume;
-                float v = masterVolume * typeVolume;
-                uint32_t vlr = cblk->getVolumeLR();
-                vl = vlr & 0xFFFF;
-                vr = vlr >> 16;
-                // track volumes come from shared memory, so can't be trusted and must be clamped
-                if (vl > MAX_GAIN_INT) {
-                    ALOGV("Track left volume out of range: %04X", vl);
-                    vl = MAX_GAIN_INT;
-                }
-                if (vr > MAX_GAIN_INT) {
-                    ALOGV("Track right volume out of range: %04X", vr);
-                    vr = MAX_GAIN_INT;
-                }
-                // now apply the master volume and stream type volume
-                vl = (uint32_t)(v * vl) << 12;
-                vr = (uint32_t)(v * vr) << 12;
-                // assuming master volume and stream type volume each go up to 1.0,
-                // vl and vr are now in 8.24 format
-
-                uint16_t sendLevel = cblk->getSendLevel_U4_12();
-                // send level comes from shared memory and so may be corrupt
-                if (sendLevel > MAX_GAIN_INT) {
-                    ALOGV("Track send level out of range: %04X", sendLevel);
-                    sendLevel = MAX_GAIN_INT;
-                }
-                va = (uint32_t)(v * sendLevel);
-            }
-            // Delegate volume control to effect in track effect chain if needed
-            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
-                // Do not ramp volume if volume is controlled by effect
-                param = AudioMixer::VOLUME;
-                track->mHasVolumeController = true;
-            } else {
-                // force no volume ramp when volume controller was just disabled or removed
-                // from effect chain to avoid volume spike
-                if (track->mHasVolumeController) {
-                    param = AudioMixer::VOLUME;
-                }
-                track->mHasVolumeController = false;
-            }
-
-            // Convert volumes from 8.24 to 4.12 format
-            // This additional clamping is needed in case chain->setVolume_l() overshot
-            vl = (vl + (1 << 11)) >> 12;
-            if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT;
-            vr = (vr + (1 << 11)) >> 12;
-            if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
-
-            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;   // va is uint32_t, so no need to check for -
-
-            // XXX: these things DON'T need to be done each time
-            mAudioMixer->setBufferProvider(name, track);
-            mAudioMixer->enable(name);
-
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
-            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::TRACK,
-                AudioMixer::FORMAT, (void *)track->format());
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::TRACK,
-                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::RESAMPLE,
-                AudioMixer::SAMPLE_RATE,
-                (void *)(cblk->sampleRate));
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::TRACK,
-                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::TRACK,
-                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
-
-            // reset retry count
-            track->mRetryCount = kMaxTrackRetries;
-
-            // If one track is ready, set the mixer ready if:
-            //  - the mixer was not ready during previous round OR
-            //  - no other track is not ready
-            if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
-                    mixerStatus != MIXER_TRACKS_ENABLED) {
-                mixerStatus = MIXER_TRACKS_READY;
-            }
-        } else {
-            // clear effect chain input buffer if an active track underruns to avoid sending
-            // previous audio buffer again to effects
-            chain = getEffectChain_l(track->sessionId());
-            if (chain != 0) {
-                chain->clearInputBuffer();
-            }
-
-            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
-            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
-                    track->isStopped() || track->isPaused()) {
-                // We have consumed all the buffers of this track.
-                // Remove it from the list of active tracks.
-                // TODO: use actual buffer filling status instead of latency when available from
-                // audio HAL
-                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
-                size_t framesWritten =
-                        mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
-                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
-                    if (track->isStopped()) {
-                        track->reset();
-                    }
-                    tracksToRemove->add(track);
-                }
-            } else {
-                track->mUnderrunCount++;
-                // No buffers for this track. Give it a few chances to
-                // fill a buffer, then remove it from active list.
-                if (--(track->mRetryCount) <= 0) {
-                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
-                    tracksToRemove->add(track);
-                    // indicate to client process that the track was disabled because of underrun;
-                    // it will then automatically call start() when data is available
-                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
-                // If one track is not ready, mark the mixer also not ready if:
-                //  - the mixer was ready during previous round OR
-                //  - no other track is ready
-                } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
-                                mixerStatus != MIXER_TRACKS_READY) {
-                    mixerStatus = MIXER_TRACKS_ENABLED;
-                }
-            }
-            mAudioMixer->disable(name);
-        }
-
-        }   // local variable scope to avoid goto warning
-track_is_ready: ;
-
-    }
-
-    // Push the new FastMixer state if necessary
-    bool pauseAudioWatchdog = false;
-    if (didModify) {
-        state->mFastTracksGen++;
-        // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
-        if (kUseFastMixer == FastMixer_Dynamic &&
-                state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
-            state->mCommand = FastMixerState::COLD_IDLE;
-            state->mColdFutexAddr = &mFastMixerFutex;
-            state->mColdGen++;
-            mFastMixerFutex = 0;
-            if (kUseFastMixer == FastMixer_Dynamic) {
-                mNormalSink = mOutputSink;
-            }
-            // If we go into cold idle, need to wait for acknowledgement
-            // so that fast mixer stops doing I/O.
-            block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
-            pauseAudioWatchdog = true;
-        }
-        sq->end();
-    }
-    if (sq != NULL) {
-        sq->end(didModify);
-        sq->push(block);
-    }
-#ifdef AUDIO_WATCHDOG
-    if (pauseAudioWatchdog && mAudioWatchdog != 0) {
-        mAudioWatchdog->pause();
-    }
-#endif
-
-    // Now perform the deferred reset on fast tracks that have stopped
-    while (resetMask != 0) {
-        size_t i = __builtin_ctz(resetMask);
-        ALOG_ASSERT(i < count);
-        resetMask &= ~(1 << i);
-        sp<Track> t = mActiveTracks[i].promote();
-        if (t == 0) continue;
-        Track* track = t.get();
-        ALOG_ASSERT(track->isFastTrack() && track->isStopped());
-        track->reset();
-    }
-
-    // remove all the tracks that need to be...
-    count = tracksToRemove->size();
-    if (CC_UNLIKELY(count)) {
-        for (size_t i=0 ; i<count ; i++) {
-            const sp<Track>& track = tracksToRemove->itemAt(i);
-            mActiveTracks.remove(track);
-            if (track->mainBuffer() != mMixBuffer) {
-                chain = getEffectChain_l(track->sessionId());
-                if (chain != 0) {
-                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
-                    chain->decActiveTrackCnt();
-                }
-            }
-            if (track->isTerminated()) {
-                removeTrack_l(track);
-            }
-        }
-    }
-
-    // mix buffer must be cleared if all tracks are connected to an
-    // effect chain as in this case the mixer will not write to
-    // mix buffer and track effects will accumulate into it
-    if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) {
-        // FIXME as a performance optimization, should remember previous zero status
-        memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
-    }
-
-    // if any fast tracks, then status is ready
-    mMixerStatusIgnoringFastTracks = mixerStatus;
-    if (fastTracks > 0) {
-        mixerStatus = MIXER_TRACKS_READY;
-    }
-    return mixerStatus;
-}
-
-/*
-The derived values that are cached:
- - mixBufferSize from frame count * frame size
- - activeSleepTime from activeSleepTimeUs()
- - idleSleepTime from idleSleepTimeUs()
- - standbyDelay from mActiveSleepTimeUs (DIRECT only)
- - maxPeriod from frame count and sample rate (MIXER only)
-
-The parameters that affect these derived values are:
- - frame count
- - frame size
- - sample rate
- - device type: A2DP or not
- - device latency
- - format: PCM or not
- - active sleep time
- - idle sleep time
-*/
-
-void AudioFlinger::PlaybackThread::cacheParameters_l()
-{
-    mixBufferSize = mNormalFrameCount * mFrameSize;
-    activeSleepTime = activeSleepTimeUs();
-    idleSleepTime = idleSleepTimeUs();
-}
-
-void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
-{
-    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
-            this,  streamType, mTracks.size());
-    Mutex::Autolock _l(mLock);
-
-    size_t size = mTracks.size();
-    for (size_t i = 0; i < size; i++) {
-        sp<Track> t = mTracks[i];
-        if (t->streamType() == streamType) {
-            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
-            t->mCblk->cv.signal();
-        }
-    }
-}
-
-// getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
-{
-    return mAudioMixer->getTrackName(channelMask, sessionId);
-}
-
-// deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::MixerThread::deleteTrackName_l(int name)
-{
-    ALOGV("remove track (%d) and delete from mixer", name);
-    mAudioMixer->deleteTrackName(name);
-}
-
-// checkForNewParameters_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MixerThread::checkForNewParameters_l()
-{
-    // if !&IDLE, holds the FastMixer state to restore after new parameters processed
-    FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
-    bool reconfig = false;
-
-    while (!mNewParameters.isEmpty()) {
-
-        if (mFastMixer != NULL) {
-            FastMixerStateQueue *sq = mFastMixer->sq();
-            FastMixerState *state = sq->begin();
-            if (!(state->mCommand & FastMixerState::IDLE)) {
-                previousCommand = state->mCommand;
-                state->mCommand = FastMixerState::HOT_IDLE;
-                sq->end();
-                sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
-            } else {
-                sq->end(false /*didModify*/);
-            }
-        }
-
-        status_t status = NO_ERROR;
-        String8 keyValuePair = mNewParameters[0];
-        AudioParameter param = AudioParameter(keyValuePair);
-        int value;
-
-        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
-            reconfig = true;
-        }
-        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
-                status = BAD_VALUE;
-            } else {
-                reconfig = true;
-            }
-        }
-        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
-            if (value != AUDIO_CHANNEL_OUT_STEREO) {
-                status = BAD_VALUE;
-            } else {
-                reconfig = true;
-            }
-        }
-        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
-            // do not accept frame count changes if tracks are open as the track buffer
-            // size depends on frame count and correct behavior would not be guaranteed
-            // if frame count is changed after track creation
-            if (!mTracks.isEmpty()) {
-                status = INVALID_OPERATION;
-            } else {
-                reconfig = true;
-            }
-        }
-        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
-#ifdef ADD_BATTERY_DATA
-            // when changing the audio output device, call addBatteryData to notify
-            // the change
-            if (mOutDevice != value) {
-                uint32_t params = 0;
-                // check whether speaker is on
-                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
-                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
-                }
-
-                audio_devices_t deviceWithoutSpeaker
-                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
-                // check if any other device (except speaker) is on
-                if (value & deviceWithoutSpeaker ) {
-                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
-                }
-
-                if (params != 0) {
-                    addBatteryData(params);
-                }
-            }
-#endif
-
-            // forward device change to effects that have requested to be
-            // aware of attached audio device.
-            mOutDevice = value;
-            for (size_t i = 0; i < mEffectChains.size(); i++) {
-                mEffectChains[i]->setDevice_l(mOutDevice);
-            }
-        }
-
-        if (status == NO_ERROR) {
-            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
-                                                    keyValuePair.string());
-            if (!mStandby && status == INVALID_OPERATION) {
-                mOutput->stream->common.standby(&mOutput->stream->common);
-                mStandby = true;
-                mBytesWritten = 0;
-                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
-                                                       keyValuePair.string());
-            }
-            if (status == NO_ERROR && reconfig) {
-                delete mAudioMixer;
-                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
-                mAudioMixer = NULL;
-                readOutputParameters();
-                mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
-                for (size_t i = 0; i < mTracks.size() ; i++) {
-                    int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
-                    if (name < 0) break;
-                    mTracks[i]->mName = name;
-                    // limit track sample rate to 2 x new output sample rate
-                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
-                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
-                    }
-                }
-                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
-            }
-        }
-
-        mNewParameters.removeAt(0);
-
-        mParamStatus = status;
-        mParamCond.signal();
-        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
-        // already timed out waiting for the status and will never signal the condition.
-        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
-    }
-
-    if (!(previousCommand & FastMixerState::IDLE)) {
-        ALOG_ASSERT(mFastMixer != NULL);
-        FastMixerStateQueue *sq = mFastMixer->sq();
-        FastMixerState *state = sq->begin();
-        ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
-        state->mCommand = previousCommand;
-        sq->end();
-        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
-    }
-
-    return reconfig;
-}
-
-void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    PlaybackThread::dumpInternals(fd, args);
-
-    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
-    FastMixerDumpState copy = mFastMixerDumpState;
-    copy.dump(fd);
-
-#ifdef STATE_QUEUE_DUMP
-    // Similar for state queue
-    StateQueueObserverDump observerCopy = mStateQueueObserverDump;
-    observerCopy.dump(fd);
-    StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
-    mutatorCopy.dump(fd);
-#endif
-
-    // Write the tee output to a .wav file
-    NBAIO_Source *teeSource = mTeeSource.get();
-    if (teeSource != NULL) {
-        char teePath[64];
-        struct timeval tv;
-        gettimeofday(&tv, NULL);
-        struct tm tm;
-        localtime_r(&tv.tv_sec, &tm);
-        strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm);
-        int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR);
-        if (teeFd >= 0) {
-            char wavHeader[44];
-            memcpy(wavHeader,
-                "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
-                sizeof(wavHeader));
-            NBAIO_Format format = teeSource->format();
-            unsigned channelCount = Format_channelCount(format);
-            ALOG_ASSERT(channelCount <= FCC_2);
-            unsigned sampleRate = Format_sampleRate(format);
-            wavHeader[22] = channelCount;       // number of channels
-            wavHeader[24] = sampleRate;         // sample rate
-            wavHeader[25] = sampleRate >> 8;
-            wavHeader[32] = channelCount * 2;   // block alignment
-            write(teeFd, wavHeader, sizeof(wavHeader));
-            size_t total = 0;
-            bool firstRead = true;
-            for (;;) {
-#define TEE_SINK_READ 1024
-                short buffer[TEE_SINK_READ * FCC_2];
-                size_t count = TEE_SINK_READ;
-                ssize_t actual = teeSource->read(buffer, count,
-                        AudioBufferProvider::kInvalidPTS);
-                bool wasFirstRead = firstRead;
-                firstRead = false;
-                if (actual <= 0) {
-                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
-                        continue;
-                    }
-                    break;
-                }
-                ALOG_ASSERT(actual <= (ssize_t)count);
-                write(teeFd, buffer, actual * channelCount * sizeof(short));
-                total += actual;
-            }
-            lseek(teeFd, (off_t) 4, SEEK_SET);
-            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
-            write(teeFd, &temp, sizeof(temp));
-            lseek(teeFd, (off_t) 40, SEEK_SET);
-            temp =  total * channelCount * sizeof(short);
-            write(teeFd, &temp, sizeof(temp));
-            close(teeFd);
-            fdprintf(fd, "FastMixer tee copied to %s\n", teePath);
-        } else {
-            fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno));
-        }
-    }
-
-#ifdef AUDIO_WATCHDOG
-    if (mAudioWatchdog != 0) {
-        // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
-        AudioWatchdogDump wdCopy = mAudioWatchdogDump;
-        wdCopy.dump(fd);
-    }
-#endif
-}
-
-uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
-{
-    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
-}
-
-uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
-{
-    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
-}
-
-void AudioFlinger::MixerThread::cacheParameters_l()
-{
-    PlaybackThread::cacheParameters_l();
-
-    // FIXME: Relaxed timing because of a certain device that can't meet latency
-    // Should be reduced to 2x after the vendor fixes the driver issue
-    // increase threshold again due to low power audio mode. The way this warning
-    // threshold is calculated and its usefulness should be reconsidered anyway.
-    maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
-}
-
-// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
-        AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
-    :   PlaybackThread(audioFlinger, output, id, device, DIRECT)
-        // mLeftVolFloat, mRightVolFloat
-{
-}
-
-AudioFlinger::DirectOutputThread::~DirectOutputThread()
-{
-}
-
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
-    Vector< sp<Track> > *tracksToRemove
-)
-{
-    sp<Track> trackToRemove;
-
-    mixer_state mixerStatus = MIXER_IDLE;
-
-    // find out which tracks need to be processed
-    if (mActiveTracks.size() != 0) {
-        sp<Track> t = mActiveTracks[0].promote();
-        // The track died recently
-        if (t == 0) return MIXER_IDLE;
-
-        Track* const track = t.get();
-        audio_track_cblk_t* cblk = track->cblk();
-
-        // The first time a track is added we wait
-        // for all its buffers to be filled before processing it
-        uint32_t minFrames;
-        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
-            minFrames = mNormalFrameCount;
-        } else {
-            minFrames = 1;
-        }
-        if ((track->framesReady() >= minFrames) && track->isReady() &&
-                !track->isPaused() && !track->isTerminated())
-        {
-            //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
-
-            if (track->mFillingUpStatus == Track::FS_FILLED) {
-                track->mFillingUpStatus = Track::FS_ACTIVE;
-                mLeftVolFloat = mRightVolFloat = 0;
-                if (track->mState == TrackBase::RESUMING) {
-                    track->mState = TrackBase::ACTIVE;
-                }
-            }
-
-            // compute volume for this track
-            float left, right;
-            if (track->isMuted() || mMasterMute || track->isPausing() ||
-                mStreamTypes[track->streamType()].mute) {
-                left = right = 0;
-                if (track->isPausing()) {
-                    track->setPaused();
-                }
-            } else {
-                float typeVolume = mStreamTypes[track->streamType()].volume;
-                float v = mMasterVolume * typeVolume;
-                uint32_t vlr = cblk->getVolumeLR();
-                float v_clamped = v * (vlr & 0xFFFF);
-                if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
-                left = v_clamped/MAX_GAIN;
-                v_clamped = v * (vlr >> 16);
-                if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
-                right = v_clamped/MAX_GAIN;
-            }
-
-            if (left != mLeftVolFloat || right != mRightVolFloat) {
-                mLeftVolFloat = left;
-                mRightVolFloat = right;
-
-                // Convert volumes from float to 8.24
-                uint32_t vl = (uint32_t)(left * (1 << 24));
-                uint32_t vr = (uint32_t)(right * (1 << 24));
-
-                // Delegate volume control to effect in track effect chain if needed
-                // only one effect chain can be present on DirectOutputThread, so if
-                // there is one, the track is connected to it
-                if (!mEffectChains.isEmpty()) {
-                    // Do not ramp volume if volume is controlled by effect
-                    mEffectChains[0]->setVolume_l(&vl, &vr);
-                    left = (float)vl / (1 << 24);
-                    right = (float)vr / (1 << 24);
-                }
-                mOutput->stream->set_volume(mOutput->stream, left, right);
-            }
-
-            // reset retry count
-            track->mRetryCount = kMaxTrackRetriesDirect;
-            mActiveTrack = t;
-            mixerStatus = MIXER_TRACKS_READY;
-        } else {
-            // clear effect chain input buffer if an active track underruns to avoid sending
-            // previous audio buffer again to effects
-            if (!mEffectChains.isEmpty()) {
-                mEffectChains[0]->clearInputBuffer();
-            }
-
-            //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
-            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
-                    track->isStopped() || track->isPaused()) {
-                // We have consumed all the buffers of this track.
-                // Remove it from the list of active tracks.
-                // TODO: implement behavior for compressed audio
-                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
-                size_t framesWritten =
-                        mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
-                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
-                    if (track->isStopped()) {
-                        track->reset();
-                    }
-                    trackToRemove = track;
-                }
-            } else {
-                // No buffers for this track. Give it a few chances to
-                // fill a buffer, then remove it from active list.
-                if (--(track->mRetryCount) <= 0) {
-                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
-                    trackToRemove = track;
-                } else {
-                    mixerStatus = MIXER_TRACKS_ENABLED;
-                }
-            }
-        }
-    }
-
-    // FIXME merge this with similar code for removing multiple tracks
-    // remove all the tracks that need to be...
-    if (CC_UNLIKELY(trackToRemove != 0)) {
-        tracksToRemove->add(trackToRemove);
-        mActiveTracks.remove(trackToRemove);
-        if (!mEffectChains.isEmpty()) {
-            ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
-                    trackToRemove->sessionId());
-            mEffectChains[0]->decActiveTrackCnt();
-        }
-        if (trackToRemove->isTerminated()) {
-            removeTrack_l(trackToRemove);
-        }
-    }
-
-    return mixerStatus;
-}
-
-void AudioFlinger::DirectOutputThread::threadLoop_mix()
-{
-    AudioBufferProvider::Buffer buffer;
-    size_t frameCount = mFrameCount;
-    int8_t *curBuf = (int8_t *)mMixBuffer;
-    // output audio to hardware
-    while (frameCount) {
-        buffer.frameCount = frameCount;
-        mActiveTrack->getNextBuffer(&buffer);
-        if (CC_UNLIKELY(buffer.raw == NULL)) {
-            memset(curBuf, 0, frameCount * mFrameSize);
-            break;
-        }
-        memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
-        frameCount -= buffer.frameCount;
-        curBuf += buffer.frameCount * mFrameSize;
-        mActiveTrack->releaseBuffer(&buffer);
-    }
-    sleepTime = 0;
-    standbyTime = systemTime() + standbyDelay;
-    mActiveTrack.clear();
-
-}
-
-void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
-{
-    if (sleepTime == 0) {
-        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
-            sleepTime = activeSleepTime;
-        } else {
-            sleepTime = idleSleepTime;
-        }
-    } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
-        memset(mMixBuffer, 0, mFrameCount * mFrameSize);
-        sleepTime = 0;
-    }
-}
-
-// getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
-        int sessionId)
-{
-    return 0;
-}
-
-// deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
-{
-}
-
-// checkForNewParameters_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
-{
-    bool reconfig = false;
-
-    while (!mNewParameters.isEmpty()) {
-        status_t status = NO_ERROR;
-        String8 keyValuePair = mNewParameters[0];
-        AudioParameter param = AudioParameter(keyValuePair);
-        int value;
-
-        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
-            // do not accept frame count changes if tracks are open as the track buffer
-            // size depends on frame count and correct behavior would not be garantied
-            // if frame count is changed after track creation
-            if (!mTracks.isEmpty()) {
-                status = INVALID_OPERATION;
-            } else {
-                reconfig = true;
-            }
-        }
-        if (status == NO_ERROR) {
-            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
-                                                    keyValuePair.string());
-            if (!mStandby && status == INVALID_OPERATION) {
-                mOutput->stream->common.standby(&mOutput->stream->common);
-                mStandby = true;
-                mBytesWritten = 0;
-                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
-                                                       keyValuePair.string());
-            }
-            if (status == NO_ERROR && reconfig) {
-                readOutputParameters();
-                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
-            }
-        }
-
-        mNewParameters.removeAt(0);
-
-        mParamStatus = status;
-        mParamCond.signal();
-        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
-        // already timed out waiting for the status and will never signal the condition.
-        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
-    }
-    return reconfig;
-}
-
-uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
-{
-    uint32_t time;
-    if (audio_is_linear_pcm(mFormat)) {
-        time = PlaybackThread::activeSleepTimeUs();
-    } else {
-        time = 10000;
-    }
-    return time;
-}
-
-uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
-{
-    uint32_t time;
-    if (audio_is_linear_pcm(mFormat)) {
-        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
-    } else {
-        time = 10000;
-    }
-    return time;
-}
-
-uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
-{
-    uint32_t time;
-    if (audio_is_linear_pcm(mFormat)) {
-        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
-    } else {
-        time = 10000;
-    }
-    return time;
-}
-
-void AudioFlinger::DirectOutputThread::cacheParameters_l()
-{
-    PlaybackThread::cacheParameters_l();
-
-    // use shorter standby delay as on normal output to release
-    // hardware resources as soon as possible
-    standbyDelay = microseconds(activeSleepTime*2);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
-        AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
-    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING),
-        mWaitTimeMs(UINT_MAX)
-{
-    addOutputTrack(mainThread);
-}
-
-AudioFlinger::DuplicatingThread::~DuplicatingThread()
-{
-    for (size_t i = 0; i < mOutputTracks.size(); i++) {
-        mOutputTracks[i]->destroy();
-    }
-}
-
-void AudioFlinger::DuplicatingThread::threadLoop_mix()
-{
-    // mix buffers...
-    if (outputsReady(outputTracks)) {
-        mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
-    } else {
-        memset(mMixBuffer, 0, mixBufferSize);
-    }
-    sleepTime = 0;
-    writeFrames = mNormalFrameCount;
-    standbyTime = systemTime() + standbyDelay;
-}
-
-void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
-{
-    if (sleepTime == 0) {
-        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
-            sleepTime = activeSleepTime;
-        } else {
-            sleepTime = idleSleepTime;
-        }
-    } else if (mBytesWritten != 0) {
-        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
-            writeFrames = mNormalFrameCount;
-            memset(mMixBuffer, 0, mixBufferSize);
-        } else {
-            // flush remaining overflow buffers in output tracks
-            writeFrames = 0;
-        }
-        sleepTime = 0;
-    }
-}
-
-void AudioFlinger::DuplicatingThread::threadLoop_write()
-{
-    for (size_t i = 0; i < outputTracks.size(); i++) {
-        outputTracks[i]->write(mMixBuffer, writeFrames);
-    }
-    mBytesWritten += mixBufferSize;
-}
-
-void AudioFlinger::DuplicatingThread::threadLoop_standby()
-{
-    // DuplicatingThread implements standby by stopping all tracks
-    for (size_t i = 0; i < outputTracks.size(); i++) {
-        outputTracks[i]->stop();
-    }
-}
-
-void AudioFlinger::DuplicatingThread::saveOutputTracks()
-{
-    outputTracks = mOutputTracks;
-}
-
-void AudioFlinger::DuplicatingThread::clearOutputTracks()
-{
-    outputTracks.clear();
-}
-
-void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
-{
-    Mutex::Autolock _l(mLock);
-    // FIXME explain this formula
-    int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
-    OutputTrack *outputTrack = new OutputTrack(thread,
-                                            this,
-                                            mSampleRate,
-                                            mFormat,
-                                            mChannelMask,
-                                            frameCount);
-    if (outputTrack->cblk() != NULL) {
-        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
-        mOutputTracks.add(outputTrack);
-        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
-        updateWaitTime_l();
-    }
-}
-
-void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
-{
-    Mutex::Autolock _l(mLock);
-    for (size_t i = 0; i < mOutputTracks.size(); i++) {
-        if (mOutputTracks[i]->thread() == thread) {
-            mOutputTracks[i]->destroy();
-            mOutputTracks.removeAt(i);
-            updateWaitTime_l();
-            return;
-        }
-    }
-    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
-}
-
-// caller must hold mLock
-void AudioFlinger::DuplicatingThread::updateWaitTime_l()
-{
-    mWaitTimeMs = UINT_MAX;
-    for (size_t i = 0; i < mOutputTracks.size(); i++) {
-        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
-        if (strong != 0) {
-            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
-            if (waitTimeMs < mWaitTimeMs) {
-                mWaitTimeMs = waitTimeMs;
-            }
-        }
-    }
-}
-
-
-bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
-{
-    for (size_t i = 0; i < outputTracks.size(); i++) {
-        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
-        if (thread == 0) {
-            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
-            return false;
-        }
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-        // see note at standby() declaration
-        if (playbackThread->standby() && !playbackThread->isSuspended()) {
-            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
-            return false;
-        }
-    }
-    return true;
-}
-
-uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
-{
-    return (mWaitTimeMs * 1000) / 2;
-}
-
-void AudioFlinger::DuplicatingThread::cacheParameters_l()
-{
-    // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
-    updateWaitTime_l();
-
-    MixerThread::cacheParameters_l();
-}
-
-// ----------------------------------------------------------------------------
-
-// TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::ThreadBase::TrackBase::TrackBase(
-            ThreadBase *thread,
-            const sp<Client>& client,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount,
-            const sp<IMemory>& sharedBuffer,
-            int sessionId)
-    :   RefBase(),
-        mThread(thread),
-        mClient(client),
-        mCblk(NULL),
-        // mBuffer
-        // mBufferEnd
-        mFrameCount(0),
-        mState(IDLE),
-        mSampleRate(sampleRate),
-        mFormat(format),
-        mStepServerFailed(false),
-        mSessionId(sessionId)
-        // mChannelCount
-        // mChannelMask
-{
-    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
-
-    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
-    size_t size = sizeof(audio_track_cblk_t);
-    uint8_t channelCount = popcount(channelMask);
-    size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
-    if (sharedBuffer == 0) {
-        size += bufferSize;
-    }
-
-    if (client != NULL) {
-        mCblkMemory = client->heap()->allocate(size);
-        if (mCblkMemory != 0) {
-            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
-            if (mCblk != NULL) { // construct the shared structure in-place.
-                new(mCblk) audio_track_cblk_t();
-                // clear all buffers
-                mCblk->frameCount = frameCount;
-                mCblk->sampleRate = sampleRate;
-// uncomment the following lines to quickly test 32-bit wraparound
-//                mCblk->user = 0xffff0000;
-//                mCblk->server = 0xffff0000;
-//                mCblk->userBase = 0xffff0000;
-//                mCblk->serverBase = 0xffff0000;
-                mChannelCount = channelCount;
-                mChannelMask = channelMask;
-                if (sharedBuffer == 0) {
-                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
-                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
-                    // Force underrun condition to avoid false underrun callback until first data is
-                    // written to buffer (other flags are cleared)
-                    mCblk->flags = CBLK_UNDERRUN_ON;
-                } else {
-                    mBuffer = sharedBuffer->pointer();
-                }
-                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
-            }
-        } else {
-            ALOGE("not enough memory for AudioTrack size=%u", size);
-            client->heap()->dump("AudioTrack");
-            return;
-        }
-    } else {
-        mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
-        // construct the shared structure in-place.
-        new(mCblk) audio_track_cblk_t();
-        // clear all buffers
-        mCblk->frameCount = frameCount;
-        mCblk->sampleRate = sampleRate;
-// uncomment the following lines to quickly test 32-bit wraparound
-//        mCblk->user = 0xffff0000;
-//        mCblk->server = 0xffff0000;
-//        mCblk->userBase = 0xffff0000;
-//        mCblk->serverBase = 0xffff0000;
-        mChannelCount = channelCount;
-        mChannelMask = channelMask;
-        mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
-        memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
-        // Force underrun condition to avoid false underrun callback until first data is
-        // written to buffer (other flags are cleared)
-        mCblk->flags = CBLK_UNDERRUN_ON;
-        mBufferEnd = (uint8_t *)mBuffer + bufferSize;
-    }
-}
-
-AudioFlinger::ThreadBase::TrackBase::~TrackBase()
-{
-    if (mCblk != NULL) {
-        if (mClient == 0) {
-            delete mCblk;
-        } else {
-            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
-        }
-    }
-    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
-    if (mClient != 0) {
-        // Client destructor must run with AudioFlinger mutex locked
-        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
-        // If the client's reference count drops to zero, the associated destructor
-        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
-        // relying on the automatic clear() at end of scope.
-        mClient.clear();
-    }
-}
-
-// AudioBufferProvider interface
-// getNextBuffer() = 0;
-// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
-void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
-{
-    buffer->raw = NULL;
-    mFrameCount = buffer->frameCount;
-    // FIXME See note at getNextBuffer()
-    (void) step();      // ignore return value of step()
-    buffer->frameCount = 0;
-}
-
-bool AudioFlinger::ThreadBase::TrackBase::step() {
-    bool result;
-    audio_track_cblk_t* cblk = this->cblk();
-
-    result = cblk->stepServer(mFrameCount);
-    if (!result) {
-        ALOGV("stepServer failed acquiring cblk mutex");
-        mStepServerFailed = true;
-    }
-    return result;
-}
-
-void AudioFlinger::ThreadBase::TrackBase::reset() {
-    audio_track_cblk_t* cblk = this->cblk();
-
-    cblk->user = 0;
-    cblk->server = 0;
-    cblk->userBase = 0;
-    cblk->serverBase = 0;
-    mStepServerFailed = false;
-    ALOGV("TrackBase::reset");
-}
-
-int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
-    return (int)mCblk->sampleRate;
-}
-
-void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
-    audio_track_cblk_t* cblk = this->cblk();
-    size_t frameSize = cblk->frameSize;
-    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
-    int8_t *bufferEnd = bufferStart + frames * frameSize;
-
-    // Check validity of returned pointer in case the track control block would have been corrupted.
-    ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
-            "TrackBase::getBuffer buffer out of range:\n"
-                "    start: %p, end %p , mBuffer %p mBufferEnd %p\n"
-                "    server %u, serverBase %u, user %u, userBase %u, frameSize %d",
-                bufferStart, bufferEnd, mBuffer, mBufferEnd,
-                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize);
-
-    return bufferStart;
-}
-
-status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
-{
-    mSyncEvents.add(event);
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
-AudioFlinger::PlaybackThread::Track::Track(
-            PlaybackThread *thread,
-            const sp<Client>& client,
-            audio_stream_type_t streamType,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount,
-            const sp<IMemory>& sharedBuffer,
-            int sessionId,
-            IAudioFlinger::track_flags_t flags)
-    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
-    mMute(false),
-    mFillingUpStatus(FS_INVALID),
-    // mRetryCount initialized later when needed
-    mSharedBuffer(sharedBuffer),
-    mStreamType(streamType),
-    mName(-1),  // see note below
-    mMainBuffer(thread->mixBuffer()),
-    mAuxBuffer(NULL),
-    mAuxEffectId(0), mHasVolumeController(false),
-    mPresentationCompleteFrames(0),
-    mFlags(flags),
-    mFastIndex(-1),
-    mUnderrunCount(0),
-    mCachedVolume(1.0)
-{
-    if (mCblk != NULL) {
-        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
-        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
-        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
-        // to avoid leaking a track name, do not allocate one unless there is an mCblk
-        mName = thread->getTrackName_l(channelMask, sessionId);
-        mCblk->mName = mName;
-        if (mName < 0) {
-            ALOGE("no more track names available");
-            return;
-        }
-        // only allocate a fast track index if we were able to allocate a normal track name
-        if (flags & IAudioFlinger::TRACK_FAST) {
-            mCblk->flags |= CBLK_FAST;  // atomic op not needed yet
-            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
-            int i = __builtin_ctz(thread->mFastTrackAvailMask);
-            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
-            // FIXME This is too eager.  We allocate a fast track index before the
-            //       fast track becomes active.  Since fast tracks are a scarce resource,
-            //       this means we are potentially denying other more important fast tracks from
-            //       being created.  It would be better to allocate the index dynamically.
-            mFastIndex = i;
-            mCblk->mName = i;
-            // Read the initial underruns because this field is never cleared by the fast mixer
-            mObservedUnderruns = thread->getFastTrackUnderruns(i);
-            thread->mFastTrackAvailMask &= ~(1 << i);
-        }
-    }
-    ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
-}
-
-AudioFlinger::PlaybackThread::Track::~Track()
-{
-    ALOGV("PlaybackThread::Track destructor");
-}
-
-void AudioFlinger::PlaybackThread::Track::destroy()
-{
-    // NOTE: destroyTrack_l() can remove a strong reference to this Track
-    // by removing it from mTracks vector, so there is a risk that this Tracks's
-    // destructor is called. As the destructor needs to lock mLock,
-    // we must acquire a strong reference on this Track before locking mLock
-    // here so that the destructor is called only when exiting this function.
-    // On the other hand, as long as Track::destroy() is only called by
-    // TrackHandle destructor, the TrackHandle still holds a strong ref on
-    // this Track with its member mTrack.
-    sp<Track> keep(this);
-    { // scope for mLock
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            if (!isOutputTrack()) {
-                if (mState == ACTIVE || mState == RESUMING) {
-                    AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
-
-#ifdef ADD_BATTERY_DATA
-                    // to track the speaker usage
-                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
-#endif
-                }
-                AudioSystem::releaseOutput(thread->id());
-            }
-            Mutex::Autolock _l(thread->mLock);
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-            playbackThread->destroyTrack_l(this);
-        }
-    }
-}
-
-/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
-{
-    result.append("   Name Client Type Fmt Chn mask   Session mFrCnt fCount S M F SRate  L dB  R dB  "
-                  "  Server      User     Main buf    Aux Buf  Flags Underruns\n");
-}
-
-void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
-{
-    uint32_t vlr = mCblk->getVolumeLR();
-    if (isFastTrack()) {
-        sprintf(buffer, "   F %2d", mFastIndex);
-    } else {
-        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
-    }
-    track_state state = mState;
-    char stateChar;
-    switch (state) {
-    case IDLE:
-        stateChar = 'I';
-        break;
-    case TERMINATED:
-        stateChar = 'T';
-        break;
-    case STOPPING_1:
-        stateChar = 's';
-        break;
-    case STOPPING_2:
-        stateChar = '5';
-        break;
-    case STOPPED:
-        stateChar = 'S';
-        break;
-    case RESUMING:
-        stateChar = 'R';
-        break;
-    case ACTIVE:
-        stateChar = 'A';
-        break;
-    case PAUSING:
-        stateChar = 'p';
-        break;
-    case PAUSED:
-        stateChar = 'P';
-        break;
-    case FLUSHED:
-        stateChar = 'F';
-        break;
-    default:
-        stateChar = '?';
-        break;
-    }
-    char nowInUnderrun;
-    switch (mObservedUnderruns.mBitFields.mMostRecent) {
-    case UNDERRUN_FULL:
-        nowInUnderrun = ' ';
-        break;
-    case UNDERRUN_PARTIAL:
-        nowInUnderrun = '<';
-        break;
-    case UNDERRUN_EMPTY:
-        nowInUnderrun = '*';
-        break;
-    default:
-        nowInUnderrun = '?';
-        break;
-    }
-    snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g  "
-            "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
-            mStreamType,
-            mFormat,
-            mChannelMask,
-            mSessionId,
-            mFrameCount,
-            mCblk->frameCount,
-            stateChar,
-            mMute,
-            mFillingUpStatus,
-            mCblk->sampleRate,
-            20.0 * log10((vlr & 0xFFFF) / 4096.0),
-            20.0 * log10((vlr >> 16) / 4096.0),
-            mCblk->server,
-            mCblk->user,
-            (int)mMainBuffer,
-            (int)mAuxBuffer,
-            mCblk->flags,
-            mUnderrunCount,
-            nowInUnderrun);
-}
-
-// AudioBufferProvider interface
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
-        AudioBufferProvider::Buffer* buffer, int64_t pts)
-{
-    audio_track_cblk_t* cblk = this->cblk();
-    uint32_t framesReady;
-    uint32_t framesReq = buffer->frameCount;
-
-    // Check if last stepServer failed, try to step now
-    if (mStepServerFailed) {
-        // FIXME When called by fast mixer, this takes a mutex with tryLock().
-        //       Since the fast mixer is higher priority than client callback thread,
-        //       it does not result in priority inversion for client.
-        //       But a non-blocking solution would be preferable to avoid
-        //       fast mixer being unable to tryLock(), and
-        //       to avoid the extra context switches if the client wakes up,
-        //       discovers the mutex is locked, then has to wait for fast mixer to unlock.
-        if (!step())  goto getNextBuffer_exit;
-        ALOGV("stepServer recovered");
-        mStepServerFailed = false;
-    }
-
-    // FIXME Same as above
-    framesReady = cblk->framesReady();
-
-    if (CC_LIKELY(framesReady)) {
-        uint32_t s = cblk->server;
-        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
-
-        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
-        if (framesReq > framesReady) {
-            framesReq = framesReady;
-        }
-        if (framesReq > bufferEnd - s) {
-            framesReq = bufferEnd - s;
-        }
-
-        buffer->raw = getBuffer(s, framesReq);
-        buffer->frameCount = framesReq;
-        return NO_ERROR;
-    }
-
-getNextBuffer_exit:
-    buffer->raw = NULL;
-    buffer->frameCount = 0;
-    ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
-    return NOT_ENOUGH_DATA;
-}
-
-// Note that framesReady() takes a mutex on the control block using tryLock().
-// This could result in priority inversion if framesReady() is called by the normal mixer,
-// as the normal mixer thread runs at lower
-// priority than the client's callback thread:  there is a short window within framesReady()
-// during which the normal mixer could be preempted, and the client callback would block.
-// Another problem can occur if framesReady() is called by the fast mixer:
-// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
-// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
-size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
-    return mCblk->framesReady();
-}
-
-// Don't call for fast tracks; the framesReady() could result in priority inversion
-bool AudioFlinger::PlaybackThread::Track::isReady() const {
-    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
-
-    if (framesReady() >= mCblk->frameCount ||
-            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
-        mFillingUpStatus = FS_FILLED;
-        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
-        return true;
-    }
-    return false;
-}
-
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
-                                                    int triggerSession)
-{
-    status_t status = NO_ERROR;
-    ALOGV("start(%d), calling pid %d session %d",
-            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
-
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        track_state state = mState;
-        // here the track could be either new, or restarted
-        // in both cases "unstop" the track
-        if (mState == PAUSED) {
-            mState = TrackBase::RESUMING;
-            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
-        } else {
-            mState = TrackBase::ACTIVE;
-            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
-        }
-
-        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
-            thread->mLock.unlock();
-            status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
-            thread->mLock.lock();
-
-#ifdef ADD_BATTERY_DATA
-            // to track the speaker usage
-            if (status == NO_ERROR) {
-                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
-            }
-#endif
-        }
-        if (status == NO_ERROR) {
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-            playbackThread->addTrack_l(this);
-        } else {
-            mState = state;
-            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
-        }
-    } else {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-void AudioFlinger::PlaybackThread::Track::stop()
-{
-    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        track_state state = mState;
-        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
-            // If the track is not active (PAUSED and buffers full), flush buffers
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
-                reset();
-                mState = STOPPED;
-            } else if (!isFastTrack()) {
-                mState = STOPPED;
-            } else {
-                // prepareTracks_l() will set state to STOPPING_2 after next underrun,
-                // and then to STOPPED and reset() when presentation is complete
-                mState = STOPPING_1;
-            }
-            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread);
-        }
-        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
-            thread->mLock.unlock();
-            AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
-            thread->mLock.lock();
-
-#ifdef ADD_BATTERY_DATA
-            // to track the speaker usage
-            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
-#endif
-        }
-    }
-}
-
-void AudioFlinger::PlaybackThread::Track::pause()
-{
-    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        if (mState == ACTIVE || mState == RESUMING) {
-            mState = PAUSING;
-            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
-            if (!isOutputTrack()) {
-                thread->mLock.unlock();
-                AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
-                thread->mLock.lock();
-
-#ifdef ADD_BATTERY_DATA
-                // to track the speaker usage
-                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
-#endif
-            }
-        }
-    }
-}
-
-void AudioFlinger::PlaybackThread::Track::flush()
-{
-    ALOGV("flush(%d)", mName);
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
-                mState != PAUSING && mState != IDLE && mState != FLUSHED) {
-            return;
-        }
-        // No point remaining in PAUSED state after a flush => go to
-        // FLUSHED state
-        mState = FLUSHED;
-        // do not reset the track if it is still in the process of being stopped or paused.
-        // this will be done by prepareTracks_l() when the track is stopped.
-        // prepareTracks_l() will see mState == FLUSHED, then
-        // remove from active track list, reset(), and trigger presentation complete
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
-            reset();
-        }
-    }
-}
-
-void AudioFlinger::PlaybackThread::Track::reset()
-{
-    // Do not reset twice to avoid discarding data written just after a flush and before
-    // the audioflinger thread detects the track is stopped.
-    if (!mResetDone) {
-        TrackBase::reset();
-        // Force underrun condition to avoid false underrun callback until first data is
-        // written to buffer
-        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
-        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
-        mFillingUpStatus = FS_FILLING;
-        mResetDone = true;
-        if (mState == FLUSHED) {
-            mState = IDLE;
-        }
-    }
-}
-
-void AudioFlinger::PlaybackThread::Track::mute(bool muted)
-{
-    mMute = muted;
-}
-
-status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
-{
-    status_t status = DEAD_OBJECT;
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-        sp<AudioFlinger> af = mClient->audioFlinger();
-
-        Mutex::Autolock _l(af->mLock);
-
-        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
-
-        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
-            Mutex::Autolock _dl(playbackThread->mLock);
-            Mutex::Autolock _sl(srcThread->mLock);
-            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
-            if (chain == 0) {
-                return INVALID_OPERATION;
-            }
-
-            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
-            if (effect == 0) {
-                return INVALID_OPERATION;
-            }
-            srcThread->removeEffect_l(effect);
-            playbackThread->addEffect_l(effect);
-            // removeEffect_l() has stopped the effect if it was active so it must be restarted
-            if (effect->state() == EffectModule::ACTIVE ||
-                    effect->state() == EffectModule::STOPPING) {
-                effect->start();
-            }
-
-            sp<EffectChain> dstChain = effect->chain().promote();
-            if (dstChain == 0) {
-                srcThread->addEffect_l(effect);
-                return INVALID_OPERATION;
-            }
-            AudioSystem::unregisterEffect(effect->id());
-            AudioSystem::registerEffect(&effect->desc(),
-                                        srcThread->id(),
-                                        dstChain->strategy(),
-                                        AUDIO_SESSION_OUTPUT_MIX,
-                                        effect->id());
-        }
-        status = playbackThread->attachAuxEffect(this, EffectId);
-    }
-    return status;
-}
-
-void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
-{
-    mAuxEffectId = EffectId;
-    mAuxBuffer = buffer;
-}
-
-bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
-                                                         size_t audioHalFrames)
-{
-    // a track is considered presented when the total number of frames written to audio HAL
-    // corresponds to the number of frames written when presentationComplete() is called for the
-    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
-    if (mPresentationCompleteFrames == 0) {
-        mPresentationCompleteFrames = framesWritten + audioHalFrames;
-        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
-                  mPresentationCompleteFrames, audioHalFrames);
-    }
-    if (framesWritten >= mPresentationCompleteFrames) {
-        ALOGV("presentationComplete() session %d complete: framesWritten %d",
-                  mSessionId, framesWritten);
-        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
-        return true;
-    }
-    return false;
-}
-
-void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
-{
-    for (int i = 0; i < (int)mSyncEvents.size(); i++) {
-        if (mSyncEvents[i]->type() == type) {
-            mSyncEvents[i]->trigger();
-            mSyncEvents.removeAt(i);
-            i--;
-        }
-    }
-}
-
-// implement VolumeBufferProvider interface
-
-uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
-{
-    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
-    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
-    uint32_t vlr = mCblk->getVolumeLR();
-    uint32_t vl = vlr & 0xFFFF;
-    uint32_t vr = vlr >> 16;
-    // track volumes come from shared memory, so can't be trusted and must be clamped
-    if (vl > MAX_GAIN_INT) {
-        vl = MAX_GAIN_INT;
-    }
-    if (vr > MAX_GAIN_INT) {
-        vr = MAX_GAIN_INT;
-    }
-    // now apply the cached master volume and stream type volume;
-    // this is trusted but lacks any synchronization or barrier so may be stale
-    float v = mCachedVolume;
-    vl *= v;
-    vr *= v;
-    // re-combine into U4.16
-    vlr = (vr << 16) | (vl & 0xFFFF);
-    // FIXME look at mute, pause, and stop flags
-    return vlr;
-}
-
-status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
-{
-    if (mState == TERMINATED || mState == PAUSED ||
-            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
-                                      (mState == STOPPED)))) {
-        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
-              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
-        event->cancel();
-        return INVALID_OPERATION;
-    }
-    (void) TrackBase::setSyncEvent(event);
-    return NO_ERROR;
-}
-
-// timed audio tracks
-
-sp<AudioFlinger::PlaybackThread::TimedTrack>
-AudioFlinger::PlaybackThread::TimedTrack::create(
-            PlaybackThread *thread,
-            const sp<Client>& client,
-            audio_stream_type_t streamType,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount,
-            const sp<IMemory>& sharedBuffer,
-            int sessionId) {
-    if (!client->reserveTimedTrack())
-        return 0;
-
-    return new TimedTrack(
-        thread, client, streamType, sampleRate, format, channelMask, frameCount,
-        sharedBuffer, sessionId);
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
-            PlaybackThread *thread,
-            const sp<Client>& client,
-            audio_stream_type_t streamType,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount,
-            const sp<IMemory>& sharedBuffer,
-            int sessionId)
-    : Track(thread, client, streamType, sampleRate, format, channelMask,
-            frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
-      mQueueHeadInFlight(false),
-      mTrimQueueHeadOnRelease(false),
-      mFramesPendingInQueue(0),
-      mTimedSilenceBuffer(NULL),
-      mTimedSilenceBufferSize(0),
-      mTimedAudioOutputOnTime(false),
-      mMediaTimeTransformValid(false)
-{
-    LocalClock lc;
-    mLocalTimeFreq = lc.getLocalFreq();
-
-    mLocalTimeToSampleTransform.a_zero = 0;
-    mLocalTimeToSampleTransform.b_zero = 0;
-    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
-    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
-    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
-                            &mLocalTimeToSampleTransform.a_to_b_denom);
-
-    mMediaTimeToSampleTransform.a_zero = 0;
-    mMediaTimeToSampleTransform.b_zero = 0;
-    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
-    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
-    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
-                            &mMediaTimeToSampleTransform.a_to_b_denom);
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
-    mClient->releaseTimedTrack();
-    delete [] mTimedSilenceBuffer;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
-    size_t size, sp<IMemory>* buffer) {
-
-    Mutex::Autolock _l(mTimedBufferQueueLock);
-
-    trimTimedBufferQueue_l();
-
-    // lazily initialize the shared memory heap for timed buffers
-    if (mTimedMemoryDealer == NULL) {
-        const int kTimedBufferHeapSize = 512 << 10;
-
-        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
-                                              "AudioFlingerTimed");
-        if (mTimedMemoryDealer == NULL)
-            return NO_MEMORY;
-    }
-
-    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
-    if (newBuffer == NULL) {
-        newBuffer = mTimedMemoryDealer->allocate(size);
-        if (newBuffer == NULL)
-            return NO_MEMORY;
-    }
-
-    *buffer = newBuffer;
-    return NO_ERROR;
-}
-
-// caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
-    int64_t mediaTimeNow;
-    {
-        Mutex::Autolock mttLock(mMediaTimeTransformLock);
-        if (!mMediaTimeTransformValid)
-            return;
-
-        int64_t targetTimeNow;
-        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
-            ? mCCHelper.getCommonTime(&targetTimeNow)
-            : mCCHelper.getLocalTime(&targetTimeNow);
-
-        if (OK != res)
-            return;
-
-        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
-                                                    &mediaTimeNow)) {
-            return;
-        }
-    }
-
-    size_t trimEnd;
-    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
-        int64_t bufEnd;
-
-        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
-            // We have a next buffer.  Just use its PTS as the PTS of the frame
-            // following the last frame in this buffer.  If the stream is sparse
-            // (ie, there are deliberate gaps left in the stream which should be
-            // filled with silence by the TimedAudioTrack), then this can result
-            // in one extra buffer being left un-trimmed when it could have
-            // been.  In general, this is not typical, and we would rather
-            // optimized away the TS calculation below for the more common case
-            // where PTSes are contiguous.
-            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
-        } else {
-            // We have no next buffer.  Compute the PTS of the frame following
-            // the last frame in this buffer by computing the duration of of
-            // this frame in media time units and adding it to the PTS of the
-            // buffer.
-            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
-                               / mCblk->frameSize;
-
-            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
-                                                                &bufEnd)) {
-                ALOGE("Failed to convert frame count of %lld to media time"
-                      " duration" " (scale factor %d/%u) in %s",
-                      frameCount,
-                      mMediaTimeToSampleTransform.a_to_b_numer,
-                      mMediaTimeToSampleTransform.a_to_b_denom,
-                      __PRETTY_FUNCTION__);
-                break;
-            }
-            bufEnd += mTimedBufferQueue[trimEnd].pts();
-        }
-
-        if (bufEnd > mediaTimeNow)
-            break;
-
-        // Is the buffer we want to use in the middle of a mix operation right
-        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
-        // from the mixer which should be coming back shortly.
-        if (!trimEnd && mQueueHeadInFlight) {
-            mTrimQueueHeadOnRelease = true;
-        }
-    }
-
-    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
-    if (trimStart < trimEnd) {
-        // Update the bookkeeping for framesReady()
-        for (size_t i = trimStart; i < trimEnd; ++i) {
-            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
-        }
-
-        // Now actually remove the buffers from the queue.
-        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
-    }
-}
-
-void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
-        const char* logTag) {
-    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
-                "%s called (reason \"%s\"), but timed buffer queue has no"
-                " elements to trim.", __FUNCTION__, logTag);
-
-    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
-    mTimedBufferQueue.removeAt(0);
-}
-
-void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
-        const TimedBuffer& buf,
-        const char* logTag) {
-    uint32_t bufBytes        = buf.buffer()->size();
-    uint32_t consumedAlready = buf.position();
-
-    ALOG_ASSERT(consumedAlready <= bufBytes,
-                "Bad bookkeeping while updating frames pending.  Timed buffer is"
-                " only %u bytes long, but claims to have consumed %u"
-                " bytes.  (update reason: \"%s\")",
-                bufBytes, consumedAlready, logTag);
-
-    uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize;
-    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
-                "Bad bookkeeping while updating frames pending.  Should have at"
-                " least %u queued frames, but we think we have only %u.  (update"
-                " reason: \"%s\")",
-                bufFrames, mFramesPendingInQueue, logTag);
-
-    mFramesPendingInQueue -= bufFrames;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
-    const sp<IMemory>& buffer, int64_t pts) {
-
-    {
-        Mutex::Autolock mttLock(mMediaTimeTransformLock);
-        if (!mMediaTimeTransformValid)
-            return INVALID_OPERATION;
-    }
-
-    Mutex::Autolock _l(mTimedBufferQueueLock);
-
-    uint32_t bufFrames = buffer->size() / mCblk->frameSize;
-    mFramesPendingInQueue += bufFrames;
-    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
-
-    return NO_ERROR;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
-    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
-
-    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
-           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
-           target);
-
-    if (!(target == TimedAudioTrack::LOCAL_TIME ||
-          target == TimedAudioTrack::COMMON_TIME)) {
-        return BAD_VALUE;
-    }
-
-    Mutex::Autolock lock(mMediaTimeTransformLock);
-    mMediaTimeTransform = xform;
-    mMediaTimeTransformTarget = target;
-    mMediaTimeTransformValid = true;
-
-    return NO_ERROR;
-}
-
-#define min(a, b) ((a) < (b) ? (a) : (b))
-
-// implementation of getNextBuffer for tracks whose buffers have timestamps
-status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
-    AudioBufferProvider::Buffer* buffer, int64_t pts)
-{
-    if (pts == AudioBufferProvider::kInvalidPTS) {
-        buffer->raw = NULL;
-        buffer->frameCount = 0;
-        mTimedAudioOutputOnTime = false;
-        return INVALID_OPERATION;
-    }
-
-    Mutex::Autolock _l(mTimedBufferQueueLock);
-
-    ALOG_ASSERT(!mQueueHeadInFlight,
-                "getNextBuffer called without releaseBuffer!");
-
-    while (true) {
-
-        // if we have no timed buffers, then fail
-        if (mTimedBufferQueue.isEmpty()) {
-            buffer->raw = NULL;
-            buffer->frameCount = 0;
-            return NOT_ENOUGH_DATA;
-        }
-
-        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
-
-        // calculate the PTS of the head of the timed buffer queue expressed in
-        // local time
-        int64_t headLocalPTS;
-        {
-            Mutex::Autolock mttLock(mMediaTimeTransformLock);
-
-            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
-
-            if (mMediaTimeTransform.a_to_b_denom == 0) {
-                // the transform represents a pause, so yield silence
-                timedYieldSilence_l(buffer->frameCount, buffer);
-                return NO_ERROR;
-            }
-
-            int64_t transformedPTS;
-            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
-                                                        &transformedPTS)) {
-                // the transform failed.  this shouldn't happen, but if it does
-                // then just drop this buffer
-                ALOGW("timedGetNextBuffer transform failed");
-                buffer->raw = NULL;
-                buffer->frameCount = 0;
-                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
-                return NO_ERROR;
-            }
-
-            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
-                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
-                                                          &headLocalPTS)) {
-                    buffer->raw = NULL;
-                    buffer->frameCount = 0;
-                    return INVALID_OPERATION;
-                }
-            } else {
-                headLocalPTS = transformedPTS;
-            }
-        }
-
-        // adjust the head buffer's PTS to reflect the portion of the head buffer
-        // that has already been consumed
-        int64_t effectivePTS = headLocalPTS +
-                ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
-
-        // Calculate the delta in samples between the head of the input buffer
-        // queue and the start of the next output buffer that will be written.
-        // If the transformation fails because of over or underflow, it means
-        // that the sample's position in the output stream is so far out of
-        // whack that it should just be dropped.
-        int64_t sampleDelta;
-        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
-            ALOGV("*** head buffer is too far from PTS: dropped buffer");
-            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
-                                       " mix");
-            continue;
-        }
-        if (!mLocalTimeToSampleTransform.doForwardTransform(
-                (effectivePTS - pts) << 32, &sampleDelta)) {
-            ALOGV("*** too late during sample rate transform: dropped buffer");
-            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
-            continue;
-        }
-
-        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
-               " sampleDelta=[%d.%08x]",
-               head.pts(), head.position(), pts,
-               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
-                   + (sampleDelta >> 32)),
-               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
-
-        // if the delta between the ideal placement for the next input sample and
-        // the current output position is within this threshold, then we will
-        // concatenate the next input samples to the previous output
-        const int64_t kSampleContinuityThreshold =
-                (static_cast<int64_t>(sampleRate()) << 32) / 250;
-
-        // if this is the first buffer of audio that we're emitting from this track
-        // then it should be almost exactly on time.
-        const int64_t kSampleStartupThreshold = 1LL << 32;
-
-        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
-           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
-            // the next input is close enough to being on time, so concatenate it
-            // with the last output
-            timedYieldSamples_l(buffer);
-
-            ALOGVV("*** on time: head.pos=%d frameCount=%u",
-                    head.position(), buffer->frameCount);
-            return NO_ERROR;
-        }
-
-        // Looks like our output is not on time.  Reset our on timed status.
-        // Next time we mix samples from our input queue, then should be within
-        // the StartupThreshold.
-        mTimedAudioOutputOnTime = false;
-        if (sampleDelta > 0) {
-            // the gap between the current output position and the proper start of
-            // the next input sample is too big, so fill it with silence
-            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
-
-            timedYieldSilence_l(framesUntilNextInput, buffer);
-            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
-            return NO_ERROR;
-        } else {
-            // the next input sample is late
-            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
-            size_t onTimeSamplePosition =
-                    head.position() + lateFrames * mCblk->frameSize;
-
-            if (onTimeSamplePosition > head.buffer()->size()) {
-                // all the remaining samples in the head are too late, so
-                // drop it and move on
-                ALOGV("*** too late: dropped buffer");
-                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
-                continue;
-            } else {
-                // skip over the late samples
-                head.setPosition(onTimeSamplePosition);
-
-                // yield the available samples
-                timedYieldSamples_l(buffer);
-
-                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
-                return NO_ERROR;
-            }
-        }
-    }
-}
-
-// Yield samples from the timed buffer queue head up to the given output
-// buffer's capacity.
-//
-// Caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
-    AudioBufferProvider::Buffer* buffer) {
-
-    const TimedBuffer& head = mTimedBufferQueue[0];
-
-    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
-                   head.position());
-
-    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
-                                 mCblk->frameSize);
-    size_t framesRequested = buffer->frameCount;
-    buffer->frameCount = min(framesLeftInHead, framesRequested);
-
-    mQueueHeadInFlight = true;
-    mTimedAudioOutputOnTime = true;
-}
-
-// Yield samples of silence up to the given output buffer's capacity
-//
-// Caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
-    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
-
-    // lazily allocate a buffer filled with silence
-    if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
-        delete [] mTimedSilenceBuffer;
-        mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
-        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
-        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
-    }
-
-    buffer->raw = mTimedSilenceBuffer;
-    size_t framesRequested = buffer->frameCount;
-    buffer->frameCount = min(numFrames, framesRequested);
-
-    mTimedAudioOutputOnTime = false;
-}
-
-// AudioBufferProvider interface
-void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
-    AudioBufferProvider::Buffer* buffer) {
-
-    Mutex::Autolock _l(mTimedBufferQueueLock);
-
-    // If the buffer which was just released is part of the buffer at the head
-    // of the queue, be sure to update the amt of the buffer which has been
-    // consumed.  If the buffer being returned is not part of the head of the
-    // queue, its either because the buffer is part of the silence buffer, or
-    // because the head of the timed queue was trimmed after the mixer called
-    // getNextBuffer but before the mixer called releaseBuffer.
-    if (buffer->raw == mTimedSilenceBuffer) {
-        ALOG_ASSERT(!mQueueHeadInFlight,
-                    "Queue head in flight during release of silence buffer!");
-        goto done;
-    }
-
-    ALOG_ASSERT(mQueueHeadInFlight,
-                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
-                " head in flight.");
-
-    if (mTimedBufferQueue.size()) {
-        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
-
-        void* start = head.buffer()->pointer();
-        void* end   = reinterpret_cast<void*>(
-                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
-                        + head.buffer()->size());
-
-        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
-                    "released buffer not within the head of the timed buffer"
-                    " queue; qHead = [%p, %p], released buffer = %p",
-                    start, end, buffer->raw);
-
-        head.setPosition(head.position() +
-                (buffer->frameCount * mCblk->frameSize));
-        mQueueHeadInFlight = false;
-
-        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
-                    "Bad bookkeeping during releaseBuffer!  Should have at"
-                    " least %u queued frames, but we think we have only %u",
-                    buffer->frameCount, mFramesPendingInQueue);
-
-        mFramesPendingInQueue -= buffer->frameCount;
-
-        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
-            || mTrimQueueHeadOnRelease) {
-            trimTimedBufferQueueHead_l("releaseBuffer");
-            mTrimQueueHeadOnRelease = false;
-        }
-    } else {
-        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
-                  " buffers in the timed buffer queue");
-    }
-
-done:
-    buffer->raw = 0;
-    buffer->frameCount = 0;
-}
-
-size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
-    Mutex::Autolock _l(mTimedBufferQueueLock);
-    return mFramesPendingInQueue;
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
-        : mPTS(0), mPosition(0) {}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
-    const sp<IMemory>& buffer, int64_t pts)
-        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
-
-// ----------------------------------------------------------------------------
-
-// RecordTrack constructor must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread::RecordTrack::RecordTrack(
-            RecordThread *thread,
-            const sp<Client>& client,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount,
-            int sessionId)
-    :   TrackBase(thread, client, sampleRate, format,
-                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
-        mOverflow(false)
-{
-    if (mCblk != NULL) {
-        ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
-        if (format == AUDIO_FORMAT_PCM_16_BIT) {
-            mCblk->frameSize = mChannelCount * sizeof(int16_t);
-        } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
-            mCblk->frameSize = mChannelCount * sizeof(int8_t);
-        } else {
-            mCblk->frameSize = sizeof(int8_t);
-        }
-    }
-}
-
-AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
-{
-    ALOGV("%s", __func__);
-}
-
-// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
-{
-    audio_track_cblk_t* cblk = this->cblk();
-    uint32_t framesAvail;
-    uint32_t framesReq = buffer->frameCount;
-
-    // Check if last stepServer failed, try to step now
-    if (mStepServerFailed) {
-        if (!step()) goto getNextBuffer_exit;
-        ALOGV("stepServer recovered");
-        mStepServerFailed = false;
-    }
-
-    framesAvail = cblk->framesAvailable_l();
-
-    if (CC_LIKELY(framesAvail)) {
-        uint32_t s = cblk->server;
-        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
-
-        if (framesReq > framesAvail) {
-            framesReq = framesAvail;
-        }
-        if (framesReq > bufferEnd - s) {
-            framesReq = bufferEnd - s;
-        }
-
-        buffer->raw = getBuffer(s, framesReq);
-        buffer->frameCount = framesReq;
-        return NO_ERROR;
-    }
-
-getNextBuffer_exit:
-    buffer->raw = NULL;
-    buffer->frameCount = 0;
-    return NOT_ENOUGH_DATA;
-}
-
-status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
-                                                        int triggerSession)
-{
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
-        return recordThread->start(this, event, triggerSession);
-    } else {
-        return BAD_VALUE;
-    }
-}
-
-void AudioFlinger::RecordThread::RecordTrack::stop()
-{
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
-        recordThread->mLock.lock();
-        bool doStop = recordThread->stop_l(this);
-        if (doStop) {
-            TrackBase::reset();
-            // Force overrun condition to avoid false overrun callback until first data is
-            // read from buffer
-            android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
-        }
-        recordThread->mLock.unlock();
-        if (doStop) {
-            AudioSystem::stopInput(recordThread->id());
-        }
-    }
-}
-
-/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
-{
-    result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User   FrameCount\n");
-}
-
-void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
-{
-    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x %05d\n",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
-            mFormat,
-            mChannelMask,
-            mSessionId,
-            mFrameCount,
-            mState,
-            mCblk->sampleRate,
-            mCblk->server,
-            mCblk->user,
-            mCblk->frameCount);
-}
-
-
-// ----------------------------------------------------------------------------
-
-AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
-            PlaybackThread *playbackThread,
-            DuplicatingThread *sourceThread,
-            uint32_t sampleRate,
-            audio_format_t format,
-            audio_channel_mask_t channelMask,
-            int frameCount)
-    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
-                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
-    mActive(false), mSourceThread(sourceThread)
-{
-
-    if (mCblk != NULL) {
-        mCblk->flags |= CBLK_DIRECTION_OUT;
-        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
-        mOutBuffer.frameCount = 0;
-        playbackThread->mTracks.add(this);
-        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
-                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
-                mCblk, mBuffer, mCblk->buffers,
-                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
-    } else {
-        ALOGW("Error creating output track on thread %p", playbackThread);
-    }
-}
-
-AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
-{
-    clearBufferQueue();
-}
-
-status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
-                                                          int triggerSession)
-{
-    status_t status = Track::start(event, triggerSession);
-    if (status != NO_ERROR) {
-        return status;
-    }
-
-    mActive = true;
-    mRetryCount = 127;
-    return status;
-}
-
-void AudioFlinger::PlaybackThread::OutputTrack::stop()
-{
-    Track::stop();
-    clearBufferQueue();
-    mOutBuffer.frameCount = 0;
-    mActive = false;
-}
-
-bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
-{
-    Buffer *pInBuffer;
-    Buffer inBuffer;
-    uint32_t channelCount = mChannelCount;
-    bool outputBufferFull = false;
-    inBuffer.frameCount = frames;
-    inBuffer.i16 = data;
-
-    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
-
-    if (!mActive && frames != 0) {
-        start();
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            MixerThread *mixerThread = (MixerThread *)thread.get();
-            if (mCblk->frameCount > frames){
-                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
-                    uint32_t startFrames = (mCblk->frameCount - frames);
-                    pInBuffer = new Buffer;
-                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
-                    pInBuffer->frameCount = startFrames;
-                    pInBuffer->i16 = pInBuffer->mBuffer;
-                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
-                    mBufferQueue.add(pInBuffer);
-                } else {
-                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
-                }
-            }
-        }
-    }
-
-    while (waitTimeLeftMs) {
-        // First write pending buffers, then new data
-        if (mBufferQueue.size()) {
-            pInBuffer = mBufferQueue.itemAt(0);
-        } else {
-            pInBuffer = &inBuffer;
-        }
-
-        if (pInBuffer->frameCount == 0) {
-            break;
-        }
-
-        if (mOutBuffer.frameCount == 0) {
-            mOutBuffer.frameCount = pInBuffer->frameCount;
-            nsecs_t startTime = systemTime();
-            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
-                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
-                outputBufferFull = true;
-                break;
-            }
-            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
-            if (waitTimeLeftMs >= waitTimeMs) {
-                waitTimeLeftMs -= waitTimeMs;
-            } else {
-                waitTimeLeftMs = 0;
-            }
-        }
-
-        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
-        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
-        mCblk->stepUser(outFrames);
-        pInBuffer->frameCount -= outFrames;
-        pInBuffer->i16 += outFrames * channelCount;
-        mOutBuffer.frameCount -= outFrames;
-        mOutBuffer.i16 += outFrames * channelCount;
-
-        if (pInBuffer->frameCount == 0) {
-            if (mBufferQueue.size()) {
-                mBufferQueue.removeAt(0);
-                delete [] pInBuffer->mBuffer;
-                delete pInBuffer;
-                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
-            } else {
-                break;
-            }
-        }
-    }
-
-    // If we could not write all frames, allocate a buffer and queue it for next time.
-    if (inBuffer.frameCount) {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0 && !thread->standby()) {
-            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
-                pInBuffer = new Buffer;
-                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
-                pInBuffer->frameCount = inBuffer.frameCount;
-                pInBuffer->i16 = pInBuffer->mBuffer;
-                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
-                mBufferQueue.add(pInBuffer);
-                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
-            } else {
-                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
-            }
-        }
-    }
-
-    // Calling write() with a 0 length buffer, means that no more data will be written:
-    // If no more buffers are pending, fill output track buffer to make sure it is started
-    // by output mixer.
-    if (frames == 0 && mBufferQueue.size() == 0) {
-        if (mCblk->user < mCblk->frameCount) {
-            frames = mCblk->frameCount - mCblk->user;
-            pInBuffer = new Buffer;
-            pInBuffer->mBuffer = new int16_t[frames * channelCount];
-            pInBuffer->frameCount = frames;
-            pInBuffer->i16 = pInBuffer->mBuffer;
-            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
-            mBufferQueue.add(pInBuffer);
-        } else if (mActive) {
-            stop();
-        }
-    }
-
-    return outputBufferFull;
-}
-
-status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
-{
-    int active;
-    status_t result;
-    audio_track_cblk_t* cblk = mCblk;
-    uint32_t framesReq = buffer->frameCount;
-
-//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
-    buffer->frameCount  = 0;
-
-    uint32_t framesAvail = cblk->framesAvailable();
-
-
-    if (framesAvail == 0) {
-        Mutex::Autolock _l(cblk->lock);
-        goto start_loop_here;
-        while (framesAvail == 0) {
-            active = mActive;
-            if (CC_UNLIKELY(!active)) {
-                ALOGV("Not active and NO_MORE_BUFFERS");
-                return NO_MORE_BUFFERS;
-            }
-            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
-            if (result != NO_ERROR) {
-                return NO_MORE_BUFFERS;
-            }
-            // read the server count again
-        start_loop_here:
-            framesAvail = cblk->framesAvailable_l();
-        }
-    }
-
-//    if (framesAvail < framesReq) {
-//        return NO_MORE_BUFFERS;
-//    }
-
-    if (framesReq > framesAvail) {
-        framesReq = framesAvail;
-    }
-
-    uint32_t u = cblk->user;
-    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
-
-    if (framesReq > bufferEnd - u) {
-        framesReq = bufferEnd - u;
-    }
-
-    buffer->frameCount  = framesReq;
-    buffer->raw         = (void *)cblk->buffer(u);
-    return NO_ERROR;
-}
-
-
-void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
-{
-    size_t size = mBufferQueue.size();
 
-    for (size_t i = 0; i < size; i++) {
-        Buffer *pBuffer = mBufferQueue.itemAt(i);
-        delete [] pBuffer->mBuffer;
-        delete pBuffer;
-    }
-    mBufferQueue.clear();
-}
 
 // ----------------------------------------------------------------------------
 
@@ -5790,88 +1116,6 @@
     mAudioFlinger->removeNotificationClient(mPid);
 }
 
-// ----------------------------------------------------------------------------
-
-AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
-    : BnAudioTrack(),
-      mTrack(track)
-{
-}
-
-AudioFlinger::TrackHandle::~TrackHandle() {
-    // just stop the track on deletion, associated resources
-    // will be freed from the main thread once all pending buffers have
-    // been played. Unless it's not in the active track list, in which
-    // case we free everything now...
-    mTrack->destroy();
-}
-
-sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
-    return mTrack->getCblk();
-}
-
-status_t AudioFlinger::TrackHandle::start() {
-    return mTrack->start();
-}
-
-void AudioFlinger::TrackHandle::stop() {
-    mTrack->stop();
-}
-
-void AudioFlinger::TrackHandle::flush() {
-    mTrack->flush();
-}
-
-void AudioFlinger::TrackHandle::mute(bool e) {
-    mTrack->mute(e);
-}
-
-void AudioFlinger::TrackHandle::pause() {
-    mTrack->pause();
-}
-
-status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
-{
-    return mTrack->attachAuxEffect(EffectId);
-}
-
-status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
-                                                         sp<IMemory>* buffer) {
-    if (!mTrack->isTimedTrack())
-        return INVALID_OPERATION;
-
-    PlaybackThread::TimedTrack* tt =
-            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
-    return tt->allocateTimedBuffer(size, buffer);
-}
-
-status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
-                                                     int64_t pts) {
-    if (!mTrack->isTimedTrack())
-        return INVALID_OPERATION;
-
-    PlaybackThread::TimedTrack* tt =
-            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
-    return tt->queueTimedBuffer(buffer, pts);
-}
-
-status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
-    const LinearTransform& xform, int target) {
-
-    if (!mTrack->isTimedTrack())
-        return INVALID_OPERATION;
-
-    PlaybackThread::TimedTrack* tt =
-            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
-    return tt->setMediaTimeTransform(
-        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
-}
-
-status_t AudioFlinger::TrackHandle::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    return BnAudioTrack::onTransact(code, data, reply, flags);
-}
 
 // ----------------------------------------------------------------------------
 
@@ -5881,7 +1125,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         IAudioFlinger::track_flags_t flags,
         pid_t tid,
         int *sessionId,
@@ -5921,13 +1165,14 @@
                 *sessionId = lSessionId;
             }
         }
-        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
+        // create new record track.
+        // The record track uses one track in mHardwareMixerThread by convention.
         recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
                                                   frameCount, lSessionId, flags, tid, &lStatus);
     }
     if (lStatus != NO_ERROR) {
-        // remove local strong reference to Client before deleting the RecordTrack so that the Client
-        // destructor is called by the TrackBase destructor with mLock held
+        // remove local strong reference to Client before deleting the RecordTrack so that the
+        // Client destructor is called by the TrackBase destructor with mLock held
         client.clear();
         recordTrack.clear();
         goto Exit;
@@ -5944,891 +1189,6 @@
     return recordHandle;
 }
 
-// ----------------------------------------------------------------------------
-
-AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
-    : BnAudioRecord(),
-    mRecordTrack(recordTrack)
-{
-}
-
-AudioFlinger::RecordHandle::~RecordHandle() {
-    stop_nonvirtual();
-    mRecordTrack->destroy();
-}
-
-sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
-    return mRecordTrack->getCblk();
-}
-
-status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) {
-    ALOGV("RecordHandle::start()");
-    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
-}
-
-void AudioFlinger::RecordHandle::stop() {
-    stop_nonvirtual();
-}
-
-void AudioFlinger::RecordHandle::stop_nonvirtual() {
-    ALOGV("RecordHandle::stop()");
-    mRecordTrack->stop();
-}
-
-status_t AudioFlinger::RecordHandle::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    return BnAudioRecord::onTransact(code, data, reply, flags);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
-                                         AudioStreamIn *input,
-                                         uint32_t sampleRate,
-                                         audio_channel_mask_t channelMask,
-                                         audio_io_handle_t id,
-                                         audio_devices_t device) :
-    ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),
-    mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
-    // mRsmpInIndex and mInputBytes set by readInputParameters()
-    mReqChannelCount(popcount(channelMask)),
-    mReqSampleRate(sampleRate)
-    // mBytesRead is only meaningful while active, and so is cleared in start()
-    // (but might be better to also clear here for dump?)
-{
-    snprintf(mName, kNameLength, "AudioIn_%X", id);
-
-    readInputParameters();
-}
-
-
-AudioFlinger::RecordThread::~RecordThread()
-{
-    delete[] mRsmpInBuffer;
-    delete mResampler;
-    delete[] mRsmpOutBuffer;
-}
-
-void AudioFlinger::RecordThread::onFirstRef()
-{
-    run(mName, PRIORITY_URGENT_AUDIO);
-}
-
-status_t AudioFlinger::RecordThread::readyToRun()
-{
-    status_t status = initCheck();
-    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
-    return status;
-}
-
-bool AudioFlinger::RecordThread::threadLoop()
-{
-    AudioBufferProvider::Buffer buffer;
-    sp<RecordTrack> activeTrack;
-    Vector< sp<EffectChain> > effectChains;
-
-    nsecs_t lastWarning = 0;
-
-    inputStandBy();
-    acquireWakeLock();
-
-    // used to verify we've read at least once before evaluating how many bytes were read
-    bool readOnce = false;
-
-    // start recording
-    while (!exitPending()) {
-
-        processConfigEvents();
-
-        { // scope for mLock
-            Mutex::Autolock _l(mLock);
-            checkForNewParameters_l();
-            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
-                standby();
-
-                if (exitPending()) break;
-
-                releaseWakeLock_l();
-                ALOGV("RecordThread: loop stopping");
-                // go to sleep
-                mWaitWorkCV.wait(mLock);
-                ALOGV("RecordThread: loop starting");
-                acquireWakeLock_l();
-                continue;
-            }
-            if (mActiveTrack != 0) {
-                if (mActiveTrack->mState == TrackBase::PAUSING) {
-                    standby();
-                    mActiveTrack.clear();
-                    mStartStopCond.broadcast();
-                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
-                    if (mReqChannelCount != mActiveTrack->channelCount()) {
-                        mActiveTrack.clear();
-                        mStartStopCond.broadcast();
-                    } else if (readOnce) {
-                        // record start succeeds only if first read from audio input
-                        // succeeds
-                        if (mBytesRead >= 0) {
-                            mActiveTrack->mState = TrackBase::ACTIVE;
-                        } else {
-                            mActiveTrack.clear();
-                        }
-                        mStartStopCond.broadcast();
-                    }
-                    mStandby = false;
-                } else if (mActiveTrack->mState == TrackBase::TERMINATED) {
-                    removeTrack_l(mActiveTrack);
-                    mActiveTrack.clear();
-                }
-            }
-            lockEffectChains_l(effectChains);
-        }
-
-        if (mActiveTrack != 0) {
-            if (mActiveTrack->mState != TrackBase::ACTIVE &&
-                mActiveTrack->mState != TrackBase::RESUMING) {
-                unlockEffectChains(effectChains);
-                usleep(kRecordThreadSleepUs);
-                continue;
-            }
-            for (size_t i = 0; i < effectChains.size(); i ++) {
-                effectChains[i]->process_l();
-            }
-
-            buffer.frameCount = mFrameCount;
-            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
-                readOnce = true;
-                size_t framesOut = buffer.frameCount;
-                if (mResampler == NULL) {
-                    // no resampling
-                    while (framesOut) {
-                        size_t framesIn = mFrameCount - mRsmpInIndex;
-                        if (framesIn) {
-                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
-                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
-                            if (framesIn > framesOut)
-                                framesIn = framesOut;
-                            mRsmpInIndex += framesIn;
-                            framesOut -= framesIn;
-                            if ((int)mChannelCount == mReqChannelCount ||
-                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-                                memcpy(dst, src, framesIn * mFrameSize);
-                            } else {
-                                if (mChannelCount == 1) {
-                                    upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
-                                            (int16_t *)src, framesIn);
-                                } else {
-                                    downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
-                                            (int16_t *)src, framesIn);
-                                }
-                            }
-                        }
-                        if (framesOut && mFrameCount == mRsmpInIndex) {
-                            if (framesOut == mFrameCount &&
-                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
-                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
-                                framesOut = 0;
-                            } else {
-                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
-                                mRsmpInIndex = 0;
-                            }
-                            if (mBytesRead <= 0) {
-                                if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
-                                {
-                                    ALOGE("Error reading audio input");
-                                    // Force input into standby so that it tries to
-                                    // recover at next read attempt
-                                    inputStandBy();
-                                    usleep(kRecordThreadSleepUs);
-                                }
-                                mRsmpInIndex = mFrameCount;
-                                framesOut = 0;
-                                buffer.frameCount = 0;
-                            }
-                        }
-                    }
-                } else {
-                    // resampling
-
-                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
-                    // alter output frame count as if we were expecting stereo samples
-                    if (mChannelCount == 1 && mReqChannelCount == 1) {
-                        framesOut >>= 1;
-                    }
-                    mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */);
-                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
-                    // are 32 bit aligned which should be always true.
-                    if (mChannelCount == 2 && mReqChannelCount == 1) {
-                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
-                        // the resampler always outputs stereo samples: do post stereo to mono conversion
-                        downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
-                                framesOut);
-                    } else {
-                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
-                    }
-
-                }
-                if (mFramestoDrop == 0) {
-                    mActiveTrack->releaseBuffer(&buffer);
-                } else {
-                    if (mFramestoDrop > 0) {
-                        mFramestoDrop -= buffer.frameCount;
-                        if (mFramestoDrop <= 0) {
-                            clearSyncStartEvent();
-                        }
-                    } else {
-                        mFramestoDrop += buffer.frameCount;
-                        if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
-                                mSyncStartEvent->isCancelled()) {
-                            ALOGW("Synced record %s, session %d, trigger session %d",
-                                  (mFramestoDrop >= 0) ? "timed out" : "cancelled",
-                                  mActiveTrack->sessionId(),
-                                  (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
-                            clearSyncStartEvent();
-                        }
-                    }
-                }
-                mActiveTrack->clearOverflow();
-            }
-            // client isn't retrieving buffers fast enough
-            else {
-                if (!mActiveTrack->setOverflow()) {
-                    nsecs_t now = systemTime();
-                    if ((now - lastWarning) > kWarningThrottleNs) {
-                        ALOGW("RecordThread: buffer overflow");
-                        lastWarning = now;
-                    }
-                }
-                // Release the processor for a while before asking for a new buffer.
-                // This will give the application more chance to read from the buffer and
-                // clear the overflow.
-                usleep(kRecordThreadSleepUs);
-            }
-        }
-        // enable changes in effect chain
-        unlockEffectChains(effectChains);
-        effectChains.clear();
-    }
-
-    standby();
-
-    {
-        Mutex::Autolock _l(mLock);
-        mActiveTrack.clear();
-        mStartStopCond.broadcast();
-    }
-
-    releaseWakeLock();
-
-    ALOGV("RecordThread %p exiting", this);
-    return false;
-}
-
-void AudioFlinger::RecordThread::standby()
-{
-    if (!mStandby) {
-        inputStandBy();
-        mStandby = true;
-    }
-}
-
-void AudioFlinger::RecordThread::inputStandBy()
-{
-    mInput->stream->common.standby(&mInput->stream->common);
-}
-
-sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
-        const sp<AudioFlinger::Client>& client,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        int frameCount,
-        int sessionId,
-        IAudioFlinger::track_flags_t flags,
-        pid_t tid,
-        status_t *status)
-{
-    sp<RecordTrack> track;
-    status_t lStatus;
-
-    lStatus = initCheck();
-    if (lStatus != NO_ERROR) {
-        ALOGE("Audio driver not initialized.");
-        goto Exit;
-    }
-
-    // FIXME use flags and tid similar to createTrack_l()
-
-    { // scope for mLock
-        Mutex::Autolock _l(mLock);
-
-        track = new RecordTrack(this, client, sampleRate,
-                      format, channelMask, frameCount, sessionId);
-
-        if (track->getCblk() == 0) {
-            lStatus = NO_MEMORY;
-            goto Exit;
-        }
-        mTracks.add(track);
-
-        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
-        bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
-                        mAudioFlinger->btNrecIsOff();
-        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
-        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
-    }
-    lStatus = NO_ERROR;
-
-Exit:
-    if (status) {
-        *status = lStatus;
-    }
-    return track;
-}
-
-status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
-                                           AudioSystem::sync_event_t event,
-                                           int triggerSession)
-{
-    ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
-    sp<ThreadBase> strongMe = this;
-    status_t status = NO_ERROR;
-
-    if (event == AudioSystem::SYNC_EVENT_NONE) {
-        clearSyncStartEvent();
-    } else if (event != AudioSystem::SYNC_EVENT_SAME) {
-        mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
-                                       triggerSession,
-                                       recordTrack->sessionId(),
-                                       syncStartEventCallback,
-                                       this);
-        // Sync event can be cancelled by the trigger session if the track is not in a
-        // compatible state in which case we start record immediately
-        if (mSyncStartEvent->isCancelled()) {
-            clearSyncStartEvent();
-        } else {
-            // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
-            mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
-        }
-    }
-
-    {
-        AutoMutex lock(mLock);
-        if (mActiveTrack != 0) {
-            if (recordTrack != mActiveTrack.get()) {
-                status = -EBUSY;
-            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
-                mActiveTrack->mState = TrackBase::ACTIVE;
-            }
-            return status;
-        }
-
-        recordTrack->mState = TrackBase::IDLE;
-        mActiveTrack = recordTrack;
-        mLock.unlock();
-        status_t status = AudioSystem::startInput(mId);
-        mLock.lock();
-        if (status != NO_ERROR) {
-            mActiveTrack.clear();
-            clearSyncStartEvent();
-            return status;
-        }
-        mRsmpInIndex = mFrameCount;
-        mBytesRead = 0;
-        if (mResampler != NULL) {
-            mResampler->reset();
-        }
-        mActiveTrack->mState = TrackBase::RESUMING;
-        // signal thread to start
-        ALOGV("Signal record thread");
-        mWaitWorkCV.broadcast();
-        // do not wait for mStartStopCond if exiting
-        if (exitPending()) {
-            mActiveTrack.clear();
-            status = INVALID_OPERATION;
-            goto startError;
-        }
-        mStartStopCond.wait(mLock);
-        if (mActiveTrack == 0) {
-            ALOGV("Record failed to start");
-            status = BAD_VALUE;
-            goto startError;
-        }
-        ALOGV("Record started OK");
-        return status;
-    }
-startError:
-    AudioSystem::stopInput(mId);
-    clearSyncStartEvent();
-    return status;
-}
-
-void AudioFlinger::RecordThread::clearSyncStartEvent()
-{
-    if (mSyncStartEvent != 0) {
-        mSyncStartEvent->cancel();
-    }
-    mSyncStartEvent.clear();
-    mFramestoDrop = 0;
-}
-
-void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
-{
-    sp<SyncEvent> strongEvent = event.promote();
-
-    if (strongEvent != 0) {
-        RecordThread *me = (RecordThread *)strongEvent->cookie();
-        me->handleSyncStartEvent(strongEvent);
-    }
-}
-
-void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
-{
-    if (event == mSyncStartEvent) {
-        // TODO: use actual buffer filling status instead of 2 buffers when info is available
-        // from audio HAL
-        mFramestoDrop = mFrameCount * 2;
-    }
-}
-
-bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) {
-    ALOGV("RecordThread::stop");
-    if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
-        return false;
-    }
-    recordTrack->mState = TrackBase::PAUSING;
-    // do not wait for mStartStopCond if exiting
-    if (exitPending()) {
-        return true;
-    }
-    mStartStopCond.wait(mLock);
-    // if we have been restarted, recordTrack == mActiveTrack.get() here
-    if (exitPending() || recordTrack != mActiveTrack.get()) {
-        ALOGV("Record stopped OK");
-        return true;
-    }
-    return false;
-}
-
-bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
-{
-    return false;
-}
-
-status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
-{
-#if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
-    if (!isValidSyncEvent(event)) {
-        return BAD_VALUE;
-    }
-
-    int eventSession = event->triggerSession();
-    status_t ret = NAME_NOT_FOUND;
-
-    Mutex::Autolock _l(mLock);
-
-    for (size_t i = 0; i < mTracks.size(); i++) {
-        sp<RecordTrack> track = mTracks[i];
-        if (eventSession == track->sessionId()) {
-            (void) track->setSyncEvent(event);
-            ret = NO_ERROR;
-        }
-    }
-    return ret;
-#else
-    return BAD_VALUE;
-#endif
-}
-
-void AudioFlinger::RecordThread::RecordTrack::destroy()
-{
-    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
-    sp<RecordTrack> keep(this);
-    {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            if (mState == ACTIVE || mState == RESUMING) {
-                AudioSystem::stopInput(thread->id());
-            }
-            AudioSystem::releaseInput(thread->id());
-            Mutex::Autolock _l(thread->mLock);
-            RecordThread *recordThread = (RecordThread *) thread.get();
-            recordThread->destroyTrack_l(this);
-        }
-    }
-}
-
-// destroyTrack_l() must be called with ThreadBase::mLock held
-void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
-{
-    track->mState = TrackBase::TERMINATED;
-    // active tracks are removed by threadLoop()
-    if (mActiveTrack != track) {
-        removeTrack_l(track);
-    }
-}
-
-void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
-{
-    mTracks.remove(track);
-    // need anything related to effects here?
-}
-
-void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
-{
-    dumpInternals(fd, args);
-    dumpTracks(fd, args);
-    dumpEffectChains(fd, args);
-}
-
-void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
-    result.append(buffer);
-
-    if (mActiveTrack != 0) {
-        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
-        result.append(buffer);
-    } else {
-        result.append("No active record client\n");
-    }
-
-    write(fd, result.string(), result.size());
-
-    dumpBase(fd, args);
-}
-
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
-    result.append(buffer);
-    RecordTrack::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<RecordTrack> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-
-    if (mActiveTrack != 0) {
-        snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
-        result.append(buffer);
-        RecordTrack::appendDumpHeader(result);
-        mActiveTrack->dump(buffer, SIZE);
-        result.append(buffer);
-
-    }
-    write(fd, result.string(), result.size());
-}
-
-// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
-{
-    size_t framesReq = buffer->frameCount;
-    size_t framesReady = mFrameCount - mRsmpInIndex;
-    int channelCount;
-
-    if (framesReady == 0) {
-        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
-        if (mBytesRead <= 0) {
-            if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
-                ALOGE("RecordThread::getNextBuffer() Error reading audio input");
-                // Force input into standby so that it tries to
-                // recover at next read attempt
-                inputStandBy();
-                usleep(kRecordThreadSleepUs);
-            }
-            buffer->raw = NULL;
-            buffer->frameCount = 0;
-            return NOT_ENOUGH_DATA;
-        }
-        mRsmpInIndex = 0;
-        framesReady = mFrameCount;
-    }
-
-    if (framesReq > framesReady) {
-        framesReq = framesReady;
-    }
-
-    if (mChannelCount == 1 && mReqChannelCount == 2) {
-        channelCount = 1;
-    } else {
-        channelCount = 2;
-    }
-    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
-    buffer->frameCount = framesReq;
-    return NO_ERROR;
-}
-
-// AudioBufferProvider interface
-void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
-{
-    mRsmpInIndex += buffer->frameCount;
-    buffer->frameCount = 0;
-}
-
-bool AudioFlinger::RecordThread::checkForNewParameters_l()
-{
-    bool reconfig = false;
-
-    while (!mNewParameters.isEmpty()) {
-        status_t status = NO_ERROR;
-        String8 keyValuePair = mNewParameters[0];
-        AudioParameter param = AudioParameter(keyValuePair);
-        int value;
-        audio_format_t reqFormat = mFormat;
-        int reqSamplingRate = mReqSampleRate;
-        int reqChannelCount = mReqChannelCount;
-
-        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
-            reqSamplingRate = value;
-            reconfig = true;
-        }
-        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-            reqFormat = (audio_format_t) value;
-            reconfig = true;
-        }
-        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
-            reqChannelCount = popcount(value);
-            reconfig = true;
-        }
-        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
-            // do not accept frame count changes if tracks are open as the track buffer
-            // size depends on frame count and correct behavior would not be guaranteed
-            // if frame count is changed after track creation
-            if (mActiveTrack != 0) {
-                status = INVALID_OPERATION;
-            } else {
-                reconfig = true;
-            }
-        }
-        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
-            // forward device change to effects that have requested to be
-            // aware of attached audio device.
-            for (size_t i = 0; i < mEffectChains.size(); i++) {
-                mEffectChains[i]->setDevice_l(value);
-            }
-
-            // store input device and output device but do not forward output device to audio HAL.
-            // Note that status is ignored by the caller for output device
-            // (see AudioFlinger::setParameters()
-            if (audio_is_output_devices(value)) {
-                mOutDevice = value;
-                status = BAD_VALUE;
-            } else {
-                mInDevice = value;
-                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
-                if (mTracks.size() > 0) {
-                    bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
-                                        mAudioFlinger->btNrecIsOff();
-                    for (size_t i = 0; i < mTracks.size(); i++) {
-                        sp<RecordTrack> track = mTracks[i];
-                        setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
-                        setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
-                    }
-                }
-            }
-        }
-        if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
-                mAudioSource != (audio_source_t)value) {
-            // forward device change to effects that have requested to be
-            // aware of attached audio device.
-            for (size_t i = 0; i < mEffectChains.size(); i++) {
-                mEffectChains[i]->setAudioSource_l((audio_source_t)value);
-            }
-            mAudioSource = (audio_source_t)value;
-        }
-        if (status == NO_ERROR) {
-            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
-            if (status == INVALID_OPERATION) {
-                inputStandBy();
-                status = mInput->stream->common.set_parameters(&mInput->stream->common,
-                        keyValuePair.string());
-            }
-            if (reconfig) {
-                if (status == BAD_VALUE &&
-                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
-                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
-                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
-                    popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
-                    (reqChannelCount <= FCC_2)) {
-                    status = NO_ERROR;
-                }
-                if (status == NO_ERROR) {
-                    readInputParameters();
-                    sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
-                }
-            }
-        }
-
-        mNewParameters.removeAt(0);
-
-        mParamStatus = status;
-        mParamCond.signal();
-        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
-        // already timed out waiting for the status and will never signal the condition.
-        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
-    }
-    return reconfig;
-}
-
-String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
-{
-    char *s;
-    String8 out_s8 = String8();
-
-    Mutex::Autolock _l(mLock);
-    if (initCheck() != NO_ERROR) {
-        return out_s8;
-    }
-
-    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
-    out_s8 = String8(s);
-    free(s);
-    return out_s8;
-}
-
-void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
-    AudioSystem::OutputDescriptor desc;
-    void *param2 = NULL;
-
-    switch (event) {
-    case AudioSystem::INPUT_OPENED:
-    case AudioSystem::INPUT_CONFIG_CHANGED:
-        desc.channels = mChannelMask;
-        desc.samplingRate = mSampleRate;
-        desc.format = mFormat;
-        desc.frameCount = mFrameCount;
-        desc.latency = 0;
-        param2 = &desc;
-        break;
-
-    case AudioSystem::INPUT_CLOSED:
-    default:
-        break;
-    }
-    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
-}
-
-void AudioFlinger::RecordThread::readInputParameters()
-{
-    delete mRsmpInBuffer;
-    // mRsmpInBuffer is always assigned a new[] below
-    delete mRsmpOutBuffer;
-    mRsmpOutBuffer = NULL;
-    delete mResampler;
-    mResampler = NULL;
-
-    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
-    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
-    mChannelCount = (uint16_t)popcount(mChannelMask);
-    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
-    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
-    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
-    mFrameCount = mInputBytes / mFrameSize;
-    mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
-    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
-
-    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
-    {
-        int channelCount;
-        // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
-        // stereo to mono post process as the resampler always outputs stereo.
-        if (mChannelCount == 1 && mReqChannelCount == 2) {
-            channelCount = 1;
-        } else {
-            channelCount = 2;
-        }
-        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
-        mResampler->setSampleRate(mSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
-        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
-
-        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
-        if (mChannelCount == 1 && mReqChannelCount == 1) {
-            mFrameCount >>= 1;
-        }
-
-    }
-    mRsmpInIndex = mFrameCount;
-}
-
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
-{
-    Mutex::Autolock _l(mLock);
-    if (initCheck() != NO_ERROR) {
-        return 0;
-    }
-
-    return mInput->stream->get_input_frames_lost(mInput->stream);
-}
-
-uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
-{
-    Mutex::Autolock _l(mLock);
-    uint32_t result = 0;
-    if (getEffectChain_l(sessionId) != 0) {
-        result = EFFECT_SESSION;
-    }
-
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        if (sessionId == mTracks[i]->sessionId()) {
-            result |= TRACK_SESSION;
-            break;
-        }
-    }
-
-    return result;
-}
-
-KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
-{
-    KeyedVector<int, bool> ids;
-    Mutex::Autolock _l(mLock);
-    for (size_t j = 0; j < mTracks.size(); ++j) {
-        sp<RecordThread::RecordTrack> track = mTracks[j];
-        int sessionId = track->sessionId();
-        if (ids.indexOfKey(sessionId) < 0) {
-            ids.add(sessionId, true);
-        }
-    }
-    return ids;
-}
-
-AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
-{
-    Mutex::Autolock _l(mLock);
-    AudioStreamIn *input = mInput;
-    mInput = NULL;
-    return input;
-}
-
-// this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::RecordThread::stream() const
-{
-    if (mInput == NULL) {
-        return NULL;
-    }
-    return &mInput->stream->common;
-}
 
 
 // ----------------------------------------------------------------------------
@@ -6924,14 +1284,14 @@
 
 // ----------------------------------------------------------------------------
 
-int32_t AudioFlinger::getPrimaryOutputSamplingRate()
+uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
 {
     Mutex::Autolock _l(mLock);
     PlaybackThread *thread = primaryPlaybackThread_l();
     return thread != NULL ? thread->sampleRate() : 0;
 }
 
-int32_t AudioFlinger::getPrimaryOutputFrameCount()
+size_t AudioFlinger::getPrimaryOutputFrameCount()
 {
     Mutex::Autolock _l(mLock);
     PlaybackThread *thread = primaryPlaybackThread_l();
@@ -6989,7 +1349,8 @@
                                           &outStream);
 
     mHardwareStatus = AUDIO_HW_IDLE;
-    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, "
+            "Channels %x, status %d",
             outStream,
             config.sample_rate,
             config.format,
@@ -7042,7 +1403,8 @@
     MixerThread *thread2 = checkMixerThread_l(output2);
 
     if (thread1 == NULL || thread2 == NULL) {
-        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
+        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
+                output2);
         return 0;
     }
 
@@ -7077,7 +1439,8 @@
         if (thread->type() == ThreadBase::MIXER) {
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
-                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
+                    DuplicatingThread *dupThread =
+                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
                     dupThread->removeOutputTrack((MixerThread *)thread.get());
                 }
             }
@@ -7164,16 +1527,17 @@
 
     status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
                                         &inStream);
-    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d",
+    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
+            "status %d",
             inStream,
             config.sample_rate,
             config.format,
             config.channel_mask,
             status);
 
-    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
-    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
-    // or stereo to mono conversions on 16 bit PCM inputs.
+    // If the input could not be opened with the requested parameters and we can handle the
+    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
+    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
     if (status == BAD_VALUE &&
         reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
         (config.sample_rate <= 2 * reqSamplingRate) &&
@@ -7184,18 +1548,66 @@
     }
 
     if (status == NO_ERROR && inStream != NULL) {
+
+        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
+        // or (re-)create if current Pipe is idle and does not match the new format
+        sp<NBAIO_Sink> teeSink;
+#ifdef TEE_SINK_INPUT_FRAMES
+        enum {
+            TEE_SINK_NO,    // don't copy input
+            TEE_SINK_NEW,   // copy input using a new pipe
+            TEE_SINK_OLD,   // copy input using an existing pipe
+        } kind;
+        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
+                                        popcount(inStream->common.get_channels(&inStream->common)));
+        if (format == Format_Invalid) {
+            kind = TEE_SINK_NO;
+        } else if (mRecordTeeSink == 0) {
+            kind = TEE_SINK_NEW;
+        } else if (mRecordTeeSink->getStrongCount() != 1) {
+            kind = TEE_SINK_NO;
+        } else if (format == mRecordTeeSink->format()) {
+            kind = TEE_SINK_OLD;
+        } else {
+            kind = TEE_SINK_NEW;
+        }
+        switch (kind) {
+        case TEE_SINK_NEW: {
+            Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format);
+            size_t numCounterOffers = 0;
+            const NBAIO_Format offers[1] = {format};
+            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
+            ALOG_ASSERT(index == 0);
+            PipeReader *pipeReader = new PipeReader(*pipe);
+            numCounterOffers = 0;
+            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
+            ALOG_ASSERT(index == 0);
+            mRecordTeeSink = pipe;
+            mRecordTeeSource = pipeReader;
+            teeSink = pipe;
+            }
+            break;
+        case TEE_SINK_OLD:
+            teeSink = mRecordTeeSink;
+            break;
+        case TEE_SINK_NO:
+        default:
+            break;
+        }
+#endif
         AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
 
         // Start record thread
         // RecorThread require both input and output device indication to forward to audio
         // pre processing modules
         audio_devices_t device = (*pDevices) | primaryOutputDevice_l();
+
         thread = new RecordThread(this,
                                   input,
                                   reqSamplingRate,
                                   reqChannels,
                                   id,
-                                  device);
+                                  device, teeSink);
         mRecordThreads.add(id, thread);
         ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
         if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
@@ -7736,2025 +2148,67 @@
     return NO_ERROR;
 }
 
-
-// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
-        const sp<AudioFlinger::Client>& client,
-        const sp<IEffectClient>& effectClient,
-        int32_t priority,
-        int sessionId,
-        effect_descriptor_t *desc,
-        int *enabled,
-        status_t *status
-        )
+void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
 {
-    sp<EffectModule> effect;
-    sp<EffectHandle> handle;
-    status_t lStatus;
-    sp<EffectChain> chain;
-    bool chainCreated = false;
-    bool effectCreated = false;
-    bool effectRegistered = false;
-
-    lStatus = initCheck();
-    if (lStatus != NO_ERROR) {
-        ALOGW("createEffect_l() Audio driver not initialized.");
-        goto Exit;
-    }
-
-    // Do not allow effects with session ID 0 on direct output or duplicating threads
-    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
-    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
-        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
-                desc->name, sessionId);
-        lStatus = BAD_VALUE;
-        goto Exit;
-    }
-    // Only Pre processor effects are allowed on input threads and only on input threads
-    if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
-        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
-                desc->name, desc->flags, mType);
-        lStatus = BAD_VALUE;
-        goto Exit;
-    }
-
-    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
-
-    { // scope for mLock
-        Mutex::Autolock _l(mLock);
-
-        // check for existing effect chain with the requested audio session
-        chain = getEffectChain_l(sessionId);
-        if (chain == 0) {
-            // create a new chain for this session
-            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
-            chain = new EffectChain(this, sessionId);
-            addEffectChain_l(chain);
-            chain->setStrategy(getStrategyForSession_l(sessionId));
-            chainCreated = true;
-        } else {
-            effect = chain->getEffectFromDesc_l(desc);
-        }
-
-        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
-
-        if (effect == 0) {
-            int id = mAudioFlinger->nextUniqueId();
-            // Check CPU and memory usage
-            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
-            if (lStatus != NO_ERROR) {
-                goto Exit;
-            }
-            effectRegistered = true;
-            // create a new effect module if none present in the chain
-            effect = new EffectModule(this, chain, desc, id, sessionId);
-            lStatus = effect->status();
-            if (lStatus != NO_ERROR) {
-                goto Exit;
-            }
-            lStatus = chain->addEffect_l(effect);
-            if (lStatus != NO_ERROR) {
-                goto Exit;
-            }
-            effectCreated = true;
-
-            effect->setDevice(mOutDevice);
-            effect->setDevice(mInDevice);
-            effect->setMode(mAudioFlinger->getMode());
-            effect->setAudioSource(mAudioSource);
-        }
-        // create effect handle and connect it to effect module
-        handle = new EffectHandle(effect, client, effectClient, priority);
-        lStatus = effect->addHandle(handle.get());
-        if (enabled != NULL) {
-            *enabled = (int)effect->isEnabled();
-        }
-    }
-
-Exit:
-    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
-        Mutex::Autolock _l(mLock);
-        if (effectCreated) {
-            chain->removeEffect_l(effect);
-        }
-        if (effectRegistered) {
-            AudioSystem::unregisterEffect(effect->id());
-        }
-        if (chainCreated) {
-            removeEffectChain_l(chain);
-        }
-        handle.clear();
-    }
-
-    if (status != NULL) {
-        *status = lStatus;
-    }
-    return handle;
-}
-
-sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
-{
-    Mutex::Autolock _l(mLock);
-    return getEffect_l(sessionId, effectId);
-}
-
-sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
-{
-    sp<EffectChain> chain = getEffectChain_l(sessionId);
-    return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
-}
-
-// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
-// PlaybackThread::mLock held
-status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
-{
-    // check for existing effect chain with the requested audio session
-    int sessionId = effect->sessionId();
-    sp<EffectChain> chain = getEffectChain_l(sessionId);
-    bool chainCreated = false;
-
-    if (chain == 0) {
-        // create a new chain for this session
-        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
-        chain = new EffectChain(this, sessionId);
-        addEffectChain_l(chain);
-        chain->setStrategy(getStrategyForSession_l(sessionId));
-        chainCreated = true;
-    }
-    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
-
-    if (chain->getEffectFromId_l(effect->id()) != 0) {
-        ALOGW("addEffect_l() %p effect %s already present in chain %p",
-                this, effect->desc().name, chain.get());
-        return BAD_VALUE;
-    }
-
-    status_t status = chain->addEffect_l(effect);
-    if (status != NO_ERROR) {
-        if (chainCreated) {
-            removeEffectChain_l(chain);
-        }
-        return status;
-    }
-
-    effect->setDevice(mOutDevice);
-    effect->setDevice(mInDevice);
-    effect->setMode(mAudioFlinger->getMode());
-    effect->setAudioSource(mAudioSource);
-    return NO_ERROR;
-}
-
-void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
-
-    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
-    effect_descriptor_t desc = effect->desc();
-    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-        detachAuxEffect_l(effect->id());
-    }
-
-    sp<EffectChain> chain = effect->chain().promote();
-    if (chain != 0) {
-        // remove effect chain if removing last effect
-        if (chain->removeEffect_l(effect) == 0) {
-            removeEffectChain_l(chain);
-        }
-    } else {
-        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
-    }
-}
-
-void AudioFlinger::ThreadBase::lockEffectChains_l(
-        Vector< sp<AudioFlinger::EffectChain> >& effectChains)
-{
-    effectChains = mEffectChains;
-    for (size_t i = 0; i < mEffectChains.size(); i++) {
-        mEffectChains[i]->lock();
-    }
-}
-
-void AudioFlinger::ThreadBase::unlockEffectChains(
-        const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
-{
-    for (size_t i = 0; i < effectChains.size(); i++) {
-        effectChains[i]->unlock();
-    }
-}
-
-sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
-{
-    Mutex::Autolock _l(mLock);
-    return getEffectChain_l(sessionId);
-}
-
-sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
-{
-    size_t size = mEffectChains.size();
-    for (size_t i = 0; i < size; i++) {
-        if (mEffectChains[i]->sessionId() == sessionId) {
-            return mEffectChains[i];
-        }
-    }
-    return 0;
-}
-
-void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
-{
-    Mutex::Autolock _l(mLock);
-    size_t size = mEffectChains.size();
-    for (size_t i = 0; i < size; i++) {
-        mEffectChains[i]->setMode_l(mode);
-    }
-}
-
-void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
-                                                    EffectHandle *handle,
-                                                    bool unpinIfLast) {
-
-    Mutex::Autolock _l(mLock);
-    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
-    // delete the effect module if removing last handle on it
-    if (effect->removeHandle(handle) == 0) {
-        if (!effect->isPinned() || unpinIfLast) {
-            removeEffect_l(effect);
-            AudioSystem::unregisterEffect(effect->id());
-        }
-    }
-}
-
-status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
-{
-    int session = chain->sessionId();
-    int16_t *buffer = mMixBuffer;
-    bool ownsBuffer = false;
-
-    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
-    if (session > 0) {
-        // Only one effect chain can be present in direct output thread and it uses
-        // the mix buffer as input
-        if (mType != DIRECT) {
-            size_t numSamples = mNormalFrameCount * mChannelCount;
-            buffer = new int16_t[numSamples];
-            memset(buffer, 0, numSamples * sizeof(int16_t));
-            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
-            ownsBuffer = true;
-        }
-
-        // Attach all tracks with same session ID to this chain.
-        for (size_t i = 0; i < mTracks.size(); ++i) {
-            sp<Track> track = mTracks[i];
-            if (session == track->sessionId()) {
-                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
-                track->setMainBuffer(buffer);
-                chain->incTrackCnt();
-            }
-        }
-
-        // indicate all active tracks in the chain
-        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
-            sp<Track> track = mActiveTracks[i].promote();
-            if (track == 0) continue;
-            if (session == track->sessionId()) {
-                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
-                chain->incActiveTrackCnt();
-            }
-        }
-    }
-
-    chain->setInBuffer(buffer, ownsBuffer);
-    chain->setOutBuffer(mMixBuffer);
-    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
-    // chains list in order to be processed last as it contains output stage effects
-    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
-    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
-    // after track specific effects and before output stage
-    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
-    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
-    // Effect chain for other sessions are inserted at beginning of effect
-    // chains list to be processed before output mix effects. Relative order between other
-    // sessions is not important
-    size_t size = mEffectChains.size();
-    size_t i = 0;
-    for (i = 0; i < size; i++) {
-        if (mEffectChains[i]->sessionId() < session) break;
-    }
-    mEffectChains.insertAt(chain, i);
-    checkSuspendOnAddEffectChain_l(chain);
-
-    return NO_ERROR;
-}
-
-size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
-{
-    int session = chain->sessionId();
-
-    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
-
-    for (size_t i = 0; i < mEffectChains.size(); i++) {
-        if (chain == mEffectChains[i]) {
-            mEffectChains.removeAt(i);
-            // detach all active tracks from the chain
-            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
-                sp<Track> track = mActiveTracks[i].promote();
-                if (track == 0) continue;
-                if (session == track->sessionId()) {
-                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
-                            chain.get(), session);
-                    chain->decActiveTrackCnt();
-                }
-            }
-
-            // detach all tracks with same session ID from this chain
-            for (size_t i = 0; i < mTracks.size(); ++i) {
-                sp<Track> track = mTracks[i];
-                if (session == track->sessionId()) {
-                    track->setMainBuffer(mMixBuffer);
-                    chain->decTrackCnt();
-                }
-            }
-            break;
-        }
-    }
-    return mEffectChains.size();
-}
-
-status_t AudioFlinger::PlaybackThread::attachAuxEffect(
-        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
-{
-    Mutex::Autolock _l(mLock);
-    return attachAuxEffect_l(track, EffectId);
-}
-
-status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
-        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
-{
-    status_t status = NO_ERROR;
-
-    if (EffectId == 0) {
-        track->setAuxBuffer(0, NULL);
-    } else {
-        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
-        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
-        if (effect != 0) {
-            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
-            } else {
-                status = INVALID_OPERATION;
-            }
-        } else {
-            status = BAD_VALUE;
-        }
-    }
-    return status;
-}
-
-void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
-{
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (track->auxEffectId() == effectId) {
-            attachAuxEffect_l(track, 0);
-        }
-    }
-}
-
-status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
-{
-    // only one chain per input thread
-    if (mEffectChains.size() != 0) {
-        return INVALID_OPERATION;
-    }
-    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
-
-    chain->setInBuffer(NULL);
-    chain->setOutBuffer(NULL);
-
-    checkSuspendOnAddEffectChain_l(chain);
-
-    mEffectChains.add(chain);
-
-    return NO_ERROR;
-}
-
-size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
-{
-    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
-    ALOGW_IF(mEffectChains.size() != 1,
-            "removeEffectChain_l() %p invalid chain size %d on thread %p",
-            chain.get(), mEffectChains.size(), this);
-    if (mEffectChains.size() == 1) {
-        mEffectChains.removeAt(0);
-    }
-    return 0;
-}
-
-// ----------------------------------------------------------------------------
-//  EffectModule implementation
-// ----------------------------------------------------------------------------
-
-#undef LOG_TAG
-#define LOG_TAG "AudioFlinger::EffectModule"
-
-AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
-                                        const wp<AudioFlinger::EffectChain>& chain,
-                                        effect_descriptor_t *desc,
-                                        int id,
-                                        int sessionId)
-    : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX),
-      mThread(thread), mChain(chain), mId(id), mSessionId(sessionId),
-      mDescriptor(*desc),
-      // mConfig is set by configure() and not used before then
-      mEffectInterface(NULL),
-      mStatus(NO_INIT), mState(IDLE),
-      // mMaxDisableWaitCnt is set by configure() and not used before then
-      // mDisableWaitCnt is set by process() and updateState() and not used before then
-      mSuspended(false)
-{
-    ALOGV("Constructor %p", this);
-    int lStatus;
-
-    // create effect engine from effect factory
-    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
-
-    if (mStatus != NO_ERROR) {
-        return;
-    }
-    lStatus = init();
-    if (lStatus < 0) {
-        mStatus = lStatus;
-        goto Error;
-    }
-
-    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
-    return;
-Error:
-    EffectRelease(mEffectInterface);
-    mEffectInterface = NULL;
-    ALOGV("Constructor Error %d", mStatus);
-}
-
-AudioFlinger::EffectModule::~EffectModule()
-{
-    ALOGV("Destructor %p", this);
-    if (mEffectInterface != NULL) {
-        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
-                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
-            sp<ThreadBase> thread = mThread.promote();
-            if (thread != 0) {
-                audio_stream_t *stream = thread->stream();
-                if (stream != NULL) {
-                    stream->remove_audio_effect(stream, mEffectInterface);
-                }
-            }
-        }
-        // release effect engine
-        EffectRelease(mEffectInterface);
-    }
-}
-
-status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle)
-{
-    status_t status;
-
-    Mutex::Autolock _l(mLock);
-    int priority = handle->priority();
-    size_t size = mHandles.size();
-    EffectHandle *controlHandle = NULL;
-    size_t i;
-    for (i = 0; i < size; i++) {
-        EffectHandle *h = mHandles[i];
-        if (h == NULL || h->destroyed_l()) continue;
-        // first non destroyed handle is considered in control
-        if (controlHandle == NULL)
-            controlHandle = h;
-        if (h->priority() <= priority) break;
-    }
-    // if inserted in first place, move effect control from previous owner to this handle
-    if (i == 0) {
-        bool enabled = false;
-        if (controlHandle != NULL) {
-            enabled = controlHandle->enabled();
-            controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
-        }
-        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
-        status = NO_ERROR;
-    } else {
-        status = ALREADY_EXISTS;
-    }
-    ALOGV("addHandle() %p added handle %p in position %d", this, handle, i);
-    mHandles.insertAt(handle, i);
-    return status;
-}
-
-size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle)
-{
-    Mutex::Autolock _l(mLock);
-    size_t size = mHandles.size();
-    size_t i;
-    for (i = 0; i < size; i++) {
-        if (mHandles[i] == handle) break;
-    }
-    if (i == size) {
-        return size;
-    }
-    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i);
-
-    mHandles.removeAt(i);
-    // if removed from first place, move effect control from this handle to next in line
-    if (i == 0) {
-        EffectHandle *h = controlHandle_l();
-        if (h != NULL) {
-            h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/);
-        }
-    }
-
-    // Prevent calls to process() and other functions on effect interface from now on.
-    // The effect engine will be released by the destructor when the last strong reference on
-    // this object is released which can happen after next process is called.
-    if (mHandles.size() == 0 && !mPinned) {
-        mState = DESTROYED;
-    }
-
-    return mHandles.size();
-}
-
-// must be called with EffectModule::mLock held
-AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l()
-{
-    // the first valid handle in the list has control over the module
-    for (size_t i = 0; i < mHandles.size(); i++) {
-        EffectHandle *h = mHandles[i];
-        if (h != NULL && !h->destroyed_l()) {
-            return h;
-        }
-    }
-
-    return NULL;
-}
-
-size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast)
-{
-    ALOGV("disconnect() %p handle %p", this, handle);
-    // keep a strong reference on this EffectModule to avoid calling the
-    // destructor before we exit
-    sp<EffectModule> keep(this);
-    {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            thread->disconnectEffect(keep, handle, unpinIfLast);
-        }
-    }
-    return mHandles.size();
-}
-
-void AudioFlinger::EffectModule::updateState() {
-    Mutex::Autolock _l(mLock);
-
-    switch (mState) {
-    case RESTART:
-        reset_l();
-        // FALL THROUGH
-
-    case STARTING:
-        // clear auxiliary effect input buffer for next accumulation
-        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-            memset(mConfig.inputCfg.buffer.raw,
-                   0,
-                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
-        }
-        start_l();
-        mState = ACTIVE;
-        break;
-    case STOPPING:
-        stop_l();
-        mDisableWaitCnt = mMaxDisableWaitCnt;
-        mState = STOPPED;
-        break;
-    case STOPPED:
-        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
-        // turn off sequence.
-        if (--mDisableWaitCnt == 0) {
-            reset_l();
-            mState = IDLE;
-        }
-        break;
-    default: //IDLE , ACTIVE, DESTROYED
-        break;
-    }
-}
-
-void AudioFlinger::EffectModule::process()
-{
-    Mutex::Autolock _l(mLock);
-
-    if (mState == DESTROYED || mEffectInterface == NULL ||
-            mConfig.inputCfg.buffer.raw == NULL ||
-            mConfig.outputCfg.buffer.raw == NULL) {
-        return;
-    }
-
-    if (isProcessEnabled()) {
-        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
-        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-            ditherAndClamp(mConfig.inputCfg.buffer.s32,
-                                        mConfig.inputCfg.buffer.s32,
-                                        mConfig.inputCfg.buffer.frameCount/2);
-        }
-
-        // do the actual processing in the effect engine
-        int ret = (*mEffectInterface)->process(mEffectInterface,
-                                               &mConfig.inputCfg.buffer,
-                                               &mConfig.outputCfg.buffer);
-
-        // force transition to IDLE state when engine is ready
-        if (mState == STOPPED && ret == -ENODATA) {
-            mDisableWaitCnt = 1;
-        }
-
-        // clear auxiliary effect input buffer for next accumulation
-        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-            memset(mConfig.inputCfg.buffer.raw, 0,
-                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
-        }
-    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
-                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
-        // If an insert effect is idle and input buffer is different from output buffer,
-        // accumulate input onto output
-        sp<EffectChain> chain = mChain.promote();
-        if (chain != 0 && chain->activeTrackCnt() != 0) {
-            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
-            int16_t *in = mConfig.inputCfg.buffer.s16;
-            int16_t *out = mConfig.outputCfg.buffer.s16;
-            for (size_t i = 0; i < frameCnt; i++) {
-                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
-            }
-        }
-    }
-}
-
-void AudioFlinger::EffectModule::reset_l()
-{
-    if (mEffectInterface == NULL) {
-        return;
-    }
-    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
-}
-
-status_t AudioFlinger::EffectModule::configure()
-{
-    if (mEffectInterface == NULL) {
-        return NO_INIT;
-    }
-
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread == 0) {
-        return DEAD_OBJECT;
-    }
-
-    // TODO: handle configuration of effects replacing track process
-    audio_channel_mask_t channelMask = thread->channelMask();
-
-    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
-    } else {
-        mConfig.inputCfg.channels = channelMask;
-    }
-    mConfig.outputCfg.channels = channelMask;
-    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
-    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
-    mConfig.inputCfg.samplingRate = thread->sampleRate();
-    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
-    mConfig.inputCfg.bufferProvider.cookie = NULL;
-    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
-    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
-    mConfig.outputCfg.bufferProvider.cookie = NULL;
-    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
-    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
-    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
-    // Insert effect:
-    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
-    // always overwrites output buffer: input buffer == output buffer
-    // - in other sessions:
-    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
-    //      other effect: overwrites output buffer: input buffer == output buffer
-    // Auxiliary effect:
-    //      accumulates in output buffer: input buffer != output buffer
-    // Therefore: accumulate <=> input buffer != output buffer
-    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
-        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
-    } else {
-        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
-    }
-    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
-    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
-    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
-    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
-
-    ALOGV("configure() %p thread %p buffer %p framecount %d",
-            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
-
-    status_t cmdStatus;
-    uint32_t size = sizeof(int);
-    status_t status = (*mEffectInterface)->command(mEffectInterface,
-                                                   EFFECT_CMD_SET_CONFIG,
-                                                   sizeof(effect_config_t),
-                                                   &mConfig,
-                                                   &size,
-                                                   &cmdStatus);
-    if (status == 0) {
-        status = cmdStatus;
-    }
-
-    if (status == 0 &&
-            (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) {
-        uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
-        effect_param_t *p = (effect_param_t *)buf32;
-
-        p->psize = sizeof(uint32_t);
-        p->vsize = sizeof(uint32_t);
-        size = sizeof(int);
-        *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY;
-
-        uint32_t latency = 0;
-        PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId);
-        if (pbt != NULL) {
-            latency = pbt->latency_l();
-        }
-
-        *((int32_t *)p->data + 1)= latency;
-        (*mEffectInterface)->command(mEffectInterface,
-                                     EFFECT_CMD_SET_PARAM,
-                                     sizeof(effect_param_t) + 8,
-                                     &buf32,
-                                     &size,
-                                     &cmdStatus);
-    }
-
-    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
-            (1000 * mConfig.outputCfg.buffer.frameCount);
-
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::init()
-{
-    Mutex::Autolock _l(mLock);
-    if (mEffectInterface == NULL) {
-        return NO_INIT;
-    }
-    status_t cmdStatus;
-    uint32_t size = sizeof(status_t);
-    status_t status = (*mEffectInterface)->command(mEffectInterface,
-                                                   EFFECT_CMD_INIT,
-                                                   0,
-                                                   NULL,
-                                                   &size,
-                                                   &cmdStatus);
-    if (status == 0) {
-        status = cmdStatus;
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::start()
-{
-    Mutex::Autolock _l(mLock);
-    return start_l();
-}
-
-status_t AudioFlinger::EffectModule::start_l()
-{
-    if (mEffectInterface == NULL) {
-        return NO_INIT;
-    }
-    status_t cmdStatus;
-    uint32_t size = sizeof(status_t);
-    status_t status = (*mEffectInterface)->command(mEffectInterface,
-                                                   EFFECT_CMD_ENABLE,
-                                                   0,
-                                                   NULL,
-                                                   &size,
-                                                   &cmdStatus);
-    if (status == 0) {
-        status = cmdStatus;
-    }
-    if (status == 0 &&
-            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
-             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            audio_stream_t *stream = thread->stream();
-            if (stream != NULL) {
-                stream->add_audio_effect(stream, mEffectInterface);
-            }
-        }
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::stop()
-{
-    Mutex::Autolock _l(mLock);
-    return stop_l();
-}
-
-status_t AudioFlinger::EffectModule::stop_l()
-{
-    if (mEffectInterface == NULL) {
-        return NO_INIT;
-    }
-    status_t cmdStatus;
-    uint32_t size = sizeof(status_t);
-    status_t status = (*mEffectInterface)->command(mEffectInterface,
-                                                   EFFECT_CMD_DISABLE,
-                                                   0,
-                                                   NULL,
-                                                   &size,
-                                                   &cmdStatus);
-    if (status == 0) {
-        status = cmdStatus;
-    }
-    if (status == 0 &&
-            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
-             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0) {
-            audio_stream_t *stream = thread->stream();
-            if (stream != NULL) {
-                stream->remove_audio_effect(stream, mEffectInterface);
-            }
-        }
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
-                                             uint32_t cmdSize,
-                                             void *pCmdData,
-                                             uint32_t *replySize,
-                                             void *pReplyData)
-{
-    Mutex::Autolock _l(mLock);
-//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
-
-    if (mState == DESTROYED || mEffectInterface == NULL) {
-        return NO_INIT;
-    }
-    status_t status = (*mEffectInterface)->command(mEffectInterface,
-                                                   cmdCode,
-                                                   cmdSize,
-                                                   pCmdData,
-                                                   replySize,
-                                                   pReplyData);
-    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
-        uint32_t size = (replySize == NULL) ? 0 : *replySize;
-        for (size_t i = 1; i < mHandles.size(); i++) {
-            EffectHandle *h = mHandles[i];
-            if (h != NULL && !h->destroyed_l()) {
-                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
-            }
-        }
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
-{
-    Mutex::Autolock _l(mLock);
-    return setEnabled_l(enabled);
-}
-
-// must be called with EffectModule::mLock held
-status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled)
-{
-
-    ALOGV("setEnabled %p enabled %d", this, enabled);
-
-    if (enabled != isEnabled()) {
-        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
-        if (enabled && status != NO_ERROR) {
-            return status;
-        }
-
-        switch (mState) {
-        // going from disabled to enabled
-        case IDLE:
-            mState = STARTING;
-            break;
-        case STOPPED:
-            mState = RESTART;
-            break;
-        case STOPPING:
-            mState = ACTIVE;
-            break;
-
-        // going from enabled to disabled
-        case RESTART:
-            mState = STOPPED;
-            break;
-        case STARTING:
-            mState = IDLE;
-            break;
-        case ACTIVE:
-            mState = STOPPING;
-            break;
-        case DESTROYED:
-            return NO_ERROR; // simply ignore as we are being destroyed
-        }
-        for (size_t i = 1; i < mHandles.size(); i++) {
-            EffectHandle *h = mHandles[i];
-            if (h != NULL && !h->destroyed_l()) {
-                h->setEnabled(enabled);
-            }
-        }
-    }
-    return NO_ERROR;
-}
-
-bool AudioFlinger::EffectModule::isEnabled() const
-{
-    switch (mState) {
-    case RESTART:
-    case STARTING:
-    case ACTIVE:
-        return true;
-    case IDLE:
-    case STOPPING:
-    case STOPPED:
-    case DESTROYED:
-    default:
-        return false;
-    }
-}
-
-bool AudioFlinger::EffectModule::isProcessEnabled() const
-{
-    switch (mState) {
-    case RESTART:
-    case ACTIVE:
-    case STOPPING:
-    case STOPPED:
-        return true;
-    case IDLE:
-    case STARTING:
-    case DESTROYED:
-    default:
-        return false;
-    }
-}
-
-status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
-{
-    Mutex::Autolock _l(mLock);
-    status_t status = NO_ERROR;
-
-    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
-    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
-    if (isProcessEnabled() &&
-            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
-            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
-        status_t cmdStatus;
-        uint32_t volume[2];
-        uint32_t *pVolume = NULL;
-        uint32_t size = sizeof(volume);
-        volume[0] = *left;
-        volume[1] = *right;
-        if (controller) {
-            pVolume = volume;
-        }
-        status = (*mEffectInterface)->command(mEffectInterface,
-                                              EFFECT_CMD_SET_VOLUME,
-                                              size,
-                                              volume,
-                                              &size,
-                                              pVolume);
-        if (controller && status == NO_ERROR && size == sizeof(volume)) {
-            *left = volume[0];
-            *right = volume[1];
-        }
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device)
-{
-    if (device == AUDIO_DEVICE_NONE) {
-        return NO_ERROR;
-    }
-
-    Mutex::Autolock _l(mLock);
-    status_t status = NO_ERROR;
-    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
-        status_t cmdStatus;
-        uint32_t size = sizeof(status_t);
-        uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE :
-                            EFFECT_CMD_SET_INPUT_DEVICE;
-        status = (*mEffectInterface)->command(mEffectInterface,
-                                              cmd,
-                                              sizeof(uint32_t),
-                                              &device,
-                                              &size,
-                                              &cmdStatus);
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
-{
-    Mutex::Autolock _l(mLock);
-    status_t status = NO_ERROR;
-    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
-        status_t cmdStatus;
-        uint32_t size = sizeof(status_t);
-        status = (*mEffectInterface)->command(mEffectInterface,
-                                              EFFECT_CMD_SET_AUDIO_MODE,
-                                              sizeof(audio_mode_t),
-                                              &mode,
-                                              &size,
-                                              &cmdStatus);
-        if (status == NO_ERROR) {
-            status = cmdStatus;
-        }
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source)
-{
-    Mutex::Autolock _l(mLock);
-    status_t status = NO_ERROR;
-    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) {
-        uint32_t size = 0;
-        status = (*mEffectInterface)->command(mEffectInterface,
-                                              EFFECT_CMD_SET_AUDIO_SOURCE,
-                                              sizeof(audio_source_t),
-                                              &source,
-                                              &size,
-                                              NULL);
-    }
-    return status;
-}
-
-void AudioFlinger::EffectModule::setSuspended(bool suspended)
-{
-    Mutex::Autolock _l(mLock);
-    mSuspended = suspended;
-}
-
-bool AudioFlinger::EffectModule::suspended() const
-{
-    Mutex::Autolock _l(mLock);
-    return mSuspended;
-}
-
-bool AudioFlinger::EffectModule::purgeHandles()
-{
-    bool enabled = false;
-    Mutex::Autolock _l(mLock);
-    for (size_t i = 0; i < mHandles.size(); i++) {
-        EffectHandle *handle = mHandles[i];
-        if (handle != NULL && !handle->destroyed_l()) {
-            handle->effect().clear();
-            if (handle->hasControl()) {
-                enabled = handle->enabled();
-            }
-        }
-    }
-    return enabled;
-}
-
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
-    result.append(buffer);
-
-    bool locked = tryLock(mLock);
-    // failed to lock - AudioFlinger is probably deadlocked
-    if (!locked) {
-        result.append("\t\tCould not lock Fx mutex:\n");
-    }
-
-    result.append("\t\tSession Status State Engine:\n");
-    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
-            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
-    result.append(buffer);
-
-    result.append("\t\tDescriptor:\n");
-    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
-            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
-            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
-            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
-                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
-                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
-                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
-            mDescriptor.apiVersion,
-            mDescriptor.flags);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- name: %s\n",
-            mDescriptor.name);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
-            mDescriptor.implementor);
-    result.append(buffer);
-
-    result.append("\t\t- Input configuration:\n");
-    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.inputCfg.buffer.raw,
-            mConfig.inputCfg.buffer.frameCount,
-            mConfig.inputCfg.samplingRate,
-            mConfig.inputCfg.channels,
-            mConfig.inputCfg.format);
-    result.append(buffer);
-
-    result.append("\t\t- Output configuration:\n");
-    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.outputCfg.buffer.raw,
-            mConfig.outputCfg.buffer.frameCount,
-            mConfig.outputCfg.samplingRate,
-            mConfig.outputCfg.channels,
-            mConfig.outputCfg.format);
-    result.append(buffer);
-
-    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
-    result.append(buffer);
-    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
-    for (size_t i = 0; i < mHandles.size(); ++i) {
-        EffectHandle *handle = mHandles[i];
-        if (handle != NULL && !handle->destroyed_l()) {
-            handle->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-
-    result.append("\n");
-
-    write(fd, result.string(), result.length());
-
-    if (locked) {
-        mLock.unlock();
-    }
-}
-
-// ----------------------------------------------------------------------------
-//  EffectHandle implementation
-// ----------------------------------------------------------------------------
-
-#undef LOG_TAG
-#define LOG_TAG "AudioFlinger::EffectHandle"
-
-AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
-                                        const sp<AudioFlinger::Client>& client,
-                                        const sp<IEffectClient>& effectClient,
-                                        int32_t priority)
-    : BnEffect(),
-    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
-    mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false)
-{
-    ALOGV("constructor %p", this);
-
-    if (client == 0) {
-        return;
-    }
-    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
-    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
-    if (mCblkMemory != 0) {
-        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
-
-        if (mCblk != NULL) {
-            new(mCblk) effect_param_cblk_t();
-            mBuffer = (uint8_t *)mCblk + bufOffset;
-        }
-    } else {
-        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
-        return;
-    }
-}
-
-AudioFlinger::EffectHandle::~EffectHandle()
-{
-    ALOGV("Destructor %p", this);
-
-    if (mEffect == 0) {
-        mDestroyed = true;
-        return;
-    }
-    mEffect->lock();
-    mDestroyed = true;
-    mEffect->unlock();
-    disconnect(false);
-}
-
-status_t AudioFlinger::EffectHandle::enable()
-{
-    ALOGV("enable %p", this);
-    if (!mHasControl) return INVALID_OPERATION;
-    if (mEffect == 0) return DEAD_OBJECT;
-
-    if (mEnabled) {
-        return NO_ERROR;
-    }
-
-    mEnabled = true;
-
-    sp<ThreadBase> thread = mEffect->thread().promote();
-    if (thread != 0) {
-        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
-    }
-
-    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
-    if (mEffect->suspended()) {
-        return NO_ERROR;
-    }
-
-    status_t status = mEffect->setEnabled(true);
-    if (status != NO_ERROR) {
-        if (thread != 0) {
-            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
-        }
-        mEnabled = false;
-    }
-    return status;
-}
-
-status_t AudioFlinger::EffectHandle::disable()
-{
-    ALOGV("disable %p", this);
-    if (!mHasControl) return INVALID_OPERATION;
-    if (mEffect == 0) return DEAD_OBJECT;
-
-    if (!mEnabled) {
-        return NO_ERROR;
-    }
-    mEnabled = false;
-
-    if (mEffect->suspended()) {
-        return NO_ERROR;
-    }
-
-    status_t status = mEffect->setEnabled(false);
-
-    sp<ThreadBase> thread = mEffect->thread().promote();
-    if (thread != 0) {
-        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
-    }
-
-    return status;
-}
-
-void AudioFlinger::EffectHandle::disconnect()
-{
-    disconnect(true);
-}
-
-void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
-{
-    ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
-    if (mEffect == 0) {
-        return;
-    }
-    // restore suspended effects if the disconnected handle was enabled and the last one.
-    if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) {
-        sp<ThreadBase> thread = mEffect->thread().promote();
-        if (thread != 0) {
-            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
-        }
-    }
-
-    // release sp on module => module destructor can be called now
-    mEffect.clear();
-    if (mClient != 0) {
-        if (mCblk != NULL) {
-            // unlike ~TrackBase(), mCblk is never a local new, so don't delete
-            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
-        }
-        mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
-        // Client destructor must run with AudioFlinger mutex locked
-        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
-        mClient.clear();
-    }
-}
-
-status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
-                                             uint32_t cmdSize,
-                                             void *pCmdData,
-                                             uint32_t *replySize,
-                                             void *pReplyData)
-{
-//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
-//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
-
-    // only get parameter command is permitted for applications not controlling the effect
-    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
-        return INVALID_OPERATION;
-    }
-    if (mEffect == 0) return DEAD_OBJECT;
-    if (mClient == 0) return INVALID_OPERATION;
-
-    // handle commands that are not forwarded transparently to effect engine
-    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
-        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
-        // no risk to block the whole media server process or mixer threads is we are stuck here
-        Mutex::Autolock _l(mCblk->lock);
-        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
-            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
-            mCblk->serverIndex = 0;
-            mCblk->clientIndex = 0;
-            return BAD_VALUE;
-        }
-        status_t status = NO_ERROR;
-        while (mCblk->serverIndex < mCblk->clientIndex) {
-            int reply;
-            uint32_t rsize = sizeof(int);
-            int *p = (int *)(mBuffer + mCblk->serverIndex);
-            int size = *p++;
-            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
-                ALOGW("command(): invalid parameter block size");
-                break;
-            }
-            effect_param_t *param = (effect_param_t *)p;
-            if (param->psize == 0 || param->vsize == 0) {
-                ALOGW("command(): null parameter or value size");
-                mCblk->serverIndex += size;
-                continue;
-            }
-            uint32_t psize = sizeof(effect_param_t) +
-                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
-                             param->vsize;
-            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
-                                            psize,
-                                            p,
-                                            &rsize,
-                                            &reply);
-            // stop at first error encountered
-            if (ret != NO_ERROR) {
-                status = ret;
-                *(int *)pReplyData = reply;
-                break;
-            } else if (reply != NO_ERROR) {
-                *(int *)pReplyData = reply;
-                break;
-            }
-            mCblk->serverIndex += size;
-        }
-        mCblk->serverIndex = 0;
-        mCblk->clientIndex = 0;
-        return status;
-    } else if (cmdCode == EFFECT_CMD_ENABLE) {
-        *(int *)pReplyData = NO_ERROR;
-        return enable();
-    } else if (cmdCode == EFFECT_CMD_DISABLE) {
-        *(int *)pReplyData = NO_ERROR;
-        return disable();
-    }
-
-    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
-}
-
-void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
-{
-    ALOGV("setControl %p control %d", this, hasControl);
-
-    mHasControl = hasControl;
-    mEnabled = enabled;
-
-    if (signal && mEffectClient != 0) {
-        mEffectClient->controlStatusChanged(hasControl);
-    }
-}
-
-void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
-                                                 uint32_t cmdSize,
-                                                 void *pCmdData,
-                                                 uint32_t replySize,
-                                                 void *pReplyData)
-{
-    if (mEffectClient != 0) {
-        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
-    }
-}
-
-
-
-void AudioFlinger::EffectHandle::setEnabled(bool enabled)
-{
-    if (mEffectClient != 0) {
-        mEffectClient->enableStatusChanged(enabled);
-    }
-}
-
-status_t AudioFlinger::EffectHandle::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    return BnEffect::onTransact(code, data, reply, flags);
-}
-
-
-void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
-{
-    bool locked = mCblk != NULL && tryLock(mCblk->lock);
-
-    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
-            mPriority,
-            mHasControl,
-            !locked,
-            mCblk ? mCblk->clientIndex : 0,
-            mCblk ? mCblk->serverIndex : 0
-            );
-
-    if (locked) {
-        mCblk->lock.unlock();
-    }
-}
-
-#undef LOG_TAG
-#define LOG_TAG "AudioFlinger::EffectChain"
-
-AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
-                                        int sessionId)
-    : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
-      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
-      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
-{
-    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
-    if (thread == NULL) {
-        return;
-    }
-    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
-                                    thread->frameCount();
-}
-
-AudioFlinger::EffectChain::~EffectChain()
-{
-    if (mOwnInBuffer) {
-        delete mInBuffer;
-    }
-
-}
-
-// getEffectFromDesc_l() must be called with ThreadBase::mLock held
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
-{
-    size_t size = mEffects.size();
-
-    for (size_t i = 0; i < size; i++) {
-        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
-            return mEffects[i];
-        }
-    }
-    return 0;
-}
-
-// getEffectFromId_l() must be called with ThreadBase::mLock held
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
-{
-    size_t size = mEffects.size();
-
-    for (size_t i = 0; i < size; i++) {
-        // by convention, return first effect if id provided is 0 (0 is never a valid id)
-        if (id == 0 || mEffects[i]->id() == id) {
-            return mEffects[i];
-        }
-    }
-    return 0;
-}
-
-// getEffectFromType_l() must be called with ThreadBase::mLock held
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
-        const effect_uuid_t *type)
-{
-    size_t size = mEffects.size();
-
-    for (size_t i = 0; i < size; i++) {
-        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
-            return mEffects[i];
-        }
-    }
-    return 0;
-}
-
-void AudioFlinger::EffectChain::clearInputBuffer()
-{
-    Mutex::Autolock _l(mLock);
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread == 0) {
-        ALOGW("clearInputBuffer(): cannot promote mixer thread");
-        return;
-    }
-    clearInputBuffer_l(thread);
-}
-
-// Must be called with EffectChain::mLock locked
-void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread)
-{
-    size_t numSamples = thread->frameCount() * thread->channelCount();
-    memset(mInBuffer, 0, numSamples * sizeof(int16_t));
-
-}
-
-// Must be called with EffectChain::mLock locked
-void AudioFlinger::EffectChain::process_l()
-{
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread == 0) {
-        ALOGW("process_l(): cannot promote mixer thread");
-        return;
-    }
-    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
-            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
-    // always process effects unless no more tracks are on the session and the effect tail
-    // has been rendered
-    bool doProcess = true;
-    if (!isGlobalSession) {
-        bool tracksOnSession = (trackCnt() != 0);
-
-        if (!tracksOnSession && mTailBufferCount == 0) {
-            doProcess = false;
-        }
-
-        if (activeTrackCnt() == 0) {
-            // if no track is active and the effect tail has not been rendered,
-            // the input buffer must be cleared here as the mixer process will not do it
-            if (tracksOnSession || mTailBufferCount > 0) {
-                clearInputBuffer_l(thread);
-                if (mTailBufferCount > 0) {
-                    mTailBufferCount--;
-                }
-            }
-        }
-    }
-
-    size_t size = mEffects.size();
-    if (doProcess) {
-        for (size_t i = 0; i < size; i++) {
-            mEffects[i]->process();
-        }
-    }
-    for (size_t i = 0; i < size; i++) {
-        mEffects[i]->updateState();
-    }
-}
-
-// addEffect_l() must be called with PlaybackThread::mLock held
-status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
-{
-    effect_descriptor_t desc = effect->desc();
-    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
-
-    Mutex::Autolock _l(mLock);
-    effect->setChain(this);
-    sp<ThreadBase> thread = mThread.promote();
-    if (thread == 0) {
-        return NO_INIT;
-    }
-    effect->setThread(thread);
-
-    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-        // Auxiliary effects are inserted at the beginning of mEffects vector as
-        // they are processed first and accumulated in chain input buffer
-        mEffects.insertAt(effect, 0);
-
-        // the input buffer for auxiliary effect contains mono samples in
-        // 32 bit format. This is to avoid saturation in AudoMixer
-        // accumulation stage. Saturation is done in EffectModule::process() before
-        // calling the process in effect engine
-        size_t numSamples = thread->frameCount();
-        int32_t *buffer = new int32_t[numSamples];
-        memset(buffer, 0, numSamples * sizeof(int32_t));
-        effect->setInBuffer((int16_t *)buffer);
-        // auxiliary effects output samples to chain input buffer for further processing
-        // by insert effects
-        effect->setOutBuffer(mInBuffer);
-    } else {
-        // Insert effects are inserted at the end of mEffects vector as they are processed
-        //  after track and auxiliary effects.
-        // Insert effect order as a function of indicated preference:
-        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
-        //  another effect is present
-        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
-        //  last effect claiming first position
-        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
-        //  first effect claiming last position
-        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
-        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
-        // already present
-
-        size_t size = mEffects.size();
-        size_t idx_insert = size;
-        ssize_t idx_insert_first = -1;
-        ssize_t idx_insert_last = -1;
-
-        for (size_t i = 0; i < size; i++) {
-            effect_descriptor_t d = mEffects[i]->desc();
-            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
-            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
-            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
-                // check invalid effect chaining combinations
-                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
-                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
-                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
-                    return INVALID_OPERATION;
-                }
-                // remember position of first insert effect and by default
-                // select this as insert position for new effect
-                if (idx_insert == size) {
-                    idx_insert = i;
-                }
-                // remember position of last insert effect claiming
-                // first position
-                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
-                    idx_insert_first = i;
-                }
-                // remember position of first insert effect claiming
-                // last position
-                if (iPref == EFFECT_FLAG_INSERT_LAST &&
-                    idx_insert_last == -1) {
-                    idx_insert_last = i;
-                }
-            }
-        }
-
-        // modify idx_insert from first position if needed
-        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
-            if (idx_insert_last != -1) {
-                idx_insert = idx_insert_last;
-            } else {
-                idx_insert = size;
-            }
-        } else {
-            if (idx_insert_first != -1) {
-                idx_insert = idx_insert_first + 1;
-            }
-        }
-
-        // always read samples from chain input buffer
-        effect->setInBuffer(mInBuffer);
-
-        // if last effect in the chain, output samples to chain
-        // output buffer, otherwise to chain input buffer
-        if (idx_insert == size) {
-            if (idx_insert != 0) {
-                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
-                mEffects[idx_insert-1]->configure();
-            }
-            effect->setOutBuffer(mOutBuffer);
-        } else {
-            effect->setOutBuffer(mInBuffer);
-        }
-        mEffects.insertAt(effect, idx_insert);
-
-        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
-    }
-    effect->configure();
-    return NO_ERROR;
-}
-
-// removeEffect_l() must be called with PlaybackThread::mLock held
-size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
-{
-    Mutex::Autolock _l(mLock);
-    size_t size = mEffects.size();
-    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
-
-    for (size_t i = 0; i < size; i++) {
-        if (effect == mEffects[i]) {
-            // calling stop here will remove pre-processing effect from the audio HAL.
-            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
-            // the middle of a read from audio HAL
-            if (mEffects[i]->state() == EffectModule::ACTIVE ||
-                    mEffects[i]->state() == EffectModule::STOPPING) {
-                mEffects[i]->stop();
-            }
-            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
-                delete[] effect->inBuffer();
-            } else {
-                if (i == size - 1 && i != 0) {
-                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
-                    mEffects[i - 1]->configure();
-                }
-            }
-            mEffects.removeAt(i);
-            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
-            break;
-        }
-    }
-
-    return mEffects.size();
-}
-
-// setDevice_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device)
-{
-    size_t size = mEffects.size();
-    for (size_t i = 0; i < size; i++) {
-        mEffects[i]->setDevice(device);
-    }
-}
-
-// setMode_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
-{
-    size_t size = mEffects.size();
-    for (size_t i = 0; i < size; i++) {
-        mEffects[i]->setMode(mode);
-    }
-}
-
-// setAudioSource_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source)
-{
-    size_t size = mEffects.size();
-    for (size_t i = 0; i < size; i++) {
-        mEffects[i]->setAudioSource(source);
-    }
-}
-
-// setVolume_l() must be called with PlaybackThread::mLock held
-bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
-{
-    uint32_t newLeft = *left;
-    uint32_t newRight = *right;
-    bool hasControl = false;
-    int ctrlIdx = -1;
-    size_t size = mEffects.size();
-
-    // first update volume controller
-    for (size_t i = size; i > 0; i--) {
-        if (mEffects[i - 1]->isProcessEnabled() &&
-            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
-            ctrlIdx = i - 1;
-            hasControl = true;
-            break;
-        }
-    }
-
-    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
-        if (hasControl) {
-            *left = mNewLeftVolume;
-            *right = mNewRightVolume;
-        }
-        return hasControl;
-    }
-
-    mVolumeCtrlIdx = ctrlIdx;
-    mLeftVolume = newLeft;
-    mRightVolume = newRight;
-
-    // second get volume update from volume controller
-    if (ctrlIdx >= 0) {
-        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
-        mNewLeftVolume = newLeft;
-        mNewRightVolume = newRight;
-    }
-    // then indicate volume to all other effects in chain.
-    // Pass altered volume to effects before volume controller
-    // and requested volume to effects after controller
-    uint32_t lVol = newLeft;
-    uint32_t rVol = newRight;
-
-    for (size_t i = 0; i < size; i++) {
-        if ((int)i == ctrlIdx) continue;
-        // this also works for ctrlIdx == -1 when there is no volume controller
-        if ((int)i > ctrlIdx) {
-            lVol = *left;
-            rVol = *right;
-        }
-        mEffects[i]->setVolume(&lVol, &rVol, false);
-    }
-    *left = newLeft;
-    *right = newRight;
-
-    return hasControl;
-}
-
-void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
-    result.append(buffer);
-
-    bool locked = tryLock(mLock);
-    // failed to lock - AudioFlinger is probably deadlocked
-    if (!locked) {
-        result.append("\tCould not lock mutex:\n");
-    }
-
-    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
-    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
-            mEffects.size(),
-            (uint32_t)mInBuffer,
-            (uint32_t)mOutBuffer,
-            mActiveTrackCnt);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    for (size_t i = 0; i < mEffects.size(); ++i) {
-        sp<EffectModule> effect = mEffects[i];
-        if (effect != 0) {
-            effect->dump(fd, args);
-        }
-    }
-
-    if (locked) {
-        mLock.unlock();
-    }
-}
-
-// must be called with ThreadBase::mLock held
-void AudioFlinger::EffectChain::setEffectSuspended_l(
-        const effect_uuid_t *type, bool suspend)
-{
-    sp<SuspendedEffectDesc> desc;
-    // use effect type UUID timelow as key as there is no real risk of identical
-    // timeLow fields among effect type UUIDs.
-    ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow);
-    if (suspend) {
-        if (index >= 0) {
-            desc = mSuspendedEffects.valueAt(index);
-        } else {
-            desc = new SuspendedEffectDesc();
-            desc->mType = *type;
-            mSuspendedEffects.add(type->timeLow, desc);
-            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
-        }
-        if (desc->mRefCount++ == 0) {
-            sp<EffectModule> effect = getEffectIfEnabled(type);
-            if (effect != 0) {
-                desc->mEffect = effect;
-                effect->setSuspended(true);
-                effect->setEnabled(false);
-            }
-        }
-    } else {
-        if (index < 0) {
-            return;
-        }
-        desc = mSuspendedEffects.valueAt(index);
-        if (desc->mRefCount <= 0) {
-            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
-            desc->mRefCount = 1;
-        }
-        if (--desc->mRefCount == 0) {
-            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
-            if (desc->mEffect != 0) {
-                sp<EffectModule> effect = desc->mEffect.promote();
-                if (effect != 0) {
-                    effect->setSuspended(false);
-                    effect->lock();
-                    EffectHandle *handle = effect->controlHandle_l();
-                    if (handle != NULL && !handle->destroyed_l()) {
-                        effect->setEnabled_l(handle->enabled());
+    NBAIO_Source *teeSource = source.get();
+    if (teeSource != NULL) {
+        char teeTime[16];
+        struct timeval tv;
+        gettimeofday(&tv, NULL);
+        struct tm tm;
+        localtime_r(&tv.tv_sec, &tm);
+        strftime(teeTime, sizeof(teeTime), "%T", &tm);
+        char teePath[64];
+        sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id);
+        int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR);
+        if (teeFd >= 0) {
+            char wavHeader[44];
+            memcpy(wavHeader,
+                "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
+                sizeof(wavHeader));
+            NBAIO_Format format = teeSource->format();
+            unsigned channelCount = Format_channelCount(format);
+            ALOG_ASSERT(channelCount <= FCC_2);
+            uint32_t sampleRate = Format_sampleRate(format);
+            wavHeader[22] = channelCount;       // number of channels
+            wavHeader[24] = sampleRate;         // sample rate
+            wavHeader[25] = sampleRate >> 8;
+            wavHeader[32] = channelCount * 2;   // block alignment
+            write(teeFd, wavHeader, sizeof(wavHeader));
+            size_t total = 0;
+            bool firstRead = true;
+            for (;;) {
+#define TEE_SINK_READ 1024
+                short buffer[TEE_SINK_READ * FCC_2];
+                size_t count = TEE_SINK_READ;
+                ssize_t actual = teeSource->read(buffer, count,
+                        AudioBufferProvider::kInvalidPTS);
+                bool wasFirstRead = firstRead;
+                firstRead = false;
+                if (actual <= 0) {
+                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
+                        continue;
                     }
-                    effect->unlock();
+                    break;
                 }
-                desc->mEffect.clear();
+                ALOG_ASSERT(actual <= (ssize_t)count);
+                write(teeFd, buffer, actual * channelCount * sizeof(short));
+                total += actual;
             }
-            mSuspendedEffects.removeItemsAt(index);
-        }
-    }
-}
-
-// must be called with ThreadBase::mLock held
-void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
-{
-    sp<SuspendedEffectDesc> desc;
-
-    ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
-    if (suspend) {
-        if (index >= 0) {
-            desc = mSuspendedEffects.valueAt(index);
+            lseek(teeFd, (off_t) 4, SEEK_SET);
+            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
+            write(teeFd, &temp, sizeof(temp));
+            lseek(teeFd, (off_t) 40, SEEK_SET);
+            temp =  total * channelCount * sizeof(short);
+            write(teeFd, &temp, sizeof(temp));
+            close(teeFd);
+            fdprintf(fd, "FastMixer tee copied to %s\n", teePath);
         } else {
-            desc = new SuspendedEffectDesc();
-            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
-            ALOGV("setEffectSuspendedAll_l() add entry for 0");
-        }
-        if (desc->mRefCount++ == 0) {
-            Vector< sp<EffectModule> > effects;
-            getSuspendEligibleEffects(effects);
-            for (size_t i = 0; i < effects.size(); i++) {
-                setEffectSuspended_l(&effects[i]->desc().type, true);
-            }
-        }
-    } else {
-        if (index < 0) {
-            return;
-        }
-        desc = mSuspendedEffects.valueAt(index);
-        if (desc->mRefCount <= 0) {
-            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
-            desc->mRefCount = 1;
-        }
-        if (--desc->mRefCount == 0) {
-            Vector<const effect_uuid_t *> types;
-            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
-                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
-                    continue;
-                }
-                types.add(&mSuspendedEffects.valueAt(i)->mType);
-            }
-            for (size_t i = 0; i < types.size(); i++) {
-                setEffectSuspended_l(types[i], false);
-            }
-            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
-            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
+            fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno));
         }
     }
 }
 
-
-// The volume effect is used for automated tests only
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
-                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
-const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
-#endif //OPENSL_ES_H_
-
-bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
-{
-    // auxiliary effects and visualizer are never suspended on output mix
-    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
-        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
-         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
-         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
-        return false;
-    }
-    return true;
-}
-
-void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
-{
-    effects.clear();
-    for (size_t i = 0; i < mEffects.size(); i++) {
-        if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
-            effects.add(mEffects[i]);
-        }
-    }
-}
-
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
-                                                            const effect_uuid_t *type)
-{
-    sp<EffectModule> effect = getEffectFromType_l(type);
-    return effect != 0 && effect->isEnabled() ? effect : 0;
-}
-
-void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
-                                                            bool enabled)
-{
-    ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
-    if (enabled) {
-        if (index < 0) {
-            // if the effect is not suspend check if all effects are suspended
-            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
-            if (index < 0) {
-                return;
-            }
-            if (!isEffectEligibleForSuspend(effect->desc())) {
-                return;
-            }
-            setEffectSuspended_l(&effect->desc().type, enabled);
-            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
-            if (index < 0) {
-                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
-                return;
-            }
-        }
-        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
-            effect->desc().type.timeLow);
-        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
-        // if effect is requested to suspended but was not yet enabled, supend it now.
-        if (desc->mEffect == 0) {
-            desc->mEffect = effect;
-            effect->setEnabled(false);
-            effect->setSuspended(true);
-        }
-    } else {
-        if (index < 0) {
-            return;
-        }
-        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
-            effect->desc().type.timeLow);
-        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
-        desc->mEffect.clear();
-        effect->setSuspended(false);
-    }
-}
-
-#undef LOG_TAG
-#define LOG_TAG "AudioFlinger"
-
 // ----------------------------------------------------------------------------
 
 status_t AudioFlinger::onTransact(
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 49e2b2c..46a8e0f 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -75,6 +75,11 @@
 
 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
 
+#define MAX_GAIN 4096.0f
+#define MAX_GAIN_INT 0x1000
+
+#define INCLUDING_FROM_AUDIOFLINGER_H
+
 class AudioFlinger :
     public BinderService<AudioFlinger>,
     public BnAudioFlinger
@@ -92,8 +97,8 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
-                                IAudioFlinger::track_flags_t flags,
+                                size_t frameCount,
+                                IAudioFlinger::track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
@@ -106,7 +111,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 IAudioFlinger::track_flags_t flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -174,7 +179,7 @@
 
     virtual status_t setVoiceVolume(float volume);
 
-    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
                                        audio_io_handle_t output) const;
 
     virtual     unsigned int  getInputFramesLost(audio_io_handle_t ioHandle) const;
@@ -207,8 +212,8 @@
 
     virtual audio_module_handle_t loadHwModule(const char *name);
 
-    virtual int32_t getPrimaryOutputSamplingRate();
-    virtual int32_t getPrimaryOutputFrameCount();
+    virtual uint32_t getPrimaryOutputSamplingRate();
+    virtual size_t getPrimaryOutputFrameCount();
 
     virtual     status_t    onTransact(
                                 uint32_t code,
@@ -269,19 +274,28 @@
     virtual                 ~AudioFlinger();
 
     // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
-    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
+    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
+                                                        NO_INIT : NO_ERROR; }
 
     // RefBase
     virtual     void        onFirstRef();
 
-    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices);
+    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
+                                                audio_devices_t devices);
     void                    purgeStaleEffects_l();
 
     // standby delay for MIXER and DUPLICATING playback threads is read from property
     // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
     static nsecs_t          mStandbyTimeInNsecs;
 
+    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
+    // AudioFlinger::setParameters() updates, other threads read w/o lock
+    static uint32_t         mScreenState;
+
     // Internal dump utilities.
+    static const int kDumpLockRetries = 50;
+    static const int kDumpLockSleepUs = 20000;
+    static bool dumpTryLock(Mutex& mutex);
     void dumpPermissionDenial(int fd, const Vector<String16>& args);
     void dumpClients(int fd, const Vector<String16>& args);
     void dumpInternals(int fd, const Vector<String16>& args);
@@ -346,409 +360,6 @@
     struct AudioStreamOut;
     struct AudioStreamIn;
 
-    class ThreadBase : public Thread {
-    public:
-
-        enum type_t {
-            MIXER,              // Thread class is MixerThread
-            DIRECT,             // Thread class is DirectOutputThread
-            DUPLICATING,        // Thread class is DuplicatingThread
-            RECORD              // Thread class is RecordThread
-        };
-
-        ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
-                    audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
-        virtual             ~ThreadBase();
-
-        void dumpBase(int fd, const Vector<String16>& args);
-        void dumpEffectChains(int fd, const Vector<String16>& args);
-
-        void clearPowerManager();
-
-        // base for record and playback
-        class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
-
-        public:
-            enum track_state {
-                IDLE,
-                TERMINATED,
-                FLUSHED,
-                STOPPED,
-                // next 2 states are currently used for fast tracks only
-                STOPPING_1,     // waiting for first underrun
-                STOPPING_2,     // waiting for presentation complete
-                RESUMING,
-                ACTIVE,
-                PAUSING,
-                PAUSED
-            };
-
-                                TrackBase(ThreadBase *thread,
-                                        const sp<Client>& client,
-                                        uint32_t sampleRate,
-                                        audio_format_t format,
-                                        audio_channel_mask_t channelMask,
-                                        int frameCount,
-                                        const sp<IMemory>& sharedBuffer,
-                                        int sessionId);
-            virtual             ~TrackBase();
-
-            virtual status_t    start(AudioSystem::sync_event_t event,
-                                     int triggerSession) = 0;
-            virtual void        stop() = 0;
-                    sp<IMemory> getCblk() const { return mCblkMemory; }
-                    audio_track_cblk_t* cblk() const { return mCblk; }
-                    int         sessionId() const { return mSessionId; }
-            virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
-
-        protected:
-                                TrackBase(const TrackBase&);
-                                TrackBase& operator = (const TrackBase&);
-
-            // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
-            virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
-
-            // ExtendedAudioBufferProvider interface is only needed for Track,
-            // but putting it in TrackBase avoids the complexity of virtual inheritance
-            virtual size_t  framesReady() const { return SIZE_MAX; }
-
-            audio_format_t format() const {
-                return mFormat;
-            }
-
-            int channelCount() const { return mChannelCount; }
-
-            audio_channel_mask_t channelMask() const { return mChannelMask; }
-
-            int sampleRate() const; // FIXME inline after cblk sr moved
-
-            // Return a pointer to the start of a contiguous slice of the track buffer.
-            // Parameter 'offset' is the requested start position, expressed in
-            // monotonically increasing frame units relative to the track epoch.
-            // Parameter 'frames' is the requested length, also in frame units.
-            // Always returns non-NULL.  It is the caller's responsibility to
-            // verify that this will be successful; the result of calling this
-            // function with invalid 'offset' or 'frames' is undefined.
-            void* getBuffer(uint32_t offset, uint32_t frames) const;
-
-            bool isStopped() const {
-                return (mState == STOPPED || mState == FLUSHED);
-            }
-
-            // for fast tracks only
-            bool isStopping() const {
-                return mState == STOPPING_1 || mState == STOPPING_2;
-            }
-            bool isStopping_1() const {
-                return mState == STOPPING_1;
-            }
-            bool isStopping_2() const {
-                return mState == STOPPING_2;
-            }
-
-            bool isTerminated() const {
-                return mState == TERMINATED;
-            }
-
-            bool step();
-            void reset();
-
-            const wp<ThreadBase> mThread;
-            /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
-            sp<IMemory>         mCblkMemory;
-            audio_track_cblk_t* mCblk;
-            void*               mBuffer;    // start of track buffer, typically in shared memory
-            void*               mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize
-                                            //   is based on mChannelCount and 16-bit samples
-            uint32_t            mFrameCount;
-            // we don't really need a lock for these
-            track_state         mState;
-            const uint32_t      mSampleRate;    // initial sample rate only; for tracks which
-                                // support dynamic rates, the current value is in control block
-            const audio_format_t mFormat;
-            bool                mStepServerFailed;
-            const int           mSessionId;
-            uint8_t             mChannelCount;
-            audio_channel_mask_t mChannelMask;
-            Vector < sp<SyncEvent> >mSyncEvents;
-        };
-
-        enum {
-            CFG_EVENT_IO,
-            CFG_EVENT_PRIO
-        };
-
-        class ConfigEvent {
-        public:
-            ConfigEvent(int type) : mType(type) {}
-            virtual ~ConfigEvent() {}
-
-                     int type() const { return mType; }
-
-            virtual  void dump(char *buffer, size_t size) = 0;
-
-        private:
-            const int mType;
-        };
-
-        class IoConfigEvent : public ConfigEvent {
-        public:
-            IoConfigEvent(int event, int param) :
-                ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
-            virtual ~IoConfigEvent() {}
-
-                    int event() const { return mEvent; }
-                    int param() const { return mParam; }
-
-            virtual  void dump(char *buffer, size_t size) {
-                snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam);
-            }
-
-        private:
-            const int mEvent;
-            const int mParam;
-        };
-
-        class PrioConfigEvent : public ConfigEvent {
-        public:
-            PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) :
-                ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {}
-            virtual ~PrioConfigEvent() {}
-
-                    pid_t pid() const { return mPid; }
-                    pid_t tid() const { return mTid; }
-                    int32_t prio() const { return mPrio; }
-
-            virtual  void dump(char *buffer, size_t size) {
-                snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio);
-            }
-
-        private:
-            const pid_t mPid;
-            const pid_t mTid;
-            const int32_t mPrio;
-        };
-
-
-        class PMDeathRecipient : public IBinder::DeathRecipient {
-        public:
-                        PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
-            virtual     ~PMDeathRecipient() {}
-
-            // IBinder::DeathRecipient
-            virtual     void        binderDied(const wp<IBinder>& who);
-
-        private:
-                        PMDeathRecipient(const PMDeathRecipient&);
-                        PMDeathRecipient& operator = (const PMDeathRecipient&);
-
-            wp<ThreadBase> mThread;
-        };
-
-        virtual     status_t    initCheck() const = 0;
-
-                    // static externally-visible
-                    type_t      type() const { return mType; }
-                    audio_io_handle_t id() const { return mId;}
-
-                    // dynamic externally-visible
-                    uint32_t    sampleRate() const { return mSampleRate; }
-                    int         channelCount() const { return mChannelCount; }
-                    audio_channel_mask_t channelMask() const { return mChannelMask; }
-                    audio_format_t format() const { return mFormat; }
-                    // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
-                    // and returns the normal mix buffer's frame count.
-                    size_t      frameCount() const { return mNormalFrameCount; }
-                    // Return's the HAL's frame count i.e. fast mixer buffer size.
-                    size_t      frameCountHAL() const { return mFrameCount; }
-
-        // Should be "virtual status_t requestExitAndWait()" and override same
-        // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
-                    void        exit();
-        virtual     bool        checkForNewParameters_l() = 0;
-        virtual     status_t    setParameters(const String8& keyValuePairs);
-        virtual     String8     getParameters(const String8& keys) = 0;
-        virtual     void        audioConfigChanged_l(int event, int param = 0) = 0;
-                    void        sendIoConfigEvent(int event, int param = 0);
-                    void        sendIoConfigEvent_l(int event, int param = 0);
-                    void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
-                    void        processConfigEvents();
-
-                    // see note at declaration of mStandby, mOutDevice and mInDevice
-                    bool        standby() const { return mStandby; }
-                    audio_devices_t outDevice() const { return mOutDevice; }
-                    audio_devices_t inDevice() const { return mInDevice; }
-
-        virtual     audio_stream_t* stream() const = 0;
-
-                    sp<EffectHandle> createEffect_l(
-                                        const sp<AudioFlinger::Client>& client,
-                                        const sp<IEffectClient>& effectClient,
-                                        int32_t priority,
-                                        int sessionId,
-                                        effect_descriptor_t *desc,
-                                        int *enabled,
-                                        status_t *status);
-                    void disconnectEffect(const sp< EffectModule>& effect,
-                                          EffectHandle *handle,
-                                          bool unpinIfLast);
-
-                    // return values for hasAudioSession (bit field)
-                    enum effect_state {
-                        EFFECT_SESSION = 0x1,   // the audio session corresponds to at least one
-                                                // effect
-                        TRACK_SESSION = 0x2     // the audio session corresponds to at least one
-                                                // track
-                    };
-
-                    // get effect chain corresponding to session Id.
-                    sp<EffectChain> getEffectChain(int sessionId);
-                    // same as getEffectChain() but must be called with ThreadBase mutex locked
-                    sp<EffectChain> getEffectChain_l(int sessionId) const;
-                    // add an effect chain to the chain list (mEffectChains)
-        virtual     status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
-                    // remove an effect chain from the chain list (mEffectChains)
-        virtual     size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
-                    // lock all effect chains Mutexes. Must be called before releasing the
-                    // ThreadBase mutex before processing the mixer and effects. This guarantees the
-                    // integrity of the chains during the process.
-                    // Also sets the parameter 'effectChains' to current value of mEffectChains.
-                    void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
-                    // unlock effect chains after process
-                    void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
-                    // set audio mode to all effect chains
-                    void setMode(audio_mode_t mode);
-                    // get effect module with corresponding ID on specified audio session
-                    sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId);
-                    sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
-                    // add and effect module. Also creates the effect chain is none exists for
-                    // the effects audio session
-                    status_t addEffect_l(const sp< EffectModule>& effect);
-                    // remove and effect module. Also removes the effect chain is this was the last
-                    // effect
-                    void removeEffect_l(const sp< EffectModule>& effect);
-                    // detach all tracks connected to an auxiliary effect
-        virtual     void detachAuxEffect_l(int effectId) {}
-                    // returns either EFFECT_SESSION if effects on this audio session exist in one
-                    // chain, or TRACK_SESSION if tracks on this audio session exist, or both
-                    virtual uint32_t hasAudioSession(int sessionId) const = 0;
-                    // the value returned by default implementation is not important as the
-                    // strategy is only meaningful for PlaybackThread which implements this method
-                    virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
-
-                    // suspend or restore effect according to the type of effect passed. a NULL
-                    // type pointer means suspend all effects in the session
-                    void setEffectSuspended(const effect_uuid_t *type,
-                                            bool suspend,
-                                            int sessionId = AUDIO_SESSION_OUTPUT_MIX);
-                    // check if some effects must be suspended/restored when an effect is enabled
-                    // or disabled
-                    void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
-                                                     bool enabled,
-                                                     int sessionId = AUDIO_SESSION_OUTPUT_MIX);
-                    void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
-                                                       bool enabled,
-                                                       int sessionId = AUDIO_SESSION_OUTPUT_MIX);
-
-                    virtual status_t    setSyncEvent(const sp<SyncEvent>& event) = 0;
-                    virtual bool        isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
-
-
-        mutable     Mutex                   mLock;
-
-    protected:
-
-                    // entry describing an effect being suspended in mSuspendedSessions keyed vector
-                    class SuspendedSessionDesc : public RefBase {
-                    public:
-                        SuspendedSessionDesc() : mRefCount(0) {}
-
-                        int mRefCount;          // number of active suspend requests
-                        effect_uuid_t mType;    // effect type UUID
-                    };
-
-                    void        acquireWakeLock();
-                    void        acquireWakeLock_l();
-                    void        releaseWakeLock();
-                    void        releaseWakeLock_l();
-                    void setEffectSuspended_l(const effect_uuid_t *type,
-                                              bool suspend,
-                                              int sessionId);
-                    // updated mSuspendedSessions when an effect suspended or restored
-                    void        updateSuspendedSessions_l(const effect_uuid_t *type,
-                                                          bool suspend,
-                                                          int sessionId);
-                    // check if some effects must be suspended when an effect chain is added
-                    void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
-
-        virtual     void        preExit() { }
-
-        friend class AudioFlinger;      // for mEffectChains
-
-                    const type_t            mType;
-
-                    // Used by parameters, config events, addTrack_l, exit
-                    Condition               mWaitWorkCV;
-
-                    const sp<AudioFlinger>  mAudioFlinger;
-                    uint32_t                mSampleRate;
-                    size_t                  mFrameCount;       // output HAL, direct output, record
-                    size_t                  mNormalFrameCount; // normal mixer and effects
-                    audio_channel_mask_t    mChannelMask;
-                    uint16_t                mChannelCount;
-                    size_t                  mFrameSize;
-                    audio_format_t          mFormat;
-
-                    // Parameter sequence by client: binder thread calling setParameters():
-                    //  1. Lock mLock
-                    //  2. Append to mNewParameters
-                    //  3. mWaitWorkCV.signal
-                    //  4. mParamCond.waitRelative with timeout
-                    //  5. read mParamStatus
-                    //  6. mWaitWorkCV.signal
-                    //  7. Unlock
-                    //
-                    // Parameter sequence by server: threadLoop calling checkForNewParameters_l():
-                    // 1. Lock mLock
-                    // 2. If there is an entry in mNewParameters proceed ...
-                    // 2. Read first entry in mNewParameters
-                    // 3. Process
-                    // 4. Remove first entry from mNewParameters
-                    // 5. Set mParamStatus
-                    // 6. mParamCond.signal
-                    // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus)
-                    // 8. Unlock
-                    Condition               mParamCond;
-                    Vector<String8>         mNewParameters;
-                    status_t                mParamStatus;
-
-                    Vector<ConfigEvent *>     mConfigEvents;
-
-                    // These fields are written and read by thread itself without lock or barrier,
-                    // and read by other threads without lock or barrier via standby() , outDevice()
-                    // and inDevice().
-                    // Because of the absence of a lock or barrier, any other thread that reads
-                    // these fields must use the information in isolation, or be prepared to deal
-                    // with possibility that it might be inconsistent with other information.
-                    bool                    mStandby;   // Whether thread is currently in standby.
-                    audio_devices_t         mOutDevice;   // output device
-                    audio_devices_t         mInDevice;    // input device
-                    audio_source_t          mAudioSource; // (see audio.h, audio_source_t)
-
-                    const audio_io_handle_t mId;
-                    Vector< sp<EffectChain> > mEffectChains;
-
-                    static const int        kNameLength = 16;   // prctl(PR_SET_NAME) limit
-                    char                    mName[kNameLength];
-                    sp<IPowerManager>       mPowerManager;
-                    sp<IBinder>             mWakeLockToken;
-                    const sp<PMDeathRecipient> mDeathRecipient;
-                    // list of suspended effects per session and per type. The first vector is
-                    // keyed by session ID, the second by type UUID timeLow field
-                    KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > >  mSuspendedSessions;
-    };
-
     struct  stream_type_t {
         stream_type_t()
             :   volume(1.0f),
@@ -760,619 +371,50 @@
     };
 
     // --- PlaybackThread ---
-    class PlaybackThread : public ThreadBase {
+
+#include "Threads.h"
+
+#include "Effects.h"
+
+    // server side of the client's IAudioTrack
+    class TrackHandle : public android::BnAudioTrack {
     public:
-
-        enum mixer_state {
-            MIXER_IDLE,             // no active tracks
-            MIXER_TRACKS_ENABLED,   // at least one active track, but no track has any data ready
-            MIXER_TRACKS_READY      // at least one active track, and at least one track has data
-            // standby mode does not have an enum value
-            // suspend by audio policy manager is orthogonal to mixer state
-        };
-
-        // playback track
-        class Track : public TrackBase, public VolumeProvider {
-        public:
-                                Track(  PlaybackThread *thread,
-                                        const sp<Client>& client,
-                                        audio_stream_type_t streamType,
-                                        uint32_t sampleRate,
-                                        audio_format_t format,
-                                        audio_channel_mask_t channelMask,
-                                        int frameCount,
-                                        const sp<IMemory>& sharedBuffer,
-                                        int sessionId,
-                                        IAudioFlinger::track_flags_t flags);
-            virtual             ~Track();
-
-            static  void        appendDumpHeader(String8& result);
-                    void        dump(char* buffer, size_t size);
-            virtual status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
-                                     int triggerSession = 0);
-            virtual void        stop();
-                    void        pause();
-
-                    void        flush();
-                    void        destroy();
-                    void        mute(bool);
-                    int         name() const { return mName; }
-
-                    audio_stream_type_t streamType() const {
-                        return mStreamType;
-                    }
-                    status_t    attachAuxEffect(int EffectId);
-                    void        setAuxBuffer(int EffectId, int32_t *buffer);
-                    int32_t     *auxBuffer() const { return mAuxBuffer; }
-                    void        setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
-                    int16_t     *mainBuffer() const { return mMainBuffer; }
-                    int         auxEffectId() const { return mAuxEffectId; }
-
-        // implement FastMixerState::VolumeProvider interface
-            virtual uint32_t    getVolumeLR();
-            virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
-
-        protected:
-            // for numerous
-            friend class PlaybackThread;
-            friend class MixerThread;
-            friend class DirectOutputThread;
-
-                                Track(const Track&);
-                                Track& operator = (const Track&);
-
-            // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
-            // releaseBuffer() not overridden
-
-            virtual size_t framesReady() const;
-
-            bool isMuted() const { return mMute; }
-            bool isPausing() const {
-                return mState == PAUSING;
-            }
-            bool isPaused() const {
-                return mState == PAUSED;
-            }
-            bool isResuming() const {
-                return mState == RESUMING;
-            }
-            bool isReady() const;
-            void setPaused() { mState = PAUSED; }
-            void reset();
-
-            bool isOutputTrack() const {
-                return (mStreamType == AUDIO_STREAM_CNT);
-            }
-
-            sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
-
-            bool presentationComplete(size_t framesWritten, size_t audioHalFrames);
-
-        public:
-            void triggerEvents(AudioSystem::sync_event_t type);
-            virtual bool isTimedTrack() const { return false; }
-            bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; }
-
-        protected:
-
-            // written by Track::mute() called by binder thread(s), without a mutex or barrier.
-            // read by Track::isMuted() called by playback thread, also without a mutex or barrier.
-            // The lack of mutex or barrier is safe because the mute status is only used by itself.
-            bool                mMute;
-
-            // FILLED state is used for suppressing volume ramp at begin of playing
-            enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE};
-            mutable uint8_t     mFillingUpStatus;
-            int8_t              mRetryCount;
-            const sp<IMemory>   mSharedBuffer;
-            bool                mResetDone;
-            const audio_stream_type_t mStreamType;
-            int                 mName;      // track name on the normal mixer,
-                                            // allocated statically at track creation time,
-                                            // and is even allocated (though unused) for fast tracks
-                                            // FIXME don't allocate track name for fast tracks
-            int16_t             *mMainBuffer;
-            int32_t             *mAuxBuffer;
-            int                 mAuxEffectId;
-            bool                mHasVolumeController;
-            size_t              mPresentationCompleteFrames; // number of frames written to the audio HAL
-                                                       // when this track will be fully rendered
-        private:
-            IAudioFlinger::track_flags_t mFlags;
-
-            // The following fields are only for fast tracks, and should be in a subclass
-            int                 mFastIndex; // index within FastMixerState::mFastTracks[];
-                                            // either mFastIndex == -1 if not isFastTrack()
-                                            // or 0 < mFastIndex < FastMixerState::kMaxFast because
-                                            // index 0 is reserved for normal mixer's submix;
-                                            // index is allocated statically at track creation time
-                                            // but the slot is only used if track is active
-            FastTrackUnderruns  mObservedUnderruns; // Most recently observed value of
-                                            // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns
-            uint32_t            mUnderrunCount; // Counter of total number of underruns, never reset
-            volatile float      mCachedVolume;  // combined master volume and stream type volume;
-                                                // 'volatile' means accessed without lock or
-                                                // barrier, but is read/written atomically
-        };  // end of Track
-
-        class TimedTrack : public Track {
-          public:
-            static sp<TimedTrack> create(PlaybackThread *thread,
-                                         const sp<Client>& client,
-                                         audio_stream_type_t streamType,
-                                         uint32_t sampleRate,
-                                         audio_format_t format,
-                                         audio_channel_mask_t channelMask,
-                                         int frameCount,
-                                         const sp<IMemory>& sharedBuffer,
-                                         int sessionId);
-            virtual ~TimedTrack();
-
-            class TimedBuffer {
-              public:
-                TimedBuffer();
-                TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
-                const sp<IMemory>& buffer() const { return mBuffer; }
-                int64_t pts() const { return mPTS; }
-                uint32_t position() const { return mPosition; }
-                void setPosition(uint32_t pos) { mPosition = pos; }
-              private:
-                sp<IMemory> mBuffer;
-                int64_t     mPTS;
-                uint32_t    mPosition;
-            };
-
-            // Mixer facing methods.
-            virtual bool isTimedTrack() const { return true; }
-            virtual size_t framesReady() const;
-
-            // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
-                                           int64_t pts);
-            virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
-
-            // Client/App facing methods.
-            status_t    allocateTimedBuffer(size_t size,
-                                            sp<IMemory>* buffer);
-            status_t    queueTimedBuffer(const sp<IMemory>& buffer,
-                                         int64_t pts);
-            status_t    setMediaTimeTransform(const LinearTransform& xform,
-                                              TimedAudioTrack::TargetTimeline target);
-
-          private:
-            TimedTrack(PlaybackThread *thread,
-                       const sp<Client>& client,
-                       audio_stream_type_t streamType,
-                       uint32_t sampleRate,
-                       audio_format_t format,
-                       audio_channel_mask_t channelMask,
-                       int frameCount,
-                       const sp<IMemory>& sharedBuffer,
-                       int sessionId);
-
-            void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
-            void timedYieldSilence_l(uint32_t numFrames,
-                                     AudioBufferProvider::Buffer* buffer);
-            void trimTimedBufferQueue_l();
-            void trimTimedBufferQueueHead_l(const char* logTag);
-            void updateFramesPendingAfterTrim_l(const TimedBuffer& buf,
-                                                const char* logTag);
-
-            uint64_t            mLocalTimeFreq;
-            LinearTransform     mLocalTimeToSampleTransform;
-            LinearTransform     mMediaTimeToSampleTransform;
-            sp<MemoryDealer>    mTimedMemoryDealer;
-
-            Vector<TimedBuffer> mTimedBufferQueue;
-            bool                mQueueHeadInFlight;
-            bool                mTrimQueueHeadOnRelease;
-            uint32_t            mFramesPendingInQueue;
-
-            uint8_t*            mTimedSilenceBuffer;
-            uint32_t            mTimedSilenceBufferSize;
-            mutable Mutex       mTimedBufferQueueLock;
-            bool                mTimedAudioOutputOnTime;
-            CCHelper            mCCHelper;
-
-            Mutex               mMediaTimeTransformLock;
-            LinearTransform     mMediaTimeTransform;
-            bool                mMediaTimeTransformValid;
-            TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
-        };
-
-
-        // playback track
-        class OutputTrack : public Track {
-        public:
-
-            class Buffer: public AudioBufferProvider::Buffer {
-            public:
-                int16_t *mBuffer;
-            };
-
-                                OutputTrack(PlaybackThread *thread,
-                                        DuplicatingThread *sourceThread,
-                                        uint32_t sampleRate,
-                                        audio_format_t format,
-                                        audio_channel_mask_t channelMask,
-                                        int frameCount);
-            virtual             ~OutputTrack();
-
-            virtual status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
-                                     int triggerSession = 0);
-            virtual void        stop();
-                    bool        write(int16_t* data, uint32_t frames);
-                    bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
-                    bool        isActive() const { return mActive; }
-            const wp<ThreadBase>& thread() const { return mThread; }
-
-        private:
-
-            enum {
-                NO_MORE_BUFFERS = 0x80000001,   // same in AudioTrack.h, ok to be different value
-            };
-
-            status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
-            void                clearBufferQueue();
-
-            // Maximum number of pending buffers allocated by OutputTrack::write()
-            static const uint8_t kMaxOverFlowBuffers = 10;
-
-            Vector < Buffer* >          mBufferQueue;
-            AudioBufferProvider::Buffer mOutBuffer;
-            bool                        mActive;
-            DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
-        };  // end of OutputTrack
-
-        PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                        audio_io_handle_t id, audio_devices_t device, type_t type);
-        virtual             ~PlaybackThread();
-
-                    void        dump(int fd, const Vector<String16>& args);
-
-        // Thread virtuals
-        virtual     status_t    readyToRun();
-        virtual     bool        threadLoop();
-
-        // RefBase
-        virtual     void        onFirstRef();
-
-protected:
-        // Code snippets that were lifted up out of threadLoop()
-        virtual     void        threadLoop_mix() = 0;
-        virtual     void        threadLoop_sleepTime() = 0;
-        virtual     void        threadLoop_write();
-        virtual     void        threadLoop_standby();
-        virtual     void        threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
-
-                    // prepareTracks_l reads and writes mActiveTracks, and returns
-                    // the pending set of tracks to remove via Vector 'tracksToRemove'.  The caller
-                    // is responsible for clearing or destroying this Vector later on, when it
-                    // is safe to do so. That will drop the final ref count and destroy the tracks.
-        virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
-
-        // ThreadBase virtuals
-        virtual     void        preExit();
-
-public:
-
-        virtual     status_t    initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
-
-                    // return estimated latency in milliseconds, as reported by HAL
-                    uint32_t    latency() const;
-                    // same, but lock must already be held
-                    uint32_t    latency_l() const;
-
-                    void        setMasterVolume(float value);
-                    void        setMasterMute(bool muted);
-
-                    void        setStreamVolume(audio_stream_type_t stream, float value);
-                    void        setStreamMute(audio_stream_type_t stream, bool muted);
-
-                    float       streamVolume(audio_stream_type_t stream) const;
-
-                    sp<Track>   createTrack_l(
-                                    const sp<AudioFlinger::Client>& client,
-                                    audio_stream_type_t streamType,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    int frameCount,
-                                    const sp<IMemory>& sharedBuffer,
-                                    int sessionId,
-                                    IAudioFlinger::track_flags_t flags,
-                                    pid_t tid,
-                                    status_t *status);
-
-                    AudioStreamOut* getOutput() const;
-                    AudioStreamOut* clearOutput();
-                    virtual audio_stream_t* stream() const;
-
-                    // a very large number of suspend() will eventually wraparound, but unlikely
-                    void        suspend() { (void) android_atomic_inc(&mSuspended); }
-                    void        restore()
-                                    {
-                                        // if restore() is done without suspend(), get back into
-                                        // range so that the next suspend() will operate correctly
-                                        if (android_atomic_dec(&mSuspended) <= 0) {
-                                            android_atomic_release_store(0, &mSuspended);
-                                        }
-                                    }
-                    bool        isSuspended() const
-                                    { return android_atomic_acquire_load(&mSuspended) > 0; }
-
-        virtual     String8     getParameters(const String8& keys);
-        virtual     void        audioConfigChanged_l(int event, int param = 0);
-                    status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
-                    int16_t     *mixBuffer() const { return mMixBuffer; };
-
-        virtual     void detachAuxEffect_l(int effectId);
-                    status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
-                            int EffectId);
-                    status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track,
-                            int EffectId);
-
-                    virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
-                    virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
-                    virtual uint32_t hasAudioSession(int sessionId) const;
-                    virtual uint32_t getStrategyForSession_l(int sessionId);
-
-
-                    virtual status_t setSyncEvent(const sp<SyncEvent>& event);
-                    virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
-                            void     invalidateTracks(audio_stream_type_t streamType);
-
-
-    protected:
-        int16_t*                        mMixBuffer;
-
-        // suspend count, > 0 means suspended.  While suspended, the thread continues to pull from
-        // tracks and mix, but doesn't write to HAL.  A2DP and SCO HAL implementations can't handle
-        // concurrent use of both of them, so Audio Policy Service suspends one of the threads to
-        // workaround that restriction.
-        // 'volatile' means accessed via atomic operations and no lock.
-        volatile int32_t                mSuspended;
-
-        int                             mBytesWritten;
+                            TrackHandle(const sp<PlaybackThread::Track>& track);
+        virtual             ~TrackHandle();
+        virtual sp<IMemory> getCblk() const;
+        virtual status_t    start();
+        virtual void        stop();
+        virtual void        flush();
+        virtual void        mute(bool);
+        virtual void        pause();
+        virtual status_t    attachAuxEffect(int effectId);
+        virtual status_t    allocateTimedBuffer(size_t size,
+                                                sp<IMemory>* buffer);
+        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
+                                             int64_t pts);
+        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
+                                                  int target);
+        virtual status_t onTransact(
+            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
-        // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
-        // PlaybackThread needs to find out if master-muted, it checks it's local
-        // copy rather than the one in AudioFlinger.  This optimization saves a lock.
-        bool                            mMasterMute;
-                    void        setMasterMute_l(bool muted) { mMasterMute = muted; }
-    protected:
-        SortedVector< wp<Track> >       mActiveTracks;  // FIXME check if this could be sp<>
-
-        // Allocate a track name for a given channel mask.
-        //   Returns name >= 0 if successful, -1 on failure.
-        virtual int             getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
-        virtual void            deleteTrackName_l(int name) = 0;
-
-        // Time to sleep between cycles when:
-        virtual uint32_t        activeSleepTimeUs() const;      // mixer state MIXER_TRACKS_ENABLED
-        virtual uint32_t        idleSleepTimeUs() const = 0;    // mixer state MIXER_IDLE
-        virtual uint32_t        suspendSleepTimeUs() const = 0; // audio policy manager suspended us
-        // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
-        // No sleep in standby mode; waits on a condition
-
-        // Code snippets that are temporarily lifted up out of threadLoop() until the merge
-                    void        checkSilentMode_l();
-
-        // Non-trivial for DUPLICATING only
-        virtual     void        saveOutputTracks() { }
-        virtual     void        clearOutputTracks() { }
-
-        // Cache various calculated values, at threadLoop() entry and after a parameter change
-        virtual     void        cacheParameters_l();
-
-        virtual     uint32_t    correctLatency(uint32_t latency) const;
-
-    private:
-
-        friend class AudioFlinger;      // for numerous
-
-        PlaybackThread(const Client&);
-        PlaybackThread& operator = (const PlaybackThread&);
-
-        status_t    addTrack_l(const sp<Track>& track);
-        void        destroyTrack_l(const sp<Track>& track);
-        void        removeTrack_l(const sp<Track>& track);
-
-        void        readOutputParameters();
-
-        virtual void dumpInternals(int fd, const Vector<String16>& args);
-        void        dumpTracks(int fd, const Vector<String16>& args);
-
-        SortedVector< sp<Track> >       mTracks;
-        // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
-        stream_type_t                   mStreamTypes[AUDIO_STREAM_CNT + 1];
-        AudioStreamOut                  *mOutput;
-
-        float                           mMasterVolume;
-        nsecs_t                         mLastWriteTime;
-        int                             mNumWrites;
-        int                             mNumDelayedWrites;
-        bool                            mInWrite;
-
-        // FIXME rename these former local variables of threadLoop to standard "m" names
-        nsecs_t                         standbyTime;
-        size_t                          mixBufferSize;
-
-        // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
-        uint32_t                        activeSleepTime;
-        uint32_t                        idleSleepTime;
-
-        uint32_t                        sleepTime;
-
-        // mixer status returned by prepareTracks_l()
-        mixer_state                     mMixerStatus; // current cycle
-                                                      // previous cycle when in prepareTracks_l()
-        mixer_state                     mMixerStatusIgnoringFastTracks;
-                                                      // FIXME or a separate ready state per track
-
-        // FIXME move these declarations into the specific sub-class that needs them
-        // MIXER only
-        uint32_t                        sleepTimeShift;
-
-        // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
-        nsecs_t                         standbyDelay;
-
-        // MIXER only
-        nsecs_t                         maxPeriod;
-
-        // DUPLICATING only
-        uint32_t                        writeFrames;
-
-    private:
-        // The HAL output sink is treated as non-blocking, but current implementation is blocking
-        sp<NBAIO_Sink>          mOutputSink;
-        // If a fast mixer is present, the blocking pipe sink, otherwise clear
-        sp<NBAIO_Sink>          mPipeSink;
-        // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
-        sp<NBAIO_Sink>          mNormalSink;
-        // For dumpsys
-        sp<NBAIO_Sink>          mTeeSink;
-        sp<NBAIO_Source>        mTeeSource;
-        uint32_t                mScreenState;   // cached copy of gScreenState
-    public:
-        virtual     bool        hasFastMixer() const = 0;
-        virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
-                                    { FastTrackUnderruns dummy; return dummy; }
-
-    protected:
-                    // accessed by both binder threads and within threadLoop(), lock on mutex needed
-                    unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
-
+        const sp<PlaybackThread::Track> mTrack;
     };
 
-    class MixerThread : public PlaybackThread {
+    // server side of the client's IAudioRecord
+    class RecordHandle : public android::BnAudioRecord {
     public:
-        MixerThread (const sp<AudioFlinger>& audioFlinger,
-                     AudioStreamOut* output,
-                     audio_io_handle_t id,
-                     audio_devices_t device,
-                     type_t type = MIXER);
-        virtual             ~MixerThread();
-
-        // Thread virtuals
-
-        virtual     bool        checkForNewParameters_l();
-        virtual     void        dumpInternals(int fd, const Vector<String16>& args);
-
-    protected:
-        virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
-        virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
-        virtual     void        deleteTrackName_l(int name);
-        virtual     uint32_t    idleSleepTimeUs() const;
-        virtual     uint32_t    suspendSleepTimeUs() const;
-        virtual     void        cacheParameters_l();
-
-        // threadLoop snippets
-        virtual     void        threadLoop_write();
-        virtual     void        threadLoop_standby();
-        virtual     void        threadLoop_mix();
-        virtual     void        threadLoop_sleepTime();
-        virtual     void        threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
-        virtual     uint32_t    correctLatency(uint32_t latency) const;
-
-                    AudioMixer* mAudioMixer;    // normal mixer
+        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
+        virtual             ~RecordHandle();
+        virtual sp<IMemory> getCblk() const;
+        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
+        virtual void        stop();
+        virtual status_t onTransact(
+            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
-                    // one-time initialization, no locks required
-                    FastMixer*  mFastMixer;         // non-NULL if there is also a fast mixer
-                    sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
+        const sp<RecordThread::RecordTrack> mRecordTrack;
 
-                    // contents are not guaranteed to be consistent, no locks required
-                    FastMixerDumpState mFastMixerDumpState;
-#ifdef STATE_QUEUE_DUMP
-                    StateQueueObserverDump mStateQueueObserverDump;
-                    StateQueueMutatorDump  mStateQueueMutatorDump;
-#endif
-                    AudioWatchdogDump mAudioWatchdogDump;
-
-                    // accessible only within the threadLoop(), no locks required
-                    //          mFastMixer->sq()    // for mutating and pushing state
-                    int32_t     mFastMixerFutex;    // for cold idle
-
-    public:
-        virtual     bool        hasFastMixer() const { return mFastMixer != NULL; }
-        virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
-                                  ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks);
-                                  return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
-                                }
-    };
-
-    class DirectOutputThread : public PlaybackThread {
-    public:
-
-        DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                            audio_io_handle_t id, audio_devices_t device);
-        virtual                 ~DirectOutputThread();
-
-        // Thread virtuals
-
-        virtual     bool        checkForNewParameters_l();
-
-    protected:
-        virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
-        virtual     void        deleteTrackName_l(int name);
-        virtual     uint32_t    activeSleepTimeUs() const;
-        virtual     uint32_t    idleSleepTimeUs() const;
-        virtual     uint32_t    suspendSleepTimeUs() const;
-        virtual     void        cacheParameters_l();
-
-        // threadLoop snippets
-        virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
-        virtual     void        threadLoop_mix();
-        virtual     void        threadLoop_sleepTime();
-
-        // volumes last sent to audio HAL with stream->set_volume()
-        float mLeftVolFloat;
-        float mRightVolFloat;
-
-private:
-        // prepareTracks_l() tells threadLoop_mix() the name of the single active track
-        sp<Track>               mActiveTrack;
-    public:
-        virtual     bool        hasFastMixer() const { return false; }
-    };
-
-    class DuplicatingThread : public MixerThread {
-    public:
-        DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
-                           audio_io_handle_t id);
-        virtual                 ~DuplicatingThread();
-
-        // Thread virtuals
-                    void        addOutputTrack(MixerThread* thread);
-                    void        removeOutputTrack(MixerThread* thread);
-                    uint32_t    waitTimeMs() const { return mWaitTimeMs; }
-    protected:
-        virtual     uint32_t    activeSleepTimeUs() const;
-
-    private:
-                    bool        outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
-    protected:
-        // threadLoop snippets
-        virtual     void        threadLoop_mix();
-        virtual     void        threadLoop_sleepTime();
-        virtual     void        threadLoop_write();
-        virtual     void        threadLoop_standby();
-        virtual     void        cacheParameters_l();
-
-    private:
-        // called from threadLoop, addOutputTrack, removeOutputTrack
-        virtual     void        updateWaitTime_l();
-    protected:
-        virtual     void        saveOutputTracks();
-        virtual     void        clearOutputTracks();
-    private:
-
-                    uint32_t    mWaitTimeMs;
-        SortedVector < sp<OutputTrack> >  outputTracks;
-        SortedVector < sp<OutputTrack> >  mOutputTracks;
-    public:
-        virtual     bool        hasFastMixer() const { return false; }
+        // for use from destructor
+        void                stop_nonvirtual();
     };
 
               PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
@@ -1399,536 +441,10 @@
 
               sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
 
-    // server side of the client's IAudioTrack
-    class TrackHandle : public android::BnAudioTrack {
-    public:
-                            TrackHandle(const sp<PlaybackThread::Track>& track);
-        virtual             ~TrackHandle();
-        virtual sp<IMemory> getCblk() const;
-        virtual status_t    start();
-        virtual void        stop();
-        virtual void        flush();
-        virtual void        mute(bool);
-        virtual void        pause();
-        virtual status_t    attachAuxEffect(int effectId);
-        virtual status_t    allocateTimedBuffer(size_t size,
-                                                sp<IMemory>* buffer);
-        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
-                                             int64_t pts);
-        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
-                                                  int target);
-        virtual status_t onTransact(
-            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
-    private:
-        const sp<PlaybackThread::Track> mTrack;
-    };
 
                 void        removeClient_l(pid_t pid);
                 void        removeNotificationClient(pid_t pid);
 
-
-    // record thread
-    class RecordThread : public ThreadBase, public AudioBufferProvider
-                            // derives from AudioBufferProvider interface for use by resampler
-    {
-    public:
-
-        // record track
-        class RecordTrack : public TrackBase {
-        public:
-                                RecordTrack(RecordThread *thread,
-                                        const sp<Client>& client,
-                                        uint32_t sampleRate,
-                                        audio_format_t format,
-                                        audio_channel_mask_t channelMask,
-                                        int frameCount,
-                                        int sessionId);
-            virtual             ~RecordTrack();
-
-            virtual status_t    start(AudioSystem::sync_event_t event, int triggerSession);
-            virtual void        stop();
-
-                    void        destroy();
-
-                    // clear the buffer overflow flag
-                    void        clearOverflow() { mOverflow = false; }
-                    // set the buffer overflow flag and return previous value
-                    bool        setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
-
-            static  void        appendDumpHeader(String8& result);
-                    void        dump(char* buffer, size_t size);
-
-        private:
-            friend class AudioFlinger;  // for mState
-
-                                RecordTrack(const RecordTrack&);
-                                RecordTrack& operator = (const RecordTrack&);
-
-            // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
-            // releaseBuffer() not overridden
-
-            bool                mOverflow;  // overflow on most recent attempt to fill client buffer
-        };
-
-                RecordThread(const sp<AudioFlinger>& audioFlinger,
-                        AudioStreamIn *input,
-                        uint32_t sampleRate,
-                        audio_channel_mask_t channelMask,
-                        audio_io_handle_t id,
-                        audio_devices_t device);
-                virtual     ~RecordThread();
-
-        // no addTrack_l ?
-        void        destroyTrack_l(const sp<RecordTrack>& track);
-        void        removeTrack_l(const sp<RecordTrack>& track);
-
-        void        dumpInternals(int fd, const Vector<String16>& args);
-        void        dumpTracks(int fd, const Vector<String16>& args);
-
-        // Thread virtuals
-        virtual bool        threadLoop();
-        virtual status_t    readyToRun();
-
-        // RefBase
-        virtual void        onFirstRef();
-
-        virtual status_t    initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
-                sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
-                        const sp<AudioFlinger::Client>& client,
-                        uint32_t sampleRate,
-                        audio_format_t format,
-                        audio_channel_mask_t channelMask,
-                        int frameCount,
-                        int sessionId,
-                        IAudioFlinger::track_flags_t flags,
-                        pid_t tid,
-                        status_t *status);
-
-                status_t    start(RecordTrack* recordTrack,
-                                  AudioSystem::sync_event_t event,
-                                  int triggerSession);
-
-                // ask the thread to stop the specified track, and
-                // return true if the caller should then do it's part of the stopping process
-                bool        stop_l(RecordTrack* recordTrack);
-
-                void        dump(int fd, const Vector<String16>& args);
-                AudioStreamIn* clearInput();
-                virtual audio_stream_t* stream() const;
-
-        // AudioBufferProvider interface
-        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
-        virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
-
-        virtual bool        checkForNewParameters_l();
-        virtual String8     getParameters(const String8& keys);
-        virtual void        audioConfigChanged_l(int event, int param = 0);
-                void        readInputParameters();
-        virtual unsigned int  getInputFramesLost();
-
-        virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
-        virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
-        virtual uint32_t hasAudioSession(int sessionId) const;
-
-                // Return the set of unique session IDs across all tracks.
-                // The keys are the session IDs, and the associated values are meaningless.
-                // FIXME replace by Set [and implement Bag/Multiset for other uses].
-                KeyedVector<int, bool> sessionIds() const;
-
-        virtual status_t setSyncEvent(const sp<SyncEvent>& event);
-        virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
-
-        static void syncStartEventCallback(const wp<SyncEvent>& event);
-               void handleSyncStartEvent(const sp<SyncEvent>& event);
-
-    private:
-                void clearSyncStartEvent();
-
-                // Enter standby if not already in standby, and set mStandby flag
-                void standby();
-
-                // Call the HAL standby method unconditionally, and don't change mStandby flag
-                void inputStandBy();
-
-                AudioStreamIn                       *mInput;
-                SortedVector < sp<RecordTrack> >    mTracks;
-                // mActiveTrack has dual roles:  it indicates the current active track, and
-                // is used together with mStartStopCond to indicate start()/stop() progress
-                sp<RecordTrack>                     mActiveTrack;
-                Condition                           mStartStopCond;
-                AudioResampler                      *mResampler;
-                int32_t                             *mRsmpOutBuffer;
-                int16_t                             *mRsmpInBuffer;
-                size_t                              mRsmpInIndex;
-                size_t                              mInputBytes;
-                const int                           mReqChannelCount;
-                const uint32_t                      mReqSampleRate;
-                ssize_t                             mBytesRead;
-                // sync event triggering actual audio capture. Frames read before this event will
-                // be dropped and therefore not read by the application.
-                sp<SyncEvent>                       mSyncStartEvent;
-                // number of captured frames to drop after the start sync event has been received.
-                // when < 0, maximum frames to drop before starting capture even if sync event is
-                // not received
-                ssize_t                             mFramestoDrop;
-    };
-
-    // server side of the client's IAudioRecord
-    class RecordHandle : public android::BnAudioRecord {
-    public:
-        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
-        virtual             ~RecordHandle();
-        virtual sp<IMemory> getCblk() const;
-        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
-        virtual void        stop();
-        virtual status_t onTransact(
-            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
-    private:
-        const sp<RecordThread::RecordTrack> mRecordTrack;
-
-        // for use from destructor
-        void                stop_nonvirtual();
-    };
-
-    //--- Audio Effect Management
-
-    // EffectModule and EffectChain classes both have their own mutex to protect
-    // state changes or resource modifications. Always respect the following order
-    // if multiple mutexes must be acquired to avoid cross deadlock:
-    // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
-
-    // The EffectModule class is a wrapper object controlling the effect engine implementation
-    // in the effect library. It prevents concurrent calls to process() and command() functions
-    // from different client threads. It keeps a list of EffectHandle objects corresponding
-    // to all client applications using this effect and notifies applications of effect state,
-    // control or parameter changes. It manages the activation state machine to send appropriate
-    // reset, enable, disable commands to effect engine and provide volume
-    // ramping when effects are activated/deactivated.
-    // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
-    // the attached track(s) to accumulate their auxiliary channel.
-    class EffectModule: public RefBase {
-    public:
-        EffectModule(ThreadBase *thread,
-                        const wp<AudioFlinger::EffectChain>& chain,
-                        effect_descriptor_t *desc,
-                        int id,
-                        int sessionId);
-        virtual ~EffectModule();
-
-        enum effect_state {
-            IDLE,
-            RESTART,
-            STARTING,
-            ACTIVE,
-            STOPPING,
-            STOPPED,
-            DESTROYED
-        };
-
-        int         id() const { return mId; }
-        void process();
-        void updateState();
-        status_t command(uint32_t cmdCode,
-                         uint32_t cmdSize,
-                         void *pCmdData,
-                         uint32_t *replySize,
-                         void *pReplyData);
-
-        void reset_l();
-        status_t configure();
-        status_t init();
-        effect_state state() const {
-            return mState;
-        }
-        uint32_t status() {
-            return mStatus;
-        }
-        int sessionId() const {
-            return mSessionId;
-        }
-        status_t    setEnabled(bool enabled);
-        status_t    setEnabled_l(bool enabled);
-        bool isEnabled() const;
-        bool isProcessEnabled() const;
-
-        void        setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
-        int16_t     *inBuffer() { return mConfig.inputCfg.buffer.s16; }
-        void        setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; }
-        int16_t     *outBuffer() { return mConfig.outputCfg.buffer.s16; }
-        void        setChain(const wp<EffectChain>& chain) { mChain = chain; }
-        void        setThread(const wp<ThreadBase>& thread) { mThread = thread; }
-        const wp<ThreadBase>& thread() { return mThread; }
-
-        status_t addHandle(EffectHandle *handle);
-        size_t disconnect(EffectHandle *handle, bool unpinIfLast);
-        size_t removeHandle(EffectHandle *handle);
-
-        const effect_descriptor_t& desc() const { return mDescriptor; }
-        wp<EffectChain>&     chain() { return mChain; }
-
-        status_t         setDevice(audio_devices_t device);
-        status_t         setVolume(uint32_t *left, uint32_t *right, bool controller);
-        status_t         setMode(audio_mode_t mode);
-        status_t         setAudioSource(audio_source_t source);
-        status_t         start();
-        status_t         stop();
-        void             setSuspended(bool suspended);
-        bool             suspended() const;
-
-        EffectHandle*    controlHandle_l();
-
-        bool             isPinned() const { return mPinned; }
-        void             unPin() { mPinned = false; }
-        bool             purgeHandles();
-        void             lock() { mLock.lock(); }
-        void             unlock() { mLock.unlock(); }
-
-        void             dump(int fd, const Vector<String16>& args);
-
-    protected:
-        friend class AudioFlinger;      // for mHandles
-        bool                mPinned;
-
-        // Maximum time allocated to effect engines to complete the turn off sequence
-        static const uint32_t MAX_DISABLE_TIME_MS = 10000;
-
-        EffectModule(const EffectModule&);
-        EffectModule& operator = (const EffectModule&);
-
-        status_t start_l();
-        status_t stop_l();
-
-mutable Mutex               mLock;      // mutex for process, commands and handles list protection
-        wp<ThreadBase>      mThread;    // parent thread
-        wp<EffectChain>     mChain;     // parent effect chain
-        const int           mId;        // this instance unique ID
-        const int           mSessionId; // audio session ID
-        const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
-        effect_config_t     mConfig;    // input and output audio configuration
-        effect_handle_t  mEffectInterface; // Effect module C API
-        status_t            mStatus;    // initialization status
-        effect_state        mState;     // current activation state
-        Vector<EffectHandle *> mHandles;    // list of client handles
-                    // First handle in mHandles has highest priority and controls the effect module
-        uint32_t mMaxDisableWaitCnt;    // maximum grace period before forcing an effect off after
-                                        // sending disable command.
-        uint32_t mDisableWaitCnt;       // current process() calls count during disable period.
-        bool     mSuspended;            // effect is suspended: temporarily disabled by framework
-    };
-
-    // The EffectHandle class implements the IEffect interface. It provides resources
-    // to receive parameter updates, keeps track of effect control
-    // ownership and state and has a pointer to the EffectModule object it is controlling.
-    // There is one EffectHandle object for each application controlling (or using)
-    // an effect module.
-    // The EffectHandle is obtained by calling AudioFlinger::createEffect().
-    class EffectHandle: public android::BnEffect {
-    public:
-
-        EffectHandle(const sp<EffectModule>& effect,
-                const sp<AudioFlinger::Client>& client,
-                const sp<IEffectClient>& effectClient,
-                int32_t priority);
-        virtual ~EffectHandle();
-
-        // IEffect
-        virtual status_t enable();
-        virtual status_t disable();
-        virtual status_t command(uint32_t cmdCode,
-                                 uint32_t cmdSize,
-                                 void *pCmdData,
-                                 uint32_t *replySize,
-                                 void *pReplyData);
-        virtual void disconnect();
-    private:
-                void disconnect(bool unpinIfLast);
-    public:
-        virtual sp<IMemory> getCblk() const { return mCblkMemory; }
-        virtual status_t onTransact(uint32_t code, const Parcel& data,
-                Parcel* reply, uint32_t flags);
-
-
-        // Give or take control of effect module
-        // - hasControl: true if control is given, false if removed
-        // - signal: true client app should be signaled of change, false otherwise
-        // - enabled: state of the effect when control is passed
-        void setControl(bool hasControl, bool signal, bool enabled);
-        void commandExecuted(uint32_t cmdCode,
-                             uint32_t cmdSize,
-                             void *pCmdData,
-                             uint32_t replySize,
-                             void *pReplyData);
-        void setEnabled(bool enabled);
-        bool enabled() const { return mEnabled; }
-
-        // Getters
-        int id() const { return mEffect->id(); }
-        int priority() const { return mPriority; }
-        bool hasControl() const { return mHasControl; }
-        sp<EffectModule> effect() const { return mEffect; }
-        // destroyed_l() must be called with the associated EffectModule mLock held
-        bool destroyed_l() const { return mDestroyed; }
-
-        void dump(char* buffer, size_t size);
-
-    protected:
-        friend class AudioFlinger;          // for mEffect, mHasControl, mEnabled
-        EffectHandle(const EffectHandle&);
-        EffectHandle& operator =(const EffectHandle&);
-
-        sp<EffectModule> mEffect;           // pointer to controlled EffectModule
-        sp<IEffectClient> mEffectClient;    // callback interface for client notifications
-        /*const*/ sp<Client> mClient;       // client for shared memory allocation, see disconnect()
-        sp<IMemory>         mCblkMemory;    // shared memory for control block
-        effect_param_cblk_t* mCblk;         // control block for deferred parameter setting via shared memory
-        uint8_t*            mBuffer;        // pointer to parameter area in shared memory
-        int mPriority;                      // client application priority to control the effect
-        bool mHasControl;                   // true if this handle is controlling the effect
-        bool mEnabled;                      // cached enable state: needed when the effect is
-                                            // restored after being suspended
-        bool mDestroyed;                    // Set to true by destructor. Access with EffectModule
-                                            // mLock held
-    };
-
-    // the EffectChain class represents a group of effects associated to one audio session.
-    // There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
-    // The EffecChain with session ID 0 contains global effects applied to the output mix.
-    // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks)
-    // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding
-    // in the effect process order. When attached to a track (session ID != 0), it also provide it's own
-    // input buffer used by the track as accumulation buffer.
-    class EffectChain: public RefBase {
-    public:
-        EffectChain(const wp<ThreadBase>& wThread, int sessionId);
-        EffectChain(ThreadBase *thread, int sessionId);
-        virtual ~EffectChain();
-
-        // special key used for an entry in mSuspendedEffects keyed vector
-        // corresponding to a suspend all request.
-        static const int        kKeyForSuspendAll = 0;
-
-        // minimum duration during which we force calling effect process when last track on
-        // a session is stopped or removed to allow effect tail to be rendered
-        static const int        kProcessTailDurationMs = 1000;
-
-        void process_l();
-
-        void lock() {
-            mLock.lock();
-        }
-        void unlock() {
-            mLock.unlock();
-        }
-
-        status_t addEffect_l(const sp<EffectModule>& handle);
-        size_t removeEffect_l(const sp<EffectModule>& handle);
-
-        int sessionId() const { return mSessionId; }
-        void setSessionId(int sessionId) { mSessionId = sessionId; }
-
-        sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
-        sp<EffectModule> getEffectFromId_l(int id);
-        sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
-        bool setVolume_l(uint32_t *left, uint32_t *right);
-        void setDevice_l(audio_devices_t device);
-        void setMode_l(audio_mode_t mode);
-        void setAudioSource_l(audio_source_t source);
-
-        void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
-            mInBuffer = buffer;
-            mOwnInBuffer = ownsBuffer;
-        }
-        int16_t *inBuffer() const {
-            return mInBuffer;
-        }
-        void setOutBuffer(int16_t *buffer) {
-            mOutBuffer = buffer;
-        }
-        int16_t *outBuffer() const {
-            return mOutBuffer;
-        }
-
-        void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
-        void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
-        int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); }
-
-        void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
-                                   mTailBufferCount = mMaxTailBuffers; }
-        void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
-        int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); }
-
-        uint32_t strategy() const { return mStrategy; }
-        void setStrategy(uint32_t strategy)
-                { mStrategy = strategy; }
-
-        // suspend effect of the given type
-        void setEffectSuspended_l(const effect_uuid_t *type,
-                                  bool suspend);
-        // suspend all eligible effects
-        void setEffectSuspendedAll_l(bool suspend);
-        // check if effects should be suspend or restored when a given effect is enable or disabled
-        void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
-                                              bool enabled);
-
-        void clearInputBuffer();
-
-        void dump(int fd, const Vector<String16>& args);
-
-    protected:
-        friend class AudioFlinger;  // for mThread, mEffects
-        EffectChain(const EffectChain&);
-        EffectChain& operator =(const EffectChain&);
-
-        class SuspendedEffectDesc : public RefBase {
-        public:
-            SuspendedEffectDesc() : mRefCount(0) {}
-
-            int mRefCount;
-            effect_uuid_t mType;
-            wp<EffectModule> mEffect;
-        };
-
-        // get a list of effect modules to suspend when an effect of the type
-        // passed is enabled.
-        void                       getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
-
-        // get an effect module if it is currently enable
-        sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
-        // true if the effect whose descriptor is passed can be suspended
-        // OEMs can modify the rules implemented in this method to exclude specific effect
-        // types or implementations from the suspend/restore mechanism.
-        bool isEffectEligibleForSuspend(const effect_descriptor_t& desc);
-
-        void clearInputBuffer_l(sp<ThreadBase> thread);
-
-        wp<ThreadBase> mThread;     // parent mixer thread
-        Mutex mLock;                // mutex protecting effect list
-        Vector< sp<EffectModule> > mEffects; // list of effect modules
-        int mSessionId;             // audio session ID
-        int16_t *mInBuffer;         // chain input buffer
-        int16_t *mOutBuffer;        // chain output buffer
-
-        // 'volatile' here means these are accessed with atomic operations instead of mutex
-        volatile int32_t mActiveTrackCnt;    // number of active tracks connected
-        volatile int32_t mTrackCnt;          // number of tracks connected
-
-        int32_t mTailBufferCount;   // current effect tail buffer count
-        int32_t mMaxTailBuffers;    // maximum effect tail buffers
-        bool mOwnInBuffer;          // true if the chain owns its input buffer
-        int mVolumeCtrlIdx;         // index of insert effect having control over volume
-        uint32_t mLeftVolume;       // previous volume on left channel
-        uint32_t mRightVolume;      // previous volume on right channel
-        uint32_t mNewLeftVolume;       // new volume on left channel
-        uint32_t mNewRightVolume;      // new volume on right channel
-        uint32_t mStrategy; // strategy for this effect chain
-        // mSuspendedEffects lists all effects currently suspended in the chain.
-        // Use effect type UUID timelow field as key. There is no real risk of identical
-        // timeLow fields among effect type UUIDs.
-        // Updated by updateSuspendedSessions_l() only.
-        KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
-    };
-
     class AudioHwDevice {
     public:
         enum Flags {
@@ -2064,8 +580,16 @@
     // for use from destructor
     status_t    closeOutput_nonvirtual(audio_io_handle_t output);
     status_t    closeInput_nonvirtual(audio_io_handle_t input);
+
+    // all record threads serially share a common tee sink, which is re-created on format change
+    sp<NBAIO_Sink>   mRecordTeeSink;
+    sp<NBAIO_Source> mRecordTeeSource;
+
+public:
+    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
 };
 
+#undef INCLUDING_FROM_AUDIOFLINGER_H
 
 // ----------------------------------------------------------------------------
 
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index af169d5..b3ca877 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -106,8 +106,16 @@
     ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
             maxNumTracks, MAX_NUM_TRACKS);
 
+    // AudioMixer is not yet capable of more than 32 active track inputs
+    ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
+
+    // AudioMixer is not yet capable of multi-channel output beyond stereo
+    ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS);
+
     LocalClock lc;
 
+    pthread_once(&sOnceControl, &sInitRoutine);
+
     mState.enabledTracks= 0;
     mState.needsChanged = 0;
     mState.frameCount   = frameCount;
@@ -121,8 +129,6 @@
     // and mTrackNames is initially 0.  However, leave it here until that's verified.
     track_t* t = mState.tracks;
     for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
-        // FIXME redundant per track
-        t->localTimeFreq = lc.getLocalFreq();
         t->resampler = NULL;
         t->downmixerBufferProvider = NULL;
         t++;
@@ -192,7 +198,6 @@
         t->sessionId = sessionId;
         // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
         t->bufferProvider = NULL;
-        t->downmixerBufferProvider = NULL;
         t->buffer.raw = NULL;
         // no initialization needed
         // t->buffer.frameCount
@@ -203,7 +208,7 @@
         // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
         t->mainBuffer = NULL;
         t->auxBuffer = NULL;
-        // see t->localTimeFreq in constructor above
+        t->downmixerBufferProvider = NULL;
 
         status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
         if (status == OK) {
@@ -556,7 +561,7 @@
                         // the resampler sees the number of channels after the downmixer, if any
                         downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
                         devSampleRate, quality);
-                resampler->setLocalTimeFreq(localTimeFreq);
+                resampler->setLocalTimeFreq(sLocalTimeFreq);
             }
             return true;
         }
@@ -760,7 +765,8 @@
 }
 
 
-void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
+        int32_t* temp, int32_t* aux)
 {
     t->resampler->setSampleRate(t->sampleRate);
 
@@ -793,11 +799,13 @@
     }
 }
 
-void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp,
+        int32_t* aux)
 {
 }
 
-void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+        int32_t* aux)
 {
     int32_t vl = t->prevVolume[0];
     int32_t vr = t->prevVolume[1];
@@ -839,7 +847,8 @@
     t->adjustVolumeRamp(aux != NULL);
 }
 
-void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+        int32_t* aux)
 {
     const int16_t vl = t->volume[0];
     const int16_t vr = t->volume[1];
@@ -867,7 +876,8 @@
     }
 }
 
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+        int32_t* aux)
 {
     const int16_t *in = static_cast<const int16_t *>(t->in);
 
@@ -957,7 +967,8 @@
     t->in = in;
 }
 
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+        int32_t* aux)
 {
     const int16_t *in = static_cast<int16_t const *>(t->in);
 
@@ -1142,7 +1153,8 @@
                 while (outFrames) {
                     size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
                     if (inFrames) {
-                        t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
+                        t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
+                                state->resampleTemp, aux);
                         t.frameCount -= inFrames;
                         outFrames -= inFrames;
                         if (CC_UNLIKELY(aux != NULL)) {
@@ -1151,7 +1163,8 @@
                     }
                     if (t.frameCount == 0 && outFrames) {
                         t.bufferProvider->releaseBuffer(&t.buffer);
-                        t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
+                        t.buffer.frameCount = (state->frameCount - numFrames) -
+                                (BLOCKSIZE - outFrames);
                         int64_t outputPTS = calculateOutputPTS(
                             t, pts, numFrames + (BLOCKSIZE - outFrames));
                         t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
@@ -1241,7 +1254,8 @@
                     if (CC_UNLIKELY(aux != NULL)) {
                         aux += outFrames;
                     }
-                    t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
+                    t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount,
+                            state->resampleTemp, aux);
                     outFrames += t.buffer.frameCount;
                     t.bufferProvider->releaseBuffer(&t.buffer);
                 }
@@ -1281,7 +1295,8 @@
         // been enabled for mixing.
         if (in == NULL || ((unsigned long)in & 3)) {
             memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
-            ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x",
+            ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: "
+                                              "buffer %p track %d, channels %d, needs %08x",
                     in, i, t.channelCount, t.needs);
             return;
         }
@@ -1423,7 +1438,16 @@
     if (AudioBufferProvider::kInvalidPTS == basePTS)
         return AudioBufferProvider::kInvalidPTS;
 
-    return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+    return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
+}
+
+/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
+/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
+
+/*static*/ void AudioMixer::sInitRoutine()
+{
+    LocalClock lc;
+    sLocalTimeFreq = lc.getLocalFreq();
 }
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 6333357..fd21fda 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -41,8 +41,15 @@
 
     /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
 
+
+    // This mixer has a hard-coded upper limit of 32 active track inputs.
+    // Adding support for > 32 tracks would require more than simply changing this value.
     static const uint32_t MAX_NUM_TRACKS = 32;
     // maximum number of channels supported by the mixer
+
+    // This mixer has a hard-coded upper limit of 2 channels for output.
+    // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
+    // Adding support for > 2 channel output would require more than simply changing this value.
     static const uint32_t MAX_NUM_CHANNELS = 2;
     // maximum number of channels supported for the content
     static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
@@ -139,7 +146,8 @@
     struct track_t;
     class DownmixerBufferProvider;
 
-    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
+                           int32_t* aux);
     static const int BLOCKSIZE = 16; // 4 cache lines
 
     struct track_t {
@@ -188,12 +196,12 @@
 
         // 16-byte boundary
 
-        uint64_t    localTimeFreq;
-
         DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
 
         int32_t     sessionId;
 
+        int32_t     padding[2];
+
         // 16-byte boundary
 
         bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
@@ -254,12 +262,17 @@
     static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
     static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
 
-    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
     static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
+    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
+            int32_t* aux);
+    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+            int32_t* aux);
+    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+            int32_t* aux);
 
     static void process__validate(state_t* state, int64_t pts);
     static void process__nop(state_t* state, int64_t pts);
@@ -274,6 +287,10 @@
 
     static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
                                       int outputFrameIndex);
+
+    static uint64_t         sLocalTimeFreq;
+    static pthread_once_t   sOnceControl;
+    static void             sInitRoutine();
 };
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 8b99bd2..b86d3ae 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -145,7 +145,7 @@
         return BAD_VALUE;
     }
 
-    ALOGV("setDeviceConnectionState() tid %d", gettid());
+    ALOGV("setDeviceConnectionState()");
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device,
                                                       state, device_address);
@@ -174,7 +174,7 @@
         return BAD_VALUE;
     }
 
-    ALOGV("setPhoneState() tid %d", gettid());
+    ALOGV("setPhoneState()");
 
     // TODO: check if it is more appropriate to do it in platform specific policy manager
     AudioSystem::setMode(state);
@@ -199,7 +199,7 @@
     if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) {
         return BAD_VALUE;
     }
-    ALOGV("setForceUse() tid %d", gettid());
+    ALOGV("setForceUse()");
     Mutex::Autolock _l(mLock);
     mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config);
     return NO_ERROR;
@@ -225,9 +225,10 @@
     if (mpAudioPolicy == NULL) {
         return 0;
     }
-    ALOGV("getOutput() tid %d", gettid());
+    ALOGV("getOutput()");
     Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, flags);
+    return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask,
+                                        flags);
 }
 
 status_t AudioPolicyService::startOutput(audio_io_handle_t output,
@@ -237,7 +238,7 @@
     if (mpAudioPolicy == NULL) {
         return NO_INIT;
     }
-    ALOGV("startOutput() tid %d", gettid());
+    ALOGV("startOutput()");
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session);
 }
@@ -249,7 +250,7 @@
     if (mpAudioPolicy == NULL) {
         return NO_INIT;
     }
-    ALOGV("stopOutput() tid %d", gettid());
+    ALOGV("stopOutput()");
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session);
 }
@@ -259,7 +260,7 @@
     if (mpAudioPolicy == NULL) {
         return;
     }
-    ALOGV("releaseOutput() tid %d", gettid());
+    ALOGV("releaseOutput()");
     Mutex::Autolock _l(mLock);
     mpAudioPolicy->release_output(mpAudioPolicy, output);
 }
@@ -280,7 +281,7 @@
     Mutex::Autolock _l(mLock);
     // the audio_in_acoustics_t parameter is ignored by get_input()
     audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
-                                                       format, channelMask, (audio_in_acoustics_t) 0);
+                                                   format, channelMask, (audio_in_acoustics_t) 0);
 
     if (input == 0) {
         return input;
@@ -533,7 +534,7 @@
 }
 
 void AudioPolicyService::binderDied(const wp<IBinder>& who) {
-    ALOGW("binderDied() %p, tid %d, calling pid %d", who.unsafe_get(), gettid(),
+    ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(),
             IPCThreadState::self()->getCallingPid());
 }
 
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 63f9549..92653c1 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -142,11 +142,11 @@
             status_t dumpInternals(int fd);
 
     // Thread used for tone playback and to send audio config commands to audio flinger
-    // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone()
-    // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause
-    // calls to AudioPolicyService and an attempt to lock mLock.
-    // For audio config commands, it is necessary because audio flinger requires that the calling process (user)
-    // has permission to modify audio settings.
+    // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because
+    // startTone() and stopTone() are normally called with mLock locked and requesting a tone start
+    // or stop will cause calls to AudioPolicyService and an attempt to lock mLock.
+    // For audio config commands, it is necessary because audio flinger requires that the calling
+    // process (user) has permission to modify audio settings.
     class AudioCommandThread : public Thread {
         class AudioCommand;
     public:
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
new file mode 100644
index 0000000..74ba59e
--- /dev/null
+++ b/services/audioflinger/Effects.cpp
@@ -0,0 +1,1684 @@
+/*
+**
+** 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.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <audio_effects/effect_visualizer.h>
+#include <audio_utils/primitives.h>
+#include <private/media/AudioEffectShared.h>
+#include <media/EffectsFactoryApi.h>
+
+#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message.  In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on.  Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+//  EffectModule implementation
+// ----------------------------------------------------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectModule"
+
+AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
+                                        const wp<AudioFlinger::EffectChain>& chain,
+                                        effect_descriptor_t *desc,
+                                        int id,
+                                        int sessionId)
+    : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX),
+      mThread(thread), mChain(chain), mId(id), mSessionId(sessionId),
+      mDescriptor(*desc),
+      // mConfig is set by configure() and not used before then
+      mEffectInterface(NULL),
+      mStatus(NO_INIT), mState(IDLE),
+      // mMaxDisableWaitCnt is set by configure() and not used before then
+      // mDisableWaitCnt is set by process() and updateState() and not used before then
+      mSuspended(false)
+{
+    ALOGV("Constructor %p", this);
+    int lStatus;
+
+    // create effect engine from effect factory
+    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
+
+    if (mStatus != NO_ERROR) {
+        return;
+    }
+    lStatus = init();
+    if (lStatus < 0) {
+        mStatus = lStatus;
+        goto Error;
+    }
+
+    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
+    return;
+Error:
+    EffectRelease(mEffectInterface);
+    mEffectInterface = NULL;
+    ALOGV("Constructor Error %d", mStatus);
+}
+
+AudioFlinger::EffectModule::~EffectModule()
+{
+    ALOGV("Destructor %p", this);
+    if (mEffectInterface != NULL) {
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
+                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
+            sp<ThreadBase> thread = mThread.promote();
+            if (thread != 0) {
+                audio_stream_t *stream = thread->stream();
+                if (stream != NULL) {
+                    stream->remove_audio_effect(stream, mEffectInterface);
+                }
+            }
+        }
+        // release effect engine
+        EffectRelease(mEffectInterface);
+    }
+}
+
+status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle)
+{
+    status_t status;
+
+    Mutex::Autolock _l(mLock);
+    int priority = handle->priority();
+    size_t size = mHandles.size();
+    EffectHandle *controlHandle = NULL;
+    size_t i;
+    for (i = 0; i < size; i++) {
+        EffectHandle *h = mHandles[i];
+        if (h == NULL || h->destroyed_l()) {
+            continue;
+        }
+        // first non destroyed handle is considered in control
+        if (controlHandle == NULL)
+            controlHandle = h;
+        if (h->priority() <= priority) {
+            break;
+        }
+    }
+    // if inserted in first place, move effect control from previous owner to this handle
+    if (i == 0) {
+        bool enabled = false;
+        if (controlHandle != NULL) {
+            enabled = controlHandle->enabled();
+            controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
+        }
+        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
+        status = NO_ERROR;
+    } else {
+        status = ALREADY_EXISTS;
+    }
+    ALOGV("addHandle() %p added handle %p in position %d", this, handle, i);
+    mHandles.insertAt(handle, i);
+    return status;
+}
+
+size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mHandles.size();
+    size_t i;
+    for (i = 0; i < size; i++) {
+        if (mHandles[i] == handle) {
+            break;
+        }
+    }
+    if (i == size) {
+        return size;
+    }
+    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i);
+
+    mHandles.removeAt(i);
+    // if removed from first place, move effect control from this handle to next in line
+    if (i == 0) {
+        EffectHandle *h = controlHandle_l();
+        if (h != NULL) {
+            h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/);
+        }
+    }
+
+    // Prevent calls to process() and other functions on effect interface from now on.
+    // The effect engine will be released by the destructor when the last strong reference on
+    // this object is released which can happen after next process is called.
+    if (mHandles.size() == 0 && !mPinned) {
+        mState = DESTROYED;
+    }
+
+    return mHandles.size();
+}
+
+// must be called with EffectModule::mLock held
+AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l()
+{
+    // the first valid handle in the list has control over the module
+    for (size_t i = 0; i < mHandles.size(); i++) {
+        EffectHandle *h = mHandles[i];
+        if (h != NULL && !h->destroyed_l()) {
+            return h;
+        }
+    }
+
+    return NULL;
+}
+
+size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast)
+{
+    ALOGV("disconnect() %p handle %p", this, handle);
+    // keep a strong reference on this EffectModule to avoid calling the
+    // destructor before we exit
+    sp<EffectModule> keep(this);
+    {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            thread->disconnectEffect(keep, handle, unpinIfLast);
+        }
+    }
+    return mHandles.size();
+}
+
+void AudioFlinger::EffectModule::updateState() {
+    Mutex::Autolock _l(mLock);
+
+    switch (mState) {
+    case RESTART:
+        reset_l();
+        // FALL THROUGH
+
+    case STARTING:
+        // clear auxiliary effect input buffer for next accumulation
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            memset(mConfig.inputCfg.buffer.raw,
+                   0,
+                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
+        }
+        start_l();
+        mState = ACTIVE;
+        break;
+    case STOPPING:
+        stop_l();
+        mDisableWaitCnt = mMaxDisableWaitCnt;
+        mState = STOPPED;
+        break;
+    case STOPPED:
+        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
+        // turn off sequence.
+        if (--mDisableWaitCnt == 0) {
+            reset_l();
+            mState = IDLE;
+        }
+        break;
+    default: //IDLE , ACTIVE, DESTROYED
+        break;
+    }
+}
+
+void AudioFlinger::EffectModule::process()
+{
+    Mutex::Autolock _l(mLock);
+
+    if (mState == DESTROYED || mEffectInterface == NULL ||
+            mConfig.inputCfg.buffer.raw == NULL ||
+            mConfig.outputCfg.buffer.raw == NULL) {
+        return;
+    }
+
+    if (isProcessEnabled()) {
+        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            ditherAndClamp(mConfig.inputCfg.buffer.s32,
+                                        mConfig.inputCfg.buffer.s32,
+                                        mConfig.inputCfg.buffer.frameCount/2);
+        }
+
+        // do the actual processing in the effect engine
+        int ret = (*mEffectInterface)->process(mEffectInterface,
+                                               &mConfig.inputCfg.buffer,
+                                               &mConfig.outputCfg.buffer);
+
+        // force transition to IDLE state when engine is ready
+        if (mState == STOPPED && ret == -ENODATA) {
+            mDisableWaitCnt = 1;
+        }
+
+        // clear auxiliary effect input buffer for next accumulation
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            memset(mConfig.inputCfg.buffer.raw, 0,
+                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
+        }
+    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
+                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
+        // If an insert effect is idle and input buffer is different from output buffer,
+        // accumulate input onto output
+        sp<EffectChain> chain = mChain.promote();
+        if (chain != 0 && chain->activeTrackCnt() != 0) {
+            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
+            int16_t *in = mConfig.inputCfg.buffer.s16;
+            int16_t *out = mConfig.outputCfg.buffer.s16;
+            for (size_t i = 0; i < frameCnt; i++) {
+                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
+            }
+        }
+    }
+}
+
+void AudioFlinger::EffectModule::reset_l()
+{
+    if (mEffectInterface == NULL) {
+        return;
+    }
+    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
+}
+
+status_t AudioFlinger::EffectModule::configure()
+{
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        return DEAD_OBJECT;
+    }
+
+    // TODO: handle configuration of effects replacing track process
+    audio_channel_mask_t channelMask = thread->channelMask();
+
+    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
+    } else {
+        mConfig.inputCfg.channels = channelMask;
+    }
+    mConfig.outputCfg.channels = channelMask;
+    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+    mConfig.inputCfg.samplingRate = thread->sampleRate();
+    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
+    mConfig.inputCfg.bufferProvider.cookie = NULL;
+    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
+    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
+    mConfig.outputCfg.bufferProvider.cookie = NULL;
+    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
+    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
+    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+    // Insert effect:
+    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
+    // always overwrites output buffer: input buffer == output buffer
+    // - in other sessions:
+    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
+    //      other effect: overwrites output buffer: input buffer == output buffer
+    // Auxiliary effect:
+    //      accumulates in output buffer: input buffer != output buffer
+    // Therefore: accumulate <=> input buffer != output buffer
+    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
+        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
+    } else {
+        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
+    }
+    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
+    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
+    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
+    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
+
+    ALOGV("configure() %p thread %p buffer %p framecount %d",
+            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
+
+    status_t cmdStatus;
+    uint32_t size = sizeof(int);
+    status_t status = (*mEffectInterface)->command(mEffectInterface,
+                                                   EFFECT_CMD_SET_CONFIG,
+                                                   sizeof(effect_config_t),
+                                                   &mConfig,
+                                                   &size,
+                                                   &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+
+    if (status == 0 &&
+            (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) {
+        uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+        effect_param_t *p = (effect_param_t *)buf32;
+
+        p->psize = sizeof(uint32_t);
+        p->vsize = sizeof(uint32_t);
+        size = sizeof(int);
+        *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY;
+
+        uint32_t latency = 0;
+        PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId);
+        if (pbt != NULL) {
+            latency = pbt->latency_l();
+        }
+
+        *((int32_t *)p->data + 1)= latency;
+        (*mEffectInterface)->command(mEffectInterface,
+                                     EFFECT_CMD_SET_PARAM,
+                                     sizeof(effect_param_t) + 8,
+                                     &buf32,
+                                     &size,
+                                     &cmdStatus);
+    }
+
+    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
+            (1000 * mConfig.outputCfg.buffer.frameCount);
+
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::init()
+{
+    Mutex::Autolock _l(mLock);
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    uint32_t size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface,
+                                                   EFFECT_CMD_INIT,
+                                                   0,
+                                                   NULL,
+                                                   &size,
+                                                   &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::start()
+{
+    Mutex::Autolock _l(mLock);
+    return start_l();
+}
+
+status_t AudioFlinger::EffectModule::start_l()
+{
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    uint32_t size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface,
+                                                   EFFECT_CMD_ENABLE,
+                                                   0,
+                                                   NULL,
+                                                   &size,
+                                                   &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    if (status == 0 &&
+            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
+             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            audio_stream_t *stream = thread->stream();
+            if (stream != NULL) {
+                stream->add_audio_effect(stream, mEffectInterface);
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::stop()
+{
+    Mutex::Autolock _l(mLock);
+    return stop_l();
+}
+
+status_t AudioFlinger::EffectModule::stop_l()
+{
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    uint32_t size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface,
+                                                   EFFECT_CMD_DISABLE,
+                                                   0,
+                                                   NULL,
+                                                   &size,
+                                                   &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    if (status == 0 &&
+            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
+             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            audio_stream_t *stream = thread->stream();
+            if (stream != NULL) {
+                stream->remove_audio_effect(stream, mEffectInterface);
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
+                                             uint32_t cmdSize,
+                                             void *pCmdData,
+                                             uint32_t *replySize,
+                                             void *pReplyData)
+{
+    Mutex::Autolock _l(mLock);
+    ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
+
+    if (mState == DESTROYED || mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t status = (*mEffectInterface)->command(mEffectInterface,
+                                                   cmdCode,
+                                                   cmdSize,
+                                                   pCmdData,
+                                                   replySize,
+                                                   pReplyData);
+    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
+        uint32_t size = (replySize == NULL) ? 0 : *replySize;
+        for (size_t i = 1; i < mHandles.size(); i++) {
+            EffectHandle *h = mHandles[i];
+            if (h != NULL && !h->destroyed_l()) {
+                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
+{
+    Mutex::Autolock _l(mLock);
+    return setEnabled_l(enabled);
+}
+
+// must be called with EffectModule::mLock held
+status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled)
+{
+
+    ALOGV("setEnabled %p enabled %d", this, enabled);
+
+    if (enabled != isEnabled()) {
+        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
+        if (enabled && status != NO_ERROR) {
+            return status;
+        }
+
+        switch (mState) {
+        // going from disabled to enabled
+        case IDLE:
+            mState = STARTING;
+            break;
+        case STOPPED:
+            mState = RESTART;
+            break;
+        case STOPPING:
+            mState = ACTIVE;
+            break;
+
+        // going from enabled to disabled
+        case RESTART:
+            mState = STOPPED;
+            break;
+        case STARTING:
+            mState = IDLE;
+            break;
+        case ACTIVE:
+            mState = STOPPING;
+            break;
+        case DESTROYED:
+            return NO_ERROR; // simply ignore as we are being destroyed
+        }
+        for (size_t i = 1; i < mHandles.size(); i++) {
+            EffectHandle *h = mHandles[i];
+            if (h != NULL && !h->destroyed_l()) {
+                h->setEnabled(enabled);
+            }
+        }
+    }
+    return NO_ERROR;
+}
+
+bool AudioFlinger::EffectModule::isEnabled() const
+{
+    switch (mState) {
+    case RESTART:
+    case STARTING:
+    case ACTIVE:
+        return true;
+    case IDLE:
+    case STOPPING:
+    case STOPPED:
+    case DESTROYED:
+    default:
+        return false;
+    }
+}
+
+bool AudioFlinger::EffectModule::isProcessEnabled() const
+{
+    switch (mState) {
+    case RESTART:
+    case ACTIVE:
+    case STOPPING:
+    case STOPPED:
+        return true;
+    case IDLE:
+    case STARTING:
+    case DESTROYED:
+    default:
+        return false;
+    }
+}
+
+status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+
+    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
+    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
+    if (isProcessEnabled() &&
+            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
+            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
+        status_t cmdStatus;
+        uint32_t volume[2];
+        uint32_t *pVolume = NULL;
+        uint32_t size = sizeof(volume);
+        volume[0] = *left;
+        volume[1] = *right;
+        if (controller) {
+            pVolume = volume;
+        }
+        status = (*mEffectInterface)->command(mEffectInterface,
+                                              EFFECT_CMD_SET_VOLUME,
+                                              size,
+                                              volume,
+                                              &size,
+                                              pVolume);
+        if (controller && status == NO_ERROR && size == sizeof(volume)) {
+            *left = volume[0];
+            *right = volume[1];
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device)
+{
+    if (device == AUDIO_DEVICE_NONE) {
+        return NO_ERROR;
+    }
+
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
+        status_t cmdStatus;
+        uint32_t size = sizeof(status_t);
+        uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE :
+                            EFFECT_CMD_SET_INPUT_DEVICE;
+        status = (*mEffectInterface)->command(mEffectInterface,
+                                              cmd,
+                                              sizeof(uint32_t),
+                                              &device,
+                                              &size,
+                                              &cmdStatus);
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
+        status_t cmdStatus;
+        uint32_t size = sizeof(status_t);
+        status = (*mEffectInterface)->command(mEffectInterface,
+                                              EFFECT_CMD_SET_AUDIO_MODE,
+                                              sizeof(audio_mode_t),
+                                              &mode,
+                                              &size,
+                                              &cmdStatus);
+        if (status == NO_ERROR) {
+            status = cmdStatus;
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) {
+        uint32_t size = 0;
+        status = (*mEffectInterface)->command(mEffectInterface,
+                                              EFFECT_CMD_SET_AUDIO_SOURCE,
+                                              sizeof(audio_source_t),
+                                              &source,
+                                              &size,
+                                              NULL);
+    }
+    return status;
+}
+
+void AudioFlinger::EffectModule::setSuspended(bool suspended)
+{
+    Mutex::Autolock _l(mLock);
+    mSuspended = suspended;
+}
+
+bool AudioFlinger::EffectModule::suspended() const
+{
+    Mutex::Autolock _l(mLock);
+    return mSuspended;
+}
+
+bool AudioFlinger::EffectModule::purgeHandles()
+{
+    bool enabled = false;
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mHandles.size(); i++) {
+        EffectHandle *handle = mHandles[i];
+        if (handle != NULL && !handle->destroyed_l()) {
+            handle->effect().clear();
+            if (handle->hasControl()) {
+                enabled = handle->enabled();
+            }
+        }
+    }
+    return enabled;
+}
+
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
+    result.append(buffer);
+
+    bool locked = AudioFlinger::dumpTryLock(mLock);
+    // failed to lock - AudioFlinger is probably deadlocked
+    if (!locked) {
+        result.append("\t\tCould not lock Fx mutex:\n");
+    }
+
+    result.append("\t\tSession Status State Engine:\n");
+    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
+            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
+    result.append(buffer);
+
+    result.append("\t\tDescriptor:\n");
+    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
+            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
+            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],
+                    mDescriptor.uuid.node[2],
+            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
+                mDescriptor.type.timeLow, mDescriptor.type.timeMid,
+                    mDescriptor.type.timeHiAndVersion,
+                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],
+                    mDescriptor.type.node[2],
+                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
+            mDescriptor.apiVersion,
+            mDescriptor.flags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- name: %s\n",
+            mDescriptor.name);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
+            mDescriptor.implementor);
+    result.append(buffer);
+
+    result.append("\t\t- Input configuration:\n");
+    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
+    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
+            (uint32_t)mConfig.inputCfg.buffer.raw,
+            mConfig.inputCfg.buffer.frameCount,
+            mConfig.inputCfg.samplingRate,
+            mConfig.inputCfg.channels,
+            mConfig.inputCfg.format);
+    result.append(buffer);
+
+    result.append("\t\t- Output configuration:\n");
+    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
+    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
+            (uint32_t)mConfig.outputCfg.buffer.raw,
+            mConfig.outputCfg.buffer.frameCount,
+            mConfig.outputCfg.samplingRate,
+            mConfig.outputCfg.channels,
+            mConfig.outputCfg.format);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
+    result.append(buffer);
+    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
+    for (size_t i = 0; i < mHandles.size(); ++i) {
+        EffectHandle *handle = mHandles[i];
+        if (handle != NULL && !handle->destroyed_l()) {
+            handle->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    result.append("\n");
+
+    write(fd, result.string(), result.length());
+
+    if (locked) {
+        mLock.unlock();
+    }
+}
+
+// ----------------------------------------------------------------------------
+//  EffectHandle implementation
+// ----------------------------------------------------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectHandle"
+
+AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
+                                        const sp<AudioFlinger::Client>& client,
+                                        const sp<IEffectClient>& effectClient,
+                                        int32_t priority)
+    : BnEffect(),
+    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
+    mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false)
+{
+    ALOGV("constructor %p", this);
+
+    if (client == 0) {
+        return;
+    }
+    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
+    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
+    if (mCblkMemory != 0) {
+        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
+
+        if (mCblk != NULL) {
+            new(mCblk) effect_param_cblk_t();
+            mBuffer = (uint8_t *)mCblk + bufOffset;
+        }
+    } else {
+        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
+                sizeof(effect_param_cblk_t));
+        return;
+    }
+}
+
+AudioFlinger::EffectHandle::~EffectHandle()
+{
+    ALOGV("Destructor %p", this);
+
+    if (mEffect == 0) {
+        mDestroyed = true;
+        return;
+    }
+    mEffect->lock();
+    mDestroyed = true;
+    mEffect->unlock();
+    disconnect(false);
+}
+
+status_t AudioFlinger::EffectHandle::enable()
+{
+    ALOGV("enable %p", this);
+    if (!mHasControl) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) {
+        return DEAD_OBJECT;
+    }
+
+    if (mEnabled) {
+        return NO_ERROR;
+    }
+
+    mEnabled = true;
+
+    sp<ThreadBase> thread = mEffect->thread().promote();
+    if (thread != 0) {
+        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
+    }
+
+    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
+    if (mEffect->suspended()) {
+        return NO_ERROR;
+    }
+
+    status_t status = mEffect->setEnabled(true);
+    if (status != NO_ERROR) {
+        if (thread != 0) {
+            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
+        }
+        mEnabled = false;
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectHandle::disable()
+{
+    ALOGV("disable %p", this);
+    if (!mHasControl) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) {
+        return DEAD_OBJECT;
+    }
+
+    if (!mEnabled) {
+        return NO_ERROR;
+    }
+    mEnabled = false;
+
+    if (mEffect->suspended()) {
+        return NO_ERROR;
+    }
+
+    status_t status = mEffect->setEnabled(false);
+
+    sp<ThreadBase> thread = mEffect->thread().promote();
+    if (thread != 0) {
+        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
+    }
+
+    return status;
+}
+
+void AudioFlinger::EffectHandle::disconnect()
+{
+    disconnect(true);
+}
+
+void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
+{
+    ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
+    if (mEffect == 0) {
+        return;
+    }
+    // restore suspended effects if the disconnected handle was enabled and the last one.
+    if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) {
+        sp<ThreadBase> thread = mEffect->thread().promote();
+        if (thread != 0) {
+            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
+        }
+    }
+
+    // release sp on module => module destructor can be called now
+    mEffect.clear();
+    if (mClient != 0) {
+        if (mCblk != NULL) {
+            // unlike ~TrackBase(), mCblk is never a local new, so don't delete
+            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
+        }
+        mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
+        // Client destructor must run with AudioFlinger mutex locked
+        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+        mClient.clear();
+    }
+}
+
+status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
+                                             uint32_t cmdSize,
+                                             void *pCmdData,
+                                             uint32_t *replySize,
+                                             void *pReplyData)
+{
+    ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
+            cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
+
+    // only get parameter command is permitted for applications not controlling the effect
+    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) {
+        return DEAD_OBJECT;
+    }
+    if (mClient == 0) {
+        return INVALID_OPERATION;
+    }
+
+    // handle commands that are not forwarded transparently to effect engine
+    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
+        // No need to trylock() here as this function is executed in the binder thread serving a
+        // particular client process:  no risk to block the whole media server process or mixer
+        // threads if we are stuck here
+        Mutex::Autolock _l(mCblk->lock);
+        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
+            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
+            mCblk->serverIndex = 0;
+            mCblk->clientIndex = 0;
+            return BAD_VALUE;
+        }
+        status_t status = NO_ERROR;
+        while (mCblk->serverIndex < mCblk->clientIndex) {
+            int reply;
+            uint32_t rsize = sizeof(int);
+            int *p = (int *)(mBuffer + mCblk->serverIndex);
+            int size = *p++;
+            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
+                ALOGW("command(): invalid parameter block size");
+                break;
+            }
+            effect_param_t *param = (effect_param_t *)p;
+            if (param->psize == 0 || param->vsize == 0) {
+                ALOGW("command(): null parameter or value size");
+                mCblk->serverIndex += size;
+                continue;
+            }
+            uint32_t psize = sizeof(effect_param_t) +
+                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
+                             param->vsize;
+            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
+                                            psize,
+                                            p,
+                                            &rsize,
+                                            &reply);
+            // stop at first error encountered
+            if (ret != NO_ERROR) {
+                status = ret;
+                *(int *)pReplyData = reply;
+                break;
+            } else if (reply != NO_ERROR) {
+                *(int *)pReplyData = reply;
+                break;
+            }
+            mCblk->serverIndex += size;
+        }
+        mCblk->serverIndex = 0;
+        mCblk->clientIndex = 0;
+        return status;
+    } else if (cmdCode == EFFECT_CMD_ENABLE) {
+        *(int *)pReplyData = NO_ERROR;
+        return enable();
+    } else if (cmdCode == EFFECT_CMD_DISABLE) {
+        *(int *)pReplyData = NO_ERROR;
+        return disable();
+    }
+
+    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+}
+
+void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
+{
+    ALOGV("setControl %p control %d", this, hasControl);
+
+    mHasControl = hasControl;
+    mEnabled = enabled;
+
+    if (signal && mEffectClient != 0) {
+        mEffectClient->controlStatusChanged(hasControl);
+    }
+}
+
+void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
+                                                 uint32_t cmdSize,
+                                                 void *pCmdData,
+                                                 uint32_t replySize,
+                                                 void *pReplyData)
+{
+    if (mEffectClient != 0) {
+        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+    }
+}
+
+
+
+void AudioFlinger::EffectHandle::setEnabled(bool enabled)
+{
+    if (mEffectClient != 0) {
+        mEffectClient->enableStatusChanged(enabled);
+    }
+}
+
+status_t AudioFlinger::EffectHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnEffect::onTransact(code, data, reply, flags);
+}
+
+
+void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
+{
+    bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
+
+    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
+            (mClient == 0) ? getpid_cached : mClient->pid(),
+            mPriority,
+            mHasControl,
+            !locked,
+            mCblk ? mCblk->clientIndex : 0,
+            mCblk ? mCblk->serverIndex : 0
+            );
+
+    if (locked) {
+        mCblk->lock.unlock();
+    }
+}
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectChain"
+
+AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
+                                        int sessionId)
+    : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
+      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
+      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
+{
+    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+    if (thread == NULL) {
+        return;
+    }
+    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
+                                    thread->frameCount();
+}
+
+AudioFlinger::EffectChain::~EffectChain()
+{
+    if (mOwnInBuffer) {
+        delete mInBuffer;
+    }
+
+}
+
+// getEffectFromDesc_l() must be called with ThreadBase::mLock held
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(
+        effect_descriptor_t *descriptor)
+{
+    size_t size = mEffects.size();
+
+    for (size_t i = 0; i < size; i++) {
+        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
+            return mEffects[i];
+        }
+    }
+    return 0;
+}
+
+// getEffectFromId_l() must be called with ThreadBase::mLock held
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
+{
+    size_t size = mEffects.size();
+
+    for (size_t i = 0; i < size; i++) {
+        // by convention, return first effect if id provided is 0 (0 is never a valid id)
+        if (id == 0 || mEffects[i]->id() == id) {
+            return mEffects[i];
+        }
+    }
+    return 0;
+}
+
+// getEffectFromType_l() must be called with ThreadBase::mLock held
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
+        const effect_uuid_t *type)
+{
+    size_t size = mEffects.size();
+
+    for (size_t i = 0; i < size; i++) {
+        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
+            return mEffects[i];
+        }
+    }
+    return 0;
+}
+
+void AudioFlinger::EffectChain::clearInputBuffer()
+{
+    Mutex::Autolock _l(mLock);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        ALOGW("clearInputBuffer(): cannot promote mixer thread");
+        return;
+    }
+    clearInputBuffer_l(thread);
+}
+
+// Must be called with EffectChain::mLock locked
+void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread)
+{
+    size_t numSamples = thread->frameCount() * thread->channelCount();
+    memset(mInBuffer, 0, numSamples * sizeof(int16_t));
+
+}
+
+// Must be called with EffectChain::mLock locked
+void AudioFlinger::EffectChain::process_l()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        ALOGW("process_l(): cannot promote mixer thread");
+        return;
+    }
+    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
+            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
+    // always process effects unless no more tracks are on the session and the effect tail
+    // has been rendered
+    bool doProcess = true;
+    if (!isGlobalSession) {
+        bool tracksOnSession = (trackCnt() != 0);
+
+        if (!tracksOnSession && mTailBufferCount == 0) {
+            doProcess = false;
+        }
+
+        if (activeTrackCnt() == 0) {
+            // if no track is active and the effect tail has not been rendered,
+            // the input buffer must be cleared here as the mixer process will not do it
+            if (tracksOnSession || mTailBufferCount > 0) {
+                clearInputBuffer_l(thread);
+                if (mTailBufferCount > 0) {
+                    mTailBufferCount--;
+                }
+            }
+        }
+    }
+
+    size_t size = mEffects.size();
+    if (doProcess) {
+        for (size_t i = 0; i < size; i++) {
+            mEffects[i]->process();
+        }
+    }
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->updateState();
+    }
+}
+
+// addEffect_l() must be called with PlaybackThread::mLock held
+status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
+{
+    effect_descriptor_t desc = effect->desc();
+    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
+
+    Mutex::Autolock _l(mLock);
+    effect->setChain(this);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        return NO_INIT;
+    }
+    effect->setThread(thread);
+
+    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        // Auxiliary effects are inserted at the beginning of mEffects vector as
+        // they are processed first and accumulated in chain input buffer
+        mEffects.insertAt(effect, 0);
+
+        // the input buffer for auxiliary effect contains mono samples in
+        // 32 bit format. This is to avoid saturation in AudoMixer
+        // accumulation stage. Saturation is done in EffectModule::process() before
+        // calling the process in effect engine
+        size_t numSamples = thread->frameCount();
+        int32_t *buffer = new int32_t[numSamples];
+        memset(buffer, 0, numSamples * sizeof(int32_t));
+        effect->setInBuffer((int16_t *)buffer);
+        // auxiliary effects output samples to chain input buffer for further processing
+        // by insert effects
+        effect->setOutBuffer(mInBuffer);
+    } else {
+        // Insert effects are inserted at the end of mEffects vector as they are processed
+        //  after track and auxiliary effects.
+        // Insert effect order as a function of indicated preference:
+        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
+        //  another effect is present
+        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
+        //  last effect claiming first position
+        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
+        //  first effect claiming last position
+        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
+        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
+        // already present
+
+        size_t size = mEffects.size();
+        size_t idx_insert = size;
+        ssize_t idx_insert_first = -1;
+        ssize_t idx_insert_last = -1;
+
+        for (size_t i = 0; i < size; i++) {
+            effect_descriptor_t d = mEffects[i]->desc();
+            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
+            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
+            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
+                // check invalid effect chaining combinations
+                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
+                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
+                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s",
+                            desc.name, d.name);
+                    return INVALID_OPERATION;
+                }
+                // remember position of first insert effect and by default
+                // select this as insert position for new effect
+                if (idx_insert == size) {
+                    idx_insert = i;
+                }
+                // remember position of last insert effect claiming
+                // first position
+                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
+                    idx_insert_first = i;
+                }
+                // remember position of first insert effect claiming
+                // last position
+                if (iPref == EFFECT_FLAG_INSERT_LAST &&
+                    idx_insert_last == -1) {
+                    idx_insert_last = i;
+                }
+            }
+        }
+
+        // modify idx_insert from first position if needed
+        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
+            if (idx_insert_last != -1) {
+                idx_insert = idx_insert_last;
+            } else {
+                idx_insert = size;
+            }
+        } else {
+            if (idx_insert_first != -1) {
+                idx_insert = idx_insert_first + 1;
+            }
+        }
+
+        // always read samples from chain input buffer
+        effect->setInBuffer(mInBuffer);
+
+        // if last effect in the chain, output samples to chain
+        // output buffer, otherwise to chain input buffer
+        if (idx_insert == size) {
+            if (idx_insert != 0) {
+                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
+                mEffects[idx_insert-1]->configure();
+            }
+            effect->setOutBuffer(mOutBuffer);
+        } else {
+            effect->setOutBuffer(mInBuffer);
+        }
+        mEffects.insertAt(effect, idx_insert);
+
+        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this,
+                idx_insert);
+    }
+    effect->configure();
+    return NO_ERROR;
+}
+
+// removeEffect_l() must be called with PlaybackThread::mLock held
+size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mEffects.size();
+    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
+
+    for (size_t i = 0; i < size; i++) {
+        if (effect == mEffects[i]) {
+            // calling stop here will remove pre-processing effect from the audio HAL.
+            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
+            // the middle of a read from audio HAL
+            if (mEffects[i]->state() == EffectModule::ACTIVE ||
+                    mEffects[i]->state() == EffectModule::STOPPING) {
+                mEffects[i]->stop();
+            }
+            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
+                delete[] effect->inBuffer();
+            } else {
+                if (i == size - 1 && i != 0) {
+                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
+                    mEffects[i - 1]->configure();
+                }
+            }
+            mEffects.removeAt(i);
+            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(),
+                    this, i);
+            break;
+        }
+    }
+
+    return mEffects.size();
+}
+
+// setDevice_l() must be called with PlaybackThread::mLock held
+void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device)
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->setDevice(device);
+    }
+}
+
+// setMode_l() must be called with PlaybackThread::mLock held
+void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->setMode(mode);
+    }
+}
+
+// setAudioSource_l() must be called with PlaybackThread::mLock held
+void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source)
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->setAudioSource(source);
+    }
+}
+
+// setVolume_l() must be called with PlaybackThread::mLock held
+bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
+{
+    uint32_t newLeft = *left;
+    uint32_t newRight = *right;
+    bool hasControl = false;
+    int ctrlIdx = -1;
+    size_t size = mEffects.size();
+
+    // first update volume controller
+    for (size_t i = size; i > 0; i--) {
+        if (mEffects[i - 1]->isProcessEnabled() &&
+            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
+            ctrlIdx = i - 1;
+            hasControl = true;
+            break;
+        }
+    }
+
+    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
+        if (hasControl) {
+            *left = mNewLeftVolume;
+            *right = mNewRightVolume;
+        }
+        return hasControl;
+    }
+
+    mVolumeCtrlIdx = ctrlIdx;
+    mLeftVolume = newLeft;
+    mRightVolume = newRight;
+
+    // second get volume update from volume controller
+    if (ctrlIdx >= 0) {
+        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
+        mNewLeftVolume = newLeft;
+        mNewRightVolume = newRight;
+    }
+    // then indicate volume to all other effects in chain.
+    // Pass altered volume to effects before volume controller
+    // and requested volume to effects after controller
+    uint32_t lVol = newLeft;
+    uint32_t rVol = newRight;
+
+    for (size_t i = 0; i < size; i++) {
+        if ((int)i == ctrlIdx) {
+            continue;
+        }
+        // this also works for ctrlIdx == -1 when there is no volume controller
+        if ((int)i > ctrlIdx) {
+            lVol = *left;
+            rVol = *right;
+        }
+        mEffects[i]->setVolume(&lVol, &rVol, false);
+    }
+    *left = newLeft;
+    *right = newRight;
+
+    return hasControl;
+}
+
+void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
+    result.append(buffer);
+
+    bool locked = AudioFlinger::dumpTryLock(mLock);
+    // failed to lock - AudioFlinger is probably deadlocked
+    if (!locked) {
+        result.append("\tCould not lock mutex:\n");
+    }
+
+    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
+    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
+            mEffects.size(),
+            (uint32_t)mInBuffer,
+            (uint32_t)mOutBuffer,
+            mActiveTrackCnt);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    for (size_t i = 0; i < mEffects.size(); ++i) {
+        sp<EffectModule> effect = mEffects[i];
+        if (effect != 0) {
+            effect->dump(fd, args);
+        }
+    }
+
+    if (locked) {
+        mLock.unlock();
+    }
+}
+
+// must be called with ThreadBase::mLock held
+void AudioFlinger::EffectChain::setEffectSuspended_l(
+        const effect_uuid_t *type, bool suspend)
+{
+    sp<SuspendedEffectDesc> desc;
+    // use effect type UUID timelow as key as there is no real risk of identical
+    // timeLow fields among effect type UUIDs.
+    ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow);
+    if (suspend) {
+        if (index >= 0) {
+            desc = mSuspendedEffects.valueAt(index);
+        } else {
+            desc = new SuspendedEffectDesc();
+            desc->mType = *type;
+            mSuspendedEffects.add(type->timeLow, desc);
+            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
+        }
+        if (desc->mRefCount++ == 0) {
+            sp<EffectModule> effect = getEffectIfEnabled(type);
+            if (effect != 0) {
+                desc->mEffect = effect;
+                effect->setSuspended(true);
+                effect->setEnabled(false);
+            }
+        }
+    } else {
+        if (index < 0) {
+            return;
+        }
+        desc = mSuspendedEffects.valueAt(index);
+        if (desc->mRefCount <= 0) {
+            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
+            desc->mRefCount = 1;
+        }
+        if (--desc->mRefCount == 0) {
+            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
+            if (desc->mEffect != 0) {
+                sp<EffectModule> effect = desc->mEffect.promote();
+                if (effect != 0) {
+                    effect->setSuspended(false);
+                    effect->lock();
+                    EffectHandle *handle = effect->controlHandle_l();
+                    if (handle != NULL && !handle->destroyed_l()) {
+                        effect->setEnabled_l(handle->enabled());
+                    }
+                    effect->unlock();
+                }
+                desc->mEffect.clear();
+            }
+            mSuspendedEffects.removeItemsAt(index);
+        }
+    }
+}
+
+// must be called with ThreadBase::mLock held
+void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
+{
+    sp<SuspendedEffectDesc> desc;
+
+    ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
+    if (suspend) {
+        if (index >= 0) {
+            desc = mSuspendedEffects.valueAt(index);
+        } else {
+            desc = new SuspendedEffectDesc();
+            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
+            ALOGV("setEffectSuspendedAll_l() add entry for 0");
+        }
+        if (desc->mRefCount++ == 0) {
+            Vector< sp<EffectModule> > effects;
+            getSuspendEligibleEffects(effects);
+            for (size_t i = 0; i < effects.size(); i++) {
+                setEffectSuspended_l(&effects[i]->desc().type, true);
+            }
+        }
+    } else {
+        if (index < 0) {
+            return;
+        }
+        desc = mSuspendedEffects.valueAt(index);
+        if (desc->mRefCount <= 0) {
+            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
+            desc->mRefCount = 1;
+        }
+        if (--desc->mRefCount == 0) {
+            Vector<const effect_uuid_t *> types;
+            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
+                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
+                    continue;
+                }
+                types.add(&mSuspendedEffects.valueAt(i)->mType);
+            }
+            for (size_t i = 0; i < types.size(); i++) {
+                setEffectSuspended_l(types[i], false);
+            }
+            ALOGV("setEffectSuspendedAll_l() remove entry for %08x",
+                    mSuspendedEffects.keyAt(index));
+            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
+        }
+    }
+}
+
+
+// The volume effect is used for automated tests only
+#ifndef OPENSL_ES_H_
+static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
+                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
+const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
+#endif //OPENSL_ES_H_
+
+bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
+{
+    // auxiliary effects and visualizer are never suspended on output mix
+    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
+        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
+         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
+         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
+        return false;
+    }
+    return true;
+}
+
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(
+        Vector< sp<AudioFlinger::EffectModule> > &effects)
+{
+    effects.clear();
+    for (size_t i = 0; i < mEffects.size(); i++) {
+        if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
+            effects.add(mEffects[i]);
+        }
+    }
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
+                                                            const effect_uuid_t *type)
+{
+    sp<EffectModule> effect = getEffectFromType_l(type);
+    return effect != 0 && effect->isEnabled() ? effect : 0;
+}
+
+void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+                                                            bool enabled)
+{
+    ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
+    if (enabled) {
+        if (index < 0) {
+            // if the effect is not suspend check if all effects are suspended
+            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
+            if (index < 0) {
+                return;
+            }
+            if (!isEffectEligibleForSuspend(effect->desc())) {
+                return;
+            }
+            setEffectSuspended_l(&effect->desc().type, enabled);
+            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
+            if (index < 0) {
+                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
+                return;
+            }
+        }
+        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
+            effect->desc().type.timeLow);
+        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
+        // if effect is requested to suspended but was not yet enabled, supend it now.
+        if (desc->mEffect == 0) {
+            desc->mEffect = effect;
+            effect->setEnabled(false);
+            effect->setSuspended(true);
+        }
+    } else {
+        if (index < 0) {
+            return;
+        }
+        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
+            effect->desc().type.timeLow);
+        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
+        desc->mEffect.clear();
+        effect->setSuspended(false);
+    }
+}
+
+}; // namespace android
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
new file mode 100644
index 0000000..91303ee
--- /dev/null
+++ b/services/audioflinger/Effects.h
@@ -0,0 +1,359 @@
+/*
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+//--- Audio Effect Management
+
+// EffectModule and EffectChain classes both have their own mutex to protect
+// state changes or resource modifications. Always respect the following order
+// if multiple mutexes must be acquired to avoid cross deadlock:
+// AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
+
+// The EffectModule class is a wrapper object controlling the effect engine implementation
+// in the effect library. It prevents concurrent calls to process() and command() functions
+// from different client threads. It keeps a list of EffectHandle objects corresponding
+// to all client applications using this effect and notifies applications of effect state,
+// control or parameter changes. It manages the activation state machine to send appropriate
+// reset, enable, disable commands to effect engine and provide volume
+// ramping when effects are activated/deactivated.
+// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
+// the attached track(s) to accumulate their auxiliary channel.
+class EffectModule : public RefBase {
+public:
+    EffectModule(ThreadBase *thread,
+                    const wp<AudioFlinger::EffectChain>& chain,
+                    effect_descriptor_t *desc,
+                    int id,
+                    int sessionId);
+    virtual ~EffectModule();
+
+    enum effect_state {
+        IDLE,
+        RESTART,
+        STARTING,
+        ACTIVE,
+        STOPPING,
+        STOPPED,
+        DESTROYED
+    };
+
+    int         id() const { return mId; }
+    void process();
+    void updateState();
+    status_t command(uint32_t cmdCode,
+                     uint32_t cmdSize,
+                     void *pCmdData,
+                     uint32_t *replySize,
+                     void *pReplyData);
+
+    void reset_l();
+    status_t configure();
+    status_t init();
+    effect_state state() const {
+        return mState;
+    }
+    uint32_t status() {
+        return mStatus;
+    }
+    int sessionId() const {
+        return mSessionId;
+    }
+    status_t    setEnabled(bool enabled);
+    status_t    setEnabled_l(bool enabled);
+    bool isEnabled() const;
+    bool isProcessEnabled() const;
+
+    void        setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
+    int16_t     *inBuffer() { return mConfig.inputCfg.buffer.s16; }
+    void        setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; }
+    int16_t     *outBuffer() { return mConfig.outputCfg.buffer.s16; }
+    void        setChain(const wp<EffectChain>& chain) { mChain = chain; }
+    void        setThread(const wp<ThreadBase>& thread) { mThread = thread; }
+    const wp<ThreadBase>& thread() { return mThread; }
+
+    status_t addHandle(EffectHandle *handle);
+    size_t disconnect(EffectHandle *handle, bool unpinIfLast);
+    size_t removeHandle(EffectHandle *handle);
+
+    const effect_descriptor_t& desc() const { return mDescriptor; }
+    wp<EffectChain>&     chain() { return mChain; }
+
+    status_t         setDevice(audio_devices_t device);
+    status_t         setVolume(uint32_t *left, uint32_t *right, bool controller);
+    status_t         setMode(audio_mode_t mode);
+    status_t         setAudioSource(audio_source_t source);
+    status_t         start();
+    status_t         stop();
+    void             setSuspended(bool suspended);
+    bool             suspended() const;
+
+    EffectHandle*    controlHandle_l();
+
+    bool             isPinned() const { return mPinned; }
+    void             unPin() { mPinned = false; }
+    bool             purgeHandles();
+    void             lock() { mLock.lock(); }
+    void             unlock() { mLock.unlock(); }
+
+    void             dump(int fd, const Vector<String16>& args);
+
+protected:
+    friend class AudioFlinger;      // for mHandles
+    bool                mPinned;
+
+    // Maximum time allocated to effect engines to complete the turn off sequence
+    static const uint32_t MAX_DISABLE_TIME_MS = 10000;
+
+    EffectModule(const EffectModule&);
+    EffectModule& operator = (const EffectModule&);
+
+    status_t start_l();
+    status_t stop_l();
+
+mutable Mutex               mLock;      // mutex for process, commands and handles list protection
+    wp<ThreadBase>      mThread;    // parent thread
+    wp<EffectChain>     mChain;     // parent effect chain
+    const int           mId;        // this instance unique ID
+    const int           mSessionId; // audio session ID
+    const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
+    effect_config_t     mConfig;    // input and output audio configuration
+    effect_handle_t  mEffectInterface; // Effect module C API
+    status_t            mStatus;    // initialization status
+    effect_state        mState;     // current activation state
+    Vector<EffectHandle *> mHandles;    // list of client handles
+                // First handle in mHandles has highest priority and controls the effect module
+    uint32_t mMaxDisableWaitCnt;    // maximum grace period before forcing an effect off after
+                                    // sending disable command.
+    uint32_t mDisableWaitCnt;       // current process() calls count during disable period.
+    bool     mSuspended;            // effect is suspended: temporarily disabled by framework
+};
+
+// The EffectHandle class implements the IEffect interface. It provides resources
+// to receive parameter updates, keeps track of effect control
+// ownership and state and has a pointer to the EffectModule object it is controlling.
+// There is one EffectHandle object for each application controlling (or using)
+// an effect module.
+// The EffectHandle is obtained by calling AudioFlinger::createEffect().
+class EffectHandle: public android::BnEffect {
+public:
+
+    EffectHandle(const sp<EffectModule>& effect,
+            const sp<AudioFlinger::Client>& client,
+            const sp<IEffectClient>& effectClient,
+            int32_t priority);
+    virtual ~EffectHandle();
+
+    // IEffect
+    virtual status_t enable();
+    virtual status_t disable();
+    virtual status_t command(uint32_t cmdCode,
+                             uint32_t cmdSize,
+                             void *pCmdData,
+                             uint32_t *replySize,
+                             void *pReplyData);
+    virtual void disconnect();
+private:
+            void disconnect(bool unpinIfLast);
+public:
+    virtual sp<IMemory> getCblk() const { return mCblkMemory; }
+    virtual status_t onTransact(uint32_t code, const Parcel& data,
+            Parcel* reply, uint32_t flags);
+
+
+    // Give or take control of effect module
+    // - hasControl: true if control is given, false if removed
+    // - signal: true client app should be signaled of change, false otherwise
+    // - enabled: state of the effect when control is passed
+    void setControl(bool hasControl, bool signal, bool enabled);
+    void commandExecuted(uint32_t cmdCode,
+                         uint32_t cmdSize,
+                         void *pCmdData,
+                         uint32_t replySize,
+                         void *pReplyData);
+    void setEnabled(bool enabled);
+    bool enabled() const { return mEnabled; }
+
+    // Getters
+    int id() const { return mEffect->id(); }
+    int priority() const { return mPriority; }
+    bool hasControl() const { return mHasControl; }
+    sp<EffectModule> effect() const { return mEffect; }
+    // destroyed_l() must be called with the associated EffectModule mLock held
+    bool destroyed_l() const { return mDestroyed; }
+
+    void dump(char* buffer, size_t size);
+
+protected:
+    friend class AudioFlinger;          // for mEffect, mHasControl, mEnabled
+    EffectHandle(const EffectHandle&);
+    EffectHandle& operator =(const EffectHandle&);
+
+    sp<EffectModule> mEffect;           // pointer to controlled EffectModule
+    sp<IEffectClient> mEffectClient;    // callback interface for client notifications
+    /*const*/ sp<Client> mClient;       // client for shared memory allocation, see disconnect()
+    sp<IMemory>         mCblkMemory;    // shared memory for control block
+    effect_param_cblk_t* mCblk;         // control block for deferred parameter setting via
+                                        // shared memory
+    uint8_t*            mBuffer;        // pointer to parameter area in shared memory
+    int mPriority;                      // client application priority to control the effect
+    bool mHasControl;                   // true if this handle is controlling the effect
+    bool mEnabled;                      // cached enable state: needed when the effect is
+                                        // restored after being suspended
+    bool mDestroyed;                    // Set to true by destructor. Access with EffectModule
+                                        // mLock held
+};
+
+// the EffectChain class represents a group of effects associated to one audio session.
+// There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
+// The EffecChain with session ID 0 contains global effects applied to the output mix.
+// Effects in this chain can be insert or auxiliary. Effects in other chains (attached to
+// tracks) are insert only. The EffectChain maintains an ordered list of effect module, the
+// order corresponding in the effect process order. When attached to a track (session ID != 0),
+// it also provide it's own input buffer used by the track as accumulation buffer.
+class EffectChain : public RefBase {
+public:
+    EffectChain(const wp<ThreadBase>& wThread, int sessionId);
+    EffectChain(ThreadBase *thread, int sessionId);
+    virtual ~EffectChain();
+
+    // special key used for an entry in mSuspendedEffects keyed vector
+    // corresponding to a suspend all request.
+    static const int        kKeyForSuspendAll = 0;
+
+    // minimum duration during which we force calling effect process when last track on
+    // a session is stopped or removed to allow effect tail to be rendered
+    static const int        kProcessTailDurationMs = 1000;
+
+    void process_l();
+
+    void lock() {
+        mLock.lock();
+    }
+    void unlock() {
+        mLock.unlock();
+    }
+
+    status_t addEffect_l(const sp<EffectModule>& handle);
+    size_t removeEffect_l(const sp<EffectModule>& handle);
+
+    int sessionId() const { return mSessionId; }
+    void setSessionId(int sessionId) { mSessionId = sessionId; }
+
+    sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
+    sp<EffectModule> getEffectFromId_l(int id);
+    sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
+    bool setVolume_l(uint32_t *left, uint32_t *right);
+    void setDevice_l(audio_devices_t device);
+    void setMode_l(audio_mode_t mode);
+    void setAudioSource_l(audio_source_t source);
+
+    void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
+        mInBuffer = buffer;
+        mOwnInBuffer = ownsBuffer;
+    }
+    int16_t *inBuffer() const {
+        return mInBuffer;
+    }
+    void setOutBuffer(int16_t *buffer) {
+        mOutBuffer = buffer;
+    }
+    int16_t *outBuffer() const {
+        return mOutBuffer;
+    }
+
+    void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
+    void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
+    int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); }
+
+    void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
+                               mTailBufferCount = mMaxTailBuffers; }
+    void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
+    int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); }
+
+    uint32_t strategy() const { return mStrategy; }
+    void setStrategy(uint32_t strategy)
+            { mStrategy = strategy; }
+
+    // suspend effect of the given type
+    void setEffectSuspended_l(const effect_uuid_t *type,
+                              bool suspend);
+    // suspend all eligible effects
+    void setEffectSuspendedAll_l(bool suspend);
+    // check if effects should be suspend or restored when a given effect is enable or disabled
+    void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+                                          bool enabled);
+
+    void clearInputBuffer();
+
+    void dump(int fd, const Vector<String16>& args);
+
+protected:
+    friend class AudioFlinger;  // for mThread, mEffects
+    EffectChain(const EffectChain&);
+    EffectChain& operator =(const EffectChain&);
+
+    class SuspendedEffectDesc : public RefBase {
+    public:
+        SuspendedEffectDesc() : mRefCount(0) {}
+
+        int mRefCount;
+        effect_uuid_t mType;
+        wp<EffectModule> mEffect;
+    };
+
+    // get a list of effect modules to suspend when an effect of the type
+    // passed is enabled.
+    void                       getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
+
+    // get an effect module if it is currently enable
+    sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
+    // true if the effect whose descriptor is passed can be suspended
+    // OEMs can modify the rules implemented in this method to exclude specific effect
+    // types or implementations from the suspend/restore mechanism.
+    bool isEffectEligibleForSuspend(const effect_descriptor_t& desc);
+
+    void clearInputBuffer_l(sp<ThreadBase> thread);
+
+    wp<ThreadBase> mThread;     // parent mixer thread
+    Mutex mLock;                // mutex protecting effect list
+    Vector< sp<EffectModule> > mEffects; // list of effect modules
+    int mSessionId;             // audio session ID
+    int16_t *mInBuffer;         // chain input buffer
+    int16_t *mOutBuffer;        // chain output buffer
+
+    // 'volatile' here means these are accessed with atomic operations instead of mutex
+    volatile int32_t mActiveTrackCnt;    // number of active tracks connected
+    volatile int32_t mTrackCnt;          // number of tracks connected
+
+    int32_t mTailBufferCount;   // current effect tail buffer count
+    int32_t mMaxTailBuffers;    // maximum effect tail buffers
+    bool mOwnInBuffer;          // true if the chain owns its input buffer
+    int mVolumeCtrlIdx;         // index of insert effect having control over volume
+    uint32_t mLeftVolume;       // previous volume on left channel
+    uint32_t mRightVolume;      // previous volume on right channel
+    uint32_t mNewLeftVolume;       // new volume on left channel
+    uint32_t mNewRightVolume;      // new volume on right channel
+    uint32_t mStrategy; // strategy for this effect chain
+    // mSuspendedEffects lists all effects currently suspended in the chain.
+    // Use effect type UUID timelow field as key. There is no real risk of identical
+    // timeLow fields among effect type UUIDs.
+    // Updated by updateSuspendedSessions_l() only.
+    KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
+};
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
new file mode 100644
index 0000000..b898924
--- /dev/null
+++ b/services/audioflinger/PlaybackTracks.h
@@ -0,0 +1,285 @@
+/*
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+// playback track
+class Track : public TrackBase, public VolumeProvider {
+public:
+                        Track(  PlaybackThread *thread,
+                                const sp<Client>& client,
+                                audio_stream_type_t streamType,
+                                uint32_t sampleRate,
+                                audio_format_t format,
+                                audio_channel_mask_t channelMask,
+                                size_t frameCount,
+                                const sp<IMemory>& sharedBuffer,
+                                int sessionId,
+                                IAudioFlinger::track_flags_t flags);
+    virtual             ~Track();
+
+    static  void        appendDumpHeader(String8& result);
+            void        dump(char* buffer, size_t size);
+    virtual status_t    start(AudioSystem::sync_event_t event =
+                                    AudioSystem::SYNC_EVENT_NONE,
+                             int triggerSession = 0);
+    virtual void        stop();
+            void        pause();
+
+            void        flush();
+            void        destroy();
+            void        mute(bool);
+            int         name() const { return mName; }
+
+            audio_stream_type_t streamType() const {
+                return mStreamType;
+            }
+            status_t    attachAuxEffect(int EffectId);
+            void        setAuxBuffer(int EffectId, int32_t *buffer);
+            int32_t     *auxBuffer() const { return mAuxBuffer; }
+            void        setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
+            int16_t     *mainBuffer() const { return mMainBuffer; }
+            int         auxEffectId() const { return mAuxEffectId; }
+
+// implement FastMixerState::VolumeProvider interface
+    virtual uint32_t    getVolumeLR();
+
+    virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
+
+protected:
+    // for numerous
+    friend class PlaybackThread;
+    friend class MixerThread;
+    friend class DirectOutputThread;
+
+                        Track(const Track&);
+                        Track& operator = (const Track&);
+
+    // AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                   int64_t pts = kInvalidPTS);
+    // releaseBuffer() not overridden
+
+    virtual size_t framesReady() const;
+
+    bool isMuted() const { return mMute; }
+    bool isPausing() const {
+        return mState == PAUSING;
+    }
+    bool isPaused() const {
+        return mState == PAUSED;
+    }
+    bool isResuming() const {
+        return mState == RESUMING;
+    }
+    bool isReady() const;
+    void setPaused() { mState = PAUSED; }
+    void reset();
+
+    bool isOutputTrack() const {
+        return (mStreamType == AUDIO_STREAM_CNT);
+    }
+
+    sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
+
+    // framesWritten is cumulative, never reset, and is shared all tracks
+    // audioHalFrames is derived from output latency
+    // FIXME parameters not needed, could get them from the thread
+    bool presentationComplete(size_t framesWritten, size_t audioHalFrames);
+
+public:
+    void triggerEvents(AudioSystem::sync_event_t type);
+    virtual bool isTimedTrack() const { return false; }
+    bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; }
+    virtual bool isOut() const;
+
+protected:
+
+    // written by Track::mute() called by binder thread(s), without a mutex or barrier.
+    // read by Track::isMuted() called by playback thread, also without a mutex or barrier.
+    // The lack of mutex or barrier is safe because the mute status is only used by itself.
+    bool                mMute;
+
+    // FILLED state is used for suppressing volume ramp at begin of playing
+    enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE};
+    mutable uint8_t     mFillingUpStatus;
+    int8_t              mRetryCount;
+    const sp<IMemory>   mSharedBuffer;
+    bool                mResetDone;
+    const audio_stream_type_t mStreamType;
+    int                 mName;      // track name on the normal mixer,
+                                    // allocated statically at track creation time,
+                                    // and is even allocated (though unused) for fast tracks
+                                    // FIXME don't allocate track name for fast tracks
+    int16_t             *mMainBuffer;
+    int32_t             *mAuxBuffer;
+    int                 mAuxEffectId;
+    bool                mHasVolumeController;
+    size_t              mPresentationCompleteFrames; // number of frames written to the
+                                    // audio HAL when this track will be fully rendered
+                                    // zero means not monitoring
+private:
+    IAudioFlinger::track_flags_t mFlags;
+
+    // The following fields are only for fast tracks, and should be in a subclass
+    int                 mFastIndex; // index within FastMixerState::mFastTracks[];
+                                    // either mFastIndex == -1 if not isFastTrack()
+                                    // or 0 < mFastIndex < FastMixerState::kMaxFast because
+                                    // index 0 is reserved for normal mixer's submix;
+                                    // index is allocated statically at track creation time
+                                    // but the slot is only used if track is active
+    FastTrackUnderruns  mObservedUnderruns; // Most recently observed value of
+                                    // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns
+    uint32_t            mUnderrunCount; // Counter of total number of underruns, never reset
+    volatile float      mCachedVolume;  // combined master volume and stream type volume;
+                                        // 'volatile' means accessed without lock or
+                                        // barrier, but is read/written atomically
+};  // end of Track
+
+class TimedTrack : public Track {
+  public:
+    static sp<TimedTrack> create(PlaybackThread *thread,
+                                 const sp<Client>& client,
+                                 audio_stream_type_t streamType,
+                                 uint32_t sampleRate,
+                                 audio_format_t format,
+                                 audio_channel_mask_t channelMask,
+                                 size_t frameCount,
+                                 const sp<IMemory>& sharedBuffer,
+                                 int sessionId);
+    virtual ~TimedTrack();
+
+    class TimedBuffer {
+      public:
+        TimedBuffer();
+        TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+        const sp<IMemory>& buffer() const { return mBuffer; }
+        int64_t pts() const { return mPTS; }
+        uint32_t position() const { return mPosition; }
+        void setPosition(uint32_t pos) { mPosition = pos; }
+      private:
+        sp<IMemory> mBuffer;
+        int64_t     mPTS;
+        uint32_t    mPosition;
+    };
+
+    // Mixer facing methods.
+    virtual bool isTimedTrack() const { return true; }
+    virtual size_t framesReady() const;
+
+    // AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                   int64_t pts);
+    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
+    // Client/App facing methods.
+    status_t    allocateTimedBuffer(size_t size,
+                                    sp<IMemory>* buffer);
+    status_t    queueTimedBuffer(const sp<IMemory>& buffer,
+                                 int64_t pts);
+    status_t    setMediaTimeTransform(const LinearTransform& xform,
+                                      TimedAudioTrack::TargetTimeline target);
+
+  private:
+    TimedTrack(PlaybackThread *thread,
+               const sp<Client>& client,
+               audio_stream_type_t streamType,
+               uint32_t sampleRate,
+               audio_format_t format,
+               audio_channel_mask_t channelMask,
+               size_t frameCount,
+               const sp<IMemory>& sharedBuffer,
+               int sessionId);
+
+    void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
+    void timedYieldSilence_l(uint32_t numFrames,
+                             AudioBufferProvider::Buffer* buffer);
+    void trimTimedBufferQueue_l();
+    void trimTimedBufferQueueHead_l(const char* logTag);
+    void updateFramesPendingAfterTrim_l(const TimedBuffer& buf,
+                                        const char* logTag);
+
+    uint64_t            mLocalTimeFreq;
+    LinearTransform     mLocalTimeToSampleTransform;
+    LinearTransform     mMediaTimeToSampleTransform;
+    sp<MemoryDealer>    mTimedMemoryDealer;
+
+    Vector<TimedBuffer> mTimedBufferQueue;
+    bool                mQueueHeadInFlight;
+    bool                mTrimQueueHeadOnRelease;
+    uint32_t            mFramesPendingInQueue;
+
+    uint8_t*            mTimedSilenceBuffer;
+    uint32_t            mTimedSilenceBufferSize;
+    mutable Mutex       mTimedBufferQueueLock;
+    bool                mTimedAudioOutputOnTime;
+    CCHelper            mCCHelper;
+
+    Mutex               mMediaTimeTransformLock;
+    LinearTransform     mMediaTimeTransform;
+    bool                mMediaTimeTransformValid;
+    TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+};
+
+
+// playback track, used by DuplicatingThread
+class OutputTrack : public Track {
+public:
+
+    class Buffer : public AudioBufferProvider::Buffer {
+    public:
+        int16_t *mBuffer;
+    };
+
+                        OutputTrack(PlaybackThread *thread,
+                                DuplicatingThread *sourceThread,
+                                uint32_t sampleRate,
+                                audio_format_t format,
+                                audio_channel_mask_t channelMask,
+                                size_t frameCount);
+    virtual             ~OutputTrack();
+
+    virtual status_t    start(AudioSystem::sync_event_t event =
+                                    AudioSystem::SYNC_EVENT_NONE,
+                             int triggerSession = 0);
+    virtual void        stop();
+            bool        write(int16_t* data, uint32_t frames);
+            bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
+            bool        isActive() const { return mActive; }
+    const wp<ThreadBase>& thread() const { return mThread; }
+
+private:
+
+    enum {
+        NO_MORE_BUFFERS = 0x80000001,   // same in AudioTrack.h, ok to be different value
+    };
+
+    status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer,
+                                     uint32_t waitTimeMs);
+    void                clearBufferQueue();
+
+    // Maximum number of pending buffers allocated by OutputTrack::write()
+    static const uint8_t kMaxOverFlowBuffers = 10;
+
+    Vector < Buffer* >          mBufferQueue;
+    AudioBufferProvider::Buffer mOutBuffer;
+    bool                        mActive;
+    DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
+    void*                       mBuffers;   // starting address of buffers in plain memory
+};  // end of OutputTrack
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
new file mode 100644
index 0000000..fe681d7
--- /dev/null
+++ b/services/audioflinger/RecordTracks.h
@@ -0,0 +1,62 @@
+/*
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+// record track
+class RecordTrack : public TrackBase {
+public:
+                        RecordTrack(RecordThread *thread,
+                                const sp<Client>& client,
+                                uint32_t sampleRate,
+                                audio_format_t format,
+                                audio_channel_mask_t channelMask,
+                                size_t frameCount,
+                                int sessionId);
+    virtual             ~RecordTrack();
+
+    virtual status_t    start(AudioSystem::sync_event_t event, int triggerSession);
+    virtual void        stop();
+
+            void        destroy();
+
+            // clear the buffer overflow flag
+            void        clearOverflow() { mOverflow = false; }
+            // set the buffer overflow flag and return previous value
+            bool        setOverflow() { bool tmp = mOverflow; mOverflow = true;
+                                                return tmp; }
+
+    static  void        appendDumpHeader(String8& result);
+            void        dump(char* buffer, size_t size);
+
+    virtual bool isOut() const;
+
+private:
+    friend class AudioFlinger;  // for mState
+
+                        RecordTrack(const RecordTrack&);
+                        RecordTrack& operator = (const RecordTrack&);
+
+    // AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                   int64_t pts = kInvalidPTS);
+    // releaseBuffer() not overridden
+
+    bool                mOverflow;  // overflow on most recent attempt to fill client buffer
+};
diff --git a/services/audioflinger/StateQueue.h b/services/audioflinger/StateQueue.h
index eba190c..e33b3c6 100644
--- a/services/audioflinger/StateQueue.h
+++ b/services/audioflinger/StateQueue.h
@@ -17,6 +17,72 @@
 #ifndef ANDROID_AUDIO_STATE_QUEUE_H
 #define ANDROID_AUDIO_STATE_QUEUE_H
 
+// The state queue template class was originally driven by this use case / requirements:
+//  There are two threads: a fast mixer, and a normal mixer, and they share state.
+//  The interesting part of the shared state is a set of active fast tracks,
+//  and the output HAL configuration (buffer size in frames, sample rate, etc.).
+//  Fast mixer thread:
+//      periodic with typical period < 10 ms
+//      FIFO/RR scheduling policy and a low fixed priority
+//      ok to block for bounded time using nanosleep() to achieve desired period
+//      must not block on condition wait, mutex lock, atomic operation spin, I/O, etc.
+//        under typical operations of mixing, writing, or adding/removing tracks
+//      ok to block for unbounded time when the output HAL configuration changes,
+//        and this may result in an audible artifact
+//      needs read-only access to a recent stable state,
+//        but not necessarily the most current one
+//  Normal mixer thread:
+//      periodic with typical period ~40 ms
+//      SCHED_OTHER scheduling policy and nice priority == urgent audio
+//      ok to block, but prefer to avoid as much as possible
+//      needs read/write access to state
+//  The normal mixer may need to temporarily suspend the fast mixer thread during mode changes.
+//  It will do this using the state -- one of the fields tells the fast mixer to idle.
+
+// Additional requirements:
+//  - observer must always be able to poll for and view the latest pushed state; it must never be
+//    blocked from seeing that state
+//  - observer does not need to see every state in sequence; it is OK for it to skip states
+//    [see below for more on this]
+//  - mutator must always be able to read/modify a state, it must never be blocked from reading or
+//    modifying state
+//  - reduce memcpy where possible
+//  - work well if the observer runs more frequently than the mutator,
+//    as is the case with fast mixer/normal mixer.
+// It is not a requirement to work well if the roles were reversed,
+// and the mutator were to run more frequently than the observer.
+// In this case, the mutator could get blocked waiting for a slot to fill up for
+// it to work with. This could be solved somewhat by increasing the depth of the queue, but it would
+// still limit the mutator to a finite number of changes before it would block.  A future
+// possibility, not implemented here, would be to allow the mutator to safely overwrite an already
+// pushed state. This could be done by the mutator overwriting mNext, but then being prepared to
+// read an mAck which is actually for the earlier mNext (since there is a race).
+
+// Solution:
+//  Let's call the fast mixer thread the "observer" and normal mixer thread the "mutator".
+//  We assume there is only a single observer and a single mutator; this is critical.
+//  Each state is of type <T>, and should contain only POD (Plain Old Data) and raw pointers, as
+//  memcpy() may be used to copy state, and the destructors are run in unpredictable order.
+//  The states in chronological order are: previous, current, next, and mutating:
+//      previous    read-only, observer can compare vs. current to see the subset that changed
+//      current     read-only, this is the primary state for observer
+//      next        read-only, when observer is ready to accept a new state it will shift it in:
+//                      previous = current
+//                      current = next
+//                  and the slot formerly used by previous is now available to the mutator.
+//      mutating    invisible to observer, read/write to mutator
+//  Initialization is tricky, especially for the observer.  If the observer starts execution
+//  before the mutator, there are no previous, current, or next states.  And even if the observer
+//  starts execution after the mutator, there is a next state but no previous or current states.
+//  To solve this, we'll have the observer idle until there is a next state,
+//  and it will have to deal with the case where there is no previous state.
+//  The states are stored in a shared FIFO queue represented using a circular array.
+//  The observer polls for mutations, and receives a new state pointer after a
+//  a mutation is pushed onto the queue.  To the observer, the state pointers are
+//  effectively in random order, that is the observer should not do address
+//  arithmetic on the state pointers.  However to the mutator, the state pointers
+//  are in a definite circular order.
+
 namespace android {
 
 #ifdef STATE_QUEUE_DUMP
@@ -108,7 +174,7 @@
 #endif
 
 private:
-    static const unsigned kN = 4;       // values != 4 are not supported by this code
+    static const unsigned kN = 4;       // values < 4 are not supported by this code
     T                 mStates[kN];      // written by mutator, read by observer
 
     // "volatile" is meaningless with SMP, but here it indicates that we're using atomic ops
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
new file mode 100644
index 0000000..1ceb850
--- /dev/null
+++ b/services/audioflinger/Threads.cpp
@@ -0,0 +1,4426 @@
+/*
+**
+** 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.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#include <cutils/properties.h>
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+
+#include <private/media/AudioTrackShared.h>
+#include <hardware/audio.h>
+#include <audio_effects/effect_ns.h>
+#include <audio_effects/effect_aec.h>
+#include <audio_utils/primitives.h>
+
+// NBAIO implementations
+#include <media/nbaio/AudioStreamOutSink.h>
+#include <media/nbaio/MonoPipe.h>
+#include <media/nbaio/MonoPipeReader.h>
+#include <media/nbaio/Pipe.h>
+#include <media/nbaio/PipeReader.h>
+#include <media/nbaio/SourceAudioBufferProvider.h>
+
+#include <powermanager/PowerManager.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
+#include "AudioFlinger.h"
+#include "AudioMixer.h"
+#include "FastMixer.h"
+#include "ServiceUtilities.h"
+#include "SchedulingPolicyService.h"
+
+#undef ADD_BATTERY_DATA
+
+#ifdef ADD_BATTERY_DATA
+#include <media/IMediaPlayerService.h>
+#include <media/IMediaDeathNotifier.h>
+#endif
+
+// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
+#ifdef DEBUG_CPU_USAGE
+#include <cpustats/CentralTendencyStatistics.h>
+#include <cpustats/ThreadCpuUsage.h>
+#endif
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message.  In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on.  Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+// allow less retry attempts on direct output thread.
+// direct outputs can be a scarce resource in audio hardware and should
+// be released as quickly as possible.
+static const int8_t kMaxTrackRetriesDirect = 2;
+
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
+
+// RecordThread loop sleep time upon application overrun or audio HAL read error
+static const int kRecordThreadSleepUs = 5000;
+
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
+
+// minimum sleep time for the mixer thread loop when tracks are active but in underrun
+static const uint32_t kMinThreadSleepTimeUs = 5000;
+// maximum divider applied to the active sleep time in the mixer thread loop
+static const uint32_t kMaxThreadSleepTimeShift = 2;
+
+// minimum normal mix buffer size, expressed in milliseconds rather than frames
+static const uint32_t kMinNormalMixBufferSizeMs = 20;
+// maximum normal mix buffer size
+static const uint32_t kMaxNormalMixBufferSizeMs = 24;
+
+// Whether to use fast mixer
+static const enum {
+    FastMixer_Never,    // never initialize or use: for debugging only
+    FastMixer_Always,   // always initialize and use, even if not needed: for debugging only
+                        // normal mixer multiplier is 1
+    FastMixer_Static,   // initialize if needed, then use all the time if initialized,
+                        // multiplier is calculated based on min & max normal mixer buffer size
+    FastMixer_Dynamic,  // initialize if needed, then use dynamically depending on track load,
+                        // multiplier is calculated based on min & max normal mixer buffer size
+    // FIXME for FastMixer_Dynamic:
+    //  Supporting this option will require fixing HALs that can't handle large writes.
+    //  For example, one HAL implementation returns an error from a large write,
+    //  and another HAL implementation corrupts memory, possibly in the sample rate converter.
+    //  We could either fix the HAL implementations, or provide a wrapper that breaks
+    //  up large writes into smaller ones, and the wrapper would need to deal with scheduler.
+} kUseFastMixer = FastMixer_Static;
+
+// Priorities for requestPriority
+static const int kPriorityAudioApp = 2;
+static const int kPriorityFastMixer = 3;
+
+// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
+// for the track.  The client then sub-divides this into smaller buffers for its use.
+// Currently the client uses double-buffering by default, but doesn't tell us about that.
+// So for now we just assume that client is double-buffered.
+// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
+// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// See the client's minBufCount and mNotificationFramesAct calculations for details.
+static const int kFastTrackMultiplier = 2;
+
+// ----------------------------------------------------------------------------
+
+#ifdef ADD_BATTERY_DATA
+// To collect the amplifier usage
+static void addBatteryData(uint32_t params) {
+    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
+    if (service == NULL) {
+        // it already logged
+        return;
+    }
+
+    service->addBatteryData(params);
+}
+#endif
+
+
+// ----------------------------------------------------------------------------
+//      CPU Stats
+// ----------------------------------------------------------------------------
+
+class CpuStats {
+public:
+    CpuStats();
+    void sample(const String8 &title);
+#ifdef DEBUG_CPU_USAGE
+private:
+    ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
+    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+
+    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+
+    int mCpuNum;                        // thread's current CPU number
+    int mCpukHz;                        // frequency of thread's current CPU in kHz
+#endif
+};
+
+CpuStats::CpuStats()
+#ifdef DEBUG_CPU_USAGE
+    : mCpuNum(-1), mCpukHz(-1)
+#endif
+{
+}
+
+void CpuStats::sample(const String8 &title) {
+#ifdef DEBUG_CPU_USAGE
+    // get current thread's delta CPU time in wall clock ns
+    double wcNs;
+    bool valid = mCpuUsage.sampleAndEnable(wcNs);
+
+    // record sample for wall clock statistics
+    if (valid) {
+        mWcStats.sample(wcNs);
+    }
+
+    // get the current CPU number
+    int cpuNum = sched_getcpu();
+
+    // get the current CPU frequency in kHz
+    int cpukHz = mCpuUsage.getCpukHz(cpuNum);
+
+    // check if either CPU number or frequency changed
+    if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
+        mCpuNum = cpuNum;
+        mCpukHz = cpukHz;
+        // ignore sample for purposes of cycles
+        valid = false;
+    }
+
+    // if no change in CPU number or frequency, then record sample for cycle statistics
+    if (valid && mCpukHz > 0) {
+        double cycles = wcNs * cpukHz * 0.000001;
+        mHzStats.sample(cycles);
+    }
+
+    unsigned n = mWcStats.n();
+    // mCpuUsage.elapsed() is expensive, so don't call it every loop
+    if ((n & 127) == 1) {
+        long long elapsed = mCpuUsage.elapsed();
+        if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
+            double perLoop = elapsed / (double) n;
+            double perLoop100 = perLoop * 0.01;
+            double perLoop1k = perLoop * 0.001;
+            double mean = mWcStats.mean();
+            double stddev = mWcStats.stddev();
+            double minimum = mWcStats.minimum();
+            double maximum = mWcStats.maximum();
+            double meanCycles = mHzStats.mean();
+            double stddevCycles = mHzStats.stddev();
+            double minCycles = mHzStats.minimum();
+            double maxCycles = mHzStats.maximum();
+            mCpuUsage.resetElapsed();
+            mWcStats.reset();
+            mHzStats.reset();
+            ALOGD("CPU usage for %s over past %.1f secs\n"
+                "  (%u mixer loops at %.1f mean ms per loop):\n"
+                "  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
+                "  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
+                "  MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
+                    title.string(),
+                    elapsed * .000000001, n, perLoop * .000001,
+                    mean * .001,
+                    stddev * .001,
+                    minimum * .001,
+                    maximum * .001,
+                    mean / perLoop100,
+                    stddev / perLoop100,
+                    minimum / perLoop100,
+                    maximum / perLoop100,
+                    meanCycles / perLoop1k,
+                    stddevCycles / perLoop1k,
+                    minCycles / perLoop1k,
+                    maxCycles / perLoop1k);
+
+        }
+    }
+#endif
+};
+
+// ----------------------------------------------------------------------------
+//      ThreadBase
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
+    :   Thread(false /*canCallJava*/),
+        mType(type),
+        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
+        // mChannelMask
+        mChannelCount(0),
+        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+        mParamStatus(NO_ERROR),
+        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
+        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
+        // mName will be set by concrete (non-virtual) subclass
+        mDeathRecipient(new PMDeathRecipient(this))
+{
+}
+
+AudioFlinger::ThreadBase::~ThreadBase()
+{
+    mParamCond.broadcast();
+    // do not lock the mutex in destructor
+    releaseWakeLock_l();
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = mPowerManager->asBinder();
+        binder->unlinkToDeath(mDeathRecipient);
+    }
+}
+
+void AudioFlinger::ThreadBase::exit()
+{
+    ALOGV("ThreadBase::exit");
+    // do any cleanup required for exit to succeed
+    preExit();
+    {
+        // This lock prevents the following race in thread (uniprocessor for illustration):
+        //  if (!exitPending()) {
+        //      // context switch from here to exit()
+        //      // exit() calls requestExit(), what exitPending() observes
+        //      // exit() calls signal(), which is dropped since no waiters
+        //      // context switch back from exit() to here
+        //      mWaitWorkCV.wait(...);
+        //      // now thread is hung
+        //  }
+        AutoMutex lock(mLock);
+        requestExit();
+        mWaitWorkCV.broadcast();
+    }
+    // When Thread::requestExitAndWait is made virtual and this method is renamed to
+    // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
+    requestExitAndWait();
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+    status_t status;
+
+    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
+    Mutex::Autolock _l(mLock);
+
+    mNewParameters.add(keyValuePairs);
+    mWaitWorkCV.signal();
+    // wait condition with timeout in case the thread loop has exited
+    // before the request could be processed
+    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
+        status = mParamStatus;
+        mWaitWorkCV.signal();
+    } else {
+        status = TIMED_OUT;
+    }
+    return status;
+}
+
+void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
+{
+    Mutex::Autolock _l(mLock);
+    sendIoConfigEvent_l(event, param);
+}
+
+// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
+{
+    IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
+    mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
+    ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
+            param);
+    mWaitWorkCV.signal();
+}
+
+// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
+{
+    PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio);
+    mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent));
+    ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d",
+          mConfigEvents.size(), pid, tid, prio);
+    mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+    mLock.lock();
+    while (!mConfigEvents.isEmpty()) {
+        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+        ConfigEvent *event = mConfigEvents[0];
+        mConfigEvents.removeAt(0);
+        // release mLock before locking AudioFlinger mLock: lock order is always
+        // AudioFlinger then ThreadBase to avoid cross deadlock
+        mLock.unlock();
+        switch(event->type()) {
+            case CFG_EVENT_PRIO: {
+                PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
+                int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
+                if (err != 0) {
+                    ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
+                          "error %d",
+                          prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
+                }
+            } break;
+            case CFG_EVENT_IO: {
+                IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
+                mAudioFlinger->mLock.lock();
+                audioConfigChanged_l(ioEvent->event(), ioEvent->param());
+                mAudioFlinger->mLock.unlock();
+            } break;
+            default:
+                ALOGE("processConfigEvents() unknown event type %d", event->type());
+                break;
+        }
+        delete event;
+        mLock.lock();
+    }
+    mLock.unlock();
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = AudioFlinger::dumpTryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
+        write(fd, buffer, strlen(buffer));
+    }
+
+    snprintf(buffer, SIZE, "io handle: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "TID: %d\n", getTid());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
+    result.append(buffer);
+    result.append(" Index Command");
+    for (size_t i = 0; i < mNewParameters.size(); ++i) {
+        snprintf(buffer, SIZE, "\n %02d    ", i);
+        result.append(buffer);
+        result.append(mNewParameters[i]);
+    }
+
+    snprintf(buffer, SIZE, "\n\nPending config events: \n");
+    result.append(buffer);
+    for (size_t i = 0; i < mConfigEvents.size(); i++) {
+        mConfigEvents[i]->dump(buffer, SIZE);
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+
+    if (locked) {
+        mLock.unlock();
+    }
+}
+
+void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+    write(fd, buffer, strlen(buffer));
+
+    for (size_t i = 0; i < mEffectChains.size(); ++i) {
+        sp<EffectChain> chain = mEffectChains[i];
+        if (chain != 0) {
+            chain->dump(fd, args);
+        }
+    }
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock()
+{
+    Mutex::Autolock _l(mLock);
+    acquireWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock_l()
+{
+    if (mPowerManager == 0) {
+        // use checkService() to avoid blocking if power service is not up yet
+        sp<IBinder> binder =
+            defaultServiceManager()->checkService(String16("power"));
+        if (binder == 0) {
+            ALOGW("Thread %s cannot connect to the power manager service", mName);
+        } else {
+            mPowerManager = interface_cast<IPowerManager>(binder);
+            binder->linkToDeath(mDeathRecipient);
+        }
+    }
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = new BBinder();
+        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
+                                                         binder,
+                                                         String16(mName));
+        if (status == NO_ERROR) {
+            mWakeLockToken = binder;
+        }
+        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+    }
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock()
+{
+    Mutex::Autolock _l(mLock);
+    releaseWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock_l()
+{
+    if (mWakeLockToken != 0) {
+        ALOGV("releaseWakeLock_l() %s", mName);
+        if (mPowerManager != 0) {
+            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
+        }
+        mWakeLockToken.clear();
+    }
+}
+
+void AudioFlinger::ThreadBase::clearPowerManager()
+{
+    Mutex::Autolock _l(mLock);
+    releaseWakeLock_l();
+    mPowerManager.clear();
+}
+
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        thread->clearPowerManager();
+    }
+    ALOGW("power manager service died !!!");
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended(
+        const effect_uuid_t *type, bool suspend, int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    setEffectSuspended_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended_l(
+        const effect_uuid_t *type, bool suspend, int sessionId)
+{
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    if (chain != 0) {
+        if (type != NULL) {
+            chain->setEffectSuspended_l(type, suspend);
+        } else {
+            chain->setEffectSuspendedAll_l(suspend);
+        }
+    }
+
+    updateSuspendedSessions_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
+{
+    ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
+    if (index < 0) {
+        return;
+    }
+
+    const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
+            mSuspendedSessions.valueAt(index);
+
+    for (size_t i = 0; i < sessionEffects.size(); i++) {
+        sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+        for (int j = 0; j < desc->mRefCount; j++) {
+            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
+                chain->setEffectSuspendedAll_l(true);
+            } else {
+                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
+                    desc->mType.timeLow);
+                chain->setEffectSuspended_l(&desc->mType, true);
+            }
+        }
+    }
+}
+
+void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
+                                                         bool suspend,
+                                                         int sessionId)
+{
+    ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
+
+    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
+
+    if (suspend) {
+        if (index >= 0) {
+            sessionEffects = mSuspendedSessions.valueAt(index);
+        } else {
+            mSuspendedSessions.add(sessionId, sessionEffects);
+        }
+    } else {
+        if (index < 0) {
+            return;
+        }
+        sessionEffects = mSuspendedSessions.valueAt(index);
+    }
+
+
+    int key = EffectChain::kKeyForSuspendAll;
+    if (type != NULL) {
+        key = type->timeLow;
+    }
+    index = sessionEffects.indexOfKey(key);
+
+    sp<SuspendedSessionDesc> desc;
+    if (suspend) {
+        if (index >= 0) {
+            desc = sessionEffects.valueAt(index);
+        } else {
+            desc = new SuspendedSessionDesc();
+            if (type != NULL) {
+                desc->mType = *type;
+            }
+            sessionEffects.add(key, desc);
+            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
+        }
+        desc->mRefCount++;
+    } else {
+        if (index < 0) {
+            return;
+        }
+        desc = sessionEffects.valueAt(index);
+        if (--desc->mRefCount == 0) {
+            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
+            sessionEffects.removeItemsAt(index);
+            if (sessionEffects.isEmpty()) {
+                ALOGV("updateSuspendedSessions_l() restore removing session %d",
+                                 sessionId);
+                mSuspendedSessions.removeItem(sessionId);
+            }
+        }
+    }
+    if (!sessionEffects.isEmpty()) {
+        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
+    }
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+                                                            bool enabled,
+                                                            int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
+                                                            bool enabled,
+                                                            int sessionId)
+{
+    if (mType != RECORD) {
+        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
+        // another session. This gives the priority to well behaved effect control panels
+        // and applications not using global effects.
+        // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
+        // global effects
+        if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
+            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
+        }
+    }
+
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    if (chain != 0) {
+        chain->checkSuspendOnEffectEnabled(effect, enabled);
+    }
+}
+
+// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
+        const sp<AudioFlinger::Client>& client,
+        const sp<IEffectClient>& effectClient,
+        int32_t priority,
+        int sessionId,
+        effect_descriptor_t *desc,
+        int *enabled,
+        status_t *status
+        )
+{
+    sp<EffectModule> effect;
+    sp<EffectHandle> handle;
+    status_t lStatus;
+    sp<EffectChain> chain;
+    bool chainCreated = false;
+    bool effectCreated = false;
+    bool effectRegistered = false;
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGW("createEffect_l() Audio driver not initialized.");
+        goto Exit;
+    }
+
+    // Do not allow effects with session ID 0 on direct output or duplicating threads
+    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
+    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
+        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
+                desc->name, sessionId);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+    // Only Pre processor effects are allowed on input threads and only on input threads
+    if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
+        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
+                desc->name, desc->flags, mType);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        // check for existing effect chain with the requested audio session
+        chain = getEffectChain_l(sessionId);
+        if (chain == 0) {
+            // create a new chain for this session
+            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
+            chain = new EffectChain(this, sessionId);
+            addEffectChain_l(chain);
+            chain->setStrategy(getStrategyForSession_l(sessionId));
+            chainCreated = true;
+        } else {
+            effect = chain->getEffectFromDesc_l(desc);
+        }
+
+        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
+
+        if (effect == 0) {
+            int id = mAudioFlinger->nextUniqueId();
+            // Check CPU and memory usage
+            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectRegistered = true;
+            // create a new effect module if none present in the chain
+            effect = new EffectModule(this, chain, desc, id, sessionId);
+            lStatus = effect->status();
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            lStatus = chain->addEffect_l(effect);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectCreated = true;
+
+            effect->setDevice(mOutDevice);
+            effect->setDevice(mInDevice);
+            effect->setMode(mAudioFlinger->getMode());
+            effect->setAudioSource(mAudioSource);
+        }
+        // create effect handle and connect it to effect module
+        handle = new EffectHandle(effect, client, effectClient, priority);
+        lStatus = effect->addHandle(handle.get());
+        if (enabled != NULL) {
+            *enabled = (int)effect->isEnabled();
+        }
+    }
+
+Exit:
+    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
+        Mutex::Autolock _l(mLock);
+        if (effectCreated) {
+            chain->removeEffect_l(effect);
+        }
+        if (effectRegistered) {
+            AudioSystem::unregisterEffect(effect->id());
+        }
+        if (chainCreated) {
+            removeEffectChain_l(chain);
+        }
+        handle.clear();
+    }
+
+    if (status != NULL) {
+        *status = lStatus;
+    }
+    return handle;
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
+{
+    Mutex::Autolock _l(mLock);
+    return getEffect_l(sessionId, effectId);
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
+{
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
+}
+
+// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
+// PlaybackThread::mLock held
+status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
+{
+    // check for existing effect chain with the requested audio session
+    int sessionId = effect->sessionId();
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    bool chainCreated = false;
+
+    if (chain == 0) {
+        // create a new chain for this session
+        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
+        chain = new EffectChain(this, sessionId);
+        addEffectChain_l(chain);
+        chain->setStrategy(getStrategyForSession_l(sessionId));
+        chainCreated = true;
+    }
+    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
+
+    if (chain->getEffectFromId_l(effect->id()) != 0) {
+        ALOGW("addEffect_l() %p effect %s already present in chain %p",
+                this, effect->desc().name, chain.get());
+        return BAD_VALUE;
+    }
+
+    status_t status = chain->addEffect_l(effect);
+    if (status != NO_ERROR) {
+        if (chainCreated) {
+            removeEffectChain_l(chain);
+        }
+        return status;
+    }
+
+    effect->setDevice(mOutDevice);
+    effect->setDevice(mInDevice);
+    effect->setMode(mAudioFlinger->getMode());
+    effect->setAudioSource(mAudioSource);
+    return NO_ERROR;
+}
+
+void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
+
+    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
+    effect_descriptor_t desc = effect->desc();
+    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        detachAuxEffect_l(effect->id());
+    }
+
+    sp<EffectChain> chain = effect->chain().promote();
+    if (chain != 0) {
+        // remove effect chain if removing last effect
+        if (chain->removeEffect_l(effect) == 0) {
+            removeEffectChain_l(chain);
+        }
+    } else {
+        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
+    }
+}
+
+void AudioFlinger::ThreadBase::lockEffectChains_l(
+        Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+    effectChains = mEffectChains;
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        mEffectChains[i]->lock();
+    }
+}
+
+void AudioFlinger::ThreadBase::unlockEffectChains(
+        const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+    for (size_t i = 0; i < effectChains.size(); i++) {
+        effectChains[i]->unlock();
+    }
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    return getEffectChain_l(sessionId);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
+{
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() == sessionId) {
+            return mEffectChains[i];
+        }
+    }
+    return 0;
+}
+
+void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffectChains[i]->setMode_l(mode);
+    }
+}
+
+void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
+                                                    EffectHandle *handle,
+                                                    bool unpinIfLast) {
+
+    Mutex::Autolock _l(mLock);
+    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
+    // delete the effect module if removing last handle on it
+    if (effect->removeHandle(handle) == 0) {
+        if (!effect->isPinned() || unpinIfLast) {
+            removeEffect_l(effect);
+            AudioSystem::unregisterEffect(effect->id());
+        }
+    }
+}
+
+// ----------------------------------------------------------------------------
+//      Playback
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
+                                             AudioStreamOut* output,
+                                             audio_io_handle_t id,
+                                             audio_devices_t device,
+                                             type_t type)
+    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
+        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+        // mStreamTypes[] initialized in constructor body
+        mOutput(output),
+        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+        mMixerStatus(MIXER_IDLE),
+        mMixerStatusIgnoringFastTracks(MIXER_IDLE),
+        standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
+        mScreenState(AudioFlinger::mScreenState),
+        // index 0 is reserved for normal mixer's submix
+        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
+{
+    snprintf(mName, kNameLength, "AudioOut_%X", id);
+
+    // Assumes constructor is called by AudioFlinger with it's mLock held, but
+    // it would be safer to explicitly pass initial masterVolume/masterMute as
+    // parameter.
+    //
+    // If the HAL we are using has support for master volume or master mute,
+    // then do not attenuate or mute during mixing (just leave the volume at 1.0
+    // and the mute set to false).
+    mMasterVolume = audioFlinger->masterVolume_l();
+    mMasterMute = audioFlinger->masterMute_l();
+    if (mOutput && mOutput->audioHwDev) {
+        if (mOutput->audioHwDev->canSetMasterVolume()) {
+            mMasterVolume = 1.0;
+        }
+
+        if (mOutput->audioHwDev->canSetMasterMute()) {
+            mMasterMute = false;
+        }
+    }
+
+    readOutputParameters();
+
+    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
+    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
+    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
+            stream = (audio_stream_type_t) (stream + 1)) {
+        mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
+        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+    }
+    // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
+    // because mAudioFlinger doesn't have one to copy from
+}
+
+AudioFlinger::PlaybackThread::~PlaybackThread()
+{
+    delete [] mMixBuffer;
+}
+
+void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
+    for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
+        const stream_type_t *st = &mStreamTypes[i];
+        if (i > 0) {
+            result.appendFormat(", ");
+        }
+        result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
+        if (st->mute) {
+            result.append("M");
+        }
+    }
+    result.append("\n");
+    write(fd, result.string(), result.length());
+    result.clear();
+
+    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
+    result.append(buffer);
+    Track::appendDumpHeader(result);
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
+    result.append(buffer);
+    Track::appendDumpHeader(result);
+    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+        sp<Track> track = mActiveTracks[i].promote();
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+    write(fd, result.string(), result.size());
+
+    // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
+    FastTrackUnderruns underruns = getFastTrackUnderruns(0);
+    fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+            underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+}
+
+void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
+            ns2ms(systemTime() - mLastWriteTime));
+    result.append(buffer);
+    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+
+    dumpBase(fd, args);
+}
+
+// Thread virtuals
+status_t AudioFlinger::PlaybackThread::readyToRun()
+{
+    status_t status = initCheck();
+    if (status == NO_ERROR) {
+        ALOGI("AudioFlinger's thread %p ready to run", this);
+    } else {
+        ALOGE("No working audio driver found.");
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// ThreadBase virtuals
+void AudioFlinger::PlaybackThread::preExit()
+{
+    ALOGV("  preExit()");
+    // FIXME this is using hard-coded strings but in the future, this functionality will be
+    //       converted to use audio HAL extensions required to support tunneling
+    mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
+}
+
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+        const sp<AudioFlinger::Client>& client,
+        audio_stream_type_t streamType,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        const sp<IMemory>& sharedBuffer,
+        int sessionId,
+        IAudioFlinger::track_flags_t *flags,
+        pid_t tid,
+        status_t *status)
+{
+    sp<Track> track;
+    status_t lStatus;
+
+    bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
+
+    // client expresses a preference for FAST, but we get the final say
+    if (*flags & IAudioFlinger::TRACK_FAST) {
+      if (
+            // not timed
+            (!isTimed) &&
+            // either of these use cases:
+            (
+              // use case 1: shared buffer with any frame count
+              (
+                (sharedBuffer != 0)
+              ) ||
+              // use case 2: callback handler and frame count is default or at least as large as HAL
+              (
+                (tid != -1) &&
+                ((frameCount == 0) ||
+                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+              )
+            ) &&
+            // PCM data
+            audio_is_linear_pcm(format) &&
+            // mono or stereo
+            ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
+              (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
+#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
+            // hardware sample rate
+            (sampleRate == mSampleRate) &&
+#endif
+            // normal mixer has an associated fast mixer
+            hasFastMixer() &&
+            // there are sufficient fast track slots available
+            (mFastTrackAvailMask != 0)
+            // FIXME test that MixerThread for this fast track has a capable output HAL
+            // FIXME add a permission test also?
+        ) {
+        // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
+        if (frameCount == 0) {
+            frameCount = mFrameCount * kFastTrackMultiplier;
+        }
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
+                frameCount, mFrameCount);
+      } else {
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
+                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
+                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
+                isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
+                audio_is_linear_pcm(format),
+                channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
+        *flags &= ~IAudioFlinger::TRACK_FAST;
+        // For compatibility with AudioTrack calculation, buffer depth is forced
+        // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+        // This is probably too conservative, but legacy application code may depend on it.
+        // If you change this calculation, also review the start threshold which is related.
+        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
+        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
+        if (minBufCount < 2) {
+            minBufCount = 2;
+        }
+        size_t minFrameCount = mNormalFrameCount * minBufCount;
+        if (frameCount < minFrameCount) {
+            frameCount = minFrameCount;
+        }
+      }
+    }
+
+    if (mType == DIRECT) {
+        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
+                        "for output %p with format %d",
+                        sampleRate, format, channelMask, mOutput, mFormat);
+                lStatus = BAD_VALUE;
+                goto Exit;
+            }
+        }
+    } else {
+        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+        if (sampleRate > mSampleRate*2) {
+            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+    }
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGE("Audio driver not initialized.");
+        goto Exit;
+    }
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        // all tracks in same audio session must share the same routing strategy otherwise
+        // conflicts will happen when tracks are moved from one output to another by audio policy
+        // manager
+        uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            sp<Track> t = mTracks[i];
+            if (t != 0 && !t->isOutputTrack()) {
+                uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
+                if (sessionId == t->sessionId() && strategy != actual) {
+                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+                            strategy, actual);
+                    lStatus = BAD_VALUE;
+                    goto Exit;
+                }
+            }
+        }
+
+        if (!isTimed) {
+            track = new Track(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId, *flags);
+        } else {
+            track = TimedTrack::create(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId);
+        }
+        if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+
+        sp<EffectChain> chain = getEffectChain_l(sessionId);
+        if (chain != 0) {
+            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
+            track->setMainBuffer(chain->inBuffer());
+            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+            chain->incTrackCnt();
+        }
+
+        if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
+            pid_t callingPid = IPCThreadState::self()->getCallingPid();
+            // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
+            // so ask activity manager to do this on our behalf
+            sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
+        }
+    }
+
+    lStatus = NO_ERROR;
+
+Exit:
+    if (status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
+{
+    return latency;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+    Mutex::Autolock _l(mLock);
+    return latency_l();
+}
+uint32_t AudioFlinger::PlaybackThread::latency_l() const
+{
+    if (initCheck() == NO_ERROR) {
+        return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
+    } else {
+        return 0;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+    Mutex::Autolock _l(mLock);
+    // Don't apply master volume in SW if our HAL can do it for us.
+    if (mOutput && mOutput->audioHwDev &&
+        mOutput->audioHwDev->canSetMasterVolume()) {
+        mMasterVolume = 1.0;
+    } else {
+        mMasterVolume = value;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+    Mutex::Autolock _l(mLock);
+    // Don't apply master mute in SW if our HAL can do it for us.
+    if (mOutput && mOutput->audioHwDev &&
+        mOutput->audioHwDev->canSetMasterMute()) {
+        mMasterMute = false;
+    } else {
+        mMasterMute = muted;
+    }
+}
+
+void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+{
+    Mutex::Autolock _l(mLock);
+    mStreamTypes[stream].volume = value;
+}
+
+void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+{
+    Mutex::Autolock _l(mLock);
+    mStreamTypes[stream].mute = muted;
+}
+
+float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
+{
+    Mutex::Autolock _l(mLock);
+    return mStreamTypes[stream].volume;
+}
+
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+{
+    status_t status = ALREADY_EXISTS;
+
+    // set retry count for buffer fill
+    track->mRetryCount = kMaxTrackStartupRetries;
+    if (mActiveTracks.indexOf(track) < 0) {
+        // the track is newly added, make sure it fills up all its
+        // buffers before playing. This is to ensure the client will
+        // effectively get the latency it requested.
+        track->mFillingUpStatus = Track::FS_FILLING;
+        track->mResetDone = false;
+        track->mPresentationCompleteFrames = 0;
+        mActiveTracks.add(track);
+        if (track->mainBuffer() != mMixBuffer) {
+            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+            if (chain != 0) {
+                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
+                        track->sessionId());
+                chain->incActiveTrackCnt();
+            }
+        }
+
+        status = NO_ERROR;
+    }
+
+    ALOGV("mWaitWorkCV.broadcast");
+    mWaitWorkCV.broadcast();
+
+    return status;
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+{
+    track->mState = TrackBase::TERMINATED;
+    // active tracks are removed by threadLoop()
+    if (mActiveTracks.indexOf(track) < 0) {
+        removeTrack_l(track);
+    }
+}
+
+void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
+{
+    track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+    mTracks.remove(track);
+    deleteTrackName_l(track->name());
+    // redundant as track is about to be destroyed, for dumpsys only
+    track->mName = -1;
+    if (track->isFastTrack()) {
+        int index = track->mFastIndex;
+        ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
+        ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
+        mFastTrackAvailMask |= 1 << index;
+        // redundant as track is about to be destroyed, for dumpsys only
+        track->mFastIndex = -1;
+    }
+    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+    if (chain != 0) {
+        chain->decTrackCnt();
+    }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+    String8 out_s8 = String8("");
+    char *s;
+
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return out_s8;
+    }
+
+    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
+    out_s8 = String8(s);
+    free(s);
+    return out_s8;
+}
+
+// audioConfigChanged_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = NULL;
+
+    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
+            param);
+
+    switch (event) {
+    case AudioSystem::OUTPUT_OPENED:
+    case AudioSystem::OUTPUT_CONFIG_CHANGED:
+        desc.channels = mChannelMask;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mNormalFrameCount; // FIXME see
+                                             // AudioFlinger::frameCount(audio_io_handle_t)
+        desc.latency = latency();
+        param2 = &desc;
+        break;
+
+    case AudioSystem::STREAM_CONFIG_CHANGED:
+        param2 = &param;
+    case AudioSystem::OUTPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
+    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
+    mChannelCount = (uint16_t)popcount(mChannelMask);
+    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
+    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+    if (mFrameCount & 15) {
+        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
+                mFrameCount);
+    }
+
+    // Calculate size of normal mix buffer relative to the HAL output buffer size
+    double multiplier = 1.0;
+    if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
+            kUseFastMixer == FastMixer_Dynamic)) {
+        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
+        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
+        // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
+        minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
+        maxNormalFrameCount = maxNormalFrameCount & ~15;
+        if (maxNormalFrameCount < minNormalFrameCount) {
+            maxNormalFrameCount = minNormalFrameCount;
+        }
+        multiplier = (double) minNormalFrameCount / (double) mFrameCount;
+        if (multiplier <= 1.0) {
+            multiplier = 1.0;
+        } else if (multiplier <= 2.0) {
+            if (2 * mFrameCount <= maxNormalFrameCount) {
+                multiplier = 2.0;
+            } else {
+                multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
+            }
+        } else {
+            // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
+            // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
+            // track, but we sometimes have to do this to satisfy the maximum frame count
+            // constraint)
+            // FIXME this rounding up should not be done if no HAL SRC
+            uint32_t truncMult = (uint32_t) multiplier;
+            if ((truncMult & 1)) {
+                if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
+                    ++truncMult;
+                }
+            }
+            multiplier = (double) truncMult;
+        }
+    }
+    mNormalFrameCount = multiplier * mFrameCount;
+    // round up to nearest 16 frames to satisfy AudioMixer
+    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+            mNormalFrameCount);
+
+    delete[] mMixBuffer;
+    mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
+    memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+
+    // force reconfiguration of effect chains and engines to take new buffer size and audio
+    // parameters into account
+    // Note that mLock is not held when readOutputParameters() is called from the constructor
+    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
+    // matter.
+    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
+    Vector< sp<EffectChain> > effectChains = mEffectChains;
+    for (size_t i = 0; i < effectChains.size(); i ++) {
+        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
+    }
+}
+
+
+status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+{
+    if (halFrames == NULL || dspFrames == NULL) {
+        return BAD_VALUE;
+    }
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return INVALID_OPERATION;
+    }
+    size_t framesWritten = mBytesWritten / mFrameSize;
+    *halFrames = framesWritten;
+
+    if (isSuspended()) {
+        // return an estimation of rendered frames when the output is suspended
+        size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
+        *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
+        return NO_ERROR;
+    } else {
+        return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+    }
+}
+
+uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
+{
+    Mutex::Autolock _l(mLock);
+    uint32_t result = 0;
+    if (getEffectChain_l(sessionId) != 0) {
+        result = EFFECT_SESSION;
+    }
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (sessionId == track->sessionId() &&
+                !(track->mCblk->flags & CBLK_INVALID)) {
+            result |= TRACK_SESSION;
+            break;
+        }
+    }
+
+    return result;
+}
+
+uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
+{
+    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
+    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
+    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
+        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+    }
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        sp<Track> track = mTracks[i];
+        if (sessionId == track->sessionId() &&
+                !(track->mCblk->flags & CBLK_INVALID)) {
+            return AudioSystem::getStrategyForStream(track->streamType());
+        }
+    }
+    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+}
+
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+{
+    Mutex::Autolock _l(mLock);
+    return mOutput;
+}
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+{
+    Mutex::Autolock _l(mLock);
+    AudioStreamOut *output = mOutput;
+    mOutput = NULL;
+    // FIXME FastMixer might also have a raw ptr to mOutputSink;
+    //       must push a NULL and wait for ack
+    mOutputSink.clear();
+    mPipeSink.clear();
+    mNormalSink.clear();
+    return output;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::PlaybackThread::stream() const
+{
+    if (mOutput == NULL) {
+        return NULL;
+    }
+    return &mOutput->stream->common;
+}
+
+uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
+{
+    return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+    if (!isValidSyncEvent(event)) {
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (event->triggerSession() == track->sessionId()) {
+            (void) track->setSyncEvent(event);
+            return NO_ERROR;
+        }
+    }
+
+    return NAME_NOT_FOUND;
+}
+
+bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+    return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
+}
+
+void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+        const Vector< sp<Track> >& tracksToRemove)
+{
+    size_t count = tracksToRemove.size();
+    if (CC_UNLIKELY(count)) {
+        for (size_t i = 0 ; i < count ; i++) {
+            const sp<Track>& track = tracksToRemove.itemAt(i);
+            if ((track->sharedBuffer() != 0) &&
+                    (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
+                AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
+            }
+        }
+    }
+
+}
+
+void AudioFlinger::PlaybackThread::checkSilentMode_l()
+{
+    if (!mMasterMute) {
+        char value[PROPERTY_VALUE_MAX];
+        if (property_get("ro.audio.silent", value, "0") > 0) {
+            char *endptr;
+            unsigned long ul = strtoul(value, &endptr, 0);
+            if (*endptr == '\0' && ul != 0) {
+                ALOGD("Silence is golden");
+                // The setprop command will not allow a property to be changed after
+                // the first time it is set, so we don't have to worry about un-muting.
+                setMasterMute_l(true);
+            }
+        }
+    }
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_write()
+{
+    // FIXME rewrite to reduce number of system calls
+    mLastWriteTime = systemTime();
+    mInWrite = true;
+    int bytesWritten;
+
+    // If an NBAIO sink is present, use it to write the normal mixer's submix
+    if (mNormalSink != 0) {
+#define mBitShift 2 // FIXME
+        size_t count = mixBufferSize >> mBitShift;
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+        Tracer::traceBegin(ATRACE_TAG, "write");
+#endif
+        // update the setpoint when AudioFlinger::mScreenState changes
+        uint32_t screenState = AudioFlinger::mScreenState;
+        if (screenState != mScreenState) {
+            mScreenState = screenState;
+            MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+            if (pipe != NULL) {
+                pipe->setAvgFrames((mScreenState & 1) ?
+                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+            }
+        }
+        ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+        Tracer::traceEnd(ATRACE_TAG);
+#endif
+        if (framesWritten > 0) {
+            bytesWritten = framesWritten << mBitShift;
+        } else {
+            bytesWritten = framesWritten;
+        }
+    // otherwise use the HAL / AudioStreamOut directly
+    } else {
+        // Direct output thread.
+        bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
+    }
+
+    if (bytesWritten > 0) {
+        mBytesWritten += mixBufferSize;
+    }
+    mNumWrites++;
+    mInWrite = false;
+}
+
+/*
+The derived values that are cached:
+ - mixBufferSize from frame count * frame size
+ - activeSleepTime from activeSleepTimeUs()
+ - idleSleepTime from idleSleepTimeUs()
+ - standbyDelay from mActiveSleepTimeUs (DIRECT only)
+ - maxPeriod from frame count and sample rate (MIXER only)
+
+The parameters that affect these derived values are:
+ - frame count
+ - frame size
+ - sample rate
+ - device type: A2DP or not
+ - device latency
+ - format: PCM or not
+ - active sleep time
+ - idle sleep time
+*/
+
+void AudioFlinger::PlaybackThread::cacheParameters_l()
+{
+    mixBufferSize = mNormalFrameCount * mFrameSize;
+    activeSleepTime = activeSleepTimeUs();
+    idleSleepTime = idleSleepTimeUs();
+}
+
+void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+{
+    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
+            this,  streamType, mTracks.size());
+    Mutex::Autolock _l(mLock);
+
+    size_t size = mTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = mTracks[i];
+        if (t->streamType() == streamType) {
+            android_atomic_or(CBLK_INVALID, &t->mCblk->flags);
+            t->mCblk->cv.signal();
+        }
+    }
+}
+
+status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+    int16_t *buffer = mMixBuffer;
+    bool ownsBuffer = false;
+
+    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
+    if (session > 0) {
+        // Only one effect chain can be present in direct output thread and it uses
+        // the mix buffer as input
+        if (mType != DIRECT) {
+            size_t numSamples = mNormalFrameCount * mChannelCount;
+            buffer = new int16_t[numSamples];
+            memset(buffer, 0, numSamples * sizeof(int16_t));
+            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
+            ownsBuffer = true;
+        }
+
+        // Attach all tracks with same session ID to this chain.
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            sp<Track> track = mTracks[i];
+            if (session == track->sessionId()) {
+                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
+                        buffer);
+                track->setMainBuffer(buffer);
+                chain->incTrackCnt();
+            }
+        }
+
+        // indicate all active tracks in the chain
+        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+            sp<Track> track = mActiveTracks[i].promote();
+            if (track == 0) {
+                continue;
+            }
+            if (session == track->sessionId()) {
+                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
+                chain->incActiveTrackCnt();
+            }
+        }
+    }
+
+    chain->setInBuffer(buffer, ownsBuffer);
+    chain->setOutBuffer(mMixBuffer);
+    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
+    // chains list in order to be processed last as it contains output stage effects
+    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
+    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
+    // after track specific effects and before output stage
+    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
+    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
+    // Effect chain for other sessions are inserted at beginning of effect
+    // chains list to be processed before output mix effects. Relative order between other
+    // sessions is not important
+    size_t size = mEffectChains.size();
+    size_t i = 0;
+    for (i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() < session) {
+            break;
+        }
+    }
+    mEffectChains.insertAt(chain, i);
+    checkSuspendOnAddEffectChain_l(chain);
+
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+
+    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
+
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        if (chain == mEffectChains[i]) {
+            mEffectChains.removeAt(i);
+            // detach all active tracks from the chain
+            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+                sp<Track> track = mActiveTracks[i].promote();
+                if (track == 0) {
+                    continue;
+                }
+                if (session == track->sessionId()) {
+                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
+                            chain.get(), session);
+                    chain->decActiveTrackCnt();
+                }
+            }
+
+            // detach all tracks with same session ID from this chain
+            for (size_t i = 0; i < mTracks.size(); ++i) {
+                sp<Track> track = mTracks[i];
+                if (session == track->sessionId()) {
+                    track->setMainBuffer(mMixBuffer);
+                    chain->decTrackCnt();
+                }
+            }
+            break;
+        }
+    }
+    return mEffectChains.size();
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect(
+        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    Mutex::Autolock _l(mLock);
+    return attachAuxEffect_l(track, EffectId);
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
+        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    status_t status = NO_ERROR;
+
+    if (EffectId == 0) {
+        track->setAuxBuffer(0, NULL);
+    } else {
+        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
+        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+        if (effect != 0) {
+            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
+            } else {
+                status = INVALID_OPERATION;
+            }
+        } else {
+            status = BAD_VALUE;
+        }
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+{
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track->auxEffectId() == effectId) {
+            attachAuxEffect_l(track, 0);
+        }
+    }
+}
+
+bool AudioFlinger::PlaybackThread::threadLoop()
+{
+    Vector< sp<Track> > tracksToRemove;
+
+    standbyTime = systemTime();
+
+    // MIXER
+    nsecs_t lastWarning = 0;
+
+    // DUPLICATING
+    // FIXME could this be made local to while loop?
+    writeFrames = 0;
+
+    cacheParameters_l();
+    sleepTime = idleSleepTime;
+
+    if (mType == MIXER) {
+        sleepTimeShift = 0;
+    }
+
+    CpuStats cpuStats;
+    const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
+
+    acquireWakeLock();
+
+    while (!exitPending())
+    {
+        cpuStats.sample(myName);
+
+        Vector< sp<EffectChain> > effectChains;
+
+        processConfigEvents();
+
+        { // scope for mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                cacheParameters_l();
+            }
+
+            saveOutputTracks();
+
+            // put audio hardware into standby after short delay
+            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+                        isSuspended())) {
+                if (!mStandby) {
+
+                    threadLoop_standby();
+
+                    mStandby = true;
+                }
+
+                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    clearOutputTracks();
+
+                    if (exitPending()) {
+                        break;
+                    }
+
+                    releaseWakeLock_l();
+                    // wait until we have something to do...
+                    ALOGV("%s going to sleep", myName.string());
+                    mWaitWorkCV.wait(mLock);
+                    ALOGV("%s waking up", myName.string());
+                    acquireWakeLock_l();
+
+                    mMixerStatus = MIXER_IDLE;
+                    mMixerStatusIgnoringFastTracks = MIXER_IDLE;
+                    mBytesWritten = 0;
+
+                    checkSilentMode_l();
+
+                    standbyTime = systemTime() + standbyDelay;
+                    sleepTime = idleSleepTime;
+                    if (mType == MIXER) {
+                        sleepTimeShift = 0;
+                    }
+
+                    continue;
+                }
+            }
+
+            // mMixerStatusIgnoringFastTracks is also updated internally
+            mMixerStatus = prepareTracks_l(&tracksToRemove);
+
+            // prevent any changes in effect chain list and in each effect chain
+            // during mixing and effect process as the audio buffers could be deleted
+            // or modified if an effect is created or deleted
+            lockEffectChains_l(effectChains);
+        }
+
+        if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
+            threadLoop_mix();
+        } else {
+            threadLoop_sleepTime();
+        }
+
+        if (isSuspended()) {
+            sleepTime = suspendSleepTimeUs();
+            mBytesWritten += mixBufferSize;
+        }
+
+        // only process effects if we're going to write
+        if (sleepTime == 0) {
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+        }
+
+        // enable changes in effect chain
+        unlockEffectChains(effectChains);
+
+        // sleepTime == 0 means we must write to audio hardware
+        if (sleepTime == 0) {
+
+            threadLoop_write();
+
+if (mType == MIXER) {
+            // write blocked detection
+            nsecs_t now = systemTime();
+            nsecs_t delta = now - mLastWriteTime;
+            if (!mStandby && delta > maxPeriod) {
+                mNumDelayedWrites++;
+                if ((now - lastWarning) > kWarningThrottleNs) {
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+                    ScopedTrace st(ATRACE_TAG, "underrun");
+#endif
+                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                            ns2ms(delta), mNumDelayedWrites, this);
+                    lastWarning = now;
+                }
+            }
+}
+
+            mStandby = false;
+        } else {
+            usleep(sleepTime);
+        }
+
+        // Finally let go of removed track(s), without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.  This will also mutate and push a new fast mixer state.
+        threadLoop_removeTracks(tracksToRemove);
+        tracksToRemove.clear();
+
+        // FIXME I don't understand the need for this here;
+        //       it was in the original code but maybe the
+        //       assignment in saveOutputTracks() makes this unnecessary?
+        clearOutputTracks();
+
+        // Effect chains will be actually deleted here if they were removed from
+        // mEffectChains list during mixing or effects processing
+        effectChains.clear();
+
+        // FIXME Note that the above .clear() is no longer necessary since effectChains
+        // is now local to this block, but will keep it for now (at least until merge done).
+    }
+
+    // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
+    if (mType == MIXER || mType == DIRECT) {
+        // put output stream into standby mode
+        if (!mStandby) {
+            mOutput->stream->common.standby(&mOutput->stream->common);
+        }
+    }
+
+    releaseWakeLock();
+
+    ALOGV("Thread %p type %d exiting", this, mType);
+    return false;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+        audio_io_handle_t id, audio_devices_t device, type_t type)
+    :   PlaybackThread(audioFlinger, output, id, device, type),
+        // mAudioMixer below
+        // mFastMixer below
+        mFastMixerFutex(0)
+        // mOutputSink below
+        // mPipeSink below
+        // mNormalSink below
+{
+    ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
+    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
+            "mFrameCount=%d, mNormalFrameCount=%d",
+            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
+            mNormalFrameCount);
+    mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+
+    // FIXME - Current mixer implementation only supports stereo output
+    if (mChannelCount != FCC_2) {
+        ALOGE("Invalid audio hardware channel count %d", mChannelCount);
+    }
+
+    // create an NBAIO sink for the HAL output stream, and negotiate
+    mOutputSink = new AudioStreamOutSink(output->stream);
+    size_t numCounterOffers = 0;
+    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
+    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
+    ALOG_ASSERT(index == 0);
+
+    // initialize fast mixer depending on configuration
+    bool initFastMixer;
+    switch (kUseFastMixer) {
+    case FastMixer_Never:
+        initFastMixer = false;
+        break;
+    case FastMixer_Always:
+        initFastMixer = true;
+        break;
+    case FastMixer_Static:
+    case FastMixer_Dynamic:
+        initFastMixer = mFrameCount < mNormalFrameCount;
+        break;
+    }
+    if (initFastMixer) {
+
+        // create a MonoPipe to connect our submix to FastMixer
+        NBAIO_Format format = mOutputSink->format();
+        // This pipe depth compensates for scheduling latency of the normal mixer thread.
+        // When it wakes up after a maximum latency, it runs a few cycles quickly before
+        // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
+        MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
+        const NBAIO_Format offers[1] = {format};
+        size_t numCounterOffers = 0;
+        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        monoPipe->setAvgFrames((mScreenState & 1) ?
+                (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+        mPipeSink = monoPipe;
+
+#ifdef TEE_SINK_FRAMES
+        // create a Pipe to archive a copy of FastMixer's output for dumpsys
+        Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
+        numCounterOffers = 0;
+        index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mTeeSink = teeSink;
+        PipeReader *teeSource = new PipeReader(*teeSink);
+        numCounterOffers = 0;
+        index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mTeeSource = teeSource;
+#endif
+
+        // create fast mixer and configure it initially with just one fast track for our submix
+        mFastMixer = new FastMixer();
+        FastMixerStateQueue *sq = mFastMixer->sq();
+#ifdef STATE_QUEUE_DUMP
+        sq->setObserverDump(&mStateQueueObserverDump);
+        sq->setMutatorDump(&mStateQueueMutatorDump);
+#endif
+        FastMixerState *state = sq->begin();
+        FastTrack *fastTrack = &state->mFastTracks[0];
+        // wrap the source side of the MonoPipe to make it an AudioBufferProvider
+        fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
+        fastTrack->mVolumeProvider = NULL;
+        fastTrack->mGeneration++;
+        state->mFastTracksGen++;
+        state->mTrackMask = 1;
+        // fast mixer will use the HAL output sink
+        state->mOutputSink = mOutputSink.get();
+        state->mOutputSinkGen++;
+        state->mFrameCount = mFrameCount;
+        state->mCommand = FastMixerState::COLD_IDLE;
+        // already done in constructor initialization list
+        //mFastMixerFutex = 0;
+        state->mColdFutexAddr = &mFastMixerFutex;
+        state->mColdGen++;
+        state->mDumpState = &mFastMixerDumpState;
+        state->mTeeSink = mTeeSink.get();
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+
+        // start the fast mixer
+        mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
+        pid_t tid = mFastMixer->getTid();
+        int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+        if (err != 0) {
+            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+                    kPriorityFastMixer, getpid_cached, tid, err);
+        }
+
+#ifdef AUDIO_WATCHDOG
+        // create and start the watchdog
+        mAudioWatchdog = new AudioWatchdog();
+        mAudioWatchdog->setDump(&mAudioWatchdogDump);
+        mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
+        tid = mAudioWatchdog->getTid();
+        err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+        if (err != 0) {
+            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+                    kPriorityFastMixer, getpid_cached, tid, err);
+        }
+#endif
+
+    } else {
+        mFastMixer = NULL;
+    }
+
+    switch (kUseFastMixer) {
+    case FastMixer_Never:
+    case FastMixer_Dynamic:
+        mNormalSink = mOutputSink;
+        break;
+    case FastMixer_Always:
+        mNormalSink = mPipeSink;
+        break;
+    case FastMixer_Static:
+        mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
+        break;
+    }
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (state->mCommand == FastMixerState::COLD_IDLE) {
+            int32_t old = android_atomic_inc(&mFastMixerFutex);
+            if (old == -1) {
+                __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+            }
+        }
+        state->mCommand = FastMixerState::EXIT;
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+        mFastMixer->join();
+        // Though the fast mixer thread has exited, it's state queue is still valid.
+        // We'll use that extract the final state which contains one remaining fast track
+        // corresponding to our sub-mix.
+        state = sq->begin();
+        ALOG_ASSERT(state->mTrackMask == 1);
+        FastTrack *fastTrack = &state->mFastTracks[0];
+        ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
+        delete fastTrack->mBufferProvider;
+        sq->end(false /*didModify*/);
+        delete mFastMixer;
+#ifdef AUDIO_WATCHDOG
+        if (mAudioWatchdog != 0) {
+            mAudioWatchdog->requestExit();
+            mAudioWatchdog->requestExitAndWait();
+            mAudioWatchdog.clear();
+        }
+#endif
+    }
+    delete mAudioMixer;
+}
+
+
+uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
+{
+    if (mFastMixer != NULL) {
+        MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+        latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
+    }
+    return latency;
+}
+
+
+void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
+{
+    PlaybackThread::threadLoop_removeTracks(tracksToRemove);
+}
+
+void AudioFlinger::MixerThread::threadLoop_write()
+{
+    // FIXME we should only do one push per cycle; confirm this is true
+    // Start the fast mixer if it's not already running
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (state->mCommand != FastMixerState::MIX_WRITE &&
+                (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
+            if (state->mCommand == FastMixerState::COLD_IDLE) {
+                int32_t old = android_atomic_inc(&mFastMixerFutex);
+                if (old == -1) {
+                    __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+                }
+#ifdef AUDIO_WATCHDOG
+                if (mAudioWatchdog != 0) {
+                    mAudioWatchdog->resume();
+                }
+#endif
+            }
+            state->mCommand = FastMixerState::MIX_WRITE;
+            sq->end();
+            sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mPipeSink;
+            }
+        } else {
+            sq->end(false /*didModify*/);
+        }
+    }
+    PlaybackThread::threadLoop_write();
+}
+
+void AudioFlinger::MixerThread::threadLoop_standby()
+{
+    // Idle the fast mixer if it's currently running
+    if (mFastMixer != NULL) {
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        if (!(state->mCommand & FastMixerState::IDLE)) {
+            state->mCommand = FastMixerState::COLD_IDLE;
+            state->mColdFutexAddr = &mFastMixerFutex;
+            state->mColdGen++;
+            mFastMixerFutex = 0;
+            sq->end();
+            // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
+            sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mOutputSink;
+            }
+#ifdef AUDIO_WATCHDOG
+            if (mAudioWatchdog != 0) {
+                mAudioWatchdog->pause();
+            }
+#endif
+        } else {
+            sq->end(false /*didModify*/);
+        }
+    }
+    PlaybackThread::threadLoop_standby();
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_standby()
+{
+    ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
+    mOutput->stream->common.standby(&mOutput->stream->common);
+}
+
+void AudioFlinger::MixerThread::threadLoop_mix()
+{
+    // obtain the presentation timestamp of the next output buffer
+    int64_t pts;
+    status_t status = INVALID_OPERATION;
+
+    if (mNormalSink != 0) {
+        status = mNormalSink->getNextWriteTimestamp(&pts);
+    } else {
+        status = mOutputSink->getNextWriteTimestamp(&pts);
+    }
+
+    if (status != NO_ERROR) {
+        pts = AudioBufferProvider::kInvalidPTS;
+    }
+
+    // mix buffers...
+    mAudioMixer->process(pts);
+    // increase sleep time progressively when application underrun condition clears.
+    // Only increase sleep time if the mixer is ready for two consecutive times to avoid
+    // that a steady state of alternating ready/not ready conditions keeps the sleep time
+    // such that we would underrun the audio HAL.
+    if ((sleepTime == 0) && (sleepTimeShift > 0)) {
+        sleepTimeShift--;
+    }
+    sleepTime = 0;
+    standbyTime = systemTime() + standbyDelay;
+    //TODO: delay standby when effects have a tail
+}
+
+void AudioFlinger::MixerThread::threadLoop_sleepTime()
+{
+    // If no tracks are ready, sleep once for the duration of an output
+    // buffer size, then write 0s to the output
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime >> sleepTimeShift;
+            if (sleepTime < kMinThreadSleepTimeUs) {
+                sleepTime = kMinThreadSleepTimeUs;
+            }
+            // reduce sleep time in case of consecutive application underruns to avoid
+            // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
+            // duration we would end up writing less data than needed by the audio HAL if
+            // the condition persists.
+            if (sleepTimeShift < kMaxThreadSleepTimeShift) {
+                sleepTimeShift++;
+            }
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
+        memset (mMixBuffer, 0, mixBufferSize);
+        sleepTime = 0;
+        ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
+                "anticipated start");
+    }
+    // TODO add standby time extension fct of effect tail
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+        Vector< sp<Track> > *tracksToRemove)
+{
+
+    mixer_state mixerStatus = MIXER_IDLE;
+    // find out which tracks need to be processed
+    size_t count = mActiveTracks.size();
+    size_t mixedTracks = 0;
+    size_t tracksWithEffect = 0;
+    // counts only _active_ fast tracks
+    size_t fastTracks = 0;
+    uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
+
+    float masterVolume = mMasterVolume;
+    bool masterMute = mMasterMute;
+
+    if (masterMute) {
+        masterVolume = 0;
+    }
+    // Delegate master volume control to effect in output mix effect chain if needed
+    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+    if (chain != 0) {
+        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
+        chain->setVolume_l(&v, &v);
+        masterVolume = (float)((v + (1 << 23)) >> 24);
+        chain.clear();
+    }
+
+    // prepare a new state to push
+    FastMixerStateQueue *sq = NULL;
+    FastMixerState *state = NULL;
+    bool didModify = false;
+    FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
+    if (mFastMixer != NULL) {
+        sq = mFastMixer->sq();
+        state = sq->begin();
+    }
+
+    for (size_t i=0 ; i<count ; i++) {
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) {
+            continue;
+        }
+
+        // this const just means the local variable doesn't change
+        Track* const track = t.get();
+
+        // process fast tracks
+        if (track->isFastTrack()) {
+
+            // It's theoretically possible (though unlikely) for a fast track to be created
+            // and then removed within the same normal mix cycle.  This is not a problem, as
+            // the track never becomes active so it's fast mixer slot is never touched.
+            // The converse, of removing an (active) track and then creating a new track
+            // at the identical fast mixer slot within the same normal mix cycle,
+            // is impossible because the slot isn't marked available until the end of each cycle.
+            int j = track->mFastIndex;
+            ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
+            ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
+            FastTrack *fastTrack = &state->mFastTracks[j];
+
+            // Determine whether the track is currently in underrun condition,
+            // and whether it had a recent underrun.
+            FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
+            FastTrackUnderruns underruns = ftDump->mUnderruns;
+            uint32_t recentFull = (underruns.mBitFields.mFull -
+                    track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
+            uint32_t recentPartial = (underruns.mBitFields.mPartial -
+                    track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
+            uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
+                    track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
+            uint32_t recentUnderruns = recentPartial + recentEmpty;
+            track->mObservedUnderruns = underruns;
+            // don't count underruns that occur while stopping or pausing
+            // or stopped which can occur when flush() is called while active
+            if (!(track->isStopping() || track->isPausing() || track->isStopped())) {
+                track->mUnderrunCount += recentUnderruns;
+            }
+
+            // This is similar to the state machine for normal tracks,
+            // with a few modifications for fast tracks.
+            bool isActive = true;
+            switch (track->mState) {
+            case TrackBase::STOPPING_1:
+                // track stays active in STOPPING_1 state until first underrun
+                if (recentUnderruns > 0) {
+                    track->mState = TrackBase::STOPPING_2;
+                }
+                break;
+            case TrackBase::PAUSING:
+                // ramp down is not yet implemented
+                track->setPaused();
+                break;
+            case TrackBase::RESUMING:
+                // ramp up is not yet implemented
+                track->mState = TrackBase::ACTIVE;
+                break;
+            case TrackBase::ACTIVE:
+                if (recentFull > 0 || recentPartial > 0) {
+                    // track has provided at least some frames recently: reset retry count
+                    track->mRetryCount = kMaxTrackRetries;
+                }
+                if (recentUnderruns == 0) {
+                    // no recent underruns: stay active
+                    break;
+                }
+                // there has recently been an underrun of some kind
+                if (track->sharedBuffer() == 0) {
+                    // were any of the recent underruns "empty" (no frames available)?
+                    if (recentEmpty == 0) {
+                        // no, then ignore the partial underruns as they are allowed indefinitely
+                        break;
+                    }
+                    // there has recently been an "empty" underrun: decrement the retry counter
+                    if (--(track->mRetryCount) > 0) {
+                        break;
+                    }
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &track->mCblk->flags);
+                    // remove from active list, but state remains ACTIVE [confusing but true]
+                    isActive = false;
+                    break;
+                }
+                // fall through
+            case TrackBase::STOPPING_2:
+            case TrackBase::PAUSED:
+            case TrackBase::TERMINATED:
+            case TrackBase::STOPPED:
+            case TrackBase::FLUSHED:   // flush() while active
+                // Check for presentation complete if track is inactive
+                // We have consumed all the buffers of this track.
+                // This would be incomplete if we auto-paused on underrun
+                {
+                    size_t audioHALFrames =
+                            (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
+                    size_t framesWritten = mBytesWritten / mFrameSize;
+                    if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
+                        // track stays in active list until presentation is complete
+                        break;
+                    }
+                }
+                if (track->isStopping_2()) {
+                    track->mState = TrackBase::STOPPED;
+                }
+                if (track->isStopped()) {
+                    // Can't reset directly, as fast mixer is still polling this track
+                    //   track->reset();
+                    // So instead mark this track as needing to be reset after push with ack
+                    resetMask |= 1 << i;
+                }
+                isActive = false;
+                break;
+            case TrackBase::IDLE:
+            default:
+                LOG_FATAL("unexpected track state %d", track->mState);
+            }
+
+            if (isActive) {
+                // was it previously inactive?
+                if (!(state->mTrackMask & (1 << j))) {
+                    ExtendedAudioBufferProvider *eabp = track;
+                    VolumeProvider *vp = track;
+                    fastTrack->mBufferProvider = eabp;
+                    fastTrack->mVolumeProvider = vp;
+                    fastTrack->mSampleRate = track->mSampleRate;
+                    fastTrack->mChannelMask = track->mChannelMask;
+                    fastTrack->mGeneration++;
+                    state->mTrackMask |= 1 << j;
+                    didModify = true;
+                    // no acknowledgement required for newly active tracks
+                }
+                // cache the combined master volume and stream type volume for fast mixer; this
+                // lacks any synchronization or barrier so VolumeProvider may read a stale value
+                track->mCachedVolume = track->isMuted() ?
+                        0 : masterVolume * mStreamTypes[track->streamType()].volume;
+                ++fastTracks;
+            } else {
+                // was it previously active?
+                if (state->mTrackMask & (1 << j)) {
+                    fastTrack->mBufferProvider = NULL;
+                    fastTrack->mGeneration++;
+                    state->mTrackMask &= ~(1 << j);
+                    didModify = true;
+                    // If any fast tracks were removed, we must wait for acknowledgement
+                    // because we're about to decrement the last sp<> on those tracks.
+                    block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+                } else {
+                    LOG_FATAL("fast track %d should have been active", j);
+                }
+                tracksToRemove->add(track);
+                // Avoids a misleading display in dumpsys
+                track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
+            }
+            continue;
+        }
+
+        {   // local variable scope to avoid goto warning
+
+        audio_track_cblk_t* cblk = track->cblk();
+
+        // The first time a track is added we wait
+        // for all its buffers to be filled before processing it
+        int name = track->name();
+        // make sure that we have enough frames to mix one full buffer.
+        // enforce this condition only once to enable draining the buffer in case the client
+        // app does not call stop() and relies on underrun to stop:
+        // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
+        // during last round
+        uint32_t minFrames = 1;
+        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
+                (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
+            if (t->sampleRate() == mSampleRate) {
+                minFrames = mNormalFrameCount;
+            } else {
+                // +1 for rounding and +1 for additional sample needed for interpolation
+                minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+                // add frames already consumed but not yet released by the resampler
+                // because cblk->framesReady() will include these frames
+                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
+                // the minimum track buffer size is normally twice the number of frames necessary
+                // to fill one buffer and the resampler should not leave more than one buffer worth
+                // of unreleased frames after each pass, but just in case...
+                ALOG_ASSERT(minFrames <= cblk->frameCount);
+            }
+        }
+        if ((track->framesReady() >= minFrames) && track->isReady() &&
+                !track->isPaused() && !track->isTerminated())
+        {
+            ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
+                    this);
+
+            mixedTracks++;
+
+            // track->mainBuffer() != mMixBuffer means there is an effect chain
+            // connected to the track
+            chain.clear();
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                // Delegate volume control to effect in track effect chain if needed
+                if (chain != 0) {
+                    tracksWithEffect++;
+                } else {
+                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
+                            "session %d",
+                            name, track->sessionId());
+                }
+            }
+
+
+            int param = AudioMixer::VOLUME;
+            if (track->mFillingUpStatus == Track::FS_FILLED) {
+                // no ramp for the first volume setting
+                track->mFillingUpStatus = Track::FS_ACTIVE;
+                if (track->mState == TrackBase::RESUMING) {
+                    track->mState = TrackBase::ACTIVE;
+                    param = AudioMixer::RAMP_VOLUME;
+                }
+                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+            } else if (cblk->server != 0) {
+                // If the track is stopped before the first frame was mixed,
+                // do not apply ramp
+                param = AudioMixer::RAMP_VOLUME;
+            }
+
+            // compute volume for this track
+            uint32_t vl, vr, va;
+            if (track->isMuted() || track->isPausing() ||
+                mStreamTypes[track->streamType()].mute) {
+                vl = vr = va = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+
+                // read original volumes with volume control
+                float typeVolume = mStreamTypes[track->streamType()].volume;
+                float v = masterVolume * typeVolume;
+                uint32_t vlr = cblk->getVolumeLR();
+                vl = vlr & 0xFFFF;
+                vr = vlr >> 16;
+                // track volumes come from shared memory, so can't be trusted and must be clamped
+                if (vl > MAX_GAIN_INT) {
+                    ALOGV("Track left volume out of range: %04X", vl);
+                    vl = MAX_GAIN_INT;
+                }
+                if (vr > MAX_GAIN_INT) {
+                    ALOGV("Track right volume out of range: %04X", vr);
+                    vr = MAX_GAIN_INT;
+                }
+                // now apply the master volume and stream type volume
+                vl = (uint32_t)(v * vl) << 12;
+                vr = (uint32_t)(v * vr) << 12;
+                // assuming master volume and stream type volume each go up to 1.0,
+                // vl and vr are now in 8.24 format
+
+                uint16_t sendLevel = cblk->getSendLevel_U4_12();
+                // send level comes from shared memory and so may be corrupt
+                if (sendLevel > MAX_GAIN_INT) {
+                    ALOGV("Track send level out of range: %04X", sendLevel);
+                    sendLevel = MAX_GAIN_INT;
+                }
+                va = (uint32_t)(v * sendLevel);
+            }
+            // Delegate volume control to effect in track effect chain if needed
+            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
+                // Do not ramp volume if volume is controlled by effect
+                param = AudioMixer::VOLUME;
+                track->mHasVolumeController = true;
+            } else {
+                // force no volume ramp when volume controller was just disabled or removed
+                // from effect chain to avoid volume spike
+                if (track->mHasVolumeController) {
+                    param = AudioMixer::VOLUME;
+                }
+                track->mHasVolumeController = false;
+            }
+
+            // Convert volumes from 8.24 to 4.12 format
+            // This additional clamping is needed in case chain->setVolume_l() overshot
+            vl = (vl + (1 << 11)) >> 12;
+            if (vl > MAX_GAIN_INT) {
+                vl = MAX_GAIN_INT;
+            }
+            vr = (vr + (1 << 11)) >> 12;
+            if (vr > MAX_GAIN_INT) {
+                vr = MAX_GAIN_INT;
+            }
+
+            if (va > MAX_GAIN_INT) {
+                va = MAX_GAIN_INT;   // va is uint32_t, so no need to check for -
+            }
+
+            // XXX: these things DON'T need to be done each time
+            mAudioMixer->setBufferProvider(name, track);
+            mAudioMixer->enable(name);
+
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT, (void *)track->format());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::RESAMPLE,
+                AudioMixer::SAMPLE_RATE,
+                (void *)(cblk->sampleRate));
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetries;
+
+            // If one track is ready, set the mixer ready if:
+            //  - the mixer was not ready during previous round OR
+            //  - no other track is not ready
+            if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
+                    mixerStatus != MIXER_TRACKS_ENABLED) {
+                mixerStatus = MIXER_TRACKS_READY;
+            }
+        } else {
+            // clear effect chain input buffer if an active track underruns to avoid sending
+            // previous audio buffer again to effects
+            chain = getEffectChain_l(track->sessionId());
+            if (chain != 0) {
+                chain->clearInputBuffer();
+            }
+
+            ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user,
+                    cblk->server, this);
+            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+                    track->isStopped() || track->isPaused()) {
+                // We have consumed all the buffers of this track.
+                // Remove it from the list of active tracks.
+                // TODO: use actual buffer filling status instead of latency when available from
+                // audio HAL
+                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                size_t framesWritten = mBytesWritten / mFrameSize;
+                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    tracksToRemove->add(track);
+                }
+            } else {
+                track->mUnderrunCount++;
+                // No buffers for this track. Give it a few chances to
+                // fill a buffer, then remove it from active list.
+                if (--(track->mRetryCount) <= 0) {
+                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
+                    tracksToRemove->add(track);
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &cblk->flags);
+                // If one track is not ready, mark the mixer also not ready if:
+                //  - the mixer was ready during previous round OR
+                //  - no other track is ready
+                } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
+                                mixerStatus != MIXER_TRACKS_READY) {
+                    mixerStatus = MIXER_TRACKS_ENABLED;
+                }
+            }
+            mAudioMixer->disable(name);
+        }
+
+        }   // local variable scope to avoid goto warning
+track_is_ready: ;
+
+    }
+
+    // Push the new FastMixer state if necessary
+    bool pauseAudioWatchdog = false;
+    if (didModify) {
+        state->mFastTracksGen++;
+        // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
+        if (kUseFastMixer == FastMixer_Dynamic &&
+                state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
+            state->mCommand = FastMixerState::COLD_IDLE;
+            state->mColdFutexAddr = &mFastMixerFutex;
+            state->mColdGen++;
+            mFastMixerFutex = 0;
+            if (kUseFastMixer == FastMixer_Dynamic) {
+                mNormalSink = mOutputSink;
+            }
+            // If we go into cold idle, need to wait for acknowledgement
+            // so that fast mixer stops doing I/O.
+            block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+            pauseAudioWatchdog = true;
+        }
+        sq->end();
+    }
+    if (sq != NULL) {
+        sq->end(didModify);
+        sq->push(block);
+    }
+#ifdef AUDIO_WATCHDOG
+    if (pauseAudioWatchdog && mAudioWatchdog != 0) {
+        mAudioWatchdog->pause();
+    }
+#endif
+
+    // Now perform the deferred reset on fast tracks that have stopped
+    while (resetMask != 0) {
+        size_t i = __builtin_ctz(resetMask);
+        ALOG_ASSERT(i < count);
+        resetMask &= ~(1 << i);
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) {
+            continue;
+        }
+        Track* track = t.get();
+        ALOG_ASSERT(track->isFastTrack() && track->isStopped());
+        track->reset();
+    }
+
+    // remove all the tracks that need to be...
+    count = tracksToRemove->size();
+    if (CC_UNLIKELY(count)) {
+        for (size_t i=0 ; i<count ; i++) {
+            const sp<Track>& track = tracksToRemove->itemAt(i);
+            mActiveTracks.remove(track);
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                if (chain != 0) {
+                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
+                            track->sessionId());
+                    chain->decActiveTrackCnt();
+                }
+            }
+            if (track->isTerminated()) {
+                removeTrack_l(track);
+            }
+        }
+    }
+
+    // mix buffer must be cleared if all tracks are connected to an
+    // effect chain as in this case the mixer will not write to
+    // mix buffer and track effects will accumulate into it
+    if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+            (mixedTracks == 0 && fastTracks > 0)) {
+        // FIXME as a performance optimization, should remember previous zero status
+        memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+    }
+
+    // if any fast tracks, then status is ready
+    mMixerStatusIgnoringFastTracks = mixerStatus;
+    if (fastTracks > 0) {
+        mixerStatus = MIXER_TRACKS_READY;
+    }
+    return mixerStatus;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
+{
+    return mAudioMixer->getTrackName(channelMask, sessionId);
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::MixerThread::deleteTrackName_l(int name)
+{
+    ALOGV("remove track (%d) and delete from mixer", name);
+    mAudioMixer->deleteTrackName(name);
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
+{
+    // if !&IDLE, holds the FastMixer state to restore after new parameters processed
+    FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+
+        if (mFastMixer != NULL) {
+            FastMixerStateQueue *sq = mFastMixer->sq();
+            FastMixerState *state = sq->begin();
+            if (!(state->mCommand & FastMixerState::IDLE)) {
+                previousCommand = state->mCommand;
+                state->mCommand = FastMixerState::HOT_IDLE;
+                sq->end();
+                sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+            } else {
+                sq->end(false /*didModify*/);
+            }
+        }
+
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            if (value != AUDIO_CHANNEL_OUT_STEREO) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be guaranteed
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+#ifdef ADD_BATTERY_DATA
+            // when changing the audio output device, call addBatteryData to notify
+            // the change
+            if (mOutDevice != value) {
+                uint32_t params = 0;
+                // check whether speaker is on
+                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
+                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
+                }
+
+                audio_devices_t deviceWithoutSpeaker
+                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
+                // check if any other device (except speaker) is on
+                if (value & deviceWithoutSpeaker ) {
+                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
+                }
+
+                if (params != 0) {
+                    addBatteryData(params);
+                }
+            }
+#endif
+
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            mOutDevice = value;
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setDevice_l(mOutDevice);
+            }
+        }
+
+        if (status == NO_ERROR) {
+            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                    keyValuePair.string());
+            if (!mStandby && status == INVALID_OPERATION) {
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                       keyValuePair.string());
+            }
+            if (status == NO_ERROR && reconfig) {
+                delete mAudioMixer;
+                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+                mAudioMixer = NULL;
+                readOutputParameters();
+                mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+                for (size_t i = 0; i < mTracks.size() ; i++) {
+                    int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
+                    if (name < 0) {
+                        break;
+                    }
+                    mTracks[i]->mName = name;
+                    // limit track sample rate to 2 x new output sample rate
+                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
+                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
+                    }
+                }
+                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+
+    if (!(previousCommand & FastMixerState::IDLE)) {
+        ALOG_ASSERT(mFastMixer != NULL);
+        FastMixerStateQueue *sq = mFastMixer->sq();
+        FastMixerState *state = sq->begin();
+        ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
+        state->mCommand = previousCommand;
+        sq->end();
+        sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+    }
+
+    return reconfig;
+}
+
+
+void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    PlaybackThread::dumpInternals(fd, args);
+
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+    FastMixerDumpState copy = mFastMixerDumpState;
+    copy.dump(fd);
+
+#ifdef STATE_QUEUE_DUMP
+    // Similar for state queue
+    StateQueueObserverDump observerCopy = mStateQueueObserverDump;
+    observerCopy.dump(fd);
+    StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
+    mutatorCopy.dump(fd);
+#endif
+
+    // Write the tee output to a .wav file
+    dumpTee(fd, mTeeSource, mId);
+
+#ifdef AUDIO_WATCHDOG
+    if (mAudioWatchdog != 0) {
+        // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
+        AudioWatchdogDump wdCopy = mAudioWatchdogDump;
+        wdCopy.dump(fd);
+    }
+#endif
+}
+
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
+{
+    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
+}
+
+uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
+{
+    return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+void AudioFlinger::MixerThread::cacheParameters_l()
+{
+    PlaybackThread::cacheParameters_l();
+
+    // FIXME: Relaxed timing because of a certain device that can't meet latency
+    // Should be reduced to 2x after the vendor fixes the driver issue
+    // increase threshold again due to low power audio mode. The way this warning
+    // threshold is calculated and its usefulness should be reconsidered anyway.
+    maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+        AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
+    :   PlaybackThread(audioFlinger, output, id, device, DIRECT)
+        // mLeftVolFloat, mRightVolFloat
+{
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
+    Vector< sp<Track> > *tracksToRemove
+)
+{
+    sp<Track> trackToRemove;
+
+    mixer_state mixerStatus = MIXER_IDLE;
+
+    // find out which tracks need to be processed
+    if (mActiveTracks.size() != 0) {
+        sp<Track> t = mActiveTracks[0].promote();
+        // The track died recently
+        if (t == 0) {
+            return MIXER_IDLE;
+        }
+
+        Track* const track = t.get();
+        audio_track_cblk_t* cblk = track->cblk();
+
+        // The first time a track is added we wait
+        // for all its buffers to be filled before processing it
+        uint32_t minFrames;
+        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
+            minFrames = mNormalFrameCount;
+        } else {
+            minFrames = 1;
+        }
+        if ((track->framesReady() >= minFrames) && track->isReady() &&
+                !track->isPaused() && !track->isTerminated())
+        {
+            ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+            if (track->mFillingUpStatus == Track::FS_FILLED) {
+                track->mFillingUpStatus = Track::FS_ACTIVE;
+                mLeftVolFloat = mRightVolFloat = 0;
+                if (track->mState == TrackBase::RESUMING) {
+                    track->mState = TrackBase::ACTIVE;
+                }
+            }
+
+            // compute volume for this track
+            float left, right;
+            if (track->isMuted() || mMasterMute || track->isPausing() ||
+                mStreamTypes[track->streamType()].mute) {
+                left = right = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+                float typeVolume = mStreamTypes[track->streamType()].volume;
+                float v = mMasterVolume * typeVolume;
+                uint32_t vlr = cblk->getVolumeLR();
+                float v_clamped = v * (vlr & 0xFFFF);
+                if (v_clamped > MAX_GAIN) {
+                    v_clamped = MAX_GAIN;
+                }
+                left = v_clamped/MAX_GAIN;
+                v_clamped = v * (vlr >> 16);
+                if (v_clamped > MAX_GAIN) {
+                    v_clamped = MAX_GAIN;
+                }
+                right = v_clamped/MAX_GAIN;
+            }
+
+            if (left != mLeftVolFloat || right != mRightVolFloat) {
+                mLeftVolFloat = left;
+                mRightVolFloat = right;
+
+                // Convert volumes from float to 8.24
+                uint32_t vl = (uint32_t)(left * (1 << 24));
+                uint32_t vr = (uint32_t)(right * (1 << 24));
+
+                // Delegate volume control to effect in track effect chain if needed
+                // only one effect chain can be present on DirectOutputThread, so if
+                // there is one, the track is connected to it
+                if (!mEffectChains.isEmpty()) {
+                    // Do not ramp volume if volume is controlled by effect
+                    mEffectChains[0]->setVolume_l(&vl, &vr);
+                    left = (float)vl / (1 << 24);
+                    right = (float)vr / (1 << 24);
+                }
+                mOutput->stream->set_volume(mOutput->stream, left, right);
+            }
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetriesDirect;
+            mActiveTrack = t;
+            mixerStatus = MIXER_TRACKS_READY;
+        } else {
+            // clear effect chain input buffer if an active track underruns to avoid sending
+            // previous audio buffer again to effects
+            if (!mEffectChains.isEmpty()) {
+                mEffectChains[0]->clearInputBuffer();
+            }
+
+            ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+                    track->isStopped() || track->isPaused()) {
+                // We have consumed all the buffers of this track.
+                // Remove it from the list of active tracks.
+                // TODO: implement behavior for compressed audio
+                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                size_t framesWritten = mBytesWritten / mFrameSize;
+                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    trackToRemove = track;
+                }
+            } else {
+                // No buffers for this track. Give it a few chances to
+                // fill a buffer, then remove it from active list.
+                if (--(track->mRetryCount) <= 0) {
+                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                    trackToRemove = track;
+                } else {
+                    mixerStatus = MIXER_TRACKS_ENABLED;
+                }
+            }
+        }
+    }
+
+    // FIXME merge this with similar code for removing multiple tracks
+    // remove all the tracks that need to be...
+    if (CC_UNLIKELY(trackToRemove != 0)) {
+        tracksToRemove->add(trackToRemove);
+        mActiveTracks.remove(trackToRemove);
+        if (!mEffectChains.isEmpty()) {
+            ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
+                    trackToRemove->sessionId());
+            mEffectChains[0]->decActiveTrackCnt();
+        }
+        if (trackToRemove->isTerminated()) {
+            removeTrack_l(trackToRemove);
+        }
+    }
+
+    return mixerStatus;
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_mix()
+{
+    AudioBufferProvider::Buffer buffer;
+    size_t frameCount = mFrameCount;
+    int8_t *curBuf = (int8_t *)mMixBuffer;
+    // output audio to hardware
+    while (frameCount) {
+        buffer.frameCount = frameCount;
+        mActiveTrack->getNextBuffer(&buffer);
+        if (CC_UNLIKELY(buffer.raw == NULL)) {
+            memset(curBuf, 0, frameCount * mFrameSize);
+            break;
+        }
+        memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+        frameCount -= buffer.frameCount;
+        curBuf += buffer.frameCount * mFrameSize;
+        mActiveTrack->releaseBuffer(&buffer);
+    }
+    sleepTime = 0;
+    standbyTime = systemTime() + standbyDelay;
+    mActiveTrack.clear();
+
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
+{
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime;
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
+        memset(mMixBuffer, 0, mFrameCount * mFrameSize);
+        sleepTime = 0;
+    }
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
+        int sessionId)
+{
+    return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                    keyValuePair.string());
+            if (!mStandby && status == INVALID_OPERATION) {
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                                                       keyValuePair.string());
+            }
+            if (status == NO_ERROR && reconfig) {
+                readOutputParameters();
+                sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+    return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = PlaybackThread::activeSleepTimeUs();
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
+{
+    uint32_t time;
+    if (audio_is_linear_pcm(mFormat)) {
+        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+void AudioFlinger::DirectOutputThread::cacheParameters_l()
+{
+    PlaybackThread::cacheParameters_l();
+
+    // use shorter standby delay as on normal output to release
+    // hardware resources as soon as possible
+    standbyDelay = microseconds(activeSleepTime*2);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+        AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+                DUPLICATING),
+        mWaitTimeMs(UINT_MAX)
+{
+    addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        mOutputTracks[i]->destroy();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_mix()
+{
+    // mix buffers...
+    if (outputsReady(outputTracks)) {
+        mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
+    } else {
+        memset(mMixBuffer, 0, mixBufferSize);
+    }
+    sleepTime = 0;
+    writeFrames = mNormalFrameCount;
+    standbyTime = systemTime() + standbyDelay;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+{
+    if (sleepTime == 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            sleepTime = activeSleepTime;
+        } else {
+            sleepTime = idleSleepTime;
+        }
+    } else if (mBytesWritten != 0) {
+        if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+            writeFrames = mNormalFrameCount;
+            memset(mMixBuffer, 0, mixBufferSize);
+        } else {
+            // flush remaining overflow buffers in output tracks
+            writeFrames = 0;
+        }
+        sleepTime = 0;
+    }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_write()
+{
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        outputTracks[i]->write(mMixBuffer, writeFrames);
+    }
+    mBytesWritten += mixBufferSize;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_standby()
+{
+    // DuplicatingThread implements standby by stopping all tracks
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        outputTracks[i]->stop();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::saveOutputTracks()
+{
+    outputTracks = mOutputTracks;
+}
+
+void AudioFlinger::DuplicatingThread::clearOutputTracks()
+{
+    outputTracks.clear();
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    // FIXME explain this formula
+    size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+    OutputTrack *outputTrack = new OutputTrack(thread,
+                                            this,
+                                            mSampleRate,
+                                            mFormat,
+                                            mChannelMask,
+                                            frameCount);
+    if (outputTrack->cblk() != NULL) {
+        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
+        mOutputTracks.add(outputTrack);
+        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+        updateWaitTime_l();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        if (mOutputTracks[i]->thread() == thread) {
+            mOutputTracks[i]->destroy();
+            mOutputTracks.removeAt(i);
+            updateWaitTime_l();
+            return;
+        }
+    }
+    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
+}
+
+// caller must hold mLock
+void AudioFlinger::DuplicatingThread::updateWaitTime_l()
+{
+    mWaitTimeMs = UINT_MAX;
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+        if (strong != 0) {
+            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
+            if (waitTimeMs < mWaitTimeMs) {
+                mWaitTimeMs = waitTimeMs;
+            }
+        }
+    }
+}
+
+
+bool AudioFlinger::DuplicatingThread::outputsReady(
+        const SortedVector< sp<OutputTrack> > &outputTracks)
+{
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
+        if (thread == 0) {
+            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
+                    outputTracks[i].get());
+            return false;
+        }
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        // see note at standby() declaration
+        if (playbackThread->standby() && !playbackThread->isSuspended()) {
+            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
+                    thread.get());
+            return false;
+        }
+    }
+    return true;
+}
+
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
+{
+    return (mWaitTimeMs * 1000) / 2;
+}
+
+void AudioFlinger::DuplicatingThread::cacheParameters_l()
+{
+    // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
+    updateWaitTime_l();
+
+    MixerThread::cacheParameters_l();
+}
+
+// ----------------------------------------------------------------------------
+//      Record
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
+                                         AudioStreamIn *input,
+                                         uint32_t sampleRate,
+                                         audio_channel_mask_t channelMask,
+                                         audio_io_handle_t id,
+                                         audio_devices_t device,
+                                         const sp<NBAIO_Sink>& teeSink) :
+    ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),
+    mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+    // mRsmpInIndex and mInputBytes set by readInputParameters()
+    mReqChannelCount(popcount(channelMask)),
+    mReqSampleRate(sampleRate),
+    // mBytesRead is only meaningful while active, and so is cleared in start()
+    // (but might be better to also clear here for dump?)
+    mTeeSink(teeSink)
+{
+    snprintf(mName, kNameLength, "AudioIn_%X", id);
+
+    readInputParameters();
+
+}
+
+
+AudioFlinger::RecordThread::~RecordThread()
+{
+    delete[] mRsmpInBuffer;
+    delete mResampler;
+    delete[] mRsmpOutBuffer;
+}
+
+void AudioFlinger::RecordThread::onFirstRef()
+{
+    run(mName, PRIORITY_URGENT_AUDIO);
+}
+
+status_t AudioFlinger::RecordThread::readyToRun()
+{
+    status_t status = initCheck();
+    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
+    return status;
+}
+
+bool AudioFlinger::RecordThread::threadLoop()
+{
+    AudioBufferProvider::Buffer buffer;
+    sp<RecordTrack> activeTrack;
+    Vector< sp<EffectChain> > effectChains;
+
+    nsecs_t lastWarning = 0;
+
+    inputStandBy();
+    acquireWakeLock();
+
+    // used to verify we've read at least once before evaluating how many bytes were read
+    bool readOnce = false;
+
+    // start recording
+    while (!exitPending()) {
+
+        processConfigEvents();
+
+        { // scope for mLock
+            Mutex::Autolock _l(mLock);
+            checkForNewParameters_l();
+            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+                standby();
+
+                if (exitPending()) {
+                    break;
+                }
+
+                releaseWakeLock_l();
+                ALOGV("RecordThread: loop stopping");
+                // go to sleep
+                mWaitWorkCV.wait(mLock);
+                ALOGV("RecordThread: loop starting");
+                acquireWakeLock_l();
+                continue;
+            }
+            if (mActiveTrack != 0) {
+                if (mActiveTrack->mState == TrackBase::PAUSING) {
+                    standby();
+                    mActiveTrack.clear();
+                    mStartStopCond.broadcast();
+                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+                    if (mReqChannelCount != mActiveTrack->channelCount()) {
+                        mActiveTrack.clear();
+                        mStartStopCond.broadcast();
+                    } else if (readOnce) {
+                        // record start succeeds only if first read from audio input
+                        // succeeds
+                        if (mBytesRead >= 0) {
+                            mActiveTrack->mState = TrackBase::ACTIVE;
+                        } else {
+                            mActiveTrack.clear();
+                        }
+                        mStartStopCond.broadcast();
+                    }
+                    mStandby = false;
+                } else if (mActiveTrack->mState == TrackBase::TERMINATED) {
+                    removeTrack_l(mActiveTrack);
+                    mActiveTrack.clear();
+                }
+            }
+            lockEffectChains_l(effectChains);
+        }
+
+        if (mActiveTrack != 0) {
+            if (mActiveTrack->mState != TrackBase::ACTIVE &&
+                mActiveTrack->mState != TrackBase::RESUMING) {
+                unlockEffectChains(effectChains);
+                usleep(kRecordThreadSleepUs);
+                continue;
+            }
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+
+            buffer.frameCount = mFrameCount;
+            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+                readOnce = true;
+                size_t framesOut = buffer.frameCount;
+                if (mResampler == NULL) {
+                    // no resampling
+                    while (framesOut) {
+                        size_t framesIn = mFrameCount - mRsmpInIndex;
+                        if (framesIn) {
+                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+                                    mActiveTrack->mFrameSize;
+                            if (framesIn > framesOut)
+                                framesIn = framesOut;
+                            mRsmpInIndex += framesIn;
+                            framesOut -= framesIn;
+                            if (mChannelCount == mReqChannelCount ||
+                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+                                memcpy(dst, src, framesIn * mFrameSize);
+                            } else {
+                                if (mChannelCount == 1) {
+                                    upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
+                                            (int16_t *)src, framesIn);
+                                } else {
+                                    downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+                                            (int16_t *)src, framesIn);
+                                }
+                            }
+                        }
+                        if (framesOut && mFrameCount == mRsmpInIndex) {
+                            void *readInto;
+                            if (framesOut == mFrameCount &&
+                                (mChannelCount == mReqChannelCount ||
+                                        mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+                                readInto = buffer.raw;
+                                framesOut = 0;
+                            } else {
+                                readInto = mRsmpInBuffer;
+                                mRsmpInIndex = 0;
+                            }
+                            mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes);
+                            if (mBytesRead <= 0) {
+                                if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
+                                {
+                                    ALOGE("Error reading audio input");
+                                    // Force input into standby so that it tries to
+                                    // recover at next read attempt
+                                    inputStandBy();
+                                    usleep(kRecordThreadSleepUs);
+                                }
+                                mRsmpInIndex = mFrameCount;
+                                framesOut = 0;
+                                buffer.frameCount = 0;
+                            } else if (mTeeSink != 0) {
+                                (void) mTeeSink->write(readInto,
+                                        mBytesRead >> Format_frameBitShift(mTeeSink->format()));
+                            }
+                        }
+                    }
+                } else {
+                    // resampling
+
+                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+                    // alter output frame count as if we were expecting stereo samples
+                    if (mChannelCount == 1 && mReqChannelCount == 1) {
+                        framesOut >>= 1;
+                    }
+                    mResampler->resample(mRsmpOutBuffer, framesOut,
+                            this /* AudioBufferProvider* */);
+                    // ditherAndClamp() works as long as all buffers returned by
+                    // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
+                    if (mChannelCount == 2 && mReqChannelCount == 1) {
+                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+                        // the resampler always outputs stereo samples:
+                        // do post stereo to mono conversion
+                        downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
+                                framesOut);
+                    } else {
+                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+                    }
+
+                }
+                if (mFramestoDrop == 0) {
+                    mActiveTrack->releaseBuffer(&buffer);
+                } else {
+                    if (mFramestoDrop > 0) {
+                        mFramestoDrop -= buffer.frameCount;
+                        if (mFramestoDrop <= 0) {
+                            clearSyncStartEvent();
+                        }
+                    } else {
+                        mFramestoDrop += buffer.frameCount;
+                        if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
+                                mSyncStartEvent->isCancelled()) {
+                            ALOGW("Synced record %s, session %d, trigger session %d",
+                                  (mFramestoDrop >= 0) ? "timed out" : "cancelled",
+                                  mActiveTrack->sessionId(),
+                                  (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
+                            clearSyncStartEvent();
+                        }
+                    }
+                }
+                mActiveTrack->clearOverflow();
+            }
+            // client isn't retrieving buffers fast enough
+            else {
+                if (!mActiveTrack->setOverflow()) {
+                    nsecs_t now = systemTime();
+                    if ((now - lastWarning) > kWarningThrottleNs) {
+                        ALOGW("RecordThread: buffer overflow");
+                        lastWarning = now;
+                    }
+                }
+                // Release the processor for a while before asking for a new buffer.
+                // This will give the application more chance to read from the buffer and
+                // clear the overflow.
+                usleep(kRecordThreadSleepUs);
+            }
+        }
+        // enable changes in effect chain
+        unlockEffectChains(effectChains);
+        effectChains.clear();
+    }
+
+    standby();
+
+    {
+        Mutex::Autolock _l(mLock);
+        mActiveTrack.clear();
+        mStartStopCond.broadcast();
+    }
+
+    releaseWakeLock();
+
+    ALOGV("RecordThread %p exiting", this);
+    return false;
+}
+
+void AudioFlinger::RecordThread::standby()
+{
+    if (!mStandby) {
+        inputStandBy();
+        mStandby = true;
+    }
+}
+
+void AudioFlinger::RecordThread::inputStandBy()
+{
+    mInput->stream->common.standby(&mInput->stream->common);
+}
+
+sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
+        const sp<AudioFlinger::Client>& client,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        int sessionId,
+        IAudioFlinger::track_flags_t flags,
+        pid_t tid,
+        status_t *status)
+{
+    sp<RecordTrack> track;
+    status_t lStatus;
+
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGE("Audio driver not initialized.");
+        goto Exit;
+    }
+
+    // FIXME use flags and tid similar to createTrack_l()
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        track = new RecordTrack(this, client, sampleRate,
+                      format, channelMask, frameCount, sessionId);
+
+        if (track->getCblk() == 0) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+
+        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+        bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+                        mAudioFlinger->btNrecIsOff();
+        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
+        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
+    }
+    lStatus = NO_ERROR;
+
+Exit:
+    if (status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
+                                           AudioSystem::sync_event_t event,
+                                           int triggerSession)
+{
+    ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
+    sp<ThreadBase> strongMe = this;
+    status_t status = NO_ERROR;
+
+    if (event == AudioSystem::SYNC_EVENT_NONE) {
+        clearSyncStartEvent();
+    } else if (event != AudioSystem::SYNC_EVENT_SAME) {
+        mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
+                                       triggerSession,
+                                       recordTrack->sessionId(),
+                                       syncStartEventCallback,
+                                       this);
+        // Sync event can be cancelled by the trigger session if the track is not in a
+        // compatible state in which case we start record immediately
+        if (mSyncStartEvent->isCancelled()) {
+            clearSyncStartEvent();
+        } else {
+            // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
+            mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
+        }
+    }
+
+    {
+        AutoMutex lock(mLock);
+        if (mActiveTrack != 0) {
+            if (recordTrack != mActiveTrack.get()) {
+                status = -EBUSY;
+            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
+                mActiveTrack->mState = TrackBase::ACTIVE;
+            }
+            return status;
+        }
+
+        recordTrack->mState = TrackBase::IDLE;
+        mActiveTrack = recordTrack;
+        mLock.unlock();
+        status_t status = AudioSystem::startInput(mId);
+        mLock.lock();
+        if (status != NO_ERROR) {
+            mActiveTrack.clear();
+            clearSyncStartEvent();
+            return status;
+        }
+        mRsmpInIndex = mFrameCount;
+        mBytesRead = 0;
+        if (mResampler != NULL) {
+            mResampler->reset();
+        }
+        mActiveTrack->mState = TrackBase::RESUMING;
+        // signal thread to start
+        ALOGV("Signal record thread");
+        mWaitWorkCV.broadcast();
+        // do not wait for mStartStopCond if exiting
+        if (exitPending()) {
+            mActiveTrack.clear();
+            status = INVALID_OPERATION;
+            goto startError;
+        }
+        mStartStopCond.wait(mLock);
+        if (mActiveTrack == 0) {
+            ALOGV("Record failed to start");
+            status = BAD_VALUE;
+            goto startError;
+        }
+        ALOGV("Record started OK");
+        return status;
+    }
+startError:
+    AudioSystem::stopInput(mId);
+    clearSyncStartEvent();
+    return status;
+}
+
+void AudioFlinger::RecordThread::clearSyncStartEvent()
+{
+    if (mSyncStartEvent != 0) {
+        mSyncStartEvent->cancel();
+    }
+    mSyncStartEvent.clear();
+    mFramestoDrop = 0;
+}
+
+void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
+{
+    sp<SyncEvent> strongEvent = event.promote();
+
+    if (strongEvent != 0) {
+        RecordThread *me = (RecordThread *)strongEvent->cookie();
+        me->handleSyncStartEvent(strongEvent);
+    }
+}
+
+void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
+{
+    if (event == mSyncStartEvent) {
+        // TODO: use actual buffer filling status instead of 2 buffers when info is available
+        // from audio HAL
+        mFramestoDrop = mFrameCount * 2;
+    }
+}
+
+bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) {
+    ALOGV("RecordThread::stop");
+    if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
+        return false;
+    }
+    recordTrack->mState = TrackBase::PAUSING;
+    // do not wait for mStartStopCond if exiting
+    if (exitPending()) {
+        return true;
+    }
+    mStartStopCond.wait(mLock);
+    // if we have been restarted, recordTrack == mActiveTrack.get() here
+    if (exitPending() || recordTrack != mActiveTrack.get()) {
+        ALOGV("Record stopped OK");
+        return true;
+    }
+    return false;
+}
+
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+    return false;
+}
+
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+#if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
+    if (!isValidSyncEvent(event)) {
+        return BAD_VALUE;
+    }
+
+    int eventSession = event->triggerSession();
+    status_t ret = NAME_NOT_FOUND;
+
+    Mutex::Autolock _l(mLock);
+
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        sp<RecordTrack> track = mTracks[i];
+        if (eventSession == track->sessionId()) {
+            (void) track->setSyncEvent(event);
+            ret = NO_ERROR;
+        }
+    }
+    return ret;
+#else
+    return BAD_VALUE;
+#endif
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
+{
+    track->mState = TrackBase::TERMINATED;
+    // active tracks are removed by threadLoop()
+    if (mActiveTrack != track) {
+        removeTrack_l(track);
+    }
+}
+
+void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
+{
+    mTracks.remove(track);
+    // need anything related to effects here?
+}
+
+void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
+    result.append(buffer);
+
+    if (mActiveTrack != 0) {
+        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
+        result.append(buffer);
+    } else {
+        result.append("No active record client\n");
+    }
+
+    write(fd, result.string(), result.size());
+
+    dumpBase(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
+    result.append(buffer);
+    RecordTrack::appendDumpHeader(result);
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<RecordTrack> track = mTracks[i];
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    if (mActiveTrack != 0) {
+        snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+        result.append(buffer);
+        RecordTrack::appendDumpHeader(result);
+        mActiveTrack->dump(buffer, SIZE);
+        result.append(buffer);
+
+    }
+    write(fd, result.string(), result.size());
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+    size_t framesReq = buffer->frameCount;
+    size_t framesReady = mFrameCount - mRsmpInIndex;
+    int channelCount;
+
+    if (framesReady == 0) {
+        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
+        if (mBytesRead <= 0) {
+            if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
+                ALOGE("RecordThread::getNextBuffer() Error reading audio input");
+                // Force input into standby so that it tries to
+                // recover at next read attempt
+                inputStandBy();
+                usleep(kRecordThreadSleepUs);
+            }
+            buffer->raw = NULL;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+        mRsmpInIndex = 0;
+        framesReady = mFrameCount;
+    }
+
+    if (framesReq > framesReady) {
+        framesReq = framesReady;
+    }
+
+    if (mChannelCount == 1 && mReqChannelCount == 2) {
+        channelCount = 1;
+    } else {
+        channelCount = 2;
+    }
+    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+    buffer->frameCount = framesReq;
+    return NO_ERROR;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    mRsmpInIndex += buffer->frameCount;
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+        audio_format_t reqFormat = mFormat;
+        uint32_t reqSamplingRate = mReqSampleRate;
+        uint32_t reqChannelCount = mReqChannelCount;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reqSamplingRate = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            reqFormat = (audio_format_t) value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            reqChannelCount = popcount(value);
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be guaranteed
+            // if frame count is changed after track creation
+            if (mActiveTrack != 0) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setDevice_l(value);
+            }
+
+            // store input device and output device but do not forward output device to audio HAL.
+            // Note that status is ignored by the caller for output device
+            // (see AudioFlinger::setParameters()
+            if (audio_is_output_devices(value)) {
+                mOutDevice = value;
+                status = BAD_VALUE;
+            } else {
+                mInDevice = value;
+                // disable AEC and NS if the device is a BT SCO headset supporting those
+                // pre processings
+                if (mTracks.size() > 0) {
+                    bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+                                        mAudioFlinger->btNrecIsOff();
+                    for (size_t i = 0; i < mTracks.size(); i++) {
+                        sp<RecordTrack> track = mTracks[i];
+                        setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
+                        setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
+                    }
+                }
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
+                mAudioSource != (audio_source_t)value) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setAudioSource_l((audio_source_t)value);
+            }
+            mAudioSource = (audio_source_t)value;
+        }
+        if (status == NO_ERROR) {
+            status = mInput->stream->common.set_parameters(&mInput->stream->common,
+                    keyValuePair.string());
+            if (status == INVALID_OPERATION) {
+                inputStandBy();
+                status = mInput->stream->common.set_parameters(&mInput->stream->common,
+                        keyValuePair.string());
+            }
+            if (reconfig) {
+                if (status == BAD_VALUE &&
+                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
+                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common)
+                            <= (2 * reqSamplingRate)) &&
+                    popcount(mInput->stream->common.get_channels(&mInput->stream->common))
+                            <= FCC_2 &&
+                    (reqChannelCount <= FCC_2)) {
+                    status = NO_ERROR;
+                }
+                if (status == NO_ERROR) {
+                    readInputParameters();
+                    sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+                }
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+        // already timed out waiting for the status and will never signal the condition.
+        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+    }
+    return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+    char *s;
+    String8 out_s8 = String8();
+
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return out_s8;
+    }
+
+    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
+    out_s8 = String8(s);
+    free(s);
+    return out_s8;
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = NULL;
+
+    switch (event) {
+    case AudioSystem::INPUT_OPENED:
+    case AudioSystem::INPUT_CONFIG_CHANGED:
+        desc.channels = mChannelMask;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = 0;
+        param2 = &desc;
+        break;
+
+    case AudioSystem::INPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+    delete mRsmpInBuffer;
+    // mRsmpInBuffer is always assigned a new[] below
+    delete mRsmpOutBuffer;
+    mRsmpOutBuffer = NULL;
+    delete mResampler;
+    mResampler = NULL;
+
+    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
+    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
+    mChannelCount = (uint16_t)popcount(mChannelMask);
+    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
+    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
+    mFrameCount = mInputBytes / mFrameSize;
+    mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
+    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
+    {
+        int channelCount;
+        // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+        // stereo to mono post process as the resampler always outputs stereo.
+        if (mChannelCount == 1 && mReqChannelCount == 2) {
+            channelCount = 1;
+        } else {
+            channelCount = 2;
+        }
+        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+        mResampler->setSampleRate(mSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+        // optmization: if mono to mono, alter input frame count as if we were inputing
+        // stereo samples
+        if (mChannelCount == 1 && mReqChannelCount == 1) {
+            mFrameCount >>= 1;
+        }
+
+    }
+    mRsmpInIndex = mFrameCount;
+}
+
+unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+{
+    Mutex::Autolock _l(mLock);
+    if (initCheck() != NO_ERROR) {
+        return 0;
+    }
+
+    return mInput->stream->get_input_frames_lost(mInput->stream);
+}
+
+uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
+{
+    Mutex::Autolock _l(mLock);
+    uint32_t result = 0;
+    if (getEffectChain_l(sessionId) != 0) {
+        result = EFFECT_SESSION;
+    }
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        if (sessionId == mTracks[i]->sessionId()) {
+            result |= TRACK_SESSION;
+            break;
+        }
+    }
+
+    return result;
+}
+
+KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
+{
+    KeyedVector<int, bool> ids;
+    Mutex::Autolock _l(mLock);
+    for (size_t j = 0; j < mTracks.size(); ++j) {
+        sp<RecordThread::RecordTrack> track = mTracks[j];
+        int sessionId = track->sessionId();
+        if (ids.indexOfKey(sessionId) < 0) {
+            ids.add(sessionId, true);
+        }
+    }
+    return ids;
+}
+
+AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
+{
+    Mutex::Autolock _l(mLock);
+    AudioStreamIn *input = mInput;
+    mInput = NULL;
+    return input;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::RecordThread::stream() const
+{
+    if (mInput == NULL) {
+        return NULL;
+    }
+    return &mInput->stream->common;
+}
+
+status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+    // only one chain per input thread
+    if (mEffectChains.size() != 0) {
+        return INVALID_OPERATION;
+    }
+    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
+
+    chain->setInBuffer(NULL);
+    chain->setOutBuffer(NULL);
+
+    checkSuspendOnAddEffectChain_l(chain);
+
+    mEffectChains.add(chain);
+
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
+    ALOGW_IF(mEffectChains.size() != 1,
+            "removeEffectChain_l() %p invalid chain size %d on thread %p",
+            chain.get(), mEffectChains.size(), this);
+    if (mEffectChains.size() == 1) {
+        mEffectChains.removeAt(0);
+    }
+    return 0;
+}
+
+}; // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
new file mode 100644
index 0000000..06a1c8c
--- /dev/null
+++ b/services/audioflinger/Threads.h
@@ -0,0 +1,801 @@
+/*
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+class ThreadBase : public Thread {
+public:
+
+#include "TrackBase.h"
+
+    enum type_t {
+        MIXER,              // Thread class is MixerThread
+        DIRECT,             // Thread class is DirectOutputThread
+        DUPLICATING,        // Thread class is DuplicatingThread
+        RECORD              // Thread class is RecordThread
+    };
+
+    ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+                audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
+    virtual             ~ThreadBase();
+
+    void dumpBase(int fd, const Vector<String16>& args);
+    void dumpEffectChains(int fd, const Vector<String16>& args);
+
+    void clearPowerManager();
+
+    // base for record and playback
+    enum {
+        CFG_EVENT_IO,
+        CFG_EVENT_PRIO
+    };
+
+    class ConfigEvent {
+    public:
+        ConfigEvent(int type) : mType(type) {}
+        virtual ~ConfigEvent() {}
+
+                 int type() const { return mType; }
+
+        virtual  void dump(char *buffer, size_t size) = 0;
+
+    private:
+        const int mType;
+    };
+
+    class IoConfigEvent : public ConfigEvent {
+    public:
+        IoConfigEvent(int event, int param) :
+            ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+        virtual ~IoConfigEvent() {}
+
+                int event() const { return mEvent; }
+                int param() const { return mParam; }
+
+        virtual  void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam);
+        }
+
+    private:
+        const int mEvent;
+        const int mParam;
+    };
+
+    class PrioConfigEvent : public ConfigEvent {
+    public:
+        PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) :
+            ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {}
+        virtual ~PrioConfigEvent() {}
+
+                pid_t pid() const { return mPid; }
+                pid_t tid() const { return mTid; }
+                int32_t prio() const { return mPrio; }
+
+        virtual  void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio);
+        }
+
+    private:
+        const pid_t mPid;
+        const pid_t mTid;
+        const int32_t mPrio;
+    };
+
+
+    class PMDeathRecipient : public IBinder::DeathRecipient {
+    public:
+                    PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
+        virtual     ~PMDeathRecipient() {}
+
+        // IBinder::DeathRecipient
+        virtual     void        binderDied(const wp<IBinder>& who);
+
+    private:
+                    PMDeathRecipient(const PMDeathRecipient&);
+                    PMDeathRecipient& operator = (const PMDeathRecipient&);
+
+        wp<ThreadBase> mThread;
+    };
+
+    virtual     status_t    initCheck() const = 0;
+
+                // static externally-visible
+                type_t      type() const { return mType; }
+                audio_io_handle_t id() const { return mId;}
+
+                // dynamic externally-visible
+                uint32_t    sampleRate() const { return mSampleRate; }
+                uint32_t    channelCount() const { return mChannelCount; }
+                audio_channel_mask_t channelMask() const { return mChannelMask; }
+                audio_format_t format() const { return mFormat; }
+                // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
+                // and returns the normal mix buffer's frame count.
+                size_t      frameCount() const { return mNormalFrameCount; }
+                // Return's the HAL's frame count i.e. fast mixer buffer size.
+                size_t      frameCountHAL() const { return mFrameCount; }
+
+    // Should be "virtual status_t requestExitAndWait()" and override same
+    // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
+                void        exit();
+    virtual     bool        checkForNewParameters_l() = 0;
+    virtual     status_t    setParameters(const String8& keyValuePairs);
+    virtual     String8     getParameters(const String8& keys) = 0;
+    virtual     void        audioConfigChanged_l(int event, int param = 0) = 0;
+                void        sendIoConfigEvent(int event, int param = 0);
+                void        sendIoConfigEvent_l(int event, int param = 0);
+                void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
+                void        processConfigEvents();
+
+                // see note at declaration of mStandby, mOutDevice and mInDevice
+                bool        standby() const { return mStandby; }
+                audio_devices_t outDevice() const { return mOutDevice; }
+                audio_devices_t inDevice() const { return mInDevice; }
+
+    virtual     audio_stream_t* stream() const = 0;
+
+                sp<EffectHandle> createEffect_l(
+                                    const sp<AudioFlinger::Client>& client,
+                                    const sp<IEffectClient>& effectClient,
+                                    int32_t priority,
+                                    int sessionId,
+                                    effect_descriptor_t *desc,
+                                    int *enabled,
+                                    status_t *status);
+                void disconnectEffect(const sp< EffectModule>& effect,
+                                      EffectHandle *handle,
+                                      bool unpinIfLast);
+
+                // return values for hasAudioSession (bit field)
+                enum effect_state {
+                    EFFECT_SESSION = 0x1,   // the audio session corresponds to at least one
+                                            // effect
+                    TRACK_SESSION = 0x2     // the audio session corresponds to at least one
+                                            // track
+                };
+
+                // get effect chain corresponding to session Id.
+                sp<EffectChain> getEffectChain(int sessionId);
+                // same as getEffectChain() but must be called with ThreadBase mutex locked
+                sp<EffectChain> getEffectChain_l(int sessionId) const;
+                // add an effect chain to the chain list (mEffectChains)
+    virtual     status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
+                // remove an effect chain from the chain list (mEffectChains)
+    virtual     size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
+                // lock all effect chains Mutexes. Must be called before releasing the
+                // ThreadBase mutex before processing the mixer and effects. This guarantees the
+                // integrity of the chains during the process.
+                // Also sets the parameter 'effectChains' to current value of mEffectChains.
+                void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
+                // unlock effect chains after process
+                void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
+                // set audio mode to all effect chains
+                void setMode(audio_mode_t mode);
+                // get effect module with corresponding ID on specified audio session
+                sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId);
+                sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
+                // add and effect module. Also creates the effect chain is none exists for
+                // the effects audio session
+                status_t addEffect_l(const sp< EffectModule>& effect);
+                // remove and effect module. Also removes the effect chain is this was the last
+                // effect
+                void removeEffect_l(const sp< EffectModule>& effect);
+                // detach all tracks connected to an auxiliary effect
+    virtual     void detachAuxEffect_l(int effectId) {}
+                // returns either EFFECT_SESSION if effects on this audio session exist in one
+                // chain, or TRACK_SESSION if tracks on this audio session exist, or both
+                virtual uint32_t hasAudioSession(int sessionId) const = 0;
+                // the value returned by default implementation is not important as the
+                // strategy is only meaningful for PlaybackThread which implements this method
+                virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+
+                // suspend or restore effect according to the type of effect passed. a NULL
+                // type pointer means suspend all effects in the session
+                void setEffectSuspended(const effect_uuid_t *type,
+                                        bool suspend,
+                                        int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+                // check if some effects must be suspended/restored when an effect is enabled
+                // or disabled
+                void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+                                                 bool enabled,
+                                                 int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+                void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
+                                                   bool enabled,
+                                                   int sessionId = AUDIO_SESSION_OUTPUT_MIX);
+
+                virtual status_t    setSyncEvent(const sp<SyncEvent>& event) = 0;
+                virtual bool        isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
+
+
+    mutable     Mutex                   mLock;
+
+protected:
+
+                // entry describing an effect being suspended in mSuspendedSessions keyed vector
+                class SuspendedSessionDesc : public RefBase {
+                public:
+                    SuspendedSessionDesc() : mRefCount(0) {}
+
+                    int mRefCount;          // number of active suspend requests
+                    effect_uuid_t mType;    // effect type UUID
+                };
+
+                void        acquireWakeLock();
+                void        acquireWakeLock_l();
+                void        releaseWakeLock();
+                void        releaseWakeLock_l();
+                void setEffectSuspended_l(const effect_uuid_t *type,
+                                          bool suspend,
+                                          int sessionId);
+                // updated mSuspendedSessions when an effect suspended or restored
+                void        updateSuspendedSessions_l(const effect_uuid_t *type,
+                                                      bool suspend,
+                                                      int sessionId);
+                // check if some effects must be suspended when an effect chain is added
+                void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
+
+    virtual     void        preExit() { }
+
+    friend class AudioFlinger;      // for mEffectChains
+
+                const type_t            mType;
+
+                // Used by parameters, config events, addTrack_l, exit
+                Condition               mWaitWorkCV;
+
+                const sp<AudioFlinger>  mAudioFlinger;
+                uint32_t                mSampleRate;
+                size_t                  mFrameCount;       // output HAL, direct output, record
+                size_t                  mNormalFrameCount; // normal mixer and effects
+                audio_channel_mask_t    mChannelMask;
+                uint16_t                mChannelCount;
+                size_t                  mFrameSize;
+                audio_format_t          mFormat;
+
+                // Parameter sequence by client: binder thread calling setParameters():
+                //  1. Lock mLock
+                //  2. Append to mNewParameters
+                //  3. mWaitWorkCV.signal
+                //  4. mParamCond.waitRelative with timeout
+                //  5. read mParamStatus
+                //  6. mWaitWorkCV.signal
+                //  7. Unlock
+                //
+                // Parameter sequence by server: threadLoop calling checkForNewParameters_l():
+                // 1. Lock mLock
+                // 2. If there is an entry in mNewParameters proceed ...
+                // 2. Read first entry in mNewParameters
+                // 3. Process
+                // 4. Remove first entry from mNewParameters
+                // 5. Set mParamStatus
+                // 6. mParamCond.signal
+                // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus)
+                // 8. Unlock
+                Condition               mParamCond;
+                Vector<String8>         mNewParameters;
+                status_t                mParamStatus;
+
+                Vector<ConfigEvent *>     mConfigEvents;
+
+                // These fields are written and read by thread itself without lock or barrier,
+                // and read by other threads without lock or barrier via standby() , outDevice()
+                // and inDevice().
+                // Because of the absence of a lock or barrier, any other thread that reads
+                // these fields must use the information in isolation, or be prepared to deal
+                // with possibility that it might be inconsistent with other information.
+                bool                    mStandby;   // Whether thread is currently in standby.
+                audio_devices_t         mOutDevice;   // output device
+                audio_devices_t         mInDevice;    // input device
+                audio_source_t          mAudioSource; // (see audio.h, audio_source_t)
+
+                const audio_io_handle_t mId;
+                Vector< sp<EffectChain> > mEffectChains;
+
+                static const int        kNameLength = 16;   // prctl(PR_SET_NAME) limit
+                char                    mName[kNameLength];
+                sp<IPowerManager>       mPowerManager;
+                sp<IBinder>             mWakeLockToken;
+                const sp<PMDeathRecipient> mDeathRecipient;
+                // list of suspended effects per session and per type. The first vector is
+                // keyed by session ID, the second by type UUID timeLow field
+                KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > >
+                                        mSuspendedSessions;
+};
+
+// --- PlaybackThread ---
+class PlaybackThread : public ThreadBase {
+public:
+
+#include "PlaybackTracks.h"
+
+    enum mixer_state {
+        MIXER_IDLE,             // no active tracks
+        MIXER_TRACKS_ENABLED,   // at least one active track, but no track has any data ready
+        MIXER_TRACKS_READY      // at least one active track, and at least one track has data
+        // standby mode does not have an enum value
+        // suspend by audio policy manager is orthogonal to mixer state
+    };
+
+    PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+                   audio_io_handle_t id, audio_devices_t device, type_t type);
+    virtual             ~PlaybackThread();
+
+                void        dump(int fd, const Vector<String16>& args);
+
+    // Thread virtuals
+    virtual     status_t    readyToRun();
+    virtual     bool        threadLoop();
+
+    // RefBase
+    virtual     void        onFirstRef();
+
+protected:
+    // Code snippets that were lifted up out of threadLoop()
+    virtual     void        threadLoop_mix() = 0;
+    virtual     void        threadLoop_sleepTime() = 0;
+    virtual     void        threadLoop_write();
+    virtual     void        threadLoop_standby();
+    virtual     void        threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
+
+                // prepareTracks_l reads and writes mActiveTracks, and returns
+                // the pending set of tracks to remove via Vector 'tracksToRemove'.  The caller
+                // is responsible for clearing or destroying this Vector later on, when it
+                // is safe to do so. That will drop the final ref count and destroy the tracks.
+    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
+
+    // ThreadBase virtuals
+    virtual     void        preExit();
+
+public:
+
+    virtual     status_t    initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
+
+                // return estimated latency in milliseconds, as reported by HAL
+                uint32_t    latency() const;
+                // same, but lock must already be held
+                uint32_t    latency_l() const;
+
+                void        setMasterVolume(float value);
+                void        setMasterMute(bool muted);
+
+                void        setStreamVolume(audio_stream_type_t stream, float value);
+                void        setStreamMute(audio_stream_type_t stream, bool muted);
+
+                float       streamVolume(audio_stream_type_t stream) const;
+
+                sp<Track>   createTrack_l(
+                                const sp<AudioFlinger::Client>& client,
+                                audio_stream_type_t streamType,
+                                uint32_t sampleRate,
+                                audio_format_t format,
+                                audio_channel_mask_t channelMask,
+                                size_t frameCount,
+                                const sp<IMemory>& sharedBuffer,
+                                int sessionId,
+                                IAudioFlinger::track_flags_t *flags,
+                                pid_t tid,
+                                status_t *status);
+
+                AudioStreamOut* getOutput() const;
+                AudioStreamOut* clearOutput();
+                virtual audio_stream_t* stream() const;
+
+                // a very large number of suspend() will eventually wraparound, but unlikely
+                void        suspend() { (void) android_atomic_inc(&mSuspended); }
+                void        restore()
+                                {
+                                    // if restore() is done without suspend(), get back into
+                                    // range so that the next suspend() will operate correctly
+                                    if (android_atomic_dec(&mSuspended) <= 0) {
+                                        android_atomic_release_store(0, &mSuspended);
+                                    }
+                                }
+                bool        isSuspended() const
+                                { return android_atomic_acquire_load(&mSuspended) > 0; }
+
+    virtual     String8     getParameters(const String8& keys);
+    virtual     void        audioConfigChanged_l(int event, int param = 0);
+                status_t    getRenderPosition(size_t *halFrames, size_t *dspFrames);
+                int16_t     *mixBuffer() const { return mMixBuffer; };
+
+    virtual     void detachAuxEffect_l(int effectId);
+                status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
+                        int EffectId);
+                status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track,
+                        int EffectId);
+
+                virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
+                virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
+                virtual uint32_t hasAudioSession(int sessionId) const;
+                virtual uint32_t getStrategyForSession_l(int sessionId);
+
+
+                virtual status_t setSyncEvent(const sp<SyncEvent>& event);
+                virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
+                        void     invalidateTracks(audio_stream_type_t streamType);
+
+
+protected:
+    int16_t*                        mMixBuffer;
+
+    // suspend count, > 0 means suspended.  While suspended, the thread continues to pull from
+    // tracks and mix, but doesn't write to HAL.  A2DP and SCO HAL implementations can't handle
+    // concurrent use of both of them, so Audio Policy Service suspends one of the threads to
+    // workaround that restriction.
+    // 'volatile' means accessed via atomic operations and no lock.
+    volatile int32_t                mSuspended;
+
+    // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples
+    // mFramesWritten would be better, or 64-bit even better
+    size_t                          mBytesWritten;
+private:
+    // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
+    // PlaybackThread needs to find out if master-muted, it checks it's local
+    // copy rather than the one in AudioFlinger.  This optimization saves a lock.
+    bool                            mMasterMute;
+                void        setMasterMute_l(bool muted) { mMasterMute = muted; }
+protected:
+    SortedVector< wp<Track> >       mActiveTracks;  // FIXME check if this could be sp<>
+
+    // Allocate a track name for a given channel mask.
+    //   Returns name >= 0 if successful, -1 on failure.
+    virtual int             getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
+    virtual void            deleteTrackName_l(int name) = 0;
+
+    // Time to sleep between cycles when:
+    virtual uint32_t        activeSleepTimeUs() const;      // mixer state MIXER_TRACKS_ENABLED
+    virtual uint32_t        idleSleepTimeUs() const = 0;    // mixer state MIXER_IDLE
+    virtual uint32_t        suspendSleepTimeUs() const = 0; // audio policy manager suspended us
+    // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
+    // No sleep in standby mode; waits on a condition
+
+    // Code snippets that are temporarily lifted up out of threadLoop() until the merge
+                void        checkSilentMode_l();
+
+    // Non-trivial for DUPLICATING only
+    virtual     void        saveOutputTracks() { }
+    virtual     void        clearOutputTracks() { }
+
+    // Cache various calculated values, at threadLoop() entry and after a parameter change
+    virtual     void        cacheParameters_l();
+
+    virtual     uint32_t    correctLatency_l(uint32_t latency) const;
+
+private:
+
+    friend class AudioFlinger;      // for numerous
+
+    PlaybackThread(const Client&);
+    PlaybackThread& operator = (const PlaybackThread&);
+
+    status_t    addTrack_l(const sp<Track>& track);
+    void        destroyTrack_l(const sp<Track>& track);
+    void        removeTrack_l(const sp<Track>& track);
+
+    void        readOutputParameters();
+
+    virtual void dumpInternals(int fd, const Vector<String16>& args);
+    void        dumpTracks(int fd, const Vector<String16>& args);
+
+    SortedVector< sp<Track> >       mTracks;
+    // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by
+    // DuplicatingThread
+    stream_type_t                   mStreamTypes[AUDIO_STREAM_CNT + 1];
+    AudioStreamOut                  *mOutput;
+
+    float                           mMasterVolume;
+    nsecs_t                         mLastWriteTime;
+    int                             mNumWrites;
+    int                             mNumDelayedWrites;
+    bool                            mInWrite;
+
+    // FIXME rename these former local variables of threadLoop to standard "m" names
+    nsecs_t                         standbyTime;
+    size_t                          mixBufferSize;
+
+    // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
+    uint32_t                        activeSleepTime;
+    uint32_t                        idleSleepTime;
+
+    uint32_t                        sleepTime;
+
+    // mixer status returned by prepareTracks_l()
+    mixer_state                     mMixerStatus; // current cycle
+                                                  // previous cycle when in prepareTracks_l()
+    mixer_state                     mMixerStatusIgnoringFastTracks;
+                                                  // FIXME or a separate ready state per track
+
+    // FIXME move these declarations into the specific sub-class that needs them
+    // MIXER only
+    uint32_t                        sleepTimeShift;
+
+    // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
+    nsecs_t                         standbyDelay;
+
+    // MIXER only
+    nsecs_t                         maxPeriod;
+
+    // DUPLICATING only
+    uint32_t                        writeFrames;
+
+private:
+    // The HAL output sink is treated as non-blocking, but current implementation is blocking
+    sp<NBAIO_Sink>          mOutputSink;
+    // If a fast mixer is present, the blocking pipe sink, otherwise clear
+    sp<NBAIO_Sink>          mPipeSink;
+    // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
+    sp<NBAIO_Sink>          mNormalSink;
+    // For dumpsys
+    sp<NBAIO_Sink>          mTeeSink;
+    sp<NBAIO_Source>        mTeeSource;
+    uint32_t                mScreenState;   // cached copy of gScreenState
+public:
+    virtual     bool        hasFastMixer() const = 0;
+    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+                                { FastTrackUnderruns dummy; return dummy; }
+
+protected:
+                // accessed by both binder threads and within threadLoop(), lock on mutex needed
+                unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
+
+};
+
+class MixerThread : public PlaybackThread {
+public:
+    MixerThread(const sp<AudioFlinger>& audioFlinger,
+                AudioStreamOut* output,
+                audio_io_handle_t id,
+                audio_devices_t device,
+                type_t type = MIXER);
+    virtual             ~MixerThread();
+
+    // Thread virtuals
+
+    virtual     bool        checkForNewParameters_l();
+    virtual     void        dumpInternals(int fd, const Vector<String16>& args);
+
+protected:
+    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+    virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+    virtual     void        deleteTrackName_l(int name);
+    virtual     uint32_t    idleSleepTimeUs() const;
+    virtual     uint32_t    suspendSleepTimeUs() const;
+    virtual     void        cacheParameters_l();
+
+    // threadLoop snippets
+    virtual     void        threadLoop_write();
+    virtual     void        threadLoop_standby();
+    virtual     void        threadLoop_mix();
+    virtual     void        threadLoop_sleepTime();
+    virtual     void        threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
+    virtual     uint32_t    correctLatency_l(uint32_t latency) const;
+
+                AudioMixer* mAudioMixer;    // normal mixer
+private:
+                // one-time initialization, no locks required
+                FastMixer*  mFastMixer;         // non-NULL if there is also a fast mixer
+                sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
+
+                // contents are not guaranteed to be consistent, no locks required
+                FastMixerDumpState mFastMixerDumpState;
+#ifdef STATE_QUEUE_DUMP
+                StateQueueObserverDump mStateQueueObserverDump;
+                StateQueueMutatorDump  mStateQueueMutatorDump;
+#endif
+                AudioWatchdogDump mAudioWatchdogDump;
+
+                // accessible only within the threadLoop(), no locks required
+                //          mFastMixer->sq()    // for mutating and pushing state
+                int32_t     mFastMixerFutex;    // for cold idle
+
+public:
+    virtual     bool        hasFastMixer() const { return mFastMixer != NULL; }
+    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
+                              ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks);
+                              return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
+                            }
+};
+
+class DirectOutputThread : public PlaybackThread {
+public:
+
+    DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+                       audio_io_handle_t id, audio_devices_t device);
+    virtual                 ~DirectOutputThread();
+
+    // Thread virtuals
+
+    virtual     bool        checkForNewParameters_l();
+
+protected:
+    virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+    virtual     void        deleteTrackName_l(int name);
+    virtual     uint32_t    activeSleepTimeUs() const;
+    virtual     uint32_t    idleSleepTimeUs() const;
+    virtual     uint32_t    suspendSleepTimeUs() const;
+    virtual     void        cacheParameters_l();
+
+    // threadLoop snippets
+    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+    virtual     void        threadLoop_mix();
+    virtual     void        threadLoop_sleepTime();
+
+private:
+    // volumes last sent to audio HAL with stream->set_volume()
+    float mLeftVolFloat;
+    float mRightVolFloat;
+
+    // prepareTracks_l() tells threadLoop_mix() the name of the single active track
+    sp<Track>               mActiveTrack;
+public:
+    virtual     bool        hasFastMixer() const { return false; }
+};
+
+class DuplicatingThread : public MixerThread {
+public:
+    DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+                      audio_io_handle_t id);
+    virtual                 ~DuplicatingThread();
+
+    // Thread virtuals
+                void        addOutputTrack(MixerThread* thread);
+                void        removeOutputTrack(MixerThread* thread);
+                uint32_t    waitTimeMs() const { return mWaitTimeMs; }
+protected:
+    virtual     uint32_t    activeSleepTimeUs() const;
+
+private:
+                bool        outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
+protected:
+    // threadLoop snippets
+    virtual     void        threadLoop_mix();
+    virtual     void        threadLoop_sleepTime();
+    virtual     void        threadLoop_write();
+    virtual     void        threadLoop_standby();
+    virtual     void        cacheParameters_l();
+
+private:
+    // called from threadLoop, addOutputTrack, removeOutputTrack
+    virtual     void        updateWaitTime_l();
+protected:
+    virtual     void        saveOutputTracks();
+    virtual     void        clearOutputTracks();
+private:
+
+                uint32_t    mWaitTimeMs;
+    SortedVector < sp<OutputTrack> >  outputTracks;
+    SortedVector < sp<OutputTrack> >  mOutputTracks;
+public:
+    virtual     bool        hasFastMixer() const { return false; }
+};
+
+
+// record thread
+class RecordThread : public ThreadBase, public AudioBufferProvider
+                        // derives from AudioBufferProvider interface for use by resampler
+{
+public:
+
+#include "RecordTracks.h"
+
+            RecordThread(const sp<AudioFlinger>& audioFlinger,
+                    AudioStreamIn *input,
+                    uint32_t sampleRate,
+                    audio_channel_mask_t channelMask,
+                    audio_io_handle_t id,
+                    audio_devices_t device,
+                    const sp<NBAIO_Sink>& teeSink);
+            virtual     ~RecordThread();
+
+    // no addTrack_l ?
+    void        destroyTrack_l(const sp<RecordTrack>& track);
+    void        removeTrack_l(const sp<RecordTrack>& track);
+
+    void        dumpInternals(int fd, const Vector<String16>& args);
+    void        dumpTracks(int fd, const Vector<String16>& args);
+
+    // Thread virtuals
+    virtual bool        threadLoop();
+    virtual status_t    readyToRun();
+
+    // RefBase
+    virtual void        onFirstRef();
+
+    virtual status_t    initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
+            sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
+                    const sp<AudioFlinger::Client>& client,
+                    uint32_t sampleRate,
+                    audio_format_t format,
+                    audio_channel_mask_t channelMask,
+                    size_t frameCount,
+                    int sessionId,
+                    IAudioFlinger::track_flags_t flags,
+                    pid_t tid,
+                    status_t *status);
+
+            status_t    start(RecordTrack* recordTrack,
+                              AudioSystem::sync_event_t event,
+                              int triggerSession);
+
+            // ask the thread to stop the specified track, and
+            // return true if the caller should then do it's part of the stopping process
+            bool        stop_l(RecordTrack* recordTrack);
+
+            void        dump(int fd, const Vector<String16>& args);
+            AudioStreamIn* clearInput();
+            virtual audio_stream_t* stream() const;
+
+    // AudioBufferProvider interface
+    virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+    virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
+    virtual bool        checkForNewParameters_l();
+    virtual String8     getParameters(const String8& keys);
+    virtual void        audioConfigChanged_l(int event, int param = 0);
+            void        readInputParameters();
+    virtual unsigned int  getInputFramesLost();
+
+    virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
+    virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
+    virtual uint32_t hasAudioSession(int sessionId) const;
+
+            // Return the set of unique session IDs across all tracks.
+            // The keys are the session IDs, and the associated values are meaningless.
+            // FIXME replace by Set [and implement Bag/Multiset for other uses].
+            KeyedVector<int, bool> sessionIds() const;
+
+    virtual status_t setSyncEvent(const sp<SyncEvent>& event);
+    virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
+
+    static void syncStartEventCallback(const wp<SyncEvent>& event);
+           void handleSyncStartEvent(const sp<SyncEvent>& event);
+
+private:
+            void clearSyncStartEvent();
+
+            // Enter standby if not already in standby, and set mStandby flag
+            void standby();
+
+            // Call the HAL standby method unconditionally, and don't change mStandby flag
+            void inputStandBy();
+
+            AudioStreamIn                       *mInput;
+            SortedVector < sp<RecordTrack> >    mTracks;
+            // mActiveTrack has dual roles:  it indicates the current active track, and
+            // is used together with mStartStopCond to indicate start()/stop() progress
+            sp<RecordTrack>                     mActiveTrack;
+            Condition                           mStartStopCond;
+            AudioResampler                      *mResampler;
+            int32_t                             *mRsmpOutBuffer;
+            int16_t                             *mRsmpInBuffer;
+            size_t                              mRsmpInIndex;
+            size_t                              mInputBytes;
+            const uint32_t                      mReqChannelCount;
+            const uint32_t                      mReqSampleRate;
+            ssize_t                             mBytesRead;
+            // sync event triggering actual audio capture. Frames read before this event will
+            // be dropped and therefore not read by the application.
+            sp<SyncEvent>                       mSyncStartEvent;
+            // number of captured frames to drop after the start sync event has been received.
+            // when < 0, maximum frames to drop before starting capture even if sync event is
+            // not received
+            ssize_t                             mFramestoDrop;
+
+            // For dumpsys
+            const sp<NBAIO_Sink>                mTeeSink;
+};
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
new file mode 100644
index 0000000..17de49b
--- /dev/null
+++ b/services/audioflinger/TrackBase.h
@@ -0,0 +1,139 @@
+/*
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+// base for record and playback
+class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
+
+public:
+    enum track_state {
+        IDLE,
+        TERMINATED,
+        FLUSHED,
+        STOPPED,
+        // next 2 states are currently used for fast tracks only
+        STOPPING_1,     // waiting for first underrun
+        STOPPING_2,     // waiting for presentation complete
+        RESUMING,
+        ACTIVE,
+        PAUSING,
+        PAUSED
+    };
+
+                        TrackBase(ThreadBase *thread,
+                                const sp<Client>& client,
+                                uint32_t sampleRate,
+                                audio_format_t format,
+                                audio_channel_mask_t channelMask,
+                                size_t frameCount,
+                                const sp<IMemory>& sharedBuffer,
+                                int sessionId);
+    virtual             ~TrackBase();
+
+    virtual status_t    start(AudioSystem::sync_event_t event,
+                             int triggerSession) = 0;
+    virtual void        stop() = 0;
+            sp<IMemory> getCblk() const { return mCblkMemory; }
+            audio_track_cblk_t* cblk() const { return mCblk; }
+            int         sessionId() const { return mSessionId; }
+    virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
+
+protected:
+                        TrackBase(const TrackBase&);
+                        TrackBase& operator = (const TrackBase&);
+
+    // AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
+    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
+    // ExtendedAudioBufferProvider interface is only needed for Track,
+    // but putting it in TrackBase avoids the complexity of virtual inheritance
+    virtual size_t  framesReady() const { return SIZE_MAX; }
+
+    audio_format_t format() const {
+        return mFormat;
+    }
+
+    uint32_t channelCount() const { return mChannelCount; }
+
+    audio_channel_mask_t channelMask() const { return mChannelMask; }
+
+    uint32_t sampleRate() const; // FIXME inline after cblk sr moved
+
+    // Return a pointer to the start of a contiguous slice of the track buffer.
+    // Parameter 'offset' is the requested start position, expressed in
+    // monotonically increasing frame units relative to the track epoch.
+    // Parameter 'frames' is the requested length, also in frame units.
+    // Always returns non-NULL.  It is the caller's responsibility to
+    // verify that this will be successful; the result of calling this
+    // function with invalid 'offset' or 'frames' is undefined.
+    void* getBuffer(uint32_t offset, uint32_t frames) const;
+
+    bool isStopped() const {
+        return (mState == STOPPED || mState == FLUSHED);
+    }
+
+    // for fast tracks only
+    bool isStopping() const {
+        return mState == STOPPING_1 || mState == STOPPING_2;
+    }
+    bool isStopping_1() const {
+        return mState == STOPPING_1;
+    }
+    bool isStopping_2() const {
+        return mState == STOPPING_2;
+    }
+
+    bool isTerminated() const {
+        return mState == TERMINATED;
+    }
+
+    bool step();    // mStepCount is an implicit input
+    void reset();
+
+    virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack,
+                                    // this could be a track type if needed later
+
+    const wp<ThreadBase> mThread;
+    /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
+    sp<IMemory>         mCblkMemory;
+    audio_track_cblk_t* mCblk;
+    void*               mBuffer;    // start of track buffer, typically in shared memory
+    void*               mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize
+                                    //   is based on mChannelCount and 16-bit samples
+    uint32_t            mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of
+                                    // time of releaseBuffer() for later use by step()
+    // we don't really need a lock for these
+    track_state         mState;
+    const uint32_t      mSampleRate;    // initial sample rate only; for tracks which
+                        // support dynamic rates, the current value is in control block
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const uint8_t       mChannelCount;
+    const size_t        mFrameSize; // AudioFlinger's view of frame size in shared memory,
+                                    // where for AudioTrack (but not AudioRecord),
+                                    // 8-bit PCM samples are stored as 16-bit
+    const size_t        mFrameCount;// size of track buffer given at createTrack() or
+                                    // openRecord(), and then adjusted as needed
+
+    bool                mStepServerFailed;
+    const int           mSessionId;
+    Vector < sp<SyncEvent> >mSyncEvents;
+};
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
new file mode 100644
index 0000000..2c6ba8b
--- /dev/null
+++ b/services/audioflinger/Tracks.cpp
@@ -0,0 +1,1789 @@
+/*
+**
+** 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.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message.  In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on.  Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+//      TrackBase
+// ----------------------------------------------------------------------------
+
+// TrackBase constructor must be called with AudioFlinger::mLock held
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+            ThreadBase *thread,
+            const sp<Client>& client,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId)
+    :   RefBase(),
+        mThread(thread),
+        mClient(client),
+        mCblk(NULL),
+        // mBuffer
+        // mBufferEnd
+        mStepCount(0),
+        mState(IDLE),
+        mSampleRate(sampleRate),
+        mFormat(format),
+        mChannelMask(channelMask),
+        mChannelCount(popcount(channelMask)),
+        mFrameSize(audio_is_linear_pcm(format) ?
+                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
+        mFrameCount(frameCount),
+        mStepServerFailed(false),
+        mSessionId(sessionId)
+{
+    // client == 0 implies sharedBuffer == 0
+    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
+
+    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
+            sharedBuffer->size());
+
+    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+    size_t size = sizeof(audio_track_cblk_t);
+    size_t bufferSize = frameCount * mFrameSize;
+    if (sharedBuffer == 0) {
+        size += bufferSize;
+    }
+
+    if (client != 0) {
+        mCblkMemory = client->heap()->allocate(size);
+        if (mCblkMemory != 0) {
+            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
+            // can't assume mCblk != NULL
+        } else {
+            ALOGE("not enough memory for AudioTrack size=%u", size);
+            client->heap()->dump("AudioTrack");
+            return;
+        }
+    } else {
+        mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+        // assume mCblk != NULL
+    }
+
+    // construct the shared structure in-place.
+    if (mCblk != NULL) {
+        new(mCblk) audio_track_cblk_t();
+        // clear all buffers
+        mCblk->frameCount_ = frameCount;
+        mCblk->sampleRate = sampleRate;
+// uncomment the following lines to quickly test 32-bit wraparound
+//      mCblk->user = 0xffff0000;
+//      mCblk->server = 0xffff0000;
+//      mCblk->userBase = 0xffff0000;
+//      mCblk->serverBase = 0xffff0000;
+        if (sharedBuffer == 0) {
+            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+            memset(mBuffer, 0, bufferSize);
+            // Force underrun condition to avoid false underrun callback until first data is
+            // written to buffer (other flags are cleared)
+            mCblk->flags = CBLK_UNDERRUN;
+        } else {
+            mBuffer = sharedBuffer->pointer();
+        }
+        mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+    }
+}
+
+AudioFlinger::ThreadBase::TrackBase::~TrackBase()
+{
+    if (mCblk != NULL) {
+        if (mClient == 0) {
+            delete mCblk;
+        } else {
+            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
+        }
+    }
+    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
+    if (mClient != 0) {
+        // Client destructor must run with AudioFlinger mutex locked
+        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+        // If the client's reference count drops to zero, the associated destructor
+        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
+        // relying on the automatic clear() at end of scope.
+        mClient.clear();
+    }
+}
+
+// AudioBufferProvider interface
+// getNextBuffer() = 0;
+// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
+void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    buffer->raw = NULL;
+    mStepCount = buffer->frameCount;
+    // FIXME See note at getNextBuffer()
+    (void) step();      // ignore return value of step()
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::ThreadBase::TrackBase::step() {
+    bool result;
+    audio_track_cblk_t* cblk = this->cblk();
+
+    result = cblk->stepServer(mStepCount, mFrameCount, isOut());
+    if (!result) {
+        ALOGV("stepServer failed acquiring cblk mutex");
+        mStepServerFailed = true;
+    }
+    return result;
+}
+
+void AudioFlinger::ThreadBase::TrackBase::reset() {
+    audio_track_cblk_t* cblk = this->cblk();
+
+    cblk->user = 0;
+    cblk->server = 0;
+    cblk->userBase = 0;
+    cblk->serverBase = 0;
+    mStepServerFailed = false;
+    ALOGV("TrackBase::reset");
+}
+
+uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
+    return mCblk->sampleRate;
+}
+
+void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+    audio_track_cblk_t* cblk = this->cblk();
+    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
+    int8_t *bufferEnd = bufferStart + frames * mFrameSize;
+
+    // Check validity of returned pointer in case the track control block would have been corrupted.
+    ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
+            "TrackBase::getBuffer buffer out of range:\n"
+                "    start: %p, end %p , mBuffer %p mBufferEnd %p\n"
+                "    server %u, serverBase %u, user %u, userBase %u, frameSize %u",
+                bufferStart, bufferEnd, mBuffer, mBufferEnd,
+                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
+
+    return bufferStart;
+}
+
+status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
+{
+    mSyncEvents.add(event);
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+//      Playback
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
+    : BnAudioTrack(),
+      mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+    // just stop the track on deletion, associated resources
+    // will be freed from the main thread once all pending buffers have
+    // been played. Unless it's not in the active track list, in which
+    // case we free everything now...
+    mTrack->destroy();
+}
+
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+    return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+    return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+    mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+    mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+    mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+    mTrack->pause();
+}
+
+status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
+{
+    return mTrack->attachAuxEffect(EffectId);
+}
+
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+                                                         sp<IMemory>* buffer) {
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+                                                     int64_t pts) {
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+    const LinearTransform& xform, int target) {
+
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->setMediaTimeTransform(
+        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
+status_t AudioFlinger::TrackHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+            PlaybackThread *thread,
+            const sp<Client>& client,
+            audio_stream_type_t streamType,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId,
+            IAudioFlinger::track_flags_t flags)
+    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
+            sessionId),
+    mMute(false),
+    mFillingUpStatus(FS_INVALID),
+    // mRetryCount initialized later when needed
+    mSharedBuffer(sharedBuffer),
+    mStreamType(streamType),
+    mName(-1),  // see note below
+    mMainBuffer(thread->mixBuffer()),
+    mAuxBuffer(NULL),
+    mAuxEffectId(0), mHasVolumeController(false),
+    mPresentationCompleteFrames(0),
+    mFlags(flags),
+    mFastIndex(-1),
+    mUnderrunCount(0),
+    mCachedVolume(1.0)
+{
+    if (mCblk != NULL) {
+        // to avoid leaking a track name, do not allocate one unless there is an mCblk
+        mName = thread->getTrackName_l(channelMask, sessionId);
+        mCblk->mName = mName;
+        if (mName < 0) {
+            ALOGE("no more track names available");
+            return;
+        }
+        // only allocate a fast track index if we were able to allocate a normal track name
+        if (flags & IAudioFlinger::TRACK_FAST) {
+            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
+            int i = __builtin_ctz(thread->mFastTrackAvailMask);
+            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
+            // FIXME This is too eager.  We allocate a fast track index before the
+            //       fast track becomes active.  Since fast tracks are a scarce resource,
+            //       this means we are potentially denying other more important fast tracks from
+            //       being created.  It would be better to allocate the index dynamically.
+            mFastIndex = i;
+            mCblk->mName = i;
+            // Read the initial underruns because this field is never cleared by the fast mixer
+            mObservedUnderruns = thread->getFastTrackUnderruns(i);
+            thread->mFastTrackAvailMask &= ~(1 << i);
+        }
+    }
+    ALOGV("Track constructor name %d, calling pid %d", mName,
+            IPCThreadState::self()->getCallingPid());
+}
+
+AudioFlinger::PlaybackThread::Track::~Track()
+{
+    ALOGV("PlaybackThread::Track destructor");
+}
+
+void AudioFlinger::PlaybackThread::Track::destroy()
+{
+    // NOTE: destroyTrack_l() can remove a strong reference to this Track
+    // by removing it from mTracks vector, so there is a risk that this Tracks's
+    // destructor is called. As the destructor needs to lock mLock,
+    // we must acquire a strong reference on this Track before locking mLock
+    // here so that the destructor is called only when exiting this function.
+    // On the other hand, as long as Track::destroy() is only called by
+    // TrackHandle destructor, the TrackHandle still holds a strong ref on
+    // this Track with its member mTrack.
+    sp<Track> keep(this);
+    { // scope for mLock
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            if (!isOutputTrack()) {
+                if (mState == ACTIVE || mState == RESUMING) {
+                    AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+
+#ifdef ADD_BATTERY_DATA
+                    // to track the speaker usage
+                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+                }
+                AudioSystem::releaseOutput(thread->id());
+            }
+            Mutex::Autolock _l(thread->mLock);
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->destroyTrack_l(this);
+        }
+    }
+}
+
+/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
+{
+    result.append("   Name Client Type Fmt Chn mask   Session StpCnt fCount S M F SRate  "
+                  "L dB  R dB    Server      User     Main buf    Aux Buf  Flags Underruns\n");
+}
+
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+{
+    uint32_t vlr = mCblk->getVolumeLR();
+    if (isFastTrack()) {
+        sprintf(buffer, "   F %2d", mFastIndex);
+    } else {
+        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
+    }
+    track_state state = mState;
+    char stateChar;
+    switch (state) {
+    case IDLE:
+        stateChar = 'I';
+        break;
+    case TERMINATED:
+        stateChar = 'T';
+        break;
+    case STOPPING_1:
+        stateChar = 's';
+        break;
+    case STOPPING_2:
+        stateChar = '5';
+        break;
+    case STOPPED:
+        stateChar = 'S';
+        break;
+    case RESUMING:
+        stateChar = 'R';
+        break;
+    case ACTIVE:
+        stateChar = 'A';
+        break;
+    case PAUSING:
+        stateChar = 'p';
+        break;
+    case PAUSED:
+        stateChar = 'P';
+        break;
+    case FLUSHED:
+        stateChar = 'F';
+        break;
+    default:
+        stateChar = '?';
+        break;
+    }
+    char nowInUnderrun;
+    switch (mObservedUnderruns.mBitFields.mMostRecent) {
+    case UNDERRUN_FULL:
+        nowInUnderrun = ' ';
+        break;
+    case UNDERRUN_PARTIAL:
+        nowInUnderrun = '<';
+        break;
+    case UNDERRUN_EMPTY:
+        nowInUnderrun = '*';
+        break;
+    default:
+        nowInUnderrun = '?';
+        break;
+    }
+    snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g  "
+            "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
+            (mClient == 0) ? getpid_cached : mClient->pid(),
+            mStreamType,
+            mFormat,
+            mChannelMask,
+            mSessionId,
+            mStepCount,
+            mFrameCount,
+            stateChar,
+            mMute,
+            mFillingUpStatus,
+            mCblk->sampleRate,
+            20.0 * log10((vlr & 0xFFFF) / 4096.0),
+            20.0 * log10((vlr >> 16) / 4096.0),
+            mCblk->server,
+            mCblk->user,
+            (int)mMainBuffer,
+            (int)mAuxBuffer,
+            mCblk->flags,
+            mUnderrunCount,
+            nowInUnderrun);
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+        AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+    audio_track_cblk_t* cblk = this->cblk();
+    uint32_t framesReady;
+    uint32_t framesReq = buffer->frameCount;
+
+    // Check if last stepServer failed, try to step now
+    if (mStepServerFailed) {
+        // FIXME When called by fast mixer, this takes a mutex with tryLock().
+        //       Since the fast mixer is higher priority than client callback thread,
+        //       it does not result in priority inversion for client.
+        //       But a non-blocking solution would be preferable to avoid
+        //       fast mixer being unable to tryLock(), and
+        //       to avoid the extra context switches if the client wakes up,
+        //       discovers the mutex is locked, then has to wait for fast mixer to unlock.
+        if (!step())  goto getNextBuffer_exit;
+        ALOGV("stepServer recovered");
+        mStepServerFailed = false;
+    }
+
+    // FIXME Same as above
+    framesReady = cblk->framesReadyOut();
+
+    if (CC_LIKELY(framesReady)) {
+        uint32_t s = cblk->server;
+        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
+
+        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
+        if (framesReq > framesReady) {
+            framesReq = framesReady;
+        }
+        if (framesReq > bufferEnd - s) {
+            framesReq = bufferEnd - s;
+        }
+
+        buffer->raw = getBuffer(s, framesReq);
+        buffer->frameCount = framesReq;
+        return NO_ERROR;
+    }
+
+getNextBuffer_exit:
+    buffer->raw = NULL;
+    buffer->frameCount = 0;
+    ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+    return NOT_ENOUGH_DATA;
+}
+
+// Note that framesReady() takes a mutex on the control block using tryLock().
+// This could result in priority inversion if framesReady() is called by the normal mixer,
+// as the normal mixer thread runs at lower
+// priority than the client's callback thread:  there is a short window within framesReady()
+// during which the normal mixer could be preempted, and the client callback would block.
+// Another problem can occur if framesReady() is called by the fast mixer:
+// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
+// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
+size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
+    return mCblk->framesReadyOut();
+}
+
+// Don't call for fast tracks; the framesReady() could result in priority inversion
+bool AudioFlinger::PlaybackThread::Track::isReady() const {
+    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+        return true;
+    }
+
+    if (framesReady() >= mFrameCount ||
+            (mCblk->flags & CBLK_FORCEREADY)) {
+        mFillingUpStatus = FS_FILLED;
+        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
+        return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
+                                                    int triggerSession)
+{
+    status_t status = NO_ERROR;
+    ALOGV("start(%d), calling pid %d session %d",
+            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
+
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        track_state state = mState;
+        // here the track could be either new, or restarted
+        // in both cases "unstop" the track
+        if (mState == PAUSED) {
+            mState = TrackBase::RESUMING;
+            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
+        } else {
+            mState = TrackBase::ACTIVE;
+            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
+        }
+
+        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
+            thread->mLock.unlock();
+            status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
+            thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+            // to track the speaker usage
+            if (status == NO_ERROR) {
+                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
+            }
+#endif
+        }
+        if (status == NO_ERROR) {
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->addTrack_l(this);
+        } else {
+            mState = state;
+            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+        }
+    } else {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::stop()
+{
+    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        track_state state = mState;
+        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
+            // If the track is not active (PAUSED and buffers full), flush buffers
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+                reset();
+                mState = STOPPED;
+            } else if (!isFastTrack()) {
+                mState = STOPPED;
+            } else {
+                // prepareTracks_l() will set state to STOPPING_2 after next underrun,
+                // and then to STOPPED and reset() when presentation is complete
+                mState = STOPPING_1;
+            }
+            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
+                    playbackThread);
+        }
+        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
+            thread->mLock.unlock();
+            AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+            thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+            // to track the speaker usage
+            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::pause()
+{
+    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState == ACTIVE || mState == RESUMING) {
+            mState = PAUSING;
+            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
+            if (!isOutputTrack()) {
+                thread->mLock.unlock();
+                AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+                thread->mLock.lock();
+
+#ifdef ADD_BATTERY_DATA
+                // to track the speaker usage
+                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
+            }
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::flush()
+{
+    ALOGV("flush(%d)", mName);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
+                mState != PAUSING && mState != IDLE && mState != FLUSHED) {
+            return;
+        }
+        // No point remaining in PAUSED state after a flush => go to
+        // FLUSHED state
+        mState = FLUSHED;
+        // do not reset the track if it is still in the process of being stopped or paused.
+        // this will be done by prepareTracks_l() when the track is stopped.
+        // prepareTracks_l() will see mState == FLUSHED, then
+        // remove from active track list, reset(), and trigger presentation complete
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+            reset();
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::reset()
+{
+    // Do not reset twice to avoid discarding data written just after a flush and before
+    // the audioflinger thread detects the track is stopped.
+    if (!mResetDone) {
+        TrackBase::reset();
+        // Force underrun condition to avoid false underrun callback until first data is
+        // written to buffer
+        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
+        android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
+        mFillingUpStatus = FS_FILLING;
+        mResetDone = true;
+        if (mState == FLUSHED) {
+            mState = IDLE;
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
+{
+    mMute = muted;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
+{
+    status_t status = DEAD_OBJECT;
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        sp<AudioFlinger> af = mClient->audioFlinger();
+
+        Mutex::Autolock _l(af->mLock);
+
+        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+
+        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
+            Mutex::Autolock _dl(playbackThread->mLock);
+            Mutex::Autolock _sl(srcThread->mLock);
+            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+            if (chain == 0) {
+                return INVALID_OPERATION;
+            }
+
+            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
+            if (effect == 0) {
+                return INVALID_OPERATION;
+            }
+            srcThread->removeEffect_l(effect);
+            playbackThread->addEffect_l(effect);
+            // removeEffect_l() has stopped the effect if it was active so it must be restarted
+            if (effect->state() == EffectModule::ACTIVE ||
+                    effect->state() == EffectModule::STOPPING) {
+                effect->start();
+            }
+
+            sp<EffectChain> dstChain = effect->chain().promote();
+            if (dstChain == 0) {
+                srcThread->addEffect_l(effect);
+                return INVALID_OPERATION;
+            }
+            AudioSystem::unregisterEffect(effect->id());
+            AudioSystem::registerEffect(&effect->desc(),
+                                        srcThread->id(),
+                                        dstChain->strategy(),
+                                        AUDIO_SESSION_OUTPUT_MIX,
+                                        effect->id());
+        }
+        status = playbackThread->attachAuxEffect(this, EffectId);
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
+{
+    mAuxEffectId = EffectId;
+    mAuxBuffer = buffer;
+}
+
+bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
+                                                         size_t audioHalFrames)
+{
+    // a track is considered presented when the total number of frames written to audio HAL
+    // corresponds to the number of frames written when presentationComplete() is called for the
+    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
+    if (mPresentationCompleteFrames == 0) {
+        mPresentationCompleteFrames = framesWritten + audioHalFrames;
+        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
+                  mPresentationCompleteFrames, audioHalFrames);
+    }
+    if (framesWritten >= mPresentationCompleteFrames) {
+        ALOGV("presentationComplete() session %d complete: framesWritten %d",
+                  mSessionId, framesWritten);
+        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+        return true;
+    }
+    return false;
+}
+
+void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
+{
+    for (int i = 0; i < (int)mSyncEvents.size(); i++) {
+        if (mSyncEvents[i]->type() == type) {
+            mSyncEvents[i]->trigger();
+            mSyncEvents.removeAt(i);
+            i--;
+        }
+    }
+}
+
+// implement VolumeBufferProvider interface
+
+uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+{
+    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
+    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
+    uint32_t vlr = mCblk->getVolumeLR();
+    uint32_t vl = vlr & 0xFFFF;
+    uint32_t vr = vlr >> 16;
+    // track volumes come from shared memory, so can't be trusted and must be clamped
+    if (vl > MAX_GAIN_INT) {
+        vl = MAX_GAIN_INT;
+    }
+    if (vr > MAX_GAIN_INT) {
+        vr = MAX_GAIN_INT;
+    }
+    // now apply the cached master volume and stream type volume;
+    // this is trusted but lacks any synchronization or barrier so may be stale
+    float v = mCachedVolume;
+    vl *= v;
+    vr *= v;
+    // re-combine into U4.16
+    vlr = (vr << 16) | (vl & 0xFFFF);
+    // FIXME look at mute, pause, and stop flags
+    return vlr;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
+{
+    if (mState == TERMINATED || mState == PAUSED ||
+            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
+                                      (mState == STOPPED)))) {
+        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
+              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
+        event->cancel();
+        return INVALID_OPERATION;
+    }
+    (void) TrackBase::setSyncEvent(event);
+    return NO_ERROR;
+}
+
+bool AudioFlinger::PlaybackThread::Track::isOut() const
+{
+    return true;
+}
+
+// ----------------------------------------------------------------------------
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+            PlaybackThread *thread,
+            const sp<Client>& client,
+            audio_stream_type_t streamType,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId) {
+    if (!client->reserveTimedTrack())
+        return 0;
+
+    return new TimedTrack(
+        thread, client, streamType, sampleRate, format, channelMask, frameCount,
+        sharedBuffer, sessionId);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+            PlaybackThread *thread,
+            const sp<Client>& client,
+            audio_stream_type_t streamType,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId)
+    : Track(thread, client, streamType, sampleRate, format, channelMask,
+            frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
+      mQueueHeadInFlight(false),
+      mTrimQueueHeadOnRelease(false),
+      mFramesPendingInQueue(0),
+      mTimedSilenceBuffer(NULL),
+      mTimedSilenceBufferSize(0),
+      mTimedAudioOutputOnTime(false),
+      mMediaTimeTransformValid(false)
+{
+    LocalClock lc;
+    mLocalTimeFreq = lc.getLocalFreq();
+
+    mLocalTimeToSampleTransform.a_zero = 0;
+    mLocalTimeToSampleTransform.b_zero = 0;
+    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+                            &mLocalTimeToSampleTransform.a_to_b_denom);
+
+    mMediaTimeToSampleTransform.a_zero = 0;
+    mMediaTimeToSampleTransform.b_zero = 0;
+    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
+    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
+    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
+                            &mMediaTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+    mClient->releaseTimedTrack();
+    delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+    size_t size, sp<IMemory>* buffer) {
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    trimTimedBufferQueue_l();
+
+    // lazily initialize the shared memory heap for timed buffers
+    if (mTimedMemoryDealer == NULL) {
+        const int kTimedBufferHeapSize = 512 << 10;
+
+        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+                                              "AudioFlingerTimed");
+        if (mTimedMemoryDealer == NULL)
+            return NO_MEMORY;
+    }
+
+    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+    if (newBuffer == NULL) {
+        newBuffer = mTimedMemoryDealer->allocate(size);
+        if (newBuffer == NULL)
+            return NO_MEMORY;
+    }
+
+    *buffer = newBuffer;
+    return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+    int64_t mediaTimeNow;
+    {
+        Mutex::Autolock mttLock(mMediaTimeTransformLock);
+        if (!mMediaTimeTransformValid)
+            return;
+
+        int64_t targetTimeNow;
+        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+            ? mCCHelper.getCommonTime(&targetTimeNow)
+            : mCCHelper.getLocalTime(&targetTimeNow);
+
+        if (OK != res)
+            return;
+
+        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+                                                    &mediaTimeNow)) {
+            return;
+        }
+    }
+
+    size_t trimEnd;
+    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
+        int64_t bufEnd;
+
+        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
+            // We have a next buffer.  Just use its PTS as the PTS of the frame
+            // following the last frame in this buffer.  If the stream is sparse
+            // (ie, there are deliberate gaps left in the stream which should be
+            // filled with silence by the TimedAudioTrack), then this can result
+            // in one extra buffer being left un-trimmed when it could have
+            // been.  In general, this is not typical, and we would rather
+            // optimized away the TS calculation below for the more common case
+            // where PTSes are contiguous.
+            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
+        } else {
+            // We have no next buffer.  Compute the PTS of the frame following
+            // the last frame in this buffer by computing the duration of of
+            // this frame in media time units and adding it to the PTS of the
+            // buffer.
+            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
+                               / mFrameSize;
+
+            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
+                                                                &bufEnd)) {
+                ALOGE("Failed to convert frame count of %lld to media time"
+                      " duration" " (scale factor %d/%u) in %s",
+                      frameCount,
+                      mMediaTimeToSampleTransform.a_to_b_numer,
+                      mMediaTimeToSampleTransform.a_to_b_denom,
+                      __PRETTY_FUNCTION__);
+                break;
+            }
+            bufEnd += mTimedBufferQueue[trimEnd].pts();
+        }
+
+        if (bufEnd > mediaTimeNow)
+            break;
+
+        // Is the buffer we want to use in the middle of a mix operation right
+        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
+        // from the mixer which should be coming back shortly.
+        if (!trimEnd && mQueueHeadInFlight) {
+            mTrimQueueHeadOnRelease = true;
+        }
+    }
+
+    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
+    if (trimStart < trimEnd) {
+        // Update the bookkeeping for framesReady()
+        for (size_t i = trimStart; i < trimEnd; ++i) {
+            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
+        }
+
+        // Now actually remove the buffers from the queue.
+        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
+    }
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
+        const char* logTag) {
+    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
+                "%s called (reason \"%s\"), but timed buffer queue has no"
+                " elements to trim.", __FUNCTION__, logTag);
+
+    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
+    mTimedBufferQueue.removeAt(0);
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
+        const TimedBuffer& buf,
+        const char* logTag) {
+    uint32_t bufBytes        = buf.buffer()->size();
+    uint32_t consumedAlready = buf.position();
+
+    ALOG_ASSERT(consumedAlready <= bufBytes,
+                "Bad bookkeeping while updating frames pending.  Timed buffer is"
+                " only %u bytes long, but claims to have consumed %u"
+                " bytes.  (update reason: \"%s\")",
+                bufBytes, consumedAlready, logTag);
+
+    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
+    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
+                "Bad bookkeeping while updating frames pending.  Should have at"
+                " least %u queued frames, but we think we have only %u.  (update"
+                " reason: \"%s\")",
+                bufFrames, mFramesPendingInQueue, logTag);
+
+    mFramesPendingInQueue -= bufFrames;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+    const sp<IMemory>& buffer, int64_t pts) {
+
+    {
+        Mutex::Autolock mttLock(mMediaTimeTransformLock);
+        if (!mMediaTimeTransformValid)
+            return INVALID_OPERATION;
+    }
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    uint32_t bufFrames = buffer->size() / mFrameSize;
+    mFramesPendingInQueue += bufFrames;
+    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
+           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+           target);
+
+    if (!(target == TimedAudioTrack::LOCAL_TIME ||
+          target == TimedAudioTrack::COMMON_TIME)) {
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock lock(mMediaTimeTransformLock);
+    mMediaTimeTransform = xform;
+    mMediaTimeTransformTarget = target;
+    mMediaTimeTransformValid = true;
+
+    return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+    AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+    if (pts == AudioBufferProvider::kInvalidPTS) {
+        buffer->raw = NULL;
+        buffer->frameCount = 0;
+        mTimedAudioOutputOnTime = false;
+        return INVALID_OPERATION;
+    }
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    ALOG_ASSERT(!mQueueHeadInFlight,
+                "getNextBuffer called without releaseBuffer!");
+
+    while (true) {
+
+        // if we have no timed buffers, then fail
+        if (mTimedBufferQueue.isEmpty()) {
+            buffer->raw = NULL;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+
+        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+        // calculate the PTS of the head of the timed buffer queue expressed in
+        // local time
+        int64_t headLocalPTS;
+        {
+            Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
+
+            if (mMediaTimeTransform.a_to_b_denom == 0) {
+                // the transform represents a pause, so yield silence
+                timedYieldSilence_l(buffer->frameCount, buffer);
+                return NO_ERROR;
+            }
+
+            int64_t transformedPTS;
+            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+                                                        &transformedPTS)) {
+                // the transform failed.  this shouldn't happen, but if it does
+                // then just drop this buffer
+                ALOGW("timedGetNextBuffer transform failed");
+                buffer->raw = NULL;
+                buffer->frameCount = 0;
+                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
+                return NO_ERROR;
+            }
+
+            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
+                                                          &headLocalPTS)) {
+                    buffer->raw = NULL;
+                    buffer->frameCount = 0;
+                    return INVALID_OPERATION;
+                }
+            } else {
+                headLocalPTS = transformedPTS;
+            }
+        }
+
+        // adjust the head buffer's PTS to reflect the portion of the head buffer
+        // that has already been consumed
+        int64_t effectivePTS = headLocalPTS +
+                ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
+
+        // Calculate the delta in samples between the head of the input buffer
+        // queue and the start of the next output buffer that will be written.
+        // If the transformation fails because of over or underflow, it means
+        // that the sample's position in the output stream is so far out of
+        // whack that it should just be dropped.
+        int64_t sampleDelta;
+        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+            ALOGV("*** head buffer is too far from PTS: dropped buffer");
+            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
+                                       " mix");
+            continue;
+        }
+        if (!mLocalTimeToSampleTransform.doForwardTransform(
+                (effectivePTS - pts) << 32, &sampleDelta)) {
+            ALOGV("*** too late during sample rate transform: dropped buffer");
+            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
+            continue;
+        }
+
+        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
+               " sampleDelta=[%d.%08x]",
+               head.pts(), head.position(), pts,
+               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
+                   + (sampleDelta >> 32)),
+               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+        // if the delta between the ideal placement for the next input sample and
+        // the current output position is within this threshold, then we will
+        // concatenate the next input samples to the previous output
+        const int64_t kSampleContinuityThreshold =
+                (static_cast<int64_t>(sampleRate()) << 32) / 250;
+
+        // if this is the first buffer of audio that we're emitting from this track
+        // then it should be almost exactly on time.
+        const int64_t kSampleStartupThreshold = 1LL << 32;
+
+        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+            // the next input is close enough to being on time, so concatenate it
+            // with the last output
+            timedYieldSamples_l(buffer);
+
+            ALOGVV("*** on time: head.pos=%d frameCount=%u",
+                    head.position(), buffer->frameCount);
+            return NO_ERROR;
+        }
+
+        // Looks like our output is not on time.  Reset our on timed status.
+        // Next time we mix samples from our input queue, then should be within
+        // the StartupThreshold.
+        mTimedAudioOutputOnTime = false;
+        if (sampleDelta > 0) {
+            // the gap between the current output position and the proper start of
+            // the next input sample is too big, so fill it with silence
+            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+            timedYieldSilence_l(framesUntilNextInput, buffer);
+            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
+            return NO_ERROR;
+        } else {
+            // the next input sample is late
+            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+            size_t onTimeSamplePosition =
+                    head.position() + lateFrames * mFrameSize;
+
+            if (onTimeSamplePosition > head.buffer()->size()) {
+                // all the remaining samples in the head are too late, so
+                // drop it and move on
+                ALOGV("*** too late: dropped buffer");
+                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
+                continue;
+            } else {
+                // skip over the late samples
+                head.setPosition(onTimeSamplePosition);
+
+                // yield the available samples
+                timedYieldSamples_l(buffer);
+
+                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+                return NO_ERROR;
+            }
+        }
+    }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
+    AudioBufferProvider::Buffer* buffer) {
+
+    const TimedBuffer& head = mTimedBufferQueue[0];
+
+    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+                   head.position());
+
+    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+                                 mFrameSize);
+    size_t framesRequested = buffer->frameCount;
+    buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+    mQueueHeadInFlight = true;
+    mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
+    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+    // lazily allocate a buffer filled with silence
+    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
+        delete [] mTimedSilenceBuffer;
+        mTimedSilenceBufferSize = numFrames * mFrameSize;
+        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+    }
+
+    buffer->raw = mTimedSilenceBuffer;
+    size_t framesRequested = buffer->frameCount;
+    buffer->frameCount = min(numFrames, framesRequested);
+
+    mTimedAudioOutputOnTime = false;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+    AudioBufferProvider::Buffer* buffer) {
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    // If the buffer which was just released is part of the buffer at the head
+    // of the queue, be sure to update the amt of the buffer which has been
+    // consumed.  If the buffer being returned is not part of the head of the
+    // queue, its either because the buffer is part of the silence buffer, or
+    // because the head of the timed queue was trimmed after the mixer called
+    // getNextBuffer but before the mixer called releaseBuffer.
+    if (buffer->raw == mTimedSilenceBuffer) {
+        ALOG_ASSERT(!mQueueHeadInFlight,
+                    "Queue head in flight during release of silence buffer!");
+        goto done;
+    }
+
+    ALOG_ASSERT(mQueueHeadInFlight,
+                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
+                " head in flight.");
+
+    if (mTimedBufferQueue.size()) {
+        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+        void* start = head.buffer()->pointer();
+        void* end   = reinterpret_cast<void*>(
+                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
+                        + head.buffer()->size());
+
+        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
+                    "released buffer not within the head of the timed buffer"
+                    " queue; qHead = [%p, %p], released buffer = %p",
+                    start, end, buffer->raw);
+
+        head.setPosition(head.position() +
+                (buffer->frameCount * mFrameSize));
+        mQueueHeadInFlight = false;
+
+        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
+                    "Bad bookkeeping during releaseBuffer!  Should have at"
+                    " least %u queued frames, but we think we have only %u",
+                    buffer->frameCount, mFramesPendingInQueue);
+
+        mFramesPendingInQueue -= buffer->frameCount;
+
+        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
+            || mTrimQueueHeadOnRelease) {
+            trimTimedBufferQueueHead_l("releaseBuffer");
+            mTrimQueueHeadOnRelease = false;
+        }
+    } else {
+        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
+                  " buffers in the timed buffer queue");
+    }
+
+done:
+    buffer->raw = 0;
+    buffer->frameCount = 0;
+}
+
+size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+    return mFramesPendingInQueue;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+        : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+    const sp<IMemory>& buffer, int64_t pts)
+        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+            PlaybackThread *playbackThread,
+            DuplicatingThread *sourceThread,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount)
+    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
+                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
+    mActive(false), mSourceThread(sourceThread), mBuffers(NULL)
+{
+
+    if (mCblk != NULL) {
+        mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+        mOutBuffer.frameCount = 0;
+        playbackThread->mTracks.add(this);
+        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \
+                "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p",
+                mCblk, mBuffer, mBuffers,
+                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
+    } else {
+        ALOGW("Error creating output track on thread %p", playbackThread);
+    }
+}
+
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
+{
+    clearBufferQueue();
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
+                                                          int triggerSession)
+{
+    status_t status = Track::start(event, triggerSession);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    mActive = true;
+    mRetryCount = 127;
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
+{
+    Track::stop();
+    clearBufferQueue();
+    mOutBuffer.frameCount = 0;
+    mActive = false;
+}
+
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+{
+    Buffer *pInBuffer;
+    Buffer inBuffer;
+    uint32_t channelCount = mChannelCount;
+    bool outputBufferFull = false;
+    inBuffer.frameCount = frames;
+    inBuffer.i16 = data;
+
+    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
+
+    if (!mActive && frames != 0) {
+        start();
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            MixerThread *mixerThread = (MixerThread *)thread.get();
+            if (mFrameCount > frames) {
+                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+                    uint32_t startFrames = (mFrameCount - frames);
+                    pInBuffer = new Buffer;
+                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+                    pInBuffer->frameCount = startFrames;
+                    pInBuffer->i16 = pInBuffer->mBuffer;
+                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+                    mBufferQueue.add(pInBuffer);
+                } else {
+                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
+                }
+            }
+        }
+    }
+
+    while (waitTimeLeftMs) {
+        // First write pending buffers, then new data
+        if (mBufferQueue.size()) {
+            pInBuffer = mBufferQueue.itemAt(0);
+        } else {
+            pInBuffer = &inBuffer;
+        }
+
+        if (pInBuffer->frameCount == 0) {
+            break;
+        }
+
+        if (mOutBuffer.frameCount == 0) {
+            mOutBuffer.frameCount = pInBuffer->frameCount;
+            nsecs_t startTime = systemTime();
+            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
+                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
+                        mThread.unsafe_get());
+                outputBufferFull = true;
+                break;
+            }
+            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+            if (waitTimeLeftMs >= waitTimeMs) {
+                waitTimeLeftMs -= waitTimeMs;
+            } else {
+                waitTimeLeftMs = 0;
+            }
+        }
+
+        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
+                pInBuffer->frameCount;
+        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+        mCblk->stepUserOut(outFrames, mFrameCount);
+        pInBuffer->frameCount -= outFrames;
+        pInBuffer->i16 += outFrames * channelCount;
+        mOutBuffer.frameCount -= outFrames;
+        mOutBuffer.i16 += outFrames * channelCount;
+
+        if (pInBuffer->frameCount == 0) {
+            if (mBufferQueue.size()) {
+                mBufferQueue.removeAt(0);
+                delete [] pInBuffer->mBuffer;
+                delete pInBuffer;
+                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
+                        mThread.unsafe_get(), mBufferQueue.size());
+            } else {
+                break;
+            }
+        }
+    }
+
+    // If we could not write all frames, allocate a buffer and queue it for next time.
+    if (inBuffer.frameCount) {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0 && !thread->standby()) {
+            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+                pInBuffer = new Buffer;
+                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+                pInBuffer->frameCount = inBuffer.frameCount;
+                pInBuffer->i16 = pInBuffer->mBuffer;
+                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
+                        sizeof(int16_t));
+                mBufferQueue.add(pInBuffer);
+                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
+                        mThread.unsafe_get(), mBufferQueue.size());
+            } else {
+                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
+                        mThread.unsafe_get(), this);
+            }
+        }
+    }
+
+    // Calling write() with a 0 length buffer, means that no more data will be written:
+    // If no more buffers are pending, fill output track buffer to make sure it is started
+    // by output mixer.
+    if (frames == 0 && mBufferQueue.size() == 0) {
+        if (mCblk->user < mFrameCount) {
+            frames = mFrameCount - mCblk->user;
+            pInBuffer = new Buffer;
+            pInBuffer->mBuffer = new int16_t[frames * channelCount];
+            pInBuffer->frameCount = frames;
+            pInBuffer->i16 = pInBuffer->mBuffer;
+            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
+            mBufferQueue.add(pInBuffer);
+        } else if (mActive) {
+            stop();
+        }
+    }
+
+    return outputBufferFull;
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
+        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
+{
+    int active;
+    status_t result;
+    audio_track_cblk_t* cblk = mCblk;
+    uint32_t framesReq = buffer->frameCount;
+
+    ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+    buffer->frameCount  = 0;
+
+    uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
+
+
+    if (framesAvail == 0) {
+        Mutex::Autolock _l(cblk->lock);
+        goto start_loop_here;
+        while (framesAvail == 0) {
+            active = mActive;
+            if (CC_UNLIKELY(!active)) {
+                ALOGV("Not active and NO_MORE_BUFFERS");
+                return NO_MORE_BUFFERS;
+            }
+            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+            if (result != NO_ERROR) {
+                return NO_MORE_BUFFERS;
+            }
+            // read the server count again
+        start_loop_here:
+            framesAvail = cblk->framesAvailableOut_l(mFrameCount);
+        }
+    }
+
+//    if (framesAvail < framesReq) {
+//        return NO_MORE_BUFFERS;
+//    }
+
+    if (framesReq > framesAvail) {
+        framesReq = framesAvail;
+    }
+
+    uint32_t u = cblk->user;
+    uint32_t bufferEnd = cblk->userBase + mFrameCount;
+
+    if (framesReq > bufferEnd - u) {
+        framesReq = bufferEnd - u;
+    }
+
+    buffer->frameCount  = framesReq;
+    buffer->raw         = cblk->buffer(mBuffers, mFrameSize, u);
+    return NO_ERROR;
+}
+
+
+void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
+{
+    size_t size = mBufferQueue.size();
+
+    for (size_t i = 0; i < size; i++) {
+        Buffer *pBuffer = mBufferQueue.itemAt(i);
+        delete [] pBuffer->mBuffer;
+        delete pBuffer;
+    }
+    mBufferQueue.clear();
+}
+
+
+// ----------------------------------------------------------------------------
+//      Record
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(
+        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+    : BnAudioRecord(),
+    mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {
+    stop_nonvirtual();
+    mRecordTrack->destroy();
+}
+
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+    return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
+        int triggerSession) {
+    ALOGV("RecordHandle::start()");
+    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
+}
+
+void AudioFlinger::RecordHandle::stop() {
+    stop_nonvirtual();
+}
+
+void AudioFlinger::RecordHandle::stop_nonvirtual() {
+    ALOGV("RecordHandle::stop()");
+    mRecordTrack->stop();
+}
+
+status_t AudioFlinger::RecordHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+// RecordTrack constructor must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+            RecordThread *thread,
+            const sp<Client>& client,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            int sessionId)
+    :   TrackBase(thread, client, sampleRate, format,
+                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
+        mOverflow(false)
+{
+    ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+}
+
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
+{
+    ALOGV("%s", __func__);
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
+        int64_t pts)
+{
+    audio_track_cblk_t* cblk = this->cblk();
+    uint32_t framesAvail;
+    uint32_t framesReq = buffer->frameCount;
+
+    // Check if last stepServer failed, try to step now
+    if (mStepServerFailed) {
+        if (!step()) {
+            goto getNextBuffer_exit;
+        }
+        ALOGV("stepServer recovered");
+        mStepServerFailed = false;
+    }
+
+    // FIXME lock is not actually held, so overrun is possible
+    framesAvail = cblk->framesAvailableIn_l(mFrameCount);
+
+    if (CC_LIKELY(framesAvail)) {
+        uint32_t s = cblk->server;
+        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
+
+        if (framesReq > framesAvail) {
+            framesReq = framesAvail;
+        }
+        if (framesReq > bufferEnd - s) {
+            framesReq = bufferEnd - s;
+        }
+
+        buffer->raw = getBuffer(s, framesReq);
+        buffer->frameCount = framesReq;
+        return NO_ERROR;
+    }
+
+getNextBuffer_exit:
+    buffer->raw = NULL;
+    buffer->frameCount = 0;
+    return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
+                                                        int triggerSession)
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        return recordThread->start(this, event, triggerSession);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::stop()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        recordThread->mLock.lock();
+        bool doStop = recordThread->stop_l(this);
+        if (doStop) {
+            TrackBase::reset();
+            // Force overrun condition to avoid false overrun callback until first data is
+            // read from buffer
+            android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
+        }
+        recordThread->mLock.unlock();
+        if (doStop) {
+            AudioSystem::stopInput(recordThread->id());
+        }
+    }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::destroy()
+{
+    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
+    sp<RecordTrack> keep(this);
+    {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            if (mState == ACTIVE || mState == RESUMING) {
+                AudioSystem::stopInput(thread->id());
+            }
+            AudioSystem::releaseInput(thread->id());
+            Mutex::Autolock _l(thread->mLock);
+            RecordThread *recordThread = (RecordThread *) thread.get();
+            recordThread->destroyTrack_l(this);
+        }
+    }
+}
+
+
+/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
+{
+    result.append("   Clien Fmt Chn mask   Session Step S SRate  Serv     User   FrameCount\n");
+}
+
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x %05d\n",
+            (mClient == 0) ? getpid_cached : mClient->pid(),
+            mFormat,
+            mChannelMask,
+            mSessionId,
+            mStepCount,
+            mState,
+            mCblk->sampleRate,
+            mCblk->server,
+            mCblk->user,
+            mFrameCount);
+}
+
+bool AudioFlinger::RecordThread::RecordTrack::isOut() const
+{
+    return false;
+}
+
+}; // namespace android
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index eff47c8..5245983 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -28,7 +28,6 @@
     libbinder \
     libcutils \
     libmedia \
-    libmedia_native \
     libcamera_client \
     libgui \
     libhardware \
@@ -40,6 +39,9 @@
     system/media/camera/include \
     external/jpeg
 
+
+LOCAL_CFLAGS += -Wall -Wextra
+
 LOCAL_MODULE:= libcameraservice
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/camera/libcameraservice/Camera2Client.cpp b/services/camera/libcameraservice/Camera2Client.cpp
index 0f1e650..5a7bb48 100644
--- a/services/camera/libcameraservice/Camera2Client.cpp
+++ b/services/camera/libcameraservice/Camera2Client.cpp
@@ -37,10 +37,6 @@
     return IPCThreadState::self()->getCallingPid();
 }
 
-static int getCallingUid() {
-    return IPCThreadState::self()->getCallingUid();
-}
-
 // Interface used by CameraService
 
 Camera2Client::Camera2Client(const sp<CameraService>& cameraService,
@@ -370,7 +366,6 @@
 void Camera2Client::disconnect() {
     ATRACE_CALL();
     Mutex::Autolock icl(mICameraLock);
-    status_t res;
 
     // Allow both client and the media server to disconnect at all times
     int callingPid = getCallingPid();
@@ -575,7 +570,7 @@
     ATRACE_CALL();
     ALOGV("%s: Camera %d: Flag 0x%x", __FUNCTION__, mCameraId, flag);
     Mutex::Autolock icl(mICameraLock);
-    status_t res;
+
     if ( checkPid(__FUNCTION__) != OK) return;
 
     SharedParameters::Lock l(mParameters);
@@ -1244,7 +1239,7 @@
     return OK;
 }
 
-status_t Camera2Client::commandStartFaceDetectionL(int type) {
+status_t Camera2Client::commandStartFaceDetectionL(int /*type*/) {
     ALOGV("%s: Camera %d: Starting face detection",
           __FUNCTION__, mCameraId);
     status_t res;
@@ -1331,6 +1326,8 @@
 }
 
 void Camera2Client::notifyShutter(int frameNumber, nsecs_t timestamp) {
+    (void)frameNumber;
+    (void)timestamp;
     ALOGV("%s: Shutter notification for frame %d at time %lld", __FUNCTION__,
             frameNumber, timestamp);
 }
@@ -1452,6 +1449,8 @@
 }
 
 void Camera2Client::notifyAutoWhitebalance(uint8_t newState, int triggerId) {
+    (void)newState;
+    (void)triggerId;
     ALOGV("%s: Auto-whitebalance state now %d, last trigger %d",
             __FUNCTION__, newState, triggerId);
 }
diff --git a/services/camera/libcameraservice/Camera2Device.cpp b/services/camera/libcameraservice/Camera2Device.cpp
index d6445c1..5bfa085 100644
--- a/services/camera/libcameraservice/Camera2Device.cpp
+++ b/services/camera/libcameraservice/Camera2Device.cpp
@@ -765,7 +765,6 @@
     ATRACE_CALL();
     ALOGV("%s: E", __FUNCTION__);
     Mutex::Autolock l(mMutex);
-    status_t res;
 
     if (mStreamSlotCount > 0) {
         freeBuffers(mStreamSlot.begin(), mStreamSlot.end());
@@ -785,7 +784,7 @@
 }
 
 status_t Camera2Device::MetadataQueue::dump(int fd,
-        const Vector<String16>& args) {
+        const Vector<String16>& /*args*/) {
     ATRACE_CALL();
     String8 result;
     status_t notLocked;
@@ -894,12 +893,13 @@
 {
     ATRACE_CALL();
     MetadataQueue *queue = getInstance(q);
+    (void)queue;
     free_camera_metadata(old_buffer);
     return OK;
 }
 
 int Camera2Device::MetadataQueue::producer_dequeue(
-        const camera2_frame_queue_dst_ops_t *q,
+        const camera2_frame_queue_dst_ops_t * /*q*/,
         size_t entries, size_t bytes,
         camera_metadata_t **buffer)
 {
@@ -912,7 +912,7 @@
 }
 
 int Camera2Device::MetadataQueue::producer_cancel(
-        const camera2_frame_queue_dst_ops_t *q,
+        const camera2_frame_queue_dst_ops_t * /*q*/,
         camera_metadata_t *old_buffer)
 {
     ATRACE_CALL();
@@ -1184,7 +1184,7 @@
 }
 
 status_t Camera2Device::StreamAdapter::dump(int fd,
-        const Vector<String16>& args) {
+        const Vector<String16>& /*args*/) {
     ATRACE_CALL();
     String8 result = String8::format("      Stream %d: %d x %d, format 0x%x\n",
             mId, mWidth, mHeight, mFormat);
@@ -1423,7 +1423,7 @@
 }
 
 status_t Camera2Device::ReprocessStreamAdapter::dump(int fd,
-        const Vector<String16>& args) {
+        const Vector<String16>& /*args*/) {
     ATRACE_CALL();
     String8 result =
             String8::format("      Reprocess stream %d: %d x %d, fmt 0x%x\n",
@@ -1444,7 +1444,7 @@
     const camera2_stream_in_ops_t *w,
         buffer_handle_t** buffer) {
     ATRACE_CALL();
-    int res;
+
     ReprocessStreamAdapter* stream =
             const_cast<ReprocessStreamAdapter*>(
                 static_cast<const ReprocessStreamAdapter*>(w));
diff --git a/services/camera/libcameraservice/CameraClient.cpp b/services/camera/libcameraservice/CameraClient.cpp
index b930c02..006a9c9 100644
--- a/services/camera/libcameraservice/CameraClient.cpp
+++ b/services/camera/libcameraservice/CameraClient.cpp
@@ -34,10 +34,6 @@
     return IPCThreadState::self()->getCallingPid();
 }
 
-static int getCallingUid() {
-    return IPCThreadState::self()->getCallingUid();
-}
-
 CameraClient::CameraClient(const sp<CameraService>& cameraService,
         const sp<ICameraClient>& cameraClient,
         int cameraId, int cameraFacing, int clientPid, int servicePid):
diff --git a/services/camera/libcameraservice/CameraHardwareInterface.h b/services/camera/libcameraservice/CameraHardwareInterface.h
index 05ac9fa..167b37c 100644
--- a/services/camera/libcameraservice/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/CameraHardwareInterface.h
@@ -427,7 +427,7 @@
     /**
      * Dump state of the camera hardware
      */
-    status_t dump(int fd, const Vector<String16>& args) const
+    status_t dump(int fd, const Vector<String16>& /*args*/) const
     {
         ALOGV("%s(%s)", __FUNCTION__, mName.string());
         if (mDevice->ops->dump)
@@ -584,9 +584,10 @@
 #endif
 
     static int __lock_buffer(struct preview_stream_ops* w,
-                      buffer_handle_t* buffer)
+                      buffer_handle_t* /*buffer*/)
     {
         ANativeWindow *a = anw(w);
+        (void)a;
         return 0;
     }
 
diff --git a/services/camera/libcameraservice/camera2/BurstCapture.cpp b/services/camera/libcameraservice/camera2/BurstCapture.cpp
index f56c50c..192d419 100644
--- a/services/camera/libcameraservice/camera2/BurstCapture.cpp
+++ b/services/camera/libcameraservice/camera2/BurstCapture.cpp
@@ -38,7 +38,8 @@
 BurstCapture::~BurstCapture() {
 }
 
-status_t BurstCapture::start(Vector<CameraMetadata> &metadatas, int32_t firstCaptureId) {
+status_t BurstCapture::start(Vector<CameraMetadata> &/*metadatas*/,
+                             int32_t /*firstCaptureId*/) {
     ALOGE("Not completely implemented");
     return INVALID_OPERATION;
 }
@@ -75,7 +76,7 @@
 
 CpuConsumer::LockedBuffer* BurstCapture::jpegEncode(
     CpuConsumer::LockedBuffer *imgBuffer,
-    int quality)
+    int /*quality*/)
 {
     ALOGV("%s", __FUNCTION__);
 
@@ -91,7 +92,7 @@
     buffers.push_back(imgEncoded);
 
     sp<JpegCompressor> jpeg = new JpegCompressor();
-    status_t res = jpeg->start(buffers, 1);
+    jpeg->start(buffers, 1);
 
     bool success = jpeg->waitForDone(10 * 1e9);
     if(success) {
@@ -103,7 +104,7 @@
     }
 }
 
-status_t BurstCapture::processFrameAvailable(sp<Camera2Client> &client) {
+status_t BurstCapture::processFrameAvailable(sp<Camera2Client> &/*client*/) {
     ALOGE("Not implemented");
     return INVALID_OPERATION;
 }
diff --git a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
index 3e9c255..307cfab 100644
--- a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
@@ -119,7 +119,6 @@
 
 status_t CallbackProcessor::deleteStream() {
     ATRACE_CALL();
-    status_t res;
 
     Mutex::Autolock l(mInputMutex);
 
@@ -144,7 +143,7 @@
     return mCallbackStreamId;
 }
 
-void CallbackProcessor::dump(int fd, const Vector<String16>& args) const {
+void CallbackProcessor::dump(int /*fd*/, const Vector<String16>& /*args*/) const {
 }
 
 bool CallbackProcessor::threadLoop() {
@@ -173,7 +172,6 @@
     ATRACE_CALL();
     status_t res;
 
-    int callbackHeapId;
     sp<Camera2Heap> callbackHeap;
     size_t heapIdx;
 
diff --git a/services/camera/libcameraservice/camera2/CaptureSequencer.cpp b/services/camera/libcameraservice/camera2/CaptureSequencer.cpp
index 072453b..513a47e 100644
--- a/services/camera/libcameraservice/camera2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/camera2/CaptureSequencer.cpp
@@ -130,7 +130,7 @@
 }
 
 
-void CaptureSequencer::dump(int fd, const Vector<String16>& args) {
+void CaptureSequencer::dump(int fd, const Vector<String16>& /*args*/) {
     String8 result;
     if (mCaptureRequest.entryCount() != 0) {
         result = "    Capture request:\n";
@@ -184,7 +184,6 @@
 };
 
 bool CaptureSequencer::threadLoop() {
-    status_t res;
 
     sp<Camera2Client> client = mClient.promote();
     if (client == 0) return false;
@@ -215,7 +214,8 @@
     return true;
 }
 
-CaptureSequencer::CaptureState CaptureSequencer::manageIdle(sp<Camera2Client> &client) {
+CaptureSequencer::CaptureState CaptureSequencer::manageIdle(
+        sp<Camera2Client> &/*client*/) {
     status_t res;
     Mutex::Autolock l(mInputMutex);
     while (!mStartCapture) {
@@ -352,13 +352,13 @@
 }
 
 CaptureSequencer::CaptureState CaptureSequencer::manageZslWaiting(
-        sp<Camera2Client> &client) {
+        sp<Camera2Client> &/*client*/) {
     ALOGV("%s", __FUNCTION__);
     return DONE;
 }
 
 CaptureSequencer::CaptureState CaptureSequencer::manageZslReprocessing(
-        sp<Camera2Client> &client) {
+        sp<Camera2Client> &/*client*/) {
     ALOGV("%s", __FUNCTION__);
     return START;
 }
@@ -380,7 +380,7 @@
 }
 
 CaptureSequencer::CaptureState CaptureSequencer::manageStandardPrecaptureWait(
-        sp<Camera2Client> &client) {
+        sp<Camera2Client> &/*client*/) {
     status_t res;
     ATRACE_CALL();
     Mutex::Autolock l(mInputMutex);
@@ -580,7 +580,7 @@
 }
 
 CaptureSequencer::CaptureState CaptureSequencer::manageBurstCaptureWait(
-        sp<Camera2Client> &client) {
+        sp<Camera2Client> &/*client*/) {
     status_t res;
     ATRACE_CALL();
 
diff --git a/services/camera/libcameraservice/camera2/FrameProcessor.cpp b/services/camera/libcameraservice/camera2/FrameProcessor.cpp
index 064607c..e032522 100644
--- a/services/camera/libcameraservice/camera2/FrameProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/FrameProcessor.cpp
@@ -62,7 +62,7 @@
     return OK;
 }
 
-void FrameProcessor::dump(int fd, const Vector<String16>& args) {
+void FrameProcessor::dump(int fd, const Vector<String16>& /*args*/) {
     String8 result("    Latest received frame:\n");
     write(fd, result.string(), result.size());
     mLastFrame.dump(fd, 2, 6);
@@ -128,7 +128,6 @@
 
 status_t FrameProcessor::processListeners(const CameraMetadata &frame,
         sp<Camera2Client> &client) {
-    status_t res;
     ATRACE_CALL();
     camera_metadata_ro_entry_t entry;
 
@@ -173,7 +172,7 @@
     ATRACE_CALL();
     camera_metadata_ro_entry_t entry;
     bool enableFaceDetect;
-    int maxFaces;
+
     {
         SharedParameters::Lock l(client->getParameters());
         enableFaceDetect = l.mParameters.enableFaceDetect;
diff --git a/services/camera/libcameraservice/camera2/JpegCompressor.cpp b/services/camera/libcameraservice/camera2/JpegCompressor.cpp
index 702ef58..c9af71e 100644
--- a/services/camera/libcameraservice/camera2/JpegCompressor.cpp
+++ b/services/camera/libcameraservice/camera2/JpegCompressor.cpp
@@ -144,7 +144,7 @@
 }
 
 // old function -- TODO: update for new buffer type
-bool JpegCompressor::isStreamInUse(uint32_t id) {
+bool JpegCompressor::isStreamInUse(uint32_t /*id*/) {
     ALOGV("%s", __FUNCTION__);
     Mutex::Autolock lock(mBusyMutex);
 
@@ -203,14 +203,14 @@
     dest->free_in_buffer = kMaxJpegSize;
 }
 
-boolean JpegCompressor::jpegEmptyOutputBuffer(j_compress_ptr cinfo) {
+boolean JpegCompressor::jpegEmptyOutputBuffer(j_compress_ptr /*cinfo*/) {
     ALOGV("%s", __FUNCTION__);
     ALOGE("%s: JPEG destination buffer overflow!",
             __FUNCTION__);
     return true;
 }
 
-void JpegCompressor::jpegTermDestination(j_compress_ptr cinfo) {
+void JpegCompressor::jpegTermDestination(j_compress_ptr /*cinfo*/) {
     ALOGV("%s", __FUNCTION__);
     ALOGV("%s: Done writing JPEG data. %d bytes left in buffer",
             __FUNCTION__, cinfo->dest->free_in_buffer);
diff --git a/services/camera/libcameraservice/camera2/JpegProcessor.cpp b/services/camera/libcameraservice/camera2/JpegProcessor.cpp
index ffc072b..6280f83 100644
--- a/services/camera/libcameraservice/camera2/JpegProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/JpegProcessor.cpp
@@ -139,7 +139,6 @@
 
 status_t JpegProcessor::deleteStream() {
     ATRACE_CALL();
-    status_t res;
 
     Mutex::Autolock l(mInputMutex);
 
@@ -164,7 +163,7 @@
     return mCaptureStreamId;
 }
 
-void JpegProcessor::dump(int fd, const Vector<String16>& args) const {
+void JpegProcessor::dump(int /*fd*/, const Vector<String16>& /*args*/) const {
 }
 
 bool JpegProcessor::threadLoop() {
@@ -356,7 +355,7 @@
     // Find End of Image
     // Scan JPEG buffer until End of Image (EOI)
     bool foundEnd = false;
-    for (size; size <= maxSize - MARKER_LENGTH; size++) {
+    for ( ; size <= maxSize - MARKER_LENGTH; size++) {
         if ( checkJpegEnd(jpegBuffer + size) ) {
             foundEnd = true;
             size += MARKER_LENGTH;
diff --git a/services/camera/libcameraservice/camera2/Parameters.cpp b/services/camera/libcameraservice/camera2/Parameters.cpp
index 9a0083a..93927e6 100644
--- a/services/camera/libcameraservice/camera2/Parameters.cpp
+++ b/services/camera/libcameraservice/camera2/Parameters.cpp
@@ -951,7 +951,6 @@
 
 camera_metadata_ro_entry_t Parameters::staticInfo(uint32_t tag,
         size_t minCount, size_t maxCount) const {
-    status_t res;
     camera_metadata_ro_entry_t entry = info->find(tag);
 
     if (CC_UNLIKELY( entry.count == 0 )) {
@@ -1567,6 +1566,10 @@
         ALOGE("%s: Video stabilization not supported", __FUNCTION__);
     }
 
+    // LIGHTFX
+    validatedParams.lightFx = lightFxStringToEnum(
+        newParams.get(CameraParameters::KEY_LIGHTFX));
+
     /** Update internal parameters */
 
     *this = validatedParams;
@@ -2094,6 +2097,18 @@
     }
 }
 
+Parameters::Parameters::lightFxMode_t Parameters::lightFxStringToEnum(
+        const char *lightFxMode) {
+    return
+        !lightFxMode ?
+            Parameters::LIGHTFX_NONE :
+        !strcmp(lightFxMode, CameraParameters::LIGHTFX_LOWLIGHT) ?
+            Parameters::LIGHTFX_LOWLIGHT :
+        !strcmp(lightFxMode, CameraParameters::LIGHTFX_HDR) ?
+            Parameters::LIGHTFX_HDR :
+        Parameters::LIGHTFX_NONE;
+}
+
 status_t Parameters::parseAreas(const char *areasCStr,
         Vector<Parameters::Area> *areas) {
     static const size_t NUM_FIELDS = 5;
@@ -2414,7 +2429,7 @@
     return crop;
 }
 
-int32_t Parameters::fpsFromRange(int32_t min, int32_t max) const {
+int32_t Parameters::fpsFromRange(int32_t /*min*/, int32_t max) const {
     return max;
 }
 
diff --git a/services/camera/libcameraservice/camera2/Parameters.h b/services/camera/libcameraservice/camera2/Parameters.h
index 54b1e8c..6d32bf6 100644
--- a/services/camera/libcameraservice/camera2/Parameters.h
+++ b/services/camera/libcameraservice/camera2/Parameters.h
@@ -261,6 +261,8 @@
     static const char* flashModeEnumToString(flashMode_t flashMode);
     static focusMode_t focusModeStringToEnum(const char *focusMode);
     static const char* focusModeEnumToString(focusMode_t focusMode);
+    static lightFxMode_t lightFxStringToEnum(const char *lightFxMode);
+
     static status_t parseAreas(const char *areasCStr,
             Vector<Area> *areas);
 
diff --git a/services/camera/libcameraservice/camera2/StreamingProcessor.cpp b/services/camera/libcameraservice/camera2/StreamingProcessor.cpp
index 207f780..6ea27b2 100644
--- a/services/camera/libcameraservice/camera2/StreamingProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/StreamingProcessor.cpp
@@ -447,7 +447,6 @@
     ATRACE_CALL();
     Mutex::Autolock m(mMutex);
 
-    status_t res;
     mPreviewRequestId++;
     if (mPreviewRequestId >= Camera2Client::kPreviewRequestIdEnd) {
         mPreviewRequestId = Camera2Client::kPreviewRequestIdStart;
@@ -628,7 +627,7 @@
 }
 
 
-status_t StreamingProcessor::dump(int fd, const Vector<String16>& args) {
+status_t StreamingProcessor::dump(int fd, const Vector<String16>& /*args*/) {
     String8 result;
 
     result.append("  Current requests:\n");
diff --git a/services/camera/libcameraservice/camera2/ZslProcessor.cpp b/services/camera/libcameraservice/camera2/ZslProcessor.cpp
index 1937955..9584028 100644
--- a/services/camera/libcameraservice/camera2/ZslProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/ZslProcessor.cpp
@@ -69,11 +69,12 @@
     }
 }
 
-void ZslProcessor::onFrameAvailable(int32_t frameId, const CameraMetadata &frame) {
+void ZslProcessor::onFrameAvailable(int32_t /*frameId*/, const CameraMetadata &frame) {
     Mutex::Autolock l(mInputMutex);
     camera_metadata_ro_entry_t entry;
     entry = frame.find(ANDROID_SENSOR_TIMESTAMP);
     nsecs_t timestamp = entry.data.i64[0];
+    (void)timestamp;
     ALOGVV("Got preview frame for timestamp %lld", timestamp);
 
     if (mState != RUNNING) return;
@@ -367,7 +368,7 @@
     return OK;
 }
 
-void ZslProcessor::dump(int fd, const Vector<String16>& args) const {
+void ZslProcessor::dump(int fd, const Vector<String16>& /*args*/) const {
     Mutex::Autolock l(mInputMutex);
     if (!mLatestCapturedRequest.isEmpty()) {
         String8 result("    Latest ZSL capture request:\n");